pax_global_header00006660000000000000000000000064140422531070014510gustar00rootroot0000000000000052 comment=801f67f671929311e0c9952c5f92d6e147c7b003 fdk-aac-2.0.2/000077500000000000000000000000001404225310700127775ustar00rootroot00000000000000fdk-aac-2.0.2/.clang-format000066400000000000000000000001761404225310700153560ustar00rootroot00000000000000BasedOnStyle: Google SortIncludes: false # Do not reformat the Doxygen-style comments in the code CommentPragmas : "^ * \\\\" fdk-aac-2.0.2/.gitignore000066400000000000000000000007371404225310700147760ustar00rootroot00000000000000*.o *.lo *.la .deps .libs .dirstamp Makefile Makefile.in aclocal.m4 autom4te.cache configure fdk-aac.pc config.guess config.log config.status config.sub depcomp install-sh libtool ltmain.sh m4/libtool.m4 m4/ltoptions.m4 m4/ltsugar.m4 m4/ltversion.m4 m4/lt~obsolete.m4 missing stamp-h1 aac-enc compile CMakeCache.txt CMakeFiles CMakeScripts Testing cmake_install.cmake install_manifest.txt compile_commands.json CTestTestfile.cmake CMakeSettings.json *[Bb]uild*/ .vs/ .vscode/ fdk-aac-2.0.2/Android.bp000066400000000000000000000055601404225310700147100ustar00rootroot00000000000000// *** THIS PACKAGE HAS SPECIAL LICENSING CONDITIONS. PLEASE // CONSULT THE OWNERS AND opensource-licensing@google.com BEFORE // DEPENDING ON IT IN YOUR PROJECT. *** package { default_applicable_licenses: ["external_aac_license"], } // Added automatically by a large-scale-change that took the approach of // 'apply every license found to every target'. While this makes sure we respect // every license restriction, it may not be entirely correct. // // e.g. GPL in an MIT project might only apply to the contrib/ directory. // // Please consider splitting the single license below into multiple licenses, // taking care not to lose any license_kind information, and overriding the // default license using the 'licenses: [...]' property on targets as needed. // // For unused files, consider creating a 'fileGroup' with "//visibility:private" // to attach the license to, and including a comment whether the files may be // used in the current project. // See: http://go/android-license-faq license { name: "external_aac_license", visibility: [":__subpackages__"], license_kinds: [ "SPDX-license-identifier-Apache-2.0", "legacy_by_exception_only", // by exception only ], license_text: [ "NOTICE", ], } cc_library_static { name: "libFraunhoferAAC", vendor_available: true, host_supported: true, srcs: [ "libAACdec/src/*.cpp", "libAACenc/src/*.cpp", "libPCMutils/src/*.cpp", "libFDK/src/*.cpp", "libSYS/src/*.cpp", "libMpegTPDec/src/*.cpp", "libMpegTPEnc/src/*.cpp", "libSBRdec/src/*.cpp", "libSBRenc/src/*.cpp", "libArithCoding/src/*.cpp", "libDRCdec/src/*.cpp", "libSACdec/src/*.cpp", "libSACenc/src/*.cpp", ], cflags: [ "-Werror", "-Wno-unused-parameter", "-Wno-#warnings", "-Wuninitialized", "-Wno-self-assign", "-Wno-implicit-fallthrough", "-DSUPPRESS_BUILD_DATE_INFO", ], sanitize: { misc_undefined: [ "unsigned-integer-overflow", "signed-integer-overflow", "bounds", ], cfi: true, }, shared_libs: [ "liblog", ], export_include_dirs: [ "libAACdec/include", "libAACenc/include", "libPCMutils/include", "libFDK/include", "libSYS/include", "libMpegTPDec/include", "libMpegTPEnc/include", "libSBRdec/include", "libSBRenc/include", "libArithCoding/include", "libDRCdec/include", "libSACdec/include", "libSACenc/include", ], target: { darwin: { enabled: false, }, }, apex_available: [ "//apex_available:platform", "com.android.bluetooth.updatable", "com.android.media.swcodec", ], min_sdk_version: "29", } fdk-aac-2.0.2/CMakeLists.txt000066400000000000000000000413551404225310700155470ustar00rootroot00000000000000# # CMake project for fdk-aac # # ## Basic usage # # 1. Download and install CMake >= 3.10 (CMake >= 3.15 is recommended) # 2. Clone fdk-aac repository # 3. In fdk-aac directory create and go to `build` subdirectory: # # mkdir build # cd build # # 4. Run cmake to configure project with desired build type (Release|Debug): # # cmake .. -DCMAKE_BUILD_TYPE=Release # # Note CMake configuration without CMAKE_BUILD_TYPE option will not set build # type to some default and it is most likely is not what you want. # # If you want to build static (default is shared library), add BUILD_SHARED_LIBS=OFF option: # # # cmake .. -DBUILD_SHARED_LIBS=OFF # # You can combine options as well: # # cmake .. -DCMAKE_BUILD_TYPE=Release -DBUILD_SHARED_LIBS=OFF # # 5. Run cmake to build project: # # cmake --build . -c Release cmake_minimum_required(VERSION 3.5.1) # Policies ## Enables CMAKE_MSVC_RUNTIME_LIBRARY option support for CMake >= 3.15 ## if you want to use a MSVC multi-threaded statically-linked runtime library ## If you enable it, CMake will build fdk-acc.dll without external dependencies. ## ## Example usage: ## ## cmake .. -DCMAKE_MSVC_RUNTIME_LIBRARY="MultiThreaded$<$:Debug> if(POLICY CMP0091) cmake_policy(SET CMP0091 NEW) endif() project(fdk-aac VERSION 2.0.2) # Includes include(CheckFunctionExists) include(CheckLibraryExists) include(GNUInstallDirs) include(CMakePackageConfigHelpers) if(CMAKE_VERSION VERSION_LESS 3.11) set(CPACK_PACKAGE_VERSION_MAJOR ${PROJECT_VERSION_MAJOR}) set(CPACK_PACKAGE_VERSION_MINOR ${PROJECT_VERSION_MINOR}) set(CPACK_PACKAGE_VERSION_PATCH ${PROJECT_VERSION_PATCH}) endif() include(CPack) # Options option(BUILD_SHARED_LIBS "Build shared library" ON) option(BUILD_PROGRAMS "Build aac-enc utility" OFF) option(FDK_AAC_INSTALL_CMAKE_CONFIG_MODULE "Install CMake package configuration file" ON) option(FDK_AAC_INSTALL_PKGCONFIG_MODULE "Install pkg-config .pc file" ON) # Checks ## Check if math functions are in separate library (Most of Linux distros, maybe some other OSes) check_function_exists(sin HAVE_DEFAULT_MATH) if(NOT HAVE_DEFAULT_MATH) check_library_exists(m sin "" HAVE_LIBM) endif() # Library ## Sources set(AACDEC_SRC libAACdec/src/FDK_delay.cpp libAACdec/src/aac_ram.cpp libAACdec/src/aac_rom.cpp libAACdec/src/aacdec_drc.cpp libAACdec/src/aacdec_hcr.cpp libAACdec/src/aacdec_hcr_bit.cpp libAACdec/src/aacdec_hcrs.cpp libAACdec/src/aacdec_pns.cpp libAACdec/src/aacdec_tns.cpp libAACdec/src/aacdecoder.cpp libAACdec/src/aacdecoder_lib.cpp libAACdec/src/block.cpp libAACdec/src/channel.cpp libAACdec/src/channelinfo.cpp libAACdec/src/conceal.cpp libAACdec/src/ldfiltbank.cpp libAACdec/src/pulsedata.cpp libAACdec/src/rvlc.cpp libAACdec/src/rvlcbit.cpp libAACdec/src/rvlcconceal.cpp libAACdec/src/stereo.cpp libAACdec/src/usacdec_ace_d4t64.cpp libAACdec/src/usacdec_ace_ltp.cpp libAACdec/src/usacdec_acelp.cpp libAACdec/src/usacdec_fac.cpp libAACdec/src/usacdec_lpc.cpp libAACdec/src/usacdec_lpd.cpp libAACdec/src/usacdec_rom.cpp libAACdec/src/aac_ram.h libAACdec/src/aac_rom.h libAACdec/src/aacdec_drc.h libAACdec/src/aacdec_drc_types.h libAACdec/src/aacdec_hcr.h libAACdec/src/aacdec_hcr_bit.h libAACdec/src/aacdec_hcr_types.h libAACdec/src/aacdec_hcrs.h libAACdec/src/aacdec_pns.h libAACdec/src/aacdec_tns.h libAACdec/src/aacdecoder.h libAACdec/src/block.h libAACdec/src/channel.h libAACdec/src/channelinfo.h libAACdec/src/conceal.h libAACdec/src/conceal_types.h libAACdec/src/FDK_delay.h libAACdec/src/ldfiltbank.h libAACdec/src/overlapadd.h libAACdec/src/pulsedata.h libAACdec/src/rvlc.h libAACdec/src/rvlc_info.h libAACdec/src/rvlcbit.h libAACdec/src/rvlcconceal.h libAACdec/src/stereo.h libAACdec/src/usacdec_ace_d4t64.h libAACdec/src/usacdec_ace_ltp.h libAACdec/src/usacdec_acelp.h libAACdec/src/usacdec_const.h libAACdec/src/usacdec_fac.h libAACdec/src/usacdec_lpc.h libAACdec/src/usacdec_lpd.h libAACdec/src/usacdec_rom.h) set(AACENC_SRC libAACenc/src/aacEnc_ram.cpp libAACenc/src/aacEnc_rom.cpp libAACenc/src/aacenc.cpp libAACenc/src/aacenc_lib.cpp libAACenc/src/aacenc_pns.cpp libAACenc/src/aacenc_tns.cpp libAACenc/src/adj_thr.cpp libAACenc/src/band_nrg.cpp libAACenc/src/bandwidth.cpp libAACenc/src/bit_cnt.cpp libAACenc/src/bitenc.cpp libAACenc/src/block_switch.cpp libAACenc/src/channel_map.cpp libAACenc/src/chaosmeasure.cpp libAACenc/src/dyn_bits.cpp libAACenc/src/grp_data.cpp libAACenc/src/intensity.cpp libAACenc/src/line_pe.cpp libAACenc/src/metadata_compressor.cpp libAACenc/src/metadata_main.cpp libAACenc/src/mps_main.cpp libAACenc/src/ms_stereo.cpp libAACenc/src/noisedet.cpp libAACenc/src/pnsparam.cpp libAACenc/src/pre_echo_control.cpp libAACenc/src/psy_configuration.cpp libAACenc/src/psy_main.cpp libAACenc/src/qc_main.cpp libAACenc/src/quantize.cpp libAACenc/src/sf_estim.cpp libAACenc/src/spreading.cpp libAACenc/src/tonality.cpp libAACenc/src/transform.cpp libAACenc/src/aacenc.h libAACenc/src/aacenc_pns.h libAACenc/src/aacEnc_ram.h libAACenc/src/aacEnc_rom.h libAACenc/src/aacenc_tns.h libAACenc/src/adj_thr.h libAACenc/src/adj_thr_data.h libAACenc/src/band_nrg.h libAACenc/src/bandwidth.h libAACenc/src/bit_cnt.h libAACenc/src/bitenc.h libAACenc/src/block_switch.h libAACenc/src/channel_map.h libAACenc/src/chaosmeasure.h libAACenc/src/dyn_bits.h libAACenc/src/grp_data.h libAACenc/src/intensity.h libAACenc/src/interface.h libAACenc/src/line_pe.h libAACenc/src/metadata_compressor.h libAACenc/src/metadata_main.h libAACenc/src/mps_main.h libAACenc/src/ms_stereo.h libAACenc/src/noisedet.h libAACenc/src/pns_func.h libAACenc/src/pnsparam.h libAACenc/src/pre_echo_control.h libAACenc/src/psy_configuration.h libAACenc/src/psy_const.h libAACenc/src/psy_data.h libAACenc/src/psy_main.h libAACenc/src/qc_data.h libAACenc/src/qc_main.h libAACenc/src/quantize.h libAACenc/src/sf_estim.h libAACenc/src/spreading.h libAACenc/src/tns_func.h libAACenc/src/tonality.h libAACenc/src/transform.h) set(ARITHCODING_SRC libArithCoding/src/ac_arith_coder.cpp) set(DRCDEC_SRC libDRCdec/src/FDK_drcDecLib.cpp libDRCdec/src/drcDec_gainDecoder.cpp libDRCdec/src/drcDec_reader.cpp libDRCdec/src/drcDec_rom.cpp libDRCdec/src/drcDec_selectionProcess.cpp libDRCdec/src/drcDec_tools.cpp libDRCdec/src/drcGainDec_init.cpp libDRCdec/src/drcGainDec_preprocess.cpp libDRCdec/src/drcGainDec_process.cpp libDRCdec/src/drcDec_gainDecoder.h libDRCdec/src/drcDec_reader.h libDRCdec/src/drcDec_rom.h libDRCdec/src/drcDec_selectionProcess.h libDRCdec/src/drcDec_tools.h libDRCdec/src/drcDec_types.h libDRCdec/src/drcDecoder.h libDRCdec/src/drcGainDec_init.h libDRCdec/src/drcGainDec_preprocess.h libDRCdec/src/drcGainDec_process.h) set(FDK_SRC libFDK/src/FDK_bitbuffer.cpp libFDK/src/FDK_core.cpp libFDK/src/FDK_crc.cpp libFDK/src/FDK_decorrelate.cpp libFDK/src/FDK_hybrid.cpp libFDK/src/FDK_lpc.cpp libFDK/src/FDK_matrixCalloc.cpp libFDK/src/FDK_qmf_domain.cpp libFDK/src/FDK_tools_rom.cpp libFDK/src/FDK_trigFcts.cpp libFDK/src/autocorr2nd.cpp libFDK/src/dct.cpp libFDK/src/fft.cpp libFDK/src/fft_rad2.cpp libFDK/src/fixpoint_math.cpp libFDK/src/huff_nodes.cpp libFDK/src/mdct.cpp libFDK/src/nlc_dec.cpp libFDK/src/qmf.cpp libFDK/src/scale.cpp) set(MPEGTPDEC_SRC libMpegTPDec/src/tpdec_adif.cpp libMpegTPDec/src/tpdec_adts.cpp libMpegTPDec/src/tpdec_asc.cpp libMpegTPDec/src/tpdec_drm.cpp libMpegTPDec/src/tpdec_latm.cpp libMpegTPDec/src/tpdec_lib.cpp libMpegTPDec/src/tp_version.h libMpegTPDec/src/tpdec_adif.h libMpegTPDec/src/tpdec_adts.h libMpegTPDec/src/tpdec_drm.h libMpegTPDec/src/tpdec_latm.h) set(MPEGTPENC_SRC libMpegTPEnc/src/tpenc_adif.cpp libMpegTPEnc/src/tpenc_adts.cpp libMpegTPEnc/src/tpenc_asc.cpp libMpegTPEnc/src/tpenc_latm.cpp libMpegTPEnc/src/tpenc_lib.cpp libMpegTPEnc/src/tp_version.h libMpegTPEnc/src/tpenc_adif.h libMpegTPEnc/src/tpenc_adts.h libMpegTPEnc/src/tpenc_asc.h libMpegTPEnc/src/tpenc_latm.h) set(PCMUTILS_SRC libPCMutils/src/limiter.cpp libPCMutils/src/pcm_utils.cpp libPCMutils/src/pcmdmx_lib.cpp libPCMutils/src/version.h) set(SACDEC_SRC libSACdec/src/sac_bitdec.cpp libSACdec/src/sac_calcM1andM2.cpp libSACdec/src/sac_dec.cpp libSACdec/src/sac_dec_conceal.cpp libSACdec/src/sac_dec_lib.cpp libSACdec/src/sac_process.cpp libSACdec/src/sac_qmf.cpp libSACdec/src/sac_reshapeBBEnv.cpp libSACdec/src/sac_rom.cpp libSACdec/src/sac_smoothing.cpp libSACdec/src/sac_stp.cpp libSACdec/src/sac_tsd.cpp libSACdec/src/sac_bitdec.h libSACdec/src/sac_calcM1andM2.h libSACdec/src/sac_dec.h libSACdec/src/sac_dec_conceal.h libSACdec/src/sac_dec_interface.h libSACdec/src/sac_dec_ssc_struct.h libSACdec/src/sac_process.h libSACdec/src/sac_qmf.h libSACdec/src/sac_reshapeBBEnv.h libSACdec/src/sac_rom.h libSACdec/src/sac_smoothing.h libSACdec/src/sac_stp.h libSACdec/src/sac_tsd.h) set(SACENC_SRC libSACenc/src/sacenc_bitstream.cpp libSACenc/src/sacenc_delay.cpp libSACenc/src/sacenc_dmx_tdom_enh.cpp libSACenc/src/sacenc_filter.cpp libSACenc/src/sacenc_framewindowing.cpp libSACenc/src/sacenc_huff_tab.cpp libSACenc/src/sacenc_lib.cpp libSACenc/src/sacenc_nlc_enc.cpp libSACenc/src/sacenc_onsetdetect.cpp libSACenc/src/sacenc_paramextract.cpp libSACenc/src/sacenc_staticgain.cpp libSACenc/src/sacenc_tree.cpp libSACenc/src/sacenc_vectorfunctions.cpp libSACenc/src/sacenc_bitstream.h libSACenc/src/sacenc_const.h libSACenc/src/sacenc_delay.h libSACenc/src/sacenc_dmx_tdom_enh.h libSACenc/src/sacenc_filter.h libSACenc/src/sacenc_framewindowing.h libSACenc/src/sacenc_huff_tab.h libSACenc/src/sacenc_nlc_enc.h libSACenc/src/sacenc_onsetdetect.h libSACenc/src/sacenc_paramextract.h libSACenc/src/sacenc_staticgain.h libSACenc/src/sacenc_tree.h libSACenc/src/sacenc_vectorfunctions.h) set(SBRDEC_SRC libSBRdec/src/HFgen_preFlat.cpp libSBRdec/src/env_calc.cpp libSBRdec/src/env_dec.cpp libSBRdec/src/env_extr.cpp libSBRdec/src/hbe.cpp libSBRdec/src/huff_dec.cpp libSBRdec/src/lpp_tran.cpp libSBRdec/src/psbitdec.cpp libSBRdec/src/psdec.cpp libSBRdec/src/psdec_drm.cpp libSBRdec/src/psdecrom_drm.cpp libSBRdec/src/pvc_dec.cpp libSBRdec/src/sbr_deb.cpp libSBRdec/src/sbr_dec.cpp libSBRdec/src/sbr_ram.cpp libSBRdec/src/sbr_rom.cpp libSBRdec/src/sbrdec_drc.cpp libSBRdec/src/sbrdec_freq_sca.cpp libSBRdec/src/sbrdecoder.cpp libSBRdec/src/env_calc.h libSBRdec/src/env_dec.h libSBRdec/src/env_extr.h libSBRdec/src/hbe.h libSBRdec/src/HFgen_preFlat.h libSBRdec/src/huff_dec.h libSBRdec/src/lpp_tran.h libSBRdec/src/psbitdec.h libSBRdec/src/psdec.h libSBRdec/src/psdec_drm.h libSBRdec/src/pvc_dec.h libSBRdec/src/sbr_deb.h libSBRdec/src/sbr_dec.h libSBRdec/src/sbr_ram.h libSBRdec/src/sbr_rom.h libSBRdec/src/sbrdec_drc.h libSBRdec/src/sbrdec_freq_sca.h libSBRdec/src/transcendent.h) set(SBRENC_SRC libSBRenc/src/bit_sbr.cpp libSBRenc/src/code_env.cpp libSBRenc/src/env_bit.cpp libSBRenc/src/env_est.cpp libSBRenc/src/fram_gen.cpp libSBRenc/src/invf_est.cpp libSBRenc/src/mh_det.cpp libSBRenc/src/nf_est.cpp libSBRenc/src/ps_bitenc.cpp libSBRenc/src/ps_encode.cpp libSBRenc/src/ps_main.cpp libSBRenc/src/resampler.cpp libSBRenc/src/sbr_encoder.cpp libSBRenc/src/sbr_misc.cpp libSBRenc/src/sbrenc_freq_sca.cpp libSBRenc/src/sbrenc_ram.cpp libSBRenc/src/sbrenc_rom.cpp libSBRenc/src/ton_corr.cpp libSBRenc/src/tran_det.cpp libSBRenc/src/bit_sbr.h libSBRenc/src/cmondata.h libSBRenc/src/code_env.h libSBRenc/src/env_bit.h libSBRenc/src/env_est.h libSBRenc/src/fram_gen.h libSBRenc/src/invf_est.h libSBRenc/src/mh_det.h libSBRenc/src/nf_est.h libSBRenc/src/ps_bitenc.h libSBRenc/src/ps_const.h libSBRenc/src/ps_encode.h libSBRenc/src/ps_main.h libSBRenc/src/resampler.h libSBRenc/src/sbr.h libSBRenc/src/sbr_def.h libSBRenc/src/sbr_misc.h libSBRenc/src/sbrenc_freq_sca.h libSBRenc/src/sbrenc_ram.h libSBRenc/src/sbrenc_rom.h libSBRenc/src/ton_corr.h libSBRenc/src/tran_det.h) set(SYS_SRC libSYS/src/genericStds.cpp libSYS/src/syslib_channelMapDescr.cpp) set(fdk_aacinclude_HEADERS libSYS/include/machine_type.h libSYS/include/genericStds.h libSYS/include/FDK_audio.h libSYS/include/syslib_channelMapDescr.h libAACenc/include/aacenc_lib.h libAACdec/include/aacdecoder_lib.h) set(libfdk_aac_SOURCES ${AACDEC_SRC} ${AACENC_SRC} ${ARITHCODING_SRC} ${DRCDEC_SRC} ${MPEGTPDEC_SRC} ${MPEGTPENC_SRC} ${SACDEC_SRC} ${SACENC_SRC} ${SBRDEC_SRC} ${SBRENC_SRC} ${PCMUTILS_SRC} ${FDK_SRC} ${SYS_SRC}) if(WIN32 AND BUILD_SHARED_LIBS) set(libfdk_aac_SOURCES ${libfdk_aac_SOURCES} fdk-aac.def) endif() ## Create Library target. Actually fdk-aac is enough, but we also create ## FDK-AAC::fdk-aac for consistence with config-file package. add_library(fdk-aac ${libfdk_aac_SOURCES}) add_library(FDK-AAC::fdk-aac ALIAS fdk-aac) ## Library target configuration ### Library target includes target_include_directories(fdk-aac PUBLIC $ $ $ $ PRIVATE libArithCoding/include libDRCdec/include libSACdec/include libSACenc/include libSBRdec/include libSBRenc/include libMpegTPDec/include libMpegTPEnc/include libFDK/include libPCMutils/include) ### Link math library if required target_link_libraries(fdk-aac PRIVATE $<$:m>) ### Set public headers and shared library version. Version info is critical for Unix-like OSes. set_target_properties(fdk-aac PROPERTIES PUBLIC_HEADER "${fdk_aacinclude_HEADERS}" VERSION 2.0.2 SOVERSION 2 MACHO_COMPATIBILITY_VERSION 3.0.0 MACHO_CURRENT_VERSION 3.2.0) ### Some compiler options from Makefile.am if(MSVC) target_compile_options(fdk-aac PRIVATE /EHsc) else() target_compile_options(fdk-aac PRIVATE -fno-exceptions -fno-rtti) endif() ### Set proper name for MinGW or Cygwin DLL if((MINGW OR CYGWIN) AND BUILD_SHARED_LIBS) set_property(TARGET fdk-aac PROPERTY RUNTIME_OUTPUT_NAME fdk-aac-2) endif() ## Library installation ### Note we export `fdk-aac-targets` to use with config-file package. install(TARGETS fdk-aac EXPORT fdk-aac-targets RUNTIME DESTINATION ${CMAKE_INSTALL_BINDIR} COMPONENT BIN ARCHIVE DESTINATION ${CMAKE_INSTALL_LIBDIR} COMPONENT DEV LIBRARY DESTINATION ${CMAKE_INSTALL_LIBDIR} COMPONENT DEV PUBLIC_HEADER DESTINATION ${CMAKE_INSTALL_INCLUDEDIR}/fdk-aac COMPONENT DEV) ### Configure and install pkg-config module. if(FDK_AAC_INSTALL_PKGCONFIG_MODULE) set(prefix ${CMAKE_INSTALL_PREFIX}) set(exec_prefix "\$\{prefix\}") set(libdir "\$\{exec_prefix\}/${CMAKE_INSTALL_LIBDIR}") set(includedir "\$\{prefix\}/${CMAKE_INSTALL_INCLUDEDIR}") set(PACKAGE_VERSION ${PROJECT_VERSION}) if(HAVE_LIBM) if(BUILD_SHARED_LIBS) set(LIBS_PRIVATE "-lm") else() set(LIBS_PUBLIC "-lm") endif() endif() configure_file(fdk-aac.pc.in fdk-aac.pc @ONLY) install( FILES ${CMAKE_CURRENT_BINARY_DIR}/fdk-aac.pc DESTINATION ${CMAKE_INSTALL_LIBDIR}/pkgconfig) endif() ### Configure and install config-file package. Something like pkg-config module, but for Windows. if(FDK_AAC_INSTALL_CMAKE_CONFIG_MODULE) set(CMAKE_INSTALL_PACKAGEDIR ${CMAKE_INSTALL_LIBDIR}/cmake/fdk-aac) configure_package_config_file(fdk-aac-config.cmake.in ${CMAKE_CURRENT_BINARY_DIR}/fdk-aac-config.cmake INSTALL_DESTINATION ${CMAKE_INSTALL_PACKAGEDIR}) if(CMAKE_VERSION VERSION_LESS 3.11) write_basic_package_version_file(fdk-aac-config-version.cmake COMPATIBILITY SameMajorVersion) else() write_basic_package_version_file(fdk-aac-config-version.cmake COMPATIBILITY SameMinorVersion) endif() install(EXPORT fdk-aac-targets NAMESPACE FDK-AAC:: DESTINATION ${CMAKE_INSTALL_PACKAGEDIR}) install( FILES ${CMAKE_CURRENT_BINARY_DIR}/fdk-aac-config.cmake ${CMAKE_CURRENT_BINARY_DIR}/fdk-aac-config-version.cmake DESTINATION ${CMAKE_INSTALL_PACKAGEDIR}) endif() # Programs if(BUILD_PROGRAMS) ## Program sources set(aac_enc_SOURCES aac-enc.c wavreader.c wavreader.h) ## Program target add_executable(aac-enc ${aac_enc_SOURCES}) ## Program target configuration target_link_libraries(aac-enc PRIVATE fdk-aac) target_compile_definitions(aac-enc PRIVATE $<$:_CRT_SECURE_NO_WARNINGS>) if(WIN32) target_sources(aac-enc PRIVATE win32/getopt.h) target_include_directories(aac-enc PRIVATE win32) endif() ## Program target installation install(TARGETS aac-enc RUNTIME DESTINATION ${CMAKE_INSTALL_BINDIR}) endif() fdk-aac-2.0.2/ChangeLog000066400000000000000000000035661404225310700145630ustar00rootroot000000000000002.0.2 - Minor upstream updates - Lots of upstream and local fuzzing fixes - Added CMake project files - Removed the MSVC specific makefile 2.0.1 - Minor release with a number of crash/fuzz fixes, primarily for the decoder 2.0.0 - Major update in the upstream source base, with support for new profiles and features, and numerous crash/fuzz fixes. The new upstream version is referred to as FDKv2, thus skipping the major version 1 and syncing the fdk-aac major version number to 2. 0.1.6 - Lots of minor assorted crash/fuzz fixes, mostly for the decoder but also some for the encoder 0.1.5 - Updated upstream sources - Fixed building with GCC 3.3 and 3.4 - Fixed building with GCC 6 - AArch64 optimizations - Makefiles for building with MSVC - Support building the code in C++11 mode 0.1.4 - Updated upstream sources, with minor changes to the decoder API breaking the ABI. (Calling code using AUDIO_CHANNEL_TYPE may need to be updated. A new option AAC_PCM_LIMITER_ENABLE has been added, enabled by default, which incurs extra decoding delay.) - PowerPC optimizations, fixes for building on AIX - Support for reading streamed wav files in the encoder example - Fix VBR encoding of sample rates over 64 kHz 0.1.3 - Updated upstream sources, with a number of crash fixes and new features (including support for encoding 7.1) 0.1.2 - Fix a few more crashes - Include dependency libs (such as -lm) in the pkg-config file 0.1.1 - Updated to a new upstream version from Android 4.2, fixing a lot of crashes - Cleanup of autotools usage - Make sure the shared library links to libm if necessary - Performance improvements on x86 - Added support for WG4/DVD audio channel mappings - Minimized the differences to upstream - Added an example encoder tool 0.1.0 - Initial release of fdk-aac - autotools based build system - Enable setting VBR bitrate modes fdk-aac-2.0.2/METADATA000066400000000000000000000000621404225310700141000ustar00rootroot00000000000000third_party { license_type: BY_EXCEPTION_ONLY } fdk-aac-2.0.2/MODULE_LICENSE_FRAUNHOFER000066400000000000000000000000001404225310700165560ustar00rootroot00000000000000fdk-aac-2.0.2/Makefile.am000066400000000000000000000227461404225310700150460ustar00rootroot00000000000000ACLOCAL_AMFLAGS = -I m4 AUTOMAKE_OPTIONS = subdir-objects AM_CPPFLAGS = \ -I$(top_srcdir)/libAACdec/include \ -I$(top_srcdir)/libAACenc/include \ -I$(top_srcdir)/libArithCoding/include \ -I$(top_srcdir)/libDRCdec/include \ -I$(top_srcdir)/libSACdec/include \ -I$(top_srcdir)/libSACenc/include \ -I$(top_srcdir)/libSBRdec/include \ -I$(top_srcdir)/libSBRenc/include \ -I$(top_srcdir)/libMpegTPDec/include \ -I$(top_srcdir)/libMpegTPEnc/include \ -I$(top_srcdir)/libSYS/include \ -I$(top_srcdir)/libFDK/include \ -I$(top_srcdir)/libPCMutils/include AM_CXXFLAGS = -fno-exceptions -fno-rtti libfdk_aac_la_LINK = $(LINK) $(libfdk_aac_la_LDFLAGS) # Mention a dummy pure C file to trigger generation of the $(LINK) variable nodist_EXTRA_libfdk_aac_la_SOURCES = dummy.c fdk_aacincludedir = $(includedir)/fdk-aac fdk_aacinclude_HEADERS = \ $(top_srcdir)/libSYS/include/machine_type.h \ $(top_srcdir)/libSYS/include/genericStds.h \ $(top_srcdir)/libSYS/include/FDK_audio.h \ $(top_srcdir)/libSYS/include/syslib_channelMapDescr.h \ $(top_srcdir)/libAACenc/include/aacenc_lib.h \ $(top_srcdir)/libAACdec/include/aacdecoder_lib.h pkgconfigdir = $(libdir)/pkgconfig pkgconfig_DATA = fdk-aac.pc lib_LTLIBRARIES = libfdk-aac.la libfdk_aac_la_LDFLAGS = -version-info @FDK_AAC_VERSION@ -no-undefined \ -export-symbols $(top_srcdir)/fdk-aac.sym if EXAMPLE bin_PROGRAMS = aac-enc$(EXEEXT) aac_enc_LDADD = libfdk-aac.la aac_enc_SOURCES = aac-enc.c wavreader.c noinst_HEADERS = wavreader.h endif AACDEC_SRC = \ libAACdec/src/FDK_delay.cpp \ libAACdec/src/aac_ram.cpp \ libAACdec/src/aac_rom.cpp \ libAACdec/src/aacdec_drc.cpp \ libAACdec/src/aacdec_hcr.cpp \ libAACdec/src/aacdec_hcr_bit.cpp \ libAACdec/src/aacdec_hcrs.cpp \ libAACdec/src/aacdec_pns.cpp \ libAACdec/src/aacdec_tns.cpp \ libAACdec/src/aacdecoder.cpp \ libAACdec/src/aacdecoder_lib.cpp \ libAACdec/src/block.cpp \ libAACdec/src/channel.cpp \ libAACdec/src/channelinfo.cpp \ libAACdec/src/conceal.cpp \ libAACdec/src/ldfiltbank.cpp \ libAACdec/src/pulsedata.cpp \ libAACdec/src/rvlc.cpp \ libAACdec/src/rvlcbit.cpp \ libAACdec/src/rvlcconceal.cpp \ libAACdec/src/stereo.cpp \ libAACdec/src/usacdec_ace_d4t64.cpp \ libAACdec/src/usacdec_ace_ltp.cpp \ libAACdec/src/usacdec_acelp.cpp \ libAACdec/src/usacdec_fac.cpp \ libAACdec/src/usacdec_lpc.cpp \ libAACdec/src/usacdec_lpd.cpp \ libAACdec/src/usacdec_rom.cpp AACENC_SRC = \ libAACenc/src/aacEnc_ram.cpp \ libAACenc/src/aacEnc_rom.cpp \ libAACenc/src/aacenc.cpp \ libAACenc/src/aacenc_lib.cpp \ libAACenc/src/aacenc_pns.cpp \ libAACenc/src/aacenc_tns.cpp \ libAACenc/src/adj_thr.cpp \ libAACenc/src/band_nrg.cpp \ libAACenc/src/bandwidth.cpp \ libAACenc/src/bit_cnt.cpp \ libAACenc/src/bitenc.cpp \ libAACenc/src/block_switch.cpp \ libAACenc/src/channel_map.cpp \ libAACenc/src/chaosmeasure.cpp \ libAACenc/src/dyn_bits.cpp \ libAACenc/src/grp_data.cpp \ libAACenc/src/intensity.cpp \ libAACenc/src/line_pe.cpp \ libAACenc/src/metadata_compressor.cpp \ libAACenc/src/metadata_main.cpp \ libAACenc/src/mps_main.cpp \ libAACenc/src/ms_stereo.cpp \ libAACenc/src/noisedet.cpp \ libAACenc/src/pnsparam.cpp \ libAACenc/src/pre_echo_control.cpp \ libAACenc/src/psy_configuration.cpp \ libAACenc/src/psy_main.cpp \ libAACenc/src/qc_main.cpp \ libAACenc/src/quantize.cpp \ libAACenc/src/sf_estim.cpp \ libAACenc/src/spreading.cpp \ libAACenc/src/tonality.cpp \ libAACenc/src/transform.cpp ARITHCODING_SRC = \ libArithCoding/src/ac_arith_coder.cpp DRCDEC_SRC = \ libDRCdec/src/FDK_drcDecLib.cpp \ libDRCdec/src/drcDec_gainDecoder.cpp \ libDRCdec/src/drcDec_reader.cpp \ libDRCdec/src/drcDec_rom.cpp \ libDRCdec/src/drcDec_selectionProcess.cpp \ libDRCdec/src/drcDec_tools.cpp \ libDRCdec/src/drcGainDec_init.cpp \ libDRCdec/src/drcGainDec_preprocess.cpp \ libDRCdec/src/drcGainDec_process.cpp FDK_SRC = \ libFDK/src/FDK_bitbuffer.cpp \ libFDK/src/FDK_core.cpp \ libFDK/src/FDK_crc.cpp \ libFDK/src/FDK_decorrelate.cpp \ libFDK/src/FDK_hybrid.cpp \ libFDK/src/FDK_lpc.cpp \ libFDK/src/FDK_matrixCalloc.cpp \ libFDK/src/FDK_qmf_domain.cpp \ libFDK/src/FDK_tools_rom.cpp \ libFDK/src/FDK_trigFcts.cpp \ libFDK/src/autocorr2nd.cpp \ libFDK/src/dct.cpp \ libFDK/src/fft.cpp \ libFDK/src/fft_rad2.cpp \ libFDK/src/fixpoint_math.cpp \ libFDK/src/huff_nodes.cpp \ libFDK/src/mdct.cpp \ libFDK/src/nlc_dec.cpp \ libFDK/src/qmf.cpp \ libFDK/src/scale.cpp MPEGTPDEC_SRC = \ libMpegTPDec/src/tpdec_adif.cpp \ libMpegTPDec/src/tpdec_adts.cpp \ libMpegTPDec/src/tpdec_asc.cpp \ libMpegTPDec/src/tpdec_drm.cpp \ libMpegTPDec/src/tpdec_latm.cpp \ libMpegTPDec/src/tpdec_lib.cpp MPEGTPENC_SRC = \ libMpegTPEnc/src/tpenc_adif.cpp \ libMpegTPEnc/src/tpenc_adts.cpp \ libMpegTPEnc/src/tpenc_asc.cpp \ libMpegTPEnc/src/tpenc_latm.cpp \ libMpegTPEnc/src/tpenc_lib.cpp PCMUTILS_SRC = \ libPCMutils/src/limiter.cpp \ libPCMutils/src/pcm_utils.cpp \ libPCMutils/src/pcmdmx_lib.cpp SACDEC_SRC = \ libSACdec/src/sac_bitdec.cpp \ libSACdec/src/sac_calcM1andM2.cpp \ libSACdec/src/sac_dec.cpp \ libSACdec/src/sac_dec_conceal.cpp \ libSACdec/src/sac_dec_lib.cpp \ libSACdec/src/sac_process.cpp \ libSACdec/src/sac_qmf.cpp \ libSACdec/src/sac_reshapeBBEnv.cpp \ libSACdec/src/sac_rom.cpp \ libSACdec/src/sac_smoothing.cpp \ libSACdec/src/sac_stp.cpp \ libSACdec/src/sac_tsd.cpp SACENC_SRC = \ libSACenc/src/sacenc_bitstream.cpp \ libSACenc/src/sacenc_delay.cpp \ libSACenc/src/sacenc_dmx_tdom_enh.cpp \ libSACenc/src/sacenc_filter.cpp \ libSACenc/src/sacenc_framewindowing.cpp \ libSACenc/src/sacenc_huff_tab.cpp \ libSACenc/src/sacenc_lib.cpp \ libSACenc/src/sacenc_nlc_enc.cpp \ libSACenc/src/sacenc_onsetdetect.cpp \ libSACenc/src/sacenc_paramextract.cpp \ libSACenc/src/sacenc_staticgain.cpp \ libSACenc/src/sacenc_tree.cpp \ libSACenc/src/sacenc_vectorfunctions.cpp SBRDEC_SRC = \ libSBRdec/src/HFgen_preFlat.cpp \ libSBRdec/src/env_calc.cpp \ libSBRdec/src/env_dec.cpp \ libSBRdec/src/env_extr.cpp \ libSBRdec/src/hbe.cpp \ libSBRdec/src/huff_dec.cpp \ libSBRdec/src/lpp_tran.cpp \ libSBRdec/src/psbitdec.cpp \ libSBRdec/src/psdec.cpp \ libSBRdec/src/psdec_drm.cpp \ libSBRdec/src/psdecrom_drm.cpp \ libSBRdec/src/pvc_dec.cpp \ libSBRdec/src/sbr_deb.cpp \ libSBRdec/src/sbr_dec.cpp \ libSBRdec/src/sbr_ram.cpp \ libSBRdec/src/sbr_rom.cpp \ libSBRdec/src/sbrdec_drc.cpp \ libSBRdec/src/sbrdec_freq_sca.cpp \ libSBRdec/src/sbrdecoder.cpp SBRENC_SRC = \ libSBRenc/src/bit_sbr.cpp \ libSBRenc/src/code_env.cpp \ libSBRenc/src/env_bit.cpp \ libSBRenc/src/env_est.cpp \ libSBRenc/src/fram_gen.cpp \ libSBRenc/src/invf_est.cpp \ libSBRenc/src/mh_det.cpp \ libSBRenc/src/nf_est.cpp \ libSBRenc/src/ps_bitenc.cpp \ libSBRenc/src/ps_encode.cpp \ libSBRenc/src/ps_main.cpp \ libSBRenc/src/resampler.cpp \ libSBRenc/src/sbr_encoder.cpp \ libSBRenc/src/sbr_misc.cpp \ libSBRenc/src/sbrenc_freq_sca.cpp \ libSBRenc/src/sbrenc_ram.cpp \ libSBRenc/src/sbrenc_rom.cpp \ libSBRenc/src/ton_corr.cpp \ libSBRenc/src/tran_det.cpp SYS_SRC = \ libSYS/src/genericStds.cpp \ libSYS/src/syslib_channelMapDescr.cpp libfdk_aac_la_SOURCES = \ $(AACDEC_SRC) $(AACENC_SRC) \ $(ARITHCODING_SRC) \ $(DRCDEC_SRC) \ $(MPEGTPDEC_SRC) $(MPEGTPENC_SRC) \ $(SACDEC_SRC) $(SACENC_SRC) \ $(SBRDEC_SRC) $(SBRENC_SRC) \ $(PCMUTILS_SRC) $(FDK_SRC) $(SYS_SRC) EXTRA_DIST = \ $(top_srcdir)/CMakeLists.txt \ $(top_srcdir)/.clang-format \ $(top_srcdir)/autogen.sh \ $(top_srcdir)/MODULE_LICENSE_FRAUNHOFER \ $(top_srcdir)/NOTICE \ $(top_srcdir)/OWNERS \ $(top_srcdir)/Android.bp \ $(top_srcdir)/fdk-aac.def \ $(top_srcdir)/fdk-aac.sym \ $(top_srcdir)/fdk-aac-config.cmake.in \ $(top_srcdir)/documentation/*.pdf \ $(top_srcdir)/fuzzer/* \ $(top_srcdir)/libAACdec/src/*.h \ $(top_srcdir)/libAACdec/src/arm/*.cpp \ $(top_srcdir)/libAACenc/src/*.h \ $(top_srcdir)/libArithCoding/include/*.h \ $(top_srcdir)/libDRCdec/include/*.h \ $(top_srcdir)/libDRCdec/src/*.h \ $(top_srcdir)/libSACdec/include/*.h \ $(top_srcdir)/libSACdec/src/*.h \ $(top_srcdir)/libSACenc/include/*.h \ $(top_srcdir)/libSACenc/src/*.h \ $(top_srcdir)/libSBRenc/src/*.h \ $(top_srcdir)/libSBRenc/include/*.h \ $(top_srcdir)/libSBRdec/src/*.h \ $(top_srcdir)/libSBRdec/src/arm/*.cpp \ $(top_srcdir)/libSBRdec/include/*.h \ $(top_srcdir)/libSYS/include/*.h \ $(top_srcdir)/libPCMutils/include/*.h \ $(top_srcdir)/libPCMutils/src/*.h \ $(top_srcdir)/libMpegTPEnc/include/*.h \ $(top_srcdir)/libMpegTPEnc/src/*.h \ $(top_srcdir)/libMpegTPDec/include/*.h \ $(top_srcdir)/libMpegTPDec/src/*.h \ $(top_srcdir)/libFDK/include/*.h \ $(top_srcdir)/libFDK/include/arm/*.h \ $(top_srcdir)/libFDK/include/mips/*.h \ $(top_srcdir)/libFDK/include/ppc/*.h \ $(top_srcdir)/libFDK/include/x86/*.h \ $(top_srcdir)/libFDK/src/arm/*.cpp \ $(top_srcdir)/libFDK/src/mips/*.cpp \ $(top_srcdir)/METADATA \ $(top_srcdir)/PREUPLOAD.cfg \ $(top_srcdir)/win32/*.h fdk-aac-2.0.2/NOTICE000066400000000000000000000105371404225310700137110ustar00rootroot00000000000000Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de fdk-aac-2.0.2/OWNERS000066400000000000000000000000471404225310700137400ustar00rootroot00000000000000jmtrivi@google.com gkasten@android.com fdk-aac-2.0.2/PREUPLOAD.cfg000066400000000000000000000001311404225310700150060ustar00rootroot00000000000000[Hook Scripts] mainline_hook = ${REPO_ROOT}/frameworks/av/tools/mainline_hook_project.sh fdk-aac-2.0.2/aac-enc.c000066400000000000000000000143441404225310700144400ustar00rootroot00000000000000/* ------------------------------------------------------------------ * Copyright (C) 2011 Martin Storsjo * * Licensed under the Apache License, Version 2.0 (the "License"); * you may not use this file except in compliance with the License. * You may obtain a copy of the License at * * http://www.apache.org/licenses/LICENSE-2.0 * * Unless required by applicable law or agreed to in writing, software * distributed under the License is distributed on an "AS IS" BASIS, * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either * express or implied. * See the License for the specific language governing permissions * and limitations under the License. * ------------------------------------------------------------------- */ #include #include #if defined(_MSC_VER) #include #else #include #endif #include #include "libAACenc/include/aacenc_lib.h" #include "wavreader.h" void usage(const char* name) { fprintf(stderr, "%s [-r bitrate] [-t aot] [-a afterburner] [-s sbr] [-v vbr] in.wav out.aac\n", name); fprintf(stderr, "Supported AOTs:\n"); fprintf(stderr, "\t2\tAAC-LC\n"); fprintf(stderr, "\t5\tHE-AAC\n"); fprintf(stderr, "\t29\tHE-AAC v2\n"); fprintf(stderr, "\t23\tAAC-LD\n"); fprintf(stderr, "\t39\tAAC-ELD\n"); } int main(int argc, char *argv[]) { int bitrate = 64000; int ch; const char *infile, *outfile; FILE *out; void *wav; int format, sample_rate, channels, bits_per_sample; int input_size; uint8_t* input_buf; int16_t* convert_buf; int aot = 2; int afterburner = 1; int eld_sbr = 0; int vbr = 0; HANDLE_AACENCODER handle; CHANNEL_MODE mode; AACENC_InfoStruct info = { 0 }; while ((ch = getopt(argc, argv, "r:t:a:s:v:")) != -1) { switch (ch) { case 'r': bitrate = atoi(optarg); break; case 't': aot = atoi(optarg); break; case 'a': afterburner = atoi(optarg); break; case 's': eld_sbr = atoi(optarg); break; case 'v': vbr = atoi(optarg); break; case '?': default: usage(argv[0]); return 1; } } if (argc - optind < 2) { usage(argv[0]); return 1; } infile = argv[optind]; outfile = argv[optind + 1]; wav = wav_read_open(infile); if (!wav) { fprintf(stderr, "Unable to open wav file %s\n", infile); return 1; } if (!wav_get_header(wav, &format, &channels, &sample_rate, &bits_per_sample, NULL)) { fprintf(stderr, "Bad wav file %s\n", infile); return 1; } if (format != 1) { fprintf(stderr, "Unsupported WAV format %d\n", format); return 1; } if (bits_per_sample != 16) { fprintf(stderr, "Unsupported WAV sample depth %d\n", bits_per_sample); return 1; } switch (channels) { case 1: mode = MODE_1; break; case 2: mode = MODE_2; break; case 3: mode = MODE_1_2; break; case 4: mode = MODE_1_2_1; break; case 5: mode = MODE_1_2_2; break; case 6: mode = MODE_1_2_2_1; break; default: fprintf(stderr, "Unsupported WAV channels %d\n", channels); return 1; } if (aacEncOpen(&handle, 0, channels) != AACENC_OK) { fprintf(stderr, "Unable to open encoder\n"); return 1; } if (aacEncoder_SetParam(handle, AACENC_AOT, aot) != AACENC_OK) { fprintf(stderr, "Unable to set the AOT\n"); return 1; } if (aot == 39 && eld_sbr) { if (aacEncoder_SetParam(handle, AACENC_SBR_MODE, 1) != AACENC_OK) { fprintf(stderr, "Unable to set SBR mode for ELD\n"); return 1; } } if (aacEncoder_SetParam(handle, AACENC_SAMPLERATE, sample_rate) != AACENC_OK) { fprintf(stderr, "Unable to set the AOT\n"); return 1; } if (aacEncoder_SetParam(handle, AACENC_CHANNELMODE, mode) != AACENC_OK) { fprintf(stderr, "Unable to set the channel mode\n"); return 1; } if (aacEncoder_SetParam(handle, AACENC_CHANNELORDER, 1) != AACENC_OK) { fprintf(stderr, "Unable to set the wav channel order\n"); return 1; } if (vbr) { if (aacEncoder_SetParam(handle, AACENC_BITRATEMODE, vbr) != AACENC_OK) { fprintf(stderr, "Unable to set the VBR bitrate mode\n"); return 1; } } else { if (aacEncoder_SetParam(handle, AACENC_BITRATE, bitrate) != AACENC_OK) { fprintf(stderr, "Unable to set the bitrate\n"); return 1; } } if (aacEncoder_SetParam(handle, AACENC_TRANSMUX, TT_MP4_ADTS) != AACENC_OK) { fprintf(stderr, "Unable to set the ADTS transmux\n"); return 1; } if (aacEncoder_SetParam(handle, AACENC_AFTERBURNER, afterburner) != AACENC_OK) { fprintf(stderr, "Unable to set the afterburner mode\n"); return 1; } if (aacEncEncode(handle, NULL, NULL, NULL, NULL) != AACENC_OK) { fprintf(stderr, "Unable to initialize the encoder\n"); return 1; } if (aacEncInfo(handle, &info) != AACENC_OK) { fprintf(stderr, "Unable to get the encoder info\n"); return 1; } out = fopen(outfile, "wb"); if (!out) { perror(outfile); return 1; } input_size = channels*2*info.frameLength; input_buf = (uint8_t*) malloc(input_size); convert_buf = (int16_t*) malloc(input_size); while (1) { AACENC_BufDesc in_buf = { 0 }, out_buf = { 0 }; AACENC_InArgs in_args = { 0 }; AACENC_OutArgs out_args = { 0 }; int in_identifier = IN_AUDIO_DATA; int in_size, in_elem_size; int out_identifier = OUT_BITSTREAM_DATA; int out_size, out_elem_size; int read, i; void *in_ptr, *out_ptr; uint8_t outbuf[20480]; AACENC_ERROR err; read = wav_read_data(wav, input_buf, input_size); for (i = 0; i < read/2; i++) { const uint8_t* in = &input_buf[2*i]; convert_buf[i] = in[0] | (in[1] << 8); } in_ptr = convert_buf; in_size = read; in_elem_size = 2; in_args.numInSamples = read <= 0 ? -1 : read/2; in_buf.numBufs = 1; in_buf.bufs = &in_ptr; in_buf.bufferIdentifiers = &in_identifier; in_buf.bufSizes = &in_size; in_buf.bufElSizes = &in_elem_size; out_ptr = outbuf; out_size = sizeof(outbuf); out_elem_size = 1; out_buf.numBufs = 1; out_buf.bufs = &out_ptr; out_buf.bufferIdentifiers = &out_identifier; out_buf.bufSizes = &out_size; out_buf.bufElSizes = &out_elem_size; if ((err = aacEncEncode(handle, &in_buf, &out_buf, &in_args, &out_args)) != AACENC_OK) { if (err == AACENC_ENCODE_EOF) break; fprintf(stderr, "Encoding failed\n"); return 1; } if (out_args.numOutBytes == 0) continue; fwrite(outbuf, 1, out_args.numOutBytes, out); } free(input_buf); free(convert_buf); fclose(out); wav_read_close(wav); aacEncClose(&handle); return 0; } fdk-aac-2.0.2/autogen.sh000077500000000000000000000000321404225310700147730ustar00rootroot00000000000000#!/bin/sh autoreconf -fiv fdk-aac-2.0.2/configure.ac000066400000000000000000000021231404225310700152630ustar00rootroot00000000000000dnl -*- Autoconf -*- dnl Process this file with autoconf to produce a configure script. AC_INIT([fdk-aac], [2.0.2], [http://sourceforge.net/projects/opencore-amr/]) AC_CONFIG_AUX_DIR(.) AC_CONFIG_MACRO_DIR([m4]) AM_INIT_AUTOMAKE([tar-ustar foreign]) m4_ifdef([AM_SILENT_RULES], [AM_SILENT_RULES([yes])]) dnl Various options for configure AC_ARG_ENABLE([example], [AS_HELP_STRING([--enable-example], [enable example encoding program (default is no)])], [example=$enableval], [example=no]) dnl Automake conditionals to set AM_CONDITIONAL(EXAMPLE, test x$example = xyes) dnl Checks for programs. 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( ]P/v(] Ђl(YR3>u1[|c7/!;d7kE_hQ|g;ԓNAԈ3Yko8qoF| b`31ٽΌٹPN_hKB{ݹE_h 'oW|u3U|]l gm 65 ‰;A_ 1YY~-̜ß6uu.V2g/o~M_.AH|{m [{<΄N|A|s{9ws o_7!_WȴR3o|# B;t1_&/o~ |7C / dᐯ?;lK͊/.W1Em=Y_^] cоǣޢ߉/$q[[V׹u nXh& ]~8_ha|\ 77-{Cr,5m57_hM|No߈/,gNok@~K&puz[-im&pG]|A|sz֎3}_G^n :Xj&Ђ:ЄW|kC]B;{72X[{~'p}rEnвv:XW|+n_8: :B+먃5QjMZ Ж:ИW|Cu2 Gj7 moו3߲#  `3[iw~@|{#a_ w|R߷W]/ Wq~ڶ=w"Q  KWG__v//٧;O|ėuėc" // / " //// " //// KmnG@V+a endstream endobj 538 0 obj << /Length 2159 /Filter /FlateDecode >> stream xڭXKs6WHU0rp<ǻښښ@ E*|ěۍ(RFh|w^pquYYmŝ0Wʜ6wyX(v/ Eq$pwb*TC֙]YQӰ} (<=ǰo 9O=Р`}dIEu$bmF#pvK}p'*D/B6 kzyy2k} Oi^g%ϽI-q̔w*Loa}Y]v!<)!BhT. &aSYw0 Fx/!I#D^@|487QELrx:#,#LiR@ X'd&o(٦Q[d]>)5lx~Bc/r?1`@6tuI~=h% o>SgFǡ΃,̳e-#lN%0r5/2xTnCyLw;ĞT"PI0JM5Ak[ek_#t(|^k9F`׾g'_('%V0,4gmRV?;j2|tVkGV[d溙fٜ]u.GZ C`_ϤXGT=U^ICcvTi &=J3)ToobQ佉^g"DR/VɁCsO~r{TݛaHbsȂ){ʼX4[ ZQw!jYi/5uW}K8\pA fTO=QRH=VPt"^:eƉnWƏC7 io2$>fo $Ma\.EJ)4g^${}h-ߑ1 |EA(qEy„ۼ5#gV*q2gq >=<>B3^W*PO?_[> stream xZs6_GjBMsDZ[⋕f>mqF"]Q JLrk'r v%4եALbu0 bDY0_BA 5gG&|wv}}5.}`Жz|\S-  ?z??:v@70H$T0Y|SX)D&xtT@r<9 v{胷˚JKD"߬i,t>0N?Rf%' ]|u& sXEpzzfEG(HP" NbohED)4l{?Bà#$#Q$DwԭUc6N؍0H1ʧ$-$bGz6^1F7۵Eax"bzXp~kݸsK\{s@ jNq=)y1`*|,OnW >V!eNIfh5<4a,)!dgj+0&RK2uhxug*AZ;/"i,]ppH['sU}waHoڨjk&L4هWE;C(t^.m"Fz* $)S;aqm<Ĕ!@3c|"r]L# QdvӐu?rnCb`Dk[]-(# Z줤 z0; T(Lp'9nft.>β 6Qƙ:߯Ye$+g7h2J$iV|H'e2gi/8u-b=BrāHEI;?ŏ y{y!1Pn5x:)5l4HA8rc0wMZL*_ k)6>Xvmf88iӴ&'iЋ8@ 69[ŲO@Ea~JuI }kB9oOp~{m=9mv(iu6b ? %rjKHu+%h]Sq56 :pr2K9NQ(qf<ڼJF@eօ]T_FAu&iSdW K(n\aSM<bQg m^[?zK蕫Sztfcpt*˧p:%oB6xsja#gKQ\ XݚϷNvl-KO~L.O*"!W3ew/6k;0)˴|Π}#k\ T09ۢUVRe't?La(ihu71T#ԫsķu[8pRj<ݧD+lZ癯j\o KW ,hXEp\l?Ul];T̟;vslS1cMNk%(W^W%:;:CJ?ǫmm &na--^W*oEzk[f;kPz?Ջ)^0_S>^e* \ sz>@ q$UFǐi$B5ev`wX۩5i"B! PBn jua `w@&Lѧ)\TMW.m^+,mg6SRc+ă[DS͎݌ _ ^@|gçW頏['lxZʐɽB%k ϋ]t.m][Xntmyѩu Z}pVO霹[NCTƆAi@ mo7pTop=Bv//JO*u˦&U6Nʪ +`vEa+U_rԪЎ B۹v~&#DF|~?XTD8utWqID endstream endobj 580 0 obj << /Length 417 /Filter /FlateDecode >> stream xڽSMO0ёg"T 7^RHPzif<#7ns|4 W"CQኣΡ|Uueu}+ItYFPTKCZ&0e)S$S:LVՌROpX 1 o1W?oߜd~*Jdt\^0]q˜JVb{| qWB_z=)ko9sumaZ1bT}*nDa~kH݇WCJ> stream xڍOn@+ ݽB@BT ҄* R"YL8F4 _1y #zTzuHcS].b~ ܫvWn28ĸ L65U78љG'j,Ywg@^f.~A endstream endobj 590 0 obj << /Length 399 /Filter /FlateDecode >> stream xŒR0 y CdGN̍eXV8f&vݷ_ÙX"yљ6QY98ˬ4kh始yE&⡽LDꦦH S Oտ/{*e*:%!zl)gRK_KIKa.[bRr'/vǰfwR?M/j9{xy@I4hWC3vf/ ClU X)= P r wٍ8QhFkUVOcQa iN^ON it6HEwJPzd=DrsU-OJ5皾pZ#/0<|(M O+ endstream endobj 594 0 obj << /Length 184 /Filter /FlateDecode >> stream xڍO90)e׎H !H  T|+!jV;9W0fjհ.NjAо̖.xd>ZjGڬ>?ƅz(I iS'o,Nwu82 %B7Fq_^!_n|/$E /_@o endstream endobj 623 0 obj << /Length 1011 /Filter /FlateDecode >> stream xXs6L/p@+ L3NmI٘:=`Pf0 v}pbٸ8}ޓD넘gG"F1G$ x(LЃu*m_ën~fA8峨:Y+HEI9vNnPBu>ץV͈Ut`uN=ڔXTyj$f>ƦJ+TX.ULLXʈj㳰VhQd6u F7QqIjO"aq9Y֭}!,1Qox8E!wB9r(Äş<Wž &{l$' QĘHPūWvv{+B ˠQsG-Hƀ3m58 ǃF6x (B\r' bI]lZ!U!ߍ}-0zí/=˛,71*&Xw@˺h]+u1y,ow7h ZDĺ]~7(bdvVS \LmzssqX6!f`[+]}# ;`JhS` *@n\yX!Tw{,7Q3Ue~,v^:K ln@t;30CHI endstream endobj 486 0 obj << /Type /ObjStm /N 100 /First 886 /Length 2562 /Filter /FlateDecode >> stream xZo~ 3  q4@quH,o(ŒbllA"wo;| ֐B(PN -.$b:95y#,!B-֐ t| *ޙ1cL6EEWXʘ6 ̣.W\$J=I H?f(j) SA+.4tTu{/T1pPZ>8u6_1Q,er* [E|Vh)Ts>Kr; e#j/bBFBCzka<к%F;5`1 n&onu_8),&U̢U70nP}Ҧz'(qz`J&q\IDz}e;De1CDn$,1Qμyh}rzr= 'p#.< .!r xᢸľjtzga÷O|U?? 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CO`=aFeLnڮ;_-XӞZs +mC|;?.b127BRA#Thdl4|MȪh=y,Re9 Ա|L,שE Vl7"J !9p!XY.ULMeY*&>f Q=rVӨ~QVVQ<ߖ* 7_}ųh]xyoDM~w2 k-KgRZ鸎^ ~,0%C[_sV=r qQӂMFlGsN ~G{UC'>eeNWuDB$Rw]̯ges K eEPOiHѼjF%BKJdzVmOYq)kqzoRdI{!c#֎kU`둕L l3z.j{Ig\ѢBYY)%9+$ML%=sr{?S;b׹8.щ}c[Q4f" 5Rո|e*J)ks5grp5A =_{n Ftel9Գ=iz| u\@,Xr}`.o`bۦl\5L=$NXmY]` z^YԞO9{R.fEV|VXr!{j,G9.ѩ:=.q xڮ*8p߽_on,$_"EBbv zv1hPHD, b-/müdZlw5UfM^K'>\% V@c*A_ i*.nRpq (.755=u_z44\B(I0 Yj(ayEJ- y#z%_&E5 Q PNחG O#Is.^o ,bϻ7S큰K_ ęY.._c endstream endobj 663 0 obj << /Length 1591 /Filter /FlateDecode >> stream xX[s6~B@/coMt} FbL{#llf!ܾsZY88OZ+diE #ˬ$mA:u<'fAhzȩ}7NpOx2O#f&Ej4z2K̥HO.ofɧ$|0БZL$|kV,J(]šC`o E!vsU5hMsWmt0.ܤLRq %J\Vggw8@IĹ0Cv:oyʕqj7\l'iM ږNı̩ ["mFaWGB. 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Ltd, Bangalore */ package { // See: http://go/android-license-faq // A large-scale-change added 'default_applicable_licenses' to import // all of the 'license_kinds' from "external_aac_license" // to get the below license kinds: // SPDX-license-identifier-Apache-2.0 default_applicable_licenses: ["external_aac_license"], } cc_defaults { name: "aac_fuzz_defaults", host_supported: true, static_libs: [ "libFraunhoferAAC", ], target: { darwin: { enabled: false, }, }, fuzz_config: { cc: [ "android-media-fuzzing-reports@google.com", ], componentid: 155276, }, } cc_fuzz { name: "aac_dec_fuzzer", srcs: [ "aac_dec_fuzzer.cpp", ], static_libs: [ "liblog", ], defaults: [ "aac_fuzz_defaults" ], } cc_fuzz { name: "aac_enc_fuzzer", srcs: [ "aac_enc_fuzzer.cpp", ], defaults: [ "aac_fuzz_defaults" ], include_dirs: [ "external/aac/libAACenc/" ], } fdk-aac-2.0.2/fuzzer/README.md000066400000000000000000000215441404225310700156110ustar00rootroot00000000000000# Fuzzer for libFraunhoferAAC decoder ## Plugin Design Considerations The fuzzer plugin for aac decoder is designed based on the understanding of the codec and tries to achieve the following: ##### Maximize code coverage This fuzzer makes use of the following config parameters: 1. Transport type (parameter name: `TRANSPORT_TYPE`) | Parameter| Valid Values| Configured Value| |------------- |-------------| ----- | | `TRANSPORT_TYPE` | 0.`TT_UNKNOWN ` 1.`TT_MP4_RAW ` 2.`TT_MP4_ADIF ` 3.`TT_MP4_ADTS ` 4.`TT_MP4_LATM_MCP1 ` 5.`TT_MP4_LATM_MCP0 ` 6.`TT_MP4_LOAS ` 7.`TT_DRM ` | `TT_MP4_ADIF ` | Note: Value of `TRANSPORT_TYPE` could be set to any of these values. It is set to `TT_MP4_ADIF` in the fuzzer plugin. ##### Maximize utilization of input data The plugin feeds the entire input data to the codec using a loop. * If the decode operation was successful, the input is advanced by an offset calculated using valid bytes. * If the decode operation was un-successful, the input is advanced by 1 byte till it reaches a valid frame or end of stream. This ensures that the plugin tolerates any kind of input (empty, huge, malformed, etc) and doesnt `exit()` on any input and thereby increasing the chance of identifying vulnerabilities. ## Build This describes steps to build aac_dec_fuzzer binary. ## Android ### Steps to build Build the fuzzer ``` $ mm -j$(nproc) aac_dec_fuzzer ``` ### Steps to run Create a directory CORPUS_DIR and copy some aac files to that folder. Push this directory to device. To run on device ``` $ adb sync data $ adb shell /data/fuzz/arm64/aac_dec_fuzzer/aac_dec_fuzzer CORPUS_DIR ``` To run on host ``` $ $ANDROID_HOST_OUT/fuzz/x86_64/aac_dec_fuzzer/aac_dec_fuzzer CORPUS_DIR ``` # Fuzzer for libFraunhoferAAC encoder ## Plugin Design Considerations The fuzzer plugin for aac encoder is designed based on the understanding of the codec and tries to achieve the following: ##### Maximize code coverage The configuration parameters are not hardcoded, but instead selected based on incoming data. This ensures more code paths are reached by the fuzzer. Following arguments are passed to aacEncoder_SetParam to set the respective AACENC_PARAM parameter: | AACENC_PARAM param| Valid Values| Configured Value| |-------------| ----- |----- | |`AACENC_SBR_MODE` | `-1 ` `0 ` `1 ` `2 ` | Calculated using first byte of data | |`AACENC_AOT` |`AOT_NONE ` `AOT_NULL_OBJECT ` `AOT_AAC_MAIN ` `AOT_AAC_LC ` `AOT_AAC_SSR ` `AOT_AAC_LTP ` `AOT_SBR ` `AOT_AAC_SCAL ` `AOT_TWIN_VQ ` `AOT_CELP ` `AOT_HVXC ` `AOT_RSVD_10 ` `AOT_RSVD_11 ` `AOT_TTSI ` `AOT_MAIN_SYNTH ` `AOT_WAV_TAB_SYNTH ` `AOT_GEN_MIDI ` `AOT_ALG_SYNTH_AUD_FX ` `AOT_ER_AAC_LC ` `AOT_RSVD_18 ` `AOT_ER_AAC_LTP ` `AOT_ER_AAC_SCAL ` `AOT_ER_TWIN_VQ ` `AOT_ER_BSAC ` `AOT_ER_AAC_LD ` `AOT_ER_CELP ` `AOT_ER_HVXC ` `AOT_ER_HILN ` `AOT_ER_PARA ` `AOT_RSVD_28 ` `AOT_PS ` `AOT_MPEGS ` `AOT_ESCAPE ` `AOT_MP3ONMP4_L1 ` `AOT_MP3ONMP4_L2 ` `AOT_MP3ONMP4_L3 ` `AOT_RSVD_35 ` `AOT_RSVD_36 ` `AOT_AAC_SLS ` `AOT_SLS ` `AOT_ER_AAC_ELD ` `AOT_USAC ` `AOT_SAOC ` `AOT_LD_MPEGS ` `AOT_MP2_AAC_LC ` `AOT_MP2_SBR ` `AOT_DRM_AAC ` `AOT_DRM_SBR ` `AOT_DRM_MPEG_PS ` `AOT_DRM_SURROUND ` `AOT_DRM_USAC ` | Calculated using second byte of data | |`AACENC_SAMPLERATE` | `8000 ` `11025 ` `12000 ` `16000 ` `22050 ` `24000 ` `32000 ` `44100 ` `48000 ` `64000 ` `88200 ` `96000 `| Calculated using third byte of data | |`AACENC_BITRATE` | In range `8000 ` to `960000 ` | Calculated using fourth, fifth and sixth byte of data | |`AACENC_CHANNELMODE` | `MODE_1 ` `MODE_2 ` `MODE_1_2 ` `MODE_1_2_1 ` `MODE_1_2_2 ` `MODE_1_2_2_1 ` `MODE_1_2_2_2_1 ` `MODE_6_1 ` `MODE_7_1_BACK ` `MODE_7_1_TOP_FRONT ` `MODE_7_1_REAR_SURROUND ` `MODE_7_1_FRONT_CENTER ` `MODE_212 ` | Calculated using seventh byte of data | |`AACENC_TRANSMUX` | `TT_MP4_RAW ` `TT_MP4_ADIF ` `TT_MP4_ADTS ` `TT_MP4_LATM_MCP1 ` `TT_MP4_LATM_MCP0 ` `TT_MP4_LOAS ` `TT_DRM ` | Calculated using eight byte of data |`AACENC_SBR_RATIO` |`-1 ` `0 ` `1 ` `2 ` | Calculated using ninth byte of data | |`AACENC_BITRATEMODE` |`AACENC_BR_MODE_INVALID ` `AACENC_BR_MODE_CBR ` `AACENC_BR_MODE_VBR_1 ` `AACENC_BR_MODE_VBR_2 ` `AACENC_BR_MODE_VBR_3 ` `AACENC_BR_MODE_VBR_4 ` `AACENC_BR_MODE_VBR_5 ` `AACENC_BR_MODE_FF ` `AACENC_BR_MODE_SFR ` | Calculated using thirty-fourth byte of data | |`AACENC_GRANULE_LENGTH` |`120 ` `128 ` `240 ` `256 ` `480 ` `512 ` `1024 ` | Calculated using thirty-fifth byte of data | |`AACENC_CHANNELORDER` |`CH_ORDER_MPEG ` `CH_ORDER_WAV ` | Calculated using thirty-sixth byte of data | |`AACENC_AFTERBURNER` |`0 ` `1 ` | Calculated using thirty-seventh byte of data | |`AACENC_BANDWIDTH` |`0 ` `1` | Calculated using thirty-eigth byte of data | |` AACENC_IDX_PEAK_BITRATE` | In range `8000 ` to `960000 ` | Calculated using thirty-ninth byte of data | |` AACENC_HEADER_PERIOD` |In range `0 ` to `255 ` | Calculated using fortieth byte of data | |` AACENC_SIGNALING_MODE` |`-1 ` `0 ` `1 ` `2 ` `3 ` | Calculated using forty-first byte of data | |` AACENC_TPSUBFRAMES` |In range `0 ` to `255 ` | Calculated using forty-second byte of data | |` AACENC_AUDIOMUXVER` |`-1 ` `0 ` `1 ` `2 ` | Calculated using forty-third byte of data | |` AACENC_PROTECTION` |`0 ` `1 ` | Calculated using forty-fourth of data | |`AACENC_ANCILLARY_BITRATE` |In range `0 ` to `960000 `| Calculated using forty-fifth byte of data | |`AACENC_METADATA_MODE ` |`0 ` `1 ` `2 ` `3 ` | Calculated using forty-sixth byte of data | Following values are configured to set up the meta data represented by the class variable `mMetaData ` : | Variable name| Possible Values| Configured Value| |------------- | ----- |----- | | `drc_profile` | `AACENC_METADATA_DRC_NONE ` `AACENC_METADATA_DRC_FILMSTANDARD ` `AACENC_METADATA_DRC_FILMLIGHT ` `AACENC_METADATA_DRC_MUSICSTANDARD ` `AACENC_METADATA_DRC_MUSICLIGHT ` `AACENC_METADATA_DRC_SPEECH ` `AACENC_METADATA_DRC_NOT_PRESENT ` | Calculated using tenth byte of data | | `comp_profile` | `AACENC_METADATA_DRC_NONE ` `AACENC_METADATA_DRC_FILMSTANDARD ` `AACENC_METADATA_DRC_FILMLIGHT ` `AACENC_METADATA_DRC_MUSICSTANDARD ` `AACENC_METADATA_DRC_MUSICLIGHT ` `AACENC_METADATA_DRC_SPEECH ` `AACENC_METADATA_DRC_NOT_PRESENT ` | Calculated using eleventh byte of data | | `drc_TargetRefLevel` | In range `0 ` to `255 ` | Calculated using twelfth byte of data | | `comp_TargetRefLevel` | In range `0 ` to `255 ` | Calculated using thirteenth byte of data | | `prog_ref_level_present` | `0 ` `1 ` | Calculated using fourteenth byte of data | | `prog_ref_level` | In range `0 ` to `255 ` | Calculated using fifteenth byte of data | | `PCE_mixdown_idx_present` | `0 ` `1 ` | Calculated using sixteenth byte of data | | `ETSI_DmxLvl_present` | `0 ` `1 ` | Calculated using seventeenth byte of data | | `centerMixLevel` | In range `0 ` to `7 ` | Calculated using eighteenth byte of data | | `surroundMixLevel` | In range `0 ` to `7 ` | Calculated using nineteenth byte of data | | `dolbySurroundMode` | In range `0 ` to `2 ` | Calculated using twentieth byte of data | | `drcPresentationMode` | In range `0 ` to `2 ` | Calculated using twenty-first byte of data | | `extAncDataEnable` | `0 ` `1 ` | Calculated using twenty-second byte of data | | `extDownmixLevelEnable` | `0 ` `1 ` | Calculated using twenty-third byte of data | | `extDownmixLevel_A` | In range `0 ` to `7 ` | Calculated using twenty-fourth byte of data | | `extDownmixLevel_B` | In range `0 ` to `7 ` | Calculated using twenty-fifth byte of data | | `dmxGainEnable` | `0 ` `1 ` | Calculated using twenty-sixth byte of data | | `dmxGain5` | In range `0 ` to `255 ` | Calculated using twenty-seventh byte of data | | `dmxGain2` | In range `0 ` to `255 ` | Calculated using twenty-eighth byte of data | | `lfeDmxEnable` | `0 ` `1 ` | Calculated using twenty-ninth byte of data | | `lfeDmxLevel` | In range `0 ` to `15 ` | Calculated using thirtieth byte of data | Indexes `mInBufferIdx_1`, `mInBufferIdx_2` and `mInBufferIdx_3`(in range `0 ` to `2`) are calculated using the thirty-first, thirty-second and thirty-third byte respectively. ##### Maximize utilization of input data The plugin feeds the entire input data to the codec and continues with the encoding even on a failure. This ensures that the plugin tolerates any kind of input (empty, huge, malformed, etc) and doesnt `exit()` on any input and thereby increasing the chance of identifying vulnerabilities. ## Build This describes steps to build aac_enc_fuzzer binary. ## Android ### Steps to build Build the fuzzer ``` $ mm -j$(nproc) aac_enc_fuzzer ``` ### Steps to run Create a directory CORPUS_DIR and copy some raw files to that folder. Push this directory to device. To run on device ``` $ adb sync data $ adb shell /data/fuzz/arm64/aac_enc_fuzzer/aac_enc_fuzzer CORPUS_DIR ``` To run on host ``` $ $ANDROID_HOST_OUT/fuzz/x86_64/aac_enc_fuzzer/aac_enc_fuzzer CORPUS_DIR ``` ## References: * http://llvm.org/docs/LibFuzzer.html * https://github.com/google/oss-fuzz fdk-aac-2.0.2/fuzzer/aac_dec_fuzzer.cpp000066400000000000000000000101531404225310700177740ustar00rootroot00000000000000/****************************************************************************** * * Copyright (C) 2020 The Android Open Source Project * * Licensed under the Apache License, Version 2.0 (the "License"); * you may not use this file except in compliance with the License. * You may obtain a copy of the License at: * * http://www.apache.org/licenses/LICENSE-2.0 * * Unless required by applicable law or agreed to in writing, software * distributed under the License is distributed on an "AS IS" BASIS, * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. * See the License for the specific language governing permissions and * limitations under the License. * ***************************************************************************** * Originally developed and contributed by Ittiam Systems Pvt. Ltd, Bangalore */ #include #include #include #include "aacdecoder_lib.h" constexpr uint8_t kNumberOfLayers = 1; constexpr uint8_t kMaxChannelCount = 8; constexpr uint32_t kMaxConfigurationSize = 1024; constexpr uint32_t kMaxOutBufferSize = 2048 * kMaxChannelCount; // Value indicating the start of AAC Header Segment constexpr const char *kAacSegStartSeq = "AAC_STRT"; constexpr uint8_t kAacSegStartSeqLen = sizeof(kAacSegStartSeq); // Value indicating the end of AAC Header Segment constexpr const char *kAacSegEndSeq = "AAC_ENDS"; constexpr uint8_t kAacSegEndSeqLen = sizeof(kAacSegEndSeq); // Number of bytes used to signal the length of the header constexpr uint8_t kHeaderLengthBytes = 2; // Minimum size of an AAC header is 2 // Minimum data required is // strlen(AAC_STRT) + strlen(AAC_ENDS) + kHeaderLengthBytes + 2; constexpr UINT kMinDataSize = kAacSegStartSeqLen + kAacSegEndSeqLen + kHeaderLengthBytes + 2; UINT getHeaderSize(UCHAR *data, UINT size) { if (size < kMinDataSize) { return 0; } int32_t result = memcmp(data, kAacSegStartSeq, kAacSegStartSeqLen); if (result) { return 0; } data += kAacSegStartSeqLen; size -= kAacSegStartSeqLen; uint32_t headerLengthInBytes = (data[0] << 8 | data[1]) & 0xFFFF; data += kHeaderLengthBytes; size -= kHeaderLengthBytes; if (headerLengthInBytes + kAacSegEndSeqLen > size) { return 0; } data += headerLengthInBytes; size -= headerLengthInBytes; result = memcmp(data, kAacSegEndSeq, kAacSegEndSeqLen); if (result) { return 0; } return std::min(headerLengthInBytes, kMaxConfigurationSize); } class Codec { public: Codec() = default; ~Codec() { deInitDecoder(); } bool initDecoder(); void decodeFrames(UCHAR *data, UINT size); void deInitDecoder(); private: HANDLE_AACDECODER mAacDecoderHandle = nullptr; AAC_DECODER_ERROR mErrorCode = AAC_DEC_OK; }; bool Codec::initDecoder() { mAacDecoderHandle = aacDecoder_Open(TT_MP4_ADIF, kNumberOfLayers); if (!mAacDecoderHandle) { return false; } return true; } void Codec::deInitDecoder() { aacDecoder_Close(mAacDecoderHandle); mAacDecoderHandle = nullptr; } void Codec::decodeFrames(UCHAR *data, UINT size) { UINT headerSize = getHeaderSize(data, size); if (headerSize != 0) { data += kAacSegStartSeqLen + kHeaderLengthBytes; size -= kAacSegStartSeqLen + kHeaderLengthBytes; aacDecoder_ConfigRaw(mAacDecoderHandle, &data, &headerSize); data += headerSize + kAacSegEndSeqLen; size -= headerSize + kAacSegEndSeqLen; } while (size > 0) { UINT inputSize = size; UINT valid = size; mErrorCode = aacDecoder_Fill(mAacDecoderHandle, &data, &inputSize, &valid); if (mErrorCode != AAC_DEC_OK) { ++data; --size; } else { INT_PCM outputBuf[kMaxOutBufferSize]; do { mErrorCode = aacDecoder_DecodeFrame(mAacDecoderHandle, outputBuf, sizeof(outputBuf), 0); } while (mErrorCode == AAC_DEC_OK); UINT offset = inputSize - valid; data += offset; size = valid; } } } extern "C" int LLVMFuzzerTestOneInput(const uint8_t *data, size_t size) { Codec *codec = new Codec(); if (!codec) { return 0; } if (codec->initDecoder()) { codec->decodeFrames((UCHAR *)(data), static_cast(size)); } delete codec; return 0; } fdk-aac-2.0.2/fuzzer/aac_enc_fuzzer.cpp000066400000000000000000000477051404225310700200230ustar00rootroot00000000000000/****************************************************************************** * * Copyright (C) 2020 The Android Open Source Project * * Licensed under the Apache License, Version 2.0 (the "License"); * you may not use this file except in compliance with the License. * You may obtain a copy of the License at: * * http://www.apache.org/licenses/LICENSE-2.0 * * Unless required by applicable law or agreed to in writing, software * distributed under the License is distributed on an "AS IS" BASIS, * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. * See the License for the specific language governing permissions and * limitations under the License. * ***************************************************************************** * Originally developed and contributed by Ittiam Systems Pvt. Ltd, Bangalore */ #include #include "aacenc_lib.h" #include "src/aacenc.h" using namespace std; // IN_AUDIO_DATA, IN_ANCILLRY_DATA and IN_METADATA_SETUP constexpr size_t kMaxBuffers = 3; constexpr size_t kMaxOutputBufferSize = 8192; constexpr uint32_t kMinBitRate = 8000; constexpr uint32_t kMaxBitRate = 960000; constexpr int32_t kSampleRates[] = {8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 64000, 88200, 96000}; constexpr size_t kSampleRatesSize = size(kSampleRates); constexpr CHANNEL_MODE kChannelModes[] = {MODE_1, MODE_2, MODE_1_2, MODE_1_2_1, MODE_1_2_2, MODE_1_2_2_1, MODE_1_2_2_2_1, MODE_6_1, MODE_7_1_BACK, MODE_7_1_TOP_FRONT, MODE_7_1_REAR_SURROUND, MODE_7_1_FRONT_CENTER, MODE_212}; constexpr size_t kChannelModesSize = size(kChannelModes); constexpr TRANSPORT_TYPE kIdentifiers[] = { TT_MP4_RAW, TT_MP4_ADIF, TT_MP4_ADTS, TT_MP4_LATM_MCP1, TT_MP4_LATM_MCP0, TT_MP4_LOAS, TT_DRM}; constexpr size_t kIdentifiersSize = size(kIdentifiers); constexpr AUDIO_OBJECT_TYPE kAudioObjectTypes[] = {AOT_NONE, AOT_NULL_OBJECT, AOT_AAC_MAIN, AOT_AAC_LC, AOT_AAC_SSR, AOT_AAC_LTP, AOT_SBR, AOT_AAC_SCAL, AOT_TWIN_VQ, AOT_CELP, AOT_HVXC, AOT_RSVD_10, AOT_RSVD_11, AOT_TTSI, AOT_MAIN_SYNTH, AOT_WAV_TAB_SYNTH, AOT_GEN_MIDI, AOT_ALG_SYNTH_AUD_FX, AOT_ER_AAC_LC, AOT_RSVD_18, AOT_ER_AAC_LTP, AOT_ER_AAC_SCAL, AOT_ER_TWIN_VQ, AOT_ER_BSAC, AOT_ER_AAC_LD, AOT_ER_CELP, AOT_ER_HVXC, AOT_ER_HILN, AOT_ER_PARA, AOT_RSVD_28, AOT_PS, AOT_MPEGS, AOT_ESCAPE, AOT_MP3ONMP4_L1, AOT_MP3ONMP4_L2, AOT_MP3ONMP4_L3, AOT_RSVD_35, AOT_RSVD_36, AOT_AAC_SLS, AOT_SLS, AOT_ER_AAC_ELD, AOT_USAC, AOT_SAOC, AOT_LD_MPEGS, AOT_MP2_AAC_LC, AOT_MP2_SBR, AOT_DRM_AAC, AOT_DRM_SBR, AOT_DRM_MPEG_PS, AOT_DRM_SURROUND, AOT_DRM_USAC}; constexpr size_t kAudioObjectTypesSize = size(kAudioObjectTypes); constexpr int32_t kSbrRatios[] = {-1, 0, 1, 2}; constexpr size_t kSbrRatiosSize = size(kSbrRatios); constexpr int32_t kBitRateModes[] = { AACENC_BR_MODE_INVALID, AACENC_BR_MODE_CBR, AACENC_BR_MODE_VBR_1, AACENC_BR_MODE_VBR_2, AACENC_BR_MODE_VBR_3, AACENC_BR_MODE_VBR_4, AACENC_BR_MODE_VBR_5, AACENC_BR_MODE_FF, AACENC_BR_MODE_SFR}; constexpr size_t kBitRateModesSize = size(kBitRateModes); constexpr int32_t kGranuleLengths[] = {120, 128, 240, 256, 480, 512, 1024}; constexpr size_t kGranuleLengthsSize = size(kGranuleLengths); constexpr int32_t kChannelOrder[] = {CH_ORDER_MPEG, CH_ORDER_WAV}; constexpr size_t kChannelOrderSize = size(kChannelOrder); constexpr int32_t kSignalingModes[] = {-1, 0, 1, 2, 3}; constexpr size_t kSignalingModesSize = size(kSignalingModes); constexpr int32_t kAudioMuxVer[] = {-1, 0, 1, 2}; constexpr size_t kAudioMuxVerSize = size(kAudioMuxVer); constexpr int32_t kSbrModes[] = {-1, 0, 1, 2}; constexpr size_t kSbrModesSize = size(kSbrModes); constexpr AACENC_METADATA_DRC_PROFILE kMetaDataDrcProfiles[] = { AACENC_METADATA_DRC_NONE, AACENC_METADATA_DRC_FILMSTANDARD, AACENC_METADATA_DRC_FILMLIGHT, AACENC_METADATA_DRC_MUSICSTANDARD, AACENC_METADATA_DRC_MUSICLIGHT, AACENC_METADATA_DRC_SPEECH, AACENC_METADATA_DRC_NOT_PRESENT}; constexpr size_t kMetaDataDrcProfilesSize = size(kMetaDataDrcProfiles); enum { IDX_SBR_MODE = 0, IDX_AAC_AOT, IDX_SAMPLE_RATE, IDX_BIT_RATE_1, IDX_BIT_RATE_2, IDX_BIT_RATE_3, IDX_CHANNEL, IDX_IDENTIFIER, IDX_SBR_RATIO, IDX_METADATA_DRC_PROFILE, IDX_METADATA_COMP_PROFILE, IDX_METADATA_DRC_TARGET_REF_LEVEL, IDX_METADATA_COMP_TARGET_REF_LEVEL, IDX_METADATA_PROG_LEVEL_PRESENT, IDX_METADATA_PROG_LEVEL, IDX_METADATA_PCE_MIXDOWN_IDX_PRESENT, IDX_METADATA_ETSI_DMXLVL_PRESENT, IDX_METADATA_CENTER_MIX_LEVEL, IDX_METADATA_SURROUND_MIX_LEVEL, IDX_METADATA_DOLBY_SURROUND_MODE, IDX_METADATA_DRC_PRESENTATION_MODE, IDX_METADATA_EXT_ANC_DATA_ENABLE, IDX_METADATA_EXT_DOWNMIX_LEVEL_ENABLE, IDX_METADATA_EXT_DOWNMIX_LEVEL_A, IDX_METADATA_EXT_DOWNMIX_LEVEL_B, IDX_METADATA_DMX_GAIN_ENABLE, IDX_METADATA_DMX_GAIN_5, IDX_METADATA_DMX_GAIN_2, IDX_METADATA_LFE_DMX_ENABLE, IDX_METADATA_LFE_DMX_LEVEL, IDX_IN_BUFFER_INDEX_1, IDX_IN_BUFFER_INDEX_2, IDX_IN_BUFFER_INDEX_3, IDX_BIT_RATE_MODE, IDX_GRANULE_LENGTH, IDX_CHANNELORDER, IDX_AFTERBURNER, IDX_BANDWIDTH, IDX_PEAK_BITRATE, IDX_HEADER_PERIOD, IDX_SIGNALING_MODE, IDX_TPSUBFRAMES, IDX_AUDIOMUXVER, IDX_PROTECTION, IDX_ANCILLARY_BITRATE, IDX_METADATA_MODE, IDX_LAST }; template auto generateNumberInRangeFromData(type1 data, type2 min, type3 max) -> decltype(max) { return (data % (1 + max - min)) + min; } class Codec { public: ~Codec() { deInitEncoder(); } bool initEncoder(uint8_t **dataPtr, size_t *sizePtr); void encodeFrames(const uint8_t *data, size_t size); void deInitEncoder(); private: template void setAACParam(type1 data, const AACENC_PARAM aacParam, type2 min, type2 max, const type3 *array = nullptr); void setupMetaData(uint8_t *data); HANDLE_AACENCODER mEncoder = nullptr; AACENC_MetaData mMetaData = {}; uint32_t mInBufferIdx_1 = 0; uint32_t mInBufferIdx_2 = 0; uint32_t mInBufferIdx_3 = 0; }; void Codec::setupMetaData(uint8_t *data) { uint32_t drcProfileIndex = generateNumberInRangeFromData(data[IDX_METADATA_DRC_PROFILE], 0, kMetaDataDrcProfilesSize - 1); AACENC_METADATA_DRC_PROFILE drcProfile = kMetaDataDrcProfiles[drcProfileIndex]; mMetaData.drc_profile = drcProfile; uint32_t compProfileIndex = generateNumberInRangeFromData(data[IDX_METADATA_COMP_PROFILE], 0, kMetaDataDrcProfilesSize - 1); AACENC_METADATA_DRC_PROFILE compProfile = kMetaDataDrcProfiles[compProfileIndex]; mMetaData.comp_profile = compProfile; INT drcTargetRefLevel = generateNumberInRangeFromData(data[IDX_METADATA_DRC_TARGET_REF_LEVEL], 0, UINT8_MAX); mMetaData.drc_TargetRefLevel = drcTargetRefLevel; INT compTargetRefLevel = generateNumberInRangeFromData(data[IDX_METADATA_COMP_TARGET_REF_LEVEL], 0, UINT8_MAX); mMetaData.comp_TargetRefLevel = compTargetRefLevel; INT isProgRefLevelPresent = generateNumberInRangeFromData(data[IDX_METADATA_PROG_LEVEL_PRESENT], 0, 1); mMetaData.prog_ref_level_present = isProgRefLevelPresent; INT progRefLevel = generateNumberInRangeFromData(data[IDX_METADATA_PROG_LEVEL], 0, UINT8_MAX); mMetaData.prog_ref_level = progRefLevel; UCHAR isPCEMixdownIdxPresent = generateNumberInRangeFromData(data[IDX_METADATA_PCE_MIXDOWN_IDX_PRESENT], 0, 1); mMetaData.PCE_mixdown_idx_present = isPCEMixdownIdxPresent; UCHAR isETSIDmxLvlPresent = generateNumberInRangeFromData(data[IDX_METADATA_ETSI_DMXLVL_PRESENT], 0, 1); mMetaData.ETSI_DmxLvl_present = isETSIDmxLvlPresent; SCHAR centerMixLevel = generateNumberInRangeFromData(data[IDX_METADATA_CENTER_MIX_LEVEL], 0, 7); mMetaData.centerMixLevel = centerMixLevel; SCHAR surroundMixLevel = generateNumberInRangeFromData(data[IDX_METADATA_SURROUND_MIX_LEVEL], 0, 7); mMetaData.surroundMixLevel = surroundMixLevel; UCHAR dolbySurroundMode = generateNumberInRangeFromData(data[IDX_METADATA_DOLBY_SURROUND_MODE], 0, 2); mMetaData.dolbySurroundMode = dolbySurroundMode; UCHAR drcPresentationMode = generateNumberInRangeFromData(data[IDX_METADATA_DRC_PRESENTATION_MODE], 0, 2); mMetaData.drcPresentationMode = drcPresentationMode; UCHAR extAncDataEnable = generateNumberInRangeFromData(data[IDX_METADATA_EXT_ANC_DATA_ENABLE], 0, 1); mMetaData.ExtMetaData.extAncDataEnable = extAncDataEnable; UCHAR extDownmixLevelEnable = generateNumberInRangeFromData(data[IDX_METADATA_EXT_DOWNMIX_LEVEL_ENABLE], 0, 1); mMetaData.ExtMetaData.extDownmixLevelEnable = extDownmixLevelEnable; UCHAR extDownmixLevel_A = generateNumberInRangeFromData(data[IDX_METADATA_EXT_DOWNMIX_LEVEL_A], 0, 7); mMetaData.ExtMetaData.extDownmixLevel_A = extDownmixLevel_A; UCHAR extDownmixLevel_B = generateNumberInRangeFromData(data[IDX_METADATA_EXT_DOWNMIX_LEVEL_B], 0, 7); mMetaData.ExtMetaData.extDownmixLevel_B = extDownmixLevel_B; UCHAR dmxGainEnable = generateNumberInRangeFromData(data[IDX_METADATA_DMX_GAIN_ENABLE], 0, 1); mMetaData.ExtMetaData.dmxGainEnable = dmxGainEnable; INT dmxGain5 = generateNumberInRangeFromData(data[IDX_METADATA_DMX_GAIN_5], 0, UINT8_MAX); mMetaData.ExtMetaData.dmxGain5 = dmxGain5; INT dmxGain2 = generateNumberInRangeFromData(data[IDX_METADATA_DMX_GAIN_2], 0, UINT8_MAX); mMetaData.ExtMetaData.dmxGain2 = dmxGain2; UCHAR lfeDmxEnable = generateNumberInRangeFromData(data[IDX_METADATA_LFE_DMX_ENABLE], 0, 1); mMetaData.ExtMetaData.lfeDmxEnable = lfeDmxEnable; UCHAR lfeDmxLevel = generateNumberInRangeFromData(data[IDX_METADATA_LFE_DMX_LEVEL], 0, 15); mMetaData.ExtMetaData.lfeDmxLevel = lfeDmxLevel; } template void Codec::setAACParam(type1 data, const AACENC_PARAM aacParam, type2 min, type2 max, const type3 *array) { auto value = 0; if (array) { uint32_t index = generateNumberInRangeFromData(data, min, max); value = array[index]; } else { value = generateNumberInRangeFromData(data, min, max); } aacEncoder_SetParam(mEncoder, aacParam, value); (void)aacEncoder_GetParam(mEncoder, aacParam); } bool Codec::initEncoder(uint8_t **dataPtr, size_t *sizePtr) { uint8_t *data = *dataPtr; if (AACENC_OK != aacEncOpen(&mEncoder, 0, 0)) { return false; } setAACParam(data[IDX_SBR_MODE], AACENC_SBR_MODE, 0, kSbrModesSize - 1, kSbrModes); setAACParam(data[IDX_SBR_RATIO], AACENC_SBR_RATIO, 0, kSbrRatiosSize - 1, kSbrRatios); setAACParam(data[IDX_AAC_AOT], AACENC_AOT, 0, kAudioObjectTypesSize - 1, kAudioObjectTypes); setAACParam(data[IDX_SAMPLE_RATE], AACENC_SAMPLERATE, 0, kSampleRatesSize - 1, kSampleRates); uint32_t tempValue = (data[IDX_BIT_RATE_1] << 16) | (data[IDX_BIT_RATE_2] << 8) | data[IDX_BIT_RATE_3]; setAACParam(tempValue, AACENC_BITRATE, kMinBitRate, kMaxBitRate); setAACParam(data[IDX_CHANNEL], AACENC_CHANNELMODE, 0, kChannelModesSize - 1, kChannelModes); setAACParam(data[IDX_IDENTIFIER], AACENC_TRANSMUX, 0, kIdentifiersSize - 1, kIdentifiers); setAACParam(data[IDX_BIT_RATE_MODE], AACENC_BITRATEMODE, 0, kBitRateModesSize - 1, kBitRateModes); setAACParam(data[IDX_GRANULE_LENGTH], AACENC_GRANULE_LENGTH, 0, kGranuleLengthsSize - 1, kGranuleLengths); setAACParam(data[IDX_CHANNELORDER], AACENC_CHANNELORDER, 0, kChannelOrderSize - 1, kChannelOrder); setAACParam(data[IDX_AFTERBURNER], AACENC_AFTERBURNER, 0, 1); setAACParam(data[IDX_BANDWIDTH], AACENC_BANDWIDTH, 0, 1); setAACParam(data[IDX_PEAK_BITRATE], AACENC_PEAK_BITRATE, kMinBitRate, kMinBitRate); setAACParam(data[IDX_HEADER_PERIOD], AACENC_HEADER_PERIOD, 0, UINT8_MAX); setAACParam(data[IDX_SIGNALING_MODE], AACENC_SIGNALING_MODE, 0, kSignalingModesSize - 1, kSignalingModes); setAACParam(data[IDX_TPSUBFRAMES], AACENC_TPSUBFRAMES, 0, UINT8_MAX); setAACParam(data[IDX_AUDIOMUXVER], AACENC_AUDIOMUXVER, 0, kAudioMuxVerSize - 1, kAudioMuxVer); setAACParam(data[IDX_PROTECTION], AACENC_PROTECTION, 0, 1); setAACParam(data[IDX_ANCILLARY_BITRATE], AACENC_ANCILLARY_BITRATE, 0, kMaxBitRate); setAACParam(data[IDX_METADATA_MODE], AACENC_METADATA_MODE, 0, 3); AACENC_InfoStruct encInfo; aacEncInfo(mEncoder, &encInfo); mInBufferIdx_1 = generateNumberInRangeFromData(data[IDX_IN_BUFFER_INDEX_1], 0, kMaxBuffers - 1); mInBufferIdx_2 = generateNumberInRangeFromData(data[IDX_IN_BUFFER_INDEX_2], 0, kMaxBuffers - 1); mInBufferIdx_3 = generateNumberInRangeFromData(data[IDX_IN_BUFFER_INDEX_3], 0, kMaxBuffers - 1); setupMetaData(data); // Not re-using the data which was used for configuration for encoding *dataPtr += IDX_LAST; *sizePtr -= IDX_LAST; return true; } static void deleteBuffers(uint8_t **buffers, size_t size) { for (size_t n = 0; n < size; ++n) { delete[] buffers[n]; } delete[] buffers; } void Codec::encodeFrames(const uint8_t *data, size_t size) { uint8_t *audioData = (uint8_t *)data; uint8_t *ancData = (uint8_t *)data; size_t audioSize = size; size_t ancSize = size; while ((audioSize > 0) && (ancSize > 0)) { AACENC_InArgs inargs; memset(&inargs, 0, sizeof(inargs)); inargs.numInSamples = audioSize / sizeof(int16_t); inargs.numAncBytes = ancSize; void *buffers[] = {(void *)audioData, (void *)ancData, &mMetaData}; INT bufferIds[] = {IN_AUDIO_DATA, IN_ANCILLRY_DATA, IN_METADATA_SETUP}; INT bufferSizes[] = {static_cast(audioSize), static_cast(ancSize), static_cast(sizeof(mMetaData))}; INT bufferElSizes[] = {sizeof(int16_t), sizeof(UCHAR), sizeof(AACENC_MetaData)}; void *inBuffer[kMaxBuffers] = {}; INT inBufferIds[kMaxBuffers] = {}; INT inBufferSize[kMaxBuffers] = {}; INT inBufferElSize[kMaxBuffers] = {}; for (int32_t buffer = 0; buffer < kMaxBuffers; ++buffer) { uint32_t Idxs[] = {mInBufferIdx_1, mInBufferIdx_2, mInBufferIdx_3}; inBuffer[buffer] = buffers[Idxs[buffer]]; inBufferIds[buffer] = bufferIds[Idxs[buffer]]; inBufferSize[buffer] = bufferSizes[Idxs[buffer]]; inBufferElSize[buffer] = bufferElSizes[Idxs[buffer]]; } AACENC_BufDesc inBufDesc; inBufDesc.numBufs = kMaxBuffers; inBufDesc.bufs = (void **)&inBuffer; inBufDesc.bufferIdentifiers = inBufferIds; inBufDesc.bufSizes = inBufferSize; inBufDesc.bufElSizes = inBufferElSize; uint8_t **outPtrRef = new uint8_t *[kMaxBuffers]; for (int32_t buffer = 0; buffer < kMaxBuffers; ++buffer) { outPtrRef[buffer] = new uint8_t[kMaxOutputBufferSize]; } void *outBuffer[kMaxBuffers]; INT outBufferIds[kMaxBuffers]; INT outBufferSize[kMaxBuffers]; INT outBufferElSize[kMaxBuffers]; for (int32_t buffer = 0; buffer < kMaxBuffers; ++buffer) { outBuffer[buffer] = outPtrRef[buffer]; outBufferIds[buffer] = OUT_BITSTREAM_DATA; outBufferSize[buffer] = (INT)kMaxOutputBufferSize; outBufferElSize[buffer] = sizeof(UCHAR); } AACENC_BufDesc outBufDesc; outBufDesc.numBufs = kMaxBuffers; outBufDesc.bufs = (void **)&outBuffer; outBufDesc.bufferIdentifiers = outBufferIds; outBufDesc.bufSizes = outBufferSize; outBufDesc.bufElSizes = outBufferElSize; AACENC_OutArgs outargs = {}; aacEncEncode(mEncoder, &inBufDesc, &outBufDesc, &inargs, &outargs); if (outargs.numOutBytes == 0) { if (audioSize > 0) { ++audioData; --audioSize; } if (ancSize > 0) { ++ancData; --ancSize; } } else { size_t audioConsumed = outargs.numInSamples * sizeof(int16_t); audioData += audioConsumed; audioSize -= audioConsumed; size_t ancConsumed = outargs.numAncBytes; ancData += ancConsumed; ancSize -= ancConsumed; } deleteBuffers(outPtrRef, kMaxBuffers); // break out of loop if only metadata was sent in all the input buffers // as sending it multiple times in a loop is redundant. if ((mInBufferIdx_1 == kMaxBuffers - 1) && (mInBufferIdx_2 == kMaxBuffers - 1) && (mInBufferIdx_3 == kMaxBuffers - 1)) { break; } } } void Codec::deInitEncoder() { aacEncClose(&mEncoder); } extern "C" int LLVMFuzzerTestOneInput(const uint8_t *data, size_t size) { if (size < IDX_LAST) { return 0; } Codec encoder; if (encoder.initEncoder(const_cast(&data), &size)) { encoder.encodeFrames(data, size); } return 0; } fdk-aac-2.0.2/libAACdec/000077500000000000000000000000001404225310700145265ustar00rootroot00000000000000fdk-aac-2.0.2/libAACdec/include/000077500000000000000000000000001404225310700161515ustar00rootroot00000000000000fdk-aac-2.0.2/libAACdec/include/aacdecoder_lib.h000066400000000000000000001534201404225310700212270ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Manuel Jander Description: *******************************************************************************/ #ifndef AACDECODER_LIB_H #define AACDECODER_LIB_H /** * \file aacdecoder_lib.h * \brief FDK AAC decoder library interface header file. * \page INTRO Introduction \section SCOPE Scope This document describes the high-level application interface and usage of the ISO/MPEG-2/4 AAC Decoder library developed by the Fraunhofer Institute for Integrated Circuits (IIS). Depending on the library configuration, decoding of AAC-LC (Low-Complexity), HE-AAC (High-Efficiency AAC v1 and v2), AAC-LD (Low-Delay) and AAC-ELD (Enhanced Low-Delay) is implemented. All references to SBR (Spectral Band Replication) are only applicable to HE-AAC and AAC-ELD configurations of the FDK library. All references to PS (Parametric Stereo) are only applicable to HE-AAC v2 decoder configuration of the library. \section DecoderBasics Decoder Basics This document can only give a rough overview about the ISO/MPEG-2, ISO/MPEG-4 AAC audio and MPEG-D USAC coding standards. To understand all details referenced in this document, you are encouraged to read the following documents. - ISO/IEC 13818-7 (MPEG-2 AAC) Standard, defines the syntax of MPEG-2 AAC audio bitstreams. - ISO/IEC 14496-3 (MPEG-4 AAC, subpart 1 and 4) Standard, defines the syntax of MPEG-4 AAC audio bitstreams. - ISO/IEC 23003-3 (MPEG-D USAC), defines MPEG-D USAC unified speech and audio codec. - Lutzky, Schuller, Gayer, Krämer, Wabnik, "A guideline to audio codec delay", 116th AES Convention, May 8, 2004 In short, MPEG Advanced Audio Coding is based on a time-to-frequency mapping of the signal. The signal is partitioned into overlapping time portions and transformed into frequency domain. The spectral components are then quantized and coded using a highly efficient coding scheme.\n Encoded MPEG-2 and MPEG-4 AAC audio bitstreams are composed of frames. Contrary to MPEG-1/2 Layer-3 (mp3), the length of individual frames is not restricted to a fixed number of bytes, but can take any length between 1 and 768 bytes. In addition to the above mentioned frequency domain coding mode, MPEG-D USAC also employs a time domain Algebraic Code-Excited Linear Prediction (ACELP) speech coder core. This operating mode is selected by the encoder in order to achieve the optimum audio quality for different content type. Several enhancements allow achieving higher quality at lower bit rates compared to MPEG-4 HE-AAC. \page LIBUSE Library Usage \section InterfaceDescritpion API Description All API header files are located in the folder /include of the release package. The contents of each file is described in detail in this document. All header files are provided for usage in specific C/C++ programs. The main AAC decoder library API functions are located in aacdecoder_lib.h header file. \section Calling_Sequence Calling Sequence The following sequence is necessary for proper decoding of ISO/MPEG-2/4 AAC, HE-AAC v2, or MPEG-D USAC bitstreams. In the following description, input stream read and output write function details are left out, since they may be implemented in a variety of configurations depending on the user's specific requirements. -# Call aacDecoder_Open() to open and retrieve a handle to a new AAC decoder instance. \code aacDecoderInfo = aacDecoder_Open(transportType, nrOfLayers); \endcode -# If out-of-band config data (Audio Specific Config (ASC) or Stream Mux Config (SMC)) is available, call aacDecoder_ConfigRaw() to pass this data to the decoder before beginning the decoding process. If this data is not available in advance, the decoder will configure itself while decoding, during the aacDecoder_DecodeFrame() function call. -# Begin decoding loop. \code do { \endcode -# Read data from bitstream file or stream buffer in to the driver program working memory (a client-supplied input buffer "inBuffer" in framework). This buffer will be used to load AAC bitstream data to the decoder. Only when all data in this buffer has been processed will the decoder signal an empty buffer. -# Call aacDecoder_Fill() to fill the decoder's internal bitstream input buffer with the client-supplied bitstream input buffer. Note, if the data loaded in to the internal buffer is not sufficient to decode a frame, aacDecoder_DecodeFrame() will return ::AAC_DEC_NOT_ENOUGH_BITS until a sufficient amount of data is loaded in to the internal buffer. For streaming formats (ADTS, LOAS), it is acceptable to load more than one frame to the decoder. However, for packed based formats, only one frame may be loaded to the decoder per aacDecoder_DecodeFrame() call. For least amount of communication delay, fill and decode should be performed on a frame by frame basis. \code ErrorStatus = aacDecoder_Fill(aacDecoderInfo, inBuffer, bytesRead, bytesValid); \endcode -# Call aacDecoder_DecodeFrame(). This function decodes one frame and writes decoded PCM audio data to a client-supplied buffer. It is the client's responsibility to allocate a buffer which is large enough to hold the decoded output data. \code ErrorStatus = aacDecoder_DecodeFrame(aacDecoderInfo, TimeData, OUT_BUF_SIZE, flags); \endcode If the bitstream configuration (number of channels, sample rate, frame size) is not known a priori, you may call aacDecoder_GetStreamInfo() to retrieve a structure that contains this information. You may use this data to initialize an audio output device. \code p_si = aacDecoder_GetStreamInfo(aacDecoderInfo); \endcode -# Repeat steps 5 to 7 until no data is available to decode any more, or in case of error. \code } while (bytesRead[0] > 0 || doFlush || doBsFlush || forceContinue); \endcode -# Call aacDecoder_Close() to de-allocate all AAC decoder and transport layer structures. \code aacDecoder_Close(aacDecoderInfo); \endcode \image latex decode.png "Decode calling sequence" width=11cm \image latex change_source.png "Change data source sequence" width=5cm \image latex conceal.png "Error concealment sequence" width=14cm \subsection Error_Concealment_Sequence Error Concealment Sequence There are different strategies to handle bit stream errors. Depending on the system properties the product designer might choose to take different actions in case a bit error occurs. In many cases the decoder might be able to do reasonable error concealment without the need of any additional actions from the system. But in some cases its not even possible to know how many decoded PCM output samples are required to fill the gap due to the data error, then the software surrounding the decoder must deal with the situation. The most simple way would be to just stop audio playback and resume once enough bit stream data and/or buffered output samples are available. More sophisticated designs might also be able to deal with sender/receiver clock drifts or data drop outs by using a closed loop control of FIFO fulness levels. The chosen strategy depends on the final product requirements. The error concealment sequence diagram illustrates the general execution paths for error handling. The macro IS_OUTPUT_VALID(err) can be used to identify if the audio output buffer contains valid audio either from error free bit stream data or successful error concealment. In case the result is false, the decoder output buffer does not contain meaningful audio samples and should not be passed to any output as it is. Most likely in case that a continuous audio output PCM stream is required, the output buffer must be filled with audio data from the calling framework. This might be e.g. an appropriate number of samples all zero. If error code ::AAC_DEC_TRANSPORT_SYNC_ERROR is returned by the decoder, under some particular conditions it is possible to estimate lost frames due to the bit stream error. In that case the bit stream is required to have a constant bitrate, and compatible transport type. Audio samples for the lost frames can be obtained by calling aacDecoder_DecodeFrame() with flag ::AACDEC_CONCEAL set n-times where n is the count of lost frames. Please note that the decoder has to have encountered valid configuration data at least once to be able to generate concealed data, because at the minimum the sampling rate, frame size and amount of audio channels needs to be known. If it is not possible to get an estimation of lost frames then a constant fullness of the audio output buffer can be achieved by implementing different FIFO control techniques e.g. just stop taking of samples from the buffer to avoid underflow or stop filling new data to the buffer to avoid overflow. But this techniques are out of scope of this document. For a detailed description of a specific error code please refer also to ::AAC_DECODER_ERROR. \section BufferSystem Buffer System There are three main buffers in an AAC decoder application. One external input buffer to hold bitstream data from file I/O or elsewhere, one decoder-internal input buffer, and one to hold the decoded output PCM sample data. In resource limited applications, the output buffer may be reused as an external input buffer prior to the subsequence aacDecoder_Fill() function call. To feed the data to the decoder-internal input buffer, use the function aacDecoder_Fill(). This function returns important information regarding the number of bytes in the external input buffer that have not yet been copied into the internal input buffer (variable bytesValid). Once the external buffer has been fully copied, it can be completely re-filled again. In case you wish to refill the buffer while there are unprocessed bytes (bytesValid is unequal 0), you should preserve the unconsumed data. However, we recommend to refill the buffer only when bytesValid returns 0. The bytesValid parameter is an input and output parameter to the FDK decoder. As an input, it signals how many valid bytes are available in the external buffer. After consumption of the external buffer using aacDecoder_Fill() function, the bytesValid parameter indicates if any of the bytes in the external buffer were not consumed. \image latex dec_buffer.png "Life cycle of the external input buffer" width=9cm \page OutputFormat Decoder audio output \section OutputFormatObtaining Obtaining channel mapping information The decoded audio output format is indicated by a set of variables of the CStreamInfo structure. While the struct members sampleRate, frameSize and numChannels might be self explanatory, pChannelType and pChannelIndices require some further explanation. These two arrays indicate the configuration of channel data within the output buffer. Both arrays have CStreamInfo::numChannels number of cells. Each cell of pChannelType indicates the channel type, which is described in the enum ::AUDIO_CHANNEL_TYPE (defined in FDK_audio.h). The cells of pChannelIndices indicate the sub index among the channels starting with 0 among channels of the same audio channel type. The indexing scheme is structured as defined in MPEG-2/4 Standards. Indices start from the front direction (a center channel if available, will always be index 0) and increment, starting with the left side, pairwise (e.g. L, R) and from front to back (Front L, Front R, Surround L, Surround R). For detailed explanation, please refer to ISO/IEC 13818-7:2005(E), chapter 8.5.3.2. In case a Program Config is included in the audio configuration, the channel mapping described within it will be adopted. The examples below explain these aspects in detail. \section OutputFormatChange Changing the audio output format For MPEG-4 audio the channel order can be changed at runtime through the parameter ::AAC_PCM_OUTPUT_CHANNEL_MAPPING. See the description of those parameters and the decoder library function aacDecoder_SetParam() for more detail. \section OutputFormatExample Channel mapping examples The following examples illustrate the location of individual audio samples in the audio buffer that is passed to aacDecoder_DecodeFrame() and the expected data in the CStreamInfo structure which can be obtained by calling aacDecoder_GetStreamInfo(). \subsection ExamplesStereo Stereo In case of ::AAC_PCM_OUTPUT_CHANNEL_MAPPING set to 1, a AAC-LC bit stream which has channelConfiguration = 2 in its audio specific config would lead to the following values in CStreamInfo: CStreamInfo::numChannels = 2 CStreamInfo::pChannelType = { ::ACT_FRONT, ::ACT_FRONT } CStreamInfo::pChannelIndices = { 0, 1 } The output buffer will be formatted as follows: \verbatim ... ... \endverbatim Where N equals to CStreamInfo::frameSize . \subsection ExamplesSurround Surround 5.1 In case of ::AAC_PCM_OUTPUT_CHANNEL_MAPPING set to 1, a AAC-LC bit stream which has channelConfiguration = 6 in its audio specific config, would lead to the following values in CStreamInfo: CStreamInfo::numChannels = 6 CStreamInfo::pChannelType = { ::ACT_FRONT, ::ACT_FRONT, ::ACT_FRONT, ::ACT_LFE, ::ACT_BACK, ::ACT_BACK } CStreamInfo::pChannelIndices = { 1, 2, 0, 0, 0, 1 } Since ::AAC_PCM_OUTPUT_CHANNEL_MAPPING is 1, WAV file channel ordering will be used. For a 5.1 channel scheme, thus the channels would be: front left, front right, center, LFE, surround left, surround right. Thus the third channel is the center channel, receiving the index 0. The other front channels are front left, front right being placed as first and second channels with indices 1 and 2 correspondingly. There is only one LFE, placed as the fourth channel and index 0. Finally both surround channels get the type definition ACT_BACK, and the indices 0 and 1. The output buffer will be formatted as follows: \verbatim

...
\endverbatim Where N equals to CStreamInfo::frameSize . \subsection ExamplesArib ARIB coding mode 2/1 In case of ::AAC_PCM_OUTPUT_CHANNEL_MAPPING set to 1, in case of a ARIB bit stream using coding mode 2/1 as described in ARIB STD-B32 Part 2 Version 2.1-E1, page 61, would lead to the following values in CStreamInfo: CStreamInfo::numChannels = 3 CStreamInfo::pChannelType = { ::ACT_FRONT, ::ACT_FRONT, ::ACT_BACK } CStreamInfo::pChannelIndices = { 0, 1, 0 } The audio channels will be placed as follows in the audio output buffer: \verbatim ... Where N equals to CStreamInfo::frameSize . \endverbatim */ #include "machine_type.h" #include "FDK_audio.h" #include "genericStds.h" #define AACDECODER_LIB_VL0 3 #define AACDECODER_LIB_VL1 2 #define AACDECODER_LIB_VL2 0 /** * \brief AAC decoder error codes. */ typedef enum { AAC_DEC_OK = 0x0000, /*!< No error occurred. Output buffer is valid and error free. */ AAC_DEC_OUT_OF_MEMORY = 0x0002, /*!< Heap returned NULL pointer. Output buffer is invalid. */ AAC_DEC_UNKNOWN = 0x0005, /*!< Error condition is of unknown reason, or from a another module. Output buffer is invalid. */ /* Synchronization errors. Output buffer is invalid. */ aac_dec_sync_error_start = 0x1000, AAC_DEC_TRANSPORT_SYNC_ERROR = 0x1001, /*!< The transport decoder had synchronization problems. Do not exit decoding. Just feed new bitstream data. */ AAC_DEC_NOT_ENOUGH_BITS = 0x1002, /*!< The input buffer ran out of bits. */ aac_dec_sync_error_end = 0x1FFF, /* Initialization errors. Output buffer is invalid. */ aac_dec_init_error_start = 0x2000, AAC_DEC_INVALID_HANDLE = 0x2001, /*!< The handle passed to the function call was invalid (NULL). */ AAC_DEC_UNSUPPORTED_AOT = 0x2002, /*!< The AOT found in the configuration is not supported. */ AAC_DEC_UNSUPPORTED_FORMAT = 0x2003, /*!< The bitstream format is not supported. */ AAC_DEC_UNSUPPORTED_ER_FORMAT = 0x2004, /*!< The error resilience tool format is not supported. */ AAC_DEC_UNSUPPORTED_EPCONFIG = 0x2005, /*!< The error protection format is not supported. */ AAC_DEC_UNSUPPORTED_MULTILAYER = 0x2006, /*!< More than one layer for AAC scalable is not supported. */ AAC_DEC_UNSUPPORTED_CHANNELCONFIG = 0x2007, /*!< The channel configuration (either number or arrangement) is not supported. */ AAC_DEC_UNSUPPORTED_SAMPLINGRATE = 0x2008, /*!< The sample rate specified in the configuration is not supported. */ AAC_DEC_INVALID_SBR_CONFIG = 0x2009, /*!< The SBR configuration is not supported. */ AAC_DEC_SET_PARAM_FAIL = 0x200A, /*!< The parameter could not be set. Either the value was out of range or the parameter does not exist. */ AAC_DEC_NEED_TO_RESTART = 0x200B, /*!< The decoder needs to be restarted, since the required configuration change cannot be performed. */ AAC_DEC_OUTPUT_BUFFER_TOO_SMALL = 0x200C, /*!< The provided output buffer is too small. */ aac_dec_init_error_end = 0x2FFF, /* Decode errors. Output buffer is valid but concealed. */ aac_dec_decode_error_start = 0x4000, AAC_DEC_TRANSPORT_ERROR = 0x4001, /*!< The transport decoder encountered an unexpected error. */ AAC_DEC_PARSE_ERROR = 0x4002, /*!< Error while parsing the bitstream. Most probably it is corrupted, or the system crashed. */ AAC_DEC_UNSUPPORTED_EXTENSION_PAYLOAD = 0x4003, /*!< Error while parsing the extension payload of the bitstream. The extension payload type found is not supported. */ AAC_DEC_DECODE_FRAME_ERROR = 0x4004, /*!< The parsed bitstream value is out of range. Most probably the bitstream is corrupt, or the system crashed. */ AAC_DEC_CRC_ERROR = 0x4005, /*!< The embedded CRC did not match. */ AAC_DEC_INVALID_CODE_BOOK = 0x4006, /*!< An invalid codebook was signaled. Most probably the bitstream is corrupt, or the system crashed. */ AAC_DEC_UNSUPPORTED_PREDICTION = 0x4007, /*!< Predictor found, but not supported in the AAC Low Complexity profile. Most probably the bitstream is corrupt, or has a wrong format. */ AAC_DEC_UNSUPPORTED_CCE = 0x4008, /*!< A CCE element was found which is not supported. Most probably the bitstream is corrupt, or has a wrong format. */ AAC_DEC_UNSUPPORTED_LFE = 0x4009, /*!< A LFE element was found which is not supported. Most probably the bitstream is corrupt, or has a wrong format. */ AAC_DEC_UNSUPPORTED_GAIN_CONTROL_DATA = 0x400A, /*!< Gain control data found but not supported. Most probably the bitstream is corrupt, or has a wrong format. */ AAC_DEC_UNSUPPORTED_SBA = 0x400B, /*!< SBA found, but currently not supported in the BSAC profile. */ AAC_DEC_TNS_READ_ERROR = 0x400C, /*!< Error while reading TNS data. Most probably the bitstream is corrupt or the system crashed. */ AAC_DEC_RVLC_ERROR = 0x400D, /*!< Error while decoding error resilient data. */ aac_dec_decode_error_end = 0x4FFF, /* Ancillary data errors. Output buffer is valid. */ aac_dec_anc_data_error_start = 0x8000, AAC_DEC_ANC_DATA_ERROR = 0x8001, /*!< Non severe error concerning the ancillary data handling. */ AAC_DEC_TOO_SMALL_ANC_BUFFER = 0x8002, /*!< The registered ancillary data buffer is too small to receive the parsed data. */ AAC_DEC_TOO_MANY_ANC_ELEMENTS = 0x8003, /*!< More than the allowed number of ancillary data elements should be written to buffer. */ aac_dec_anc_data_error_end = 0x8FFF } AAC_DECODER_ERROR; /** Macro to identify initialization errors. Output buffer is invalid. */ #define IS_INIT_ERROR(err) \ ((((err) >= aac_dec_init_error_start) && ((err) <= aac_dec_init_error_end)) \ ? 1 \ : 0) /** Macro to identify decode errors. Output buffer is valid but concealed. */ #define IS_DECODE_ERROR(err) \ ((((err) >= aac_dec_decode_error_start) && \ ((err) <= aac_dec_decode_error_end)) \ ? 1 \ : 0) /** * Macro to identify if the audio output buffer contains valid samples after * calling aacDecoder_DecodeFrame(). Output buffer is valid but can be * concealed. */ #define IS_OUTPUT_VALID(err) (((err) == AAC_DEC_OK) || IS_DECODE_ERROR(err)) /*! \enum AAC_MD_PROFILE * \brief The available metadata profiles which are mostly related to downmixing. The values define the arguments * for the use with parameter ::AAC_METADATA_PROFILE. */ typedef enum { AAC_MD_PROFILE_MPEG_STANDARD = 0, /*!< The standard profile creates a mixdown signal based on the advanced downmix metadata (from a DSE). The equations and default values are defined in ISO/IEC 14496:3 Ammendment 4. Any other (legacy) downmix metadata will be ignored. No other parameter will be modified. */ AAC_MD_PROFILE_MPEG_LEGACY = 1, /*!< This profile behaves identical to the standard profile if advanced downmix metadata (from a DSE) is available. If not, the matrix_mixdown information embedded in the program configuration element (PCE) will be applied. If neither is the case, the module creates a mixdown using the default coefficients as defined in ISO/IEC 14496:3 AMD 4. The profile can be used to support legacy digital TV (e.g. DVB) streams. */ AAC_MD_PROFILE_MPEG_LEGACY_PRIO = 2, /*!< Similar to the ::AAC_MD_PROFILE_MPEG_LEGACY profile but if both the advanced (ISO/IEC 14496:3 AMD 4) and the legacy (PCE) MPEG downmix metadata are available the latter will be applied. */ AAC_MD_PROFILE_ARIB_JAPAN = 3 /*!< Downmix creation as described in ABNT NBR 15602-2. But if advanced downmix metadata (ISO/IEC 14496:3 AMD 4) is available it will be preferred because of the higher resolutions. In addition the metadata expiry time will be set to the value defined in the ARIB standard (see ::AAC_METADATA_EXPIRY_TIME). */ } AAC_MD_PROFILE; /*! \enum AAC_DRC_DEFAULT_PRESENTATION_MODE_OPTIONS * \brief Options for handling of DRC parameters, if presentation mode is not indicated in bitstream */ typedef enum { AAC_DRC_PARAMETER_HANDLING_DISABLED = -1, /*!< DRC parameter handling disabled, all parameters are applied as requested. */ AAC_DRC_PARAMETER_HANDLING_ENABLED = 0, /*!< Apply changes to requested DRC parameters to prevent clipping. */ AAC_DRC_PRESENTATION_MODE_1_DEFAULT = 1, /*!< Use DRC presentation mode 1 as default (e.g. for Nordig) */ AAC_DRC_PRESENTATION_MODE_2_DEFAULT = 2 /*!< Use DRC presentation mode 2 as default (e.g. for DTG DBook) */ } AAC_DRC_DEFAULT_PRESENTATION_MODE_OPTIONS; /** * \brief AAC decoder setting parameters */ typedef enum { AAC_PCM_DUAL_CHANNEL_OUTPUT_MODE = 0x0002, /*!< Defines how the decoder processes two channel signals: \n 0: Leave both signals as they are (default). \n 1: Create a dual mono output signal from channel 1. \n 2: Create a dual mono output signal from channel 2. \n 3: Create a dual mono output signal by mixing both channels (L' = R' = 0.5*Ch1 + 0.5*Ch2). */ AAC_PCM_OUTPUT_CHANNEL_MAPPING = 0x0003, /*!< Output buffer channel ordering. 0: MPEG PCE style order, 1: WAV file channel order (default). */ AAC_PCM_LIMITER_ENABLE = 0x0004, /*!< Enable signal level limiting. \n -1: Auto-config. Enable limiter for all non-lowdelay configurations by default. \n 0: Disable limiter in general. \n 1: Enable limiter always. It is recommended to call the decoder with a AACDEC_CLRHIST flag to reset all states when the limiter switch is changed explicitly. */ AAC_PCM_LIMITER_ATTACK_TIME = 0x0005, /*!< Signal level limiting attack time in ms. Default configuration is 15 ms. Adjustable range from 1 ms to 15 ms. */ AAC_PCM_LIMITER_RELEAS_TIME = 0x0006, /*!< Signal level limiting release time in ms. Default configuration is 50 ms. Adjustable time must be larger than 0 ms. */ AAC_PCM_MIN_OUTPUT_CHANNELS = 0x0011, /*!< Minimum number of PCM output channels. If higher than the number of encoded audio channels, a simple channel extension is applied (see note 4 for exceptions). \n -1, 0: Disable channel extension feature. The decoder output contains the same number of channels as the encoded bitstream. \n 1: This value is currently needed only together with the mix-down feature. See ::AAC_PCM_MAX_OUTPUT_CHANNELS and note 2 below. \n 2: Encoded mono signals will be duplicated to achieve a 2/0/0.0 channel output configuration. \n 6: The decoder tries to reorder encoded signals with less than six channels to achieve a 3/0/2.1 channel output signal. Missing channels will be filled with a zero signal. If reordering is not possible the empty channels will simply be appended. Only available if instance is configured to support multichannel output. \n 8: The decoder tries to reorder encoded signals with less than eight channels to achieve a 3/0/4.1 channel output signal. Missing channels will be filled with a zero signal. If reordering is not possible the empty channels will simply be appended. Only available if instance is configured to support multichannel output.\n NOTE: \n 1. The channel signaling (CStreamInfo::pChannelType and CStreamInfo::pChannelIndices) will not be modified. Added empty channels will be signaled with channel type AUDIO_CHANNEL_TYPE::ACT_NONE. \n 2. If the parameter value is greater than that of ::AAC_PCM_MAX_OUTPUT_CHANNELS both will be set to the same value. \n 3. This parameter will be ignored if the number of encoded audio channels is greater than 8. */ AAC_PCM_MAX_OUTPUT_CHANNELS = 0x0012, /*!< Maximum number of PCM output channels. If lower than the number of encoded audio channels, downmixing is applied accordingly (see note 5 for exceptions). If dedicated metadata is available in the stream it will be used to achieve better mixing results. \n -1, 0: Disable downmixing feature. The decoder output contains the same number of channels as the encoded bitstream. \n 1: All encoded audio configurations with more than one channel will be mixed down to one mono output signal. \n 2: The decoder performs a stereo mix-down if the number encoded audio channels is greater than two. \n 6: If the number of encoded audio channels is greater than six the decoder performs a mix-down to meet the target output configuration of 3/0/2.1 channels. Only available if instance is configured to support multichannel output. \n 8: This value is currently needed only together with the channel extension feature. See ::AAC_PCM_MIN_OUTPUT_CHANNELS and note 2 below. Only available if instance is configured to support multichannel output. \n NOTE: \n 1. Down-mixing of any seven or eight channel configuration not defined in ISO/IEC 14496-3 PDAM 4 is not supported by this software version. \n 2. If the parameter value is greater than zero but smaller than ::AAC_PCM_MIN_OUTPUT_CHANNELS both will be set to same value. \n 3. This parameter will be ignored if the number of encoded audio channels is greater than 8. */ AAC_METADATA_PROFILE = 0x0020, /*!< See ::AAC_MD_PROFILE for all available values. */ AAC_METADATA_EXPIRY_TIME = 0x0021, /*!< Defines the time in ms after which all the bitstream associated meta-data (DRC, downmix coefficients, ...) will be reset to default if no update has been received. Negative values disable the feature. */ AAC_CONCEAL_METHOD = 0x0100, /*!< Error concealment: Processing method. \n 0: Spectral muting. \n 1: Noise substitution (see ::CONCEAL_NOISE). \n 2: Energy interpolation (adds additional signal delay of one frame, see ::CONCEAL_INTER. only some AOTs are supported). \n */ AAC_DRC_BOOST_FACTOR = 0x0200, /*!< MPEG-4 / MPEG-D Dynamic Range Control (DRC): Scaling factor for boosting gain values. Defines how the boosting DRC factors (conveyed in the bitstream) will be applied to the decoded signal. The valid values range from 0 (don't apply boost factors) to 127 (fully apply boost factors). Default value is 0 for MPEG-4 DRC and 127 for MPEG-D DRC. */ AAC_DRC_ATTENUATION_FACTOR = 0x0201, /*!< MPEG-4 / MPEG-D DRC: Scaling factor for attenuating gain values. Same as ::AAC_DRC_BOOST_FACTOR but for attenuating DRC factors. */ AAC_DRC_REFERENCE_LEVEL = 0x0202, /*!< MPEG-4 / MPEG-D DRC: Target reference level / decoder target loudness.\n Defines the level below full-scale (quantized in steps of 0.25dB) to which the output audio signal will be normalized to by the DRC module.\n The parameter controls loudness normalization for both MPEG-4 DRC and MPEG-D DRC. The valid values range from 40 (-10 dBFS) to 127 (-31.75 dBFS).\n Example values:\n 124 (-31 dBFS) for audio/video receivers (AVR) or other devices allowing audio playback with high dynamic range,\n 96 (-24 dBFS) for TV sets or equivalent devices (default),\n 64 (-16 dBFS) for mobile devices where the dynamic range of audio playback is restricted.\n Any value smaller than 0 switches off loudness normalization and MPEG-4 DRC. */ AAC_DRC_HEAVY_COMPRESSION = 0x0203, /*!< MPEG-4 DRC: En-/Disable DVB specific heavy compression (aka RF mode). If set to 1, the decoder will apply the compression values from the DVB specific ancillary data field. At the same time the MPEG-4 Dynamic Range Control tool will be disabled. By default, heavy compression is disabled. */ AAC_DRC_DEFAULT_PRESENTATION_MODE = 0x0204, /*!< MPEG-4 DRC: Default presentation mode (DRC parameter handling). \n Defines the handling of the DRC parameters boost factor, attenuation factor and heavy compression, if no presentation mode is indicated in the bitstream.\n For options, see ::AAC_DRC_DEFAULT_PRESENTATION_MODE_OPTIONS.\n Default: ::AAC_DRC_PARAMETER_HANDLING_DISABLED */ AAC_DRC_ENC_TARGET_LEVEL = 0x0205, /*!< MPEG-4 DRC: Encoder target level for light (i.e. not heavy) compression.\n If known, this declares the target reference level that was assumed at the encoder for calculation of limiting gains. The valid values range from 0 (full-scale) to 127 (31.75 dB below full-scale). This parameter is used only with ::AAC_DRC_PARAMETER_HANDLING_ENABLED and ignored otherwise.\n Default: 127 (worst-case assumption).\n */ AAC_UNIDRC_SET_EFFECT = 0x0206, /*!< MPEG-D DRC: Request a DRC effect type for selection of a DRC set.\n Supported indices are:\n -1: DRC off. Completely disables MPEG-D DRC.\n 0: None (default). Disables MPEG-D DRC, but automatically enables DRC if necessary to prevent clipping.\n 1: Late night\n 2: Noisy environment\n 3: Limited playback range\n 4: Low playback level\n 5: Dialog enhancement\n 6: General compression. Used for generally enabling MPEG-D DRC without particular request.\n */ AAC_UNIDRC_ALBUM_MODE = 0x0207, /*!< MPEG-D DRC: Enable album mode. 0: Disabled (default), 1: Enabled.\n Disabled album mode leads to application of gain sequences for fading in and out, if provided in the bitstream.\n Enabled album mode makes use of dedicated album loudness information, if provided in the bitstream.\n */ AAC_QMF_LOWPOWER = 0x0300, /*!< Quadrature Mirror Filter (QMF) Bank processing mode. \n -1: Use internal default. \n 0: Use complex QMF data mode. \n 1: Use real (low power) QMF data mode. \n */ AAC_TPDEC_CLEAR_BUFFER = 0x0603 /*!< Clear internal bit stream buffer of transport layers. The decoder will start decoding at new data passed after this event and any previous data is discarded. */ } AACDEC_PARAM; /** * \brief This structure gives information about the currently decoded audio * data. All fields are read-only. */ typedef struct { /* These five members are the only really relevant ones for the user. */ INT sampleRate; /*!< The sample rate in Hz of the decoded PCM audio signal. */ INT frameSize; /*!< The frame size of the decoded PCM audio signal. \n Typically this is: \n 1024 or 960 for AAC-LC \n 2048 or 1920 for HE-AAC (v2) \n 512 or 480 for AAC-LD and AAC-ELD \n 768, 1024, 2048 or 4096 for USAC */ INT numChannels; /*!< The number of output audio channels before the rendering module, i.e. the original channel configuration. */ AUDIO_CHANNEL_TYPE *pChannelType; /*!< Audio channel type of each output audio channel. */ UCHAR *pChannelIndices; /*!< Audio channel index for each output audio channel. See ISO/IEC 13818-7:2005(E), 8.5.3.2 Explicit channel mapping using a program_config_element() */ /* Decoder internal members. */ INT aacSampleRate; /*!< Sampling rate in Hz without SBR (from configuration info) divided by a (ELD) downscale factor if present. */ INT profile; /*!< MPEG-2 profile (from file header) (-1: not applicable (e. g. MPEG-4)). */ AUDIO_OBJECT_TYPE aot; /*!< Audio Object Type (from ASC): is set to the appropriate value for MPEG-2 bitstreams (e. g. 2 for AAC-LC). */ INT channelConfig; /*!< Channel configuration (0: PCE defined, 1: mono, 2: stereo, ... */ INT bitRate; /*!< Instantaneous bit rate. */ INT aacSamplesPerFrame; /*!< Samples per frame for the AAC core (from ASC) divided by a (ELD) downscale factor if present. \n Typically this is (with a downscale factor of 1): \n 1024 or 960 for AAC-LC \n 512 or 480 for AAC-LD and AAC-ELD */ INT aacNumChannels; /*!< The number of audio channels after AAC core processing (before PS or MPS processing). CAUTION: This are not the final number of output channels! */ AUDIO_OBJECT_TYPE extAot; /*!< Extension Audio Object Type (from ASC) */ INT extSamplingRate; /*!< Extension sampling rate in Hz (from ASC) divided by a (ELD) downscale factor if present. */ UINT outputDelay; /*!< The number of samples the output is additionally delayed by.the decoder. */ UINT flags; /*!< Copy of internal flags. Only to be written by the decoder, and only to be read externally. */ SCHAR epConfig; /*!< epConfig level (from ASC): only level 0 supported, -1 means no ER (e. g. AOT=2, MPEG-2 AAC, etc.) */ /* Statistics */ INT numLostAccessUnits; /*!< This integer will reflect the estimated amount of lost access units in case aacDecoder_DecodeFrame() returns AAC_DEC_TRANSPORT_SYNC_ERROR. It will be < 0 if the estimation failed. */ INT64 numTotalBytes; /*!< This is the number of total bytes that have passed through the decoder. */ INT64 numBadBytes; /*!< This is the number of total bytes that were considered with errors from numTotalBytes. */ INT64 numTotalAccessUnits; /*!< This is the number of total access units that have passed through the decoder. */ INT64 numBadAccessUnits; /*!< This is the number of total access units that were considered with errors from numTotalBytes. */ /* Metadata */ SCHAR drcProgRefLev; /*!< DRC program reference level. Defines the reference level below full-scale. It is quantized in steps of 0.25dB. The valid values range from 0 (0 dBFS) to 127 (-31.75 dBFS). It is used to reflect the average loudness of the audio in LKFS according to ITU-R BS 1770. If no level has been found in the bitstream the value is -1. */ SCHAR drcPresMode; /*!< DRC presentation mode. According to ETSI TS 101 154, this field indicates whether light (MPEG-4 Dynamic Range Control tool) or heavy compression (DVB heavy compression) dynamic range control shall take priority on the outputs. For details, see ETSI TS 101 154, table C.33. Possible values are: \n -1: No corresponding metadata found in the bitstream \n 0: DRC presentation mode not indicated \n 1: DRC presentation mode 1 \n 2: DRC presentation mode 2 \n 3: Reserved */ INT outputLoudness; /*!< Audio output loudness in steps of -0.25 dB. Range: 0 (0 dBFS) to 231 (-57.75 dBFS).\n A value of -1 indicates that no loudness metadata is present.\n If loudness normalization is active, the value corresponds to the target loudness value set with ::AAC_DRC_REFERENCE_LEVEL.\n If loudness normalization is not active, the output loudness value corresponds to the loudness metadata given in the bitstream.\n Loudness metadata can originate from MPEG-4 DRC or MPEG-D DRC. */ } CStreamInfo; typedef struct AAC_DECODER_INSTANCE *HANDLE_AACDECODER; /*!< Pointer to a AAC decoder instance. */ #ifdef __cplusplus extern "C" { #endif /** * \brief Initialize ancillary data buffer. * * \param self AAC decoder handle. * \param buffer Pointer to (external) ancillary data buffer. * \param size Size of the buffer pointed to by buffer. * \return Error code. */ LINKSPEC_H AAC_DECODER_ERROR aacDecoder_AncDataInit(HANDLE_AACDECODER self, UCHAR *buffer, int size); /** * \brief Get one ancillary data element. * * \param self AAC decoder handle. * \param index Index of the ancillary data element to get. * \param ptr Pointer to a buffer receiving a pointer to the requested * ancillary data element. * \param size Pointer to a buffer receiving the length of the requested * ancillary data element. * \return Error code. */ LINKSPEC_H AAC_DECODER_ERROR aacDecoder_AncDataGet(HANDLE_AACDECODER self, int index, UCHAR **ptr, int *size); /** * \brief Set one single decoder parameter. * * \param self AAC decoder handle. * \param param Parameter to be set. * \param value Parameter value. * \return Error code. */ LINKSPEC_H AAC_DECODER_ERROR aacDecoder_SetParam(const HANDLE_AACDECODER self, const AACDEC_PARAM param, const INT value); /** * \brief Get free bytes inside decoder internal buffer. * \param self Handle of AAC decoder instance. * \param pFreeBytes Pointer to variable receiving amount of free bytes inside * decoder internal buffer. * \return Error code. */ LINKSPEC_H AAC_DECODER_ERROR aacDecoder_GetFreeBytes(const HANDLE_AACDECODER self, UINT *pFreeBytes); /** * \brief Open an AAC decoder instance. * \param transportFmt The transport type to be used. * \param nrOfLayers Number of transport layers. * \return AAC decoder handle. */ LINKSPEC_H HANDLE_AACDECODER aacDecoder_Open(TRANSPORT_TYPE transportFmt, UINT nrOfLayers); /** * \brief Explicitly configure the decoder by passing a raw AudioSpecificConfig * (ASC) or a StreamMuxConfig (SMC), contained in a binary buffer. This is * required for MPEG-4 and Raw Packets file format bitstreams as well as for * LATM bitstreams with no in-band SMC. If the transport format is LATM with or * without LOAS, configuration is assumed to be an SMC, for all other file * formats an ASC. * * \param self AAC decoder handle. * \param conf Pointer to an unsigned char buffer containing the binary * configuration buffer (either ASC or SMC). * \param length Length of the configuration buffer in bytes. * \return Error code. */ LINKSPEC_H AAC_DECODER_ERROR aacDecoder_ConfigRaw(HANDLE_AACDECODER self, UCHAR *conf[], const UINT length[]); /** * \brief Submit raw ISO base media file format boxes to decoder for parsing * (only some box types are recognized). * * \param self AAC decoder handle. * \param buffer Pointer to an unsigned char buffer containing the binary box * data (including size and type, can be a sequence of multiple boxes). * \param length Length of the data in bytes. * \return Error code. */ LINKSPEC_H AAC_DECODER_ERROR aacDecoder_RawISOBMFFData(HANDLE_AACDECODER self, UCHAR *buffer, UINT length); /** * \brief Fill AAC decoder's internal input buffer with bitstream data from the * external input buffer. The function only copies such data as long as the * decoder-internal input buffer is not full. So it grabs whatever it can from * pBuffer and returns information (bytesValid) so that at a subsequent call of * %aacDecoder_Fill(), the right position in pBuffer can be determined to grab * the next data. * * \param self AAC decoder handle. * \param pBuffer Pointer to external input buffer. * \param bufferSize Size of external input buffer. This argument is required * because decoder-internally we need the information to calculate the offset to * pBuffer, where the next available data is, which is then * fed into the decoder-internal buffer (as much as * possible). Our example framework implementation fills the * buffer at pBuffer again, once it contains no available valid bytes anymore * (meaning bytesValid equal 0). * \param bytesValid Number of bitstream bytes in the external bitstream buffer * that have not yet been copied into the decoder's internal bitstream buffer by * calling this function. The value is updated according to * the amount of newly copied bytes. * \return Error code. */ LINKSPEC_H AAC_DECODER_ERROR aacDecoder_Fill(HANDLE_AACDECODER self, UCHAR *pBuffer[], const UINT bufferSize[], UINT *bytesValid); /** Flag for aacDecoder_DecodeFrame(): Trigger the built-in error concealment * module to generate a substitute signal for one lost frame. New input data * will not be considered. */ #define AACDEC_CONCEAL 1 /** Flag for aacDecoder_DecodeFrame(): Flush all filterbanks to get all delayed * audio without having new input data. Thus new input data will not be * considered. */ #define AACDEC_FLUSH 2 /** Flag for aacDecoder_DecodeFrame(): Signal an input bit stream data * discontinuity. Resync any internals as necessary. */ #define AACDEC_INTR 4 /** Flag for aacDecoder_DecodeFrame(): Clear all signal delay lines and history * buffers. CAUTION: This can cause discontinuities in the output signal. */ #define AACDEC_CLRHIST 8 /** * \brief Decode one audio frame * * \param self AAC decoder handle. * \param pTimeData Pointer to external output buffer where the decoded PCM * samples will be stored into. * \param timeDataSize Size of external output buffer. * \param flags Bit field with flags for the decoder: \n * (flags & AACDEC_CONCEAL) == 1: Do concealment. \n * (flags & AACDEC_FLUSH) == 2: Discard input data. Flush * filter banks (output delayed audio). \n (flags & AACDEC_INTR) == 4: Input * data is discontinuous. Resynchronize any internals as * necessary. \n (flags & AACDEC_CLRHIST) == 8: Clear all signal delay lines and * history buffers. * \return Error code. */ LINKSPEC_H AAC_DECODER_ERROR aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData, const INT timeDataSize, const UINT flags); /** * \brief De-allocate all resources of an AAC decoder instance. * * \param self AAC decoder handle. * \return void. */ LINKSPEC_H void aacDecoder_Close(HANDLE_AACDECODER self); /** * \brief Get CStreamInfo handle from decoder. * * \param self AAC decoder handle. * \return Reference to requested CStreamInfo. */ LINKSPEC_H CStreamInfo *aacDecoder_GetStreamInfo(HANDLE_AACDECODER self); /** * \brief Get decoder library info. * * \param info Pointer to an allocated LIB_INFO structure. * \return 0 on success. */ LINKSPEC_H INT aacDecoder_GetLibInfo(LIB_INFO *info); #ifdef __cplusplus } #endif #endif /* AACDECODER_LIB_H */ fdk-aac-2.0.2/libAACdec/src/000077500000000000000000000000001404225310700153155ustar00rootroot00000000000000fdk-aac-2.0.2/libAACdec/src/FDK_delay.cpp000066400000000000000000000157301404225310700176110ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Matthias Hildenbrand Description: *******************************************************************************/ #include "FDK_delay.h" #include "genericStds.h" #define MAX_FRAME_LENGTH (1024) INT FDK_Delay_Create(FDK_SignalDelay* data, const USHORT delay, const UCHAR num_channels) { FDK_ASSERT(data != NULL); FDK_ASSERT(num_channels > 0); if (delay > 0) { data->delay_line = (PCM_DEC*)FDKcalloc(num_channels * delay, sizeof(PCM_DEC)); if (data->delay_line == NULL) { return -1; } } else { data->delay_line = NULL; } data->num_channels = num_channels; data->delay = delay; return 0; } void FDK_Delay_Apply(FDK_SignalDelay* data, PCM_DEC* time_buffer, const UINT frame_length, const UCHAR channel) { FDK_ASSERT(data != NULL); if (data->delay > 0) { C_ALLOC_SCRATCH_START(tmp, PCM_DEC, MAX_FRAME_LENGTH) FDK_ASSERT(frame_length <= MAX_FRAME_LENGTH); FDK_ASSERT(channel < data->num_channels); FDK_ASSERT(time_buffer != NULL); if (frame_length >= data->delay) { FDKmemcpy(tmp, &time_buffer[frame_length - data->delay], data->delay * sizeof(PCM_DEC)); FDKmemmove(&time_buffer[data->delay], &time_buffer[0], (frame_length - data->delay) * sizeof(PCM_DEC)); FDKmemcpy(&time_buffer[0], &data->delay_line[channel * data->delay], data->delay * sizeof(PCM_DEC)); FDKmemcpy(&data->delay_line[channel * data->delay], tmp, data->delay * sizeof(PCM_DEC)); } else { FDKmemcpy(tmp, &time_buffer[0], frame_length * sizeof(PCM_DEC)); FDKmemcpy(&time_buffer[0], &data->delay_line[channel * data->delay], frame_length * sizeof(PCM_DEC)); FDKmemcpy(&data->delay_line[channel * data->delay], &data->delay_line[channel * data->delay + frame_length], (data->delay - frame_length) * sizeof(PCM_DEC)); FDKmemcpy(&data->delay_line[channel * data->delay + (data->delay - frame_length)], tmp, frame_length * sizeof(PCM_DEC)); } C_ALLOC_SCRATCH_END(tmp, PCM_DEC, MAX_FRAME_LENGTH) } return; } void FDK_Delay_Destroy(FDK_SignalDelay* data) { if (data->delay_line != NULL) { FDKfree(data->delay_line); } data->delay_line = NULL; data->delay = 0; data->num_channels = 0; return; } fdk-aac-2.0.2/libAACdec/src/FDK_delay.h000066400000000000000000000142341404225310700172540ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Matthias Hildenbrand Description: *******************************************************************************/ #ifndef FDK_DELAY_H #define FDK_DELAY_H #include "aac_rom.h" /** * Structure representing one delay element for multiple channels. */ typedef struct { PCM_DEC* delay_line; /*!< Pointer which stores allocated delay line. */ USHORT delay; /*!< Delay required in samples (per channel). */ UCHAR num_channels; /*!< Number of channels to delay. */ } FDK_SignalDelay; /** * \brief Create delay element for multiple channels with fixed delay value. * * \param data Pointer delay element structure. * \param delay Required delay value in samples per channel. * \param num_channels Required number of channels. * * \return -1 on out of memory, else 0 */ INT FDK_Delay_Create(FDK_SignalDelay* data, const USHORT delay, const UCHAR num_channels); /** * \brief Apply delay to one channel (non-interleaved storage assumed). * * \param data Pointer delay element structure. * \param time_buffer Pointer to signal to delay. * \param frame_length Frame length of input/output signal (needs to be >= * delay). * \param channel Index of current channel (0 <= channel < num_channels). * * \return void */ void FDK_Delay_Apply(FDK_SignalDelay* data, PCM_DEC* time_buffer, const UINT frame_length, const UCHAR channel); /** * \brief Destroy delay element. * * \param data Pointer delay element structure. * * \return void */ void FDK_Delay_Destroy(FDK_SignalDelay* data); #endif /* #ifndef FDK_DELAY_H */ fdk-aac-2.0.2/libAACdec/src/aac_ram.cpp000066400000000000000000000164251404225310700174140ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Josef Hoepfl Description: *******************************************************************************/ #include "aac_ram.h" #include "aac_rom.h" #define WORKBUFFER1_TAG 0 #define WORKBUFFER2_TAG 1 #define WORKBUFFER5_TAG 6 #define WORKBUFFER6_TAG 7 /*! The structure AAC_DECODER_INSTANCE is the top level structure holding all decoder configurations, handles and structs. */ C_ALLOC_MEM(AacDecoder, struct AAC_DECODER_INSTANCE, 1) /*! \name StaticAacData Static memory areas, must not be overwritten in other sections of the decoder */ /* @{ */ /*! The structure CAacDecoderStaticChannelInfo contains the static sideinfo which is needed for the decoding of one aac channel.
Dimension: #AacDecoderChannels */ C_ALLOC_MEM2(AacDecoderStaticChannelInfo, CAacDecoderStaticChannelInfo, 1, (8)) /*! The structure CAacDecoderChannelInfo contains the dynamic sideinfo which is needed for the decoding of one aac channel.
Dimension: #AacDecoderChannels */ C_AALLOC_MEM2(AacDecoderChannelInfo, CAacDecoderChannelInfo, 1, (8)) /*! Overlap buffer */ C_AALLOC_MEM2(OverlapBuffer, FIXP_DBL, OverlapBufferSize, (8)) C_ALLOC_MEM(DrcInfo, CDrcInfo, 1) /*! The structure CpePersistentData holds the persistent data shared by both channels of a CPE.
It needs to be allocated for each CPE.
Dimension: 1 */ C_ALLOC_MEM(CpePersistentData, CpePersistentData, 1) /*! The structure CCplxPredictionData holds data for complex stereo prediction.
Dimension: 1 */ C_ALLOC_MEM(CplxPredictionData, CCplxPredictionData, 1) /*! The buffer holds time samples for the crossfade in case of an USAC DASH IPF config change Dimension: (8) */ C_ALLOC_MEM2(TimeDataFlush, INT_PCM, TIME_DATA_FLUSH_SIZE, (8)) /* @} */ /*! \name DynamicAacData Dynamic memory areas, might be reused in other algorithm sections, e.g. the sbr decoder */ /* Take into consideration to make use of the WorkBufferCore[3/4] for decoder * configurations with more than 2 channels */ C_ALLOC_MEM_OVERLAY(WorkBufferCore2, FIXP_DBL, ((8) * 1024), SECT_DATA_L2, WORKBUFFER2_TAG) C_ALLOC_MEM_OVERLAY(WorkBufferCore6, SCHAR, fMax((INT)(sizeof(FIXP_DBL) * WB_SECTION_SIZE), (INT)sizeof(CAacDecoderCommonData)), SECT_DATA_L2, WORKBUFFER6_TAG) C_ALLOC_MEM_OVERLAY(WorkBufferCore1, CWorkBufferCore1, 1, SECT_DATA_L1, WORKBUFFER1_TAG) /* double buffer size needed for de-/interleaving */ C_ALLOC_MEM_OVERLAY(WorkBufferCore5, PCM_DEC, (8) * (1024 * 4) * 2, SECT_DATA_EXTERN, WORKBUFFER5_TAG) fdk-aac-2.0.2/libAACdec/src/aac_ram.h000066400000000000000000000132061404225310700170530ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Josef Hoepfl Description: *******************************************************************************/ #ifndef AAC_RAM_H #define AAC_RAM_H #include "common_fix.h" #include "aacdecoder.h" #include "channel.h" #include "ac_arith_coder.h" #include "aacdec_hcr_types.h" #include "aacdec_hcr.h" /* End of formal fix.h */ #define MAX_SYNCHS 10 #define SAMPL_FREQS 12 H_ALLOC_MEM(AacDecoder, AAC_DECODER_INSTANCE) H_ALLOC_MEM(DrcInfo, CDrcInfo) H_ALLOC_MEM(AacDecoderStaticChannelInfo, CAacDecoderStaticChannelInfo) H_ALLOC_MEM(AacDecoderChannelInfo, CAacDecoderChannelInfo) H_ALLOC_MEM(OverlapBuffer, FIXP_DBL) H_ALLOC_MEM(CpePersistentData, CpePersistentData) H_ALLOC_MEM(CplxPredictionData, CCplxPredictionData) H_ALLOC_MEM(SpectralCoeffs, FIXP_DBL) H_ALLOC_MEM(SpecScale, SHORT) H_ALLOC_MEM(TimeDataFlush, INT_PCM) H_ALLOC_MEM_OVERLAY(WorkBufferCore1, CWorkBufferCore1) H_ALLOC_MEM_OVERLAY(WorkBufferCore2, FIXP_DBL) H_ALLOC_MEM_OVERLAY(WorkBufferCore5, PCM_DEC) H_ALLOC_MEM_OVERLAY(WorkBufferCore6, SCHAR) #endif /* #ifndef AAC_RAM_H */ fdk-aac-2.0.2/libAACdec/src/aac_rom.cpp000066400000000000000000006444361404225310700174430ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Josef Hoepfl, Tobias Chalupka Description: Definition of constant tables *******************************************************************************/ #include "aac_rom.h" /* Prescale InverseQuantTable by 4 to save redundant shifts in invers quantization */ #define SCL_TAB(a) (a >> 4) const FIXP_DBL InverseQuantTable[INV_QUANT_TABLESIZE + 1] = { SCL_TAB(0x32CBFD40), SCL_TAB(0x330FC340), SCL_TAB(0x33539FC0), SCL_TAB(0x33979280), SCL_TAB(0x33DB9BC0), SCL_TAB(0x341FBB80), SCL_TAB(0x3463F180), SCL_TAB(0x34A83DC0), SCL_TAB(0x34ECA000), SCL_TAB(0x35311880), SCL_TAB(0x3575A700), SCL_TAB(0x35BA4B80), SCL_TAB(0x35FF0600), SCL_TAB(0x3643D680), SCL_TAB(0x3688BCC0), SCL_TAB(0x36CDB880), SCL_TAB(0x3712CA40), SCL_TAB(0x3757F1C0), SCL_TAB(0x379D2F00), SCL_TAB(0x37E28180), SCL_TAB(0x3827E9C0), SCL_TAB(0x386D6740), SCL_TAB(0x38B2FA40), SCL_TAB(0x38F8A2C0), SCL_TAB(0x393E6080), SCL_TAB(0x39843380), SCL_TAB(0x39CA1BC0), SCL_TAB(0x3A101940), SCL_TAB(0x3A562BC0), SCL_TAB(0x3A9C5340), SCL_TAB(0x3AE28FC0), SCL_TAB(0x3B28E180), SCL_TAB(0x3B6F4800), SCL_TAB(0x3BB5C340), SCL_TAB(0x3BFC5380), SCL_TAB(0x3C42F880), SCL_TAB(0x3C89B200), SCL_TAB(0x3CD08080), SCL_TAB(0x3D176340), SCL_TAB(0x3D5E5B00), SCL_TAB(0x3DA56700), SCL_TAB(0x3DEC87C0), SCL_TAB(0x3E33BCC0), SCL_TAB(0x3E7B0640), SCL_TAB(0x3EC26400), SCL_TAB(0x3F09D640), SCL_TAB(0x3F515C80), SCL_TAB(0x3F98F740), SCL_TAB(0x3FE0A600), SCL_TAB(0x40286900), SCL_TAB(0x40704000), SCL_TAB(0x40B82B00), SCL_TAB(0x41002A00), SCL_TAB(0x41483D00), SCL_TAB(0x41906400), SCL_TAB(0x41D89F00), SCL_TAB(0x4220ED80), SCL_TAB(0x42695000), SCL_TAB(0x42B1C600), SCL_TAB(0x42FA5000), SCL_TAB(0x4342ED80), SCL_TAB(0x438B9E80), SCL_TAB(0x43D46380), SCL_TAB(0x441D3B80), SCL_TAB(0x44662780), SCL_TAB(0x44AF2680), SCL_TAB(0x44F83900), SCL_TAB(0x45415F00), SCL_TAB(0x458A9880), SCL_TAB(0x45D3E500), SCL_TAB(0x461D4500), SCL_TAB(0x4666B800), SCL_TAB(0x46B03E80), SCL_TAB(0x46F9D800), SCL_TAB(0x47438480), SCL_TAB(0x478D4400), SCL_TAB(0x47D71680), SCL_TAB(0x4820FC00), SCL_TAB(0x486AF500), SCL_TAB(0x48B50000), SCL_TAB(0x48FF1E80), SCL_TAB(0x49494F80), SCL_TAB(0x49939380), SCL_TAB(0x49DDEA80), SCL_TAB(0x4A285400), SCL_TAB(0x4A72D000), SCL_TAB(0x4ABD5E80), SCL_TAB(0x4B080000), SCL_TAB(0x4B52B400), SCL_TAB(0x4B9D7A80), SCL_TAB(0x4BE85380), SCL_TAB(0x4C333F00), SCL_TAB(0x4C7E3D00), SCL_TAB(0x4CC94D00), SCL_TAB(0x4D146F80), SCL_TAB(0x4D5FA500), SCL_TAB(0x4DAAEC00), SCL_TAB(0x4DF64580), SCL_TAB(0x4E41B180), SCL_TAB(0x4E8D2F00), SCL_TAB(0x4ED8BF80), SCL_TAB(0x4F246180), SCL_TAB(0x4F701600), SCL_TAB(0x4FBBDC00), SCL_TAB(0x5007B480), SCL_TAB(0x50539F00), SCL_TAB(0x509F9B80), SCL_TAB(0x50EBA980), SCL_TAB(0x5137C980), SCL_TAB(0x5183FB80), SCL_TAB(0x51D03F80), SCL_TAB(0x521C9500), SCL_TAB(0x5268FC80), SCL_TAB(0x52B57580), SCL_TAB(0x53020000), SCL_TAB(0x534E9C80), SCL_TAB(0x539B4A80), SCL_TAB(0x53E80A80), SCL_TAB(0x5434DB80), SCL_TAB(0x5481BE80), SCL_TAB(0x54CEB280), SCL_TAB(0x551BB880), SCL_TAB(0x5568CF80), SCL_TAB(0x55B5F800), SCL_TAB(0x56033200), SCL_TAB(0x56507D80), SCL_TAB(0x569DDA00), SCL_TAB(0x56EB4800), SCL_TAB(0x5738C700), SCL_TAB(0x57865780), SCL_TAB(0x57D3F900), SCL_TAB(0x5821AC00), SCL_TAB(0x586F7000), SCL_TAB(0x58BD4500), SCL_TAB(0x590B2B00), SCL_TAB(0x59592200), SCL_TAB(0x59A72A80), SCL_TAB(0x59F54380), SCL_TAB(0x5A436D80), SCL_TAB(0x5A91A900), SCL_TAB(0x5ADFF500), SCL_TAB(0x5B2E5180), SCL_TAB(0x5B7CBF80), SCL_TAB(0x5BCB3E00), SCL_TAB(0x5C19CD00), SCL_TAB(0x5C686D80), SCL_TAB(0x5CB71E00), SCL_TAB(0x5D05DF80), SCL_TAB(0x5D54B200), SCL_TAB(0x5DA39500), SCL_TAB(0x5DF28880), SCL_TAB(0x5E418C80), SCL_TAB(0x5E90A100), SCL_TAB(0x5EDFC680), SCL_TAB(0x5F2EFC00), SCL_TAB(0x5F7E4280), SCL_TAB(0x5FCD9900), SCL_TAB(0x601D0080), SCL_TAB(0x606C7800), SCL_TAB(0x60BC0000), SCL_TAB(0x610B9800), SCL_TAB(0x615B4100), SCL_TAB(0x61AAF980), SCL_TAB(0x61FAC300), SCL_TAB(0x624A9C80), SCL_TAB(0x629A8600), SCL_TAB(0x62EA8000), SCL_TAB(0x633A8A00), SCL_TAB(0x638AA480), SCL_TAB(0x63DACF00), SCL_TAB(0x642B0980), SCL_TAB(0x647B5400), SCL_TAB(0x64CBAE80), SCL_TAB(0x651C1900), SCL_TAB(0x656C9400), SCL_TAB(0x65BD1E80), SCL_TAB(0x660DB900), SCL_TAB(0x665E6380), SCL_TAB(0x66AF1E00), SCL_TAB(0x66FFE880), SCL_TAB(0x6750C280), SCL_TAB(0x67A1AC80), SCL_TAB(0x67F2A600), SCL_TAB(0x6843B000), SCL_TAB(0x6894C900), SCL_TAB(0x68E5F200), SCL_TAB(0x69372B00), SCL_TAB(0x69887380), SCL_TAB(0x69D9CB80), SCL_TAB(0x6A2B3300), SCL_TAB(0x6A7CAA80), SCL_TAB(0x6ACE3180), SCL_TAB(0x6B1FC800), SCL_TAB(0x6B716E00), SCL_TAB(0x6BC32400), SCL_TAB(0x6C14E900), SCL_TAB(0x6C66BD80), SCL_TAB(0x6CB8A180), SCL_TAB(0x6D0A9500), SCL_TAB(0x6D5C9800), SCL_TAB(0x6DAEAA00), SCL_TAB(0x6E00CB80), SCL_TAB(0x6E52FC80), SCL_TAB(0x6EA53D00), SCL_TAB(0x6EF78C80), SCL_TAB(0x6F49EB80), SCL_TAB(0x6F9C5980), SCL_TAB(0x6FEED700), SCL_TAB(0x70416380), SCL_TAB(0x7093FF00), SCL_TAB(0x70E6AA00), SCL_TAB(0x71396400), SCL_TAB(0x718C2D00), SCL_TAB(0x71DF0580), SCL_TAB(0x7231ED00), SCL_TAB(0x7284E300), SCL_TAB(0x72D7E880), SCL_TAB(0x732AFD00), SCL_TAB(0x737E2080), SCL_TAB(0x73D15300), SCL_TAB(0x74249480), SCL_TAB(0x7477E480), SCL_TAB(0x74CB4400), SCL_TAB(0x751EB200), SCL_TAB(0x75722F00), SCL_TAB(0x75C5BB00), SCL_TAB(0x76195580), SCL_TAB(0x766CFF00), SCL_TAB(0x76C0B700), SCL_TAB(0x77147E00), SCL_TAB(0x77685400), SCL_TAB(0x77BC3880), SCL_TAB(0x78102B80), SCL_TAB(0x78642D80), SCL_TAB(0x78B83E00), SCL_TAB(0x790C5D00), SCL_TAB(0x79608B00), SCL_TAB(0x79B4C780), SCL_TAB(0x7A091280), SCL_TAB(0x7A5D6C00), SCL_TAB(0x7AB1D400), SCL_TAB(0x7B064A80), SCL_TAB(0x7B5ACF80), SCL_TAB(0x7BAF6380), SCL_TAB(0x7C040580), SCL_TAB(0x7C58B600), SCL_TAB(0x7CAD7500), SCL_TAB(0x7D024200), SCL_TAB(0x7D571E00), SCL_TAB(0x7DAC0800), SCL_TAB(0x7E010080), SCL_TAB(0x7E560780), SCL_TAB(0x7EAB1C80), SCL_TAB(0x7F004000), SCL_TAB(0x7F557200), SCL_TAB(0x7FAAB200), SCL_TAB(0x7FFFFFFF)}; /** * \brief Table representing scale factor gains. Given a scale factor sf, and a * value pSpec[i] the gain is given by: MantissaTable[sf % 4][msb] = 2^(sf % 4) * / (1<> 2)) The corresponding * exponents for the values in this tables are stored in ExponentTable[sf % * 4][msb] below. */ const FIXP_DBL MantissaTable[4][14] = { {0x40000000, 0x50A28C00, 0x6597FA80, 0x40000000, 0x50A28C00, 0x6597FA80, 0x40000000, 0x50A28C00, 0x6597FA80, 0x40000000, 0x50A28C00, 0x6597FA80, 0x40000000, 0x50A28C00}, {0x4C1BF800, 0x5FE44380, 0x78D0DF80, 0x4C1BF800, 0x5FE44380, 0x78D0DF80, 0x4C1BF800, 0x5FE44380, 0x78D0DF80, 0x4C1BF800, 0x5FE44380, 0x78D0DF80, 0x4C1BF800, 0x5FE44380}, {0x5A827980, 0x7208F800, 0x47D66B00, 0x5A827980, 0x7208F800, 0x47D66B00, 0x5A827980, 0x7208F800, 0x47D66B00, 0x5A827980, 0x7208F800, 0x47D66B00, 0x5A827980, 0x7208F800}, {0x6BA27E80, 0x43CE3E80, 0x556E0400, 0x6BA27E80, 0x43CE3E80, 0x556E0400, 0x6BA27E80, 0x43CE3E80, 0x556E0400, 0x6BA27E80, 0x43CE3E80, 0x556E0400, 0x6BA27E80, 0x43CE3E80}}; const SCHAR ExponentTable[4][14] = { {1, 2, 3, 5, 6, 7, 9, 10, 11, 13, 14, 15, 17, 18}, {1, 2, 3, 5, 6, 7, 9, 10, 11, 13, 14, 15, 17, 18}, {1, 2, 4, 5, 6, 8, 9, 10, 12, 13, 14, 16, 17, 18}, {1, 3, 4, 5, 7, 8, 9, 11, 12, 13, 15, 16, 17, 19}}; /* 41 scfbands */ static const SHORT sfb_96_1024[42] = { 0, 4, 8, 12, 16, 20, 24, 28, 32, 36, 40, 44, 48, 52, 56, 64, 72, 80, 88, 96, 108, 120, 132, 144, 156, 172, 188, 212, 240, 276, 320, 384, 448, 512, 576, 640, 704, 768, 832, 896, 960, 1024}; /* 12 scfbands */ static const SHORT sfb_96_128[13] = {0, 4, 8, 12, 16, 20, 24, 32, 40, 48, 64, 92, 128}; /* 47 scfbands*/ static const SHORT sfb_64_1024[48] = { 0, 4, 8, 12, 16, 20, 24, 28, 32, 36, 40, 44, 48, 52, 56, 64, 72, 80, 88, 100, 112, 124, 140, 156, 172, 192, 216, 240, 268, 304, 344, 384, 424, 464, 504, 544, 584, 624, 664, 704, 744, 784, 824, 864, 904, 944, 984, 1024}; /* 12 scfbands */ static const SHORT sfb_64_128[13] = {0, 4, 8, 12, 16, 20, 24, 32, 40, 48, 64, 92, 128}; /* 49 scfbands */ static const SHORT sfb_48_1024[50] = { 0, 4, 8, 12, 16, 20, 24, 28, 32, 36, 40, 48, 56, 64, 72, 80, 88, 96, 108, 120, 132, 144, 160, 176, 196, 216, 240, 264, 292, 320, 352, 384, 416, 448, 480, 512, 544, 576, 608, 640, 672, 704, 736, 768, 800, 832, 864, 896, 928, 1024}; /* 14 scfbands */ static const SHORT sfb_48_128[15] = {0, 4, 8, 12, 16, 20, 28, 36, 44, 56, 68, 80, 96, 112, 128}; /* 51 scfbands */ static const SHORT sfb_32_1024[52] = { 0, 4, 8, 12, 16, 20, 24, 28, 32, 36, 40, 48, 56, 64, 72, 80, 88, 96, 108, 120, 132, 144, 160, 176, 196, 216, 240, 264, 292, 320, 352, 384, 416, 448, 480, 512, 544, 576, 608, 640, 672, 704, 736, 768, 800, 832, 864, 896, 928, 960, 992, 1024}; /* 47 scfbands */ static const SHORT sfb_24_1024[48] = { 0, 4, 8, 12, 16, 20, 24, 28, 32, 36, 40, 44, 52, 60, 68, 76, 84, 92, 100, 108, 116, 124, 136, 148, 160, 172, 188, 204, 220, 240, 260, 284, 308, 336, 364, 396, 432, 468, 508, 552, 600, 652, 704, 768, 832, 896, 960, 1024}; /* 15 scfbands */ static const SHORT sfb_24_128[16] = {0, 4, 8, 12, 16, 20, 24, 28, 36, 44, 52, 64, 76, 92, 108, 128}; /* 43 scfbands */ static const SHORT sfb_16_1024[44] = { 0, 8, 16, 24, 32, 40, 48, 56, 64, 72, 80, 88, 100, 112, 124, 136, 148, 160, 172, 184, 196, 212, 228, 244, 260, 280, 300, 320, 344, 368, 396, 424, 456, 492, 532, 572, 616, 664, 716, 772, 832, 896, 960, 1024}; /* 15 scfbands */ static const SHORT sfb_16_128[16] = {0, 4, 8, 12, 16, 20, 24, 28, 32, 40, 48, 60, 72, 88, 108, 128}; /* 40 scfbands */ static const SHORT sfb_8_1024[41] = { 0, 12, 24, 36, 48, 60, 72, 84, 96, 108, 120, 132, 144, 156, 172, 188, 204, 220, 236, 252, 268, 288, 308, 328, 348, 372, 396, 420, 448, 476, 508, 544, 580, 620, 664, 712, 764, 820, 880, 944, 1024}; /* 15 scfbands */ static const SHORT sfb_8_128[16] = {0, 4, 8, 12, 16, 20, 24, 28, 36, 44, 52, 60, 72, 88, 108, 128}; static const SHORT sfb_96_960[42] = {0, 4, 8, 12, 16, 20, 24, 28, 32, 36, 40, 44, 48, 52, 56, 64, 72, 80, 88, 96, 108, 120, 132, 144, 156, 172, 188, 212, 240, 276, 320, 384, 448, 512, 576, 640, 704, 768, 832, 896, 960}; /* 40 scfbands */ static const SHORT sfb_96_120[13] = {0, 4, 8, 12, 16, 20, 24, 32, 40, 48, 64, 92, 120}; /* 12 scfbands */ static const SHORT sfb_64_960[47] = { 0, 4, 8, 12, 16, 20, 24, 28, 32, 36, 40, 44, 48, 52, 56, 64, 72, 80, 88, 100, 112, 124, 140, 156, 172, 192, 216, 240, 268, 304, 344, 384, 424, 464, 504, 544, 584, 624, 664, 704, 744, 784, 824, 864, 904, 944, 960}; /* 46 scfbands */ static const SHORT sfb_64_120[13] = {0, 4, 8, 12, 16, 20, 24, 32, 40, 48, 64, 92, 120}; /* 12 scfbands */ static const SHORT sfb_48_960[50] = { 0, 4, 8, 12, 16, 20, 24, 28, 32, 36, 40, 48, 56, 64, 72, 80, 88, 96, 108, 120, 132, 144, 160, 176, 196, 216, 240, 264, 292, 320, 352, 384, 416, 448, 480, 512, 544, 576, 608, 640, 672, 704, 736, 768, 800, 832, 864, 896, 928, 960}; /* 49 scfbands */ static const SHORT sfb_48_120[15] = { 0, 4, 8, 12, 16, 20, 28, 36, 44, 56, 68, 80, 96, 112, 120}; /* 14 scfbands */ static const SHORT sfb_32_960[50] = { 0, 4, 8, 12, 16, 20, 24, 28, 32, 36, 40, 48, 56, 64, 72, 80, 88, 96, 108, 120, 132, 144, 160, 176, 196, 216, 240, 264, 292, 320, 352, 384, 416, 448, 480, 512, 544, 576, 608, 640, 672, 704, 736, 768, 800, 832, 864, 896, 928, 960}; /* 49 scfbands */ static const SHORT sfb_24_960[47] = { 0, 4, 8, 12, 16, 20, 24, 28, 32, 36, 40, 44, 52, 60, 68, 76, 84, 92, 100, 108, 116, 124, 136, 148, 160, 172, 188, 204, 220, 240, 260, 284, 308, 336, 364, 396, 432, 468, 508, 552, 600, 652, 704, 768, 832, 896, 960}; /* 46 scfbands */ static const SHORT sfb_24_120[16] = { 0, 4, 8, 12, 16, 20, 24, 28, 36, 44, 52, 64, 76, 92, 108, 120}; /* 15 scfbands */ static const SHORT sfb_16_960[43] = {0, 8, 16, 24, 32, 40, 48, 56, 64, 72, 80, 88, 100, 112, 124, 136, 148, 160, 172, 184, 196, 212, 228, 244, 260, 280, 300, 320, 344, 368, 396, 424, 456, 492, 532, 572, 616, 664, 716, 772, 832, 896, 960}; /* 42 scfbands */ static const SHORT sfb_16_120[16] = { 0, 4, 8, 12, 16, 20, 24, 28, 32, 40, 48, 60, 72, 88, 108, 120}; /* 15 scfbands */ static const SHORT sfb_8_960[41] = {0, 12, 24, 36, 48, 60, 72, 84, 96, 108, 120, 132, 144, 156, 172, 188, 204, 220, 236, 252, 268, 288, 308, 328, 348, 372, 396, 420, 448, 476, 508, 544, 580, 620, 664, 712, 764, 820, 880, 944, 960}; /* 40 scfbands */ static const SHORT sfb_8_120[16] = { 0, 4, 8, 12, 16, 20, 24, 28, 36, 44, 52, 60, 72, 88, 108, 120}; /* 15 scfbands */ static const SHORT sfb_96_768[] = {0, 4, 8, 12, 16, 20, 24, 28, 32, 36, 40, 44, 48, 52, 56, 64, 72, 80, 88, 96, 108, 120, 132, 144, 156, 172, 188, 212, 240, 276, 320, 384, 448, 512, 576, 640, 704, 768}; /* 37 scfbands */ static const SHORT sfb_96_96[] = {0, 4, 8, 12, 16, 20, 24, 32, 40, 48, 64, 92, 96}; /* 12 scfbands */ static const SHORT sfb_64_768[] = { 0, 4, 8, 12, 16, 20, 24, 28, 32, 36, 40, 44, 48, 52, 56, 64, 72, 80, 88, 100, 112, 124, 140, 156, 172, 192, 216, 240, 268, 304, 344, 384, 424, 464, 504, 544, 584, 624, 664, 704, 744, 768}; /* 41 scfbands */ static const SHORT sfb_64_96[] = {0, 4, 8, 12, 16, 20, 24, 32, 40, 48, 64, 92, 96}; /* 12 scfbands */ static const SHORT sfb_48_768[] = {0, 4, 8, 12, 16, 20, 24, 28, 32, 36, 40, 48, 56, 64, 72, 80, 88, 96, 108, 120, 132, 144, 160, 176, 196, 216, 240, 264, 292, 320, 352, 384, 416, 448, 480, 512, 544, 576, 608, 640, 672, 704, 736, 768}; /* 43 scfbands */ static const SHORT sfb_48_96[] = {0, 4, 8, 12, 16, 20, 28, 36, 44, 56, 68, 80, 96}; /* 12 scfbands */ static const SHORT sfb_32_768[] = {0, 4, 8, 12, 16, 20, 24, 28, 32, 36, 40, 48, 56, 64, 72, 80, 88, 96, 108, 120, 132, 144, 160, 176, 196, 216, 240, 264, 292, 320, 352, 384, 416, 448, 480, 512, 544, 576, 608, 640, 672, 704, 736, 768}; /* 43 scfbands */ static const SHORT sfb_24_768[] = {0, 4, 8, 12, 16, 20, 24, 28, 32, 36, 40, 44, 52, 60, 68, 76, 84, 92, 100, 108, 116, 124, 136, 148, 160, 172, 188, 204, 220, 240, 260, 284, 308, 336, 364, 396, 432, 468, 508, 552, 600, 652, 704, 768}; /* 43 scfbands */ static const SHORT sfb_24_96[] = {0, 4, 8, 12, 16, 20, 24, 28, 36, 44, 52, 64, 76, 92, 96}; /* 14 scfbands */ static const SHORT sfb_16_768[] = {0, 8, 16, 24, 32, 40, 48, 56, 64, 72, 80, 88, 100, 112, 124, 136, 148, 160, 172, 184, 196, 212, 228, 244, 260, 280, 300, 320, 344, 368, 396, 424, 456, 492, 532, 572, 616, 664, 716, 768}; /* 39 scfbands */ static const SHORT sfb_16_96[] = {0, 4, 8, 12, 16, 20, 24, 28, 32, 40, 48, 60, 72, 88, 96}; /* 14 scfbands */ static const SHORT sfb_8_768[] = {0, 12, 24, 36, 48, 60, 72, 84, 96, 108, 120, 132, 144, 156, 172, 188, 204, 220, 236, 252, 268, 288, 308, 328, 348, 372, 396, 420, 448, 476, 508, 544, 580, 620, 664, 712, 764, 768}; /* 37 scfbands */ static const SHORT sfb_8_96[] = {0, 4, 8, 12, 16, 20, 24, 28, 36, 44, 52, 60, 72, 88, 96}; /* 14 scfbands */ static const SHORT sfb_48_512[37] = { 0, 4, 8, 12, 16, 20, 24, 28, 32, 36, 40, 44, 48, 52, 56, 60, 68, 76, 84, 92, 100, 112, 124, 136, 148, 164, 184, 208, 236, 268, 300, 332, 364, 396, 428, 460, 512}; /* 36 scfbands */ static const SHORT sfb_32_512[38] = {0, 4, 8, 12, 16, 20, 24, 28, 32, 36, 40, 44, 48, 52, 56, 64, 72, 80, 88, 96, 108, 120, 132, 144, 160, 176, 192, 212, 236, 260, 288, 320, 352, 384, 416, 448, 480, 512}; /* 37 scfbands */ static const SHORT sfb_24_512[32] = { 0, 4, 8, 12, 16, 20, 24, 28, 32, 36, 40, 44, 52, 60, 68, 80, 92, 104, 120, 140, 164, 192, 224, 256, 288, 320, 352, 384, 416, 448, 480, 512}; /* 31 scfbands */ static const SHORT sfb_48_480[36] = { 0, 4, 8, 12, 16, 20, 24, 28, 32, 36, 40, 44, 48, 52, 56, 64, 72, 80, 88, 96, 108, 120, 132, 144, 156, 172, 188, 212, 240, 272, 304, 336, 368, 400, 432, 480}; /* 35 scfbands */ static const SHORT sfb_32_480[38] = {0, 4, 8, 12, 16, 20, 24, 28, 32, 36, 40, 44, 48, 52, 56, 60, 64, 72, 80, 88, 96, 104, 112, 124, 136, 148, 164, 180, 200, 224, 256, 288, 320, 352, 384, 416, 448, 480}; /* 37 scfbands */ static const SHORT sfb_24_480[31] = {0, 4, 8, 12, 16, 20, 24, 28, 32, 36, 40, 44, 52, 60, 68, 80, 92, 104, 120, 140, 164, 192, 224, 256, 288, 320, 352, 384, 416, 448, 480}; /* 30 scfbands */ const SFB_INFO sfbOffsetTables[5][16] = {{ {sfb_96_1024, sfb_96_128, 41, 12}, {sfb_96_1024, sfb_96_128, 41, 12}, {sfb_64_1024, sfb_64_128, 47, 12}, {sfb_48_1024, sfb_48_128, 49, 14}, {sfb_48_1024, sfb_48_128, 49, 14}, {sfb_32_1024, sfb_48_128, 51, 14}, {sfb_24_1024, sfb_24_128, 47, 15}, {sfb_24_1024, sfb_24_128, 47, 15}, {sfb_16_1024, sfb_16_128, 43, 15}, {sfb_16_1024, sfb_16_128, 43, 15}, {sfb_16_1024, sfb_16_128, 43, 15}, {sfb_8_1024, sfb_8_128, 40, 15}, {sfb_8_1024, sfb_8_128, 40, 15}, }, { {sfb_96_960, sfb_96_120, 40, 12}, {sfb_96_960, sfb_96_120, 40, 12}, {sfb_64_960, sfb_64_120, 46, 12}, {sfb_48_960, sfb_48_120, 49, 14}, {sfb_48_960, sfb_48_120, 49, 14}, {sfb_32_960, sfb_48_120, 49, 14}, {sfb_24_960, sfb_24_120, 46, 15}, {sfb_24_960, sfb_24_120, 46, 15}, {sfb_16_960, sfb_16_120, 42, 15}, {sfb_16_960, sfb_16_120, 42, 15}, {sfb_16_960, sfb_16_120, 42, 15}, {sfb_8_960, sfb_8_120, 40, 15}, {sfb_8_960, sfb_8_120, 40, 15}, }, { {sfb_96_768, sfb_96_96, 37, 12}, {sfb_96_768, sfb_96_96, 37, 12}, {sfb_64_768, sfb_64_96, 41, 12}, {sfb_48_768, sfb_48_96, 43, 12}, {sfb_48_768, sfb_48_96, 43, 12}, {sfb_32_768, sfb_48_96, 43, 12}, {sfb_24_768, sfb_24_96, 43, 14}, {sfb_24_768, sfb_24_96, 43, 14}, {sfb_16_768, sfb_16_96, 39, 14}, {sfb_16_768, sfb_16_96, 39, 14}, {sfb_16_768, sfb_16_96, 39, 14}, {sfb_8_768, sfb_8_96, 37, 14}, {sfb_8_768, sfb_8_96, 37, 14}, }, { {sfb_48_512, NULL, 36, 0}, {sfb_48_512, NULL, 36, 0}, {sfb_48_512, NULL, 36, 0}, {sfb_48_512, NULL, 36, 0}, {sfb_48_512, NULL, 36, 0}, {sfb_32_512, NULL, 37, 0}, {sfb_24_512, NULL, 31, 0}, {sfb_24_512, NULL, 31, 0}, {sfb_24_512, NULL, 31, 0}, {sfb_24_512, NULL, 31, 0}, {sfb_24_512, NULL, 31, 0}, {sfb_24_512, NULL, 31, 0}, {sfb_24_512, NULL, 31, 0}, }, { {sfb_48_480, NULL, 35, 0}, {sfb_48_480, NULL, 35, 0}, {sfb_48_480, NULL, 35, 0}, {sfb_48_480, NULL, 35, 0}, {sfb_48_480, NULL, 35, 0}, {sfb_32_480, NULL, 37, 0}, {sfb_24_480, NULL, 30, 0}, {sfb_24_480, NULL, 30, 0}, {sfb_24_480, NULL, 30, 0}, {sfb_24_480, NULL, 30, 0}, {sfb_24_480, NULL, 30, 0}, {sfb_24_480, NULL, 30, 0}, {sfb_24_480, NULL, 30, 0}, }}; /*# don't use 1 bit hufman tables */ /* MPEG-2 AAC 2 BITS parallel Hufman Tables Bit 0: = 1=ENDNODE, 0=INDEX Bit 1: = CODEWORD LEN MOD 2 Bit 2..9: = VALUE/REF Tables 1..10,SCL Bit 2..11: = VALUE/REF Table 11 */ const USHORT HuffmanCodeBook_1[51][4] = { {0x0157, 0x0157, 0x0004, 0x0018}, {0x0008, 0x000c, 0x0010, 0x0014}, {0x015b, 0x015b, 0x0153, 0x0153}, {0x0057, 0x0057, 0x0167, 0x0167}, {0x0257, 0x0257, 0x0117, 0x0117}, {0x0197, 0x0197, 0x0147, 0x0147}, {0x001c, 0x0030, 0x0044, 0x0058}, {0x0020, 0x0024, 0x0028, 0x002c}, {0x014b, 0x014b, 0x0163, 0x0163}, {0x0217, 0x0217, 0x0127, 0x0127}, {0x0187, 0x0187, 0x0097, 0x0097}, {0x016b, 0x016b, 0x0017, 0x0017}, {0x0034, 0x0038, 0x003c, 0x0040}, {0x0143, 0x0143, 0x0107, 0x0107}, {0x011b, 0x011b, 0x0067, 0x0067}, {0x0193, 0x0193, 0x0297, 0x0297}, {0x019b, 0x019b, 0x0247, 0x0247}, {0x0048, 0x004c, 0x0050, 0x0054}, {0x01a7, 0x01a7, 0x0267, 0x0267}, {0x0113, 0x0113, 0x025b, 0x025b}, {0x0053, 0x0053, 0x005b, 0x005b}, {0x0253, 0x0253, 0x0047, 0x0047}, {0x005c, 0x0070, 0x0084, 0x0098}, {0x0060, 0x0064, 0x0068, 0x006c}, {0x012b, 0x012b, 0x0123, 0x0123}, {0x018b, 0x018b, 0x00a7, 0x00a7}, {0x0227, 0x0227, 0x0287, 0x0287}, {0x0087, 0x0087, 0x010b, 0x010b}, {0x0074, 0x0078, 0x007c, 0x0080}, {0x021b, 0x021b, 0x0027, 0x0027}, {0x01a3, 0x01a3, 0x0093, 0x0093}, {0x0183, 0x0183, 0x0207, 0x0207}, {0x024b, 0x024b, 0x004b, 0x004b}, {0x0088, 0x008c, 0x0090, 0x0094}, {0x0063, 0x0063, 0x0103, 0x0103}, {0x0007, 0x0007, 0x02a7, 0x02a7}, {0x009b, 0x009b, 0x026b, 0x026b}, {0x0263, 0x0263, 0x01ab, 0x01ab}, {0x009c, 0x00a0, 0x00a4, 0x00b8}, {0x0241, 0x0011, 0x0069, 0x0019}, {0x0211, 0x0041, 0x0291, 0x0299}, {0x00a8, 0x00ac, 0x00b0, 0x00b4}, {0x008b, 0x008b, 0x0223, 0x0223}, {0x00a3, 0x00a3, 0x020b, 0x020b}, {0x02ab, 0x02ab, 0x0283, 0x0283}, {0x002b, 0x002b, 0x0083, 0x0083}, {0x00bc, 0x00c0, 0x00c4, 0x00c8}, {0x0003, 0x0003, 0x022b, 0x022b}, {0x028b, 0x028b, 0x02a3, 0x02a3}, {0x0023, 0x0023, 0x0203, 0x0203}, {0x000b, 0x000b, 0x00ab, 0x00ab}}; const USHORT HuffmanCodeBook_2[39][4] = { {0x0004, 0x000c, 0x0020, 0x0034}, {0x0157, 0x0157, 0x0159, 0x0008}, {0x0153, 0x0153, 0x0257, 0x0257}, {0x0010, 0x0014, 0x0018, 0x001c}, {0x0117, 0x0117, 0x0057, 0x0057}, {0x0147, 0x0147, 0x0197, 0x0197}, {0x0167, 0x0167, 0x0185, 0x0161}, {0x0125, 0x0095, 0x0065, 0x0215}, {0x0024, 0x0028, 0x002c, 0x0030}, {0x0051, 0x0149, 0x0119, 0x0141}, {0x0015, 0x0199, 0x0259, 0x0245}, {0x0191, 0x0265, 0x0105, 0x0251}, {0x0045, 0x0111, 0x0169, 0x01a5}, {0x0038, 0x0044, 0x0058, 0x006c}, {0x0295, 0x0059, 0x003c, 0x0040}, {0x0227, 0x0227, 0x021b, 0x021b}, {0x0123, 0x0123, 0x0087, 0x0087}, {0x0048, 0x004c, 0x0050, 0x0054}, {0x018b, 0x018b, 0x006b, 0x006b}, {0x029b, 0x029b, 0x01a3, 0x01a3}, {0x0207, 0x0207, 0x01ab, 0x01ab}, {0x0093, 0x0093, 0x0103, 0x0103}, {0x005c, 0x0060, 0x0064, 0x0068}, {0x0213, 0x0213, 0x010b, 0x010b}, {0x012b, 0x012b, 0x0249, 0x0061}, {0x0181, 0x0291, 0x0241, 0x0041}, {0x0005, 0x0099, 0x0019, 0x0025}, {0x0070, 0x0074, 0x0078, 0x0088}, {0x02a5, 0x0261, 0x0011, 0x00a5}, {0x0049, 0x0285, 0x0269, 0x0089}, {0x0221, 0x007c, 0x0080, 0x0084}, {0x020b, 0x020b, 0x0003, 0x0003}, {0x00a3, 0x00a3, 0x02a3, 0x02a3}, {0x02ab, 0x02ab, 0x0083, 0x0083}, {0x008c, 0x0090, 0x0094, 0x0098}, {0x028b, 0x028b, 0x0023, 0x0023}, {0x0283, 0x0283, 0x002b, 0x002b}, {0x000b, 0x000b, 0x0203, 0x0203}, {0x022b, 0x022b, 0x00ab, 0x00ab}}; const USHORT HuffmanCodeBook_3[39][4] = { {0x0003, 0x0003, 0x0004, 0x0008}, {0x0005, 0x0101, 0x0011, 0x0041}, {0x000c, 0x0010, 0x0014, 0x0020}, {0x0017, 0x0017, 0x0143, 0x0143}, {0x0051, 0x0111, 0x0045, 0x0151}, {0x0105, 0x0055, 0x0018, 0x001c}, {0x0157, 0x0157, 0x0147, 0x0147}, {0x0117, 0x0117, 0x0009, 0x0201}, {0x0024, 0x002c, 0x0040, 0x0054}, {0x0241, 0x0019, 0x0065, 0x0028}, {0x0183, 0x0183, 0x0193, 0x0193}, {0x0030, 0x0034, 0x0038, 0x003c}, {0x0027, 0x0027, 0x0253, 0x0253}, {0x005b, 0x005b, 0x0083, 0x0083}, {0x0063, 0x0063, 0x0093, 0x0093}, {0x0023, 0x0023, 0x0213, 0x0213}, {0x0044, 0x0048, 0x004c, 0x0050}, {0x004b, 0x004b, 0x0167, 0x0167}, {0x0163, 0x0163, 0x0097, 0x0097}, {0x0197, 0x0197, 0x0125, 0x0085}, {0x0185, 0x0121, 0x0159, 0x0255}, {0x0058, 0x005c, 0x0060, 0x0070}, {0x0119, 0x0245, 0x0281, 0x0291}, {0x0069, 0x00a5, 0x0205, 0x0109}, {0x01a1, 0x0064, 0x0068, 0x006c}, {0x002b, 0x002b, 0x01a7, 0x01a7}, {0x0217, 0x0217, 0x014b, 0x014b}, {0x0297, 0x0297, 0x016b, 0x016b}, {0x0074, 0x0078, 0x007c, 0x0080}, {0x00a3, 0x00a3, 0x0263, 0x0263}, {0x0285, 0x0129, 0x0099, 0x00a9}, {0x02a1, 0x01a9, 0x0199, 0x0265}, {0x02a5, 0x0084, 0x0088, 0x008c}, {0x0223, 0x0223, 0x008b, 0x008b}, {0x0227, 0x0227, 0x0189, 0x0259}, {0x0219, 0x0090, 0x0094, 0x0098}, {0x02ab, 0x02ab, 0x026b, 0x026b}, {0x029b, 0x029b, 0x024b, 0x024b}, {0x020b, 0x020b, 0x0229, 0x0289}}; const USHORT HuffmanCodeBook_4[38][4] = { {0x0004, 0x0008, 0x000c, 0x0018}, {0x0155, 0x0151, 0x0115, 0x0055}, {0x0145, 0x0005, 0x0015, 0x0001}, {0x0141, 0x0045, 0x0010, 0x0014}, {0x0107, 0x0107, 0x0053, 0x0053}, {0x0103, 0x0103, 0x0113, 0x0113}, {0x001c, 0x0020, 0x0034, 0x0048}, {0x0043, 0x0043, 0x0013, 0x0013}, {0x0024, 0x0028, 0x002c, 0x0030}, {0x015b, 0x015b, 0x0197, 0x0197}, {0x0167, 0x0167, 0x0257, 0x0257}, {0x005b, 0x005b, 0x011b, 0x011b}, {0x0067, 0x0067, 0x014b, 0x014b}, {0x0038, 0x003c, 0x0040, 0x0044}, {0x0193, 0x0193, 0x0251, 0x0095}, {0x0161, 0x0245, 0x0125, 0x0215}, {0x0185, 0x0019, 0x0049, 0x0025}, {0x0109, 0x0211, 0x0061, 0x0241}, {0x004c, 0x0050, 0x0058, 0x006c}, {0x0091, 0x0121, 0x0205, 0x0181}, {0x0085, 0x0009, 0x0201, 0x0054}, {0x0023, 0x0023, 0x0083, 0x0083}, {0x005c, 0x0060, 0x0064, 0x0068}, {0x01a7, 0x01a7, 0x016b, 0x016b}, {0x019b, 0x019b, 0x0297, 0x0297}, {0x0267, 0x0267, 0x025b, 0x025b}, {0x00a5, 0x0069, 0x0099, 0x01a1}, {0x0070, 0x0074, 0x0078, 0x0084}, {0x0291, 0x0129, 0x0261, 0x0189}, {0x0285, 0x01a9, 0x0225, 0x0249}, {0x0219, 0x02a5, 0x007c, 0x0080}, {0x029b, 0x029b, 0x026b, 0x026b}, {0x00a3, 0x00a3, 0x002b, 0x002b}, {0x0088, 0x008c, 0x0090, 0x0094}, {0x0283, 0x0283, 0x008b, 0x008b}, {0x0223, 0x0223, 0x020b, 0x020b}, {0x02ab, 0x02ab, 0x02a3, 0x02a3}, {0x00ab, 0x00ab, 0x0229, 0x0289}}; const USHORT HuffmanCodeBook_5[41][4] = { {0x0113, 0x0113, 0x0004, 0x0008}, {0x010d, 0x0115, 0x0151, 0x00d1}, {0x000c, 0x0010, 0x0014, 0x0028}, {0x00d7, 0x00d7, 0x014f, 0x014f}, {0x00cf, 0x00cf, 0x0157, 0x0157}, {0x0018, 0x001c, 0x0020, 0x0024}, {0x010b, 0x010b, 0x0193, 0x0193}, {0x011b, 0x011b, 0x0093, 0x0093}, {0x00c9, 0x0159, 0x008d, 0x0195}, {0x0149, 0x00d9, 0x018d, 0x0095}, {0x002c, 0x0030, 0x0044, 0x0058}, {0x0105, 0x011d, 0x0051, 0x01d1}, {0x0034, 0x0038, 0x003c, 0x0040}, {0x00c7, 0x00c7, 0x01d7, 0x01d7}, {0x015f, 0x015f, 0x004f, 0x004f}, {0x0147, 0x0147, 0x00df, 0x00df}, {0x0057, 0x0057, 0x01cf, 0x01cf}, {0x0048, 0x004c, 0x0050, 0x0054}, {0x018b, 0x018b, 0x019b, 0x019b}, {0x008b, 0x008b, 0x009b, 0x009b}, {0x0085, 0x009d, 0x01c9, 0x0059}, {0x019d, 0x01d9, 0x0185, 0x0049}, {0x005c, 0x0060, 0x0074, 0x0088}, {0x0011, 0x0101, 0x0161, 0x0121}, {0x0064, 0x0068, 0x006c, 0x0070}, {0x00c3, 0x00c3, 0x0213, 0x0213}, {0x00e3, 0x00e3, 0x000f, 0x000f}, {0x0217, 0x0217, 0x020f, 0x020f}, {0x0143, 0x0143, 0x0017, 0x0017}, {0x0078, 0x007c, 0x0080, 0x0084}, {0x005f, 0x005f, 0x0047, 0x0047}, {0x01c7, 0x01c7, 0x020b, 0x020b}, {0x0083, 0x0083, 0x01a3, 0x01a3}, {0x001b, 0x001b, 0x021b, 0x021b}, {0x008c, 0x0090, 0x0094, 0x0098}, {0x01df, 0x01df, 0x0183, 0x0183}, {0x0009, 0x00a1, 0x001d, 0x0041}, {0x01c1, 0x021d, 0x0205, 0x01e1}, {0x0061, 0x0005, 0x009c, 0x00a0}, {0x0023, 0x0023, 0x0203, 0x0203}, {0x0223, 0x0223, 0x0003, 0x0003}}; const USHORT HuffmanCodeBook_6[40][4] = { {0x0004, 0x0008, 0x000c, 0x001c}, {0x0111, 0x0115, 0x00d1, 0x0151}, {0x010d, 0x0155, 0x014d, 0x00d5}, {0x00cd, 0x0010, 0x0014, 0x0018}, {0x00d9, 0x0159, 0x0149, 0x00c9}, {0x0109, 0x018d, 0x0119, 0x0095}, {0x0195, 0x0091, 0x008d, 0x0191}, {0x0020, 0x0024, 0x0038, 0x004c}, {0x0099, 0x0189, 0x0089, 0x0199}, {0x0028, 0x002c, 0x0030, 0x0034}, {0x0147, 0x0147, 0x015f, 0x015f}, {0x00df, 0x00df, 0x01cf, 0x01cf}, {0x00c7, 0x00c7, 0x01d7, 0x01d7}, {0x0057, 0x0057, 0x004f, 0x004f}, {0x003c, 0x0040, 0x0044, 0x0048}, {0x011f, 0x011f, 0x0107, 0x0107}, {0x0053, 0x0053, 0x01d3, 0x01d3}, {0x019f, 0x019f, 0x0085, 0x01c9}, {0x01d9, 0x009d, 0x0059, 0x0049}, {0x0050, 0x005c, 0x0070, 0x0084}, {0x0185, 0x01dd, 0x0054, 0x0058}, {0x005f, 0x005f, 0x0047, 0x0047}, {0x01c7, 0x01c7, 0x0017, 0x0017}, {0x0060, 0x0064, 0x0068, 0x006c}, {0x000f, 0x000f, 0x0163, 0x0163}, {0x0143, 0x0143, 0x00c3, 0x00c3}, {0x0217, 0x0217, 0x00e3, 0x00e3}, {0x020f, 0x020f, 0x0013, 0x0013}, {0x0074, 0x0078, 0x007c, 0x0080}, {0x0183, 0x0183, 0x0083, 0x0083}, {0x021b, 0x021b, 0x000b, 0x000b}, {0x0103, 0x0103, 0x01a3, 0x01a3}, {0x00a3, 0x00a3, 0x020b, 0x020b}, {0x0088, 0x008c, 0x0090, 0x0094}, {0x0123, 0x0123, 0x001b, 0x001b}, {0x0213, 0x0213, 0x0005, 0x0205}, {0x001d, 0x0061, 0x021d, 0x01e1}, {0x01c1, 0x0041, 0x0098, 0x009c}, {0x0223, 0x0223, 0x0203, 0x0203}, {0x0003, 0x0003, 0x0023, 0x0023}}; const USHORT HuffmanCodeBook_7[31][4] = { {0x0003, 0x0003, 0x0004, 0x0008}, {0x0007, 0x0007, 0x0043, 0x0043}, {0x0045, 0x000c, 0x0010, 0x0024}, {0x0049, 0x0085, 0x0009, 0x0081}, {0x0014, 0x0018, 0x001c, 0x0020}, {0x004f, 0x004f, 0x00c7, 0x00c7}, {0x008b, 0x008b, 0x000f, 0x000f}, {0x00c3, 0x00c3, 0x00c9, 0x008d}, {0x0105, 0x0051, 0x0145, 0x0055}, {0x0028, 0x002c, 0x0040, 0x0054}, {0x00cd, 0x0109, 0x0101, 0x0011}, {0x0030, 0x0034, 0x0038, 0x003c}, {0x0093, 0x0093, 0x014b, 0x014b}, {0x0097, 0x0097, 0x0143, 0x0143}, {0x005b, 0x005b, 0x0017, 0x0017}, {0x0187, 0x0187, 0x00d3, 0x00d3}, {0x0044, 0x0048, 0x004c, 0x0050}, {0x014f, 0x014f, 0x010f, 0x010f}, {0x00d7, 0x00d7, 0x018b, 0x018b}, {0x009b, 0x009b, 0x01c7, 0x01c7}, {0x018d, 0x0181, 0x0019, 0x0111}, {0x0058, 0x005c, 0x0060, 0x0068}, {0x005d, 0x0151, 0x009d, 0x0115}, {0x00d9, 0x01c9, 0x00dd, 0x0119}, {0x0155, 0x0191, 0x01cd, 0x0064}, {0x001f, 0x001f, 0x01c3, 0x01c3}, {0x006c, 0x0070, 0x0074, 0x0078}, {0x015b, 0x015b, 0x0197, 0x0197}, {0x011f, 0x011f, 0x01d3, 0x01d3}, {0x01d7, 0x01d7, 0x015f, 0x015f}, {0x019d, 0x0199, 0x01d9, 0x01dd}}; const USHORT HuffmanCodeBook_8[31][4] = { {0x0004, 0x0008, 0x0010, 0x0024}, {0x0047, 0x0047, 0x0049, 0x0005}, {0x0085, 0x0041, 0x0089, 0x000c}, {0x0003, 0x0003, 0x000b, 0x000b}, {0x0014, 0x0018, 0x001c, 0x0020}, {0x0083, 0x0083, 0x004f, 0x004f}, {0x00c7, 0x00c7, 0x008f, 0x008f}, {0x00cb, 0x00cb, 0x00cd, 0x0051}, {0x0105, 0x0091, 0x0109, 0x000d}, {0x0028, 0x002c, 0x0040, 0x0054}, {0x00c1, 0x00d1, 0x010d, 0x0095}, {0x0030, 0x0034, 0x0038, 0x003c}, {0x0057, 0x0057, 0x014b, 0x014b}, {0x0147, 0x0147, 0x00d7, 0x00d7}, {0x014f, 0x014f, 0x0113, 0x0113}, {0x0117, 0x0117, 0x0103, 0x0103}, {0x0044, 0x0048, 0x004c, 0x0050}, {0x0153, 0x0153, 0x0013, 0x0013}, {0x018b, 0x018b, 0x009b, 0x009b}, {0x005b, 0x005b, 0x0187, 0x0187}, {0x018d, 0x00d9, 0x0155, 0x0015}, {0x0058, 0x005c, 0x0060, 0x0068}, {0x0119, 0x0141, 0x0191, 0x005d}, {0x009d, 0x01c9, 0x0159, 0x00dd}, {0x01c5, 0x0195, 0x01cd, 0x0064}, {0x019b, 0x019b, 0x011f, 0x011f}, {0x006c, 0x0070, 0x0074, 0x0078}, {0x001b, 0x001b, 0x01d3, 0x01d3}, {0x0183, 0x0183, 0x015f, 0x015f}, {0x019f, 0x019f, 0x01db, 0x01db}, {0x01d5, 0x001d, 0x01c1, 0x01dd}}; const USHORT HuffmanCodeBook_9[84][4] = { {0x0003, 0x0003, 0x0004, 0x0008}, {0x0007, 0x0007, 0x0043, 0x0043}, {0x0045, 0x000c, 0x0010, 0x002c}, {0x0049, 0x0085, 0x0009, 0x0081}, {0x0014, 0x0018, 0x001c, 0x0020}, {0x004f, 0x004f, 0x008b, 0x008b}, {0x00c7, 0x00c7, 0x000d, 0x00c1}, {0x00c9, 0x008d, 0x0105, 0x0051}, {0x0109, 0x0145, 0x0024, 0x0028}, {0x0093, 0x0093, 0x00cf, 0x00cf}, {0x0103, 0x0103, 0x0013, 0x0013}, {0x0030, 0x0044, 0x0058, 0x00a4}, {0x0034, 0x0038, 0x003c, 0x0040}, {0x0057, 0x0057, 0x014b, 0x014b}, {0x0187, 0x0187, 0x010f, 0x010f}, {0x0097, 0x0097, 0x005b, 0x005b}, {0x00d3, 0x00d3, 0x0141, 0x0189}, {0x0048, 0x004c, 0x0050, 0x0054}, {0x0015, 0x01c5, 0x014d, 0x0205}, {0x0061, 0x0111, 0x00d5, 0x0099}, {0x005d, 0x0181, 0x00a1, 0x0209}, {0x018d, 0x01c9, 0x0151, 0x0065}, {0x005c, 0x0068, 0x007c, 0x0090}, {0x0245, 0x009d, 0x0060, 0x0064}, {0x001b, 0x001b, 0x0117, 0x0117}, {0x00db, 0x00db, 0x00e3, 0x00e3}, {0x006c, 0x0070, 0x0074, 0x0078}, {0x01c3, 0x01c3, 0x00a7, 0x00a7}, {0x020f, 0x020f, 0x0193, 0x0193}, {0x01cf, 0x01cf, 0x0203, 0x0203}, {0x006b, 0x006b, 0x011b, 0x011b}, {0x0080, 0x0084, 0x0088, 0x008c}, {0x024b, 0x024b, 0x0157, 0x0157}, {0x0023, 0x0023, 0x001f, 0x001f}, {0x00df, 0x00df, 0x00ab, 0x00ab}, {0x00e7, 0x00e7, 0x0123, 0x0123}, {0x0094, 0x0098, 0x009c, 0x00a0}, {0x0287, 0x0287, 0x011f, 0x011f}, {0x015b, 0x015b, 0x0197, 0x0197}, {0x0213, 0x0213, 0x01d3, 0x01d3}, {0x024f, 0x024f, 0x006f, 0x006f}, {0x00a8, 0x00bc, 0x00d0, 0x00f4}, {0x00ac, 0x00b0, 0x00b4, 0x00b8}, {0x0217, 0x0217, 0x0027, 0x0027}, {0x0163, 0x0163, 0x00e9, 0x0289}, {0x0241, 0x00ad, 0x0125, 0x0199}, {0x0071, 0x0251, 0x01a1, 0x02c5}, {0x00c0, 0x00c4, 0x00c8, 0x00cc}, {0x0165, 0x0129, 0x01d5, 0x015d}, {0x02c9, 0x0305, 0x00b1, 0x00ed}, {0x028d, 0x0255, 0x01d9, 0x01e1}, {0x012d, 0x0281, 0x019d, 0x00f1}, {0x00d4, 0x00d8, 0x00dc, 0x00e0}, {0x0029, 0x0169, 0x0291, 0x0219}, {0x0309, 0x01a5, 0x01e5, 0x02d1}, {0x002d, 0x0259, 0x02cd, 0x0295}, {0x00e4, 0x00e8, 0x00ec, 0x00f0}, {0x0223, 0x0223, 0x021f, 0x021f}, {0x0173, 0x0173, 0x030f, 0x030f}, {0x016f, 0x016f, 0x01df, 0x01df}, {0x0133, 0x0133, 0x01af, 0x01af}, {0x00f8, 0x010c, 0x0120, 0x0134}, {0x00fc, 0x0100, 0x0104, 0x0108}, {0x01ab, 0x01ab, 0x0313, 0x0313}, {0x025f, 0x025f, 0x02d7, 0x02d7}, {0x02c3, 0x02c3, 0x01b3, 0x01b3}, {0x029b, 0x029b, 0x0033, 0x0033}, {0x0110, 0x0114, 0x0118, 0x011c}, {0x01eb, 0x01eb, 0x0317, 0x0317}, {0x029f, 0x029f, 0x0227, 0x0227}, {0x0303, 0x0303, 0x01ef, 0x01ef}, {0x0263, 0x0263, 0x0267, 0x0267}, {0x0124, 0x0128, 0x012c, 0x0130}, {0x022b, 0x022b, 0x02df, 0x02df}, {0x01f3, 0x01f3, 0x02db, 0x02db}, {0x02e3, 0x02e3, 0x022f, 0x022f}, {0x031f, 0x031f, 0x031b, 0x031b}, {0x0138, 0x013c, 0x0140, 0x0144}, {0x02a1, 0x0269, 0x0321, 0x02a5}, {0x02e5, 0x0325, 0x02e9, 0x0271}, {0x02a9, 0x026d, 0x0231, 0x02ad}, {0x02b1, 0x02f1, 0x0148, 0x014c}, {0x032b, 0x032b, 0x02ef, 0x02ef}, {0x032f, 0x032f, 0x0333, 0x0333}}; const USHORT HuffmanCodeBook_10[82][4] = { {0x0004, 0x000c, 0x0020, 0x004c}, {0x0045, 0x0085, 0x0049, 0x0008}, {0x008b, 0x008b, 0x0007, 0x0007}, {0x0010, 0x0014, 0x0018, 0x001c}, {0x0043, 0x0043, 0x00c7, 0x00c7}, {0x008f, 0x008f, 0x004f, 0x004f}, {0x00cb, 0x00cb, 0x00cf, 0x00cf}, {0x0009, 0x0081, 0x0109, 0x0091}, {0x0024, 0x0028, 0x002c, 0x0038}, {0x0105, 0x0051, 0x0001, 0x00d1}, {0x010d, 0x000d, 0x00c1, 0x0111}, {0x0149, 0x0095, 0x0030, 0x0034}, {0x0147, 0x0147, 0x0057, 0x0057}, {0x00d7, 0x00d7, 0x014f, 0x014f}, {0x003c, 0x0040, 0x0044, 0x0048}, {0x0117, 0x0117, 0x0153, 0x0153}, {0x009b, 0x009b, 0x018b, 0x018b}, {0x00db, 0x00db, 0x0013, 0x0013}, {0x005b, 0x005b, 0x0103, 0x0103}, {0x0050, 0x0064, 0x0078, 0x00c0}, {0x0054, 0x0058, 0x005c, 0x0060}, {0x0187, 0x0187, 0x018f, 0x018f}, {0x0157, 0x0157, 0x011b, 0x011b}, {0x0193, 0x0193, 0x0159, 0x009d}, {0x01cd, 0x01c9, 0x0195, 0x00a1}, {0x0068, 0x006c, 0x0070, 0x0074}, {0x00dd, 0x0015, 0x005d, 0x0141}, {0x0061, 0x01c5, 0x00e1, 0x011d}, {0x01d1, 0x0209, 0x0199, 0x015d}, {0x0205, 0x020d, 0x0121, 0x0211}, {0x007c, 0x0084, 0x0098, 0x00ac}, {0x01d5, 0x0161, 0x0215, 0x0080}, {0x019f, 0x019f, 0x01db, 0x01db}, {0x0088, 0x008c, 0x0090, 0x0094}, {0x00a7, 0x00a7, 0x001b, 0x001b}, {0x021b, 0x021b, 0x00e7, 0x00e7}, {0x024f, 0x024f, 0x0067, 0x0067}, {0x024b, 0x024b, 0x0183, 0x0183}, {0x009c, 0x00a0, 0x00a4, 0x00a8}, {0x01a3, 0x01a3, 0x0127, 0x0127}, {0x0253, 0x0253, 0x00ab, 0x00ab}, {0x0247, 0x0247, 0x01df, 0x01df}, {0x01e3, 0x01e3, 0x0167, 0x0167}, {0x00b0, 0x00b4, 0x00b8, 0x00bc}, {0x021f, 0x021f, 0x00eb, 0x00eb}, {0x0257, 0x0257, 0x012b, 0x012b}, {0x028b, 0x028b, 0x006b, 0x006b}, {0x028f, 0x028f, 0x01a7, 0x01a7}, {0x00c4, 0x00d8, 0x00ec, 0x0100}, {0x00c8, 0x00cc, 0x00d0, 0x00d4}, {0x025b, 0x025b, 0x0023, 0x0023}, {0x0293, 0x0293, 0x001f, 0x001f}, {0x00af, 0x00af, 0x025d, 0x00ed}, {0x01a9, 0x0285, 0x006d, 0x01e5}, {0x00dc, 0x00e0, 0x00e4, 0x00e8}, {0x01c1, 0x0221, 0x0169, 0x02cd}, {0x0295, 0x0261, 0x016d, 0x0201}, {0x012d, 0x02c9, 0x029d, 0x0299}, {0x01e9, 0x02d1, 0x02c5, 0x00b1}, {0x00f0, 0x00f4, 0x00f8, 0x00fc}, {0x0225, 0x00f1, 0x01ad, 0x02d5}, {0x0131, 0x01ed, 0x0171, 0x030d}, {0x02d9, 0x0025, 0x0229, 0x0029}, {0x0071, 0x0241, 0x0311, 0x0265}, {0x0104, 0x010c, 0x0120, 0x0134}, {0x01b1, 0x0309, 0x02a1, 0x0108}, {0x02a7, 0x02a7, 0x0307, 0x0307}, {0x0110, 0x0114, 0x0118, 0x011c}, {0x022f, 0x022f, 0x01f3, 0x01f3}, {0x02df, 0x02df, 0x0317, 0x0317}, {0x031b, 0x031b, 0x026b, 0x026b}, {0x02e3, 0x02e3, 0x0233, 0x0233}, {0x0124, 0x0128, 0x012c, 0x0130}, {0x0283, 0x0283, 0x031f, 0x031f}, {0x002f, 0x002f, 0x02ab, 0x02ab}, {0x026f, 0x026f, 0x02af, 0x02af}, {0x02c3, 0x02c3, 0x02ef, 0x02ef}, {0x0138, 0x013c, 0x0140, 0x0144}, {0x02e7, 0x02e7, 0x02eb, 0x02eb}, {0x0033, 0x0033, 0x0323, 0x0323}, {0x0271, 0x0329, 0x0325, 0x032d}, {0x02f1, 0x0301, 0x02b1, 0x0331}}; const USHORT HuffmanCodeBook_11[152][4] = { {0x0004, 0x0010, 0x0038, 0x008c}, {0x0001, 0x0085, 0x0008, 0x000c}, {0x0843, 0x0843, 0x0007, 0x0007}, {0x0083, 0x0083, 0x008b, 0x008b}, {0x0014, 0x0018, 0x001c, 0x0024}, {0x0107, 0x0107, 0x010b, 0x010b}, {0x0185, 0x008d, 0x010d, 0x0009}, {0x0189, 0x0101, 0x018d, 0x0020}, {0x0093, 0x0093, 0x0207, 0x0207}, {0x0028, 0x002c, 0x0030, 0x0034}, {0x0113, 0x0113, 0x020b, 0x020b}, {0x0193, 0x0193, 0x020f, 0x020f}, {0x000f, 0x000f, 0x0183, 0x0183}, {0x0097, 0x0097, 0x0117, 0x0117}, {0x003c, 0x0050, 0x0064, 0x0078}, {0x0040, 0x0044, 0x0048, 0x004c}, {0x028b, 0x028b, 0x0213, 0x0213}, {0x0287, 0x0287, 0x0197, 0x0197}, {0x028f, 0x028f, 0x0217, 0x0217}, {0x0291, 0x0119, 0x0309, 0x0099}, {0x0054, 0x0058, 0x005c, 0x0060}, {0x0199, 0x030d, 0x0305, 0x0811}, {0x080d, 0x02c1, 0x01c1, 0x0241}, {0x0219, 0x0341, 0x0011, 0x0311}, {0x0201, 0x0809, 0x0295, 0x0815}, {0x0068, 0x006c, 0x0070, 0x0074}, {0x03c1, 0x0141, 0x0441, 0x0389}, {0x011d, 0x038d, 0x0299, 0x0315}, {0x0819, 0x0541, 0x019d, 0x009d}, {0x04c1, 0x081d, 0x0805, 0x0385}, {0x007c, 0x0080, 0x0084, 0x0088}, {0x0391, 0x05c1, 0x021d, 0x0641}, {0x0821, 0x00c1, 0x0319, 0x0825}, {0x0409, 0x0395, 0x0829, 0x06c1}, {0x01a1, 0x0121, 0x040d, 0x0015}, {0x0090, 0x00c8, 0x011c, 0x0170}, {0x0094, 0x0098, 0x00a0, 0x00b4}, {0x0741, 0x082d, 0x029d, 0x0411}, {0x0399, 0x031d, 0x0281, 0x009c}, {0x0223, 0x0223, 0x07c3, 0x07c3}, {0x00a4, 0x00a8, 0x00ac, 0x00b0}, {0x0833, 0x0833, 0x0407, 0x0407}, {0x00a3, 0x00a3, 0x083b, 0x083b}, {0x0417, 0x0417, 0x0837, 0x0837}, {0x048f, 0x048f, 0x02a3, 0x02a3}, {0x00b8, 0x00bc, 0x00c0, 0x00c4}, {0x039f, 0x039f, 0x048b, 0x048b}, {0x0323, 0x0323, 0x0127, 0x0127}, {0x01a7, 0x01a7, 0x083f, 0x083f}, {0x0493, 0x0493, 0x041b, 0x041b}, {0x00cc, 0x00e0, 0x00f4, 0x0108}, {0x00d0, 0x00d4, 0x00d8, 0x00dc}, {0x001b, 0x001b, 0x0227, 0x0227}, {0x0497, 0x0497, 0x03a3, 0x03a3}, {0x041f, 0x041f, 0x0487, 0x0487}, {0x01ab, 0x01ab, 0x0303, 0x0303}, {0x00e4, 0x00e8, 0x00ec, 0x00f0}, {0x012b, 0x012b, 0x00a7, 0x00a7}, {0x02a7, 0x02a7, 0x0513, 0x0513}, {0x050b, 0x050b, 0x0327, 0x0327}, {0x050f, 0x050f, 0x049b, 0x049b}, {0x00f8, 0x00fc, 0x0100, 0x0104}, {0x022b, 0x022b, 0x0423, 0x0423}, {0x02ab, 0x02ab, 0x03a7, 0x03a7}, {0x01af, 0x01af, 0x0507, 0x0507}, {0x001f, 0x001f, 0x032b, 0x032b}, {0x010c, 0x0110, 0x0114, 0x0118}, {0x049f, 0x049f, 0x058f, 0x058f}, {0x0517, 0x0517, 0x00ab, 0x00ab}, {0x0593, 0x0593, 0x012f, 0x012f}, {0x0137, 0x0137, 0x051b, 0x051b}, {0x0120, 0x0134, 0x0148, 0x015c}, {0x0124, 0x0128, 0x012c, 0x0130}, {0x01b7, 0x01b7, 0x058b, 0x058b}, {0x0043, 0x0043, 0x0597, 0x0597}, {0x02af, 0x02af, 0x022d, 0x0425}, {0x051d, 0x04a1, 0x0801, 0x0691}, {0x0138, 0x013c, 0x0140, 0x0144}, {0x0381, 0x068d, 0x032d, 0x00b5}, {0x0235, 0x01b1, 0x0689, 0x02b5}, {0x0521, 0x0599, 0x0429, 0x03a9}, {0x0139, 0x0231, 0x0585, 0x0611}, {0x014c, 0x0150, 0x0154, 0x0158}, {0x00ad, 0x060d, 0x0685, 0x0131}, {0x059d, 0x070d, 0x0615, 0x0695}, {0x0239, 0x0711, 0x03ad, 0x01b9}, {0x02b1, 0x0335, 0x0331, 0x0021}, {0x0160, 0x0164, 0x0168, 0x016c}, {0x042d, 0x0609, 0x04a5, 0x02b9}, {0x0699, 0x0529, 0x013d, 0x05a1}, {0x0525, 0x0339, 0x04a9, 0x0715}, {0x04ad, 0x00b9, 0x0709, 0x0619}, {0x0174, 0x0188, 0x019c, 0x01cc}, {0x0178, 0x017c, 0x0180, 0x0184}, {0x0605, 0x0435, 0x0401, 0x03b5}, {0x061d, 0x03b1, 0x069d, 0x01bd}, {0x00b1, 0x0719, 0x0789, 0x02bd}, {0x023d, 0x0705, 0x05a5, 0x0791}, {0x018c, 0x0190, 0x0194, 0x0198}, {0x03b9, 0x06a1, 0x04b5, 0x0621}, {0x0795, 0x078d, 0x05a9, 0x052d}, {0x0431, 0x033d, 0x03bd, 0x0721}, {0x00bd, 0x071d, 0x0025, 0x0481}, {0x01a0, 0x01a4, 0x01a8, 0x01b8}, {0x06a5, 0x0625, 0x04b1, 0x0439}, {0x06a9, 0x04b9, 0x0531, 0x0799}, {0x079d, 0x01ac, 0x01b0, 0x01b4}, {0x0727, 0x0727, 0x043f, 0x043f}, {0x05af, 0x05af, 0x072f, 0x072f}, {0x0787, 0x0787, 0x062b, 0x062b}, {0x01bc, 0x01c0, 0x01c4, 0x01c8}, {0x072b, 0x072b, 0x05b7, 0x05b7}, {0x0537, 0x0537, 0x06af, 0x06af}, {0x062f, 0x062f, 0x07a3, 0x07a3}, {0x05bb, 0x05bb, 0x0637, 0x0637}, {0x01d0, 0x01e4, 0x01f8, 0x020c}, {0x01d4, 0x01d8, 0x01dc, 0x01e0}, {0x06b3, 0x06b3, 0x04bf, 0x04bf}, {0x053b, 0x053b, 0x002b, 0x002b}, {0x05b3, 0x05b3, 0x07a7, 0x07a7}, {0x0503, 0x0503, 0x0633, 0x0633}, {0x01e8, 0x01ec, 0x01f0, 0x01f4}, {0x002f, 0x002f, 0x0733, 0x0733}, {0x07ab, 0x07ab, 0x06b7, 0x06b7}, {0x0683, 0x0683, 0x063b, 0x063b}, {0x053f, 0x053f, 0x05bf, 0x05bf}, {0x01fc, 0x0200, 0x0204, 0x0208}, {0x07af, 0x07af, 0x06bb, 0x06bb}, {0x0037, 0x0037, 0x0583, 0x0583}, {0x0737, 0x0737, 0x063f, 0x063f}, {0x06bf, 0x06bf, 0x07b3, 0x07b3}, {0x0210, 0x0214, 0x0218, 0x021c}, {0x003b, 0x003b, 0x073b, 0x073b}, {0x07b7, 0x07b7, 0x0033, 0x0033}, {0x07bb, 0x07bb, 0x0701, 0x0601}, {0x073d, 0x003d, 0x0781, 0x07bd}, {0x0118, 0x0117, 0x0100, 0x0109}, {0x05a5, 0x05a1, 0x05b7, 0x0513}, {0x08f9, 0x08ff, 0x0821, 0x08ff}, {0x084f, 0x08ff, 0x08bc, 0x08ff}, {0x0815, 0x08ff, 0x0837, 0x08ff}, {0x080d, 0x08ff, 0x085f, 0x08ff}, {0x084a, 0x08ff, 0x087d, 0x08ff}, {0x08ff, 0x08ff, 0x08a8, 0x08ff}, {0x0815, 0x08ff, 0x083f, 0x08ff}, {0x0830, 0x08ff, 0x0894, 0x08ff}, {0x08d4, 0x08ff, 0x0825, 0x08ff}, {0x08ef, 0x08ff, 0x083f, 0x08ff}, {0x0809, 0x08ff, 0x08fc, 0x08ff}, {0x0842, 0x08ff, 0x08b3, 0x08ff}, {0x070d, 0x07a9, 0x060e, 0x06e2}, {0x06c7, 0x06d0, 0x04b2, 0x0407}}; const USHORT HuffmanCodeBook_SCL[65][4] = { {0x00f3, 0x00f3, 0x0004, 0x0008}, {0x00ef, 0x00ef, 0x00f5, 0x00e9}, {0x00f9, 0x000c, 0x0010, 0x0014}, {0x00e7, 0x00e7, 0x00ff, 0x00ff}, {0x00e1, 0x0101, 0x00dd, 0x0105}, {0x0018, 0x001c, 0x0020, 0x0028}, {0x010b, 0x010b, 0x00db, 0x00db}, {0x010f, 0x010f, 0x00d5, 0x0111}, {0x00d1, 0x0115, 0x00cd, 0x0024}, {0x011b, 0x011b, 0x00cb, 0x00cb}, {0x002c, 0x0030, 0x0034, 0x0040}, {0x00c7, 0x00c7, 0x011f, 0x011f}, {0x0121, 0x00c1, 0x0125, 0x00bd}, {0x0129, 0x00b9, 0x0038, 0x003c}, {0x0133, 0x0133, 0x012f, 0x012f}, {0x0137, 0x0137, 0x013b, 0x013b}, {0x0044, 0x0048, 0x004c, 0x0058}, {0x00b7, 0x00b7, 0x00af, 0x00af}, {0x00b1, 0x013d, 0x00a9, 0x00a5}, {0x0141, 0x00a1, 0x0050, 0x0054}, {0x0147, 0x0147, 0x009f, 0x009f}, {0x014b, 0x014b, 0x009b, 0x009b}, {0x005c, 0x0060, 0x0064, 0x0070}, {0x014f, 0x014f, 0x0095, 0x008d}, {0x0155, 0x0085, 0x0091, 0x0089}, {0x0151, 0x0081, 0x0068, 0x006c}, {0x015f, 0x015f, 0x0167, 0x0167}, {0x007b, 0x007b, 0x007f, 0x007f}, {0x0074, 0x0078, 0x0080, 0x00b0}, {0x0159, 0x0075, 0x0069, 0x006d}, {0x0071, 0x0061, 0x0161, 0x007c}, {0x0067, 0x0067, 0x005b, 0x005b}, {0x0084, 0x0088, 0x008c, 0x009c}, {0x005f, 0x005f, 0x0169, 0x0055}, {0x004d, 0x000d, 0x0005, 0x0009}, {0x0001, 0x0090, 0x0094, 0x0098}, {0x018b, 0x018b, 0x018f, 0x018f}, {0x0193, 0x0193, 0x0197, 0x0197}, {0x019b, 0x019b, 0x01d7, 0x01d7}, {0x00a0, 0x00a4, 0x00a8, 0x00ac}, {0x0187, 0x0187, 0x016f, 0x016f}, {0x0173, 0x0173, 0x0177, 0x0177}, {0x017b, 0x017b, 0x017f, 0x017f}, {0x0183, 0x0183, 0x01a3, 0x01a3}, {0x00b4, 0x00c8, 0x00dc, 0x00f0}, {0x00b8, 0x00bc, 0x00c0, 0x00c4}, {0x01bf, 0x01bf, 0x01c3, 0x01c3}, {0x01c7, 0x01c7, 0x01cb, 0x01cb}, {0x01cf, 0x01cf, 0x01d3, 0x01d3}, {0x01bb, 0x01bb, 0x01a7, 0x01a7}, {0x00cc, 0x00d0, 0x00d4, 0x00d8}, {0x01ab, 0x01ab, 0x01af, 0x01af}, {0x01b3, 0x01b3, 0x01b7, 0x01b7}, {0x01db, 0x01db, 0x001b, 0x001b}, {0x0023, 0x0023, 0x0027, 0x0027}, {0x00e0, 0x00e4, 0x00e8, 0x00ec}, {0x002b, 0x002b, 0x0017, 0x0017}, {0x019f, 0x019f, 0x01e3, 0x01e3}, {0x01df, 0x01df, 0x0013, 0x0013}, {0x001f, 0x001f, 0x003f, 0x003f}, {0x00f4, 0x00f8, 0x00fc, 0x0100}, {0x0043, 0x0043, 0x004b, 0x004b}, {0x0053, 0x0053, 0x0047, 0x0047}, {0x002f, 0x002f, 0x0033, 0x0033}, {0x003b, 0x003b, 0x0037, 0x0037}}; /* .CodeBook = HuffmanCodeBook_x, .Dimension = 4, .numBits = 2, .Offset = 0 */ const CodeBookDescription AACcodeBookDescriptionTable[13] = { {NULL, 0, 0, 0}, {HuffmanCodeBook_1, 4, 2, 1}, {HuffmanCodeBook_2, 4, 2, 1}, {HuffmanCodeBook_3, 4, 2, 0}, {HuffmanCodeBook_4, 4, 2, 0}, {HuffmanCodeBook_5, 2, 4, 4}, {HuffmanCodeBook_6, 2, 4, 4}, {HuffmanCodeBook_7, 2, 4, 0}, {HuffmanCodeBook_8, 2, 4, 0}, {HuffmanCodeBook_9, 2, 4, 0}, {HuffmanCodeBook_10, 2, 4, 0}, {HuffmanCodeBook_11, 2, 5, 0}, {HuffmanCodeBook_SCL, 1, 8, 60}}; const CodeBookDescription AACcodeBookDescriptionSCL = {HuffmanCodeBook_SCL, 1, 8, 60}; /* ********************************************************************************************* */ /* Table: HuffTree41 */ /* --------------------------------------------------------------------------------------------- */ /* description: This table contains the decode tree for spectral data * (Codebook 1). */ /* bit 23 and 11 not used */ /* bit 22 and 10 determine end value */ /* bit 21-12 and 9-0 (offset to next node) or (index value * * 4) */ /* --------------------------------------------------------------------------------------------- */ /* input: codeword */ /* --------------------------------------------------------------------------------------------- */ /* output: index * 4 */ /* --------------------------------------------------------------------------------------------- */ /* HuffTree */ const UINT aHuffTree41[80] = { 0x4a0001, 0x026002, 0x013003, 0x021004, 0x01c005, 0x00b006, 0x010007, 0x019008, 0x00900e, 0x00a03a, 0x400528, 0x00c037, 0x00d03b, 0x454404, 0x00f04c, 0x448408, 0x017011, 0x01202e, 0x42c40c, 0x034014, 0x01502c, 0x016049, 0x410470, 0x01804e, 0x414424, 0x03201a, 0x02001b, 0x520418, 0x02f01d, 0x02a01e, 0x01f04d, 0x41c474, 0x540420, 0x022024, 0x04a023, 0x428510, 0x025029, 0x430508, 0x02703c, 0x028047, 0x50c434, 0x438478, 0x04802b, 0x46443c, 0x02d03e, 0x4404b0, 0x44451c, 0x03003f, 0x03104b, 0x52444c, 0x033039, 0x4f0450, 0x035041, 0x036046, 0x4e8458, 0x04f038, 0x45c53c, 0x4604e0, 0x4f8468, 0x46c4d4, 0x04503d, 0x4ac47c, 0x518480, 0x043040, 0x4844dc, 0x042044, 0x4884a8, 0x4bc48c, 0x530490, 0x4a4494, 0x4984b8, 0x49c4c4, 0x5044b4, 0x5004c0, 0x4d04c8, 0x4f44cc, 0x4d8538, 0x4ec4e4, 0x52c4fc, 0x514534}; /* ********************************************************************************************* */ /* Table: HuffTree42 */ /* --------------------------------------------------------------------------------------------- */ /* description: This table contains the decode tree for spectral data * (Codebook 2). */ /* bit 23 and 11 not used */ /* bit 22 and 10 determine end value */ /* bit 21-12 and 9-0 (offset to next node) or (index value * * 4) */ /* --------------------------------------------------------------------------------------------- */ /* input: codeword */ /* --------------------------------------------------------------------------------------------- */ /* output: index * 4 */ /* --------------------------------------------------------------------------------------------- */ const UINT aHuffTree42[80] = { 0x026001, 0x014002, 0x009003, 0x010004, 0x01d005, 0x00600d, 0x007018, 0x450008, 0x4e0400, 0x02e00a, 0x03900b, 0x03d00c, 0x43c404, 0x01b00e, 0x00f04f, 0x4d8408, 0x023011, 0x01203b, 0x01a013, 0x41440c, 0x015020, 0x016040, 0x025017, 0x500410, 0x038019, 0x540418, 0x41c444, 0x02d01c, 0x420520, 0x01e042, 0x03701f, 0x4244cc, 0x02a021, 0x02204c, 0x478428, 0x024031, 0x42c4dc, 0x4304e8, 0x027033, 0x4a0028, 0x50c029, 0x4344a4, 0x02c02b, 0x470438, 0x4404c8, 0x4f8448, 0x04902f, 0x04b030, 0x44c484, 0x524032, 0x4ec454, 0x03e034, 0x035046, 0x4c4036, 0x488458, 0x4d445c, 0x460468, 0x04e03a, 0x51c464, 0x03c04a, 0x46c514, 0x47453c, 0x04503f, 0x47c4ac, 0x044041, 0x510480, 0x04304d, 0x4e448c, 0x490518, 0x49449c, 0x048047, 0x4c0498, 0x4b84a8, 0x4b0508, 0x4fc4b4, 0x4bc504, 0x5304d0, 0x5344f0, 0x4f452c, 0x528538}; /* ********************************************************************************************* */ /* Table: HuffTree43 */ /* --------------------------------------------------------------------------------------------- */ /* description: This table contains the decode tree for spectral data * (Codebook 3). */ /* bit 23 and 11 not used */ /* bit 22 and 10 determine end value */ /* bit 21-12 and 9-0 (offset to next node) or (index value * * 4) */ /* --------------------------------------------------------------------------------------------- */ /* input: codeword */ /* --------------------------------------------------------------------------------------------- */ /* output: index * 4 */ /* --------------------------------------------------------------------------------------------- */ const UINT aHuffTree43[80] = { 0x400001, 0x002004, 0x00300a, 0x46c404, 0x00b005, 0x00600d, 0x034007, 0x037008, 0x494009, 0x4d8408, 0x42440c, 0x00c01b, 0x490410, 0x00e016, 0x00f011, 0x010014, 0x4144fc, 0x01201d, 0x020013, 0x508418, 0x4c0015, 0x41c440, 0x022017, 0x018026, 0x019035, 0x03801a, 0x420444, 0x01c01f, 0x430428, 0x02101e, 0x44842c, 0x478434, 0x4b4438, 0x45443c, 0x02c023, 0x039024, 0x02503f, 0x48844c, 0x030027, 0x02e028, 0x032029, 0x02a041, 0x4d402b, 0x4504f0, 0x04302d, 0x4584a8, 0x02f03b, 0x46045c, 0x03103d, 0x464046, 0x033044, 0x46853c, 0x47049c, 0x045036, 0x4744dc, 0x4a047c, 0x500480, 0x4ac03a, 0x4b8484, 0x03c04e, 0x48c524, 0x03e040, 0x4984e8, 0x50c4a4, 0x4b0530, 0x042047, 0x4bc04b, 0x4e44c4, 0x5184c8, 0x52c4cc, 0x5204d0, 0x04d048, 0x04a049, 0x4e004c, 0x51c4ec, 0x4f4510, 0x5284f8, 0x50404f, 0x514538, 0x540534}; /* ********************************************************************************************* */ /* Table: HuffTree44 */ /* --------------------------------------------------------------------------------------------- */ /* description: This table contains the decode tree for spectral data * (Codebook 4). */ /* bit 23 and 11 not used */ /* bit 22 and 10 determine end value */ /* bit 21-12 and 9-0 (offset to next node) or (index value * * 4) */ /* --------------------------------------------------------------------------------------------- */ /* input: codeword */ /* --------------------------------------------------------------------------------------------- */ /* output: index * 4 */ /* --------------------------------------------------------------------------------------------- */ const UINT aHuffTree44[80] = { 0x001004, 0x020002, 0x036003, 0x490400, 0x005008, 0x010006, 0x01f007, 0x404428, 0x00e009, 0x01100a, 0x00b018, 0x01600c, 0x03700d, 0x408015, 0x00f03e, 0x40c424, 0x410478, 0x022012, 0x038013, 0x01e014, 0x454414, 0x448418, 0x025017, 0x47441c, 0x030019, 0x02601a, 0x02d01b, 0x01c034, 0x01d029, 0x4204f0, 0x4dc42c, 0x470430, 0x02103c, 0x4a0434, 0x02302a, 0x440024, 0x4384a8, 0x43c44c, 0x02703a, 0x02802c, 0x444524, 0x4504e0, 0x02b03d, 0x458480, 0x45c4f4, 0x04b02e, 0x04f02f, 0x460520, 0x042031, 0x048032, 0x049033, 0x514464, 0x03504c, 0x540468, 0x47c46c, 0x4844d8, 0x039044, 0x4884fc, 0x03b045, 0x48c53c, 0x49449c, 0x4b8498, 0x03f046, 0x041040, 0x4c44a4, 0x50c4ac, 0x04a043, 0x5184b0, 0x4e44b4, 0x4bc4ec, 0x04e047, 0x4c04e8, 0x4c8510, 0x4cc52c, 0x4d0530, 0x5044d4, 0x53804d, 0x5284f8, 0x508500, 0x51c534}; /* ********************************************************************************************* */ /* Table: HuffTree21 */ /* --------------------------------------------------------------------------------------------- */ /* description: This table contains the decode tree for spectral data * (Codebook 5). */ /* bit 23 and 11 not used */ /* bit 22 and 10 determine end value */ /* bit 21-12 and 9-0 (offset to next node) or (index value * * 2) */ /* --------------------------------------------------------------------------------------------- */ /* input: codeword */ /* --------------------------------------------------------------------------------------------- */ /* output: index * 2 */ /* --------------------------------------------------------------------------------------------- */ const UINT aHuffTree21[80] = { 0x450001, 0x044002, 0x042003, 0x035004, 0x026005, 0x022006, 0x013007, 0x010008, 0x00d009, 0x01c00a, 0x01f00b, 0x01e00c, 0x4a0400, 0x01b00e, 0x03200f, 0x47e402, 0x020011, 0x01204d, 0x40449c, 0x017014, 0x015019, 0x01603f, 0x406458, 0x01804f, 0x448408, 0x04901a, 0x40a45a, 0x48c40c, 0x01d031, 0x40e48e, 0x490410, 0x492412, 0x021030, 0x480414, 0x033023, 0x02402e, 0x02503e, 0x416482, 0x02a027, 0x02802c, 0x029040, 0x418468, 0x02b04a, 0x41a486, 0x02d048, 0x41c484, 0x04e02f, 0x41e426, 0x420434, 0x42249e, 0x424494, 0x03d034, 0x428470, 0x039036, 0x03703b, 0x038041, 0x42a476, 0x03a04b, 0x42c454, 0x03c047, 0x42e472, 0x430478, 0x43246e, 0x496436, 0x488438, 0x43a466, 0x046043, 0x43c464, 0x04504c, 0x43e462, 0x460440, 0x44245e, 0x45c444, 0x46a446, 0x44a456, 0x47444c, 0x45244e, 0x46c47c, 0x48a47a, 0x49a498}; /* ********************************************************************************************* */ /* Table: HuffTree22 */ /* --------------------------------------------------------------------------------------------- */ /* description: This table contains the decode tree for spectral data * (Codebook 6). */ /* bit 23 and 11 not used */ /* bit 22 and 10 determine end value */ /* bit 21-12 and 9-0 (offset to next node) or (index value * * 2) */ /* --------------------------------------------------------------------------------------------- */ /* input: codeword */ /* --------------------------------------------------------------------------------------------- */ /* output: index * 2 */ /* --------------------------------------------------------------------------------------------- */ const UINT aHuffTree22[80] = { 0x03c001, 0x02f002, 0x020003, 0x01c004, 0x00f005, 0x00c006, 0x016007, 0x04d008, 0x00b009, 0x01500a, 0x400490, 0x40e402, 0x00d013, 0x00e02a, 0x40c404, 0x019010, 0x011041, 0x038012, 0x40a406, 0x014037, 0x40849c, 0x4a0410, 0x04a017, 0x458018, 0x412422, 0x02801a, 0x01b029, 0x480414, 0x02401d, 0x01e02b, 0x48a01f, 0x416432, 0x02d021, 0x026022, 0x023039, 0x418468, 0x025043, 0x48641a, 0x027040, 0x41c488, 0x41e48c, 0x42045a, 0x47c424, 0x04c02c, 0x46e426, 0x03602e, 0x428478, 0x030033, 0x43c031, 0x04b032, 0x42e42a, 0x03403a, 0x035048, 0x42c442, 0x470430, 0x494434, 0x43649a, 0x45c438, 0x04403b, 0x43a454, 0x04503d, 0x03e03f, 0x43e464, 0x440460, 0x484444, 0x049042, 0x446448, 0x44a456, 0x46644c, 0x047046, 0x44e452, 0x450462, 0x47445e, 0x46a496, 0x49846c, 0x472476, 0x47a482, 0x04e04f, 0x47e492, 0x48e49e}; /* ********************************************************************************************* */ /* Table: HuffTree23 */ /* --------------------------------------------------------------------------------------------- */ /* description: This table contains the decode tree for spectral data * (Codebook 7). */ /* bit 23 and 11 not used */ /* bit 22 and 10 determine end value */ /* bit 21-12 and 9-0 (offset to next node) or (index value * * 2) */ /* --------------------------------------------------------------------------------------------- */ /* input: codeword */ /* --------------------------------------------------------------------------------------------- */ /* output: index * 2 */ /* --------------------------------------------------------------------------------------------- */ const UINT aHuffTree23[63] = { 0x400001, 0x002003, 0x410402, 0x004007, 0x412005, 0x01c006, 0x420404, 0x00800b, 0x01d009, 0x00a01f, 0x406026, 0x00c012, 0x00d00f, 0x02700e, 0x408440, 0x010022, 0x028011, 0x45440a, 0x013017, 0x029014, 0x024015, 0x01602f, 0x43c40c, 0x02b018, 0x019033, 0x03201a, 0x43e01b, 0x47040e, 0x422414, 0x01e025, 0x432416, 0x020021, 0x418442, 0x41a452, 0x036023, 0x41c446, 0x46441e, 0x424430, 0x426434, 0x436428, 0x44442a, 0x02e02a, 0x45642c, 0x03002c, 0x02d03b, 0x46642e, 0x43a438, 0x460448, 0x031037, 0x47244a, 0x45a44c, 0x034039, 0x038035, 0x47844e, 0x462450, 0x474458, 0x46a45c, 0x03a03c, 0x45e47a, 0x476468, 0x03d03e, 0x47c46c, 0x46e47e}; /* ********************************************************************************************* */ /* Table: HuffTree24 */ /* --------------------------------------------------------------------------------------------- */ /* description: This table contains the decode tree for spectral data * (Codebook 8). */ /* bit 23 and 11 not used */ /* bit 22 and 10 determine end value */ /* bit 21-12 and 9-0 (offset to next node) or (index value * * 2) */ /* --------------------------------------------------------------------------------------------- */ /* input: codeword */ /* --------------------------------------------------------------------------------------------- */ /* output: index * 2 */ /* --------------------------------------------------------------------------------------------- */ const UINT aHuffTree24[63] = { 0x001006, 0x01d002, 0x005003, 0x424004, 0x400420, 0x414402, 0x00700a, 0x008020, 0x00901f, 0x404432, 0x00b011, 0x00c00e, 0x00d032, 0x406446, 0x02300f, 0x033010, 0x458408, 0x025012, 0x013016, 0x01402f, 0x015038, 0x46840a, 0x028017, 0x01801a, 0x039019, 0x40c47a, 0x03e01b, 0x03b01c, 0x40e47e, 0x41201e, 0x422410, 0x416434, 0x02a021, 0x02202b, 0x418444, 0x02c024, 0x41a456, 0x02d026, 0x027034, 0x46241c, 0x029036, 0x41e45c, 0x426031, 0x428430, 0x45242a, 0x03702e, 0x42c464, 0x03003c, 0x47442e, 0x436442, 0x438454, 0x43a448, 0x03503a, 0x43c466, 0x43e03d, 0x44a440, 0x44c472, 0x46044e, 0x45a450, 0x45e470, 0x46a476, 0x46c478, 0x47c46e}; /* ********************************************************************************************* */ /* Table: HuffTree25 */ /* --------------------------------------------------------------------------------------------- */ /* description: This table contains the decode tree for spectral data * (Codebook 9). */ /* bit 23 and 11 not used */ /* bit 22 and 10 determine end value */ /* bit 21-12 and 9-0 (offset to next node) or (index value * * 2) */ /* --------------------------------------------------------------------------------------------- */ /* input: codeword */ /* --------------------------------------------------------------------------------------------- */ /* output: index * 2 */ /* --------------------------------------------------------------------------------------------- */ const UINT aHuffTree25[168] = { 0x400001, 0x002003, 0x41a402, 0x004007, 0x41c005, 0x035006, 0x434404, 0x008010, 0x00900c, 0x04a00a, 0x42000b, 0x44e406, 0x03600d, 0x03800e, 0x05a00f, 0x408468, 0x01101a, 0x012016, 0x039013, 0x070014, 0x46e015, 0x40a440, 0x03b017, 0x01804d, 0x01904f, 0x4b840c, 0x01b022, 0x01c041, 0x03f01d, 0x01e020, 0x01f05b, 0x40e4ee, 0x02107c, 0x45c410, 0x02302c, 0x024028, 0x053025, 0x026045, 0x02707d, 0x412522, 0x047029, 0x05e02a, 0x02b08a, 0x526414, 0x05602d, 0x02e081, 0x02f032, 0x06e030, 0x031080, 0x416544, 0x079033, 0x034091, 0x41852c, 0x43641e, 0x04b037, 0x42246a, 0x43c424, 0x04c03a, 0x426456, 0x03c066, 0x03d03e, 0x482428, 0x45842a, 0x040072, 0x42c4ba, 0x050042, 0x04305c, 0x044074, 0x42e4be, 0x06a046, 0x4dc430, 0x075048, 0x0490a3, 0x44a432, 0x450438, 0x43a452, 0x48443e, 0x04e068, 0x45a442, 0x4d4444, 0x051088, 0x052087, 0x44648c, 0x077054, 0x4da055, 0x50a448, 0x057060, 0x06b058, 0x05906d, 0x44c4f6, 0x46c454, 0x45e474, 0x06905d, 0x460520, 0x05f07e, 0x462494, 0x061063, 0x07f062, 0x464496, 0x06408b, 0x08d065, 0x542466, 0x067071, 0x4d2470, 0x4724ec, 0x478476, 0x53a47a, 0x09b06c, 0x47c4ac, 0x4f847e, 0x06f078, 0x510480, 0x48649e, 0x4884a0, 0x07307b, 0x49c48a, 0x4a648e, 0x098076, 0x4904c0, 0x4924ea, 0x4c8498, 0x07a08e, 0x51249a, 0x4a24d6, 0x5064a4, 0x4f24a8, 0x4aa4de, 0x51e4ae, 0x4b0538, 0x082092, 0x083085, 0x08f084, 0x5464b2, 0x096086, 0x4ce4b4, 0x4d04b6, 0x089090, 0x4bc508, 0x4c253e, 0x08c0a4, 0x5284c4, 0x4e04c6, 0x4ca4fa, 0x5144cc, 0x4f04d8, 0x4e24fc, 0x09309c, 0x094099, 0x095097, 0x4e4516, 0x4e652e, 0x4e84fe, 0x4f450c, 0x09a09f, 0x500502, 0x50450e, 0x09d0a0, 0x09e0a5, 0x518530, 0x51a54a, 0x0a70a1, 0x0a20a6, 0x51c534, 0x53c524, 0x54052a, 0x548532, 0x536550, 0x54c54e}; /* ********************************************************************************************* */ /* Table: HuffTree26 */ /* --------------------------------------------------------------------------------------------- */ /* description: This table contains the decode tree for spectral data * (Codebook 10). */ /* bit 23 and 11 not used */ /* bit 22 and 10 determine end value */ /* bit 21-12 and 9-0 (offset to next node) or (index value * * 2) */ /* --------------------------------------------------------------------------------------------- */ /* input: codeword */ /* --------------------------------------------------------------------------------------------- */ /* output: index * 2 */ /* --------------------------------------------------------------------------------------------- */ const UINT aHuffTree26[168] = { 0x006001, 0x002013, 0x00300f, 0x00400d, 0x03b005, 0x40046e, 0x037007, 0x00800a, 0x009067, 0x402420, 0x05600b, 0x00c057, 0x434404, 0x06600e, 0x406470, 0x03c010, 0x059011, 0x06f012, 0x49e408, 0x014019, 0x03f015, 0x016044, 0x017042, 0x079018, 0x4b840a, 0x01a01f, 0x01b047, 0x07c01c, 0x08701d, 0x06901e, 0x44640c, 0x020027, 0x04b021, 0x02204f, 0x023025, 0x02406b, 0x40e4e0, 0x081026, 0x528410, 0x02802c, 0x06c029, 0x08f02a, 0x02b078, 0x53a412, 0x05202d, 0x02e033, 0x02f031, 0x0300a2, 0x4144ce, 0x0a6032, 0x416534, 0x09a034, 0x09f035, 0x0360a7, 0x54e418, 0x03a038, 0x436039, 0x43841a, 0x41c41e, 0x42246a, 0x05803d, 0x03e068, 0x424484, 0x04005b, 0x04107a, 0x42645a, 0x043093, 0x4d2428, 0x05e045, 0x046072, 0x42a45e, 0x048060, 0x073049, 0x04a098, 0x42c4c4, 0x07504c, 0x09504d, 0x04e09c, 0x51042e, 0x063050, 0x077051, 0x43053c, 0x053084, 0x065054, 0x4e4055, 0x4fe432, 0x43a454, 0x43c46c, 0x43e486, 0x07005a, 0x4a0440, 0x07105c, 0x05d07b, 0x45c442, 0x05f08a, 0x476444, 0x07f061, 0x06206a, 0x448506, 0x06408e, 0x52644a, 0x54444c, 0x45644e, 0x452450, 0x488458, 0x4604ec, 0x4624f6, 0x50e464, 0x08206d, 0x0a406e, 0x542466, 0x4a2468, 0x48a472, 0x474089, 0x4d8478, 0x097074, 0x47a508, 0x08d076, 0x47c4b6, 0x51247e, 0x4804fc, 0x4bc482, 0x48c4a4, 0x48e4d4, 0x07d07e, 0x4904da, 0x49208b, 0x094080, 0x49450c, 0x4964e2, 0x09d083, 0x52a498, 0x085091, 0x0a5086, 0x4cc49a, 0x08808c, 0x4ee49c, 0x4a64ba, 0x4a84c0, 0x4c24aa, 0x4ac4f0, 0x4ae4d0, 0x4ca4b0, 0x0900a1, 0x4b24ea, 0x092099, 0x4b4516, 0x4d64be, 0x4c650a, 0x522096, 0x4c8524, 0x4dc4f2, 0x4de4f4, 0x4e6548, 0x09e09b, 0x5384e8, 0x5204f8, 0x4fa53e, 0x50051a, 0x0a30a0, 0x502536, 0x514504, 0x51e518, 0x54a51c, 0x54052c, 0x52e546, 0x530532, 0x54c550}; /* ********************************************************************************************* */ /* Table: HuffTree27 */ /* --------------------------------------------------------------------------------------------- */ /* description: This table contains the decode tree for spectral data * (Codebook 11). */ /* bit 23 and 11 not used */ /* bit 22 and 10 determine end value */ /* bit 21-12 and 9-0 (offset to next node) or (index value * * 2) */ /* --------------------------------------------------------------------------------------------- */ /* input: codeword */ /* --------------------------------------------------------------------------------------------- */ /* output: index * 2 */ /* --------------------------------------------------------------------------------------------- */ const UINT aHuffTree27[288] = { 0x00100d, 0x002006, 0x003004, 0x400424, 0x047005, 0x402446, 0x048007, 0x00800a, 0x00904c, 0x44a404, 0x07400b, 0x00c0bb, 0x466406, 0x00e014, 0x00f054, 0x04e010, 0x051011, 0x0a9012, 0x0130bc, 0x408464, 0x01501f, 0x01601a, 0x017059, 0x0af018, 0x0ca019, 0x40a0e4, 0x01b05e, 0x01c084, 0x0bf01d, 0x05d01e, 0x55a40c, 0x020026, 0x021066, 0x043022, 0x023062, 0x02408d, 0x025108, 0x40e480, 0x027030, 0x02802c, 0x02906b, 0x02a0da, 0x06502b, 0x4105c8, 0x0a402d, 0x0ec02e, 0x0dd02f, 0x532412, 0x06e031, 0x032036, 0x03303e, 0x0fd034, 0x0fc035, 0x4145b0, 0x03703a, 0x038117, 0x10d039, 0x5ba416, 0x10f03b, 0x03c041, 0x5fa03d, 0x41c418, 0x10403f, 0x04011d, 0x41a5f4, 0x11c042, 0x41e61c, 0x087044, 0x0f5045, 0x0d9046, 0x4204a2, 0x640422, 0x04904a, 0x426448, 0x04b073, 0x428468, 0x46c04d, 0x48a42a, 0x04f077, 0x076050, 0x42c4b0, 0x0520a7, 0x096053, 0x42e4a8, 0x05507d, 0x07a056, 0x0d4057, 0x0df058, 0x442430, 0x05a081, 0x05b09b, 0x05c0e2, 0x5b8432, 0x4fe434, 0x05f09e, 0x0e6060, 0x0610d6, 0x57c436, 0x0cc063, 0x112064, 0x4384a0, 0x43a5ca, 0x067089, 0x0680b7, 0x0690a2, 0x0a106a, 0x43c59c, 0x09206c, 0x06d0ba, 0x60643e, 0x0d106f, 0x0700ee, 0x0de071, 0x10b072, 0x44056c, 0x46a444, 0x075094, 0x48c44c, 0x44e490, 0x095078, 0x0ab079, 0x4504ce, 0x07b097, 0x11e07c, 0x630452, 0x0ac07e, 0x07f099, 0x080106, 0x4544b8, 0x0820b1, 0x0830e5, 0x4fc456, 0x0b3085, 0x08609d, 0x45853e, 0x0880c2, 0x5c045a, 0x08a08f, 0x08b0ce, 0x08c0f7, 0x58645c, 0x11108e, 0x45e5c4, 0x0c4090, 0x10a091, 0x4604e4, 0x0d0093, 0x462608, 0x48e46e, 0x4704b2, 0x4d2472, 0x0980bd, 0x4f2474, 0x0e309a, 0x4764aa, 0x0be09c, 0x47851a, 0x47a4de, 0x09f0b5, 0x0a00c1, 0x50047c, 0x57847e, 0x0a30c3, 0x504482, 0x0e90a5, 0x0a6100, 0x4c8484, 0x0a811f, 0x48662a, 0x0c70aa, 0x488494, 0x4924d0, 0x0ad0c8, 0x0ae0d8, 0x496636, 0x10e0b0, 0x4f8498, 0x0f30b2, 0x49a4dc, 0x0f20b4, 0x53c49c, 0x0b60cb, 0x49e57a, 0x0b80e0, 0x0b9109, 0x5e44a4, 0x5484a6, 0x4ac4ae, 0x4b44ca, 0x4d64b6, 0x4ba5da, 0x0c60c0, 0x4bc51e, 0x4be556, 0x6204c0, 0x4c24c4, 0x0f80c5, 0x5664c6, 0x4cc53a, 0x4d462c, 0x0f10c9, 0x4d8552, 0x4da4fa, 0x5be4e0, 0x0cd0ff, 0x5244e2, 0x0cf0e8, 0x4e6568, 0x59a4e8, 0x0f90d2, 0x1010d3, 0x5ac4ea, 0x0d50d7, 0x4ec634, 0x4ee560, 0x4f44f0, 0x4f6638, 0x502522, 0x0db0dc, 0x5065a6, 0x508604, 0x60050a, 0x50c0fb, 0x63250e, 0x1130e1, 0x5a4510, 0x5125fc, 0x516514, 0x51863e, 0x51c536, 0x0e70f4, 0x55c520, 0x602526, 0x0eb0ea, 0x5cc528, 0x5ea52a, 0x1140ed, 0x60c52c, 0x1020ef, 0x0f0119, 0x58e52e, 0x530622, 0x558534, 0x53861e, 0x55e540, 0x5800f6, 0x57e542, 0x5445e6, 0x5465e8, 0x0fa115, 0x54c54a, 0x54e60e, 0x5ae550, 0x1160fe, 0x5f0554, 0x564562, 0x56a58a, 0x56e5ee, 0x10310c, 0x5705d0, 0x107105, 0x5725d4, 0x57463a, 0x5765b4, 0x5825bc, 0x5845e2, 0x5885de, 0x58c592, 0x5ce590, 0x5945f6, 0x63c596, 0x11b110, 0x5d8598, 0x5c259e, 0x5e05a0, 0x5a25c6, 0x5a860a, 0x5aa5ec, 0x5b2610, 0x11a118, 0x6185b6, 0x5f25d2, 0x5d6616, 0x5dc5f8, 0x61a5fe, 0x612614, 0x62e624, 0x626628}; /* get starting addresses of huffman tables into an array [convert codebook into * starting address] */ /* cb tree */ const UINT *aHuffTable[MAX_CB] = { aHuffTree41, /* 0 - */ /* use tree 1 as dummy here */ aHuffTree41, /* 1 1 */ aHuffTree42, /* 2 2 */ aHuffTree43, /* 3 3 */ aHuffTree44, /* 4 4 */ aHuffTree21, /* 5 5 */ aHuffTree22, /* 6 6 */ aHuffTree23, /* 7 7 */ aHuffTree24, /* 8 8 */ aHuffTree25, /* 9 9 */ aHuffTree26, /* 10 10 */ aHuffTree27, /* 11 11 */ aHuffTree41, /* 12 - */ /* use tree 1 as dummy here */ aHuffTree41, /* 13 - */ /* use tree 1 as dummy here */ aHuffTree41, /* 14 - */ /* use tree 1 as dummy here */ aHuffTree41, /* 15 - */ /* use tree 1 as dummy here */ aHuffTree27, /* 16 11 */ aHuffTree27, /* 17 11 */ aHuffTree27, /* 18 11 */ aHuffTree27, /* 19 11 */ aHuffTree27, /* 20 11 */ aHuffTree27, /* 21 11 */ aHuffTree27, /* 22 11 */ aHuffTree27, /* 23 11 */ aHuffTree27, /* 24 11 */ aHuffTree27, /* 25 11 */ aHuffTree27, /* 26 11 */ aHuffTree27, /* 27 11 */ aHuffTree27, /* 28 11 */ aHuffTree27, /* 29 11 */ aHuffTree27, /* 30 11 */ aHuffTree27}; /* 31 11 */ /*--------------------------------------------------------------------------------------------- data-description: The following tables contain the quantized values. Two or four of the quantized values are indexed by the result of the decoding in the decoding tree (see tables above). -------------------------------------------------------------------------------------------- */ /* ********************************************************************************************* */ /* Table: ValTab41 */ /* --------------------------------------------------------------------------------------------- */ /* description: This table contains the quantized values for codebooks * 1-2. */ /* --------------------------------------------------------------------------------------------- */ const SCHAR aValTab41[324] = { -1, -1, -1, -1, -1, -1, -1, 0, -1, -1, -1, 1, -1, -1, 0, -1, -1, -1, 0, 0, -1, -1, 0, 1, -1, -1, 1, -1, -1, -1, 1, 0, -1, -1, 1, 1, -1, 0, -1, -1, -1, 0, -1, 0, -1, 0, -1, 1, -1, 0, 0, -1, -1, 0, 0, 0, -1, 0, 0, 1, -1, 0, 1, -1, -1, 0, 1, 0, -1, 0, 1, 1, -1, 1, -1, -1, -1, 1, -1, 0, -1, 1, -1, 1, -1, 1, 0, -1, -1, 1, 0, 0, -1, 1, 0, 1, -1, 1, 1, -1, -1, 1, 1, 0, -1, 1, 1, 1, 0, -1, -1, -1, 0, -1, -1, 0, 0, -1, -1, 1, 0, -1, 0, -1, 0, -1, 0, 0, 0, -1, 0, 1, 0, -1, 1, -1, 0, -1, 1, 0, 0, -1, 1, 1, 0, 0, -1, -1, 0, 0, -1, 0, 0, 0, -1, 1, 0, 0, 0, -1, 0, 0, 0, 0, 0, 0, 0, 1, 0, 0, 1, -1, 0, 0, 1, 0, 0, 0, 1, 1, 0, 1, -1, -1, 0, 1, -1, 0, 0, 1, -1, 1, 0, 1, 0, -1, 0, 1, 0, 0, 0, 1, 0, 1, 0, 1, 1, -1, 0, 1, 1, 0, 0, 1, 1, 1, 1, -1, -1, -1, 1, -1, -1, 0, 1, -1, -1, 1, 1, -1, 0, -1, 1, -1, 0, 0, 1, -1, 0, 1, 1, -1, 1, -1, 1, -1, 1, 0, 1, -1, 1, 1, 1, 0, -1, -1, 1, 0, -1, 0, 1, 0, -1, 1, 1, 0, 0, -1, 1, 0, 0, 0, 1, 0, 0, 1, 1, 0, 1, -1, 1, 0, 1, 0, 1, 0, 1, 1, 1, 1, -1, -1, 1, 1, -1, 0, 1, 1, -1, 1, 1, 1, 0, -1, 1, 1, 0, 0, 1, 1, 0, 1, 1, 1, 1, -1, 1, 1, 1, 0, 1, 1, 1, 1}; /* ********************************************************************************************* */ /* Table: ValTab42 */ /* --------------------------------------------------------------------------------------------- */ /* description: This table contains the quantized values for codebooks * 3-4. */ /* --------------------------------------------------------------------------------------------- */ const SCHAR aValTab42[324] = { 0, 0, 0, 0, 0, 0, 0, 1, 0, 0, 0, 2, 0, 0, 1, 0, 0, 0, 1, 1, 0, 0, 1, 2, 0, 0, 2, 0, 0, 0, 2, 1, 0, 0, 2, 2, 0, 1, 0, 0, 0, 1, 0, 1, 0, 1, 0, 2, 0, 1, 1, 0, 0, 1, 1, 1, 0, 1, 1, 2, 0, 1, 2, 0, 0, 1, 2, 1, 0, 1, 2, 2, 0, 2, 0, 0, 0, 2, 0, 1, 0, 2, 0, 2, 0, 2, 1, 0, 0, 2, 1, 1, 0, 2, 1, 2, 0, 2, 2, 0, 0, 2, 2, 1, 0, 2, 2, 2, 1, 0, 0, 0, 1, 0, 0, 1, 1, 0, 0, 2, 1, 0, 1, 0, 1, 0, 1, 1, 1, 0, 1, 2, 1, 0, 2, 0, 1, 0, 2, 1, 1, 0, 2, 2, 1, 1, 0, 0, 1, 1, 0, 1, 1, 1, 0, 2, 1, 1, 1, 0, 1, 1, 1, 1, 1, 1, 1, 2, 1, 1, 2, 0, 1, 1, 2, 1, 1, 1, 2, 2, 1, 2, 0, 0, 1, 2, 0, 1, 1, 2, 0, 2, 1, 2, 1, 0, 1, 2, 1, 1, 1, 2, 1, 2, 1, 2, 2, 0, 1, 2, 2, 1, 1, 2, 2, 2, 2, 0, 0, 0, 2, 0, 0, 1, 2, 0, 0, 2, 2, 0, 1, 0, 2, 0, 1, 1, 2, 0, 1, 2, 2, 0, 2, 0, 2, 0, 2, 1, 2, 0, 2, 2, 2, 1, 0, 0, 2, 1, 0, 1, 2, 1, 0, 2, 2, 1, 1, 0, 2, 1, 1, 1, 2, 1, 1, 2, 2, 1, 2, 0, 2, 1, 2, 1, 2, 1, 2, 2, 2, 2, 0, 0, 2, 2, 0, 1, 2, 2, 0, 2, 2, 2, 1, 0, 2, 2, 1, 1, 2, 2, 1, 2, 2, 2, 2, 0, 2, 2, 2, 1, 2, 2, 2, 2}; /* ********************************************************************************************* */ /* Table: ValTab21 */ /* --------------------------------------------------------------------------------------------- */ /* description: This table contains the quantized values for codebooks * 5-6. */ /* --------------------------------------------------------------------------------------------- */ const SCHAR aValTab21[162] = { -4, -4, -4, -3, -4, -2, -4, -1, -4, 0, -4, 1, -4, 2, -4, 3, -4, 4, -3, -4, -3, -3, -3, -2, -3, -1, -3, 0, -3, 1, -3, 2, -3, 3, -3, 4, -2, -4, -2, -3, -2, -2, -2, -1, -2, 0, -2, 1, -2, 2, -2, 3, -2, 4, -1, -4, -1, -3, -1, -2, -1, -1, -1, 0, -1, 1, -1, 2, -1, 3, -1, 4, 0, -4, 0, -3, 0, -2, 0, -1, 0, 0, 0, 1, 0, 2, 0, 3, 0, 4, 1, -4, 1, -3, 1, -2, 1, -1, 1, 0, 1, 1, 1, 2, 1, 3, 1, 4, 2, -4, 2, -3, 2, -2, 2, -1, 2, 0, 2, 1, 2, 2, 2, 3, 2, 4, 3, -4, 3, -3, 3, -2, 3, -1, 3, 0, 3, 1, 3, 2, 3, 3, 3, 4, 4, -4, 4, -3, 4, -2, 4, -1, 4, 0, 4, 1, 4, 2, 4, 3, 4, 4}; /* ********************************************************************************************* */ /* Table: ValTab22 */ /* --------------------------------------------------------------------------------------------- */ /* description: This table contains the quantized values for codebooks * 7-8. */ /* --------------------------------------------------------------------------------------------- */ const SCHAR aValTab22[128] = { 0, 0, 0, 1, 0, 2, 0, 3, 0, 4, 0, 5, 0, 6, 0, 7, 1, 0, 1, 1, 1, 2, 1, 3, 1, 4, 1, 5, 1, 6, 1, 7, 2, 0, 2, 1, 2, 2, 2, 3, 2, 4, 2, 5, 2, 6, 2, 7, 3, 0, 3, 1, 3, 2, 3, 3, 3, 4, 3, 5, 3, 6, 3, 7, 4, 0, 4, 1, 4, 2, 4, 3, 4, 4, 4, 5, 4, 6, 4, 7, 5, 0, 5, 1, 5, 2, 5, 3, 5, 4, 5, 5, 5, 6, 5, 7, 6, 0, 6, 1, 6, 2, 6, 3, 6, 4, 6, 5, 6, 6, 6, 7, 7, 0, 7, 1, 7, 2, 7, 3, 7, 4, 7, 5, 7, 6, 7, 7}; /* ********************************************************************************************* */ /* Table: ValTab23 */ /* --------------------------------------------------------------------------------------------- */ /* description: This table contains the quantized values for codebooks * 9-10. */ /* --------------------------------------------------------------------------------------------- */ const SCHAR aValTab23[338] = { 0, 0, 0, 1, 0, 2, 0, 3, 0, 4, 0, 5, 0, 6, 0, 7, 0, 8, 0, 9, 0, 10, 0, 11, 0, 12, 1, 0, 1, 1, 1, 2, 1, 3, 1, 4, 1, 5, 1, 6, 1, 7, 1, 8, 1, 9, 1, 10, 1, 11, 1, 12, 2, 0, 2, 1, 2, 2, 2, 3, 2, 4, 2, 5, 2, 6, 2, 7, 2, 8, 2, 9, 2, 10, 2, 11, 2, 12, 3, 0, 3, 1, 3, 2, 3, 3, 3, 4, 3, 5, 3, 6, 3, 7, 3, 8, 3, 9, 3, 10, 3, 11, 3, 12, 4, 0, 4, 1, 4, 2, 4, 3, 4, 4, 4, 5, 4, 6, 4, 7, 4, 8, 4, 9, 4, 10, 4, 11, 4, 12, 5, 0, 5, 1, 5, 2, 5, 3, 5, 4, 5, 5, 5, 6, 5, 7, 5, 8, 5, 9, 5, 10, 5, 11, 5, 12, 6, 0, 6, 1, 6, 2, 6, 3, 6, 4, 6, 5, 6, 6, 6, 7, 6, 8, 6, 9, 6, 10, 6, 11, 6, 12, 7, 0, 7, 1, 7, 2, 7, 3, 7, 4, 7, 5, 7, 6, 7, 7, 7, 8, 7, 9, 7, 10, 7, 11, 7, 12, 8, 0, 8, 1, 8, 2, 8, 3, 8, 4, 8, 5, 8, 6, 8, 7, 8, 8, 8, 9, 8, 10, 8, 11, 8, 12, 9, 0, 9, 1, 9, 2, 9, 3, 9, 4, 9, 5, 9, 6, 9, 7, 9, 8, 9, 9, 9, 10, 9, 11, 9, 12, 10, 0, 10, 1, 10, 2, 10, 3, 10, 4, 10, 5, 10, 6, 10, 7, 10, 8, 10, 9, 10, 10, 10, 11, 10, 12, 11, 0, 11, 1, 11, 2, 11, 3, 11, 4, 11, 5, 11, 6, 11, 7, 11, 8, 11, 9, 11, 10, 11, 11, 11, 12, 12, 0, 12, 1, 12, 2, 12, 3, 12, 4, 12, 5, 12, 6, 12, 7, 12, 8, 12, 9, 12, 10, 12, 11, 12, 12}; /* ********************************************************************************************* */ /* Table: ValTab24 */ /* --------------------------------------------------------------------------------------------- */ /* description: This table contains the quantized values for codebooks 11. */ /* --------------------------------------------------------------------------------------------- */ const SCHAR aValTab24[578] = { 0, 0, 0, 1, 0, 2, 0, 3, 0, 4, 0, 5, 0, 6, 0, 7, 0, 8, 0, 9, 0, 10, 0, 11, 0, 12, 0, 13, 0, 14, 0, 15, 0, 16, 1, 0, 1, 1, 1, 2, 1, 3, 1, 4, 1, 5, 1, 6, 1, 7, 1, 8, 1, 9, 1, 10, 1, 11, 1, 12, 1, 13, 1, 14, 1, 15, 1, 16, 2, 0, 2, 1, 2, 2, 2, 3, 2, 4, 2, 5, 2, 6, 2, 7, 2, 8, 2, 9, 2, 10, 2, 11, 2, 12, 2, 13, 2, 14, 2, 15, 2, 16, 3, 0, 3, 1, 3, 2, 3, 3, 3, 4, 3, 5, 3, 6, 3, 7, 3, 8, 3, 9, 3, 10, 3, 11, 3, 12, 3, 13, 3, 14, 3, 15, 3, 16, 4, 0, 4, 1, 4, 2, 4, 3, 4, 4, 4, 5, 4, 6, 4, 7, 4, 8, 4, 9, 4, 10, 4, 11, 4, 12, 4, 13, 4, 14, 4, 15, 4, 16, 5, 0, 5, 1, 5, 2, 5, 3, 5, 4, 5, 5, 5, 6, 5, 7, 5, 8, 5, 9, 5, 10, 5, 11, 5, 12, 5, 13, 5, 14, 5, 15, 5, 16, 6, 0, 6, 1, 6, 2, 6, 3, 6, 4, 6, 5, 6, 6, 6, 7, 6, 8, 6, 9, 6, 10, 6, 11, 6, 12, 6, 13, 6, 14, 6, 15, 6, 16, 7, 0, 7, 1, 7, 2, 7, 3, 7, 4, 7, 5, 7, 6, 7, 7, 7, 8, 7, 9, 7, 10, 7, 11, 7, 12, 7, 13, 7, 14, 7, 15, 7, 16, 8, 0, 8, 1, 8, 2, 8, 3, 8, 4, 8, 5, 8, 6, 8, 7, 8, 8, 8, 9, 8, 10, 8, 11, 8, 12, 8, 13, 8, 14, 8, 15, 8, 16, 9, 0, 9, 1, 9, 2, 9, 3, 9, 4, 9, 5, 9, 6, 9, 7, 9, 8, 9, 9, 9, 10, 9, 11, 9, 12, 9, 13, 9, 14, 9, 15, 9, 16, 10, 0, 10, 1, 10, 2, 10, 3, 10, 4, 10, 5, 10, 6, 10, 7, 10, 8, 10, 9, 10, 10, 10, 11, 10, 12, 10, 13, 10, 14, 10, 15, 10, 16, 11, 0, 11, 1, 11, 2, 11, 3, 11, 4, 11, 5, 11, 6, 11, 7, 11, 8, 11, 9, 11, 10, 11, 11, 11, 12, 11, 13, 11, 14, 11, 15, 11, 16, 12, 0, 12, 1, 12, 2, 12, 3, 12, 4, 12, 5, 12, 6, 12, 7, 12, 8, 12, 9, 12, 10, 12, 11, 12, 12, 12, 13, 12, 14, 12, 15, 12, 16, 13, 0, 13, 1, 13, 2, 13, 3, 13, 4, 13, 5, 13, 6, 13, 7, 13, 8, 13, 9, 13, 10, 13, 11, 13, 12, 13, 13, 13, 14, 13, 15, 13, 16, 14, 0, 14, 1, 14, 2, 14, 3, 14, 4, 14, 5, 14, 6, 14, 7, 14, 8, 14, 9, 14, 10, 14, 11, 14, 12, 14, 13, 14, 14, 14, 15, 14, 16, 15, 0, 15, 1, 15, 2, 15, 3, 15, 4, 15, 5, 15, 6, 15, 7, 15, 8, 15, 9, 15, 10, 15, 11, 15, 12, 15, 13, 15, 14, 15, 15, 15, 16, 16, 0, 16, 1, 16, 2, 16, 3, 16, 4, 16, 5, 16, 6, 16, 7, 16, 8, 16, 9, 16, 10, 16, 11, 16, 12, 16, 13, 16, 14, 16, 15, 16, 16}; /* cb quant. val table */ const SCHAR *aQuantTable[] = { aValTab41, /* 0 - */ /* use quant. val talble 1 as dummy here */ aValTab41, /* 1 1 */ aValTab41, /* 2 1 */ aValTab42, /* 3 2 */ aValTab42, /* 4 2 */ aValTab21, /* 5 3 */ aValTab21, /* 6 3 */ aValTab22, /* 7 4 */ aValTab22, /* 8 4 */ aValTab23, /* 9 5 */ aValTab23, /* 10 5 */ aValTab24, /* 11 6 */ aValTab41, /* 12 - */ /* use quant. val talble 1 as dummy here */ aValTab41, /* 13 - */ /* use quant. val talble 1 as dummy here */ aValTab41, /* 14 - */ /* use quant. val talble 1 as dummy here */ aValTab41, /* 15 - */ /* use quant. val talble 1 as dummy here */ aValTab24, /* 16 6 */ aValTab24, /* 17 6 */ aValTab24, /* 18 6 */ aValTab24, /* 19 6 */ aValTab24, /* 20 6 */ aValTab24, /* 21 6 */ aValTab24, /* 22 6 */ aValTab24, /* 23 6 */ aValTab24, /* 24 6 */ aValTab24, /* 25 6 */ aValTab24, /* 26 6 */ aValTab24, /* 27 6 */ aValTab24, /* 28 6 */ aValTab24, /* 29 6 */ aValTab24, /* 30 6 */ aValTab24}; /* 31 6 */ /* arrays for HCR_TABLE_INFO structures */ /* maximum length of codeword in each codebook */ /* codebook: 0,1, 2,3, 4, 5, 6, 7, 8, 9, * 10,11,12,13,14,15,16,17,18,19,20,21,22,23,24,25,26,27,28,29,30,31 */ const UCHAR aMaxCwLen[MAX_CB] = {0, 11, 9, 20, 16, 13, 11, 14, 12, 17, 14, 49, 0, 0, 0, 0, 14, 17, 21, 21, 25, 25, 29, 29, 29, 29, 33, 33, 33, 37, 37, 41}; /* 11 13 15 17 19 * 21 23 25 27 39 31 */ /* CB: 0 1 2 3 4 5 6 7 8 9 10 12 14 16 18 20 * 22 24 26 28 30 */ const UCHAR aDimCb[MAX_CB] = { 2, 4, 4, 4, 4, 2, 2, 2, 2, 2, 2, 2, 1, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2}; /* codebook dimension - zero cb got a dimension of 2 */ /* 11 13 15 17 19 * 21 23 25 27 39 31 */ /* CB: 0 1 2 3 4 5 6 7 8 9 10 12 14 16 18 20 * 22 24 26 28 30 */ const UCHAR aDimCbShift[MAX_CB] = { 1, 2, 2, 2, 2, 1, 1, 1, 1, 1, 1, 1, 0, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1}; /* codebook dimension */ /* 1 -> decode sign bits */ /* 0 -> decode no sign bits 11 13 15 17 19 21 * 23 25 27 39 31 */ /* CB: 0 1 2 3 4 5 6 7 8 9 10 12 14 16 18 20 22 * 24 26 28 30 */ const UCHAR aSignCb[MAX_CB] = {0, 0, 0, 1, 1, 0, 0, 1, 1, 1, 1, 1, 0, 0, 0, 0, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1}; /* arrays for HCR_CB_PAIRS structures */ const UCHAR aMinOfCbPair[MAX_CB_PAIRS] = {0, 1, 3, 5, 7, 9, 16, 17, 18, 19, 20, 21, 22, 23, 24, 25, 26, 27, 28, 29, 30, 31, 11}; const UCHAR aMaxOfCbPair[MAX_CB_PAIRS] = {0, 2, 4, 6, 8, 10, 16, 17, 18, 19, 20, 21, 22, 23, 24, 25, 26, 27, 28, 29, 30, 31, 11}; /* priorities of codebooks */ const UCHAR aCbPriority[MAX_CB] = {0, 1, 1, 2, 2, 3, 3, 4, 4, 5, 5, 22, 0, 0, 0, 0, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 20, 21}; const SCHAR aCodebook2StartInt[] = {STOP_THIS_STATE, /* cb 0 */ BODY_ONLY, /* cb 1 */ BODY_ONLY, /* cb 2 */ BODY_SIGN__BODY, /* cb 3 */ BODY_SIGN__BODY, /* cb 4 */ BODY_ONLY, /* cb 5 */ BODY_ONLY, /* cb 6 */ BODY_SIGN__BODY, /* cb 7 */ BODY_SIGN__BODY, /* cb 8 */ BODY_SIGN__BODY, /* cb 9 */ BODY_SIGN__BODY, /* cb 10 */ BODY_SIGN_ESC__BODY, /* cb 11 */ STOP_THIS_STATE, /* cb 12 */ STOP_THIS_STATE, /* cb 13 */ STOP_THIS_STATE, /* cb 14 */ STOP_THIS_STATE, /* cb 15 */ BODY_SIGN_ESC__BODY, /* cb 16 */ BODY_SIGN_ESC__BODY, /* cb 17 */ BODY_SIGN_ESC__BODY, /* cb 18 */ BODY_SIGN_ESC__BODY, /* cb 19 */ BODY_SIGN_ESC__BODY, /* cb 20 */ BODY_SIGN_ESC__BODY, /* cb 21 */ BODY_SIGN_ESC__BODY, /* cb 22 */ BODY_SIGN_ESC__BODY, /* cb 23 */ BODY_SIGN_ESC__BODY, /* cb 24 */ BODY_SIGN_ESC__BODY, /* cb 25 */ BODY_SIGN_ESC__BODY, /* cb 26 */ BODY_SIGN_ESC__BODY, /* cb 27 */ BODY_SIGN_ESC__BODY, /* cb 28 */ BODY_SIGN_ESC__BODY, /* cb 29 */ BODY_SIGN_ESC__BODY, /* cb 30 */ BODY_SIGN_ESC__BODY}; /* cb 31 */ const STATEFUNC aStateConstant2State[] = { NULL, /* 0 = STOP_THIS_STATE */ Hcr_State_BODY_ONLY, /* 1 = BODY_ONLY */ Hcr_State_BODY_SIGN__BODY, /* 2 = BODY_SIGN__BODY */ Hcr_State_BODY_SIGN__SIGN, /* 3 = BODY_SIGN__SIGN */ Hcr_State_BODY_SIGN_ESC__BODY, /* 4 = BODY_SIGN_ESC__BODY */ Hcr_State_BODY_SIGN_ESC__SIGN, /* 5 = BODY_SIGN_ESC__SIGN */ Hcr_State_BODY_SIGN_ESC__ESC_PREFIX, /* 6 = BODY_SIGN_ESC__ESC_PREFIX */ Hcr_State_BODY_SIGN_ESC__ESC_WORD}; /* 7 = BODY_SIGN_ESC__ESC_WORD */ /* CB: 0 1 2 3 4 5 6 7 8 9 10 12 * 14 16 18 20 22 24 26 28 30 */ const USHORT aLargestAbsoluteValue[MAX_CB] = { 0, 1, 1, 2, 2, 4, 4, 7, 7, 12, 12, 8191, 0, 0, 0, 0, 15, 31, 47, 63, 95, 127, 159, 191, 223, 255, 319, 383, 511, 767, 1023, 2047}; /* lav */ /* CB: 11 13 * 15 17 19 21 23 25 27 39 31 */ /* ------------------------------------------------------------------------------------------ description: The table 'HuffTreeRvlcEscape' contains the decode tree for the rvlc escape sequences. bit 23 and 11 not used bit 22 and 10 determine end value --> if set codeword is decoded bit 21-12 and 9-0 (offset to next node) or (index value) The escape sequence is the index value. input: codeword output: index ------------------------------------------------------------------------------------------ */ const UINT aHuffTreeRvlcEscape[53] = { 0x002001, 0x400003, 0x401004, 0x402005, 0x403007, 0x404006, 0x00a405, 0x009008, 0x00b406, 0x00c407, 0x00d408, 0x00e409, 0x40b40a, 0x40c00f, 0x40d010, 0x40e011, 0x40f012, 0x410013, 0x411014, 0x412015, 0x016413, 0x414415, 0x017416, 0x417018, 0x419019, 0x01a418, 0x01b41a, 0x01c023, 0x03201d, 0x01e020, 0x43501f, 0x41b41c, 0x021022, 0x41d41e, 0x41f420, 0x02402b, 0x025028, 0x026027, 0x421422, 0x423424, 0x02902a, 0x425426, 0x427428, 0x02c02f, 0x02d02e, 0x42942a, 0x42b42c, 0x030031, 0x42d42e, 0x42f430, 0x033034, 0x431432, 0x433434}; /* ------------------------------------------------------------------------------------------ description: The table 'HuffTreeRvlc' contains the huffman decoding tree for the RVLC scale factors. The table contains 15 allowed, symmetric codewords and 8 forbidden codewords, which are used for error detection. usage of bits: bit 23 and 11 not used bit 22 and 10 determine end value --> if set codeword is decoded bit 21-12 and 9-0 (offset to next node within the table) or (index+7). The decoded (index+7) is in the range from 0,1,..,22. If the (index+7) is in the range 15,16,..,22, then a forbidden codeword is decoded. input: A single bit from a RVLC scalefactor codeword output: [if codeword is not completely decoded:] offset to next node within table or [if codeword is decoded:] A dpcm value i.e. (index+7) in range from 0,1,..,22. The differential scalefactor (DPCM value) named 'index' is calculated by subtracting 7 from the decoded value (index+7). ------------------------------------------------------------------------------------------ */ const UINT aHuffTreeRvlCodewds[22] = { 0x407001, 0x002009, 0x003406, 0x004405, 0x005404, 0x006403, 0x007400, 0x008402, 0x411401, 0x00a408, 0x00c00b, 0x00e409, 0x01000d, 0x40f40a, 0x41400f, 0x01340b, 0x011015, 0x410012, 0x41240c, 0x416014, 0x41540d, 0x41340e}; const FIXP_WTB LowDelaySynthesis256[768] = { WTC(0xdaecb88a), WTC(0xdb7a5230), WTC(0xdc093961), WTC(0xdc9977b5), WTC(0xdd2b11b4), WTC(0xddbe06c1), WTC(0xde525277), WTC(0xdee7f167), WTC(0xdf7ee2f9), WTC(0xe0173207), WTC(0xe0b0e70e), WTC(0xe14beb4d), WTC(0xe1e82002), WTC(0xe285693d), WTC(0xe323ba3c), WTC(0xe3c33cdb), WTC(0xe4640c93), WTC(0xe5060b3f), WTC(0xe5a915ce), WTC(0xe64cffc2), WTC(0xe6f19868), WTC(0xe796b4d7), WTC(0xe83c2ebf), WTC(0xe8e1eea1), WTC(0xe987f784), WTC(0xea2e8014), WTC(0xead5a2b6), WTC(0xeb7d3476), WTC(0xec24ebfb), WTC(0xeccc4e9a), WTC(0xed72f723), WTC(0xee18b585), WTC(0xeebd6902), WTC(0xef610661), WTC(0xf0037f41), WTC(0xf0a4c139), WTC(0xf144c5bc), WTC(0xf1e395c4), WTC(0xf2812c5d), WTC(0xf31d6db4), WTC(0xf3b83ed0), WTC(0xf4517963), WTC(0xf4e8d672), WTC(0xf57de495), WTC(0xf610395e), WTC(0xf69f7e84), WTC(0xf72b9152), WTC(0xf7b495b5), WTC(0xf83af453), WTC(0xf8bf7118), WTC(0xf9429a33), WTC(0xf9c4c183), WTC(0xfa466592), WTC(0xfac80867), WTC(0xfb49d2bd), WTC(0xfbcb7294), WTC(0xfc4d0e35), WTC(0xfccf7e7b), WTC(0xfd53af97), WTC(0xfddab5ca), WTC(0xfe629a56), WTC(0xfee5c7c9), WTC(0xff5b6311), WTC(0xffb6bd45), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0xbff6c845), WTC(0xbfe461ee), WTC(0xbfd1f586), WTC(0xbfbf85b2), WTC(0xbfad1518), WTC(0xbf9aa640), WTC(0xbf883911), WTC(0xbf75caf0), WTC(0xbf635c4b), WTC(0xbf50f09b), WTC(0xbf3e886d), WTC(0xbf2c2167), WTC(0xbf19bc2c), WTC(0xbf075c64), WTC(0xbef502db), WTC(0xbee2ad83), WTC(0xbed05d67), WTC(0xbebe16a7), WTC(0xbeabda46), WTC(0xbe99a62e), WTC(0xbe877b90), WTC(0xbe755ee8), WTC(0xbe63517d), WTC(0xbe51515b), WTC(0xbe3f5fc9), WTC(0xbe2d8141), WTC(0xbe1bb721), WTC(0xbe09ffa3), WTC(0xbdf85c17), WTC(0xbde6d0e3), WTC(0xbdd55f47), WTC(0xbdc4055d), WTC(0xbdb2c438), WTC(0xbda19fe0), WTC(0xbd909942), WTC(0xbd7fae2f), WTC(0xbd6edf8d), WTC(0xbd5e3153), WTC(0xbd4da45b), WTC(0xbd3d365a), WTC(0xbd2ce7eb), WTC(0xbd1cbc87), WTC(0xbd0cb466), WTC(0xbcfccc80), WTC(0xbced04c5), WTC(0xbcdd6022), WTC(0xbccdde49), WTC(0xbcbe7bb4), WTC(0xbcaf37ec), WTC(0xbca01594), WTC(0xbc91149b), WTC(0xbc823234), WTC(0xbc736d81), WTC(0xbc64c7a4), WTC(0xbc564085), WTC(0xbc47d6a9), WTC(0xbc398617), WTC(0xbc2b4915), WTC(0xbc1d2a92), WTC(0xbc0f4370), WTC(0xbc016785), WTC(0xbbf32f6c), WTC(0xbbe53648), WTC(0xbbd91eb5), WTC(0xbbd02f7c), WTC(0xbbcb58db), WTC(0xbbca5ac2), WTC(0xbbcbdea2), WTC(0xbbcee192), WTC(0xbbd2c7c2), WTC(0xbbd7a648), WTC(0xbbde3c5a), WTC(0xbbe7079f), WTC(0xbbf238bf), WTC(0xbbff8eec), WTC(0xbc0e5d22), WTC(0xbc1e2c8b), WTC(0xbc2ec3a4), WTC(0xbc402f19), WTC(0xbc52c1ac), WTC(0xbc6709aa), WTC(0xbc7dcbce), WTC(0xbc979383), WTC(0xbcb4ae30), WTC(0xbcd52aeb), WTC(0xbcf8db63), WTC(0xbd1f6993), WTC(0xbd485b46), WTC(0xbd735007), WTC(0xbda003fc), WTC(0xbdce556c), WTC(0xbdfe453b), WTC(0xbe2ffd4e), WTC(0xbe63ceb7), WTC(0xbe9a006f), WTC(0xbed2ce2b), WTC(0xbf0e7d8e), WTC(0xbf4d5cfc), WTC(0xbf8f92bd), WTC(0xbfd5160c), WTC(0xc01d3b27), WTC(0xc066b8f5), WTC(0xc0b0a3b8), WTC(0xc0fa7cdd), WTC(0xc144b0e8), WTC(0xc1908d71), WTC(0xc1ded403), WTC(0xc22fae49), WTC(0xc2830092), WTC(0xc2d86ca8), WTC(0xc32f4341), WTC(0xc38687e5), WTC(0xc3dd5c55), WTC(0xc43310e4), WTC(0xc4883eca), WTC(0xc4deba3d), WTC(0xc5372c82), WTC(0xc59102fb), WTC(0xc5eb416a), WTC(0xc644986f), WTC(0xc69cabb2), WTC(0xc6f41290), WTC(0xc74b22bf), WTC(0xc7a1e4b9), WTC(0xc7f83500), WTC(0xc84dc737), WTC(0xc8a24e07), WTC(0xc8f5776f), WTC(0xb4db4a8b), WTC(0xb3c58a32), WTC(0xb2b2acb4), WTC(0xb1a2afc0), WTC(0xb0959107), WTC(0xaf8b4e0a), WTC(0xae83dfef), WTC(0xad7f3ba6), WTC(0xac7d5a57), WTC(0xab7e397d), WTC(0xaa81d5a7), WTC(0xa9882a64), WTC(0xa89135bc), WTC(0xa79cf84e), WTC(0xa6ab74fc), WTC(0xa5bcb0d0), WTC(0xa4d0b1b8), WTC(0xa3e77e6d), WTC(0xa3011e16), WTC(0xa21d986c), WTC(0xa13cf921), WTC(0xa05f4fb9), WTC(0x9f84a850), WTC(0x9ead0baf), WTC(0x9dd8888c), WTC(0x9d073385), WTC(0x9c391db5), WTC(0x9b6e5484), WTC(0x9aa6e656), WTC(0x99e2e2c5), WTC(0x99225ac4), WTC(0x9865608a), WTC(0x97ac0564), WTC(0x96f65984), WTC(0x96446a25), WTC(0x95964196), WTC(0x94ebe9ec), WTC(0x94456d19), WTC(0x93a2d46a), WTC(0x93042864), WTC(0x92696dea), WTC(0x91d2a637), WTC(0x913fd120), WTC(0x90b0ecfe), WTC(0x9025f239), WTC(0x8f9ed33e), WTC(0x8f1b804b), WTC(0x8e9be72b), WTC(0x8e1fe984), WTC(0x8da75ce4), WTC(0x8d321511), WTC(0x8cbfe381), WTC(0x8c508090), WTC(0x8be38b7b), WTC(0x8b78a10a), WTC(0x8b0f5bb4), WTC(0x8aa740f3), WTC(0x8a3fc439), WTC(0x89d8a093), WTC(0x8971e7bf), WTC(0x890d0fa8), WTC(0x88acfc32), WTC(0x8855e07a), WTC(0x880d5010), WTC(0x87cc444c), WTC(0x87b6e84e), WTC(0x879e3713), WTC(0x87850a87), WTC(0x876c76ee), WTC(0x8753c55a), WTC(0x873aa70d), WTC(0x872147e7), WTC(0x8707bb23), WTC(0x86edf64f), WTC(0x86d3f20d), WTC(0x86b9ab68), WTC(0x869f21e7), WTC(0x86845744), WTC(0x8669499e), WTC(0x864df395), WTC(0x863254b2), WTC(0x8616715e), WTC(0x85fa4870), WTC(0x85ddd324), WTC(0x85c1108b), WTC(0x85a40597), WTC(0x8586b1d1), WTC(0x85690f48), WTC(0x854b1dfb), WTC(0x852ce3e7), WTC(0x850e61c8), WTC(0x84ef9303), WTC(0x84d078c2), WTC(0x84b119fa), WTC(0x849177fa), WTC(0x84718e56), WTC(0x84515e65), WTC(0x8430ef53), WTC(0x841042cb), WTC(0x83ef54e8), WTC(0x83ce27ab), WTC(0x83acc30a), WTC(0x838b2934), WTC(0x83695686), WTC(0x83474d47), WTC(0x832515b6), WTC(0x8302b202), WTC(0x82e01e3b), WTC(0x82bd5c8e), WTC(0x829a755a), WTC(0x82776abf), WTC(0x8254388a), WTC(0x8230e07e), WTC(0x820d6a69), WTC(0x81e9d82d), WTC(0x81c625ae), WTC(0x81a25455), WTC(0x817e6b1f), WTC(0x815a6b39), WTC(0x81364fec), WTC(0x81121a34), WTC(0x80edd0c9), WTC(0x80c97479), WTC(0x80a5000a), WTC(0x80807332), WTC(0x805bd2e6), WTC(0x80372460), WTC(0x80126cc4), WTC(0x0a8a8431), WTC(0x0ae3c329), WTC(0x0b3fdab2), WTC(0x0b9e6e44), WTC(0x0bff2161), WTC(0x0c619a72), WTC(0x0cc5c66d), WTC(0x0d2bd6df), WTC(0x0d93cf64), WTC(0x0dfd8545), WTC(0x0e69093a), WTC(0x0ed6a636), WTC(0x0f465aea), WTC(0x0fb7d810), WTC(0x102ade99), WTC(0x109f4245), WTC(0x1114c9e8), WTC(0x118b31bc), WTC(0x120282b8), WTC(0x127b0be1), WTC(0x12f46b9a), WTC(0x136d8dde), WTC(0x13e57912), WTC(0x145b55cb), WTC(0x14ce5b18), WTC(0x153dccd0), WTC(0x15a8e724), WTC(0x160edefe), WTC(0x166f004f), WTC(0x16c8aff4), WTC(0x171b7afc), WTC(0x17673db3), WTC(0x17afb798), WTC(0x17fc62a2), WTC(0x185114c1), WTC(0x18add722), WTC(0x1912798d), WTC(0x197eb9a2), WTC(0x19f25657), WTC(0x1a6d113e), WTC(0x1aeeb824), WTC(0x1b7723ab), WTC(0x1c060b77), WTC(0x1c9b09a1), WTC(0x1d361822), WTC(0x1dd7933f), WTC(0x1e7ff592), WTC(0x1f2fd0b5), WTC(0x1fe762a9), WTC(0x20a68700), WTC(0x216bbf5e), WTC(0x22344798), WTC(0x22feebdb), WTC(0x23cc22a6), WTC(0x249da10b), WTC(0x25762e41), WTC(0x2655ebc8), WTC(0x273a4e66), WTC(0x28212eaa), WTC(0x2908ff30), WTC(0x29f2ca00), WTC(0x2ae221c4), WTC(0x2bd9c6fc), WTC(0x2cdb967e), WTC(0x2de64c86), WTC(0x2ef61340), WTC(0x3006852b), WTC(0x31131a1b), WTC(0x321aec79), WTC(0x3320d4ed), WTC(0x342a2cb3), WTC(0x353e77f7), WTC(0x3660aaba), WTC(0x378ef057), WTC(0x38c42495), WTC(0x39f81cec), WTC(0x3b250148), WTC(0x3c47960e), WTC(0x3d60c625), WTC(0x3e75864a), WTC(0x3f8a9ef7), WTC(0x40a46d36), WTC(0x41c537d7), WTC(0x42ed2889), WTC(0x441b52d1), WTC(0x454dc37f), WTC(0x4681e9fa), WTC(0x47b4a9fe), WTC(0x48e39960), WTC(0x4a0d10fd), WTC(0x4b30824b), WTC(0x4c4e759e), WTC(0x4d680b1c), WTC(0x4e7eecaa), WTC(0x4f947d1e), WTC(0x50a9d1f4), WTC(0x51bfee19), WTC(0x52d7be4d), WTC(0x53f187eb), WTC(0x550cd3c8), WTC(0x56270706), WTC(0x573b7c75), WTC(0x58474819), WTC(0x59496ae2), WTC(0x5a43dfa0), WTC(0x5b3b721f), WTC(0x5c32e266), WTC(0x5d2abea1), WTC(0x5e22b479), WTC(0x5f199f8b), WTC(0x600d9194), WTC(0x60fbe188), WTC(0x61e289de), WTC(0x62c0520c), WTC(0x63974d1e), WTC(0x646cb022), WTC(0x6542745a), WTC(0x66172e2f), WTC(0x66e897d0), WTC(0x67b3cc52), WTC(0x68783ebd), WTC(0x6937b9ed), WTC(0x69f365dd), WTC(0x6aabab1a), WTC(0x6b60849c), WTC(0x6c11876e), WTC(0x6cbe46f0), WTC(0x6d6645bc), WTC(0xae238366), WTC(0xb047d99a), WTC(0xb26207ba), WTC(0xb4733ab5), WTC(0xb67c9f5f), WTC(0xb87f5bce), WTC(0xba7bf18d), WTC(0xbc723dd6), WTC(0xbe621be8), WTC(0xc04b6b3d), WTC(0xc22e00ff), WTC(0xc409a8ad), WTC(0xc5de425b), WTC(0xc7abc2a7), WTC(0xc9721421), WTC(0xcb31173c), WTC(0xcce8c08d), WTC(0xce991894), WTC(0xd04218cd), WTC(0xd1e3aba9), WTC(0xd37dcf0c), WTC(0xd510942f), WTC(0xd69bfa86), WTC(0xd81fefcc), WTC(0xd99c766d), WTC(0xdb11a570), WTC(0xdc7f806e), WTC(0xdde5f79d), WTC(0xdf451092), WTC(0xe09ce645), WTC(0xe1ed7fff), WTC(0xe336d16a), WTC(0xe478e4f4), WTC(0xe5b3dbbb), WTC(0xe6e7c1ac), WTC(0xe8148d53), WTC(0xe93a4835), WTC(0xea590eb0), WTC(0xeb70e4f8), WTC(0xec81b75d), WTC(0xed8b8b6c), WTC(0xee8e8068), WTC(0xef8aa970), WTC(0xf0800dfd), WTC(0xf16edc4d), WTC(0xf2576853), WTC(0xf339e058), WTC(0xf4164a51), WTC(0xf4ec8fb8), WTC(0xf5bc7cef), WTC(0xf685a697), WTC(0xf74771c9), WTC(0xf801f31b), WTC(0xf8b5eeb1), WTC(0xf963fadb), WTC(0xfa0c7427), WTC(0xfaaf3e0f), WTC(0xfb4bcb6d), WTC(0xfbe2276f), WTC(0xfc72f578), WTC(0xfcfe65d9), WTC(0xfd842e5b), WTC(0xfe03e14e), WTC(0xfe7ce07d), WTC(0xfeef932b), WTC(0xff5d0236), WTC(0xffc68fee), WTC(0x002dbccc), WTC(0x00927c78), WTC(0x00f32cbd), WTC(0x014d7209), WTC(0x019e5f51), WTC(0x01e550aa), WTC(0x0223fc07), WTC(0x025d2618), WTC(0x02947b12), WTC(0x02cc33db), WTC(0x0304fa92), WTC(0x033db713), WTC(0x0373a431), WTC(0x03a47d96), WTC(0x03ce9b0f), WTC(0x03f14dc9), WTC(0x040ce0bc), WTC(0x042245a9), WTC(0x043309a5), WTC(0x0440981a), WTC(0x044c30f1), WTC(0x0456bb74), WTC(0x0460c1b4), WTC(0x046a2fdd), WTC(0x0472573f), WTC(0x047877b6), WTC(0x047bc673), WTC(0x047b8615), WTC(0x04770be2), WTC(0x046e23fc), WTC(0x04611460), WTC(0x04507fbd), WTC(0x043d6170), WTC(0x0428ca31), WTC(0x0413d39a), WTC(0x03feb9c3), WTC(0x03e8d946), WTC(0x03d16667), WTC(0x03b77aba), WTC(0x039aa384), WTC(0x037ae75c), WTC(0x0358be80), WTC(0x03350af6), WTC(0x031064fa), WTC(0x02eb13f6), WTC(0x02c51a7b), WTC(0x029e38b0), WTC(0x027619ef), WTC(0x024c5c09), WTC(0x02210ea3), WTC(0x01f4b19e), WTC(0x01c78f79), WTC(0x0199b8ad), WTC(0x016b3aef), WTC(0x013c2366), WTC(0x010c7be8), WTC(0x00dc4bb5), WTC(0x00abab29), WTC(0x007ac3e1), WTC(0x0049c02f), WTC(0x0018ca71), WTC(0xd6208221), WTC(0xd54e9f5b), WTC(0xd47fa35f), WTC(0xd3b35bdb), WTC(0xd2e99693), WTC(0xd2222685), WTC(0xd15d5e4c), WTC(0xd09c06ff), WTC(0xcfde0c24), WTC(0xcf2281d2), WTC(0xce698b2a), WTC(0xcdb455e0), WTC(0xcd02c9b7), WTC(0xcc5381e5), WTC(0xcba580c3), WTC(0xcaf839ea), WTC(0xca4ad0b7), WTC(0xc99c2153), WTC(0xc8ec5a61), WTC(0xc83ce258), WTC(0xc78c42cd), WTC(0xc6d6213b), WTC(0xc616a6e0), WTC(0xc54a9f27), WTC(0xc46ee44d), WTC(0xc38058a2), WTC(0xc27bd88d), WTC(0xc15e3d07), WTC(0xc024a4d9), WTC(0xbecc7ca2), WTC(0xbd53f2b9), WTC(0xbbba98b4), WTC(0xba0ffbc3), WTC(0xb872fb24), WTC(0xb6f36547), WTC(0xb5915345), WTC(0xb44c3975), WTC(0xb3238942), WTC(0xb216b3fb), WTC(0xb1252b0b), WTC(0xb04e5fc8), WTC(0xaf91c370), WTC(0xaeeec760), WTC(0xae64dd0b), WTC(0xadf375ce), WTC(0xad9a02f0), WTC(0xad57f5bc), WTC(0xad2cbf87), WTC(0xad17d19d), WTC(0xad189d42), WTC(0xad2e93d6), WTC(0xad5926d3), WTC(0xad97c78a), WTC(0xade9e726), WTC(0xae4ef6fc), WTC(0xaec6688c), WTC(0xaf4fad2c), WTC(0xafea3605), WTC(0xb0957469), WTC(0xb150d9d2), WTC(0xb21bd793), WTC(0xb2f5de5d), WTC(0xb3de573d), WTC(0xb4d4de47), WTC(0xb5d89c80), WTC(0xb6e937c2), WTC(0xb8061354), WTC(0xb92ea07c), WTC(0xba62508e), WTC(0xbba094e4), WTC(0xbce8dee0), WTC(0xbe3a9fe3), WTC(0xbf954939), WTC(0xc0f84c11), WTC(0xc26319c2), WTC(0xc3d523c5), WTC(0xc54ddb77), WTC(0xc6ccb217), WTC(0xc85118f8), WTC(0xc9da8182), WTC(0xcb685d07), WTC(0xccfa1cc7), WTC(0xce8f3211), WTC(0xd0270e48), WTC(0xd1c122c7), WTC(0xd35ce0de), WTC(0xd4f9b9e1), WTC(0xd6971f23), WTC(0xd83481f8), WTC(0xd9d153bb), WTC(0xdb6d05c0), WTC(0xdd070956), WTC(0xde9ecfd2), WTC(0xe033ca91), WTC(0xe1c56ae7), WTC(0xe3532223), WTC(0xe4dc6199), WTC(0xe6609aa5), WTC(0xe7df3e9a), WTC(0xe957bec9), WTC(0xeac98c84), WTC(0xec341927), WTC(0xed96d607), WTC(0xeef13474), WTC(0xf042a5c5), WTC(0xf18a9b4e), WTC(0xf2c88667), WTC(0xf3fbd863), WTC(0xf5240296), WTC(0xf6407658), WTC(0xf750a4fe), WTC(0xf853ffda), WTC(0xf949f840), WTC(0xfa31ff83), WTC(0xfb0b86fb), WTC(0xfbd5fffe), WTC(0xfc90dbe1), WTC(0xfd3b8bf8), WTC(0xfdd58197), WTC(0xfe5e2e14), WTC(0xfed502c4), WTC(0xff3970fc), WTC(0xff8aea0f), WTC(0xffc8df55), WTC(0xfff2c233), WTC(0x000804e6), WTC(0x0008256a), WTC(0xfff25358), }; const FIXP_WTB LowDelaySynthesis240[720] = { WTC(0xdaf16ba1), WTC(0xdb888d4d), WTC(0xdc212bc1), WTC(0xdcbb51d0), WTC(0xdd5703be), WTC(0xddf43f3b), WTC(0xde92fee5), WTC(0xdf333f92), WTC(0xdfd505d1), WTC(0xe078624a), WTC(0xe11d48ff), WTC(0xe1c39358), WTC(0xe26b2066), WTC(0xe313d8a9), WTC(0xe3bde63c), WTC(0xe4696e45), WTC(0xe5164d95), WTC(0xe5c4597d), WTC(0xe673596f), WTC(0xe723153c), WTC(0xe7d35813), WTC(0xe883fa49), WTC(0xe934e7a3), WTC(0xe9e64205), WTC(0xea984aa8), WTC(0xeb4ae6ef), WTC(0xebfdcbf6), WTC(0xecb0744b), WTC(0xed625671), WTC(0xee133361), WTC(0xeec2e1b4), WTC(0xef715334), WTC(0xf01e7588), WTC(0xf0ca33ec), WTC(0xf1748a4e), WTC(0xf21d83ed), WTC(0xf2c50dbe), WTC(0xf36b0639), WTC(0xf40f4892), WTC(0xf4b1944d), WTC(0xf5517289), WTC(0xf5ee573d), WTC(0xf687d70d), WTC(0xf71db368), WTC(0xf7b01057), WTC(0xf83f6570), WTC(0xf8cc9d0a), WTC(0xf9585a73), WTC(0xf9e307f8), WTC(0xfa6d44d2), WTC(0xfaf79d75), WTC(0xfb8208e8), WTC(0xfc0c33c8), WTC(0xfc96d00c), WTC(0xfd22f257), WTC(0xfdb1e623), WTC(0xfe4318a3), WTC(0xfed08c1d), WTC(0xff5091f1), WTC(0xffb4390a), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0xbff62b62), WTC(0xbfe28a84), WTC(0xbfcee30d), WTC(0xbfbb3832), WTC(0xbfa78d29), WTC(0xbf93e490), WTC(0xbf803cc6), WTC(0xbf6c9376), WTC(0xbf58eb89), WTC(0xbf4547f8), WTC(0xbf31a6aa), WTC(0xbf1e06a3), WTC(0xbf0a6bdf), WTC(0xbef6d863), WTC(0xbee349e5), WTC(0xbecfc145), WTC(0xbebc4362), WTC(0xbea8d111), WTC(0xbe956806), WTC(0xbe820aeb), WTC(0xbe6ebeb9), WTC(0xbe5b8313), WTC(0xbe485678), WTC(0xbe353cfb), WTC(0xbe223ab8), WTC(0xbe0f4e5f), WTC(0xbdfc77b4), WTC(0xbde9bb99), WTC(0xbdd71cbe), WTC(0xbdc4990d), WTC(0xbdb23175), WTC(0xbd9feab6), WTC(0xbd8dc58d), WTC(0xbd7bbf8c), WTC(0xbd69daec), WTC(0xbd581c07), WTC(0xbd46820c), WTC(0xbd350af8), WTC(0xbd23b9cc), WTC(0xbd129158), WTC(0xbd018ece), WTC(0xbcf0b057), WTC(0xbcdff912), WTC(0xbccf69cf), WTC(0xbcbefe81), WTC(0xbcaeb63a), WTC(0xbc9e9406), WTC(0xbc8e977c), WTC(0xbc7ebd58), WTC(0xbc6f0544), WTC(0xbc5f7077), WTC(0xbc4ffe1d), WTC(0xbc40ab99), WTC(0xbc317239), WTC(0xbc2252ad), WTC(0xbc136963), WTC(0xbc04a822), WTC(0xbbf5941e), WTC(0xbbe68dab), WTC(0xbbd97a36), WTC(0xbbcff44f), WTC(0xbbcb1891), WTC(0xbbca7818), WTC(0xbbcc765f), WTC(0xbbcffaa5), WTC(0xbbd46a0f), WTC(0xbbda4060), WTC(0xbbe257f1), WTC(0xbbed131c), WTC(0xbbfa7628), WTC(0xbc09cdd6), WTC(0xbc1a685b), WTC(0xbc2bf2fd), WTC(0xbc3e6557), WTC(0xbc521d37), WTC(0xbc67c093), WTC(0xbc803b98), WTC(0xbc9c3060), WTC(0xbcbbf6da), WTC(0xbcdf8c51), WTC(0xbd06afb5), WTC(0xbd30e36d), WTC(0xbd5d99a0), WTC(0xbd8c71a6), WTC(0xbdbd29c3), WTC(0xbdefb72a), WTC(0xbe243660), WTC(0xbe5b0335), WTC(0xbe9477fd), WTC(0xbed0de00), WTC(0xbf108881), WTC(0xbf53d585), WTC(0xbf9aeeba), WTC(0xbfe5bb38), WTC(0xc033318c), WTC(0xc081cdf2), WTC(0xc0d0a9ee), WTC(0xc11f76d5), WTC(0xc16f66c3), WTC(0xc1c1d5c1), WTC(0xc21726d4), WTC(0xc26f5bba), WTC(0xc2ca1036), WTC(0xc3268b0f), WTC(0xc3839ff3), WTC(0xc3e03d22), WTC(0xc43b93ef), WTC(0xc4968594), WTC(0xc4f3353c), WTC(0xc55202ab), WTC(0xc5b21fcf), WTC(0xc61224e6), WTC(0xc670c168), WTC(0xc6ce3f1e), WTC(0xc72b3fbb), WTC(0xc787e688), WTC(0xc7e41f56), WTC(0xc83f94a9), WTC(0xc899e833), WTC(0xc8f2b832), WTC(0xb4d1fc78), WTC(0xb3a9ec70), WTC(0xb28524ca), WTC(0xb163a2ba), WTC(0xb0456373), WTC(0xaf2a6327), WTC(0xae129708), WTC(0xacfdf2ca), WTC(0xabec7168), WTC(0xaade0f8c), WTC(0xa9d2c80a), WTC(0xa8ca9711), WTC(0xa7c57cdc), WTC(0xa6c37c29), WTC(0xa5c49ae6), WTC(0xa4c8e02c), WTC(0xa3d05422), WTC(0xa2daff83), WTC(0xa1e8ec04), WTC(0xa0fa293f), WTC(0xa00ec98a), WTC(0x9f26d990), WTC(0x9e4265a5), WTC(0x9d618437), WTC(0x9c844ce0), WTC(0x9baad0fa), WTC(0x9ad52154), WTC(0x9a03509b), WTC(0x993572ed), WTC(0x986b9e4d), WTC(0x97a5e7d8), WTC(0x96e46329), WTC(0x96271ff8), WTC(0x956e2ab8), WTC(0x94b98f9c), WTC(0x94095a97), WTC(0x935d969b), WTC(0x92b64cdf), WTC(0x9213809b), WTC(0x9175328a), WTC(0x90db6153), WTC(0x90460712), WTC(0x8fb5146f), WTC(0x8f2876f4), WTC(0x8ea018e5), WTC(0x8e1bd6df), WTC(0x8d9b7d9b), WTC(0x8d1ed745), WTC(0x8ca5a942), WTC(0x8c2f93ce), WTC(0x8bbc204c), WTC(0x8b4ad58d), WTC(0x8adb31e4), WTC(0x8a6c909b), WTC(0x89fe673d), WTC(0x8990a478), WTC(0x89244670), WTC(0x88bc84cb), WTC(0x885e0963), WTC(0x880f73f5), WTC(0x87cba2c0), WTC(0x87b48a6f), WTC(0x8799fb25), WTC(0x877f46f2), WTC(0x876519cd), WTC(0x874a9a11), WTC(0x872fae01), WTC(0x871487d2), WTC(0x86f9287b), WTC(0x86dd83b5), WTC(0x86c1947c), WTC(0x86a558f1), WTC(0x8688d2e3), WTC(0x866c015b), WTC(0x864ede0c), WTC(0x863167cc), WTC(0x8613a3b8), WTC(0x85f58fb2), WTC(0x85d723f0), WTC(0x85b86176), WTC(0x85994d82), WTC(0x8579e443), WTC(0x855a201b), WTC(0x853a05b7), WTC(0x85199a26), WTC(0x84f8d93f), WTC(0x84d7c100), WTC(0x84b65901), WTC(0x8494a4ee), WTC(0x84729fd4), WTC(0x84504a9b), WTC(0x842dadb3), WTC(0x840aca71), WTC(0x83e79c87), WTC(0x83c42892), WTC(0x83a07767), WTC(0x837c883a), WTC(0x83585835), WTC(0x8333eeca), WTC(0x830f5368), WTC(0x82ea82ec), WTC(0x82c57c57), WTC(0x82a048ea), WTC(0x827aed86), WTC(0x82556588), WTC(0x822fb255), WTC(0x8209dd1e), WTC(0x81e3e76f), WTC(0x81bdccac), WTC(0x81979098), WTC(0x81713ad7), WTC(0x814ac95d), WTC(0x812437f2), WTC(0x80fd8c4b), WTC(0x80d6cc0a), WTC(0x80aff283), WTC(0x8088fc64), WTC(0x8061eebe), WTC(0x803acfe9), WTC(0x8013a62d), WTC(0x0a8d710a), WTC(0x0aecd974), WTC(0x0b4f7449), WTC(0x0bb4d13b), WTC(0x0c1c7fee), WTC(0x0c86202a), WTC(0x0cf1c2e7), WTC(0x0d5f98d1), WTC(0x0dcf816a), WTC(0x0e4162ae), WTC(0x0eb588ad), WTC(0x0f2c1e88), WTC(0x0fa4d14f), WTC(0x101f4d80), WTC(0x109b5bfd), WTC(0x1118b908), WTC(0x11971466), WTC(0x12168249), WTC(0x129754ab), WTC(0x1318d5c0), WTC(0x1399b5d7), WTC(0x1418d8ec), WTC(0x14953f24), WTC(0x150dfe8a), WTC(0x15822faa), WTC(0x15f0dd70), WTC(0x16592014), WTC(0x16ba35b2), WTC(0x171384e7), WTC(0x1764cf1f), WTC(0x17b22a28), WTC(0x18047d40), WTC(0x185ffc05), WTC(0x18c4a075), WTC(0x19322980), WTC(0x19a84643), WTC(0x1a26a8f4), WTC(0x1aad099d), WTC(0x1b3b3493), WTC(0x1bd0eb32), WTC(0x1c6db754), WTC(0x1d115a8c), WTC(0x1dbc329b), WTC(0x1e6ecb33), WTC(0x1f29d42a), WTC(0x1feda2e4), WTC(0x20ba056e), WTC(0x218d02a2), WTC(0x22635c75), WTC(0x233c2e7f), WTC(0x24184da1), WTC(0x24fa799d), WTC(0x25e54e95), WTC(0x26d6edd8), WTC(0x27cc57a5), WTC(0x28c36349), WTC(0x29bbdfdf), WTC(0x2ab9b7c2), WTC(0x2bc09790), WTC(0x2cd2d465), WTC(0x2def4cb9), WTC(0x2f115dcc), WTC(0x3033a99c), WTC(0x3150fd8d), WTC(0x32698a94), WTC(0x338152e6), WTC(0x34a02dcd), WTC(0x35cdedd1), WTC(0x370aa9fa), WTC(0x38527a79), WTC(0x399c4a4f), WTC(0x3adf9c27), WTC(0x3c17edc7), WTC(0x3d44f05d), WTC(0x3e6c519e), WTC(0x3f93ea5b), WTC(0x40c0fc6d), WTC(0x41f60bdd), WTC(0x43332d15), WTC(0x4476e6ea), WTC(0x45beb7e1), WTC(0x47072f2b), WTC(0x484cb71a), WTC(0x498ceafd), WTC(0x4ac650c3), WTC(0x4bf93458), WTC(0x4d26a4c7), WTC(0x4e5090eb), WTC(0x4f78c2ac), WTC(0x50a0918a), WTC(0x51c93995), WTC(0x52f3d6c7), WTC(0x5420a9a2), WTC(0x554ef122), WTC(0x567ac4a9), WTC(0x579ebeb9), WTC(0x58b8569c), WTC(0x59c74ea4), WTC(0x5ad03f31), WTC(0x5bd821c8), WTC(0x5ce05502), WTC(0x5de8e582), WTC(0x5ef09b49), WTC(0x5ff56247), WTC(0x60f40d81), WTC(0x61ea1450), WTC(0x62d60a2d), WTC(0x63bad7f1), WTC(0x649e942e), WTC(0x65827556), WTC(0x66648178), WTC(0x67418c31), WTC(0x6816c1ee), WTC(0x68e5411d), WTC(0x69aeffdb), WTC(0x6a74bb0e), WTC(0x6b36a5ae), WTC(0x6bf44fd1), WTC(0x6cad341f), WTC(0x6d60c0ca), WTC(0xae35f79b), WTC(0xb07e1b0d), WTC(0xb2bad26d), WTC(0xb4ed8af7), WTC(0xb717b207), WTC(0xb93a8f5b), WTC(0xbb56631e), WTC(0xbd6afcab), WTC(0xbf7832db), WTC(0xc17dd996), WTC(0xc37bb355), WTC(0xc5718d72), WTC(0xc75f56cc), WTC(0xc944f93d), WTC(0xcb224f17), WTC(0xccf74805), WTC(0xcec3ed8c), WTC(0xd08835d9), WTC(0xd2440837), WTC(0xd3f7692d), WTC(0xd5a26b17), WTC(0xd74502b6), WTC(0xd8df1f82), WTC(0xda70d495), WTC(0xdbfa35a1), WTC(0xdd7b3498), WTC(0xdef3cba3), WTC(0xe064184e), WTC(0xe1cc2ab9), WTC(0xe32bf548), WTC(0xe48381cd), WTC(0xe5d2f779), WTC(0xe71a6218), WTC(0xe859b789), WTC(0xe9910a60), WTC(0xeac07956), WTC(0xebe7fb55), WTC(0xed077f41), WTC(0xee1f1f9d), WTC(0xef2efd99), WTC(0xf0372472), WTC(0xf137b605), WTC(0xf23112b3), WTC(0xf323808a), WTC(0xf40f0a86), WTC(0xf4f39883), WTC(0xf5d0eb3c), WTC(0xf6a679c6), WTC(0xf7739640), WTC(0xf8389849), WTC(0xf8f66b2a), WTC(0xf9ada7d6), WTC(0xfa5e97ed), WTC(0xfb08becd), WTC(0xfbabb380), WTC(0xfc48125f), WTC(0xfcde5b40), WTC(0xfd6e4aac), WTC(0xfdf76560), WTC(0xfe78f3ab), WTC(0xfef34cf0), WTC(0xff67b689), WTC(0xffd7e342), WTC(0x004584ef), WTC(0x00b00074), WTC(0x01152d47), WTC(0x0171ccea), WTC(0x01c2fdbe), WTC(0x0209b563), WTC(0x02489832), WTC(0x0283e2cd), WTC(0x02bf2050), WTC(0x02fb72fb), WTC(0x03381f29), WTC(0x0371ea7c), WTC(0x03a602f3), WTC(0x03d267e0), WTC(0x03f6621b), WTC(0x04125dc8), WTC(0x0427b7a5), WTC(0x04385062), WTC(0x0445caca), WTC(0x04518ee3), WTC(0x045c745e), WTC(0x0466d714), WTC(0x0470125c), WTC(0x047742d7), WTC(0x047b73cf), WTC(0x047bba82), WTC(0x047744aa), WTC(0x046dc536), WTC(0x045f9068), WTC(0x044d7632), WTC(0x0438aa92), WTC(0x04227f39), WTC(0x040c2331), WTC(0x03f561d8), WTC(0x03dd60b4), WTC(0x03c31064), WTC(0x03a58d7d), WTC(0x0384b74f), WTC(0x0360e29a), WTC(0x033b1151), WTC(0x031417f0), WTC(0x02ec54ca), WTC(0x02c3d2ce), WTC(0x029a458d), WTC(0x026f4411), WTC(0x02426093), WTC(0x0213d67f), WTC(0x01e43b01), WTC(0x01b3c7c1), WTC(0x01828dd8), WTC(0x01509d92), WTC(0x011e0556), WTC(0x00eace1d), WTC(0x00b70bb3), WTC(0x0082ed8b), WTC(0x004ea707), WTC(0x001a6b8f), WTC(0xd619769b), WTC(0xd539cbf6), WTC(0xd45d68b9), WTC(0xd3840fce), WTC(0xd2ad840b), WTC(0xd1d9a580), WTC(0xd1092160), WTC(0xd03cacdc), WTC(0xcf7383ba), WTC(0xceacfd46), WTC(0xcdea429a), WTC(0xcd2bf5ec), WTC(0xcc709dc1), WTC(0xcbb6dc91), WTC(0xcafdff71), WTC(0xca4504cd), WTC(0xc98a93f4), WTC(0xc8cf0eb0), WTC(0xc813ee7f), WTC(0xc75665f3), WTC(0xc6912d63), WTC(0xc5bff74a), WTC(0xc4dee9fa), WTC(0xc3ea39ef), WTC(0xc2de1c94), WTC(0xc1b6bd08), WTC(0xc07074da), WTC(0xbf081038), WTC(0xbd7b166b), WTC(0xbbc8afd7), WTC(0xba01d47c), WTC(0xb84b481f), WTC(0xb6b65953), WTC(0xb542e5d2), WTC(0xb3f04291), WTC(0xb2bdc25a), WTC(0xb1aab810), WTC(0xb0b676a0), WTC(0xafe050d5), WTC(0xaf27997d), WTC(0xae8ba390), WTC(0xae0bc202), WTC(0xada7479d), WTC(0xad5d872f), WTC(0xad2dd392), WTC(0xad177f97), WTC(0xad19de04), WTC(0xad3441c5), WTC(0xad65fddc), WTC(0xadae6514), WTC(0xae0cca18), WTC(0xae807fde), WTC(0xaf08d967), WTC(0xafa5296f), WTC(0xb054c2ac), WTC(0xb116f81b), WTC(0xb1eb1cb1), WTC(0xb2d082fe), WTC(0xb3c677e5), WTC(0xb4cc6e9c), WTC(0xb5e17eb4), WTC(0xb7052956), WTC(0xb836b427), WTC(0xb97571f3), WTC(0xbac0b594), WTC(0xbc17d1ee), WTC(0xbd7a19eb), WTC(0xbee6e071), WTC(0xc05d7837), WTC(0xc1dd33fe), WTC(0xc36566c2), WTC(0xc4f56377), WTC(0xc68c7ce5), WTC(0xc82a05db), WTC(0xc9cd5148), WTC(0xcb75b207), WTC(0xcd227ad9), WTC(0xced2fe95), WTC(0xd0869026), WTC(0xd23c8268), WTC(0xd3f42834), WTC(0xd5acd460), WTC(0xd765d9c5), WTC(0xd91e8b42), WTC(0xdad63bb5), WTC(0xdc8c3df2), WTC(0xde3fe4d1), WTC(0xdff08333), WTC(0xe19d6bf6), WTC(0xe345f1ee), WTC(0xe4e967f3), WTC(0xe68720e7), WTC(0xe81e6fa3), WTC(0xe9aea6fb), WTC(0xeb3719cb), WTC(0xecb71af3), WTC(0xee2dfd4b), WTC(0xef9b13ab), WTC(0xf0fdb0ee), WTC(0xf25527f1), WTC(0xf3a0cb8e), WTC(0xf4dfee9f), WTC(0xf611e3ff), WTC(0xf735fe8b), WTC(0xf84b911a), WTC(0xf951ee85), WTC(0xfa4869a5), WTC(0xfb2e5557), WTC(0xfc030477), WTC(0xfcc5c9e0), WTC(0xfd75f86b), WTC(0xfe12e2f2), WTC(0xfe9bdc51), WTC(0xff103761), WTC(0xff6f46fc), WTC(0xffb85dfe), WTC(0xffeacf42), WTC(0x0005ee03), WTC(0x0009162b), WTC(0xfff368d1), }; const FIXP_WTB LowDelaySynthesis160[480] = { WTC(0xdb171130), WTC(0xdbfadfbd), WTC(0xdce2192a), WTC(0xddcccbc8), WTC(0xdebaeb0c), WTC(0xdfac6ebd), WTC(0xe0a17875), WTC(0xe199e10c), WTC(0xe29531fd), WTC(0xe3933ef6), WTC(0xe49487be), WTC(0xe598bcf5), WTC(0xe69f38b0), WTC(0xe7a73d45), WTC(0xe8b02e94), WTC(0xe9b9dc97), WTC(0xeac4e62c), WTC(0xebd111fc), WTC(0xecdd0242), WTC(0xede7178d), WTC(0xeeee9c24), WTC(0xeff34dba), WTC(0xf0f4eaf5), WTC(0xf1f36695), WTC(0xf2eeb2b2), WTC(0xf3e663cb), WTC(0xf4d9cbfe), WTC(0xf5c76b3c), WTC(0xf6ada4cb), WTC(0xf78bc92d), WTC(0xf862dc57), WTC(0xf93589ca), WTC(0xfa059b44), WTC(0xfad501fc), WTC(0xfba49819), WTC(0xfc740f58), WTC(0xfd465b6d), WTC(0xfe1ee06f), WTC(0xfef2581b), WTC(0xff9ec7d9), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0xbff143f6), WTC(0xbfd3cd5c), WTC(0xbfb64d54), WTC(0xbf98ce80), WTC(0xbf7b5291), WTC(0xbf5dd523), WTC(0xbf405f8e), WTC(0xbf22ee55), WTC(0xbf058664), WTC(0xbee82d22), WTC(0xbecae0a1), WTC(0xbeadacb5), WTC(0xbe908f68), WTC(0xbe73901d), WTC(0xbe56b673), WTC(0xbe3a013e), WTC(0xbe1d7dad), WTC(0xbe012b19), WTC(0xbde51134), WTC(0xbdc93781), WTC(0xbdad9c88), WTC(0xbd924bd8), WTC(0xbd774311), WTC(0xbd5c882b), WTC(0xbd4220fa), WTC(0xbd280a49), WTC(0xbd0e4d50), WTC(0xbcf4e46e), WTC(0xbcdbd1a4), WTC(0xbcc31624), WTC(0xbcaaaa02), WTC(0xbc929348), WTC(0xbc7acc12), WTC(0xbc63525e), WTC(0xbc4c26b1), WTC(0xbc353ea9), WTC(0xbc1e91e2), WTC(0xbc085aea), WTC(0xbbf1c04a), WTC(0xbbdc7521), WTC(0xbbce4740), WTC(0xbbca5187), WTC(0xbbcd3a7c), WTC(0xbbd33634), WTC(0xbbdc0a34), WTC(0xbbea218a), WTC(0xbbfe25b7), WTC(0xbc162887), WTC(0xbc3076c5), WTC(0xbc4d09f6), WTC(0xbc6d925c), WTC(0xbc94da6e), WTC(0xbcc48300), WTC(0xbcfc9763), WTC(0xbd3bd94f), WTC(0xbd808d05), WTC(0xbdc9a11b), WTC(0xbe16e339), WTC(0xbe691db9), WTC(0xbec179c1), WTC(0xbf210140), WTC(0xbf88cd62), WTC(0xbff8e8d3), WTC(0xc06e1aaa), WTC(0xc0e45951), WTC(0xc15b3820), WTC(0xc1d6e2ff), WTC(0xc2590e0a), WTC(0xc2e10f83), WTC(0xc36c5c9a), WTC(0xc3f735f3), WTC(0xc47fb2d1), WTC(0xc50abce5), WTC(0xc59a0a25), WTC(0xc629f21c), WTC(0xc6b6ef89), WTC(0xc7427180), WTC(0xc7cd1fc4), WTC(0xc856485f), WTC(0xc8dcae93), WTC(0xb487a986), WTC(0xb2ce0812), WTC(0xb11bc4c2), WTC(0xaf70d5cb), WTC(0xadcd228e), WTC(0xac3086a2), WTC(0xaa9af36b), WTC(0xa90c5935), WTC(0xa784b214), WTC(0xa60407e9), WTC(0xa48a7076), WTC(0xa31806f9), WTC(0xa1aceb03), WTC(0xa0494f63), WTC(0x9eed6840), WTC(0x9d996570), WTC(0x9c4d93b4), WTC(0x9b0a3114), WTC(0x99cf798d), WTC(0x989db16a), WTC(0x97752146), WTC(0x965609f1), WTC(0x95409b05), WTC(0x9434fda9), WTC(0x9333587b), WTC(0x923bc7d6), WTC(0x914e5299), WTC(0x906af345), WTC(0x8f9185e6), WTC(0x8ec1cbe9), WTC(0x8dfb64c9), WTC(0x8d3dab18), WTC(0x8c87de34), WTC(0x8bd8c494), WTC(0x8b2ecacc), WTC(0x8a882a5f), WTC(0x89e2ecd1), WTC(0x893f12b2), WTC(0x88a3c878), WTC(0x882141c7), WTC(0x87c65dcd), WTC(0x87a0c07b), WTC(0x8778b859), WTC(0x87514698), WTC(0x8728e900), WTC(0x87000679), WTC(0x86d68f05), WTC(0x86ac6ee7), WTC(0x8681a5ce), WTC(0x86562ed0), WTC(0x8629fdeb), WTC(0x85fd1ca8), WTC(0x85cf7b32), WTC(0x85a11a2f), WTC(0x8571fbbe), WTC(0x854213f5), WTC(0x8511722b), WTC(0x84e00ee3), WTC(0x84adf346), WTC(0x847b28e5), WTC(0x8447a9d0), WTC(0x84138a20), WTC(0x83dec5b6), WTC(0x83a9694f), WTC(0x83738231), WTC(0x833d0def), WTC(0x8306247b), WTC(0x82cec29b), WTC(0x8296f5f3), WTC(0x825ecbe4), WTC(0x82263fe8), WTC(0x81ed682f), WTC(0x81b4406d), WTC(0x817ad2a1), WTC(0x814127f4), WTC(0x8107392d), WTC(0x80cd184d), WTC(0x8092bc5f), WTC(0x80582866), WTC(0x801d714f), WTC(0x0aa4f846), WTC(0x0b3686fe), WTC(0x0bce899e), WTC(0x0c6b8a7e), WTC(0x0d0d036f), WTC(0x0db358b8), WTC(0x0e5e31dd), WTC(0x0f0e4270), WTC(0x0fc347c1), WTC(0x107c35cc), WTC(0x11383a63), WTC(0x11f68759), WTC(0x12b7b1e3), WTC(0x13799d61), WTC(0x14383db4), WTC(0x14f032c0), WTC(0x159e6920), WTC(0x163fb449), WTC(0x16d14820), WTC(0x17512c3f), WTC(0x17c5f622), WTC(0x18483f2a), WTC(0x18df3074), WTC(0x1989f589), WTC(0x1a4783af), WTC(0x1b16f152), WTC(0x1bf7795a), WTC(0x1ce7c950), WTC(0x1de817ec), WTC(0x1efa3f4a), WTC(0x201ff37a), WTC(0x2157d4e1), WTC(0x22995036), WTC(0x23e0d882), WTC(0x25345db2), WTC(0x269a10c3), WTC(0x280a0798), WTC(0x297d6cf9), WTC(0x2afa7e1b), WTC(0x2c8d210a), WTC(0x2e377db2), WTC(0x2feb60cc), WTC(0x31977111), WTC(0x333b1637), WTC(0x34ea14df), WTC(0x36ba32d4), WTC(0x38a524c8), WTC(0x3a8fa891), WTC(0x3c640cb3), WTC(0x3e22ab11), WTC(0x3fde7cc9), WTC(0x41a80486), WTC(0x438394e4), WTC(0x456c8d8f), WTC(0x4758f4da), WTC(0x493d7a7b), WTC(0x4b139f32), WTC(0x4cdbb199), WTC(0x4e9ab8d1), WTC(0x50569530), WTC(0x5213a89e), WTC(0x53d5462c), WTC(0x559a5aa7), WTC(0x5756acf5), WTC(0x58fcfb38), WTC(0x5a8e3e07), WTC(0x5c1a20e8), WTC(0x5da6c7da), WTC(0x5f3231f3), WTC(0x60b51fb6), WTC(0x62260848), WTC(0x6381f5a9), WTC(0x64d79ac7), WTC(0x662c51c0), WTC(0x67779c07), WTC(0x68b22184), WTC(0x69e0ca28), WTC(0x6b068bcf), WTC(0x6c23008e), WTC(0x6d346856), WTC(0xaec92693), WTC(0xb22ca2bd), WTC(0xb578d421), WTC(0xb8b27edb), WTC(0xbbdc38b5), WTC(0xbef598bb), WTC(0xc1fe0dcd), WTC(0xc4f4d7ff), WTC(0xc7d98479), WTC(0xcaabc289), WTC(0xcd6b38f6), WTC(0xd017edff), WTC(0xd2b1aa37), WTC(0xd538739f), WTC(0xd7ac554d), WTC(0xda0d2f5e), WTC(0xdc5b3f69), WTC(0xde966e30), WTC(0xe0bee2c8), WTC(0xe2d4c9cd), WTC(0xe4d82000), WTC(0xe6c94926), WTC(0xe8a84b14), WTC(0xea755ac3), WTC(0xec309bdc), WTC(0xedd9f2a8), WTC(0xef71c040), WTC(0xf0f842ad), WTC(0xf26e53cb), WTC(0xf3d4cd9f), WTC(0xf52b9ddf), WTC(0xf671db4d), WTC(0xf7a58faa), WTC(0xf8c79907), WTC(0xf9da7c73), WTC(0xfadee240), WTC(0xfbd360e6), WTC(0xfcb95db4), WTC(0xfd913bb1), WTC(0xfe594e2e), WTC(0xff10e67c), WTC(0xffbc2326), WTC(0x00608350), WTC(0x00fc8a2d), WTC(0x01873313), WTC(0x01f8e4fe), WTC(0x02579318), WTC(0x02b03ec3), WTC(0x030ab2da), WTC(0x0363ea2a), WTC(0x03b1e60d), WTC(0x03ee2920), WTC(0x0418405c), WTC(0x04348449), WTC(0x0448d9f7), WTC(0x0459c65f), WTC(0x04694a29), WTC(0x0475b506), WTC(0x047bebeb), WTC(0x0478dcd2), WTC(0x046aa475), WTC(0x045249a9), WTC(0x043332f0), WTC(0x0411bb77), WTC(0x03ef893b), WTC(0x03c9ebb8), WTC(0x039da778), WTC(0x036a1273), WTC(0x03316a60), WTC(0x02f6553a), WTC(0x02b98be9), WTC(0x027a2e2c), WTC(0x0236df64), WTC(0x01f036bf), WTC(0x01a78b41), WTC(0x015d29da), WTC(0x01114624), WTC(0x00c406e9), WTC(0x0075ddb1), WTC(0x00277692), WTC(0xd5e139e9), WTC(0xd494362e), WTC(0xd34e2bff), WTC(0xd20e56f8), WTC(0xd0d5a3dc), WTC(0xcfa5904f), WTC(0xce7be3cb), WTC(0xcd5b3086), WTC(0xcc420f0f), WTC(0xcb2c2bf6), WTC(0xca1695c3), WTC(0xc8fdf020), WTC(0xc7e4fc07), WTC(0xc6c374c6), WTC(0xc5895653), WTC(0xc4297218), WTC(0xc296f958), WTC(0xc0c51a3c), WTC(0xbea84ee5), WTC(0xbc38842f), WTC(0xb9915966), WTC(0xb7186ae5), WTC(0xb4eb3040), WTC(0xb3076897), WTC(0xb16acba3), WTC(0xb013112a), WTC(0xaefdf0a2), WTC(0xae2921f5), WTC(0xad925cd6), WTC(0xad3758be), WTC(0xad15cd3d), WTC(0xad2b71ca), WTC(0xad75fe65), WTC(0xadf32a84), WTC(0xaea0ada9), WTC(0xaf7c3fcd), WTC(0xb0839825), WTC(0xb1b46e9c), WTC(0xb30c7a20), WTC(0xb48970be), WTC(0xb62912e0), WTC(0xb7e90de7), WTC(0xb9c71d12), WTC(0xbbc0f7fd), WTC(0xbdd45674), WTC(0xbffef022), WTC(0xc23e7c49), WTC(0xc490b2d4), WTC(0xc6f34b70), WTC(0xc963fda9), WTC(0xcbe0813e), WTC(0xce668d98), WTC(0xd0f3da69), WTC(0xd3861f64), WTC(0xd61b141f), WTC(0xd8b07038), WTC(0xdb43eb5d), WTC(0xddd33d22), WTC(0xe05c1d2e), WTC(0xe2dc432c), WTC(0xe55166ad), WTC(0xe7b93f62), WTC(0xea1184df), WTC(0xec57eec9), WTC(0xee8a34c6), WTC(0xf0a60e70), WTC(0xf2a9336e), WTC(0xf4915b60), WTC(0xf65c3dea), WTC(0xf80792ae), WTC(0xf9911147), WTC(0xfaf67154), WTC(0xfc356a7e), WTC(0xfd4bb465), WTC(0xfe3706a9), WTC(0xfef518ec), WTC(0xff83a2cf), WTC(0xffe05bed), WTC(0x0008fd26), WTC(0xfffb4037), }; const FIXP_WTB LowDelaySynthesis128[384] = { WTC(0xdb335c78), WTC(0xdc512d40), WTC(0xdd746116), WTC(0xde9cf7ae), WTC(0xdfcadda8), WTC(0xe0fe4196), WTC(0xe236a409), WTC(0xe373518f), WTC(0xe4b4e870), WTC(0xe5faf25e), WTC(0xe744190a), WTC(0xe88f06f4), WTC(0xe9db2796), WTC(0xeb29613f), WTC(0xec78b08e), WTC(0xedc5f65d), WTC(0xef0f5af4), WTC(0xf05447dd), WTC(0xf1945392), WTC(0xf2cf795a), WTC(0xf405123d), WTC(0xf533af3d), WTC(0xf65842dd), WTC(0xf77071ae), WTC(0xf87d623f), WTC(0xf983c711), WTC(0xfa8730ce), WTC(0xfb8aad4b), WTC(0xfc8e1dc6), WTC(0xfd96d19a), WTC(0xfea5325a), WTC(0xff8d21da), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0xbfed9606), WTC(0xbfc8bddf), WTC(0xbfa3dd57), WTC(0xbf7f023c), WTC(0xbf5a25e8), WTC(0xbf3554df), WTC(0xbf108b6b), WTC(0xbeebd7bd), WTC(0xbec738a6), WTC(0xbea2bf45), WTC(0xbe7e6b42), WTC(0xbe5a4fd5), WTC(0xbe366de8), WTC(0xbe12d91e), WTC(0xbdef9338), WTC(0xbdccaf6e), WTC(0xbdaa2e3f), WTC(0xbd88205e), WTC(0xbd668430), WTC(0xbd456994), WTC(0xbd24cdad), WTC(0xbd04bc91), WTC(0xbce52def), WTC(0xbcc62942), WTC(0xbca7a273), WTC(0xbc899fba), WTC(0xbc6c16ab), WTC(0xbc4f0812), WTC(0xbc326537), WTC(0xbc162fb3), WTC(0xbbfa5915), WTC(0xbbded462), WTC(0xbbcd3956), WTC(0xbbcae0f5), WTC(0xbbd0beb7), WTC(0xbbdaaf42), WTC(0xbbec5289), WTC(0xbc06d115), WTC(0xbc266162), WTC(0xbc494d27), WTC(0xbc721013), WTC(0xbca5b2d9), WTC(0xbce6a063), WTC(0xbd339d3d), WTC(0xbd8975b2), WTC(0xbde618b1), WTC(0xbe499de1), WTC(0xbeb60feb), WTC(0xbf2d830e), WTC(0xbfb1ee2e), WTC(0xc041e22f), WTC(0xc0d59618), WTC(0xc16a574d), WTC(0xc206ed94), WTC(0xc2ad7ad1), WTC(0xc35ae733), WTC(0xc4085fbf), WTC(0xc4b33a35), WTC(0xc563f6ce), WTC(0xc6181ab7), WTC(0xc6c86beb), WTC(0xc7768de4), WTC(0xc8231ab3), WTC(0xc8cc145a), WTC(0xb4500dde), WTC(0xb22a524e), WTC(0xb010144c), WTC(0xae013530), WTC(0xabfd7206), WTC(0xaa04a92f), WTC(0xa816c008), WTC(0xa633baab), WTC(0xa45bbe2b), WTC(0xa28eff5e), WTC(0xa0cdc4c6), WTC(0x9f187801), WTC(0x9d6f7708), WTC(0x9bd34eb0), WTC(0x9a44763a), WTC(0x98c36ace), WTC(0x9750b896), WTC(0x95ecdc61), WTC(0x94982f8c), WTC(0x9353005a), WTC(0x921d8bac), WTC(0x90f7e126), WTC(0x8fe1e828), WTC(0x8edb3ddc), WTC(0x8de337c8), WTC(0x8cf89cdb), WTC(0x8c19be6d), WTC(0x8b43d072), WTC(0x8a737663), WTC(0x89a52f4d), WTC(0x88dc38e8), WTC(0x882f6f3f), WTC(0x87c22c55), WTC(0x87918a21), WTC(0x87602c61), WTC(0x872dfd0a), WTC(0x86fae063), WTC(0x86c6d729), WTC(0x8691c4ad), WTC(0x865ba7e8), WTC(0x86246b66), WTC(0x85ec17ab), WTC(0x85b293e2), WTC(0x8577eaac), WTC(0x853c09be), WTC(0x84ff042d), WTC(0x84c0d195), WTC(0x84818c4c), WTC(0x84412e63), WTC(0x83ffd42c), WTC(0x83bd7bdf), WTC(0x837a471b), WTC(0x833636dc), WTC(0x82f16e48), WTC(0x82abed37), WTC(0x8265d6c4), WTC(0x821f28cf), WTC(0x81d80322), WTC(0x8190625c), WTC(0x81486134), WTC(0x80fff7a0), WTC(0x80b73d86), WTC(0x806e2527), WTC(0x8024c969), WTC(0x0ab6c2d2), WTC(0x0b6edac2), WTC(0x0c302988), WTC(0x0cf88fab), WTC(0x0dc87461), WTC(0x0e9f9122), WTC(0x0f7ee53f), WTC(0x1064e7bf), WTC(0x114fe4b7), WTC(0x123e9e4c), WTC(0x13311589), WTC(0x1420b664), WTC(0x15069245), WTC(0x15dc93ad), WTC(0x169caf41), WTC(0x17422e16), WTC(0x17d51de4), WTC(0x187e7622), WTC(0x1947aa0b), WTC(0x1a2ed3b4), WTC(0x1b321858), WTC(0x1c4fcc7f), WTC(0x1d8601a3), WTC(0x1ed6ec5e), WTC(0x20460b07), WTC(0x21cfbf59), WTC(0x23652732), WTC(0x2508e4b1), WTC(0x26c7b0a6), WTC(0x289505da), WTC(0x2a698dd8), WTC(0x2c5954d2), WTC(0x2e6dd135), WTC(0x308d838f), WTC(0x329de377), WTC(0x34b28f13), WTC(0x36f67988), WTC(0x395ec1e5), WTC(0x3bb7b587), WTC(0x3deb63cc), WTC(0x4016b320), WTC(0x42584eb2), WTC(0x44b424ca), WTC(0x471ba5b2), WTC(0x49791954), WTC(0x4bc02004), WTC(0x4df3b8c0), WTC(0x501f1e1e), WTC(0x524b93e2), WTC(0x547f0eef), WTC(0x56b23d03), WTC(0x58c99052), WTC(0x5abfc042), WTC(0x5caebf1a), WTC(0x5e9e618b), WTC(0x608595d7), WTC(0x62528e67), WTC(0x6401eb53), WTC(0x65ad0eb1), WTC(0x674f1cd2), WTC(0x68d8804e), WTC(0x6a4ff10e), WTC(0x6bb987a5), WTC(0x6d12e937), WTC(0xaf370652), WTC(0xb36badcf), WTC(0xb77ec321), WTC(0xbb77e364), WTC(0xbf57979e), WTC(0xc31cb589), WTC(0xc6c5e686), WTC(0xca52811d), WTC(0xcdc1d66b), WTC(0xd113d0f0), WTC(0xd4481c98), WTC(0xd75ee41b), WTC(0xda57f7bf), WTC(0xdd33a926), WTC(0xdff1e272), WTC(0xe2931227), WTC(0xe5174232), WTC(0xe77f0b15), WTC(0xe9ca889d), WTC(0xebfa2f7c), WTC(0xee0de002), WTC(0xf0063326), WTC(0xf1e3e5f1), WTC(0xf3a8d749), WTC(0xf5555599), WTC(0xf6e77eb2), WTC(0xf85cb5d6), WTC(0xf9b8e64d), WTC(0xfafe52ad), WTC(0xfc2b37b8), WTC(0xfd420467), WTC(0xfe41412c), WTC(0xff26ded5), WTC(0xfffa6150), WTC(0x00c374a4), WTC(0x01773884), WTC(0x02058de3), WTC(0x0278d689), WTC(0x02e87372), WTC(0x03593268), WTC(0x03ba79ab), WTC(0x03fff32b), WTC(0x042b2341), WTC(0x044690de), WTC(0x045bc96a), WTC(0x046e77e9), WTC(0x047a85c1), WTC(0x0479d8c4), WTC(0x04681aec), WTC(0x0447318f), WTC(0x041e4d13), WTC(0x03f3edad), WTC(0x03c4cc17), WTC(0x038b1f6f), WTC(0x03470909), WTC(0x02fdcebd), WTC(0x02b1cb18), WTC(0x0261730d), WTC(0x020afad0), WTC(0x01b0b9c8), WTC(0x0153c1a0), WTC(0x00f47426), WTC(0x00933db7), WTC(0x003140e9), WTC(0xd5b730bf), WTC(0xd4192c32), WTC(0xd2859479), WTC(0xd0fc3b9d), WTC(0xcf800352), WTC(0xce0e6ab3), WTC(0xccaaf25f), WTC(0xcb4ed003), WTC(0xc9f3ae44), WTC(0xc8948a56), WTC(0xc73226fd), WTC(0xc5b2676a), WTC(0xc3fa2a82), WTC(0xc1f05f70), WTC(0xbf7c884b), WTC(0xbc8b1e86), WTC(0xb93e1633), WTC(0xb63eb483), WTC(0xb3b45d14), WTC(0xb19a8ee2), WTC(0xafecd4aa), WTC(0xaea6b8e7), WTC(0xadc3c6bf), WTC(0xad3f88ac), WTC(0xad158929), WTC(0xad4152b1), WTC(0xadbe7068), WTC(0xae886c76), WTC(0xaf9ad1fe), WTC(0xb0f12b26), WTC(0xb2870347), WTC(0xb457d633), WTC(0xb65f592c), WTC(0xb898eca0), WTC(0xbb00291b), WTC(0xbd90996b), WTC(0xc045c861), WTC(0xc31b4022), WTC(0xc60c8bd5), WTC(0xc91535f2), WTC(0xcc30c94c), WTC(0xcf5ad039), WTC(0xd28ed58a), WTC(0xd5c863e5), WTC(0xd90305e6), WTC(0xdc3a4644), WTC(0xdf69af93), WTC(0xe28ccc93), WTC(0xe59f27d7), WTC(0xe89c4c15), WTC(0xeb7fc3e3), WTC(0xee4519f7), WTC(0xf0e7d8ed), WTC(0xf3638b73), WTC(0xf5b3bc30), WTC(0xf7d3f5d0), WTC(0xf9bfc2f0), WTC(0xfb72ae35), WTC(0xfce84251), WTC(0xfe1c09e5), WTC(0xff098f9a), WTC(0xffac5e15), WTC(0xffffffb2), WTC(0x0000159b), }; const FIXP_WTB LowDelaySynthesis120[360] = { WTC(0xdb3ccdcd), WTC(0xdc6e0d69), WTC(0xdda570a7), WTC(0xdee2ef33), WTC(0xe02680f4), WTC(0xe1704397), WTC(0xe2bf5626), WTC(0xe4137c80), WTC(0xe56d30cd), WTC(0xe6cb22fe), WTC(0xe82b9f14), WTC(0xe98d8220), WTC(0xeaf18a17), WTC(0xec57328f), WTC(0xedbae901), WTC(0xef1a42b9), WTC(0xf07481f7), WTC(0xf1c932d4), WTC(0xf3183e19), WTC(0xf460b35f), WTC(0xf5a04ac7), WTC(0xf6d33845), WTC(0xf7f80ec0), WTC(0xf912a5cd), WTC(0xfa282a2e), WTC(0xfb3cd81c), WTC(0xfc51629b), WTC(0xfd69f81f), WTC(0xfe8abdeb), WTC(0xff86f173), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0xbfec5bfa), WTC(0xbfc50dd8), WTC(0xbf9db88b), WTC(0xbf76683c), WTC(0xbf4f1944), WTC(0xbf27d649), WTC(0xbf00a158), WTC(0xbed9848c), WTC(0xbeb288f1), WTC(0xbe8bb79f), WTC(0xbe651f00), WTC(0xbe3ec6f8), WTC(0xbe18c1f6), WTC(0xbdf315fe), WTC(0xbdcdd79d), WTC(0xbda909c1), WTC(0xbd84beae), WTC(0xbd60f6ab), WTC(0xbd3dc210), WTC(0xbd1b2093), WTC(0xbcf91ae9), WTC(0xbcd7acab), WTC(0xbcb6d5cc), WTC(0xbc969143), WTC(0xbc76dcf4), WTC(0xbc57b317), WTC(0xbc390c4d), WTC(0xbc1ad514), WTC(0xbbfd2fdb), WTC(0xbbdfa548), WTC(0xbbcce946), WTC(0xbbcb39d9), WTC(0xbbd214cf), WTC(0xbbddfb60), WTC(0xbbf3783e), WTC(0xbc11f8d2), WTC(0xbc3509f8), WTC(0xbc5ca2b0), WTC(0xbc8dc0b0), WTC(0xbccd4b2f), WTC(0xbd1b7107), WTC(0xbd74c6a4), WTC(0xbdd635c0), WTC(0xbe3f4cd7), WTC(0xbeb24836), WTC(0xbf31b5df), WTC(0xbfbfe6e2), WTC(0xc05a6dbf), WTC(0xc0f80700), WTC(0xc198449c), WTC(0xc242ea15), WTC(0xc2f8270d), WTC(0xc3b20bd2), WTC(0xc468f554), WTC(0xc522637e), WTC(0xc5e2403b), WTC(0xc69f9771), WTC(0xc7599dd5), WTC(0xc811f85a), WTC(0xc8c687db), WTC(0xb43d8b3b), WTC(0xb1f3fb3c), WTC(0xafb77be9), WTC(0xad87e063), WTC(0xab64dcd5), WTC(0xa94e4cc2), WTC(0xa74418f8), WTC(0xa5465837), WTC(0xa3554252), WTC(0xa1711f59), WTC(0x9f9a62f5), WTC(0x9dd18104), WTC(0x9c17167e), WTC(0x9a6bbbe9), WTC(0x98d0061c), WTC(0x97449e23), WTC(0x95ca1ad6), WTC(0x9460e7a1), WTC(0x93096222), WTC(0x91c3c9da), WTC(0x90902633), WTC(0x8f6e3c5f), WTC(0x8e5d7788), WTC(0x8d5cb789), WTC(0x8c6a42d5), WTC(0x8b833d6f), WTC(0x8aa3cc10), WTC(0x89c7789a), WTC(0x88ef9802), WTC(0x883452c3), WTC(0x87c0bb9a), WTC(0x878c8326), WTC(0x8757eb4c), WTC(0x87222119), WTC(0x86eb5f31), WTC(0x86b3802c), WTC(0x867a73ee), WTC(0x86402cf8), WTC(0x8604a439), WTC(0x85c7cd22), WTC(0x8589a40b), WTC(0x854a1d2f), WTC(0x850944c1), WTC(0x84c71670), WTC(0x8483acc8), WTC(0x843f04b6), WTC(0x83f93cd9), WTC(0x83b2576c), WTC(0x836a7812), WTC(0x8321a764), WTC(0x82d805e8), WTC(0x828da076), WTC(0x824290c3), WTC(0x81f6e6a8), WTC(0x81aab23c), WTC(0x815e05f8), WTC(0x8110e4d6), WTC(0x80c362df), WTC(0x8075781d), WTC(0x80273c0c), WTC(0x0abcb7ed), WTC(0x0b81d183), WTC(0x0c5113e3), WTC(0x0d2867d5), WTC(0x0e082fa5), WTC(0x0ef089ad), WTC(0x0fe1d9cd), WTC(0x10d9e5f4), WTC(0x11d6a2a0), WTC(0x12d814b1), WTC(0x13d98f52), WTC(0x14d221e1), WTC(0x15ba467a), WTC(0x168a9c18), WTC(0x173d1fef), WTC(0x17da3bff), WTC(0x18912cac), WTC(0x196c2a5e), WTC(0x1a68dd2e), WTC(0x1b85250c), WTC(0x1cbea9a1), WTC(0x1e147be7), WTC(0x1f8aa446), WTC(0x2122e73a), WTC(0x22cf6de3), WTC(0x248867f3), WTC(0x265d7937), WTC(0x2847c91b), WTC(0x2a39d98e), WTC(0x2c482abf), WTC(0x2e7ff439), WTC(0x30c31f46), WTC(0x32f5245d), WTC(0x3535070e), WTC(0x37adae38), WTC(0x3a3f151a), WTC(0x3caf861b), WTC(0x3effc08b), WTC(0x415a803b), WTC(0x43d45ca9), WTC(0x46631c69), WTC(0x48ed9c7e), WTC(0x4b608807), WTC(0x4dbbead3), WTC(0x500ca1df), WTC(0x525e3c5d), WTC(0x54b7c199), WTC(0x570df9a2), WTC(0x5940f661), WTC(0x5b542f13), WTC(0x5d64a202), WTC(0x5f738e39), WTC(0x61704027), WTC(0x6348ef5e), WTC(0x65109c5f), WTC(0x66d3e29d), WTC(0x687eb29a), WTC(0x6a125643), WTC(0x6b960b3a), WTC(0x6d07b283), WTC(0xaf5b8daa), WTC(0xb3d557ba), WTC(0xb829feb2), WTC(0xbc6199e7), WTC(0xc07bfbc1), WTC(0xc477a1b6), WTC(0xc8532f30), WTC(0xcc0dd698), WTC(0xcfa71f06), WTC(0xd31ec578), WTC(0xd674c5b9), WTC(0xd9a90516), WTC(0xdcbbc2b6), WTC(0xdfacf934), WTC(0xe27d19ab), WTC(0xe52c3d89), WTC(0xe7bb0f52), WTC(0xea29bdd0), WTC(0xec78bc4a), WTC(0xeea805e6), WTC(0xf0b85a68), WTC(0xf2ab266e), WTC(0xf48234a7), WTC(0xf63cb733), WTC(0xf7d70b3f), WTC(0xf952d4d3), WTC(0xfab4906d), WTC(0xfbfaa858), WTC(0xfd272437), WTC(0xfe392bea), WTC(0xff2e26f1), WTC(0x000efac0), WTC(0x00e36d26), WTC(0x019bd803), WTC(0x0229e357), WTC(0x02a16b25), WTC(0x0319ee5e), WTC(0x038ccea0), WTC(0x03e56d3d), WTC(0x041dc072), WTC(0x043f58cc), WTC(0x04571273), WTC(0x046ba761), WTC(0x0479caa5), WTC(0x047a2279), WTC(0x04673950), WTC(0x04435263), WTC(0x041750da), WTC(0x03e99926), WTC(0x03b4ba1f), WTC(0x03731e2b), WTC(0x0327b303), WTC(0x02d8301b), WTC(0x0284fb11), WTC(0x022b464a), WTC(0x01cc1b40), WTC(0x0169ab7d), WTC(0x01047d1e), WTC(0x009d04e0), WTC(0x003484b0), WTC(0xd5a9348a), WTC(0xd3f05eca), WTC(0xd24339e7), WTC(0xd0a269b7), WTC(0xcf0fe026), WTC(0xcd8aa14b), WTC(0xcc13963e), WTC(0xcaa19c5a), WTC(0xc92cd701), WTC(0xc7b5cd7b), WTC(0xc62a4fe3), WTC(0xc46742be), WTC(0xc24e128c), WTC(0xbfc0b758), WTC(0xbca661e3), WTC(0xb922924b), WTC(0xb5f87ac2), WTC(0xb35308e7), WTC(0xb12cc90f), WTC(0xaf80522c), WTC(0xae483b10), WTC(0xad7f1af9), WTC(0xad1f8874), WTC(0xad241a09), WTC(0xad8766ea), WTC(0xae4405b1), WTC(0xaf548d72), WTC(0xb0b394ad), WTC(0xb25bb297), WTC(0xb4476fa1), WTC(0xb6718d2c), WTC(0xb8d47781), WTC(0xbb6ad37b), WTC(0xbe2f3831), WTC(0xc11c3c6c), WTC(0xc42c76b1), WTC(0xc75a7e40), WTC(0xcaa0e9d1), WTC(0xcdfa502b), WTC(0xd1614803), WTC(0xd4d06850), WTC(0xd84247d3), WTC(0xdbb17d6d), WTC(0xdf189fe3), WTC(0xe272461e), WTC(0xe5b906e1), WTC(0xe8e7790e), WTC(0xebf83366), WTC(0xeee5cccc), WTC(0xf1aadc0a), WTC(0xf441f7fa), WTC(0xf6a5b772), WTC(0xf8d0b146), WTC(0xfabd7c3e), WTC(0xfc66af35), WTC(0xfdc6e101), WTC(0xfed8a875), WTC(0xff969c63), WTC(0xfffb5390), WTC(0x00017ad8), }; const FIXP_WTB LowDelaySynthesis512[1536] = { /* part 0 */ WTC(0xdac984c0), WTC(0xdb100080), WTC(0xdb56cd00), WTC(0xdb9dec40), WTC(0xdbe55fc0), WTC(0xdc2d2880), WTC(0xdc754780), WTC(0xdcbdbd80), WTC(0xdd068a80), WTC(0xdd4fae80), WTC(0xdd992940), WTC(0xdde2f9c0), WTC(0xde2d1fc0), WTC(0xde779a80), WTC(0xdec26a00), WTC(0xdf0d8e00), WTC(0xdf590680), WTC(0xdfa4d540), WTC(0xdff0fc80), WTC(0xe03d7e20), WTC(0xe08a5900), WTC(0xe0d78a20), WTC(0xe1250cc0), WTC(0xe172dcc0), WTC(0xe1c0f7a0), WTC(0xe20f59a0), WTC(0xe25dfea0), WTC(0xe2ace400), WTC(0xe2fc0be0), WTC(0xe34b7bc0), WTC(0xe39b3c80), WTC(0xe3eb5260), WTC(0xe43bbac0), WTC(0xe48c7160), WTC(0xe4dd7140), WTC(0xe52eb600), WTC(0xe5803c00), WTC(0xe5d1fda0), WTC(0xe623f360), WTC(0xe6761700), WTC(0xe6c86400), WTC(0xe71ad500), WTC(0xe76d63e0), WTC(0xe7c00ba0), WTC(0xe812c8e0), WTC(0xe86598e0), WTC(0xe8b878e0), WTC(0xe90b68a0), WTC(0xe95e6c40), WTC(0xe9b18ae0), WTC(0xea04ce80), WTC(0xea583ba0), WTC(0xeaabcda0), WTC(0xeaff7ee0), WTC(0xeb5348e0), WTC(0xeba722c0), WTC(0xebfb0060), WTC(0xec4ed240), WTC(0xeca28540), WTC(0xecf60c20), WTC(0xed496120), WTC(0xed9c7e80), WTC(0xedef5e40), WTC(0xee41fc00), WTC(0xee945600), WTC(0xeee66ac0), WTC(0xef3839a0), WTC(0xef89c0e0), WTC(0xefdafda0), WTC(0xf02bed60), WTC(0xf07c8e80), WTC(0xf0cce000), WTC(0xf11ce220), WTC(0xf16c9620), WTC(0xf1bbfe30), WTC(0xf20b19e0), WTC(0xf259e5a0), WTC(0xf2a85dc0), WTC(0xf2f67ed0), WTC(0xf34445b0), WTC(0xf391aed0), WTC(0xf3deb590), WTC(0xf42b53e0), WTC(0xf4778140), WTC(0xf4c33190), WTC(0xf50e5660), WTC(0xf558df30), WTC(0xf5a2be50), WTC(0xf5ebea10), WTC(0xf6345780), WTC(0xf67bfab0), WTC(0xf6c2cee0), WTC(0xf708d7b0), WTC(0xf74e19c0), WTC(0xf7929a70), WTC(0xf7d66630), WTC(0xf8199268), WTC(0xf85c3860), WTC(0xf89e7480), WTC(0xf8e058c0), WTC(0xf921ec08), WTC(0xf9633800), WTC(0xf9a44980), WTC(0xf9e53158), WTC(0xfa260158), WTC(0xfa66ca18), WTC(0xfaa79ac0), WTC(0xfae87920), WTC(0xfb295fa0), WTC(0xfb6a42b8), WTC(0xfbab1240), WTC(0xfbebd1c0), WTC(0xfc2c9c24), WTC(0xfc6d8d90), WTC(0xfcaec240), WTC(0xfcf05684), WTC(0xfd326a98), WTC(0xfd75254c), WTC(0xfdb8afd4), WTC(0xfdfccdfc), WTC(0xfe40d694), WTC(0xfe84161c), WTC(0xfec5cf5a), WTC(0xff04e7fc), WTC(0xff3fdfe3), WTC(0xff751ddf), WTC(0xffa2fb0f), WTC(0xffc87c42), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0xbffb6081), WTC(0xbff22f81), WTC(0xbfe8fc01), WTC(0xbfdfc781), WTC(0xbfd69101), WTC(0xbfcd5a01), WTC(0xbfc42201), WTC(0xbfbae981), WTC(0xbfb1b101), WTC(0xbfa87901), WTC(0xbf9f4181), WTC(0xbf960b01), WTC(0xbf8cd481), WTC(0xbf839d81), WTC(0xbf7a6681), WTC(0xbf712f01), WTC(0xbf67f801), WTC(0xbf5ec101), WTC(0xbf558b01), WTC(0xbf4c5681), WTC(0xbf432281), WTC(0xbf39ee81), WTC(0xbf30bb01), WTC(0xbf278801), WTC(0xbf1e5501), WTC(0xbf152381), WTC(0xbf0bf381), WTC(0xbf02c581), WTC(0xbef99901), WTC(0xbef06d01), WTC(0xbee74281), WTC(0xbede1901), WTC(0xbed4f081), WTC(0xbecbca81), WTC(0xbec2a781), WTC(0xbeb98681), WTC(0xbeb06881), WTC(0xbea74c81), WTC(0xbe9e3281), WTC(0xbe951a81), WTC(0xbe8c0501), WTC(0xbe82f301), WTC(0xbe79e481), WTC(0xbe70da01), WTC(0xbe67d381), WTC(0xbe5ed081), WTC(0xbe55d001), WTC(0xbe4cd381), WTC(0xbe43da81), WTC(0xbe3ae601), WTC(0xbe31f701), WTC(0xbe290d01), WTC(0xbe202801), WTC(0xbe174781), WTC(0xbe0e6c01), WTC(0xbe059481), WTC(0xbdfcc301), WTC(0xbdf3f701), WTC(0xbdeb3101), WTC(0xbde27201), WTC(0xbdd9b981), WTC(0xbdd10681), WTC(0xbdc85981), WTC(0xbdbfb281), WTC(0xbdb71201), WTC(0xbdae7881), WTC(0xbda5e601), WTC(0xbd9d5b81), WTC(0xbd94d801), WTC(0xbd8c5c01), WTC(0xbd83e681), WTC(0xbd7b7781), WTC(0xbd731081), WTC(0xbd6ab101), WTC(0xbd625981), WTC(0xbd5a0b01), WTC(0xbd51c481), WTC(0xbd498601), WTC(0xbd414f01), WTC(0xbd391f81), WTC(0xbd30f881), WTC(0xbd28d981), WTC(0xbd20c401), WTC(0xbd18b781), WTC(0xbd10b381), WTC(0xbd08b781), WTC(0xbd00c381), WTC(0xbcf8d781), WTC(0xbcf0f381), WTC(0xbce91801), WTC(0xbce14601), WTC(0xbcd97c81), WTC(0xbcd1bb81), WTC(0xbcca0301), WTC(0xbcc25181), WTC(0xbcbaa801), WTC(0xbcb30601), WTC(0xbcab6c01), WTC(0xbca3db01), WTC(0xbc9c5281), WTC(0xbc94d201), WTC(0xbc8d5901), WTC(0xbc85e801), WTC(0xbc7e7e01), WTC(0xbc771c01), WTC(0xbc6fc101), WTC(0xbc686e01), WTC(0xbc612301), WTC(0xbc59df81), WTC(0xbc52a381), WTC(0xbc4b6e81), WTC(0xbc444081), WTC(0xbc3d1801), WTC(0xbc35f501), WTC(0xbc2ed681), WTC(0xbc27bd81), WTC(0xbc20ae01), WTC(0xbc19ab01), WTC(0xbc12b801), WTC(0xbc0bcf81), WTC(0xbc04e381), WTC(0xbbfde481), WTC(0xbbf6c601), WTC(0xbbef9b81), WTC(0xbbe89901), WTC(0xbbe1f401), WTC(0xbbdbe201), WTC(0xbbd68c81), WTC(0xbbd21281), WTC(0xbbce9181), WTC(0xbbcc2681), WTC(0xbbcaca01), WTC(0xbbca5081), WTC(0xbbca8d01), WTC(0xbbcb5301), WTC(0xbbcc8201), WTC(0xbbce0601), WTC(0xbbcfca81), WTC(0xbbd1bd81), WTC(0xbbd3e101), WTC(0xbbd64d01), WTC(0xbbd91b81), WTC(0xbbdc6481), WTC(0xbbe03801), WTC(0xbbe49d01), WTC(0xbbe99981), WTC(0xbbef3301), WTC(0xbbf56181), WTC(0xbbfc0f81), WTC(0xbc032601), WTC(0xbc0a8f01), WTC(0xbc123b81), WTC(0xbc1a2401), WTC(0xbc224181), WTC(0xbc2a8c81), WTC(0xbc330781), WTC(0xbc3bbc01), WTC(0xbc44b481), WTC(0xbc4dfb81), WTC(0xbc57a301), WTC(0xbc61c401), WTC(0xbc6c7781), WTC(0xbc77d601), WTC(0xbc83f201), WTC(0xbc90d481), WTC(0xbc9e8801), WTC(0xbcad1501), WTC(0xbcbc7e01), WTC(0xbcccbd01), WTC(0xbcddcc81), WTC(0xbcefa601), WTC(0xbd023f01), WTC(0xbd158801), WTC(0xbd297181), WTC(0xbd3deb81), WTC(0xbd52eb01), WTC(0xbd686681), WTC(0xbd7e5581), WTC(0xbd94b001), WTC(0xbdab7181), WTC(0xbdc29a81), WTC(0xbdda2a01), WTC(0xbdf22181), WTC(0xbe0a8581), WTC(0xbe236001), WTC(0xbe3cbc01), WTC(0xbe56a381), WTC(0xbe712001), WTC(0xbe8c3781), WTC(0xbea7f301), WTC(0xbec45881), WTC(0xbee17201), WTC(0xbeff4801), WTC(0xbf1de601), WTC(0xbf3d5501), WTC(0xbf5d9a81), WTC(0xbf7eb581), WTC(0xbfa0a581), WTC(0xbfc36a01), WTC(0xbfe6ed01), WTC(0xc00b04c0), WTC(0xc02f86c0), WTC(0xc0544940), WTC(0xc0792ec0), WTC(0xc09e2640), WTC(0xc0c31f00), WTC(0xc0e80a00), WTC(0xc10cf480), WTC(0xc1320940), WTC(0xc15773c0), WTC(0xc17d5f00), WTC(0xc1a3e340), WTC(0xc1cb05c0), WTC(0xc1f2cbc0), WTC(0xc21b3940), WTC(0xc2444b00), WTC(0xc26df5c0), WTC(0xc2982d80), WTC(0xc2c2e640), WTC(0xc2ee0a00), WTC(0xc3197940), WTC(0xc34513c0), WTC(0xc370b9c0), WTC(0xc39c4f00), WTC(0xc3c7bc00), WTC(0xc3f2e940), WTC(0xc41dc140), WTC(0xc44856c0), WTC(0xc472e640), WTC(0xc49dad80), WTC(0xc4c8e880), WTC(0xc4f4acc0), WTC(0xc520e840), WTC(0xc54d8780), WTC(0xc57a76c0), WTC(0xc5a79640), WTC(0xc5d4bac0), WTC(0xc601b880), WTC(0xc62e6580), WTC(0xc65ab600), WTC(0xc686bd40), WTC(0xc6b28fc0), WTC(0xc6de41c0), WTC(0xc709de40), WTC(0xc7356640), WTC(0xc760da80), WTC(0xc78c3c40), WTC(0xc7b78640), WTC(0xc7e2afc0), WTC(0xc80dae80), WTC(0xc83878c0), WTC(0xc86304c0), WTC(0xc88d4900), WTC(0xc8b73b80), WTC(0xc8e0d280), WTC(0xc90a0440), /* part 1 */ WTC(0xb5212e81), WTC(0xb4959501), WTC(0xb40ab501), WTC(0xb3808d81), WTC(0xb2f71f01), WTC(0xb26e6881), WTC(0xb1e66a01), WTC(0xb15f2381), WTC(0xb0d89401), WTC(0xb052bc01), WTC(0xafcd9a81), WTC(0xaf492f01), WTC(0xaec57801), WTC(0xae427481), WTC(0xadc02281), WTC(0xad3e8101), WTC(0xacbd9081), WTC(0xac3d5001), WTC(0xabbdc001), WTC(0xab3edf01), WTC(0xaac0ad01), WTC(0xaa432981), WTC(0xa9c65401), WTC(0xa94a2c01), WTC(0xa8ceb201), WTC(0xa853e501), WTC(0xa7d9c681), WTC(0xa7605601), WTC(0xa6e79401), WTC(0xa66f8201), WTC(0xa5f81f81), WTC(0xa5816e81), WTC(0xa50b6e81), WTC(0xa4962181), WTC(0xa4218801), WTC(0xa3ada281), WTC(0xa33a7201), WTC(0xa2c7f801), WTC(0xa2563501), WTC(0xa1e52a81), WTC(0xa174da81), WTC(0xa1054701), WTC(0xa0967201), WTC(0xa0285d81), WTC(0x9fbb0981), WTC(0x9f4e7801), WTC(0x9ee2a901), WTC(0x9e779f81), WTC(0x9e0d5e01), WTC(0x9da3e601), WTC(0x9d3b3b81), WTC(0x9cd35f81), WTC(0x9c6c5481), WTC(0x9c061b81), WTC(0x9ba0b701), WTC(0x9b3c2801), WTC(0x9ad87081), WTC(0x9a759301), WTC(0x9a139101), WTC(0x99b26c81), WTC(0x99522801), WTC(0x98f2c601), WTC(0x98944901), WTC(0x9836b201), WTC(0x97da0481), WTC(0x977e4181), WTC(0x97236b01), WTC(0x96c98381), WTC(0x96708b81), WTC(0x96188501), WTC(0x95c17081), WTC(0x956b4f81), WTC(0x95162381), WTC(0x94c1ee01), WTC(0x946eaf81), WTC(0x941c6901), WTC(0x93cb1c81), WTC(0x937acb01), WTC(0x932b7501), WTC(0x92dd1b01), WTC(0x928fbe01), WTC(0x92435d01), WTC(0x91f7f981), WTC(0x91ad9281), WTC(0x91642781), WTC(0x911bb981), WTC(0x90d44781), WTC(0x908dd101), WTC(0x90485401), WTC(0x9003ce81), WTC(0x8fc03f01), WTC(0x8f7da401), WTC(0x8f3bfb01), WTC(0x8efb4181), WTC(0x8ebb7581), WTC(0x8e7c9301), WTC(0x8e3e9481), WTC(0x8e017581), WTC(0x8dc53001), WTC(0x8d89be81), WTC(0x8d4f1b01), WTC(0x8d154081), WTC(0x8cdc2901), WTC(0x8ca3cb01), WTC(0x8c6c1b01), WTC(0x8c350d01), WTC(0x8bfe9401), WTC(0x8bc8a401), WTC(0x8b933001), WTC(0x8b5e2c81), WTC(0x8b298b81), WTC(0x8af53e81), WTC(0x8ac13381), WTC(0x8a8d5801), WTC(0x8a599a81), WTC(0x8a25f301), WTC(0x89f26101), WTC(0x89bee581), WTC(0x898b8301), WTC(0x89586901), WTC(0x8925f101), WTC(0x88f47901), WTC(0x88c45e81), WTC(0x88962981), WTC(0x886a8a81), WTC(0x88423301), WTC(0x881dd301), WTC(0x87fdd781), WTC(0x87d0ca81), WTC(0x87c76201), WTC(0x87bcab81), WTC(0x87b0ef01), WTC(0x87a48b01), WTC(0x8797dd81), WTC(0x878b4301), WTC(0x877ede01), WTC(0x87729701), WTC(0x87665481), WTC(0x8759fd01), WTC(0x874d8681), WTC(0x8740f681), WTC(0x87345381), WTC(0x8727a381), WTC(0x871ae981), WTC(0x870e2301), WTC(0x87014f81), WTC(0x86f46d81), WTC(0x86e77b81), WTC(0x86da7901), WTC(0x86cd6681), WTC(0x86c04381), WTC(0x86b30f01), WTC(0x86a5ca81), WTC(0x86987581), WTC(0x868b1001), WTC(0x867d9a81), WTC(0x86701381), WTC(0x86627b01), WTC(0x8654d001), WTC(0x86471281), WTC(0x86394301), WTC(0x862b6201), WTC(0x861d7081), WTC(0x860f6e01), WTC(0x86015981), WTC(0x85f33281), WTC(0x85e4f801), WTC(0x85d6a981), WTC(0x85c84801), WTC(0x85b9d481), WTC(0x85ab4f01), WTC(0x859cb781), WTC(0x858e0e01), WTC(0x857f5101), WTC(0x85707f81), WTC(0x85619a01), WTC(0x8552a181), WTC(0x85439601), WTC(0x85347901), WTC(0x85254a81), WTC(0x85160981), WTC(0x8506b581), WTC(0x84f74e01), WTC(0x84e7d381), WTC(0x84d84601), WTC(0x84c8a701), WTC(0x84b8f801), WTC(0x84a93801), WTC(0x84996701), WTC(0x84898481), WTC(0x84798f81), WTC(0x84698881), WTC(0x84597081), WTC(0x84494881), WTC(0x84391081), WTC(0x8428ca01), WTC(0x84187401), WTC(0x84080d81), WTC(0x83f79681), WTC(0x83e70f01), WTC(0x83d67881), WTC(0x83c5d381), WTC(0x83b52101), WTC(0x83a46181), WTC(0x83939501), WTC(0x8382ba01), WTC(0x8371d081), WTC(0x8360d901), WTC(0x834fd481), WTC(0x833ec381), WTC(0x832da781), WTC(0x831c8101), WTC(0x830b4f81), WTC(0x82fa1181), WTC(0x82e8c801), WTC(0x82d77201), WTC(0x82c61101), WTC(0x82b4a601), WTC(0x82a33281), WTC(0x8291b601), WTC(0x82803101), WTC(0x826ea201), WTC(0x825d0901), WTC(0x824b6601), WTC(0x8239b981), WTC(0x82280581), WTC(0x82164a81), WTC(0x82048881), WTC(0x81f2bf81), WTC(0x81e0ee81), WTC(0x81cf1581), WTC(0x81bd3401), WTC(0x81ab4b01), WTC(0x81995c01), WTC(0x81876781), WTC(0x81756d81), WTC(0x81636d81), WTC(0x81516701), WTC(0x813f5981), WTC(0x812d4481), WTC(0x811b2981), WTC(0x81090981), WTC(0x80f6e481), WTC(0x80e4bb81), WTC(0x80d28d81), WTC(0x80c05a01), WTC(0x80ae1f81), WTC(0x809bdf01), WTC(0x80899881), WTC(0x80774c81), WTC(0x8064fc81), WTC(0x8052a881), WTC(0x80405101), WTC(0x802df701), WTC(0x801b9b01), WTC(0x80093e01), WTC(0x0a74b120), WTC(0x0aa08a90), WTC(0x0acd2b80), WTC(0x0afa8860), WTC(0x0b289590), WTC(0x0b574790), WTC(0x0b8692d0), WTC(0x0bb66bb0), WTC(0x0be6c6b0), WTC(0x0c179830), WTC(0x0c48d500), WTC(0x0c7a7ad0), WTC(0x0cac9000), WTC(0x0cdf1b60), WTC(0x0d122390), WTC(0x0d45a8f0), WTC(0x0d79a5e0), WTC(0x0dae1480), WTC(0x0de2ef30), WTC(0x0e183800), WTC(0x0e4df8c0), WTC(0x0e843b90), WTC(0x0ebb0a20), WTC(0x0ef26430), WTC(0x0f2a3fc0), WTC(0x0f629280), WTC(0x0f9b5210), WTC(0x0fd47690), WTC(0x100dfa80), WTC(0x1047d8a0), WTC(0x10820b40), WTC(0x10bc8b80), WTC(0x10f75080), WTC(0x11325100), WTC(0x116d84e0), WTC(0x11a8ece0), WTC(0x11e49420), WTC(0x122085a0), WTC(0x125ccbc0), WTC(0x12995a40), WTC(0x12d60e80), WTC(0x1312c4c0), WTC(0x134f59e0), WTC(0x138bae60), WTC(0x13c7a740), WTC(0x140329e0), WTC(0x143e1b60), WTC(0x147862a0), WTC(0x14b1e840), WTC(0x14ea94c0), WTC(0x152250a0), WTC(0x15590380), WTC(0x158e93e0), WTC(0x15c2e820), WTC(0x15f5e6e0), WTC(0x162779a0), WTC(0x16578ca0), WTC(0x16860ca0), WTC(0x16b2e640), WTC(0x16de0b00), WTC(0x17077140), WTC(0x172f0fa0), WTC(0x1754e200), WTC(0x17796080), WTC(0x179d7f20), WTC(0x17c23760), WTC(0x17e87da0), WTC(0x1810cc80), WTC(0x183b25a0), WTC(0x18678520), WTC(0x1895e700), WTC(0x18c64540), WTC(0x18f89780), WTC(0x192cd560), WTC(0x1962f680), WTC(0x199af2a0), WTC(0x19d4c1e0), WTC(0x1a105ca0), WTC(0x1a4dbae0), WTC(0x1a8cd660), WTC(0x1acdaa60), WTC(0x1b103260), WTC(0x1b546940), WTC(0x1b9a4600), WTC(0x1be1bb80), WTC(0x1c2abc60), WTC(0x1c753b80), WTC(0x1cc13860), WTC(0x1d0ebe20), WTC(0x1d5dd8c0), WTC(0x1dae9480), WTC(0x1e010060), WTC(0x1e552f40), WTC(0x1eab33e0), WTC(0x1f032060), WTC(0x1f5cfce0), WTC(0x1fb8c660), WTC(0x201679c0), WTC(0x207611c0), WTC(0x20d75f00), WTC(0x213a0640), WTC(0x219dab80), WTC(0x2201f480), WTC(0x2266ba80), WTC(0x22cc0ac0), WTC(0x2331f4c0), WTC(0x23988940), WTC(0x23ffff40), WTC(0x2468b340), WTC(0x24d30300), WTC(0x253f4900), WTC(0x25ad8980), WTC(0x261d72c0), WTC(0x268eaec0), WTC(0x2700e880), WTC(0x2773db40), WTC(0x27e751c0), WTC(0x285b1780), WTC(0x28cefbc0), WTC(0x29431f80), WTC(0x29b7f680), WTC(0x2a2df780), WTC(0x2aa59880), WTC(0x2b1f3280), WTC(0x2b9b0140), WTC(0x2c194000), WTC(0x2c9a2540), WTC(0x2d1d8dc0), WTC(0x2da2fc40), WTC(0x2e29ee80), WTC(0x2eb1e340), WTC(0x2f3a4e40), WTC(0x2fc29980), WTC(0x304a2ec0), WTC(0x30d07cc0), WTC(0x315566c0), WTC(0x31d94480), WTC(0x325c72c0), WTC(0x32df51c0), WTC(0x33628c80), WTC(0x33e71a00), WTC(0x346df400), WTC(0x34f80dc0), WTC(0x3585c640), WTC(0x3616e700), WTC(0x36ab3380), WTC(0x37426ac0), WTC(0x37dbe840), WTC(0x3876a340), WTC(0x39118f40), WTC(0x39aba2c0), WTC(0x3a4422c0), WTC(0x3adaa200), WTC(0x3b6eb6c0), WTC(0x3bfffd80), WTC(0x3c8e9380), WTC(0x3d1b1780), WTC(0x3da62e00), WTC(0x3e307b00), WTC(0x3eba97c0), WTC(0x3f451280), WTC(0x3fd07940), WTC(0x405d577f), WTC(0x40ebf57f), WTC(0x417c59ff), WTC(0x420e897f), WTC(0x42a2857f), WTC(0x4338307f), WTC(0x43cf4d7f), WTC(0x44679cff), WTC(0x4500dfff), WTC(0x459ac2ff), WTC(0x4634e2ff), WTC(0x46ced9ff), WTC(0x4768437f), WTC(0x4800d27f), WTC(0x489850ff), WTC(0x492e88ff), WTC(0x49c346ff), WTC(0x4a5678ff), WTC(0x4ae82f7f), WTC(0x4b787c7f), WTC(0x4c07717f), WTC(0x4c95337f), WTC(0x4d21f77f), WTC(0x4dadf3ff), WTC(0x4e395eff), WTC(0x4ec4657f), WTC(0x4f4f297f), WTC(0x4fd9cd7f), WTC(0x5064737f), WTC(0x50ef3cff), WTC(0x517a46ff), WTC(0x5205b0ff), WTC(0x529197ff), WTC(0x531e04ff), WTC(0x53aaeb7f), WTC(0x54383eff), WTC(0x54c5ef7f), WTC(0x5553a8ff), WTC(0x55e0d57f), WTC(0x566cda7f), WTC(0x56f720ff), WTC(0x577f4aff), WTC(0x580534ff), WTC(0x5888bd7f), WTC(0x5909c6ff), WTC(0x598890ff), WTC(0x5a05b7ff), WTC(0x5a81db7f), WTC(0x5afd99ff), WTC(0x5b794a7f), WTC(0x5bf5007f), WTC(0x5c70cbff), WTC(0x5cecbb7f), WTC(0x5d68c47f), WTC(0x5de4c3ff), WTC(0x5e6094ff), WTC(0x5edc127f), WTC(0x5f56fdff), WTC(0x5fd1017f), WTC(0x6049c67f), WTC(0x60c0f67f), WTC(0x613650ff), WTC(0x61a9a9ff), WTC(0x621ad77f), WTC(0x6289b37f), WTC(0x62f67fff), WTC(0x6361e87f), WTC(0x63cc9bff), WTC(0x6437457f), WTC(0x64a2247f), WTC(0x650d0c7f), WTC(0x6577cc7f), WTC(0x65e2327f), WTC(0x664bf57f), WTC(0x66b4b5ff), WTC(0x671c137f), WTC(0x6781afff), WTC(0x67e579ff), WTC(0x6847abff), WTC(0x68a882ff), WTC(0x69083bff), WTC(0x6966fbff), WTC(0x69c4cfff), WTC(0x6a21c57f), WTC(0x6a7de87f), WTC(0x6ad9377f), WTC(0x6b33a5ff), WTC(0x6b8d257f), WTC(0x6be5a8ff), WTC(0x6c3d20ff), WTC(0x6c9380ff), WTC(0x6ce8ba7f), WTC(0x6d3cbfff), WTC(0x6d8f827f), /* part 2 */ WTC(0xad98b481), WTC(0xaead9d01), WTC(0xafbfc381), WTC(0xb0cf4d01), WTC(0xb1dc5f81), WTC(0xb2e72081), WTC(0xb3efb501), WTC(0xb4f64381), WTC(0xb5faf101), WTC(0xb6fde401), WTC(0xb7ff4001), WTC(0xb8ff1601), WTC(0xb9fd6181), WTC(0xbafa1d01), WTC(0xbbf54401), WTC(0xbceed101), WTC(0xbde6c081), WTC(0xbedd0e81), WTC(0xbfd1b701), WTC(0xc0c4b440), WTC(0xc1b5ffc0), WTC(0xc2a59340), WTC(0xc3936780), WTC(0xc47f78c0), WTC(0xc569c600), WTC(0xc6524d40), WTC(0xc7390dc0), WTC(0xc81e04c0), WTC(0xc9012e00), WTC(0xc9e28540), WTC(0xcac20700), WTC(0xcb9fb1c0), WTC(0xcc7b8640), WTC(0xcd558600), WTC(0xce2db200), WTC(0xcf0409c0), WTC(0xcfd88a40), WTC(0xd0ab3080), WTC(0xd17bfa00), WTC(0xd24ae640), WTC(0xd317f7c0), WTC(0xd3e33080), WTC(0xd4ac9340), WTC(0xd5741f40), WTC(0xd639d2c0), WTC(0xd6fdab00), WTC(0xd7bfa5c0), WTC(0xd87fc300), WTC(0xd93e0600), WTC(0xd9fa7180), WTC(0xdab50900), WTC(0xdb6dccc0), WTC(0xdc24ba80), WTC(0xdcd9d000), WTC(0xdd8d0b80), WTC(0xde3e6dc0), WTC(0xdeedf9c0), WTC(0xdf9bb340), WTC(0xe0479e20), WTC(0xe0f1bac0), WTC(0xe19a07e0), WTC(0xe2408380), WTC(0xe2e52c00), WTC(0xe38802e0), WTC(0xe4290c00), WTC(0xe4c84c20), WTC(0xe565c760), WTC(0xe6017f20), WTC(0xe69b7240), WTC(0xe7339f60), WTC(0xe7ca0500), WTC(0xe85ea480), WTC(0xe8f18180), WTC(0xe9829fc0), WTC(0xea1202e0), WTC(0xea9fab80), WTC(0xeb2b9700), WTC(0xebb5c2a0), WTC(0xec3e2bc0), WTC(0xecc4d300), WTC(0xed49bc80), WTC(0xedccec60), WTC(0xee4e66a0), WTC(0xeece2d80), WTC(0xef4c41e0), WTC(0xefc8a480), WTC(0xf0435610), WTC(0xf0bc5c60), WTC(0xf133c230), WTC(0xf1a99270), WTC(0xf21dd7b0), WTC(0xf29097e0), WTC(0xf301d3d0), WTC(0xf3718c20), WTC(0xf3dfc180), WTC(0xf44c7100), WTC(0xf4b79480), WTC(0xf52125b0), WTC(0xf5891df0), WTC(0xf5ef6fe0), WTC(0xf6540730), WTC(0xf6b6cf50), WTC(0xf717b490), WTC(0xf776b9a0), WTC(0xf7d3f720), WTC(0xf82f86e8), WTC(0xf8898260), WTC(0xf8e1fc50), WTC(0xf93900f0), WTC(0xf98e9c28), WTC(0xf9e2d940), WTC(0xfa35b4a0), WTC(0xfa871bd8), WTC(0xfad6fbd0), WTC(0xfb254250), WTC(0xfb71f0c0), WTC(0xfbbd1c28), WTC(0xfc06da60), WTC(0xfc4f40a4), WTC(0xfc965500), WTC(0xfcdc0e5c), WTC(0xfd2062f4), WTC(0xfd6348d0), WTC(0xfda4b1b8), WTC(0xfde48b2c), WTC(0xfe22c280), WTC(0xfe5f462a), WTC(0xfe9a1f2e), WTC(0xfed3711c), WTC(0xff0b60ac), WTC(0xff4212dd), WTC(0xff77b344), WTC(0xffac7407), WTC(0xffe08796), WTC(0x00141e37), WTC(0x00473665), WTC(0x00799cd0), WTC(0x00ab1bff), WTC(0x00db7d8b), WTC(0x010a75ea), WTC(0x0137a46e), WTC(0x0162a77a), WTC(0x018b20ac), WTC(0x01b0fb7a), WTC(0x01d46d3c), WTC(0x01f5ae7c), WTC(0x0214f91c), WTC(0x0232a5cc), WTC(0x024f2c04), WTC(0x026b048c), WTC(0x0286a628), WTC(0x02a25808), WTC(0x02be31c0), WTC(0x02da48e0), WTC(0x02f6b09c), WTC(0x031345dc), WTC(0x032faf50), WTC(0x034b9148), WTC(0x036690e8), WTC(0x0380658c), WTC(0x0398d8e4), WTC(0x03afb568), WTC(0x03c4c6e0), WTC(0x03d7f770), WTC(0x03e94f9c), WTC(0x03f8d938), WTC(0x04069ee8), WTC(0x0412bef8), WTC(0x041d6b30), WTC(0x0426d638), WTC(0x042f3288), WTC(0x0436ad98), WTC(0x043d6fd0), WTC(0x0443a170), WTC(0x04496a40), WTC(0x044ee728), WTC(0x04542a40), WTC(0x04594520), WTC(0x045e4890), WTC(0x04633210), WTC(0x0467ebe8), WTC(0x046c5f80), WTC(0x04707630), WTC(0x047417f0), WTC(0x04772b58), WTC(0x047996e8), WTC(0x047b4140), WTC(0x047c12a0), WTC(0x047bf520), WTC(0x047ad2e0), WTC(0x04789690), WTC(0x047539c8), WTC(0x0470c4b8), WTC(0x046b4058), WTC(0x0464b600), WTC(0x045d3a08), WTC(0x0454ebc8), WTC(0x044beb00), WTC(0x04425798), WTC(0x043853b0), WTC(0x042e0398), WTC(0x04238bd8), WTC(0x04190f98), WTC(0x040e9670), WTC(0x04040c18), WTC(0x03f95b30), WTC(0x03ee6e20), WTC(0x03e32b64), WTC(0x03d77598), WTC(0x03cb2f24), WTC(0x03be3b18), WTC(0x03b08b18), WTC(0x03a21f64), WTC(0x0392f8d4), WTC(0x038318e0), WTC(0x03728e94), WTC(0x03617694), WTC(0x034fee18), WTC(0x033e11f4), WTC(0x032bf530), WTC(0x0319a114), WTC(0x03071e80), WTC(0x02f475f4), WTC(0x02e1a7c0), WTC(0x02ceac04), WTC(0x02bb7a84), WTC(0x02a80af0), WTC(0x029452b0), WTC(0x028044e0), WTC(0x026bd488), WTC(0x0256f558), WTC(0x0241a940), WTC(0x022c0084), WTC(0x02160c08), WTC(0x01ffdc5a), WTC(0x01e97ad2), WTC(0x01d2e982), WTC(0x01bc2a2a), WTC(0x01a53e8c), WTC(0x018e2860), WTC(0x0176e94c), WTC(0x015f82fa), WTC(0x0147f70e), WTC(0x013046c2), WTC(0x011872e8), WTC(0x01007c4a), WTC(0x00e863cf), WTC(0x00d02c81), WTC(0x00b7db94), WTC(0x009f7651), WTC(0x00870204), WTC(0x006e83f8), WTC(0x00560176), WTC(0x003d7fcb), WTC(0x0025043f), WTC(0x000c941f), WTC(0xd65574c0), WTC(0xd5ebc100), WTC(0xd582d080), WTC(0xd51a9cc0), WTC(0xd4b31f80), WTC(0xd44c5280), WTC(0xd3e62f80), WTC(0xd380b040), WTC(0xd31bce40), WTC(0xd2b78380), WTC(0xd253ca40), WTC(0xd1f0acc0), WTC(0xd18e4580), WTC(0xd12caf40), WTC(0xd0cc0400), WTC(0xd06c40c0), WTC(0xd00d4740), WTC(0xcfaef6c0), WTC(0xcf513140), WTC(0xcef3fa80), WTC(0xce977a40), WTC(0xce3bd980), WTC(0xcde13f40), WTC(0xcd87a880), WTC(0xcd2ee800), WTC(0xccd6cf00), WTC(0xcc7f2f40), WTC(0xcc27e880), WTC(0xcbd0ea00), WTC(0xcb7a2380), WTC(0xcb238380), WTC(0xcaccee80), WTC(0xca763ec0), WTC(0xca1f4d00), WTC(0xc9c7f480), WTC(0xc9703b40), WTC(0xc9185200), WTC(0xc8c06b00), WTC(0xc868b4c0), WTC(0xc81100c0), WTC(0xc7b8c280), WTC(0xc75f6a40), WTC(0xc7046900), WTC(0xc6a74340), WTC(0xc6479300), WTC(0xc5e4f200), WTC(0xc57efac0), WTC(0xc5154880), WTC(0xc4a77780), WTC(0xc4352440), WTC(0xc3bdeac0), WTC(0xc3416740), WTC(0xc2bf33c0), WTC(0xc236eb40), WTC(0xc1a82900), WTC(0xc11290c0), WTC(0xc075cf00), WTC(0xbfd19081), WTC(0xbf258401), WTC(0xbe716d81), WTC(0xbdb52b81), WTC(0xbcf09a81), WTC(0xbc23af81), WTC(0xbb505c01), WTC(0xba7a9081), WTC(0xb9a65281), WTC(0xb8d79301), WTC(0xb8104c01), WTC(0xb7508181), WTC(0xb6982201), WTC(0xb5e71b01), WTC(0xb53d5b01), WTC(0xb49ad081), WTC(0xb3ff6901), WTC(0xb36b1301), WTC(0xb2ddbd01), WTC(0xb2575481), WTC(0xb1d7c801), WTC(0xb15f0601), WTC(0xb0ecfc01), WTC(0xb0819881), WTC(0xb01cca01), WTC(0xafbe7e01), WTC(0xaf66a301), WTC(0xaf152701), WTC(0xaec9f881), WTC(0xae850601), WTC(0xae463c81), WTC(0xae0d8b01), WTC(0xaddae001), WTC(0xadae2881), WTC(0xad875381), WTC(0xad664f81), WTC(0xad4b0981), WTC(0xad357081), WTC(0xad257301), WTC(0xad1afe01), WTC(0xad160081), WTC(0xad166901), WTC(0xad1c2481), WTC(0xad272201), WTC(0xad374f81), WTC(0xad4c9b01), WTC(0xad66f381), WTC(0xad864601), WTC(0xadaa8101), WTC(0xadd39301), WTC(0xae016a01), WTC(0xae33f481), WTC(0xae6b2001), WTC(0xaea6db01), WTC(0xaee71381), WTC(0xaf2bb801), WTC(0xaf74b681), WTC(0xafc1fd01), WTC(0xb0137a01), WTC(0xb0691b81), WTC(0xb0c2cf81), WTC(0xb1208481), WTC(0xb1822881), WTC(0xb1e7a981), WTC(0xb250f601), WTC(0xb2bdfc01), WTC(0xb32eaa01), WTC(0xb3a2ed01), WTC(0xb41ab481), WTC(0xb495ee01), WTC(0xb5148801), WTC(0xb5967081), WTC(0xb61b9581), WTC(0xb6a3e581), WTC(0xb72f4e01), WTC(0xb7bdbe01), WTC(0xb84f2381), WTC(0xb8e36c81), WTC(0xb97a8701), WTC(0xba146101), WTC(0xbab0e981), WTC(0xbb500d81), WTC(0xbbf1bc81), WTC(0xbc95e381), WTC(0xbd3c7181), WTC(0xbde55481), WTC(0xbe907a01), WTC(0xbf3dd101), WTC(0xbfed4701), WTC(0xc09ecac0), WTC(0xc1524a00), WTC(0xc207b300), WTC(0xc2bef440), WTC(0xc377fb80), WTC(0xc432b700), WTC(0xc4ef1500), WTC(0xc5ad03c0), WTC(0xc66c7140), WTC(0xc72d4bc0), WTC(0xc7ef8180), WTC(0xc8b30080), WTC(0xc977b700), WTC(0xca3d9340), WTC(0xcb048340), WTC(0xcbcc7540), WTC(0xcc955740), WTC(0xcd5f17c0), WTC(0xce29a480), WTC(0xcef4ec00), WTC(0xcfc0dc80), WTC(0xd08d63c0), WTC(0xd15a7040), WTC(0xd227f000), WTC(0xd2f5d140), WTC(0xd3c40240), WTC(0xd4927100), WTC(0xd5610b80), WTC(0xd62fc080), WTC(0xd6fe7dc0), WTC(0xd7cd3140), WTC(0xd89bc980), WTC(0xd96a34c0), WTC(0xda3860c0), WTC(0xdb063c00), WTC(0xdbd3b480), WTC(0xdca0b880), WTC(0xdd6d3640), WTC(0xde391bc0), WTC(0xdf045740), WTC(0xdfced6c0), WTC(0xe09888c0), WTC(0xe1615b20), WTC(0xe2293c20), WTC(0xe2f01a00), WTC(0xe3b5e2c0), WTC(0xe47a84c0), WTC(0xe53dee00), WTC(0xe6000cc0), WTC(0xe6c0cf20), WTC(0xe7802360), WTC(0xe83df7a0), WTC(0xe8fa39e0), WTC(0xe9b4d880), WTC(0xea6dc1a0), WTC(0xeb24e360), WTC(0xebda2be0), WTC(0xec8d8960), WTC(0xed3eea20), WTC(0xedee3c00), WTC(0xee9b6d80), WTC(0xef466ca0), WTC(0xefef2780), WTC(0xf0958c50), WTC(0xf1398950), WTC(0xf1db0ca0), WTC(0xf27a0470), WTC(0xf3165ed0), WTC(0xf3b00a10), WTC(0xf446f440), WTC(0xf4db0b90), WTC(0xf56c3e30), WTC(0xf5fa7a50), WTC(0xf685ae10), WTC(0xf70dc7a0), WTC(0xf792b520), WTC(0xf81464c8), WTC(0xf892c4c0), WTC(0xf90dc330), WTC(0xf9854e40), WTC(0xf9f95418), WTC(0xfa69c2f0), WTC(0xfad688e8), WTC(0xfb3f9428), WTC(0xfba4d2e8), WTC(0xfc063344), WTC(0xfc63a370), WTC(0xfcbd1194), WTC(0xfd126bdc), WTC(0xfd63a06c), WTC(0xfdb09d78), WTC(0xfdf95124), WTC(0xfe3da99e), WTC(0xfe7d950e), WTC(0xfeb901a2), WTC(0xfeefdd80), WTC(0xff2216d7), WTC(0xff4f9bcf), WTC(0xff785a93), WTC(0xff9c414e), WTC(0xffbb3e2b), WTC(0xffd53f54), WTC(0xffea32f4), WTC(0xfffa0735), WTC(0x0004aa43), WTC(0x000a0a47), WTC(0x000a156c), WTC(0x0004b9de), WTC(0xfff9e5c5), WTC(0xffe9874e)}; const FIXP_WTB LowDelaySynthesis480[1440] = { WTC(0xdad2e6c0), WTC(0xdb1da900), WTC(0xdb68ce40), WTC(0xdbb45840), WTC(0xdc004940), WTC(0xdc4ca280), WTC(0xdc996500), WTC(0xdce69140), WTC(0xdd342780), WTC(0xdd822700), WTC(0xddd08a80), WTC(0xde1f4d00), WTC(0xde6e6ec0), WTC(0xdebdec40), WTC(0xdf0dba80), WTC(0xdf5dd540), WTC(0xdfae3cc0), WTC(0xdfff0500), WTC(0xe0505140), WTC(0xe0a22980), WTC(0xe0f488e0), WTC(0xe1476180), WTC(0xe19aa480), WTC(0xe1ee4d80), WTC(0xe2425400), WTC(0xe29689a0), WTC(0xe2eacd60), WTC(0xe33f2420), WTC(0xe393a300), WTC(0xe3e87f20), WTC(0xe43dcee0), WTC(0xe4938a80), WTC(0xe4e9b0a0), WTC(0xe5404300), WTC(0xe5973e60), WTC(0xe5ee9b80), WTC(0xe64649e0), WTC(0xe69e37e0), WTC(0xe6f65ec0), WTC(0xe74eb6c0), WTC(0xe7a73000), WTC(0xe7ffbe40), WTC(0xe8585ee0), WTC(0xe8b10740), WTC(0xe9099c40), WTC(0xe96214e0), WTC(0xe9ba79a0), WTC(0xea12e7c0), WTC(0xea6b89c0), WTC(0xeac46580), WTC(0xeb1d7260), WTC(0xeb76b620), WTC(0xebd036c0), WTC(0xec29e520), WTC(0xec83aa60), WTC(0xecdd5a00), WTC(0xed36d500), WTC(0xed901540), WTC(0xede91160), WTC(0xee41bc20), WTC(0xee9a0ee0), WTC(0xeef20860), WTC(0xef49a7e0), WTC(0xefa0ec00), WTC(0xeff7d1c0), WTC(0xf04e56b0), WTC(0xf0a476e0), WTC(0xf0fa2f60), WTC(0xf14f80e0), WTC(0xf1a46e10), WTC(0xf1f8fe80), WTC(0xf24d34a0), WTC(0xf2a10bb0), WTC(0xf2f48210), WTC(0xf3479cc0), WTC(0xf39a5be0), WTC(0xf3ecb8f0), WTC(0xf43eafa0), WTC(0xf4903b50), WTC(0xf4e14e80), WTC(0xf531d6a0), WTC(0xf581bc10), WTC(0xf5d0e9c0), WTC(0xf61f5250), WTC(0xf66ce6e0), WTC(0xf6b99330), WTC(0xf7054eb0), WTC(0xf7501f20), WTC(0xf79a0750), WTC(0xf7e30700), WTC(0xf82b2fc0), WTC(0xf872a138), WTC(0xf8b97f18), WTC(0xf8ffe668), WTC(0xf945e538), WTC(0xf98b8860), WTC(0xf9d0f380), WTC(0xfa165148), WTC(0xfa5bb8a8), WTC(0xfaa13df8), WTC(0xfae6fb00), WTC(0xfb2cf8c8), WTC(0xfb732a80), WTC(0xfbb97910), WTC(0xfbffcd10), WTC(0xfc463478), WTC(0xfc8cd3fc), WTC(0xfcd3be5c), WTC(0xfd1afa90), WTC(0xfd62aa84), WTC(0xfdab0288), WTC(0xfdf404b4), WTC(0xfe3d3006), WTC(0xfe85b20e), WTC(0xfecca4cc), WTC(0xff10d559), WTC(0xff50579b), WTC(0xff8a40d2), WTC(0xffb7d86e), WTC(0xffef6bbb), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0xbff67a01), WTC(0xbfecaa81), WTC(0xbfe2d901), WTC(0xbfd90601), WTC(0xbfcf3181), WTC(0xbfc55c81), WTC(0xbfbb8701), WTC(0xbfb1b101), WTC(0xbfa7dc01), WTC(0xbf9e0701), WTC(0xbf943301), WTC(0xbf8a5f81), WTC(0xbf808b81), WTC(0xbf76b701), WTC(0xbf6ce201), WTC(0xbf630d81), WTC(0xbf593a01), WTC(0xbf4f6801), WTC(0xbf459681), WTC(0xbf3bc601), WTC(0xbf31f501), WTC(0xbf282501), WTC(0xbf1e5501), WTC(0xbf148681), WTC(0xbf0aba01), WTC(0xbf00ef81), WTC(0xbef72681), WTC(0xbeed5f01), WTC(0xbee39801), WTC(0xbed9d281), WTC(0xbed00f81), WTC(0xbec64e81), WTC(0xbebc9181), WTC(0xbeb2d681), WTC(0xbea91f01), WTC(0xbe9f6901), WTC(0xbe95b581), WTC(0xbe8c0501), WTC(0xbe825801), WTC(0xbe78b001), WTC(0xbe6f0c01), WTC(0xbe656c01), WTC(0xbe5bd001), WTC(0xbe523781), WTC(0xbe48a301), WTC(0xbe3f1381), WTC(0xbe358901), WTC(0xbe2c0501), WTC(0xbe228681), WTC(0xbe190d81), WTC(0xbe0f9a01), WTC(0xbe062b81), WTC(0xbdfcc301), WTC(0xbdf36101), WTC(0xbdea0681), WTC(0xbde0b301), WTC(0xbdd76701), WTC(0xbdce2181), WTC(0xbdc4e301), WTC(0xbdbbab01), WTC(0xbdb27b01), WTC(0xbda95301), WTC(0xbda03381), WTC(0xbd971c81), WTC(0xbd8e0e01), WTC(0xbd850701), WTC(0xbd7c0781), WTC(0xbd731081), WTC(0xbd6a2201), WTC(0xbd613d81), WTC(0xbd586281), WTC(0xbd4f9101), WTC(0xbd46c801), WTC(0xbd3e0801), WTC(0xbd355081), WTC(0xbd2ca281), WTC(0xbd23ff01), WTC(0xbd1b6501), WTC(0xbd12d581), WTC(0xbd0a4f81), WTC(0xbd01d281), WTC(0xbcf95e81), WTC(0xbcf0f381), WTC(0xbce89281), WTC(0xbce03b81), WTC(0xbcd7ef01), WTC(0xbccfac01), WTC(0xbcc77181), WTC(0xbcbf4001), WTC(0xbcb71701), WTC(0xbcaef701), WTC(0xbca6e101), WTC(0xbc9ed481), WTC(0xbc96d101), WTC(0xbc8ed701), WTC(0xbc86e581), WTC(0xbc7efc81), WTC(0xbc771c01), WTC(0xbc6f4401), WTC(0xbc677501), WTC(0xbc5fae81), WTC(0xbc57f101), WTC(0xbc503b81), WTC(0xbc488e81), WTC(0xbc40e881), WTC(0xbc394901), WTC(0xbc31af01), WTC(0xbc2a1a81), WTC(0xbc228f01), WTC(0xbc1b1081), WTC(0xbc13a481), WTC(0xbc0c4581), WTC(0xbc04e381), WTC(0xbbfd6c01), WTC(0xbbf5d181), WTC(0xbbee2f81), WTC(0xbbe6c801), WTC(0xbbdfdb81), WTC(0xbbd9a781), WTC(0xbbd45881), WTC(0xbbd01301), WTC(0xbbccfc81), WTC(0xbbcb2281), WTC(0xbbca5d01), WTC(0xbbca7481), WTC(0xbbcb3201), WTC(0xbbcc6b01), WTC(0xbbce0601), WTC(0xbbcfea81), WTC(0xbbd20301), WTC(0xbbd45601), WTC(0xbbd70201), WTC(0xbbda2501), WTC(0xbbdddb01), WTC(0xbbe23281), WTC(0xbbe73201), WTC(0xbbece281), WTC(0xbbf34281), WTC(0xbbfa3c01), WTC(0xbc01b381), WTC(0xbc098d81), WTC(0xbc11b681), WTC(0xbc1a2401), WTC(0xbc22cd81), WTC(0xbc2bab01), WTC(0xbc34c081), WTC(0xbc3e1981), WTC(0xbc47c281), WTC(0xbc51cb01), WTC(0xbc5c4c81), WTC(0xbc676501), WTC(0xbc733401), WTC(0xbc7fd301), WTC(0xbc8d5101), WTC(0xbc9bb901), WTC(0xbcab1781), WTC(0xbcbb7001), WTC(0xbcccbd01), WTC(0xbcdef701), WTC(0xbcf21601), WTC(0xbd060c81), WTC(0xbd1ac801), WTC(0xbd303581), WTC(0xbd464281), WTC(0xbd5ce281), WTC(0xbd740b81), WTC(0xbd8bb281), WTC(0xbda3d081), WTC(0xbdbc6381), WTC(0xbdd56b81), WTC(0xbdeee981), WTC(0xbe08e181), WTC(0xbe236001), WTC(0xbe3e7201), WTC(0xbe5a2301), WTC(0xbe767e81), WTC(0xbe938c81), WTC(0xbeb15701), WTC(0xbecfe601), WTC(0xbeef4601), WTC(0xbf0f8301), WTC(0xbf30a901), WTC(0xbf52c101), WTC(0xbf75cc81), WTC(0xbf99cb01), WTC(0xbfbebb81), WTC(0xbfe48981), WTC(0xc00b04c0), WTC(0xc031f880), WTC(0xc0593340), WTC(0xc0809280), WTC(0xc0a802c0), WTC(0xc0cf6ec0), WTC(0xc0f6cc00), WTC(0xc11e3a80), WTC(0xc145f040), WTC(0xc16e22c0), WTC(0xc196fb00), WTC(0xc1c08680), WTC(0xc1eaca00), WTC(0xc215cbc0), WTC(0xc2418940), WTC(0xc26df5c0), WTC(0xc29b02c0), WTC(0xc2c8a140), WTC(0xc2f6b500), WTC(0xc3251740), WTC(0xc353a0c0), WTC(0xc3822c00), WTC(0xc3b09940), WTC(0xc3deccc0), WTC(0xc40ca800), WTC(0xc43a28c0), WTC(0xc4678a00), WTC(0xc4951780), WTC(0xc4c31d00), WTC(0xc4f1bdc0), WTC(0xc520e840), WTC(0xc5508440), WTC(0xc5807900), WTC(0xc5b09e80), WTC(0xc5e0bfc0), WTC(0xc610a740), WTC(0xc64029c0), WTC(0xc66f49c0), WTC(0xc69e2180), WTC(0xc6ccca40), WTC(0xc6fb5700), WTC(0xc729cc80), WTC(0xc7582b40), WTC(0xc7867480), WTC(0xc7b4a480), WTC(0xc7e2afc0), WTC(0xc8108a80), WTC(0xc83e28c0), WTC(0xc86b7f00), WTC(0xc8988100), WTC(0xc8c52340), WTC(0xc8f15980), WTC(0xc91d1840), WTC(0xb4d6a381), WTC(0xb4422b81), WTC(0xb3ae8601), WTC(0xb31bb301), WTC(0xb289b181), WTC(0xb1f88181), WTC(0xb1682281), WTC(0xb0d89401), WTC(0xb049d601), WTC(0xafbbe801), WTC(0xaf2ec901), WTC(0xaea27681), WTC(0xae16f001), WTC(0xad8c3301), WTC(0xad023f01), WTC(0xac791401), WTC(0xabf0b181), WTC(0xab691681), WTC(0xaae24301), WTC(0xaa5c3601), WTC(0xa9d6ef01), WTC(0xa9526d81), WTC(0xa8ceb201), WTC(0xa84bbb81), WTC(0xa7c98b01), WTC(0xa7482101), WTC(0xa6c77e01), WTC(0xa647a301), WTC(0xa5c89001), WTC(0xa54a4701), WTC(0xa4ccc901), WTC(0xa4501601), WTC(0xa3d43001), WTC(0xa3591801), WTC(0xa2dece81), WTC(0xa2655581), WTC(0xa1ecae01), WTC(0xa174da81), WTC(0xa0fddd81), WTC(0xa087b981), WTC(0xa0127081), WTC(0x9f9e0301), WTC(0x9f2a7281), WTC(0x9eb7c101), WTC(0x9e45f081), WTC(0x9dd50481), WTC(0x9d650081), WTC(0x9cf5e701), WTC(0x9c87ba81), WTC(0x9c1a7c81), WTC(0x9bae2f81), WTC(0x9b42d581), WTC(0x9ad87081), WTC(0x9a6f0381), WTC(0x9a069001), WTC(0x999f1981), WTC(0x9938a281), WTC(0x98d32d81), WTC(0x986ebd81), WTC(0x980b5501), WTC(0x97a8f681), WTC(0x9747a481), WTC(0x96e76101), WTC(0x96882e01), WTC(0x962a0c81), WTC(0x95ccff01), WTC(0x95710601), WTC(0x95162381), WTC(0x94bc5981), WTC(0x9463a881), WTC(0x940c1281), WTC(0x93b59901), WTC(0x93603d01), WTC(0x930bff81), WTC(0x92b8e101), WTC(0x9266e281), WTC(0x92160301), WTC(0x91c64301), WTC(0x9177a301), WTC(0x912a2201), WTC(0x90ddc001), WTC(0x90927b81), WTC(0x90485401), WTC(0x8fff4601), WTC(0x8fb74f81), WTC(0x8f706f01), WTC(0x8f2aa101), WTC(0x8ee5e301), WTC(0x8ea23201), WTC(0x8e5f8881), WTC(0x8e1de001), WTC(0x8ddd3201), WTC(0x8d9d7781), WTC(0x8d5eaa01), WTC(0x8d20c301), WTC(0x8ce3ba81), WTC(0x8ca78781), WTC(0x8c6c1b01), WTC(0x8c316681), WTC(0x8bf75b01), WTC(0x8bbde981), WTC(0x8b850281), WTC(0x8b4c9701), WTC(0x8b149701), WTC(0x8adcee01), WTC(0x8aa58681), WTC(0x8a6e4a01), WTC(0x8a372881), WTC(0x8a001f01), WTC(0x89c92f81), WTC(0x89925a81), WTC(0x895bcd01), WTC(0x8925f101), WTC(0x88f13801), WTC(0x88be1681), WTC(0x888d3181), WTC(0x885f8481), WTC(0x88353501), WTC(0x88124281), WTC(0x87e73d81), WTC(0x87d4ac81), WTC(0x87cb5101), WTC(0x87c05e81), WTC(0x87b42481), WTC(0x87a70e81), WTC(0x87998f01), WTC(0x878c1881), WTC(0x877ede01), WTC(0x8771c601), WTC(0x8764b101), WTC(0x87578181), WTC(0x874a2f01), WTC(0x873cc201), WTC(0x872f4201), WTC(0x8721b481), WTC(0x87141b01), WTC(0x87067281), WTC(0x86f8ba81), WTC(0x86eaf081), WTC(0x86dd1481), WTC(0x86cf2601), WTC(0x86c12401), WTC(0x86b30f01), WTC(0x86a4e781), WTC(0x8696ad01), WTC(0x86886001), WTC(0x867a0081), WTC(0x866b8d81), WTC(0x865d0581), WTC(0x864e6901), WTC(0x863fb701), WTC(0x8630f181), WTC(0x86221801), WTC(0x86132c01), WTC(0x86042c01), WTC(0x85f51681), WTC(0x85e5eb81), WTC(0x85d6a981), WTC(0x85c75201), WTC(0x85b7e601), WTC(0x85a86581), WTC(0x8598d081), WTC(0x85892681), WTC(0x85796601), WTC(0x85698e81), WTC(0x8559a081), WTC(0x85499d01), WTC(0x85398481), WTC(0x85295881), WTC(0x85191801), WTC(0x8508c181), WTC(0x84f85581), WTC(0x84e7d381), WTC(0x84d73c01), WTC(0x84c69101), WTC(0x84b5d301), WTC(0x84a50201), WTC(0x84941d81), WTC(0x84832481), WTC(0x84721701), WTC(0x8460f581), WTC(0x844fc081), WTC(0x843e7a81), WTC(0x842d2281), WTC(0x841bb981), WTC(0x840a3e81), WTC(0x83f8b001), WTC(0x83e70f01), WTC(0x83d55d01), WTC(0x83c39a81), WTC(0x83b1c881), WTC(0x839fe801), WTC(0x838df801), WTC(0x837bf801), WTC(0x8369e781), WTC(0x8357c701), WTC(0x83459881), WTC(0x83335c81), WTC(0x83211501), WTC(0x830ec081), WTC(0x82fc5f01), WTC(0x82e9ef01), WTC(0x82d77201), WTC(0x82c4e801), WTC(0x82b25301), WTC(0x829fb401), WTC(0x828d0b01), WTC(0x827a5801), WTC(0x82679901), WTC(0x8254cf01), WTC(0x8241fa01), WTC(0x822f1b01), WTC(0x821c3401), WTC(0x82094581), WTC(0x81f64f01), WTC(0x81e34f81), WTC(0x81d04681), WTC(0x81bd3401), WTC(0x81aa1981), WTC(0x8196f781), WTC(0x8183cf81), WTC(0x8170a181), WTC(0x815d6c01), WTC(0x814a2f81), WTC(0x8136ea01), WTC(0x81239d81), WTC(0x81104a01), WTC(0x80fcf181), WTC(0x80e99401), WTC(0x80d63101), WTC(0x80c2c781), WTC(0x80af5701), WTC(0x809bdf01), WTC(0x80886081), WTC(0x8074dc01), WTC(0x80615281), WTC(0x804dc481), WTC(0x803a3381), WTC(0x80269f81), WTC(0x80130981), WTC(0x0a608220), WTC(0x0a8ee7d0), WTC(0x0abe35c0), WTC(0x0aee5de0), WTC(0x0b1f5230), WTC(0x0b5104a0), WTC(0x0b836720), WTC(0x0bb66bb0), WTC(0x0bea0440), WTC(0x0c1e22c0), WTC(0x0c52ba70), WTC(0x0c87ca90), WTC(0x0cbd5ba0), WTC(0x0cf375e0), WTC(0x0d2a1f50), WTC(0x0d615480), WTC(0x0d990e40), WTC(0x0dd14500), WTC(0x0e09f730), WTC(0x0e432e90), WTC(0x0e7cf790), WTC(0x0eb75e50), WTC(0x0ef26430), WTC(0x0f2dfd70), WTC(0x0f6a1d70), WTC(0x0fa6b7e0), WTC(0x0fe3c3d0), WTC(0x10213ac0), WTC(0x105f1640), WTC(0x109d4f20), WTC(0x10dbdb80), WTC(0x111ab0c0), WTC(0x1159c360), WTC(0x11990fc0), WTC(0x11d8a060), WTC(0x121882c0), WTC(0x1258c480), WTC(0x12995a40), WTC(0x12da1b00), WTC(0x131adb60), WTC(0x135b70c0), WTC(0x139bb680), WTC(0x13db8c00), WTC(0x141ad080), WTC(0x14596460), WTC(0x149729e0), WTC(0x14d404e0), WTC(0x150fd8e0), WTC(0x154a88c0), WTC(0x1583f5e0), WTC(0x15bc0120), WTC(0x15f28ba0), WTC(0x162779a0), WTC(0x165ab300), WTC(0x168c2040), WTC(0x16bbaa80), WTC(0x16e94120), WTC(0x1714d9e0), WTC(0x173e6440), WTC(0x17660680), WTC(0x178ca020), WTC(0x17b36400), WTC(0x17db84e0), WTC(0x1805d920), WTC(0x18328400), WTC(0x18617cc0), WTC(0x1892bfa0), WTC(0x18c64540), WTC(0x18fc0400), WTC(0x1933f140), WTC(0x196e0320), WTC(0x19aa2fc0), WTC(0x19e86d80), WTC(0x1a28b2e0), WTC(0x1a6af700), WTC(0x1aaf3320), WTC(0x1af56180), WTC(0x1b3d7ce0), WTC(0x1b877c40), WTC(0x1bd350c0), WTC(0x1c20ea40), WTC(0x1c703840), WTC(0x1cc13860), WTC(0x1d13f760), WTC(0x1d688420), WTC(0x1dbeed40), WTC(0x1e174660), WTC(0x1e71a640), WTC(0x1ece2400), WTC(0x1f2cd220), WTC(0x1f8db3c0), WTC(0x1ff0c3e0), WTC(0x20560080), WTC(0x20bd46c0), WTC(0x21263400), WTC(0x21905740), WTC(0x21fb4100), WTC(0x2266ba80), WTC(0x22d2d140), WTC(0x233f9780), WTC(0x23ad25c0), WTC(0x241bc800), WTC(0x248bf040), WTC(0x24fe1380), WTC(0x25728180), WTC(0x25e90a00), WTC(0x26614080), WTC(0x26dabdc0), WTC(0x27552540), WTC(0x27d03200), WTC(0x284ba580), WTC(0x28c740c0), WTC(0x29431f80), WTC(0x29bfc9c0), WTC(0x2a3dd080), WTC(0x2abdc000), WTC(0x2b3ffd00), WTC(0x2bc4cd80), WTC(0x2c4c7d40), WTC(0x2cd72ec0), WTC(0x2d647f80), WTC(0x2df3cd80), WTC(0x2e847d80), WTC(0x2f15ea40), WTC(0x2fa760c0), WTC(0x30382b80), WTC(0x30c79440), WTC(0x315566c0), WTC(0x31e20800), WTC(0x326de7c0), WTC(0x32f98200), WTC(0x3385ba00), WTC(0x3413bec0), WTC(0x34a4c480), WTC(0x3539bf00), WTC(0x35d2c4c0), WTC(0x366f8340), WTC(0x370fb800), WTC(0x37b2cf80), WTC(0x3857a480), WTC(0x38fcee80), WTC(0x39a16840), WTC(0x3a4422c0), WTC(0x3ae495c0), WTC(0x3b824000), WTC(0x3c1cb500), WTC(0x3cb438c0), WTC(0x3d4994c0), WTC(0x3ddd8f40), WTC(0x3e70ec00), WTC(0x3f045e40), WTC(0x3f989080), WTC(0x402e32ff), WTC(0x40c5c07f), WTC(0x415f547f), WTC(0x41faf07f), WTC(0x4298997f), WTC(0x4338307f), WTC(0x43d96bff), WTC(0x447bffff), WTC(0x451f9cff), WTC(0x45c3daff), WTC(0x46683eff), WTC(0x470c4cff), WTC(0x47af93ff), WTC(0x4851c3ff), WTC(0x48f29d7f), WTC(0x4991de7f), WTC(0x4a2f5e7f), WTC(0x4acb287f), WTC(0x4b65537f), WTC(0x4bfdf37f), WTC(0x4c95337f), WTC(0x4d2b51ff), WTC(0x4dc091ff), WTC(0x4e5533ff), WTC(0x4ee96b7f), WTC(0x4f7d61ff), WTC(0x501140ff), WTC(0x50a5317f), WTC(0x51395a7f), WTC(0x51cddf7f), WTC(0x5262e6ff), WTC(0x52f885ff), WTC(0x538eb47f), WTC(0x542560ff), WTC(0x54bc7b7f), WTC(0x5553a8ff), WTC(0x55ea35ff), WTC(0x567f66ff), WTC(0x5712897f), WTC(0x57a33a7f), WTC(0x583152ff), WTC(0x58bca5ff), WTC(0x594530ff), WTC(0x59cb79ff), WTC(0x5a5047ff), WTC(0x5ad45eff), WTC(0x5b584e7f), WTC(0x5bdc417f), WTC(0x5c60487f), WTC(0x5ce476ff), WTC(0x5d68c47f), WTC(0x5ded06ff), WTC(0x5e7111ff), WTC(0x5ef4b5ff), WTC(0x5f77a17f), WTC(0x5ff96aff), WTC(0x6079a7ff), WTC(0x60f7f7ff), WTC(0x617417ff), WTC(0x61edd87f), WTC(0x6264ffff), WTC(0x62d9a6ff), WTC(0x634c817f), WTC(0x63be657f), WTC(0x6430277f), WTC(0x64a2247f), WTC(0x65142bff), WTC(0x6586027f), WTC(0x65f7697f), WTC(0x666801ff), WTC(0x66d756ff), WTC(0x6744f0ff), WTC(0x67b0787f), WTC(0x681a077f), WTC(0x6881ebff), WTC(0x68e8707f), WTC(0x694dceff), WTC(0x69b21e7f), WTC(0x6a156cff), WTC(0x6a77ca7f), WTC(0x6ad9377f), WTC(0x6b39a4ff), WTC(0x6b9901ff), WTC(0x6bf73cff), WTC(0x6c54457f), WTC(0x6cb00aff), WTC(0x6d0a7bff), WTC(0x6d6387ff), WTC(0xae2cbe01), WTC(0xaf526d01), WTC(0xb0751201), WTC(0xb194da81), WTC(0xb2b1f401), WTC(0xb3cc8d01), WTC(0xb4e4d201), WTC(0xb5faf101), WTC(0xb70f1881), WTC(0xb8217301), WTC(0xb9321181), WTC(0xba40ee01), WTC(0xbb4e0201), WTC(0xbc594781), WTC(0xbd62b881), WTC(0xbe6a5181), WTC(0xbf700d01), WTC(0xc073e4c0), WTC(0xc175d240), WTC(0xc275cc80), WTC(0xc373cb80), WTC(0xc46fca00), WTC(0xc569c600), WTC(0xc661bdc0), WTC(0xc757af80), WTC(0xc84b9840), WTC(0xc93d7300), WTC(0xca2d3a40), WTC(0xcb1aea40), WTC(0xcc068280), WTC(0xccf00480), WTC(0xcdd77200), WTC(0xcebccb40), WTC(0xcfa00d80), WTC(0xd0813540), WTC(0xd1603f00), WTC(0xd23d2980), WTC(0xd317f7c0), WTC(0xd3f0ac40), WTC(0xd4c74980), WTC(0xd59bcf80), WTC(0xd66e3b00), WTC(0xd73e8900), WTC(0xd80cb740), WTC(0xd8d8c7c0), WTC(0xd9a2be00), WTC(0xda6a9e40), WTC(0xdb306a40), WTC(0xdbf42080), WTC(0xdcb5be80), WTC(0xdd754140), WTC(0xde32a900), WTC(0xdeedf9c0), WTC(0xdfa737c0), WTC(0xe05e6740), WTC(0xe1138900), WTC(0xe1c69ac0), WTC(0xe2779a40), WTC(0xe3268680), WTC(0xe3d36260), WTC(0xe47e33a0), WTC(0xe526ff80), WTC(0xe5cdc960), WTC(0xe6729100), WTC(0xe7155460), WTC(0xe7b611c0), WTC(0xe854ca20), WTC(0xe8f18180), WTC(0xe98c3ca0), WTC(0xea24ffe0), WTC(0xeabbcb20), WTC(0xeb509b60), WTC(0xebe36d00), WTC(0xec743e00), WTC(0xed0310e0), WTC(0xed8feaa0), WTC(0xee1ad060), WTC(0xeea3c640), WTC(0xef2acd60), WTC(0xefafe6a0), WTC(0xf03312f0), WTC(0xf0b45800), WTC(0xf133c230), WTC(0xf1b15ef0), WTC(0xf22d3af0), WTC(0xf2a75c80), WTC(0xf31fc460), WTC(0xf39673b0), WTC(0xf40b6a00), WTC(0xf47ea230), WTC(0xf4f01450), WTC(0xf55fb930), WTC(0xf5cd84c0), WTC(0xf6396090), WTC(0xf6a333e0), WTC(0xf70ae540), WTC(0xf7707260), WTC(0xf7d3f720), WTC(0xf83592f0), WTC(0xf8956450), WTC(0xf8f38120), WTC(0xf94ff7c8), WTC(0xf9aad740), WTC(0xfa042920), WTC(0xfa5be110), WTC(0xfab1e778), WTC(0xfb062478), WTC(0xfb588d78), WTC(0xfba93530), WTC(0xfbf836c8), WTC(0xfc45ace0), WTC(0xfc91a294), WTC(0xfcdc0e5c), WTC(0xfd24e438), WTC(0xfd6c17dc), WTC(0xfdb19758), WTC(0xfdf54c3c), WTC(0xfe371ef8), WTC(0xfe7701aa), WTC(0xfeb50d62), WTC(0xfef1700a), WTC(0xff2c5574), WTC(0xff65ee7b), WTC(0xff9e75de), WTC(0xffd62863), WTC(0x000d4401), WTC(0x0043d345), WTC(0x00799cd0), WTC(0x00ae5f49), WTC(0x00e1d7a4), WTC(0x0113a6f2), WTC(0x0143575c), WTC(0x01707024), WTC(0x019a9346), WTC(0x01c1cf08), WTC(0x01e66c12), WTC(0x0208ac48), WTC(0x0228e868), WTC(0x0247a6c8), WTC(0x02657aa0), WTC(0x0282f710), WTC(0x02a07e50), WTC(0x02be31c0), WTC(0x02dc2b30), WTC(0x02fa7f34), WTC(0x0318fb10), WTC(0x03372fdc), WTC(0x0354ae54), WTC(0x03710d18), WTC(0x038bfdb4), WTC(0x03a54084), WTC(0x03bc92b8), WTC(0x03d1c710), WTC(0x03e4dd20), WTC(0x03f5e25c), WTC(0x0404e218), WTC(0x0411fc30), WTC(0x041d6b30), WTC(0x04276cd0), WTC(0x04303e00), WTC(0x04381528), WTC(0x043f2310), WTC(0x04459908), WTC(0x044ba430), WTC(0x045161f8), WTC(0x0456e6f8), WTC(0x045c49a8), WTC(0x046192f8), WTC(0x0466af40), WTC(0x046b8240), WTC(0x046ff0d8), WTC(0x0473de18), WTC(0x04772b58), WTC(0x0479b9a0), WTC(0x047b6a30), WTC(0x047c2088), WTC(0x047bc230), WTC(0x047a3418), WTC(0x04776098), WTC(0x04734790), WTC(0x046df4c0), WTC(0x04677220), WTC(0x045fd1b0), WTC(0x04573588), WTC(0x044dc4b8), WTC(0x0443a5b8), WTC(0x04390160), WTC(0x042e0398), WTC(0x0422d8c0), WTC(0x0417aa30), WTC(0x040c7ce0), WTC(0x040136e0), WTC(0x03f5beb0), WTC(0x03e9f8ec), WTC(0x03ddc484), WTC(0x03d0fd9c), WTC(0x03c37fa0), WTC(0x03b53014), WTC(0x03a60a18), WTC(0x03960f88), WTC(0x03854110), WTC(0x0373ad9c), WTC(0x03617694), WTC(0x034ebf9c), WTC(0x033bab30), WTC(0x03284ef0), WTC(0x0314b598), WTC(0x0300ea54), WTC(0x02ecf524), WTC(0x02d8d210), WTC(0x02c476ac), WTC(0x02afd940), WTC(0x029aee4c), WTC(0x0285a6f4), WTC(0x026ff398), WTC(0x0259c448), WTC(0x024317cc), WTC(0x022c0084), WTC(0x02149310), WTC(0x01fce334), WTC(0x01e4fb24), WTC(0x01ccdd0a), WTC(0x01b48b20), WTC(0x019c077e), WTC(0x01835432), WTC(0x016a733c), WTC(0x015166a6), WTC(0x0138302e), WTC(0x011ed0f6), WTC(0x010549f8), WTC(0x00eb9c25), WTC(0x00d1caa6), WTC(0x00b7db94), WTC(0x009dd560), WTC(0x0083be75), WTC(0x00699d41), WTC(0x004f782f), WTC(0x003555ab), WTC(0x001b3c21), WTC(0x000131fe), WTC(0xd61cfc40), WTC(0xd5acb340), WTC(0xd53d4400), WTC(0xd4cea6c0), WTC(0xd460d440), WTC(0xd3f3c440), WTC(0xd3876f80), WTC(0xd31bce40), WTC(0xd2b0d900), WTC(0xd2468980), WTC(0xd1dcef00), WTC(0xd17429c0), WTC(0xd10c5b80), WTC(0xd0a59b80), WTC(0xd03fd780), WTC(0xcfdae780), WTC(0xcf76a380), WTC(0xcf12fac0), WTC(0xceb01100), WTC(0xce4e18c0), WTC(0xcded4440), WTC(0xcd8d9a40), WTC(0xcd2ee800), WTC(0xccd0f440), WTC(0xcc738780), WTC(0xcc167d40), WTC(0xcbb9c180), WTC(0xcb5d4040), WTC(0xcb00e240), WTC(0xcaa48000), WTC(0xca47eac0), WTC(0xc9eaf1c0), WTC(0xc98d8100), WTC(0xc92fc580), WTC(0xc8d1fc80), WTC(0xc8746480), WTC(0xc816dc40), WTC(0xc7b8c280), WTC(0xc7596800), WTC(0xc6f81f80), WTC(0xc6945740), WTC(0xc62d93c0), WTC(0xc5c358c0), WTC(0xc5552b80), WTC(0xc4e29240), WTC(0xc46b1440), WTC(0xc3ee3840), WTC(0xc36b8500), WTC(0xc2e28040), WTC(0xc252ae80), WTC(0xc1bb9540), WTC(0xc11cc200), WTC(0xc075cf00), WTC(0xbfc65781), WTC(0xbf0df881), WTC(0xbe4c6f01), WTC(0xbd819401), WTC(0xbcad2d01), WTC(0xbbcfb981), WTC(0xbaeca681), WTC(0xba08e781), WTC(0xb9297081), WTC(0xb851e081), WTC(0xb782ed01), WTC(0xb6bc6a81), WTC(0xb5fe4981), WTC(0xb5487281), WTC(0xb49ad081), WTC(0xb3f54d81), WTC(0xb357d401), WTC(0xb2c24e01), WTC(0xb234a681), WTC(0xb1aec701), WTC(0xb1309b01), WTC(0xb0ba0c01), WTC(0xb04b0481), WTC(0xafe36f01), WTC(0xaf833601), WTC(0xaf2a4381), WTC(0xaed88201), WTC(0xae8ddb81), WTC(0xae4a3b81), WTC(0xae0d8b01), WTC(0xadd7b581), WTC(0xada8a481), WTC(0xad804281), WTC(0xad5e7a81), WTC(0xad433601), WTC(0xad2e6001), WTC(0xad1fe281), WTC(0xad17a801), WTC(0xad159a81), WTC(0xad19a501), WTC(0xad23b101), WTC(0xad33aa01), WTC(0xad497981), WTC(0xad650a01), WTC(0xad864601), WTC(0xadad1781), WTC(0xadd96981), WTC(0xae0b2601), WTC(0xae423781), WTC(0xae7e8801), WTC(0xaec00201), WTC(0xaf069081), WTC(0xaf521c81), WTC(0xafa29201), WTC(0xaff7da01), WTC(0xb051df01), WTC(0xb0b08c81), WTC(0xb113cb81), WTC(0xb17b8701), WTC(0xb1e7a981), WTC(0xb2581d81), WTC(0xb2cccc81), WTC(0xb345a181), WTC(0xb3c28701), WTC(0xb4436681), WTC(0xb4c82b81), WTC(0xb550bf81), WTC(0xb5dd0d01), WTC(0xb66cff01), WTC(0xb7007f01), WTC(0xb7977781), WTC(0xb831d381), WTC(0xb8cf7d01), WTC(0xb9705e01), WTC(0xba146101), WTC(0xbabb7081), WTC(0xbb657781), WTC(0xbc125f01), WTC(0xbcc21281), WTC(0xbd747b81), WTC(0xbe298581), WTC(0xbee11981), WTC(0xbf9b2301), WTC(0xc0578b80), WTC(0xc1163dc0), WTC(0xc1d72400), WTC(0xc29a28c0), WTC(0xc35f3640), WTC(0xc42636c0), WTC(0xc4ef1500), WTC(0xc5b9bb00), WTC(0xc6861340), WTC(0xc7540840), WTC(0xc8238400), WTC(0xc8f47100), WTC(0xc9c6b9c0), WTC(0xca9a4840), WTC(0xcb6f0780), WTC(0xcc44e140), WTC(0xcd1bc000), WTC(0xcdf38e00), WTC(0xcecc3600), WTC(0xcfa5a240), WTC(0xd07fbcc0), WTC(0xd15a7040), WTC(0xd235a6c0), WTC(0xd3114b00), WTC(0xd3ed4740), WTC(0xd4c98580), WTC(0xd5a5f080), WTC(0xd6827280), WTC(0xd75ef600), WTC(0xd83b6500), WTC(0xd917aa00), WTC(0xd9f3af80), WTC(0xdacf5fc0), WTC(0xdbaaa540), WTC(0xdc856a00), WTC(0xdd5f98c0), WTC(0xde391bc0), WTC(0xdf11dd40), WTC(0xdfe9c780), WTC(0xe0c0c540), WTC(0xe196c080), WTC(0xe26ba3c0), WTC(0xe33f5960), WTC(0xe411cba0), WTC(0xe4e2e500), WTC(0xe5b28fc0), WTC(0xe680b640), WTC(0xe74d42e0), WTC(0xe8181fe0), WTC(0xe8e137e0), WTC(0xe9a87500), WTC(0xea6dc1a0), WTC(0xeb310820), WTC(0xebf23300), WTC(0xecb12c60), WTC(0xed6ddee0), WTC(0xee2834a0), WTC(0xeee01800), WTC(0xef957380), WTC(0xf0483160), WTC(0xf0f83c00), WTC(0xf1a57db0), WTC(0xf24fe0f0), WTC(0xf2f74ff0), WTC(0xf39bb530), WTC(0xf43cfaf0), WTC(0xf4db0b90), WTC(0xf575d180), WTC(0xf60d3700), WTC(0xf6a12680), WTC(0xf7318a50), WTC(0xf7be4cc0), WTC(0xf8475850), WTC(0xf8cc9738), WTC(0xf94df3e0), WTC(0xf9cb58a8), WTC(0xfa44afe0), WTC(0xfab9e3e8), WTC(0xfb2adf20), WTC(0xfb978be8), WTC(0xfbffd488), WTC(0xfc63a370), WTC(0xfcc2e2f0), WTC(0xfd1d7d64), WTC(0xfd735d2c), WTC(0xfdc46c9c), WTC(0xfe109618), WTC(0xfe57c3f4), WTC(0xfe99e090), WTC(0xfed6d644), WTC(0xff0e8f6e), WTC(0xff40f667), WTC(0xff6df58c), WTC(0xff957738), WTC(0xffb765c5), WTC(0xffd3ab90), WTC(0xffea32f4), WTC(0xfffae64c), WTC(0x0005aff3), WTC(0x000a7a44), WTC(0x00092f9c), WTC(0x0001ba54), WTC(0xfff404ca), WTC(0xffdff957)}; /* * TNS_MAX_BANDS * entry for each sampling rate * 1 long window * 2 SHORT window */ const UCHAR tns_max_bands_tbl[13][2] = { {31, 9}, /* 96000 */ {31, 9}, /* 88200 */ {34, 10}, /* 64000 */ {40, 14}, /* 48000 */ {42, 14}, /* 44100 */ {51, 14}, /* 32000 */ {46, 14}, /* 24000 */ {46, 14}, /* 22050 */ {42, 14}, /* 16000 */ {42, 14}, /* 12000 */ {42, 14}, /* 11025 */ {39, 14}, /* 8000 */ {39, 14}, /* 7350 */ }; /* TNS_MAX_BANDS for low delay. The array index is the sampleRateIndex */ const UCHAR tns_max_bands_tbl_480[13] = { 31, /* 96000 */ 31, /* 88200 */ 31, /* 64000 */ 31, /* 48000 */ 32, /* 44100 */ 37, /* 32000 */ 30, /* 24000 */ 30, /* 22050 */ 30, /* 16000 */ 30, /* 12000 */ 30, /* 11025 */ 30, /* 8000 */ 30 /* 7350 */ }; const UCHAR tns_max_bands_tbl_512[13] = { 31, /* 96000 */ 31, /* 88200 */ 31, /* 64000 */ 31, /* 48000 */ 32, /* 44100 */ 37, /* 32000 */ 31, /* 24000 */ 31, /* 22050 */ 31, /* 16000 */ 31, /* 12000 */ 31, /* 11025 */ 31, /* 8000 */ 31 /* 7350 */ }; #define TCC(x) (FIXP_DBL(x)) const FIXP_TCC FDKaacDec_tnsCoeff3[8] = { TCC(0x81f1d1d4), TCC(0x9126146c), TCC(0xadb922c4), TCC(0xd438af1f), TCC(0x00000000), TCC(0x3789809b), TCC(0x64130dd4), TCC(0x7cca7016)}; const FIXP_TCC FDKaacDec_tnsCoeff4[16] = { TCC(0x808bc842), TCC(0x84e2e58c), TCC(0x8d6b49d1), TCC(0x99da920a), TCC(0xa9c45713), TCC(0xbc9ddeb9), TCC(0xd1c2d51b), TCC(0xe87ae53d), TCC(0x00000000), TCC(0x1a9cd9b6), TCC(0x340ff254), TCC(0x4b3c8c29), TCC(0x5f1f5ebb), TCC(0x6ed9ebba), TCC(0x79bc385f), TCC(0x7f4c7e5b)}; const UCHAR FDKaacDec_tnsCoeff3_gain_ld[] = { 3, 1, 1, 1, 0, 1, 1, 3, }; const UCHAR FDKaacDec_tnsCoeff4_gain_ld[] = { 4, 2, 2, 1, 1, 1, 1, 1, 0, 1, 1, 1, 1, 2, 2, 4, }; /* Lookup tables for elements in ER bitstream */ const MP4_ELEMENT_ID elementsTab[AACDEC_MAX_CH_CONF][AACDEC_CH_ELEMENTS_TAB_SIZE] = { /* 1 */ {ID_SCE, ID_EXT, ID_END, ID_NONE, ID_NONE, ID_NONE, ID_NONE}, /* 1 channel */ /* 2 */ {ID_CPE, ID_EXT, ID_END, ID_NONE, ID_NONE, ID_NONE, ID_NONE} /* 2 channels */ #if (AACDEC_MAX_CH_CONF > 2) /* 3 */, {ID_SCE, ID_CPE, ID_EXT, ID_END, ID_NONE, ID_NONE, ID_NONE}, /* 3 channels */ /* 4 */ {ID_SCE, ID_CPE, ID_SCE, ID_EXT, ID_END, ID_NONE, ID_NONE}, /* 4 channels */ /* 5 */ {ID_SCE, ID_CPE, ID_CPE, ID_EXT, ID_END, ID_NONE, ID_NONE}, /* 5 channels */ /* 6 */ {ID_SCE, ID_CPE, ID_CPE, ID_LFE, ID_EXT, ID_END, ID_NONE} /* 6 channels */ #endif #if (AACDEC_MAX_CH_CONF > 6) /* 7 */, {ID_SCE, ID_CPE, ID_CPE, ID_CPE, ID_LFE, ID_EXT, ID_END}, /* 8 channels */ /* 8 */ {ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE}, /* reserved */ /* 9 */ {ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE}, /* reserved */ /* 10 */ {ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE}, /* reserved */ /* 11 */ {ID_SCE, ID_CPE, ID_CPE, ID_SCE, ID_LFE, ID_EXT, ID_END}, /* 7 channels */ /* 12 */ {ID_SCE, ID_CPE, ID_CPE, ID_CPE, ID_LFE, ID_EXT, ID_END}, /* 8 channels */ /* 13 */ {ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE}, /* see elementsChCfg13 */ /* 14 */ {ID_SCE, ID_CPE, ID_CPE, ID_LFE, ID_CPE, ID_EXT, ID_END} /* 8 channels */ #endif }; /*! Random sign bit used for concealment */ const USHORT AacDec_randomSign[AAC_NF_NO_RANDOM_VAL / 16] = { /* sign bits of FDK_sbrDecoder_sbr_randomPhase[] entries: LSB ........... MSB -> MSB ... LSB */ /* 1001 0111 0011 1100 -> */ 0x3ce9, /* 0100 0111 0111 1011 -> */ 0xdee2, /* 0001 1100 1110 1011 -> */ 0xd738, /* 0001 0011 0110 1001 -> */ 0x96c8, /* 0101 0011 1101 0000 -> */ 0x0bca, /* 0001 0001 1111 0100 -> */ 0x2f88, /* 1110 1100 1110 1101 -> */ 0xb737, /* 0010 1010 1011 1001 -> */ 0x9d54, /* 0111 1100 0110 1010 -> */ 0x563e, /* 1101 0111 0010 0101 -> */ 0xa4eb, /* 0001 0101 1011 1100 -> */ 0x3da8, /* 0101 0111 1001 1011 -> */ 0xd9ea, /* 1101 0100 0101 0101 -> */ 0xaa2b, /* 1000 1001 0100 0011 -> */ 0xc291, /* 1100 1111 1010 1100 -> */ 0x35f3, /* 1100 1010 1110 0010 -> */ 0x4753, /* 0110 0001 1010 1000 -> */ 0x1586, /* 0011 0101 1111 1100 -> */ 0x3fac, /* 0001 0110 1010 0001 -> */ 0x8568, /* 0010 1101 0111 0010 -> */ 0x4eb4, /* 1101 1010 0100 1001 -> */ 0x925b, /* 1100 1001 0000 1110 -> */ 0x7093, /* 1000 1100 0110 1010 -> */ 0x5631, /* 0000 1000 0110 1101 -> */ 0xb610, /* 1000 0001 1111 1011 -> */ 0xdf81, /* 1111 0011 0100 0111 -> */ 0xe2cf, /* 1000 0001 0010 1010 -> */ 0x5481, /* 1101 0101 1100 1111 -> */ 0xf3ab, /* 0110 0001 0110 1000 -> */ 0x1686, /* 0011 0011 1100 0110 -> */ 0x63cc, /* 0011 0111 0101 0110 -> */ 0x6aec, /* 1011 0001 1010 0010 -> */ 0x458d}; /* MDST filter coefficients for current window * max: 0.635722 => 20 bits (unsigned) necessary for representation * min: = -max */ const FIXP_FILT mdst_filt_coef_curr[20][3] = { {FILT(0.000000f), FILT(0.000000f), FILT(0.500000f)}, /*, FILT( 0.000000f), FILT(-0.500000f), FILT( 0.000000f), FILT( 0.000000f) }, */ /* only long / eight short l:sine r:sine */ {FILT(0.091497f), FILT(0.000000f), FILT(0.581427f)}, /*, FILT( 0.000000f), FILT(-0.581427f), FILT( 0.000000f), FILT(-0.091497f) }, */ /* l:kbd r:kbd */ {FILT(0.045748f), FILT(0.057238f), FILT(0.540714f)}, /*, FILT( 0.000000f), FILT(-0.540714f), FILT(-0.057238f), FILT(-0.045748f) }, */ /* l:sine r:kbd */ {FILT(0.045748f), FILT(-0.057238f), FILT(0.540714f)}, /*, FILT( 0.000000f), FILT(-0.540714f), FILT( 0.057238f), FILT(-0.045748f) }, */ /* l:kbd r:sine */ {FILT(0.102658f), FILT(0.103791f), FILT(0.567149f)}, /*, FILT( 0.000000f), FILT(-0.567149f), FILT(-0.103791f), FILT(-0.102658f) }, */ /* long start */ {FILT(0.150512f), FILT(0.047969f), FILT(0.608574f)}, /*, FILT( 0.000000f), FILT(-0.608574f), FILT(-0.047969f), FILT(-0.150512f) }, */ {FILT(0.104763f), FILT(0.105207f), FILT(0.567861f)}, /*, FILT( 0.000000f), FILT(-0.567861f), FILT(-0.105207f), FILT(-0.104763f) }, */ {FILT(0.148406f), FILT(0.046553f), FILT(0.607863f)}, /*, FILT( 0.000000f), FILT(-0.607863f), FILT(-0.046553f), FILT(-0.148406f) }, */ {FILT(0.102658f), FILT(-0.103791f), FILT(0.567149f)}, /*, FILT( 0.000000f), FILT(-0.567149f), FILT( 0.103791f), FILT(-0.102658f) }, */ /* long stop */ {FILT(0.150512f), FILT(-0.047969f), FILT(0.608574f)}, /*, FILT( 0.000000f), FILT(-0.608574f), FILT( 0.047969f), FILT(-0.150512f) }, */ {FILT(0.148406f), FILT(-0.046553f), FILT(0.607863f)}, /*, FILT( 0.000000f), FILT(-0.607863f), FILT( 0.046553f), FILT(-0.148406f) }, */ {FILT(0.104763f), FILT(-0.105207f), FILT(0.567861f)}, /*, FILT( 0.000000f), FILT(-0.567861f), FILT( 0.105207f), FILT(-0.104763f) }, */ {FILT(0.205316f), FILT(0.000000f), FILT(0.634298f)}, /*, FILT( 0.000000f), FILT(-0.634298f), FILT( 0.000000f), FILT(-0.205316f) }, */ /* stop start */ {FILT(0.209526f), FILT(0.000000f), FILT(0.635722f)}, /*, FILT( 0.000000f), FILT(-0.635722f), FILT( 0.000000f), FILT(-0.209526f) }, */ {FILT(0.207421f), FILT(0.001416f), FILT(0.635010f)}, /*, FILT( 0.000000f), FILT(-0.635010f), FILT(-0.001416f), FILT(-0.207421f) }, */ {FILT(0.207421f), FILT(-0.001416f), FILT(0.635010f)}, /*, FILT( 0.000000f), FILT(-0.635010f), FILT( 0.001416f), FILT(-0.207421f) } */ {FILT(0.185618f), FILT(0.000000f), FILT(0.627371f)}, /*, FILT( 0.000000f), FILT(-0.634298f), FILT( 0.000000f), FILT(-0.205316f) }, */ /* stop start Transform Splitting */ {FILT(0.204932f), FILT(0.000000f), FILT(0.634159f)}, /*, FILT( 0.000000f), FILT(-0.635722f), FILT( 0.000000f), FILT(-0.209526f) }, */ {FILT(0.194609f), FILT(0.006202f), FILT(0.630536f)}, /*, FILT( 0.000000f), FILT(-0.635010f), FILT(-0.001416f), FILT(-0.207421f) }, */ {FILT(0.194609f), FILT(-0.006202f), FILT(0.630536f)}, /*, FILT( 0.000000f), FILT(-0.635010f), FILT( 0.001416f), FILT(-0.207421f) } */ }; /* MDST filter coefficients for previous window * max: 0.31831 => 15 bits (unsigned) necessary for representation * min: 0.0 */ const FIXP_FILT mdst_filt_coef_prev[6][4] = { {FILT(0.000000f), FILT(0.106103f), FILT(0.250000f), FILT(0.318310f)}, /*, FILT( 0.250000f), FILT( 0.106103f), FILT( 0.000000f) }, */ /* only long / long start / eight short l:sine */ {FILT(0.059509f), FILT(0.123714f), FILT(0.186579f), FILT(0.213077f)}, /*, FILT( 0.186579f), FILT( 0.123714f), FILT( 0.059509f) }, */ /* l:kbd */ {FILT(0.038498f), FILT(0.039212f), FILT(0.039645f), FILT(0.039790f)}, /*, FILT( 0.039645f), FILT( 0.039212f), FILT( 0.038498f) }, */ /* long stop / stop start l:sine */ {FILT(0.026142f), FILT(0.026413f), FILT(0.026577f), FILT(0.026631f)}, /*, FILT( 0.026577f), FILT( 0.026413f), FILT( 0.026142f) } */ /* l:kbd */ {FILT(0.069608f), FILT(0.075028f), FILT(0.078423f), FILT(0.079580f)}, /*, FILT( 0.039645f), FILT( 0.039212f), FILT( 0.038498f) }, */ /* Transform splitting l:sine */ {FILT(0.042172f), FILT(0.043458f), FILT(0.044248f), FILT(0.044514f)}, /*, FILT( 0.026577f), FILT( 0.026413f), FILT( 0.026142f) } */ /* l:kbd */ }; fdk-aac-2.0.2/libAACdec/src/aac_rom.h000066400000000000000000000216441404225310700170760ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Josef Hoepfl Description: Definition of constant tables *******************************************************************************/ #ifndef AAC_ROM_H #define AAC_ROM_H #include "common_fix.h" #include "FDK_audio.h" #include "aacdec_hcr_types.h" #include "aacdec_hcrs.h" #define PCM_AAC LONG #define PCM_DEC FIXP_DBL #define MAXVAL_PCM_DEC MAXVAL_DBL #define MINVAL_PCM_DEC MINVAL_DBL #define FIXP_DBL2PCM_DEC(x) (x) #define PCM_DEC2FIXP_DBL(x) (x) #define PCM_DEC_BITS DFRACT_BITS #define PCM_DEC2FX_PCM(x) FX_DBL2FX_PCM(x) #define FX_PCM2PCM_DEC(x) FX_PCM2FX_DBL(x) #define AACDEC_MAX_CH_CONF 14 #define AACDEC_CH_ELEMENTS_TAB_SIZE 7 /*!< Size of element tables */ #define AAC_NF_NO_RANDOM_VAL \ 512 /*!< Size of random number array for noise floor */ #define INV_QUANT_TABLESIZE 256 extern const FIXP_DBL InverseQuantTable[INV_QUANT_TABLESIZE + 1]; extern const FIXP_DBL MantissaTable[4][14]; extern const SCHAR ExponentTable[4][14]; #define NUM_LD_COEF_512 1536 #define NUM_LD_COEF_480 1440 /* Window table partition exponents. */ #define WTS0 (1) #define WTS1 (0) #define WTS2 (-2) extern const FIXP_WTB LowDelaySynthesis512[1536]; extern const FIXP_WTB LowDelaySynthesis480[1440]; extern const FIXP_WTB LowDelaySynthesis256[768]; extern const FIXP_WTB LowDelaySynthesis240[720]; extern const FIXP_WTB LowDelaySynthesis160[480]; extern const FIXP_WTB LowDelaySynthesis128[384]; extern const FIXP_WTB LowDelaySynthesis120[360]; typedef struct { const SHORT *sfbOffsetLong; const SHORT *sfbOffsetShort; UCHAR numberOfSfbLong; UCHAR numberOfSfbShort; } SFB_INFO; extern const SFB_INFO sfbOffsetTables[5][16]; /* Huffman tables */ enum { HuffmanBits = 2, HuffmanEntries = (1 << HuffmanBits) }; typedef struct { const USHORT (*CodeBook)[HuffmanEntries]; UCHAR Dimension; UCHAR numBits; UCHAR Offset; } CodeBookDescription; extern const CodeBookDescription AACcodeBookDescriptionTable[13]; extern const CodeBookDescription AACcodeBookDescriptionSCL; extern const STATEFUNC aStateConstant2State[]; extern const SCHAR aCodebook2StartInt[]; extern const UCHAR aMinOfCbPair[]; extern const UCHAR aMaxOfCbPair[]; extern const UCHAR aMaxCwLen[]; extern const UCHAR aDimCb[]; extern const UCHAR aDimCbShift[]; extern const UCHAR aSignCb[]; extern const UCHAR aCbPriority[]; extern const UINT *aHuffTable[]; extern const SCHAR *aQuantTable[]; extern const USHORT aLargestAbsoluteValue[]; extern const UINT aHuffTreeRvlcEscape[]; extern const UINT aHuffTreeRvlCodewds[]; extern const UCHAR tns_max_bands_tbl[13][2]; extern const UCHAR tns_max_bands_tbl_480[13]; extern const UCHAR tns_max_bands_tbl_512[13]; #define FIXP_TCC FIXP_DBL extern const FIXP_TCC FDKaacDec_tnsCoeff3[8]; extern const FIXP_TCC FDKaacDec_tnsCoeff4[16]; extern const UCHAR FDKaacDec_tnsCoeff3_gain_ld[]; extern const UCHAR FDKaacDec_tnsCoeff4_gain_ld[]; extern const USHORT AacDec_randomSign[AAC_NF_NO_RANDOM_VAL / 16]; extern const FIXP_DBL pow2_div24minus1[47]; extern const int offsetTab[2][16]; /* Channel mapping indices for time domain I/O. The first dimension is the channel configuration index. */ extern const UCHAR channelMappingTablePassthrough[15][8]; extern const UCHAR channelMappingTableWAV[15][8]; /* Lookup tables for elements in ER bitstream */ extern const MP4_ELEMENT_ID elementsTab[AACDEC_MAX_CH_CONF] [AACDEC_CH_ELEMENTS_TAB_SIZE]; #define SF_FNA_COEFFS \ 1 /* Compile-time prescaler for MDST-filter coefficients. */ /* SF_FNA_COEFFS > 0 should only be considered for FIXP_DBL-coefficients */ /* (i.e. if CPLX_PRED_FILTER_16BIT is not defined). */ /* With FIXP_DBL loss of precision is possible for SF_FNA_COEFFS > 11. */ #ifdef CPLX_PRED_FILTER_16BIT #define FIXP_FILT FIXP_SGL #define FILT(a) ((FL2FXCONST_SGL(a)) >> SF_FNA_COEFFS) #else #define FIXP_FILT FIXP_DBL #define FILT(a) ((FL2FXCONST_DBL(a)) >> SF_FNA_COEFFS) #endif extern const FIXP_FILT mdst_filt_coef_curr[20][3]; /* MDST-filter coefficient tables used for current window */ extern const FIXP_FILT mdst_filt_coef_prev[6][4]; /* MDST-filter coefficient tables used for previous window */ #endif /* #ifndef AAC_ROM_H */ fdk-aac-2.0.2/libAACdec/src/aacdec_drc.cpp000066400000000000000000001427041404225310700200610ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Christian Griebel Description: Dynamic range control (DRC) decoder tool for AAC *******************************************************************************/ #include "aacdec_drc.h" #include "channelinfo.h" #include "aac_rom.h" #include "sbrdecoder.h" /* * Dynamic Range Control */ /* For parameter conversion */ #define DRC_PARAMETER_BITS (7) #define DRC_MAX_QUANT_STEPS (1 << DRC_PARAMETER_BITS) #define DRC_MAX_QUANT_FACTOR (DRC_MAX_QUANT_STEPS - 1) #define DRC_PARAM_QUANT_STEP \ (FL2FXCONST_DBL(1.0f / (float)DRC_MAX_QUANT_FACTOR)) #define DRC_PARAM_SCALE (1) #define DRC_SCALING_MAX \ ((FIXP_DBL)((INT)(DRC_PARAM_QUANT_STEP >> DRC_PARAM_SCALE) * (INT)127)) #define DRC_BLOCK_LEN (1024) #define DRC_BAND_MULT (4) #define DRC_BLOCK_LEN_DIV_BAND_MULT (DRC_BLOCK_LEN / DRC_BAND_MULT) #define MAX_REFERENCE_LEVEL (127) #define DRC_HEAVY_THRESHOLD_DB (10) #define DVB_ANC_DATA_SYNC_BYTE (0xBC) /* DVB ancillary data sync byte. */ #define OFF 0 #define ON 1 static INT convert_drcParam(FIXP_DBL param_dbl) { /* converts an internal DRC boost/cut scaling factor in FIXP_DBL (which is downscaled by DRC_PARAM_SCALE) back to an integer value between 0 and 127. */ LONG param_long; param_long = (LONG)param_dbl >> 7; param_long = param_long * (INT)DRC_MAX_QUANT_FACTOR; param_long >>= 31 - 7 - DRC_PARAM_SCALE - 1; param_long += 1; /* for rounding */ param_long >>= 1; return (INT)param_long; } /*! \brief Reset DRC information \self Handle of DRC info \return none */ void aacDecoder_drcReset(HANDLE_AAC_DRC self) { self->applyExtGain = 0; self->additionalGainPrev = AACDEC_DRC_GAIN_INIT_VALUE; self->additionalGainFilterState = AACDEC_DRC_GAIN_INIT_VALUE; self->additionalGainFilterState1 = AACDEC_DRC_GAIN_INIT_VALUE; } /*! \brief Initialize DRC information \self Handle of DRC info \return none */ void aacDecoder_drcInit(HANDLE_AAC_DRC self) { CDrcParams *pParams; if (self == NULL) { return; } /* init control fields */ self->enable = OFF; self->numThreads = 0; /* init params */ pParams = &self->params; pParams->bsDelayEnable = 0; pParams->cut = FL2FXCONST_DBL(0.0f); pParams->usrCut = FL2FXCONST_DBL(0.0f); pParams->boost = FL2FXCONST_DBL(0.0f); pParams->usrBoost = FL2FXCONST_DBL(0.0f); pParams->targetRefLevel = 96; pParams->expiryFrame = AACDEC_DRC_DFLT_EXPIRY_FRAMES; pParams->applyHeavyCompression = OFF; pParams->usrApplyHeavyCompression = OFF; pParams->defaultPresentationMode = DISABLED_PARAMETER_HANDLING; pParams->encoderTargetLevel = MAX_REFERENCE_LEVEL; /* worst case assumption */ self->update = 1; self->numOutChannels = 0; self->prevAacNumChannels = 0; /* initial program ref level = target ref level */ self->progRefLevel = pParams->targetRefLevel; self->progRefLevelPresent = 0; self->presMode = -1; self->uniDrcPrecedence = 0; aacDecoder_drcReset(self); } /*! \brief Initialize DRC control data for one channel \self Handle of DRC info \return none */ void aacDecoder_drcInitChannelData(CDrcChannelData *pDrcChData) { if (pDrcChData != NULL) { pDrcChData->expiryCount = 0; pDrcChData->numBands = 1; pDrcChData->bandTop[0] = DRC_BLOCK_LEN_DIV_BAND_MULT - 1; pDrcChData->drcValue[0] = 0; pDrcChData->drcInterpolationScheme = 0; pDrcChData->drcDataType = UNKNOWN_PAYLOAD; } } /*! \brief Set one single DRC parameter \self Handle of DRC info. \param Parameter to be set. \value Value to be set. \return an error code. */ AAC_DECODER_ERROR aacDecoder_drcSetParam(HANDLE_AAC_DRC self, AACDEC_DRC_PARAM param, INT value) { AAC_DECODER_ERROR ErrorStatus = AAC_DEC_OK; switch (param) { case DRC_CUT_SCALE: /* set attenuation scale factor */ if ((value < 0) || (value > DRC_MAX_QUANT_FACTOR)) { return AAC_DEC_SET_PARAM_FAIL; } if (self == NULL) { return AAC_DEC_INVALID_HANDLE; } self->params.usrCut = (FIXP_DBL)( (INT)(DRC_PARAM_QUANT_STEP >> DRC_PARAM_SCALE) * (INT)value); self->update = 1; break; case DRC_BOOST_SCALE: /* set boost factor */ if ((value < 0) || (value > DRC_MAX_QUANT_FACTOR)) { return AAC_DEC_SET_PARAM_FAIL; } if (self == NULL) { return AAC_DEC_INVALID_HANDLE; } self->params.usrBoost = (FIXP_DBL)( (INT)(DRC_PARAM_QUANT_STEP >> DRC_PARAM_SCALE) * (INT)value); self->update = 1; break; case TARGET_REF_LEVEL: if (value > MAX_REFERENCE_LEVEL || value < -MAX_REFERENCE_LEVEL) { return AAC_DEC_SET_PARAM_FAIL; } if (self == NULL) { return AAC_DEC_INVALID_HANDLE; } if (value < 0) { self->params.targetRefLevel = -1; } else { if (self->params.targetRefLevel != (SCHAR)value) { self->params.targetRefLevel = (SCHAR)value; self->progRefLevel = (SCHAR)value; /* Always set the program reference level equal to the target level according to 4.5.2.7.3 of ISO/IEC 14496-3. */ } self->update = 1; } break; case APPLY_HEAVY_COMPRESSION: if ((value != OFF) && (value != ON)) { return AAC_DEC_SET_PARAM_FAIL; } if (self == NULL) { return AAC_DEC_INVALID_HANDLE; } /* Store new parameter value */ self->params.usrApplyHeavyCompression = (UCHAR)value; self->update = 1; break; case DEFAULT_PRESENTATION_MODE: if (value < AAC_DRC_PARAMETER_HANDLING_DISABLED || value > AAC_DRC_PRESENTATION_MODE_2_DEFAULT) { return AAC_DEC_SET_PARAM_FAIL; } if (self == NULL) { return AAC_DEC_INVALID_HANDLE; } self->params.defaultPresentationMode = (AACDEC_DRC_PARAMETER_HANDLING)value; self->update = 1; break; case ENCODER_TARGET_LEVEL: if (value > MAX_REFERENCE_LEVEL || value < 0) { return AAC_DEC_SET_PARAM_FAIL; } if (self == NULL) { return AAC_DEC_INVALID_HANDLE; } self->params.encoderTargetLevel = (UCHAR)value; self->update = 1; break; case DRC_BS_DELAY: if (value < 0 || value > 1) { return AAC_DEC_SET_PARAM_FAIL; } if (self == NULL) { return AAC_DEC_INVALID_HANDLE; } self->params.bsDelayEnable = value; break; case DRC_DATA_EXPIRY_FRAME: if (self == NULL) { return AAC_DEC_INVALID_HANDLE; } self->params.expiryFrame = (value > 0) ? (UINT)value : 0; break; case MAX_OUTPUT_CHANNELS: if (self == NULL) { return AAC_DEC_INVALID_HANDLE; } self->numOutChannels = (INT)value; self->update = 1; break; case UNIDRC_PRECEDENCE: if (self == NULL) { return AAC_DEC_INVALID_HANDLE; } self->uniDrcPrecedence = (UCHAR)value; break; default: return AAC_DEC_SET_PARAM_FAIL; } /* switch(param) */ return ErrorStatus; } static int parseExcludedChannels(UINT *excludedChnsMask, HANDLE_FDK_BITSTREAM bs) { UINT excludeMask = 0; UINT i, j; int bitCnt = 9; for (i = 0, j = 1; i < 7; i++, j <<= 1) { if (FDKreadBits(bs, 1)) { excludeMask |= j; } } /* additional_excluded_chns */ while (FDKreadBits(bs, 1)) { for (i = 0; i < 7; i++, j <<= 1) { if (FDKreadBits(bs, 1)) { excludeMask |= j; } } bitCnt += 9; FDK_ASSERT(j < (UINT)-1); } *excludedChnsMask = excludeMask; return (bitCnt); } /*! \brief Save DRC payload bitstream position \self Handle of DRC info \bs Handle of FDK bitstream \return The number of DRC payload bits */ int aacDecoder_drcMarkPayload(HANDLE_AAC_DRC self, HANDLE_FDK_BITSTREAM bs, AACDEC_DRC_PAYLOAD_TYPE type) { UINT bsStartPos; int i, numBands = 1, bitCnt = 0; if (self == NULL) { return 0; } bsStartPos = FDKgetValidBits(bs); switch (type) { case MPEG_DRC_EXT_DATA: { bitCnt = 4; if (FDKreadBits(bs, 1)) { /* pce_tag_present */ FDKreadBits(bs, 8); /* pce_instance_tag + drc_tag_reserved_bits */ bitCnt += 8; } if (FDKreadBits(bs, 1)) { /* excluded_chns_present */ FDKreadBits(bs, 7); /* exclude mask [0..7] */ bitCnt += 8; while (FDKreadBits(bs, 1)) { /* additional_excluded_chns */ FDKreadBits(bs, 7); /* exclude mask [x..y] */ bitCnt += 8; } } if (FDKreadBits(bs, 1)) { /* drc_bands_present */ numBands += FDKreadBits(bs, 4); /* drc_band_incr */ FDKreadBits(bs, 4); /* reserved */ bitCnt += 8; for (i = 0; i < numBands; i++) { FDKreadBits(bs, 8); /* drc_band_top[i] */ bitCnt += 8; } } if (FDKreadBits(bs, 1)) { /* prog_ref_level_present */ FDKreadBits(bs, 8); /* prog_ref_level + prog_ref_level_reserved_bits */ bitCnt += 8; } for (i = 0; i < numBands; i++) { FDKreadBits(bs, 8); /* dyn_rng_sgn[i] + dyn_rng_ctl[i] */ bitCnt += 8; } if ((self->numPayloads < MAX_DRC_THREADS) && ((INT)FDKgetValidBits(bs) >= 0)) { self->drcPayloadPosition[self->numPayloads++] = bsStartPos; } } break; case DVB_DRC_ANC_DATA: bitCnt += 8; /* check sync word */ if (FDKreadBits(bs, 8) == DVB_ANC_DATA_SYNC_BYTE) { int dmxLevelsPresent, compressionPresent; int coarseGrainTcPresent, fineGrainTcPresent; /* bs_info field */ FDKreadBits( bs, 8); /* mpeg_audio_type, dolby_surround_mode, presentation_mode */ bitCnt += 8; /* Evaluate ancillary_data_status */ FDKreadBits(bs, 3); /* reserved, set to 0 */ dmxLevelsPresent = FDKreadBits(bs, 1); /* downmixing_levels_MPEG4_status */ FDKreadBits(bs, 1); /* reserved, set to 0 */ compressionPresent = FDKreadBits(bs, 1); /* audio_coding_mode_and_compression status */ coarseGrainTcPresent = FDKreadBits(bs, 1); /* coarse_grain_timecode_status */ fineGrainTcPresent = FDKreadBits(bs, 1); /* fine_grain_timecode_status */ bitCnt += 8; /* MPEG4 downmixing levels */ if (dmxLevelsPresent) { FDKreadBits(bs, 8); /* downmixing_levels_MPEG4 */ bitCnt += 8; } /* audio coding mode and compression status */ if (compressionPresent) { FDKreadBits(bs, 16); /* audio_coding_mode, Compression_value */ bitCnt += 16; } /* coarse grain timecode */ if (coarseGrainTcPresent) { FDKreadBits(bs, 16); /* coarse_grain_timecode */ bitCnt += 16; } /* fine grain timecode */ if (fineGrainTcPresent) { FDKreadBits(bs, 16); /* fine_grain_timecode */ bitCnt += 16; } if (!self->dvbAncDataAvailable && ((INT)FDKgetValidBits(bs) >= 0)) { self->dvbAncDataPosition = bsStartPos; self->dvbAncDataAvailable = 1; } } break; default: break; } return (bitCnt); } /*! \brief Parse DRC parameters from bitstream \bs Handle of FDK bitstream (in) \pDrcBs Pointer to DRC payload data container (out) \payloadPosition Bitstream position of MPEG DRC data chunk (in) \return Flag telling whether new DRC data has been found or not. */ static int aacDecoder_drcParse(HANDLE_FDK_BITSTREAM bs, CDrcPayload *pDrcBs, UINT payloadPosition) { int i, numBands; /* Move to the beginning of the DRC payload field */ FDKpushBiDirectional(bs, (INT)FDKgetValidBits(bs) - (INT)payloadPosition); /* pce_tag_present */ if (FDKreadBits(bs, 1)) { pDrcBs->pceInstanceTag = FDKreadBits(bs, 4); /* pce_instance_tag */ /* only one program supported */ FDKreadBits(bs, 4); /* drc_tag_reserved_bits */ } else { pDrcBs->pceInstanceTag = -1; /* not present */ } if (FDKreadBits(bs, 1)) { /* excluded_chns_present */ /* get excluded_chn_mask */ parseExcludedChannels(&pDrcBs->excludedChnsMask, bs); } else { pDrcBs->excludedChnsMask = 0; } numBands = 1; if (FDKreadBits(bs, 1)) /* drc_bands_present */ { /* get band_incr */ numBands += FDKreadBits(bs, 4); /* drc_band_incr */ pDrcBs->channelData.drcInterpolationScheme = FDKreadBits(bs, 4); /* drc_interpolation_scheme */ /* band_top */ for (i = 0; i < numBands; i++) { pDrcBs->channelData.bandTop[i] = FDKreadBits(bs, 8); /* drc_band_top[i] */ } } else { pDrcBs->channelData.bandTop[0] = DRC_BLOCK_LEN_DIV_BAND_MULT - 1; /* ... comprising the whole spectrum. */ ; } pDrcBs->channelData.numBands = numBands; if (FDKreadBits(bs, 1)) /* prog_ref_level_present */ { pDrcBs->progRefLevel = FDKreadBits(bs, 7); /* prog_ref_level */ FDKreadBits(bs, 1); /* prog_ref_level_reserved_bits */ } else { pDrcBs->progRefLevel = -1; } for (i = 0; i < numBands; i++) { pDrcBs->channelData.drcValue[i] = FDKreadBits(bs, 1) << 7; /* dyn_rng_sgn[i] */ pDrcBs->channelData.drcValue[i] |= FDKreadBits(bs, 7) & 0x7F; /* dyn_rng_ctl[i] */ } /* Set DRC payload type */ pDrcBs->channelData.drcDataType = MPEG_DRC_EXT_DATA; return (1); } /*! \brief Parse heavy compression value transported in DSEs of DVB streams with MPEG-4 content. \bs Handle of FDK bitstream (in) \pDrcBs Pointer to DRC payload data container (out) \payloadPosition Bitstream position of DVB ancillary data chunk \return Flag telling whether new DRC data has been found or not. */ #define DVB_COMPRESSION_SCALE (8) /* 48,164 dB */ static int aacDecoder_drcReadCompression(HANDLE_FDK_BITSTREAM bs, CDrcPayload *pDrcBs, UINT payloadPosition) { int foundDrcData = 0; int dmxLevelsPresent, compressionPresent; /* Move to the beginning of the DRC payload field */ FDKpushBiDirectional(bs, (INT)FDKgetValidBits(bs) - (INT)payloadPosition); /* Sanity checks */ if (FDKgetValidBits(bs) < 24) { return 0; } /* Check sync word */ if (FDKreadBits(bs, 8) != DVB_ANC_DATA_SYNC_BYTE) { return 0; } /* Evaluate bs_info field */ if (FDKreadBits(bs, 2) != 3) { /* mpeg_audio_type */ /* No MPEG-4 audio data */ return 0; } FDKreadBits(bs, 2); /* dolby_surround_mode */ pDrcBs->presMode = FDKreadBits(bs, 2); /* presentation_mode */ FDKreadBits(bs, 1); /* stereo_downmix_mode */ if (FDKreadBits(bs, 1) != 0) { /* reserved, set to 0 */ return 0; } /* Evaluate ancillary_data_status */ if (FDKreadBits(bs, 3) != 0) { /* reserved, set to 0 */ return 0; } dmxLevelsPresent = FDKreadBits(bs, 1); /* downmixing_levels_MPEG4_status */ /*extensionPresent =*/FDKreadBits(bs, 1); /* ancillary_data_extension_status; */ compressionPresent = FDKreadBits(bs, 1); /* audio_coding_mode_and_compression status */ /*coarseGrainTcPresent =*/FDKreadBits(bs, 1); /* coarse_grain_timecode_status */ /*fineGrainTcPresent =*/FDKreadBits(bs, 1); /* fine_grain_timecode_status */ if (dmxLevelsPresent) { FDKreadBits(bs, 8); /* downmixing_levels_MPEG4 */ } /* audio_coding_mode_and_compression_status */ if (compressionPresent) { UCHAR compressionOn, compressionValue; /* audio_coding_mode */ if (FDKreadBits(bs, 7) != 0) { /* The reserved bits shall be set to "0". */ return 0; } compressionOn = (UCHAR)FDKreadBits(bs, 1); /* compression_on */ compressionValue = (UCHAR)FDKreadBits(bs, 8); /* Compression_value */ if (compressionOn) { /* A compression value is available so store the data just like MPEG DRC * data */ pDrcBs->channelData.numBands = 1; /* One band ... */ pDrcBs->channelData.drcValue[0] = compressionValue; /* ... with one value ... */ pDrcBs->channelData.bandTop[0] = DRC_BLOCK_LEN_DIV_BAND_MULT - 1; /* ... comprising the whole spectrum. */ ; pDrcBs->pceInstanceTag = -1; /* Not present */ pDrcBs->progRefLevel = -1; /* Not present */ pDrcBs->channelData.drcDataType = DVB_DRC_ANC_DATA; /* Set DRC payload type to DVB. */ foundDrcData = 1; } } return (foundDrcData); } /* * Extract DRC payload from bitstream and map it to channels. * Valid return values are: * -1 : An unexpected error occured. * 0 : No error and no valid DRC data available. * 1 : No error and valid DRC data has been mapped. */ static int aacDecoder_drcExtractAndMap( HANDLE_AAC_DRC self, HANDLE_FDK_BITSTREAM hBs, CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo[], UCHAR pceInstanceTag, UCHAR channelMapping[], /* Channel mapping translating drcChannel index to canonical channel index */ int validChannels) { CDrcPayload threadBs[MAX_DRC_THREADS]; CDrcPayload *validThreadBs[MAX_DRC_THREADS]; CDrcParams *pParams; UINT backupBsPosition; int result = 0; int i, thread, validThreads = 0; FDK_ASSERT(self != NULL); FDK_ASSERT(hBs != NULL); FDK_ASSERT(pAacDecoderStaticChannelInfo != NULL); pParams = &self->params; self->numThreads = 0; backupBsPosition = FDKgetValidBits(hBs); for (i = 0; i < self->numPayloads && self->numThreads < MAX_DRC_THREADS; i++) { /* Init payload data chunk. The memclear is very important because it initializes the most values. Without it the module wouldn't work properly or crash. */ FDKmemclear(&threadBs[self->numThreads], sizeof(CDrcPayload)); threadBs[self->numThreads].channelData.bandTop[0] = DRC_BLOCK_LEN_DIV_BAND_MULT - 1; /* Extract payload */ self->numThreads += aacDecoder_drcParse(hBs, &threadBs[self->numThreads], self->drcPayloadPosition[i]); } self->numPayloads = 0; if (self->dvbAncDataAvailable && self->numThreads < MAX_DRC_THREADS) { /* Append a DVB heavy compression payload thread if available. */ /* Init payload data chunk. The memclear is very important because it initializes the most values. Without it the module wouldn't work properly or crash. */ FDKmemclear(&threadBs[self->numThreads], sizeof(CDrcPayload)); threadBs[self->numThreads].channelData.bandTop[0] = DRC_BLOCK_LEN_DIV_BAND_MULT - 1; /* Extract payload */ self->numThreads += aacDecoder_drcReadCompression( hBs, &threadBs[self->numThreads], self->dvbAncDataPosition); } self->dvbAncDataAvailable = 0; /* Reset the bitbufffer */ FDKpushBiDirectional(hBs, (INT)FDKgetValidBits(hBs) - (INT)backupBsPosition); /* calculate number of valid bits in excl_chn_mask */ /* coupling channels not supported */ /* check for valid threads */ for (thread = 0; thread < self->numThreads; thread++) { CDrcPayload *pThreadBs = &threadBs[thread]; int numExclChns = 0; switch ((AACDEC_DRC_PAYLOAD_TYPE)pThreadBs->channelData.drcDataType) { default: continue; case MPEG_DRC_EXT_DATA: case DVB_DRC_ANC_DATA: break; } if (pThreadBs->pceInstanceTag >= 0) { /* if PCE tag present */ if (pThreadBs->pceInstanceTag != pceInstanceTag) { continue; /* don't accept */ } } /* calculate number of excluded channels */ if (pThreadBs->excludedChnsMask > 0) { INT exclMask = pThreadBs->excludedChnsMask; int ch; for (ch = 0; ch < validChannels; ch++) { numExclChns += exclMask & 0x1; exclMask >>= 1; } } if (numExclChns < validChannels) { validThreadBs[validThreads] = pThreadBs; validThreads++; } } /* map DRC bitstream information onto DRC channel information */ for (thread = 0; thread < validThreads; thread++) { CDrcPayload *pThreadBs = validThreadBs[thread]; INT exclMask = pThreadBs->excludedChnsMask; AACDEC_DRC_PAYLOAD_TYPE drcPayloadType = (AACDEC_DRC_PAYLOAD_TYPE)pThreadBs->channelData.drcDataType; int ch; /* last progRefLevel transmitted is the one that is used * (but it should really only be transmitted once per block!) */ if (pThreadBs->progRefLevel >= 0) { self->progRefLevel = pThreadBs->progRefLevel; self->progRefLevelPresent = 1; self->prlExpiryCount = 0; /* Got a new value -> Reset counter */ } if (drcPayloadType == DVB_DRC_ANC_DATA) { /* Announce the presentation mode of this valid thread. */ self->presMode = pThreadBs->presMode; } /* SCE, CPE and LFE */ for (ch = 0; ch < validChannels; ch++) { AACDEC_DRC_PAYLOAD_TYPE prvPayloadType = UNKNOWN_PAYLOAD; int mapedChannel = channelMapping[ch]; if ((mapedChannel >= validChannels) || ((exclMask & (1 << mapedChannel)) != 0)) continue; if ((pParams->expiryFrame <= 0) || (pAacDecoderStaticChannelInfo[ch]->drcData.expiryCount < pParams->expiryFrame)) { prvPayloadType = (AACDEC_DRC_PAYLOAD_TYPE)pAacDecoderStaticChannelInfo[ch] ->drcData.drcDataType; } if (((drcPayloadType == MPEG_DRC_EXT_DATA) && (prvPayloadType != DVB_DRC_ANC_DATA)) || ((drcPayloadType == DVB_DRC_ANC_DATA) && (pParams->applyHeavyCompression == ON))) { /* copy thread to channel */ pAacDecoderStaticChannelInfo[ch]->drcData = pThreadBs->channelData; result = 1; } } /* CCEs not supported by now */ } /* Increment and check expiry counter for the program reference level: */ if ((pParams->expiryFrame > 0) && (self->prlExpiryCount++ > pParams->expiryFrame)) { /* The program reference level is too old, so set it back to the target level. */ self->progRefLevelPresent = 0; self->progRefLevel = pParams->targetRefLevel; self->prlExpiryCount = 0; } return result; } void aacDecoder_drcApply(HANDLE_AAC_DRC self, void *pSbrDec, CAacDecoderChannelInfo *pAacDecoderChannelInfo, CDrcChannelData *pDrcChData, FIXP_DBL *extGain, int ch, /* needed only for SBR */ int aacFrameSize, int bSbrPresent) { int band, bin, numBands; int bottom = 0; int modifyBins = 0; FIXP_DBL max_mantissa; INT max_exponent; FIXP_DBL norm_mantissa = FL2FXCONST_DBL(0.5f); INT norm_exponent = 1; FIXP_DBL fact_mantissa[MAX_DRC_BANDS]; INT fact_exponent[MAX_DRC_BANDS]; CDrcParams *pParams = &self->params; FIXP_DBL *pSpectralCoefficient = SPEC_LONG(pAacDecoderChannelInfo->pSpectralCoefficient); CIcsInfo *pIcsInfo = &pAacDecoderChannelInfo->icsInfo; SHORT *pSpecScale = pAacDecoderChannelInfo->specScale; int winSeq = pIcsInfo->WindowSequence; /* Increment and check expiry counter */ if ((pParams->expiryFrame > 0) && (++pDrcChData->expiryCount > pParams->expiryFrame)) { /* The DRC data is too old, so delete it. */ aacDecoder_drcInitChannelData(pDrcChData); } if (self->enable != ON) { sbrDecoder_drcDisable((HANDLE_SBRDECODER)pSbrDec, ch); if (extGain != NULL) { INT gainScale = (INT)*extGain; /* The gain scaling must be passed to the function in the buffer pointed * on by extGain. */ if (gainScale >= 0 && gainScale <= DFRACT_BITS) { *extGain = scaleValue(norm_mantissa, norm_exponent - gainScale); } else { FDK_ASSERT(0); } } return; } numBands = pDrcChData->numBands; /* If program reference normalization is done in the digital domain, modify factor to perform normalization. prog_ref_level can alternatively be passed to the system for modification of the level in the analog domain. Analog level modification avoids problems with reduced DAC SNR (if signal is attenuated) or clipping (if signal is boosted) */ if (pParams->targetRefLevel >= 0) { /* 0.5^((targetRefLevel - progRefLevel)/24) */ norm_mantissa = fLdPow(FL2FXCONST_DBL(-1.0), /* log2(0.5) */ 0, (FIXP_DBL)((INT)(FL2FXCONST_DBL(1.0f / 24.0) >> 3) * (INT)(pParams->targetRefLevel - self->progRefLevel)), 3, &norm_exponent); } /* Always export the normalization gain (if possible). */ if (extGain != NULL) { INT gainScale = (INT)*extGain; /* The gain scaling must be passed to the function in the buffer pointed on * by extGain. */ if (gainScale >= 0 && gainScale <= DFRACT_BITS) { *extGain = scaleValue(norm_mantissa, norm_exponent - gainScale); } else { FDK_ASSERT(0); } } /* Reset normalization gain since this module must not apply it */ norm_mantissa = FL2FXCONST_DBL(0.5f); norm_exponent = 1; /* calc scale factors */ for (band = 0; band < numBands; band++) { UCHAR drcVal = pDrcChData->drcValue[band]; fact_mantissa[band] = FL2FXCONST_DBL(0.5f); fact_exponent[band] = 1; if ((pParams->applyHeavyCompression == ON) && ((AACDEC_DRC_PAYLOAD_TYPE)pDrcChData->drcDataType == DVB_DRC_ANC_DATA)) { INT compressionFactorVal_e; int valX, valY; valX = drcVal >> 4; valY = drcVal & 0x0F; /* calculate the unscaled heavy compression factor. compressionFactor = 48.164 - 6.0206*valX - 0.4014*valY dB range: -48.166 dB to 48.164 dB */ if (drcVal != 0x7F) { fact_mantissa[band] = fPowInt( FL2FXCONST_DBL(0.95483867181), /* -0.4014dB = 0.95483867181 */ 0, valY, &compressionFactorVal_e); /* -0.0008dB (48.164 - 6.0206*8 = -0.0008) */ fact_mantissa[band] = fMult(FL2FXCONST_DBL(0.99990790084), fact_mantissa[band]); fact_exponent[band] = DVB_COMPRESSION_SCALE - valX + compressionFactorVal_e; } } else if ((AACDEC_DRC_PAYLOAD_TYPE)pDrcChData->drcDataType == MPEG_DRC_EXT_DATA) { /* apply the scaled dynamic range control words to factor. * if scaling drc_cut (or drc_boost), or control word drc_mantissa is 0 * then there is no dynamic range compression * * if pDrcChData->drcSgn[band] is * 1 then gain is < 1 : factor = 2^(-self->cut * * pDrcChData->drcMag[band] / 24) 0 then gain is > 1 : factor = 2^( * self->boost * pDrcChData->drcMag[band] / 24) */ if ((drcVal & 0x7F) > 0) { FIXP_DBL tParamVal = (drcVal & 0x80) ? -pParams->cut : pParams->boost; fact_mantissa[band] = f2Pow( (FIXP_DBL)((INT)fMult(FL2FXCONST_DBL(1.0f / 192.0f), tParamVal) * (drcVal & 0x7F)), 3 + DRC_PARAM_SCALE, &fact_exponent[band]); } } fact_mantissa[band] = fMult(fact_mantissa[band], norm_mantissa); fact_exponent[band] += norm_exponent; } /* end loop over bands */ /* normalizations */ { int res; max_mantissa = FL2FXCONST_DBL(0.0f); max_exponent = 0; for (band = 0; band < numBands; band++) { max_mantissa = fixMax(max_mantissa, fact_mantissa[band]); max_exponent = fixMax(max_exponent, fact_exponent[band]); } /* left shift factors to gain accurancy */ res = CntLeadingZeros(max_mantissa) - 1; /* above topmost DRC band gain factor is 1 */ if (((pDrcChData->bandTop[fMax(0, numBands - 1)] + 1) << 2) < aacFrameSize) res = 0; if (res > 0) { res = fixMin(res, max_exponent); max_exponent -= res; for (band = 0; band < numBands; band++) { fact_mantissa[band] <<= res; fact_exponent[band] -= res; } } /* normalize magnitudes to one scale factor */ for (band = 0; band < numBands; band++) { if (fact_exponent[band] < max_exponent) { fact_mantissa[band] >>= max_exponent - fact_exponent[band]; } if (fact_mantissa[band] != FL2FXCONST_DBL(0.5f)) { modifyBins = 1; } } if (max_exponent != 1) { modifyBins = 1; } } /* apply factor to spectral lines * short blocks must take care that bands fall on * block boundaries! */ if (!bSbrPresent) { bottom = 0; if (!modifyBins) { /* We don't have to modify the spectral bins because the fractional part of all factors is 0.5. In order to keep accurancy we don't apply the factor but decrease the exponent instead. */ max_exponent -= 1; } else { for (band = 0; band < numBands; band++) { int top = fixMin((int)((pDrcChData->bandTop[band] + 1) << 2), aacFrameSize); /* ... * DRC_BAND_MULT; */ for (bin = bottom; bin < top; bin++) { pSpectralCoefficient[bin] = fMult(pSpectralCoefficient[bin], fact_mantissa[band]); } bottom = top; } } /* above topmost DRC band gain factor is 1 */ if (max_exponent > 0) { for (bin = bottom; bin < aacFrameSize; bin += 1) { pSpectralCoefficient[bin] >>= max_exponent; } } /* adjust scaling */ pSpecScale[0] += max_exponent; if (winSeq == BLOCK_SHORT) { int win; for (win = 1; win < 8; win++) { pSpecScale[win] += max_exponent; } } } else { HANDLE_SBRDECODER hSbrDecoder = (HANDLE_SBRDECODER)pSbrDec; numBands = pDrcChData->numBands; /* feed factors into SBR decoder for application in QMF domain. */ sbrDecoder_drcFeedChannel(hSbrDecoder, ch, numBands, fact_mantissa, max_exponent, pDrcChData->drcInterpolationScheme, winSeq, pDrcChData->bandTop); } return; } /* * DRC parameter and presentation mode handling */ static void aacDecoder_drcParameterHandling(HANDLE_AAC_DRC self, INT aacNumChannels, SCHAR prevDrcProgRefLevel, SCHAR prevDrcPresMode) { int isDownmix, isMonoDownmix, isStereoDownmix; int dDmx, dHr; AACDEC_DRC_PARAMETER_HANDLING drcParameterHandling; CDrcParams *p; FDK_ASSERT(self != NULL); p = &self->params; if (self->progRefLevel != prevDrcProgRefLevel) self->update = 1; if (self->presMode != prevDrcPresMode) self->update = 1; if (self->prevAacNumChannels != aacNumChannels) self->update = 1; /* return if no relevant parameter has changed */ if (!self->update) { return; } /* derive downmix property. aacNumChannels: number of channels in aac stream, * numOutChannels: number of output channels */ isDownmix = (aacNumChannels > self->numOutChannels); isDownmix = (isDownmix && (self->numOutChannels > 0)); isMonoDownmix = (isDownmix && (self->numOutChannels == 1)); isStereoDownmix = (isDownmix && (self->numOutChannels == 2)); if ((self->presMode == 1) || (self->presMode == 2)) { drcParameterHandling = (AACDEC_DRC_PARAMETER_HANDLING)self->presMode; } else { /* no presentation mode -> use parameter handling specified by AAC_DRC_DEFAULT_PRESENTATION_MODE */ drcParameterHandling = p->defaultPresentationMode; } /* by default, do as desired */ p->cut = p->usrCut; p->boost = p->usrBoost; p->applyHeavyCompression = p->usrApplyHeavyCompression; switch (drcParameterHandling) { case DISABLED_PARAMETER_HANDLING: default: /* use drc parameters as requested */ break; case ENABLED_PARAMETER_HANDLING: /* dDmx: estimated headroom reduction due to downmix, format: -1/4*dB dDmx = floor(-4*20*log10(aacNumChannels/numOutChannels)) */ if (isDownmix) { FIXP_DBL dmxTmp; int e_log, e_mult; dmxTmp = fDivNorm(self->numOutChannels, aacNumChannels); /* inverse division -> negative sign after logarithm */ dmxTmp = fLog2(dmxTmp, 0, &e_log); dmxTmp = fMultNorm( dmxTmp, FL2FXCONST_DBL(4.0f * 20.0f * 0.30103f / (float)(1 << 5)), &e_mult); /* e = e_log + e_mult + 5 */ dDmx = (int)scaleValue(dmxTmp, e_log + e_mult + 5 - (DFRACT_BITS - 1)); } else { dDmx = 0; } /* dHr: Full estimated (decoder) headroom reduction due to loudness * normalisation (DTL - PRL) and downmix. Format: -1/4*dB */ if (p->targetRefLevel >= 0) { /* if target level is provided */ dHr = p->targetRefLevel + dDmx - self->progRefLevel; } else { dHr = dDmx; } if (dHr < 0) { /* if headroom is reduced */ /* Use compression, but as little as possible. */ /* eHr: Headroom provided by encoder, format: -1/4 dB */ int eHr = fixMin(p->encoderTargetLevel - self->progRefLevel, 0); if (eHr < dHr) { /* if encoder provides more headroom than decoder needs */ /* derive scaling of light DRC */ FIXP_DBL calcFactor_norm; INT calcFactor; /* fraction of DRC gains that is minimally needed for clipping prevention */ calcFactor_norm = fDivNorm(-dHr, -eHr); /* 0.0 < calcFactor_norm < 1.0 */ calcFactor_norm = calcFactor_norm >> DRC_PARAM_SCALE; /* quantize to 128 steps */ calcFactor = convert_drcParam( calcFactor_norm); /* convert to integer value between 0 and 127 */ calcFactor_norm = (FIXP_DBL)( (INT)(DRC_PARAM_QUANT_STEP >> DRC_PARAM_SCALE) * calcFactor); p->cut = (calcFactor_norm > p->cut) ? calcFactor_norm : p->cut; /* use calcFactor_norm as lower limit */ } else { /* encoder provides equal or less headroom than decoder needs */ /* the time domain limiter must always be active in this case. It is * assumed that the framework activates it by default */ p->cut = DRC_SCALING_MAX; if ((dHr - eHr) <= -4 * DRC_HEAVY_THRESHOLD_DB) { /* use heavy compression if headroom deficit is equal or higher than DRC_HEAVY_THRESHOLD_DB */ p->applyHeavyCompression = ON; } } } else { /* dHr >= 0 */ /* no restrictions required, as headroom is not reduced. */ /* p->cut = p->usrCut; */ } break; /* presentation mode 1 and 2 according to ETSI TS 101 154: Digital Video Broadcasting (DVB); Specification for the use of Video and Audio Coding in Broadcasting Applications based on the MPEG-2 Transport Stream, section C.5.4., "Decoding", and Table C.33. Also according to amendment 4 to ISO/IEC 14496-3, section 4.5.2.14.2.4, and Table AMD4.11. ISO DRC -> applyHeavyCompression = OFF (Use light compression, MPEG-style) Compression_value -> applyHeavyCompression = ON (Use heavy compression, DVB-style) scaling restricted -> p->cut = DRC_SCALING_MAX */ case DRC_PRESENTATION_MODE_1: /* presentation mode 1, Light:-31/Heavy:-23 */ if ((p->targetRefLevel >= 0) && (p->targetRefLevel < 124)) { /* if target level is provided and > -31 dB */ /* playback up to -23 dB */ p->applyHeavyCompression = ON; } else { /* target level <= -31 dB or not provided */ /* playback -31 dB */ if (isMonoDownmix || isStereoDownmix) { /* stereo or mono downmixing */ p->cut = DRC_SCALING_MAX; } } break; case DRC_PRESENTATION_MODE_2: /* presentation mode 2, Light:-23/Heavy:-23 */ if ((p->targetRefLevel >= 0) && (p->targetRefLevel < 124)) { /* if target level is provided and > -31 dB */ /* playback up to -23 dB */ if (isMonoDownmix) { /* if mono downmix */ p->applyHeavyCompression = ON; } else { p->applyHeavyCompression = OFF; p->cut = DRC_SCALING_MAX; } } else { /* target level <= -31 dB or not provided */ /* playback -31 dB */ p->applyHeavyCompression = OFF; if (isMonoDownmix || isStereoDownmix) { /* stereo or mono downmixing */ p->cut = DRC_SCALING_MAX; } } break; } /* switch (drcParameterHandling) */ /* With heavy compression, there is no scaling. Scaling factors are set for notification only. */ if (p->applyHeavyCompression == ON) { p->boost = DRC_SCALING_MAX; p->cut = DRC_SCALING_MAX; } /* switch on/off processing */ self->enable = ((p->boost > (FIXP_DBL)0) || (p->cut > (FIXP_DBL)0) || (p->applyHeavyCompression == ON) || (p->targetRefLevel >= 0)); self->enable = (self->enable && !self->uniDrcPrecedence); self->prevAacNumChannels = aacNumChannels; self->update = 0; } /* * Prepare DRC processing * Valid return values are: * -1 : An unexpected error occured. * 0 : No error and no valid DRC data available. * 1 : No error and valid DRC data has been mapped. */ int aacDecoder_drcProlog( HANDLE_AAC_DRC self, HANDLE_FDK_BITSTREAM hBs, CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo[], UCHAR pceInstanceTag, UCHAR channelMapping[], /* Channel mapping translating drcChannel index to canonical channel index */ int validChannels) { int result = 0; if (self == NULL) { return -1; } if (!self->params.bsDelayEnable) { /* keep previous progRefLevel and presMode for update flag in * drcParameterHandling */ INT prevPRL, prevPM = 0; prevPRL = self->progRefLevel; prevPM = self->presMode; result = aacDecoder_drcExtractAndMap( self, hBs, pAacDecoderStaticChannelInfo, pceInstanceTag, channelMapping, validChannels); if (result < 0) { return result; } /* Drc parameter handling */ aacDecoder_drcParameterHandling(self, validChannels, prevPRL, prevPM); } return result; } /* * Finalize DRC processing * Valid return values are: * -1 : An unexpected error occured. * 0 : No error and no valid DRC data available. * 1 : No error and valid DRC data has been mapped. */ int aacDecoder_drcEpilog( HANDLE_AAC_DRC self, HANDLE_FDK_BITSTREAM hBs, CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo[], UCHAR pceInstanceTag, UCHAR channelMapping[], /* Channel mapping translating drcChannel index to canonical channel index */ int validChannels) { int result = 0; if (self == NULL) { return -1; } if (self->params.bsDelayEnable) { /* keep previous progRefLevel and presMode for update flag in * drcParameterHandling */ INT prevPRL, prevPM = 0; prevPRL = self->progRefLevel; prevPM = self->presMode; result = aacDecoder_drcExtractAndMap( self, hBs, pAacDecoderStaticChannelInfo, pceInstanceTag, channelMapping, validChannels); if (result < 0) { return result; } /* Drc parameter handling */ aacDecoder_drcParameterHandling(self, validChannels, prevPRL, prevPM); } return result; } /* * Export relevant metadata info from bitstream payload. */ void aacDecoder_drcGetInfo(HANDLE_AAC_DRC self, SCHAR *pPresMode, SCHAR *pProgRefLevel) { if (self != NULL) { if (pPresMode != NULL) { *pPresMode = self->presMode; } if (pProgRefLevel != NULL) { if (self->progRefLevelPresent) { *pProgRefLevel = self->progRefLevel; } else { *pProgRefLevel = -1; } } } } /** * \brief Apply DRC Level Normalization. * * This function prepares/applies the gain values for the DRC Level * Normalization and returns the exponent of the time data. The following two * cases are handled: * * - Limiter enabled: * The input data must be interleaved. * One gain per sample is written to the buffer pGainPerSample. * If necessary the time data is rescaled. * * - Limiter disabled: * The input data can be interleaved or deinterleaved. * The gain values are applied to the time data. * If necessary the time data is rescaled. * * \param hDrcInfo [i/o] handle to drc data structure. * \param samplesIn [i/o] pointer to time data. * \param pGain [i ] pointer to gain to be applied to * the time data. * \param pGainPerSample [o ] pointer to the gain per sample to * be applied to the time data in the limiter. * \param gain_scale [i ] exponent to be applied to the time * data. * \param gain_delay [i ] delay[samples] with which the gains * in pGain shall be applied (gain_delay <= nSamples). * \param nSamples [i ] number of samples per frame. * \param channels [i ] number of channels. * \param stride [i ] channel stride of time data. * \param limiterEnabled [i ] 1 if limiter is enabled, otherwise * 0. * * \return exponent of time data */ INT applyDrcLevelNormalization(HANDLE_AAC_DRC hDrcInfo, PCM_DEC *samplesIn, FIXP_DBL *pGain, FIXP_DBL *pGainPerSample, const INT gain_scale, const UINT gain_delay, const UINT nSamples, const UINT channels, const UINT stride, const UINT limiterEnabled) { UINT i; INT additionalGain_scaling; FIXP_DBL additionalGain; FDK_ASSERT(gain_delay <= nSamples); FIXP_DBL additionalGainSmoothState = hDrcInfo->additionalGainFilterState; FIXP_DBL additionalGainSmoothState1 = hDrcInfo->additionalGainFilterState1; if (!gain_delay) { additionalGain = pGain[0]; /* Apply the additional scaling gain_scale[0] that has no delay and no * smoothing */ additionalGain_scaling = fMin(gain_scale, CntLeadingZeros(additionalGain) - 1); additionalGain = scaleValue(additionalGain, additionalGain_scaling); /* if it's not possible to fully apply gain_scale to additionalGain, apply * it to the input signal */ additionalGain_scaling -= gain_scale; if (additionalGain_scaling) { scaleValuesSaturate(samplesIn, channels * nSamples, -additionalGain_scaling); } if (limiterEnabled) { FDK_ASSERT(pGainPerSample != NULL); for (i = 0; i < nSamples; i++) { pGainPerSample[i] = additionalGain; } } else { for (i = 0; i < channels * nSamples; i++) { samplesIn[i] = FIXP_DBL2PCM_DEC(fMult(samplesIn[i], additionalGain)); } } } else { UINT inc; FIXP_DBL additionalGainUnfiltered; inc = (stride == 1) ? channels : 1; for (i = 0; i < nSamples; i++) { if (i < gain_delay) { additionalGainUnfiltered = hDrcInfo->additionalGainPrev; } else { additionalGainUnfiltered = pGain[0]; } /* Smooth additionalGain */ /* [b,a] = butter(1, 0.01) */ static const FIXP_SGL b[] = {FL2FXCONST_SGL(0.015466 * 2.0), FL2FXCONST_SGL(0.015466 * 2.0)}; static const FIXP_SGL a[] = {(FIXP_SGL)MAXVAL_SGL, FL2FXCONST_SGL(-0.96907)}; additionalGain = -fMult(additionalGainSmoothState, a[1]) + fMultDiv2(additionalGainUnfiltered, b[0]) + fMultDiv2(additionalGainSmoothState1, b[1]); additionalGainSmoothState1 = additionalGainUnfiltered; additionalGainSmoothState = additionalGain; /* Apply the additional scaling gain_scale[0] that has no delay and no * smoothing */ additionalGain_scaling = fMin(gain_scale, CntLeadingZeros(additionalGain) - 1); additionalGain = scaleValue(additionalGain, additionalGain_scaling); /* if it's not possible to fully apply gain_scale[0] to additionalGain, * apply it to the input signal */ additionalGain_scaling -= gain_scale; if (limiterEnabled) { FDK_ASSERT(stride == 1); FDK_ASSERT(pGainPerSample != NULL); if (additionalGain_scaling) { scaleValuesSaturate(samplesIn, channels, -additionalGain_scaling); } pGainPerSample[i] = additionalGain; } else { if (additionalGain_scaling) { for (UINT k = 0; k < channels; k++) { scaleValuesSaturate(&samplesIn[k * stride], 1, -additionalGain_scaling); } } for (UINT k = 0; k < channels; k++) { samplesIn[k * stride] = FIXP_DBL2PCM_DEC(fMult(samplesIn[k * stride], additionalGain)); } } samplesIn += inc; } } hDrcInfo->additionalGainPrev = pGain[0]; hDrcInfo->additionalGainFilterState = additionalGainSmoothState; hDrcInfo->additionalGainFilterState1 = additionalGainSmoothState1; return (AACDEC_DRC_GAIN_SCALING); } fdk-aac-2.0.2/libAACdec/src/aacdec_drc.h000066400000000000000000000240041404225310700175160ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Christian Griebel Description: Dynamic range control (DRC) decoder tool for AAC *******************************************************************************/ #ifndef AACDEC_DRC_H #define AACDEC_DRC_H #include "tp_data.h" /* for program config element support */ #include "aacdec_drc_types.h" #include "channel.h" #include "FDK_bitstream.h" #define AACDEC_DRC_GAIN_SCALING (11) /* Scaling of DRC gains */ #define AACDEC_DRC_GAIN_INIT_VALUE \ (FL2FXCONST_DBL( \ 1.0f / (1 << AACDEC_DRC_GAIN_SCALING))) /* Init value for DRC gains */ #define AACDEC_DRC_DFLT_EXPIRY_FRAMES \ (0) /* Default DRC data expiry time in AAC frames */ /* #define AACDEC_DRC_IGNORE_FRAMES_WITH_MULTIPLE_CH_THREADS */ /* The name says it all. */ /* #define AACDEC_DRC_DEBUG */ /** * \brief DRC module setting parameters */ typedef enum { DRC_CUT_SCALE = 0, DRC_BOOST_SCALE, TARGET_REF_LEVEL, DRC_BS_DELAY, DRC_DATA_EXPIRY_FRAME, APPLY_HEAVY_COMPRESSION, DEFAULT_PRESENTATION_MODE, ENCODER_TARGET_LEVEL, MAX_OUTPUT_CHANNELS, UNIDRC_PRECEDENCE } AACDEC_DRC_PARAM; /** * \brief DRC module interface functions */ void aacDecoder_drcReset(HANDLE_AAC_DRC self); void aacDecoder_drcInit(HANDLE_AAC_DRC self); void aacDecoder_drcInitChannelData(CDrcChannelData *pDrcChannel); AAC_DECODER_ERROR aacDecoder_drcSetParam(HANDLE_AAC_DRC self, AACDEC_DRC_PARAM param, INT value); int aacDecoder_drcMarkPayload(HANDLE_AAC_DRC self, HANDLE_FDK_BITSTREAM hBs, AACDEC_DRC_PAYLOAD_TYPE type); int aacDecoder_drcProlog( HANDLE_AAC_DRC self, HANDLE_FDK_BITSTREAM hBs, CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo[], UCHAR pceInstanceTag, UCHAR channelMapping[], int validChannels); /** * \brief Apply DRC. If SBR is present, DRC data is handed over to the SBR * decoder. * \param self AAC decoder instance * \param pSbrDec pointer to SBR decoder instance * \param pAacDecoderChannelInfo AAC decoder channel instance to be processed * \param pDrcDat DRC channel data * \param extGain Pointer to a FIXP_DBL where a externally applyable gain will * be stored into (independently on whether it will be apply internally or not). * At function call the buffer must hold the scale (0 >= scale < * DFRACT_BITS) to be applied on the gain value. * \param ch channel index * \param aacFrameSize AAC frame size * \param bSbrPresent flag indicating that SBR is present, in which case DRC is * handed over to the SBR instance pSbrDec */ void aacDecoder_drcApply(HANDLE_AAC_DRC self, void *pSbrDec, CAacDecoderChannelInfo *pAacDecoderChannelInfo, CDrcChannelData *pDrcDat, FIXP_DBL *extGain, int ch, int aacFrameSize, int bSbrPresent); int aacDecoder_drcEpilog( HANDLE_AAC_DRC self, HANDLE_FDK_BITSTREAM hBs, CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo[], UCHAR pceInstanceTag, UCHAR channelMapping[], int validChannels); /** * \brief Get metadata information found in bitstream. * \param self DRC module instance handle. * \param pPresMode Pointer to field where the presentation mode will be written * to. * \param pProgRefLevel Pointer to field where the program reference level will * be written to. * \return Nothing. */ void aacDecoder_drcGetInfo(HANDLE_AAC_DRC self, SCHAR *pPresMode, SCHAR *pProgRefLevel); /** * \brief Apply DRC Level Normalization. * * This function prepares/applies the gain values for the DRC Level * Normalization and returns the exponent of the time data. The following two * cases are handled: * * - Limiter enabled: * The input data must be interleaved. * One gain per sample is written to the buffer pGainPerSample. * If necessary the time data is rescaled. * * - Limiter disabled: * The input data can be interleaved or deinterleaved. * The gain values are applied to the time data. * If necessary the time data is rescaled. * * \param hDrcInfo [i/o] handle to drc data structure. * \param samplesIn [i/o] pointer to time data. * \param pGain [i ] pointer to gain to be applied to * the time data. * \param pGainPerSample [o ] pointer to the gain per sample to * be applied to the time data in the limiter. * \param gain_scale [i ] exponent to be applied to the time * data. * \param gain_delay [i ] delay[samples] with which the gains * in pGain shall be applied (gain_delay <= nSamples). * \param nSamples [i ] number of samples per frame. * \param channels [i ] number of channels. * \param stride [i ] channel stride of time data. * \param limiterEnabled [i ] 1 if limiter is enabled, otherwise * 0. * * \return exponent of time data */ INT applyDrcLevelNormalization(HANDLE_AAC_DRC hDrcInfo, PCM_DEC *samplesIn, FIXP_DBL *pGain, FIXP_DBL *pGainPerSample, const INT gain_scale, const UINT gain_delay, const UINT nSamples, const UINT channels, const UINT stride, const UINT limiterEnabled); #endif /* AACDEC_DRC_H */ fdk-aac-2.0.2/libAACdec/src/aacdec_drc_types.h000066400000000000000000000217321404225310700207470ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Christian Griebel Description: Dynamic range control (DRC) global data types *******************************************************************************/ #ifndef AACDEC_DRC_TYPES_H #define AACDEC_DRC_TYPES_H #include "common_fix.h" #define MAX_DRC_THREADS \ ((8) + 1) /* Heavy compression value is handled just like MPEG DRC data */ #define MAX_DRC_BANDS (16) /* 2^LEN_DRC_BAND_INCR (LEN_DRC_BAND_INCR = 4) */ /** * \brief DRC module global data types */ typedef enum { UNKNOWN_PAYLOAD = 0, MPEG_DRC_EXT_DATA = 1, DVB_DRC_ANC_DATA = 2 } AACDEC_DRC_PAYLOAD_TYPE; /** * \brief Options for parameter handling / presentation mode */ typedef enum { DISABLED_PARAMETER_HANDLING = -1, /*!< DRC parameter handling disabled, all parameters are applied as requested. */ ENABLED_PARAMETER_HANDLING = 0, /*!< Apply changes to requested DRC parameters to prevent clipping */ DRC_PRESENTATION_MODE_1 = 1, /*!< DRC Presentation mode 1*/ DRC_PRESENTATION_MODE_2 = 2 /*!< DRC Presentation mode 2*/ } AACDEC_DRC_PARAMETER_HANDLING; typedef struct { UINT expiryCount; UINT numBands; USHORT bandTop[MAX_DRC_BANDS]; SHORT drcInterpolationScheme; UCHAR drcValue[MAX_DRC_BANDS]; SCHAR drcDataType; } CDrcChannelData; typedef struct { UINT excludedChnsMask; SCHAR progRefLevel; SCHAR presMode; /* Presentation mode: 0 (not indicated), 1, 2, and 3 (reserved). */ SCHAR pceInstanceTag; CDrcChannelData channelData; } CDrcPayload; typedef struct { /* DRC parameters: Latest user requests */ FIXP_DBL usrCut; FIXP_DBL usrBoost; UCHAR usrApplyHeavyCompression; /* DRC parameters: Currently used, possibly changed by * aacDecoder_drcParameterHandling */ FIXP_DBL cut; /* attenuation scale factor */ FIXP_DBL boost; /* boost scale factor */ SCHAR targetRefLevel; /* target reference level for loudness normalization */ UCHAR applyHeavyCompression; /* heavy compression (DVB) flag */ UINT expiryFrame; UCHAR bsDelayEnable; AACDEC_DRC_PARAMETER_HANDLING defaultPresentationMode; UCHAR encoderTargetLevel; } CDrcParams; typedef struct { CDrcParams params; /* Module parameters that can be set by user (via SetParam API function) */ UCHAR enable; /* Switch that controls dynamic range processing */ UCHAR digitalNorm; /* Switch to en-/disable reference level normalization in digital domain */ UCHAR update; /* Flag indicating the change of a user or bitstream parameter which affects aacDecoder_drcParameterHandling */ INT numOutChannels; /* Number of output channels */ INT prevAacNumChannels; /* Previous number of channels of aac bitstream, used for update flag */ USHORT numPayloads; /* The number of DRC data payload elements found within frame */ USHORT numThreads; /* The number of DRC data threads extracted from the found payload elements */ SCHAR progRefLevel; /* Program reference level for all channels */ UCHAR progRefLevelPresent; /* Program reference level found in bitstream */ UINT prlExpiryCount; /* Counter that can be used to monitor the life time of the program reference level. */ SCHAR presMode; /* Presentation mode as defined in ETSI TS 101 154 */ UCHAR dvbAncDataAvailable; /* Flag that indicates whether DVB ancillary data is present or not */ UINT dvbAncDataPosition; /* Used to store the DVB ancillary data payload position in the bitstream (only one per frame) */ UINT drcPayloadPosition[MAX_DRC_THREADS]; /* Used to store the DRC payload positions in the bitstream */ UCHAR uniDrcPrecedence; /* Flag for signalling that uniDrc is active and takes precedence over legacy DRC */ UCHAR applyExtGain; /* Flag is 1 if extGain has to be applied, otherwise 0. */ FIXP_DBL additionalGainPrev; /* Gain of previous frame to be applied to the time data */ FIXP_DBL additionalGainFilterState; /* Filter state for the gain smoothing */ FIXP_DBL additionalGainFilterState1; /* Filter state for the gain smoothing */ } CDrcInfo; typedef CDrcInfo *HANDLE_AAC_DRC; #endif /* AACDEC_DRC_TYPES_H */ fdk-aac-2.0.2/libAACdec/src/aacdec_hcr.cpp000066400000000000000000001667531404225310700200770ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Robert Weidner (DSP Solutions) Description: HCR Decoder: HCR initialization, preprocess HCR sideinfo, decode priority codewords (PCWs) *******************************************************************************/ #include "aacdec_hcr.h" #include "aacdec_hcr_types.h" #include "aacdec_hcr_bit.h" #include "aacdec_hcrs.h" #include "aac_ram.h" #include "aac_rom.h" #include "channel.h" #include "block.h" #include "aacdecoder.h" /* for ID_CPE, ID_SCE ... */ #include "FDK_bitstream.h" extern int mlFileChCurr; static void errDetectorInHcrSideinfoShrt(SCHAR cb, SHORT numLine, UINT *errorWord); static void errDetectorInHcrLengths(SCHAR lengthOfLongestCodeword, SHORT lengthOfReorderedSpectralData, UINT *errorWord); static void HcrCalcNumCodeword(H_HCR_INFO pHcr); static void HcrSortCodebookAndNumCodewordInSection(H_HCR_INFO pHcr); static void HcrPrepareSegmentationGrid(H_HCR_INFO pHcr); static void HcrExtendedSectionInfo(H_HCR_INFO pHcr); static void DeriveNumberOfExtendedSortedSectionsInSets( UINT numSegment, USHORT *pNumExtendedSortedCodewordInSection, int numExtendedSortedCodewordInSectionIdx, USHORT *pNumExtendedSortedSectionsInSets, int numExtendedSortedSectionsInSetsIdx); static INT DecodeEscapeSequence(HANDLE_FDK_BITSTREAM bs, const INT bsAnchor, INT quantSpecCoef, INT *pLeftStartOfSegment, SCHAR *pRemainingBitsInSegment, int *pNumDecodedBits, UINT *errorWord); static int DecodePCW_Sign(HANDLE_FDK_BITSTREAM bs, const INT bsAnchor, UINT codebookDim, const SCHAR *pQuantVal, FIXP_DBL *pQuantSpecCoef, int *quantSpecCoefIdx, INT *pLeftStartOfSegment, SCHAR *pRemainingBitsInSegment, int *pNumDecodedBits); static const SCHAR *DecodePCW_Body(HANDLE_FDK_BITSTREAM bs, const INT bsAnchor, const UINT *pCurrentTree, const SCHAR *pQuantValBase, INT *pLeftStartOfSegment, SCHAR *pRemainingBitsInSegment, int *pNumDecodedBits); static void DecodePCWs(HANDLE_FDK_BITSTREAM bs, H_HCR_INFO pHcr); static void HcrReorderQuantizedSpectralCoefficients( H_HCR_INFO pHcr, CAacDecoderChannelInfo *pAacDecoderChannelInfo, const SamplingRateInfo *pSamplingRateInfo); static UCHAR errDetectPcwSegmentation(SCHAR remainingBitsInSegment, H_HCR_INFO pHcr, PCW_TYPE kind, FIXP_DBL *qsc_base_of_cw, UCHAR dimension); static void errDetectWithinSegmentationFinal(H_HCR_INFO pHcr); /*--------------------------------------------------------------------------------------------- description: Check if codebook and numSect are within allowed range (short only) -------------------------------------------------------------------------------------------- */ static void errDetectorInHcrSideinfoShrt(SCHAR cb, SHORT numLine, UINT *errorWord) { if (cb < ZERO_HCB || cb >= MAX_CB_CHECK || cb == BOOKSCL) { *errorWord |= CB_OUT_OF_RANGE_SHORT_BLOCK; } if (numLine < 0 || numLine > 1024) { *errorWord |= LINE_IN_SECT_OUT_OF_RANGE_SHORT_BLOCK; } } /*--------------------------------------------------------------------------------------------- description: Check both HCR lengths -------------------------------------------------------------------------------------------- */ static void errDetectorInHcrLengths(SCHAR lengthOfLongestCodeword, SHORT lengthOfReorderedSpectralData, UINT *errorWord) { if (lengthOfReorderedSpectralData < lengthOfLongestCodeword) { *errorWord |= HCR_SI_LENGTHS_FAILURE; } } /*--------------------------------------------------------------------------------------------- description: Decode (and adapt if necessary) the two HCR sideinfo components: 'reordered_spectral_data_length' and 'longest_codeword_length' -------------------------------------------------------------------------------------------- */ void CHcr_Read(HANDLE_FDK_BITSTREAM bs, CAacDecoderChannelInfo *pAacDecoderChannelInfo, const MP4_ELEMENT_ID globalHcrType) { SHORT lengOfReorderedSpectralData; SCHAR lengOfLongestCodeword; pAacDecoderChannelInfo->pDynData->specificTo.aac.lenOfReorderedSpectralData = 0; pAacDecoderChannelInfo->pDynData->specificTo.aac.lenOfLongestCodeword = 0; /* ------- SI-Value No 1 ------- */ lengOfReorderedSpectralData = FDKreadBits(bs, 14) + ERROR_LORSD; if (globalHcrType == ID_CPE) { if ((lengOfReorderedSpectralData >= 0) && (lengOfReorderedSpectralData <= CPE_TOP_LENGTH)) { pAacDecoderChannelInfo->pDynData->specificTo.aac .lenOfReorderedSpectralData = lengOfReorderedSpectralData; /* the decoded value is within range */ } else { if (lengOfReorderedSpectralData > CPE_TOP_LENGTH) { pAacDecoderChannelInfo->pDynData->specificTo.aac .lenOfReorderedSpectralData = CPE_TOP_LENGTH; /* use valid maximum */ } } } else if (globalHcrType == ID_SCE || globalHcrType == ID_LFE || globalHcrType == ID_CCE) { if ((lengOfReorderedSpectralData >= 0) && (lengOfReorderedSpectralData <= SCE_TOP_LENGTH)) { pAacDecoderChannelInfo->pDynData->specificTo.aac .lenOfReorderedSpectralData = lengOfReorderedSpectralData; /* the decoded value is within range */ } else { if (lengOfReorderedSpectralData > SCE_TOP_LENGTH) { pAacDecoderChannelInfo->pDynData->specificTo.aac .lenOfReorderedSpectralData = SCE_TOP_LENGTH; /* use valid maximum */ } } } /* ------- SI-Value No 2 ------- */ lengOfLongestCodeword = FDKreadBits(bs, 6) + ERROR_LOLC; if ((lengOfLongestCodeword >= 0) && (lengOfLongestCodeword <= LEN_OF_LONGEST_CW_TOP_LENGTH)) { pAacDecoderChannelInfo->pDynData->specificTo.aac.lenOfLongestCodeword = lengOfLongestCodeword; /* the decoded value is within range */ } else { if (lengOfLongestCodeword > LEN_OF_LONGEST_CW_TOP_LENGTH) { pAacDecoderChannelInfo->pDynData->specificTo.aac.lenOfLongestCodeword = LEN_OF_LONGEST_CW_TOP_LENGTH; /* use valid maximum */ } } } /*--------------------------------------------------------------------------------------------- description: Set up HCR - must be called before every call to HcrDecoder(). For short block a sorting algorithm is applied to get the SI in the order that HCR could assemble the qsc's as if it is a long block. ----------------------------------------------------------------------------------------------- return: error log -------------------------------------------------------------------------------------------- */ UINT HcrInit(H_HCR_INFO pHcr, CAacDecoderChannelInfo *pAacDecoderChannelInfo, const SamplingRateInfo *pSamplingRateInfo, HANDLE_FDK_BITSTREAM bs) { CIcsInfo *pIcsInfo = &pAacDecoderChannelInfo->icsInfo; SHORT *pNumLinesInSec; UCHAR *pCodeBk; SHORT numSection; SCHAR cb; int numLine; int i; pHcr->decInOut.lengthOfReorderedSpectralData = pAacDecoderChannelInfo->pDynData->specificTo.aac .lenOfReorderedSpectralData; pHcr->decInOut.lengthOfLongestCodeword = pAacDecoderChannelInfo->pDynData->specificTo.aac.lenOfLongestCodeword; pHcr->decInOut.pQuantizedSpectralCoefficientsBase = pAacDecoderChannelInfo->pSpectralCoefficient; pHcr->decInOut.quantizedSpectralCoefficientsIdx = 0; pHcr->decInOut.pCodebook = pAacDecoderChannelInfo->pDynData->specificTo.aac.aCodeBooks4Hcr; pHcr->decInOut.pNumLineInSect = pAacDecoderChannelInfo->pDynData->specificTo.aac.aNumLineInSec4Hcr; pHcr->decInOut.numSection = pAacDecoderChannelInfo->pDynData->specificTo.aac.numberSection; pHcr->decInOut.errorLog = 0; pHcr->nonPcwSideinfo.pResultBase = SPEC_LONG(pAacDecoderChannelInfo->pSpectralCoefficient); FDKsyncCache(bs); pHcr->decInOut.bitstreamAnchor = (INT)FDKgetValidBits(bs); if (!IsLongBlock(&pAacDecoderChannelInfo->icsInfo)) /* short block */ { SHORT band; SHORT maxBand; SCHAR group; SCHAR winGroupLen; SCHAR window; SCHAR numUnitInBand; SCHAR cntUnitInBand; SCHAR groupWin; SCHAR cb_prev; UCHAR *pCodeBook; const SHORT *BandOffsets; SCHAR numOfGroups; pCodeBook = pAacDecoderChannelInfo->pDynData->aCodeBook; /* in */ pNumLinesInSec = pHcr->decInOut.pNumLineInSect; /* out */ pCodeBk = pHcr->decInOut.pCodebook; /* out */ BandOffsets = GetScaleFactorBandOffsets(pIcsInfo, pSamplingRateInfo); /* aux */ numOfGroups = GetWindowGroups(pIcsInfo); numLine = 0; numSection = 0; cb = pCodeBook[0]; cb_prev = pCodeBook[0]; /* convert HCR-sideinfo into a unitwise manner: When the cb changes, a new * section starts */ *pCodeBk++ = cb_prev; maxBand = GetScaleFactorBandsTransmitted(&pAacDecoderChannelInfo->icsInfo); for (band = 0; band < maxBand; band++) { /* from low to high sfbs i.e. from low to high frequencies */ numUnitInBand = ((BandOffsets[band + 1] - BandOffsets[band]) >> FOUR_LOG_DIV_TWO_LOG); /* get the number of units in current sfb */ for (cntUnitInBand = numUnitInBand; cntUnitInBand != 0; cntUnitInBand--) { /* for every unit in the band */ for (window = 0, group = 0; group < numOfGroups; group++) { winGroupLen = (SCHAR)GetWindowGroupLength( &pAacDecoderChannelInfo->icsInfo, group); for (groupWin = winGroupLen; groupWin != 0; groupWin--, window++) { cb = pCodeBook[group * 16 + band]; if (cb != cb_prev) { errDetectorInHcrSideinfoShrt(cb, numLine, &pHcr->decInOut.errorLog); if (pHcr->decInOut.errorLog != 0) { return (pHcr->decInOut.errorLog); } *pCodeBk++ = cb; *pNumLinesInSec++ = numLine; numSection++; cb_prev = cb; numLine = LINES_PER_UNIT; } else { numLine += LINES_PER_UNIT; } } } } } numSection++; errDetectorInHcrSideinfoShrt(cb, numLine, &pHcr->decInOut.errorLog); if (numSection <= 0 || numSection > 1024 / 2) { pHcr->decInOut.errorLog |= NUM_SECT_OUT_OF_RANGE_SHORT_BLOCK; } errDetectorInHcrLengths(pHcr->decInOut.lengthOfLongestCodeword, pHcr->decInOut.lengthOfReorderedSpectralData, &pHcr->decInOut.errorLog); if (pHcr->decInOut.errorLog != 0) { return (pHcr->decInOut.errorLog); } *pCodeBk = cb; *pNumLinesInSec = numLine; pHcr->decInOut.numSection = numSection; } else /* end short block prepare SI */ { /* long block */ errDetectorInHcrLengths(pHcr->decInOut.lengthOfLongestCodeword, pHcr->decInOut.lengthOfReorderedSpectralData, &pHcr->decInOut.errorLog); numSection = pHcr->decInOut.numSection; pNumLinesInSec = pHcr->decInOut.pNumLineInSect; pCodeBk = pHcr->decInOut.pCodebook; if (numSection <= 0 || numSection > 64) { pHcr->decInOut.errorLog |= NUM_SECT_OUT_OF_RANGE_LONG_BLOCK; numSection = 0; } for (i = numSection; i != 0; i--) { cb = *pCodeBk++; if (cb < ZERO_HCB || cb >= MAX_CB_CHECK || cb == BOOKSCL) { pHcr->decInOut.errorLog |= CB_OUT_OF_RANGE_LONG_BLOCK; } numLine = *pNumLinesInSec++; /* FDK_ASSERT(numLine > 0); */ if ((numLine <= 0) || (numLine > 1024)) { pHcr->decInOut.errorLog |= LINE_IN_SECT_OUT_OF_RANGE_LONG_BLOCK; } } if (pHcr->decInOut.errorLog != 0) { return (pHcr->decInOut.errorLog); } } pCodeBk = pHcr->decInOut.pCodebook; for (i = 0; i < numSection; i++) { if ((*pCodeBk == NOISE_HCB) || (*pCodeBk == INTENSITY_HCB2) || (*pCodeBk == INTENSITY_HCB)) { *pCodeBk = 0; } pCodeBk++; } /* HCR-sideinfo-input is complete and seems to be valid */ return (pHcr->decInOut.errorLog); } /*--------------------------------------------------------------------------------------------- description: This function decodes the codewords of the spectral coefficients from the bitstream according to the HCR algorithm and stores the quantized spectral coefficients in correct order in the output buffer. -------------------------------------------------------------------------------------------- */ UINT HcrDecoder(H_HCR_INFO pHcr, CAacDecoderChannelInfo *pAacDecoderChannelInfo, const SamplingRateInfo *pSamplingRateInfo, HANDLE_FDK_BITSTREAM bs) { int pTmp1, pTmp2, pTmp3, pTmp4; int pTmp5; INT bitCntOffst; INT saveBitCnt = (INT)FDKgetValidBits(bs); /* save bitstream position */ HcrCalcNumCodeword(pHcr); HcrSortCodebookAndNumCodewordInSection(pHcr); HcrPrepareSegmentationGrid(pHcr); HcrExtendedSectionInfo(pHcr); if ((pHcr->decInOut.errorLog & HCR_FATAL_PCW_ERROR_MASK) != 0) { return (pHcr->decInOut.errorLog); /* sideinfo is massively corrupt, return from HCR without having decoded anything */ } DeriveNumberOfExtendedSortedSectionsInSets( pHcr->segmentInfo.numSegment, pHcr->sectionInfo.pNumExtendedSortedCodewordInSection, pHcr->sectionInfo.numExtendedSortedCodewordInSectionIdx, pHcr->sectionInfo.pNumExtendedSortedSectionsInSets, pHcr->sectionInfo.numExtendedSortedSectionsInSetsIdx); /* store */ pTmp1 = pHcr->sectionInfo.numExtendedSortedCodewordInSectionIdx; pTmp2 = pHcr->sectionInfo.extendedSortedCodebookIdx; pTmp3 = pHcr->sectionInfo.numExtendedSortedSectionsInSetsIdx; pTmp4 = pHcr->decInOut.quantizedSpectralCoefficientsIdx; pTmp5 = pHcr->sectionInfo.maxLenOfCbInExtSrtSecIdx; /* ------- decode meaningful PCWs ------ */ DecodePCWs(bs, pHcr); if ((pHcr->decInOut.errorLog & HCR_FATAL_PCW_ERROR_MASK) == 0) { /* ------ decode the non-PCWs -------- */ DecodeNonPCWs(bs, pHcr); } errDetectWithinSegmentationFinal(pHcr); /* restore */ pHcr->sectionInfo.numExtendedSortedCodewordInSectionIdx = pTmp1; pHcr->sectionInfo.extendedSortedCodebookIdx = pTmp2; pHcr->sectionInfo.numExtendedSortedSectionsInSetsIdx = pTmp3; pHcr->decInOut.quantizedSpectralCoefficientsIdx = pTmp4; pHcr->sectionInfo.maxLenOfCbInExtSrtSecIdx = pTmp5; HcrReorderQuantizedSpectralCoefficients(pHcr, pAacDecoderChannelInfo, pSamplingRateInfo); /* restore bitstream position */ bitCntOffst = (INT)FDKgetValidBits(bs) - saveBitCnt; if (bitCntOffst) { FDKpushBiDirectional(bs, bitCntOffst); } return (pHcr->decInOut.errorLog); } /*--------------------------------------------------------------------------------------------- description: This function reorders the quantized spectral coefficients sectionwise for long- and short-blocks and compares to the LAV (Largest Absolute Value of the current codebook) -- a counter is incremented if there is an error detected. Additional for short-blocks a unit-based-deinterleaving is applied. Moreover (for short blocks) the scaling is derived (compare plain huffman decoder). -------------------------------------------------------------------------------------------- */ static void HcrReorderQuantizedSpectralCoefficients( H_HCR_INFO pHcr, CAacDecoderChannelInfo *pAacDecoderChannelInfo, const SamplingRateInfo *pSamplingRateInfo) { INT qsc; UINT abs_qsc; UINT i, j; USHORT numSpectralValuesInSection; FIXP_DBL *pTeVa; USHORT lavErrorCnt = 0; UINT numSection = pHcr->decInOut.numSection; SPECTRAL_PTR pQuantizedSpectralCoefficientsBase = pHcr->decInOut.pQuantizedSpectralCoefficientsBase; FIXP_DBL *pQuantizedSpectralCoefficients = SPEC_LONG(pHcr->decInOut.pQuantizedSpectralCoefficientsBase); const UCHAR *pCbDimShift = aDimCbShift; const USHORT *pLargestAbsVal = aLargestAbsoluteValue; UCHAR *pSortedCodebook = pHcr->sectionInfo.pSortedCodebook; USHORT *pNumSortedCodewordInSection = pHcr->sectionInfo.pNumSortedCodewordInSection; USHORT *pReorderOffset = pHcr->sectionInfo.pReorderOffset; FIXP_DBL pTempValues[1024]; FIXP_DBL *pBak = pTempValues; FDKmemclear(pTempValues, 1024 * sizeof(FIXP_DBL)); /* long and short: check if decoded huffman-values (quantized spectral * coefficients) are within range */ for (i = numSection; i != 0; i--) { numSpectralValuesInSection = *pNumSortedCodewordInSection++ << pCbDimShift[*pSortedCodebook]; pTeVa = &pTempValues[*pReorderOffset++]; for (j = numSpectralValuesInSection; j != 0; j--) { qsc = *pQuantizedSpectralCoefficients++; abs_qsc = fAbs(qsc); if (abs_qsc <= pLargestAbsVal[*pSortedCodebook]) { *pTeVa++ = (FIXP_DBL)qsc; /* the qsc value is within range */ } else { /* line is too high .. */ if (abs_qsc == Q_VALUE_INVALID) { /* .. because of previous marking --> dont set LAV flag (would be confusing), just copy out the already marked value */ *pTeVa++ = (FIXP_DBL)qsc; } else { /* .. because a too high value was decoded for this cb --> set LAV flag */ *pTeVa++ = (FIXP_DBL)Q_VALUE_INVALID; lavErrorCnt += 1; } } } pSortedCodebook++; } if (!IsLongBlock(&pAacDecoderChannelInfo->icsInfo)) { FIXP_DBL *pOut; FIXP_DBL locMax; FIXP_DBL tmp; SCHAR groupoffset; SCHAR group; SCHAR band; SCHAR groupwin; SCHAR window; SCHAR numWinGroup; SHORT interm; SCHAR numSfbTransm; SCHAR winGroupLen; SHORT index; INT msb; INT lsb; SHORT *pScaleFacHcr = pAacDecoderChannelInfo->pDynData->aScaleFactor; SHORT *pSfbSclHcr = pAacDecoderChannelInfo->pDynData->aSfbScale; const SHORT *BandOffsets = GetScaleFactorBandOffsets( &pAacDecoderChannelInfo->icsInfo, pSamplingRateInfo); pBak = pTempValues; /* deinterleave unitwise for short blocks */ for (window = 0; window < (8); window++) { pOut = SPEC(pQuantizedSpectralCoefficientsBase, window, pAacDecoderChannelInfo->granuleLength); for (i = 0; i < (LINES_PER_UNIT_GROUP); i++) { pTeVa = pBak + (window << FOUR_LOG_DIV_TWO_LOG) + i * 32; /* distance of lines between unit groups has to be constant for every framelength (32)! */ for (j = (LINES_PER_UNIT); j != 0; j--) { *pOut++ = *pTeVa++; } } } /* short blocks only */ /* derive global scaling-value for every sfb and every window (as it is done * in plain-huffman-decoder at short blocks) */ groupoffset = 0; numWinGroup = GetWindowGroups(&pAacDecoderChannelInfo->icsInfo); numSfbTransm = GetScaleFactorBandsTransmitted(&pAacDecoderChannelInfo->icsInfo); for (group = 0; group < numWinGroup; group++) { winGroupLen = GetWindowGroupLength(&pAacDecoderChannelInfo->icsInfo, group); for (band = 0; band < numSfbTransm; band++) { interm = group * 16 + band; msb = pScaleFacHcr[interm] >> 2; lsb = pScaleFacHcr[interm] & 3; for (groupwin = 0; groupwin < winGroupLen; groupwin++) { window = groupoffset + groupwin; pBak = SPEC(pQuantizedSpectralCoefficientsBase, window, pAacDecoderChannelInfo->granuleLength); locMax = FL2FXCONST_DBL(0.0f); for (index = BandOffsets[band]; index < BandOffsets[band + 1]; index += LINES_PER_UNIT) { pTeVa = &pBak[index]; for (i = LINES_PER_UNIT; i != 0; i--) { tmp = (*pTeVa < FL2FXCONST_DBL(0.0f)) ? -*pTeVa++ : *pTeVa++; locMax = fixMax(tmp, locMax); } } if (fixp_abs(locMax) > (FIXP_DBL)MAX_QUANTIZED_VALUE) { locMax = (FIXP_DBL)MAX_QUANTIZED_VALUE; } pSfbSclHcr[window * 16 + band] = msb - GetScaleFromValue( locMax, lsb); /* save global scale maxima in this sfb */ } } groupoffset += GetWindowGroupLength(&pAacDecoderChannelInfo->icsInfo, group); } } else { /* copy straight for long-blocks */ pQuantizedSpectralCoefficients = SPEC_LONG(pQuantizedSpectralCoefficientsBase); for (i = 1024; i != 0; i--) { *pQuantizedSpectralCoefficients++ = *pBak++; } } if (lavErrorCnt != 0) { pHcr->decInOut.errorLog |= LAV_VIOLATION; } } /*--------------------------------------------------------------------------------------------- description: This function calculates the number of codewords for each section (numCodewordInSection) and the number of codewords for all sections (numCodeword). For zero and intensity codebooks a entry is also done in the variable numCodewordInSection. It is assumed that the codebook is a two tuples codebook. This is needed later for the calculation of the base addresses for the reordering of the quantize spectral coefficients at the end of the hcr tool. The variable numCodeword contain the number of codewords which are really in the bitstream. Zero or intensity codebooks does not increase the variable numCodewords. ----------------------------------------------------------------------------------------------- return: - -------------------------------------------------------------------------------------------- */ static void HcrCalcNumCodeword(H_HCR_INFO pHcr) { int hcrSection; UINT numCodeword; UINT numSection = pHcr->decInOut.numSection; UCHAR *pCodebook = pHcr->decInOut.pCodebook; SHORT *pNumLineInSection = pHcr->decInOut.pNumLineInSect; const UCHAR *pCbDimShift = aDimCbShift; USHORT *pNumCodewordInSection = pHcr->sectionInfo.pNumCodewordInSection; numCodeword = 0; for (hcrSection = numSection; hcrSection != 0; hcrSection--) { *pNumCodewordInSection = *pNumLineInSection++ >> pCbDimShift[*pCodebook]; if (*pCodebook != 0) { numCodeword += *pNumCodewordInSection; } pNumCodewordInSection++; pCodebook++; } pHcr->sectionInfo.numCodeword = numCodeword; } /*--------------------------------------------------------------------------------------------- description: This function calculates the number of sorted codebooks and sorts the codebooks and the numCodewordInSection according to the priority. -------------------------------------------------------------------------------------------- */ static void HcrSortCodebookAndNumCodewordInSection(H_HCR_INFO pHcr) { UINT i, j, k; UCHAR temp; UINT counter; UINT startOffset; UINT numZeroSection; UCHAR *pDest; UINT numSectionDec; UINT numSection = pHcr->decInOut.numSection; UCHAR *pCodebook = pHcr->decInOut.pCodebook; UCHAR *pSortedCodebook = pHcr->sectionInfo.pSortedCodebook; USHORT *pNumCodewordInSection = pHcr->sectionInfo.pNumCodewordInSection; USHORT *pNumSortedCodewordInSection = pHcr->sectionInfo.pNumSortedCodewordInSection; UCHAR *pCodebookSwitch = pHcr->sectionInfo.pCodebookSwitch; USHORT *pReorderOffset = pHcr->sectionInfo.pReorderOffset; const UCHAR *pCbPriority = aCbPriority; const UCHAR *pMinOfCbPair = aMinOfCbPair; const UCHAR *pMaxOfCbPair = aMaxOfCbPair; const UCHAR *pCbDimShift = aDimCbShift; UINT searchStart = 0; /* calculate *pNumSortedSection and store the priorities in array * pSortedCdebook */ pDest = pSortedCodebook; numZeroSection = 0; for (i = numSection; i != 0; i--) { if (pCbPriority[*pCodebook] == 0) { numZeroSection += 1; } *pDest++ = pCbPriority[*pCodebook++]; } pHcr->sectionInfo.numSortedSection = numSection - numZeroSection; /* numSortedSection contains no zero or intensity section */ pCodebook = pHcr->decInOut.pCodebook; /* sort priorities of the codebooks in array pSortedCdebook[] */ numSectionDec = numSection - 1; if (numSectionDec > 0) { counter = numSectionDec; for (j = numSectionDec; j != 0; j--) { for (i = 0; i < counter; i++) { /* swap priorities */ if (pSortedCodebook[i + 1] > pSortedCodebook[i]) { temp = pSortedCodebook[i]; pSortedCodebook[i] = pSortedCodebook[i + 1]; pSortedCodebook[i + 1] = temp; } } counter -= 1; } } /* clear codebookSwitch array */ for (i = numSection; i != 0; i--) { *pCodebookSwitch++ = 0; } pCodebookSwitch = pHcr->sectionInfo.pCodebookSwitch; /* sort sectionCodebooks and numCodwordsInSection and calculate * pReorderOffst[j] */ for (j = 0; j < numSection; j++) { for (i = searchStart; i < numSection; i++) { if (pCodebookSwitch[i] == 0 && (pMinOfCbPair[pSortedCodebook[j]] == pCodebook[i] || pMaxOfCbPair[pSortedCodebook[j]] == pCodebook[i])) { pCodebookSwitch[i] = 1; pSortedCodebook[j] = pCodebook[i]; /* sort codebook */ pNumSortedCodewordInSection[j] = pNumCodewordInSection[i]; /* sort NumCodewordInSection */ startOffset = 0; for (k = 0; k < i; k++) { /* make entry in pReorderOffst */ startOffset += pNumCodewordInSection[k] << pCbDimShift[pCodebook[k]]; } pReorderOffset[j] = startOffset; /* offset for reordering the codewords */ if (i == searchStart) { k = i; while (pCodebookSwitch[k++] == 1) searchStart++; } break; } } } } /*--------------------------------------------------------------------------------------------- description: This function calculates the segmentation, which includes numSegment, leftStartOfSegment, rightStartOfSegment and remainingBitsInSegment. The segmentation could be visualized a as kind of 'overlay-grid' for the bitstream-block holding the HCR-encoded quantized-spectral-coefficients. -------------------------------------------------------------------------------------------- */ static void HcrPrepareSegmentationGrid(H_HCR_INFO pHcr) { USHORT i, j; USHORT numSegment = 0; INT segmentStart = 0; UCHAR segmentWidth; UCHAR lastSegmentWidth; UCHAR sortedCodebook; UCHAR endFlag = 0; INT intermediateResult; SCHAR lengthOfLongestCodeword = pHcr->decInOut.lengthOfLongestCodeword; SHORT lengthOfReorderedSpectralData = pHcr->decInOut.lengthOfReorderedSpectralData; UINT numSortedSection = pHcr->sectionInfo.numSortedSection; UCHAR *pSortedCodebook = pHcr->sectionInfo.pSortedCodebook; USHORT *pNumSortedCodewordInSection = pHcr->sectionInfo.pNumSortedCodewordInSection; INT *pLeftStartOfSegment = pHcr->segmentInfo.pLeftStartOfSegment; INT *pRightStartOfSegment = pHcr->segmentInfo.pRightStartOfSegment; SCHAR *pRemainingBitsInSegment = pHcr->segmentInfo.pRemainingBitsInSegment; const UCHAR *pMaxCwLength = aMaxCwLen; for (i = numSortedSection; i != 0; i--) { sortedCodebook = *pSortedCodebook++; segmentWidth = fMin((INT)pMaxCwLength[sortedCodebook], (INT)lengthOfLongestCodeword); for (j = *pNumSortedCodewordInSection; j != 0; j--) { /* width allows a new segment */ intermediateResult = segmentStart; if ((segmentStart + segmentWidth) <= lengthOfReorderedSpectralData) { /* store segment start, segment length and increment the number of * segments */ *pLeftStartOfSegment++ = intermediateResult; *pRightStartOfSegment++ = intermediateResult + segmentWidth - 1; *pRemainingBitsInSegment++ = segmentWidth; segmentStart += segmentWidth; numSegment += 1; } /* width does not allow a new segment */ else { /* correct the last segment length */ pLeftStartOfSegment--; pRightStartOfSegment--; pRemainingBitsInSegment--; segmentStart = *pLeftStartOfSegment; lastSegmentWidth = lengthOfReorderedSpectralData - segmentStart; *pRemainingBitsInSegment = lastSegmentWidth; *pRightStartOfSegment = segmentStart + lastSegmentWidth - 1; endFlag = 1; break; } } pNumSortedCodewordInSection++; if (endFlag != 0) { break; } } pHcr->segmentInfo.numSegment = numSegment; } /*--------------------------------------------------------------------------------------------- description: This function adapts the sorted section boundaries to the boundaries of segmentation. If the section lengths does not fit completely into the current segment, the section is spitted into two so called 'extended sections'. The extended-section-info (pNumExtendedSortedCodewordInSectin and pExtendedSortedCodebook) is updated in this case. -------------------------------------------------------------------------------------------- */ static void HcrExtendedSectionInfo(H_HCR_INFO pHcr) { UINT srtSecCnt = 0; /* counter for sorted sections */ UINT xSrtScCnt = 0; /* counter for extended sorted sections */ UINT remainNumCwInSortSec; UINT inSegmentRemainNumCW; UINT numSortedSection = pHcr->sectionInfo.numSortedSection; UCHAR *pSortedCodebook = pHcr->sectionInfo.pSortedCodebook; USHORT *pNumSortedCodewordInSection = pHcr->sectionInfo.pNumSortedCodewordInSection; UCHAR *pExtendedSortedCoBo = pHcr->sectionInfo.pExtendedSortedCodebook; USHORT *pNumExtSortCwInSect = pHcr->sectionInfo.pNumExtendedSortedCodewordInSection; UINT numSegment = pHcr->segmentInfo.numSegment; UCHAR *pMaxLenOfCbInExtSrtSec = pHcr->sectionInfo.pMaxLenOfCbInExtSrtSec; SCHAR lengthOfLongestCodeword = pHcr->decInOut.lengthOfLongestCodeword; const UCHAR *pMaxCwLength = aMaxCwLen; remainNumCwInSortSec = pNumSortedCodewordInSection[srtSecCnt]; inSegmentRemainNumCW = numSegment; while (srtSecCnt < numSortedSection) { if (inSegmentRemainNumCW < remainNumCwInSortSec) { pNumExtSortCwInSect[xSrtScCnt] = inSegmentRemainNumCW; pExtendedSortedCoBo[xSrtScCnt] = pSortedCodebook[srtSecCnt]; remainNumCwInSortSec -= inSegmentRemainNumCW; inSegmentRemainNumCW = numSegment; /* data of a sorted section was not integrated in extended sorted section */ } else if (inSegmentRemainNumCW == remainNumCwInSortSec) { pNumExtSortCwInSect[xSrtScCnt] = inSegmentRemainNumCW; pExtendedSortedCoBo[xSrtScCnt] = pSortedCodebook[srtSecCnt]; srtSecCnt++; remainNumCwInSortSec = pNumSortedCodewordInSection[srtSecCnt]; inSegmentRemainNumCW = numSegment; /* data of a sorted section was integrated in extended sorted section */ } else { /* inSegmentRemainNumCW > remainNumCwInSortSec */ pNumExtSortCwInSect[xSrtScCnt] = remainNumCwInSortSec; pExtendedSortedCoBo[xSrtScCnt] = pSortedCodebook[srtSecCnt]; inSegmentRemainNumCW -= remainNumCwInSortSec; srtSecCnt++; remainNumCwInSortSec = pNumSortedCodewordInSection[srtSecCnt]; /* data of a sorted section was integrated in extended sorted section */ } pMaxLenOfCbInExtSrtSec[xSrtScCnt] = fMin((INT)pMaxCwLength[pExtendedSortedCoBo[xSrtScCnt]], (INT)lengthOfLongestCodeword); xSrtScCnt += 1; if (xSrtScCnt >= (MAX_SFB_HCR + MAX_HCR_SETS)) { pHcr->decInOut.errorLog |= EXTENDED_SORTED_COUNTER_OVERFLOW; return; } } pNumExtSortCwInSect[xSrtScCnt] = 0; } /*--------------------------------------------------------------------------------------------- description: This function calculates the number of extended sorted sections which belong to the sets. Each set from set 0 (one and only set for the PCWs) till to the last set gets a entry in the array to which 'pNumExtendedSortedSectinsInSets' points to. Calculation: The entrys in pNumExtendedSortedCodewordInSectin are added untill the value numSegment is reached. Then the sum_variable is cleared and the calculation starts from the beginning. As much extended sorted Sections are summed up to reach the value numSegment, as much is the current entry in *pNumExtendedSortedCodewordInSectin. -------------------------------------------------------------------------------------------- */ static void DeriveNumberOfExtendedSortedSectionsInSets( UINT numSegment, USHORT *pNumExtendedSortedCodewordInSection, int numExtendedSortedCodewordInSectionIdx, USHORT *pNumExtendedSortedSectionsInSets, int numExtendedSortedSectionsInSetsIdx) { USHORT counter = 0; UINT cwSum = 0; USHORT *pNumExSortCwInSec = pNumExtendedSortedCodewordInSection; USHORT *pNumExSortSecInSets = pNumExtendedSortedSectionsInSets; while (pNumExSortCwInSec[numExtendedSortedCodewordInSectionIdx] != 0) { cwSum += pNumExSortCwInSec[numExtendedSortedCodewordInSectionIdx]; numExtendedSortedCodewordInSectionIdx++; if (numExtendedSortedCodewordInSectionIdx >= (MAX_SFB_HCR + MAX_HCR_SETS)) { return; } if (cwSum > numSegment) { return; } counter++; if (counter > 1024 / 4) { return; } if (cwSum == numSegment) { pNumExSortSecInSets[numExtendedSortedSectionsInSetsIdx] = counter; numExtendedSortedSectionsInSetsIdx++; if (numExtendedSortedSectionsInSetsIdx >= MAX_HCR_SETS) { return; } counter = 0; cwSum = 0; } } pNumExSortSecInSets[numExtendedSortedSectionsInSetsIdx] = counter; /* save last entry for the last - probably shorter - set */ } /*--------------------------------------------------------------------------------------------- description: This function decodes all priority codewords (PCWs) in a spectrum (within set 0). The calculation of the PCWs is managed in two loops. The loopcounter of the outer loop is set to the first value pointer pNumExtendedSortedSectionsInSets points to. This value represents the number of extended sorted sections within set 0. The loopcounter of the inner loop is set to the first value pointer pNumExtendedSortedCodewordInSectin points to. The value represents the number of extended sorted codewords in sections (the original sections have been splitted to go along with the borders of the sets). Each time the number of the extended sorted codewords in sections are de- coded, the pointer 'pNumExtendedSortedCodewordInSectin' is incremented by one. -------------------------------------------------------------------------------------------- */ static void DecodePCWs(HANDLE_FDK_BITSTREAM bs, H_HCR_INFO pHcr) { UINT i; USHORT extSortSec; USHORT curExtSortCwInSec; UCHAR codebook; UCHAR dimension; const UINT *pCurrentTree; const SCHAR *pQuantValBase; const SCHAR *pQuantVal; USHORT *pNumExtendedSortedCodewordInSection = pHcr->sectionInfo.pNumExtendedSortedCodewordInSection; int numExtendedSortedCodewordInSectionIdx = pHcr->sectionInfo.numExtendedSortedCodewordInSectionIdx; UCHAR *pExtendedSortedCodebook = pHcr->sectionInfo.pExtendedSortedCodebook; int extendedSortedCodebookIdx = pHcr->sectionInfo.extendedSortedCodebookIdx; USHORT *pNumExtendedSortedSectionsInSets = pHcr->sectionInfo.pNumExtendedSortedSectionsInSets; int numExtendedSortedSectionsInSetsIdx = pHcr->sectionInfo.numExtendedSortedSectionsInSetsIdx; FIXP_DBL *pQuantizedSpectralCoefficients = SPEC_LONG(pHcr->decInOut.pQuantizedSpectralCoefficientsBase); int quantizedSpectralCoefficientsIdx = pHcr->decInOut.quantizedSpectralCoefficientsIdx; INT *pLeftStartOfSegment = pHcr->segmentInfo.pLeftStartOfSegment; SCHAR *pRemainingBitsInSegment = pHcr->segmentInfo.pRemainingBitsInSegment; UCHAR *pMaxLenOfCbInExtSrtSec = pHcr->sectionInfo.pMaxLenOfCbInExtSrtSec; int maxLenOfCbInExtSrtSecIdx = pHcr->sectionInfo.maxLenOfCbInExtSrtSecIdx; UCHAR maxAllowedCwLen; int numDecodedBits; const UCHAR *pCbDimension = aDimCb; const UCHAR *pCbSign = aSignCb; /* clear result array */ FDKmemclear(pQuantizedSpectralCoefficients + quantizedSpectralCoefficientsIdx, 1024 * sizeof(FIXP_DBL)); /* decode all PCWs in the extended sorted section(s) belonging to set 0 */ for (extSortSec = pNumExtendedSortedSectionsInSets[numExtendedSortedSectionsInSetsIdx]; extSortSec != 0; extSortSec--) { codebook = pExtendedSortedCodebook[extendedSortedCodebookIdx]; /* get codebook for this extended sorted section and increment ptr to cb of next ext. sort sec */ extendedSortedCodebookIdx++; if (extendedSortedCodebookIdx >= (MAX_SFB_HCR + MAX_HCR_SETS)) { return; } dimension = pCbDimension[codebook]; /* get dimension of codebook of this extended sort. sec. */ pCurrentTree = aHuffTable[codebook]; /* convert codebook to pointer to QSCs */ pQuantValBase = aQuantTable[codebook]; /* convert codebook to index to table of QSCs */ maxAllowedCwLen = pMaxLenOfCbInExtSrtSec[maxLenOfCbInExtSrtSecIdx]; maxLenOfCbInExtSrtSecIdx++; if (maxLenOfCbInExtSrtSecIdx >= (MAX_SFB_HCR + MAX_HCR_SETS)) { return; } /* switch for decoding with different codebooks: */ if (pCbSign[codebook] == 0) { /* no sign bits follow after the codeword-body */ /* PCW_BodyONLY */ /*==============*/ for (curExtSortCwInSec = pNumExtendedSortedCodewordInSection [numExtendedSortedCodewordInSectionIdx]; curExtSortCwInSec != 0; curExtSortCwInSec--) { numDecodedBits = 0; /* decode PCW_BODY */ pQuantVal = DecodePCW_Body( bs, pHcr->decInOut.bitstreamAnchor, pCurrentTree, pQuantValBase, pLeftStartOfSegment, pRemainingBitsInSegment, &numDecodedBits); /* result is written out here because NO sign bits follow the body */ for (i = dimension; i != 0; i--) { pQuantizedSpectralCoefficients[quantizedSpectralCoefficientsIdx] = (FIXP_DBL)*pQuantVal++; /* write quant. spec. coef. into spectrum; sign is already valid */ quantizedSpectralCoefficientsIdx++; if (quantizedSpectralCoefficientsIdx >= 1024) { return; } } /* one more PCW should be decoded */ if (maxAllowedCwLen < (numDecodedBits + ERROR_PCW_BODY_ONLY_TOO_LONG)) { pHcr->decInOut.errorLog |= TOO_MANY_PCW_BODY_BITS_DECODED; } if (1 == errDetectPcwSegmentation( *pRemainingBitsInSegment - ERROR_PCW_BODY, pHcr, PCW_BODY, pQuantizedSpectralCoefficients + quantizedSpectralCoefficientsIdx - dimension, dimension)) { return; } pLeftStartOfSegment++; /* update pointer for decoding the next PCW */ pRemainingBitsInSegment++; /* update pointer for decoding the next PCW */ } } else if ((codebook < 11) && (pCbSign[codebook] == 1)) { /* possibly there follow 1,2,3 or 4 sign bits after the codeword-body */ /* PCW_Body and PCW_Sign */ /*=======================*/ for (curExtSortCwInSec = pNumExtendedSortedCodewordInSection [numExtendedSortedCodewordInSectionIdx]; curExtSortCwInSec != 0; curExtSortCwInSec--) { int err; numDecodedBits = 0; pQuantVal = DecodePCW_Body( bs, pHcr->decInOut.bitstreamAnchor, pCurrentTree, pQuantValBase, pLeftStartOfSegment, pRemainingBitsInSegment, &numDecodedBits); err = DecodePCW_Sign( bs, pHcr->decInOut.bitstreamAnchor, dimension, pQuantVal, pQuantizedSpectralCoefficients, &quantizedSpectralCoefficientsIdx, pLeftStartOfSegment, pRemainingBitsInSegment, &numDecodedBits); if (err != 0) { return; } /* one more PCW should be decoded */ if (maxAllowedCwLen < (numDecodedBits + ERROR_PCW_BODY_SIGN_TOO_LONG)) { pHcr->decInOut.errorLog |= TOO_MANY_PCW_BODY_SIGN_BITS_DECODED; } if (1 == errDetectPcwSegmentation( *pRemainingBitsInSegment - ERROR_PCW_BODY_SIGN, pHcr, PCW_BODY_SIGN, pQuantizedSpectralCoefficients + quantizedSpectralCoefficientsIdx - dimension, dimension)) { return; } pLeftStartOfSegment++; pRemainingBitsInSegment++; } } else if ((pCbSign[codebook] == 1) && (codebook >= 11)) { /* possibly there follow some sign bits and maybe one or two escape sequences after the cw-body */ /* PCW_Body, PCW_Sign and maybe PCW_Escape */ /*=========================================*/ for (curExtSortCwInSec = pNumExtendedSortedCodewordInSection [numExtendedSortedCodewordInSectionIdx]; curExtSortCwInSec != 0; curExtSortCwInSec--) { int err; numDecodedBits = 0; /* decode PCW_BODY */ pQuantVal = DecodePCW_Body( bs, pHcr->decInOut.bitstreamAnchor, pCurrentTree, pQuantValBase, pLeftStartOfSegment, pRemainingBitsInSegment, &numDecodedBits); err = DecodePCW_Sign( bs, pHcr->decInOut.bitstreamAnchor, dimension, pQuantVal, pQuantizedSpectralCoefficients, &quantizedSpectralCoefficientsIdx, pLeftStartOfSegment, pRemainingBitsInSegment, &numDecodedBits); if (err != 0) { return; } /* decode PCW_ESCAPE if present */ quantizedSpectralCoefficientsIdx -= DIMENSION_OF_ESCAPE_CODEBOOK; if (fixp_abs(pQuantizedSpectralCoefficients [quantizedSpectralCoefficientsIdx]) == (FIXP_DBL)ESCAPE_VALUE) { pQuantizedSpectralCoefficients[quantizedSpectralCoefficientsIdx] = (FIXP_DBL)DecodeEscapeSequence( bs, pHcr->decInOut.bitstreamAnchor, pQuantizedSpectralCoefficients [quantizedSpectralCoefficientsIdx], pLeftStartOfSegment, pRemainingBitsInSegment, &numDecodedBits, &pHcr->decInOut.errorLog); } quantizedSpectralCoefficientsIdx++; if (quantizedSpectralCoefficientsIdx >= 1024) { return; } if (fixp_abs(pQuantizedSpectralCoefficients [quantizedSpectralCoefficientsIdx]) == (FIXP_DBL)ESCAPE_VALUE) { pQuantizedSpectralCoefficients[quantizedSpectralCoefficientsIdx] = (FIXP_DBL)DecodeEscapeSequence( bs, pHcr->decInOut.bitstreamAnchor, pQuantizedSpectralCoefficients [quantizedSpectralCoefficientsIdx], pLeftStartOfSegment, pRemainingBitsInSegment, &numDecodedBits, &pHcr->decInOut.errorLog); } quantizedSpectralCoefficientsIdx++; if (quantizedSpectralCoefficientsIdx >= 1024) { return; } /* one more PCW should be decoded */ if (maxAllowedCwLen < (numDecodedBits + ERROR_PCW_BODY_SIGN_ESC_TOO_LONG)) { pHcr->decInOut.errorLog |= TOO_MANY_PCW_BODY_SIGN_ESC_BITS_DECODED; } if (1 == errDetectPcwSegmentation( *pRemainingBitsInSegment - ERROR_PCW_BODY_SIGN_ESC, pHcr, PCW_BODY_SIGN_ESC, pQuantizedSpectralCoefficients + quantizedSpectralCoefficientsIdx - DIMENSION_OF_ESCAPE_CODEBOOK, DIMENSION_OF_ESCAPE_CODEBOOK)) { return; } pLeftStartOfSegment++; pRemainingBitsInSegment++; } } /* all PCWs belonging to this extended section should be decoded */ numExtendedSortedCodewordInSectionIdx++; if (numExtendedSortedCodewordInSectionIdx >= MAX_SFB_HCR + MAX_HCR_SETS) { return; } } /* all PCWs should be decoded */ numExtendedSortedSectionsInSetsIdx++; if (numExtendedSortedSectionsInSetsIdx >= MAX_HCR_SETS) { return; } /* Write back indexes into structure */ pHcr->sectionInfo.numExtendedSortedCodewordInSectionIdx = numExtendedSortedCodewordInSectionIdx; pHcr->sectionInfo.extendedSortedCodebookIdx = extendedSortedCodebookIdx; pHcr->sectionInfo.numExtendedSortedSectionsInSetsIdx = numExtendedSortedSectionsInSetsIdx; pHcr->decInOut.quantizedSpectralCoefficientsIdx = quantizedSpectralCoefficientsIdx; pHcr->sectionInfo.maxLenOfCbInExtSrtSecIdx = maxLenOfCbInExtSrtSecIdx; } /*--------------------------------------------------------------------------------------------- description: This function checks immediately after every decoded PCW, whether out of the current segment too many bits have been read or not. If an error occurrs, probably the sideinfo or the HCR-bitstream block holding the huffman encoded quantized spectral coefficients is distorted. In this case the two or four quantized spectral coefficients belonging to the current codeword are marked (for being detected by concealment later). -------------------------------------------------------------------------------------------- */ static UCHAR errDetectPcwSegmentation(SCHAR remainingBitsInSegment, H_HCR_INFO pHcr, PCW_TYPE kind, FIXP_DBL *qsc_base_of_cw, UCHAR dimension) { SCHAR i; if (remainingBitsInSegment < 0) { /* log the error */ switch (kind) { case PCW_BODY: pHcr->decInOut.errorLog |= SEGMENT_OVERRIDE_ERR_PCW_BODY; break; case PCW_BODY_SIGN: pHcr->decInOut.errorLog |= SEGMENT_OVERRIDE_ERR_PCW_BODY_SIGN; break; case PCW_BODY_SIGN_ESC: pHcr->decInOut.errorLog |= SEGMENT_OVERRIDE_ERR_PCW_BODY_SIGN_ESC; break; } /* mark the erred lines */ for (i = dimension; i != 0; i--) { *qsc_base_of_cw++ = (FIXP_DBL)Q_VALUE_INVALID; } return 1; } return 0; } /*--------------------------------------------------------------------------------------------- description: This function checks if all segments are empty after decoding. There are _no lines markded_ as invalid because it could not be traced back where from the remaining bits are. -------------------------------------------------------------------------------------------- */ static void errDetectWithinSegmentationFinal(H_HCR_INFO pHcr) { UCHAR segmentationErrorFlag = 0; USHORT i; SCHAR *pRemainingBitsInSegment = pHcr->segmentInfo.pRemainingBitsInSegment; UINT numSegment = pHcr->segmentInfo.numSegment; for (i = numSegment; i != 0; i--) { if (*pRemainingBitsInSegment++ != 0) { segmentationErrorFlag = 1; } } if (segmentationErrorFlag == 1) { pHcr->decInOut.errorLog |= BIT_IN_SEGMENTATION_ERROR; } } /*--------------------------------------------------------------------------------------------- description: This function walks one step within the decoding tree. Which branch is taken depends on the decoded carryBit input parameter. -------------------------------------------------------------------------------------------- */ void CarryBitToBranchValue(UCHAR carryBit, UINT treeNode, UINT *branchValue, UINT *branchNode) { if (carryBit == 0) { *branchNode = (treeNode & MASK_LEFT) >> LEFT_OFFSET; /* MASK_LEFT: 00FFF000 */ } else { *branchNode = treeNode & MASK_RIGHT; /* MASK_RIGHT: 00000FFF */ } *branchValue = *branchNode & CLR_BIT_10; /* clear bit 10 (if set) */ } /*--------------------------------------------------------------------------------------------- description: Decodes the body of a priority codeword (PCW) ----------------------------------------------------------------------------------------------- return: - return value is pointer to first of two or four quantized spectral coefficients -------------------------------------------------------------------------------------------- */ static const SCHAR *DecodePCW_Body(HANDLE_FDK_BITSTREAM bs, const INT bsAnchor, const UINT *pCurrentTree, const SCHAR *pQuantValBase, INT *pLeftStartOfSegment, SCHAR *pRemainingBitsInSegment, int *pNumDecodedBits) { UCHAR carryBit; UINT branchNode; UINT treeNode; UINT branchValue; const SCHAR *pQuantVal; /* decode PCW_BODY */ treeNode = *pCurrentTree; /* get first node of current tree belonging to current codebook */ /* decode whole PCW-codeword-body */ while (1) { carryBit = HcrGetABitFromBitstream(bs, bsAnchor, pLeftStartOfSegment, pLeftStartOfSegment, /* dummy */ FROM_LEFT_TO_RIGHT); *pRemainingBitsInSegment -= 1; *pNumDecodedBits += 1; CarryBitToBranchValue(carryBit, treeNode, &branchValue, &branchNode); if ((branchNode & TEST_BIT_10) == TEST_BIT_10) { /* test bit 10 ; if set --> codeword-body is complete */ break; /* end of branch in tree reached i.e. a whole PCW-Body is decoded */ } else { treeNode = *( pCurrentTree + branchValue); /* update treeNode for further step in decoding tree */ } } pQuantVal = pQuantValBase + branchValue; /* update pointer to valid first of 2 or 4 quantized values */ return pQuantVal; } /*--------------------------------------------------------------------------------------------- description: This function decodes one escape sequence. In case of a escape codebook and in case of the absolute value of the quantized spectral value == 16, a escapeSequence is decoded in two steps: 1. escape prefix 2. escape word -------------------------------------------------------------------------------------------- */ static INT DecodeEscapeSequence(HANDLE_FDK_BITSTREAM bs, const INT bsAnchor, INT quantSpecCoef, INT *pLeftStartOfSegment, SCHAR *pRemainingBitsInSegment, int *pNumDecodedBits, UINT *errorWord) { UINT i; INT sign; UINT escapeOnesCounter = 0; UINT carryBit; INT escape_word = 0; /* decode escape prefix */ while (1) { carryBit = HcrGetABitFromBitstream(bs, bsAnchor, pLeftStartOfSegment, pLeftStartOfSegment, /* dummy */ FROM_LEFT_TO_RIGHT); *pRemainingBitsInSegment -= 1; *pNumDecodedBits += 1; if (*pRemainingBitsInSegment < 0) { return Q_VALUE_INVALID; } if (carryBit != 0) { escapeOnesCounter += 1; } else { escapeOnesCounter += 4; break; } } /* decode escape word */ for (i = escapeOnesCounter; i != 0; i--) { carryBit = HcrGetABitFromBitstream(bs, bsAnchor, pLeftStartOfSegment, pLeftStartOfSegment, /* dummy */ FROM_LEFT_TO_RIGHT); *pRemainingBitsInSegment -= 1; *pNumDecodedBits += 1; if (*pRemainingBitsInSegment < 0) { return Q_VALUE_INVALID; } escape_word <<= 1; escape_word = escape_word | carryBit; } sign = (quantSpecCoef >= 0) ? 1 : -1; if (escapeOnesCounter < 13) { quantSpecCoef = sign * (((INT)1 << escapeOnesCounter) + escape_word); } else { *errorWord |= TOO_MANY_PCW_BODY_SIGN_ESC_BITS_DECODED; quantSpecCoef = Q_VALUE_INVALID; } return quantSpecCoef; } /*--------------------------------------------------------------------------------------------- description: Decodes the Signbits of a priority codeword (PCW) and writes out the resulting quantized spectral values into unsorted sections ----------------------------------------------------------------------------------------------- output: - two or four lines at position in corresponding section (which are not located at the desired position, i.e. they must be reordered in the last of eight function of HCR) ----------------------------------------------------------------------------------------------- return: - updated pQuantSpecCoef pointer (to next empty storage for a line) -------------------------------------------------------------------------------------------- */ static int DecodePCW_Sign(HANDLE_FDK_BITSTREAM bs, const INT bsAnchor, UINT codebookDim, const SCHAR *pQuantVal, FIXP_DBL *pQuantSpecCoef, int *quantSpecCoefIdx, INT *pLeftStartOfSegment, SCHAR *pRemainingBitsInSegment, int *pNumDecodedBits) { UINT i; UINT carryBit; INT quantSpecCoef; for (i = codebookDim; i != 0; i--) { quantSpecCoef = *pQuantVal++; if (quantSpecCoef != 0) { carryBit = HcrGetABitFromBitstream(bs, bsAnchor, pLeftStartOfSegment, pLeftStartOfSegment, /* dummy */ FROM_LEFT_TO_RIGHT); *pRemainingBitsInSegment -= 1; *pNumDecodedBits += 1; if (*pRemainingBitsInSegment < 0 || *pNumDecodedBits >= (1024 >> 1)) { return -1; } /* adapt sign of values according to the decoded sign bit */ if (carryBit != 0) { pQuantSpecCoef[*quantSpecCoefIdx] = -(FIXP_DBL)quantSpecCoef; } else { pQuantSpecCoef[*quantSpecCoefIdx] = (FIXP_DBL)quantSpecCoef; } } else { pQuantSpecCoef[*quantSpecCoefIdx] = FL2FXCONST_DBL(0.0f); } *quantSpecCoefIdx += 1; if (*quantSpecCoefIdx >= 1024) { return -1; } } return 0; } /*--------------------------------------------------------------------------------------------- description: Mutes spectral lines which have been marked as erroneous (Q_VALUE_INVALID) -------------------------------------------------------------------------------------------- */ void HcrMuteErroneousLines(H_HCR_INFO hHcr) { int c; FIXP_DBL *RESTRICT pLong = SPEC_LONG(hHcr->decInOut.pQuantizedSpectralCoefficientsBase); /* if there is a line with value Q_VALUE_INVALID mute it */ for (c = 0; c < 1024; c++) { if (pLong[c] == (FIXP_DBL)Q_VALUE_INVALID) { pLong[c] = FL2FXCONST_DBL(0.0f); /* muting */ } } } fdk-aac-2.0.2/libAACdec/src/aacdec_hcr.h000066400000000000000000000134201404225310700175220ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Robert Weidner (DSP Solutions) Description: HCR Decoder: Interface function declaration; common defines and structures; defines for switching error-generator, -detector, and -concealment *******************************************************************************/ #ifndef AACDEC_HCR_H #define AACDEC_HCR_H #include "channelinfo.h" #include "FDK_bitstream.h" UINT HcrInit(H_HCR_INFO pHcr, CAacDecoderChannelInfo *pAacDecoderChannelInfo, const SamplingRateInfo *pSamplingRateInfo, HANDLE_FDK_BITSTREAM bs); UINT HcrDecoder(H_HCR_INFO hHcr, CAacDecoderChannelInfo *pAacDecoderChannelInfo, const SamplingRateInfo *pSamplingRateInfo, HANDLE_FDK_BITSTREAM bs); void CarryBitToBranchValue(UCHAR carryBit, UINT treeNode, UINT *branchValue, UINT *branchNode); void CHcr_Read(HANDLE_FDK_BITSTREAM bs, CAacDecoderChannelInfo *pAacDecoderChannelInfo, const MP4_ELEMENT_ID globalHcrType); void HcrMuteErroneousLines(H_HCR_INFO hHcr); void setHcrType(H_HCR_INFO hHcr, MP4_ELEMENT_ID type); INT getHcrType(H_HCR_INFO hHcr); #endif /* AACDEC_HCR_H */ fdk-aac-2.0.2/libAACdec/src/aacdec_hcr_bit.cpp000066400000000000000000000161531404225310700207210ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Robert Weidner (DSP Solutions) Description: HCR Decoder: Bitstream reading *******************************************************************************/ #include "aacdec_hcr_bit.h" /*--------------------------------------------------------------------------------------------- description: This function toggles the read direction. ----------------------------------------------------------------------------------------------- input: current read direction ----------------------------------------------------------------------------------------------- return: new read direction -------------------------------------------------------------------------------------------- */ UCHAR ToggleReadDirection(UCHAR readDirection) { if (readDirection == FROM_LEFT_TO_RIGHT) { return FROM_RIGHT_TO_LEFT; } else { return FROM_LEFT_TO_RIGHT; } } /*--------------------------------------------------------------------------------------------- description: This function returns a bit from the bitstream according to read direction. It is called very often, therefore it makes sense to inline it (runtime). ----------------------------------------------------------------------------------------------- input: - handle to FDK bitstream - reference value marking start of bitfield - pLeftStartOfSegment - pRightStartOfSegment - readDirection ----------------------------------------------------------------------------------------------- return: - bit from bitstream -------------------------------------------------------------------------------------------- */ UINT HcrGetABitFromBitstream(HANDLE_FDK_BITSTREAM bs, const INT bsAnchor, INT *pLeftStartOfSegment, INT *pRightStartOfSegment, UCHAR readDirection) { UINT bit; INT readBitOffset; if (readDirection == FROM_LEFT_TO_RIGHT) { readBitOffset = (INT)FDKgetValidBits(bs) - bsAnchor + *pLeftStartOfSegment; if (readBitOffset) { FDKpushBiDirectional(bs, readBitOffset); } bit = FDKreadBits(bs, 1); *pLeftStartOfSegment += 1; } else { readBitOffset = (INT)FDKgetValidBits(bs) - bsAnchor + *pRightStartOfSegment; if (readBitOffset) { FDKpushBiDirectional(bs, readBitOffset); } /* to be replaced with a brother function of FDKreadBits() */ bit = FDKreadBits(bs, 1); FDKpushBack(bs, 2); *pRightStartOfSegment -= 1; } return (bit); } fdk-aac-2.0.2/libAACdec/src/aacdec_hcr_bit.h000066400000000000000000000121731404225310700203640ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Robert Weidner (DSP Solutions) Description: HCR Decoder: Bitstream reading prototypes *******************************************************************************/ #ifndef AACDEC_HCR_BIT_H #define AACDEC_HCR_BIT_H #include "aacdec_hcr.h" UCHAR ToggleReadDirection(UCHAR readDirection); UINT HcrGetABitFromBitstream(HANDLE_FDK_BITSTREAM bs, const INT bsAnchor, INT *pLeftStartOfSegment, INT *pRightStartOfSegment, UCHAR readDirection); #endif /* AACDEC_HCR_BIT_H */ fdk-aac-2.0.2/libAACdec/src/aacdec_hcr_types.h000066400000000000000000000441741404225310700207600ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Robert Weidner (DSP Solutions) Description: HCR Decoder: Common defines and structures; defines for switching error-generator, -detector, and -concealment; *******************************************************************************/ #ifndef AACDEC_HCR_TYPES_H #define AACDEC_HCR_TYPES_H #include "FDK_bitstream.h" #include "overlapadd.h" /* ------------------------------------------------ */ /* ------------------------------------------------ */ #define LINES_PER_UNIT 4 /* ------------------------------------------------ */ /* ------------------------------------------------ */ /* ----------- basic HCR configuration ------------ */ #define MAX_SFB_HCR \ (((1024 / 8) / LINES_PER_UNIT) * 8) /* (8 * 16) is not enough because sfbs \ are split in units for blocktype \ short */ #define NUMBER_OF_UNIT_GROUPS (LINES_PER_UNIT * 8) #define LINES_PER_UNIT_GROUP (1024 / NUMBER_OF_UNIT_GROUPS) /* 15 16 30 32 */ /* ------------------------------------------------ */ /* ------------------------------------------------ */ /* ------------------------------------------------ */ #define FROM_LEFT_TO_RIGHT 0 #define FROM_RIGHT_TO_LEFT 1 #define MAX_CB_PAIRS 23 #define MAX_HCR_SETS 14 #define ESCAPE_VALUE 16 #define POSITION_OF_FLAG_A 21 #define POSITION_OF_FLAG_B 20 #define MAX_CB 32 /* last used CB is cb #31 when VCB11 is used */ #define MAX_CB_CHECK \ 32 /* support for VCB11 available -- is more general, could therefore used \ in both cases */ #define NUMBER_OF_BIT_IN_WORD 32 /* log */ #define THIRTYTWO_LOG_DIV_TWO_LOG 5 #define EIGHT_LOG_DIV_TWO_LOG 3 #define FOUR_LOG_DIV_TWO_LOG 2 /* borders */ #define CPE_TOP_LENGTH 12288 #define SCE_TOP_LENGTH 6144 #define LEN_OF_LONGEST_CW_TOP_LENGTH 49 /* qsc's of high level */ #define Q_VALUE_INVALID \ 8192 /* mark a invalid line with this value (to be concealed later on) */ #define HCR_DIRAC 500 /* a line of high level */ /* masks */ #define MASK_LEFT 0xFFF000 #define MASK_RIGHT 0xFFF #define CLR_BIT_10 0x3FF #define TEST_BIT_10 0x400 #define LEFT_OFFSET 12 /* when set HCR is replaced by a dummy-module which just fills the outputbuffer * with a dirac sequence */ /* use this if HCR is suspected to write in other modules -- if error is stell * there, HCR is innocent */ /* ------------------------------ */ /* - insert HCR errors - */ /* ------------------------------ */ /* modify input lengths -- high protected */ #define ERROR_LORSD 0 /* offset: error if different from zero */ #define ERROR_LOLC 0 /* offset: error if different from zero */ /* segments are earlier empty as expected when decoding PCWs */ #define ERROR_PCW_BODY \ 0 /* set a positive values to trigger the error (make segments earlyer \ appear to be empty) */ #define ERROR_PCW_BODY_SIGN \ 0 /* set a positive values to trigger the error (make segments earlyer \ appear to be empty) */ #define ERROR_PCW_BODY_SIGN_ESC \ 0 /* set a positive values to trigger the error (make segments earlyer \ appear to be empty) */ /* pretend there are too many bits decoded (enlarge length of codeword) at PCWs * -- use a positive value */ #define ERROR_PCW_BODY_ONLY_TOO_LONG \ 0 /* set a positive values to trigger the error */ #define ERROR_PCW_BODY_SIGN_TOO_LONG \ 0 /* set a positive values to trigger the error */ #define ERROR_PCW_BODY_SIGN_ESC_TOO_LONG \ 0 /* set a positive values to trigger the error */ /* modify HCR bitstream block */ #define MODULO_DIVISOR_HCR 30 /* ------------------------------ */ /* - detect HCR errors - */ /* ------------------------------ */ /* check input data */ /* during decoding */ /* all the segments are checked -- therefore -- if this check passes, its a kind of evidence that the decoded PCWs and non-PCWs are fine */ /* if a codeword is decoded there exists a border for the number of bits, which are allowed to read for this codeword. This border is the minimum of the length of the longest codeword (for the currently used codebook) and the separately transmitted 'lengthOfLongestCodeword' in this frame and channel. The number of decoded bits is counted (for PCWs only -- there it makes really sense in my opinion). If this number exceeds the border (derived as minimum -- see above), a error is detected. */ /* ----------------------------------------------------------------------------------------------------- This error check could be set to zero because due to a test within RVLC-Escape-huffman-Decoder a too long codeword could not be detected -- it seems that for RVLC-Escape-Codeword the coderoom is used to 100%. Therefore I assume that the coderoom is used to 100% also for the codebooks 1..11 used at HCR Therefore this test is deactivated pending further notice ----------------------------------------------------------------------------------------------------- */ /* test if the number of remaining bits in a segment is _below_ zero. If there are no errors the lowest allowed value for remainingBitsInSegment is zero. This check also could be set to zero (save runtime) */ /* other */ /* when set to '1', avoid setting the LAV-Flag in errorLog due to a previous-line-marking (at PCW decoder). A little more runtime is needed then when writing values out into output-buffer. */ /* ------------------------------ */ /* - conceal HCR errors - */ /* ------------------------------ */ #define HCR_ERROR_CONCEALMENT \ 1 /* if set to '1', HCR _mutes_ the erred quantized spectral coefficients */ // ------------------------------------------------------------------------------------------------------------------ // errorLog: A word of 32 bits used for // logging possible errors within HCR // in case of distorted // bitstreams. Table of all // known errors: // ------------------------------------------------------------------------------------------------------------------------ // bit fatal location meaning // ----+-----+-----------+-------------------------------------- #define SEGMENT_OVERRIDE_ERR_PCW_BODY \ 0x80000000 // 31 no PCW-Dec During PCW decoding it is checked after // every PCW if there are too many bits decoded (immediate // check). #define SEGMENT_OVERRIDE_ERR_PCW_BODY_SIGN \ 0x40000000 // 30 no PCW-Dec During PCW decoding it is checked after // every PCW if there are too many bits decoded (immediate // check). #define SEGMENT_OVERRIDE_ERR_PCW_BODY_SIGN_ESC \ 0x20000000 // 29 no PCW-Dec During PCW decoding it is checked after // every PCW if there are too many bits decoded (immediate // check). #define EXTENDED_SORTED_COUNTER_OVERFLOW \ 0x10000000 // 28 yes Init-Dec Error during extending sideinfo // (neither a PCW nor a nonPCW was decoded so far) // 0x08000000 // 27 reserved // 0x04000000 // 26 reserved // 0x02000000 // 25 reserved // 0x01000000 // 24 reserved // 0x00800000 // 23 reserved // 0x00400000 // 22 reserved // 0x00200000 // 21 reserved // 0x00100000 // 20 reserved /* special errors */ #define TOO_MANY_PCW_BODY_BITS_DECODED \ 0x00080000 // 19 yes PCW-Dec During PCW-body-decoding too many bits // have been read from bitstream -- advice: skip non-PCW decoding #define TOO_MANY_PCW_BODY_SIGN_BITS_DECODED \ 0x00040000 // 18 yes PCW-Dec During PCW-body-sign-decoding too many // bits have been read from bitstream -- advice: skip non-PCW // decoding #define TOO_MANY_PCW_BODY_SIGN_ESC_BITS_DECODED \ 0x00020000 // 17 yes PCW-Dec During PCW-body-sign-esc-decoding too // many bits have been read from bitstream -- advice: skip // non-PCW decoding // 0x00010000 // 16 reserved #define STATE_ERROR_BODY_ONLY \ 0x00008000 // 15 no NonPCW-Dec State machine returned with error #define STATE_ERROR_BODY_SIGN__BODY \ 0x00004000 // 14 no NonPCW-Dec State machine returned with error #define STATE_ERROR_BODY_SIGN__SIGN \ 0x00002000 // 13 no NonPCW-Dec State machine returned with error #define STATE_ERROR_BODY_SIGN_ESC__BODY \ 0x00001000 // 12 no NonPCW-Dec State machine returned with error #define STATE_ERROR_BODY_SIGN_ESC__SIGN \ 0x00000800 // 11 no NonPCW-Dec State machine returned with error #define STATE_ERROR_BODY_SIGN_ESC__ESC_PREFIX \ 0x00000400 // 10 no NonPCW-Dec State machine returned with error #define STATE_ERROR_BODY_SIGN_ESC__ESC_WORD \ 0x00000200 // 9 no NonPCW-Dec State machine returned with error #define HCR_SI_LENGTHS_FAILURE \ 0x00000100 // 8 yes Init-Dec LengthOfLongestCodeword must not be // less than lenghtOfReorderedSpectralData #define NUM_SECT_OUT_OF_RANGE_SHORT_BLOCK \ 0x00000080 // 7 yes Init-Dec The number of sections is not within // the allowed range (short block) #define NUM_SECT_OUT_OF_RANGE_LONG_BLOCK \ 0x00000040 // 6 yes Init-Dec The number of sections is not within // the allowed range (long block) #define LINE_IN_SECT_OUT_OF_RANGE_SHORT_BLOCK \ 0x00000020 // 5 yes Init-Dec The number of lines per section is not // within the allowed range (short block) #define CB_OUT_OF_RANGE_SHORT_BLOCK \ 0x00000010 // 4 yes Init-Dec The codebook is not within the allowed // range (short block) #define LINE_IN_SECT_OUT_OF_RANGE_LONG_BLOCK \ 0x00000008 // 3 yes Init-Dec The number of lines per section is not // within the allowed range (long block) #define CB_OUT_OF_RANGE_LONG_BLOCK \ 0x00000004 // 2 yes Init-Dec The codebook is not within the allowed // range (long block) #define LAV_VIOLATION \ 0x00000002 // 1 no Final The absolute value of at least one // decoded line was too high for the according codebook. #define BIT_IN_SEGMENTATION_ERROR \ 0x00000001 // 0 no Final After PCW and non-PWC-decoding at least // one segment is not zero (global check). /*----------*/ #define HCR_FATAL_PCW_ERROR_MASK 0x100E01FC typedef enum { PCW_BODY, PCW_BODY_SIGN, PCW_BODY_SIGN_ESC } PCW_TYPE; /* interface Decoder <---> HCR */ typedef struct { UINT errorLog; SPECTRAL_PTR pQuantizedSpectralCoefficientsBase; int quantizedSpectralCoefficientsIdx; SHORT lengthOfReorderedSpectralData; SHORT numSection; SHORT *pNumLineInSect; INT bitstreamAnchor; SCHAR lengthOfLongestCodeword; UCHAR *pCodebook; } HCR_INPUT_OUTPUT; typedef struct { const UCHAR *pMinOfCbPair; const UCHAR *pMaxOfCbPair; } HCR_CB_PAIRS; typedef struct { const USHORT *pLargestAbsVal; const UCHAR *pMaxCwLength; const UCHAR *pCbDimension; const UCHAR *pCbDimShift; const UCHAR *pCbSign; const UCHAR *pCbPriority; } HCR_TABLE_INFO; typedef struct { UINT numSegment; UINT pSegmentBitfield[((1024 >> 1) / NUMBER_OF_BIT_IN_WORD + 1)]; UINT pCodewordBitfield[((1024 >> 1) / NUMBER_OF_BIT_IN_WORD + 1)]; UINT segmentOffset; INT pLeftStartOfSegment[1024 >> 1]; INT pRightStartOfSegment[1024 >> 1]; SCHAR pRemainingBitsInSegment[1024 >> 1]; UCHAR readDirection; UCHAR numWordForBitfield; USHORT pNumBitValidInLastWord; } HCR_SEGMENT_INFO; typedef struct { UINT numCodeword; UINT numSortedSection; USHORT pNumCodewordInSection[MAX_SFB_HCR]; USHORT pNumSortedCodewordInSection[MAX_SFB_HCR]; USHORT pNumExtendedSortedCodewordInSection[MAX_SFB_HCR + MAX_HCR_SETS]; int numExtendedSortedCodewordInSectionIdx; USHORT pNumExtendedSortedSectionsInSets[MAX_HCR_SETS]; int numExtendedSortedSectionsInSetsIdx; USHORT pReorderOffset[MAX_SFB_HCR]; UCHAR pSortedCodebook[MAX_SFB_HCR]; UCHAR pExtendedSortedCodebook[MAX_SFB_HCR + MAX_HCR_SETS]; int extendedSortedCodebookIdx; UCHAR pMaxLenOfCbInExtSrtSec[MAX_SFB_HCR + MAX_HCR_SETS]; int maxLenOfCbInExtSrtSecIdx; UCHAR pCodebookSwitch[MAX_SFB_HCR]; } HCR_SECTION_INFO; typedef UINT (*STATEFUNC)(HANDLE_FDK_BITSTREAM, void *); typedef struct { /* worst-case and 1024/4 non-PCWs exist in worst-case */ FIXP_DBL *pResultBase; /* Base address for spectral data output target buffer */ UINT iNode[1024 >> 2]; /* Helper indices for code books */ USHORT iResultPointer[1024 >> 2]; /* Helper indices for accessing pResultBase */ UINT pEscapeSequenceInfo[1024 >> 2]; UINT codewordOffset; STATEFUNC pState; UCHAR pCodebook[1024 >> 2]; UCHAR pCntSign[1024 >> 2]; /* this array holds the states coded as integer values within the range * [0,1,..,7] */ SCHAR pSta[1024 >> 2]; } HCR_NON_PCW_SIDEINFO; typedef struct { HCR_INPUT_OUTPUT decInOut; HCR_SEGMENT_INFO segmentInfo; HCR_SECTION_INFO sectionInfo; HCR_NON_PCW_SIDEINFO nonPcwSideinfo; } CErHcrInfo; typedef CErHcrInfo *H_HCR_INFO; #endif /* AACDEC_HCR_TYPES_H */ fdk-aac-2.0.2/libAACdec/src/aacdec_hcrs.cpp000066400000000000000000002007051404225310700202440ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Robert Weidner (DSP Solutions) Description: HCR Decoder: Prepare decoding of non-PCWs, segmentation- and bitfield-handling, HCR-Statemachine *******************************************************************************/ #include "aacdec_hcrs.h" #include "aacdec_hcr.h" #include "aacdec_hcr_bit.h" #include "aac_rom.h" #include "aac_ram.h" static UINT InitSegmentBitfield(UINT *pNumSegment, SCHAR *pRemainingBitsInSegment, UINT *pSegmentBitfield, UCHAR *pNumWordForBitfield, USHORT *pNumBitValidInLastWord); static void InitNonPCWSideInformationForCurrentSet(H_HCR_INFO pHcr); static INT ModuloValue(INT input, INT bufferlength); static void ClearBitFromBitfield(STATEFUNC *ptrState, UINT offset, UINT *pBitfield); /*--------------------------------------------------------------------------------------------- description: This function decodes all non-priority codewords (non-PCWs) by using a state-machine. -------------------------------------------------------------------------------------------- */ void DecodeNonPCWs(HANDLE_FDK_BITSTREAM bs, H_HCR_INFO pHcr) { UINT numValidSegment; INT segmentOffset; INT codewordOffsetBase; INT codewordOffset; UINT trial; UINT *pNumSegment; SCHAR *pRemainingBitsInSegment; UINT *pSegmentBitfield; UCHAR *pNumWordForBitfield; USHORT *pNumBitValidInLastWord; UINT *pCodewordBitfield; INT bitfieldWord; INT bitInWord; UINT tempWord; UINT interMediateWord; INT tempBit; INT carry; UINT numCodeword; UCHAR numSet; UCHAR currentSet; UINT codewordInSet; UINT remainingCodewordsInSet; SCHAR *pSta; UINT ret; pNumSegment = &(pHcr->segmentInfo.numSegment); pRemainingBitsInSegment = pHcr->segmentInfo.pRemainingBitsInSegment; pSegmentBitfield = pHcr->segmentInfo.pSegmentBitfield; pNumWordForBitfield = &(pHcr->segmentInfo.numWordForBitfield); pNumBitValidInLastWord = &(pHcr->segmentInfo.pNumBitValidInLastWord); pSta = pHcr->nonPcwSideinfo.pSta; numValidSegment = InitSegmentBitfield(pNumSegment, pRemainingBitsInSegment, pSegmentBitfield, pNumWordForBitfield, pNumBitValidInLastWord); if (numValidSegment != 0) { numCodeword = pHcr->sectionInfo.numCodeword; numSet = ((numCodeword - 1) / *pNumSegment) + 1; pHcr->segmentInfo.readDirection = FROM_RIGHT_TO_LEFT; /* Process sets subsequently */ for (currentSet = 1; currentSet < numSet; currentSet++) { /* step 1 */ numCodeword -= *pNumSegment; /* number of remaining non PCWs [for all sets] */ if (numCodeword < *pNumSegment) { codewordInSet = numCodeword; /* for last set */ } else { codewordInSet = *pNumSegment; /* for all sets except last set */ } /* step 2 */ /* prepare array 'CodewordBitfield'; as much ones are written from left in * all words, as much decodedCodewordInSetCounter nonPCWs exist in this * set */ tempWord = 0xFFFFFFFF; pCodewordBitfield = pHcr->segmentInfo.pCodewordBitfield; for (bitfieldWord = *pNumWordForBitfield; bitfieldWord != 0; bitfieldWord--) { /* loop over all used words */ if (codewordInSet > NUMBER_OF_BIT_IN_WORD) { /* more codewords than number of bits => fill ones */ /* fill a whole word with ones */ *pCodewordBitfield++ = tempWord; codewordInSet -= NUMBER_OF_BIT_IN_WORD; /* subtract number of bits */ } else { /* prepare last tempWord */ for (remainingCodewordsInSet = codewordInSet; remainingCodewordsInSet < NUMBER_OF_BIT_IN_WORD; remainingCodewordsInSet++) { tempWord = tempWord & ~(1 << (NUMBER_OF_BIT_IN_WORD - 1 - remainingCodewordsInSet)); /* set a zero at bit number (NUMBER_OF_BIT_IN_WORD-1-i) in tempWord */ } *pCodewordBitfield++ = tempWord; tempWord = 0x00000000; } } pCodewordBitfield = pHcr->segmentInfo.pCodewordBitfield; /* step 3 */ /* build non-PCW sideinfo for each non-PCW of the current set */ InitNonPCWSideInformationForCurrentSet(pHcr); /* step 4 */ /* decode all non-PCWs belonging to this set */ /* loop over trials */ codewordOffsetBase = 0; for (trial = *pNumSegment; trial > 0; trial--) { /* loop over number of words in bitfields */ segmentOffset = 0; /* start at zero in every segment */ pHcr->segmentInfo.segmentOffset = segmentOffset; /* store in structure for states */ codewordOffset = codewordOffsetBase; pHcr->nonPcwSideinfo.codewordOffset = codewordOffset; /* store in structure for states */ for (bitfieldWord = 0; bitfieldWord < *pNumWordForBitfield; bitfieldWord++) { /* derive tempWord with bitwise and */ tempWord = pSegmentBitfield[bitfieldWord] & pCodewordBitfield[bitfieldWord]; /* if tempWord is not zero, decode something */ if (tempWord != 0) { /* loop over all bits in tempWord; start state machine if & is true */ for (bitInWord = NUMBER_OF_BIT_IN_WORD; bitInWord > 0; bitInWord--) { interMediateWord = ((UINT)1 << (bitInWord - 1)); if ((tempWord & interMediateWord) == interMediateWord) { /* get state and start state machine */ pHcr->nonPcwSideinfo.pState = aStateConstant2State[pSta[codewordOffset]]; while (pHcr->nonPcwSideinfo.pState) { ret = ((STATEFUNC)pHcr->nonPcwSideinfo.pState)(bs, pHcr); if (ret != 0) { return; } } } /* update both offsets */ segmentOffset += 1; /* add NUMBER_OF_BIT_IN_WORD times one */ pHcr->segmentInfo.segmentOffset = segmentOffset; codewordOffset += 1; /* add NUMBER_OF_BIT_IN_WORD times one */ codewordOffset = ModuloValue(codewordOffset, *pNumSegment); /* index of the current codeword lies within modulo range */ pHcr->nonPcwSideinfo.codewordOffset = codewordOffset; } } else { segmentOffset += NUMBER_OF_BIT_IN_WORD; /* add NUMBER_OF_BIT_IN_WORD at once */ pHcr->segmentInfo.segmentOffset = segmentOffset; codewordOffset += NUMBER_OF_BIT_IN_WORD; /* add NUMBER_OF_BIT_IN_WORD at once */ codewordOffset = ModuloValue( codewordOffset, *pNumSegment); /* index of the current codeword lies within modulo range */ pHcr->nonPcwSideinfo.codewordOffset = codewordOffset; } } /* end of bitfield word loop */ /* decrement codeword - pointer */ codewordOffsetBase -= 1; codewordOffsetBase = ModuloValue(codewordOffsetBase, *pNumSegment); /* index of the current codeword base lies within modulo range */ /* rotate numSegment bits in codewordBitfield */ /* rotation of *numSegment bits in bitfield of codewords * (circle-rotation) */ /* get last valid bit */ tempBit = pCodewordBitfield[*pNumWordForBitfield - 1] & (1 << (NUMBER_OF_BIT_IN_WORD - *pNumBitValidInLastWord)); tempBit = tempBit >> (NUMBER_OF_BIT_IN_WORD - *pNumBitValidInLastWord); /* write zero into place where tempBit was fetched from */ pCodewordBitfield[*pNumWordForBitfield - 1] = pCodewordBitfield[*pNumWordForBitfield - 1] & ~(1 << (NUMBER_OF_BIT_IN_WORD - *pNumBitValidInLastWord)); /* rotate last valid word */ pCodewordBitfield[*pNumWordForBitfield - 1] = pCodewordBitfield[*pNumWordForBitfield - 1] >> 1; /* transfare carry bit 0 from current word into bitposition 31 from next * word and rotate current word */ for (bitfieldWord = *pNumWordForBitfield - 2; bitfieldWord > -1; bitfieldWord--) { /* get carry (=bit at position 0) from current word */ carry = pCodewordBitfield[bitfieldWord] & 1; /* put the carry bit at position 31 into word right from current word */ pCodewordBitfield[bitfieldWord + 1] = pCodewordBitfield[bitfieldWord + 1] | (carry << (NUMBER_OF_BIT_IN_WORD - 1)); /* shift current word */ pCodewordBitfield[bitfieldWord] = pCodewordBitfield[bitfieldWord] >> 1; } /* put tempBit into free bit-position 31 from first word */ pCodewordBitfield[0] = pCodewordBitfield[0] | (tempBit << (NUMBER_OF_BIT_IN_WORD - 1)); } /* end of trial loop */ /* toggle read direction */ pHcr->segmentInfo.readDirection = ToggleReadDirection(pHcr->segmentInfo.readDirection); } /* end of set loop */ /* all non-PCWs of this spectrum are decoded */ } /* all PCWs and all non PCWs are decoded. They are unbacksorted in output * buffer. Here is the Interface with comparing QSCs to asm decoding */ } /*--------------------------------------------------------------------------------------------- description: This function prepares the bitfield used for the segments. The list is set up once to be used in all following sets. If a segment is decoded empty, the according bit from the Bitfield is removed. ----------------------------------------------------------------------------------------------- return: numValidSegment = the number of valid segments -------------------------------------------------------------------------------------------- */ static UINT InitSegmentBitfield(UINT *pNumSegment, SCHAR *pRemainingBitsInSegment, UINT *pSegmentBitfield, UCHAR *pNumWordForBitfield, USHORT *pNumBitValidInLastWord) { SHORT i; USHORT r; UCHAR bitfieldWord; UINT tempWord; USHORT numValidSegment; *pNumWordForBitfield = (*pNumSegment == 0) ? 0 : ((*pNumSegment - 1) >> THIRTYTWO_LOG_DIV_TWO_LOG) + 1; /* loop over all words, which are completely used or only partial */ /* bit in pSegmentBitfield is zero if segment is empty; bit in * pSegmentBitfield is one if segment is not empty */ numValidSegment = 0; *pNumBitValidInLastWord = *pNumSegment; /* loop over words */ for (bitfieldWord = 0; bitfieldWord < *pNumWordForBitfield - 1; bitfieldWord++) { tempWord = 0xFFFFFFFF; /* set ones */ r = bitfieldWord << THIRTYTWO_LOG_DIV_TWO_LOG; for (i = 0; i < NUMBER_OF_BIT_IN_WORD; i++) { if (pRemainingBitsInSegment[r + i] == 0) { tempWord = tempWord & ~(1 << (NUMBER_OF_BIT_IN_WORD - 1 - i)); /* set a zero at bit number (NUMBER_OF_BIT_IN_WORD-1-i) in tempWord */ } else { numValidSegment += 1; /* count segments which are not empty */ } } pSegmentBitfield[bitfieldWord] = tempWord; /* store result */ *pNumBitValidInLastWord -= NUMBER_OF_BIT_IN_WORD; /* calculate number of zeros on LSB side in the last word */ } /* calculate last word: prepare special tempWord */ tempWord = 0xFFFFFFFF; for (i = 0; i < (NUMBER_OF_BIT_IN_WORD - *pNumBitValidInLastWord); i++) { tempWord = tempWord & ~(1 << i); /* clear bit i in tempWord */ } /* calculate last word */ r = bitfieldWord << THIRTYTWO_LOG_DIV_TWO_LOG; for (i = 0; i < *pNumBitValidInLastWord; i++) { if (pRemainingBitsInSegment[r + i] == 0) { tempWord = tempWord & ~(1 << (NUMBER_OF_BIT_IN_WORD - 1 - i)); /* set a zero at bit number (NUMBER_OF_BIT_IN_WORD-1-i) in tempWord */ } else { numValidSegment += 1; /* count segments which are not empty */ } } pSegmentBitfield[bitfieldWord] = tempWord; /* store result */ return numValidSegment; } /*--------------------------------------------------------------------------------------------- description: This function sets up sideinfo for the non-PCW decoder (for the current set). ---------------------------------------------------------------------------------------------*/ static void InitNonPCWSideInformationForCurrentSet(H_HCR_INFO pHcr) { USHORT i, k; UCHAR codebookDim; UINT startNode; UCHAR *pCodebook = pHcr->nonPcwSideinfo.pCodebook; UINT *iNode = pHcr->nonPcwSideinfo.iNode; UCHAR *pCntSign = pHcr->nonPcwSideinfo.pCntSign; USHORT *iResultPointer = pHcr->nonPcwSideinfo.iResultPointer; UINT *pEscapeSequenceInfo = pHcr->nonPcwSideinfo.pEscapeSequenceInfo; SCHAR *pSta = pHcr->nonPcwSideinfo.pSta; USHORT *pNumExtendedSortedCodewordInSection = pHcr->sectionInfo.pNumExtendedSortedCodewordInSection; int numExtendedSortedCodewordInSectionIdx = pHcr->sectionInfo.numExtendedSortedCodewordInSectionIdx; UCHAR *pExtendedSortedCodebook = pHcr->sectionInfo.pExtendedSortedCodebook; int extendedSortedCodebookIdx = pHcr->sectionInfo.extendedSortedCodebookIdx; USHORT *pNumExtendedSortedSectionsInSets = pHcr->sectionInfo.pNumExtendedSortedSectionsInSets; int numExtendedSortedSectionsInSetsIdx = pHcr->sectionInfo.numExtendedSortedSectionsInSetsIdx; int quantizedSpectralCoefficientsIdx = pHcr->decInOut.quantizedSpectralCoefficientsIdx; const UCHAR *pCbDimension = aDimCb; int iterationCounter = 0; /* loop over number of extended sorted sections in the current set so all * codewords sideinfo variables within this set can be prepared for decoding */ for (i = pNumExtendedSortedSectionsInSets[numExtendedSortedSectionsInSetsIdx]; i != 0; i--) { codebookDim = pCbDimension[pExtendedSortedCodebook[extendedSortedCodebookIdx]]; startNode = *aHuffTable[pExtendedSortedCodebook[extendedSortedCodebookIdx]]; for (k = pNumExtendedSortedCodewordInSection [numExtendedSortedCodewordInSectionIdx]; k != 0; k--) { iterationCounter++; if (iterationCounter > (1024 >> 2)) { return; } *pSta++ = aCodebook2StartInt [pExtendedSortedCodebook[extendedSortedCodebookIdx]]; *pCodebook++ = pExtendedSortedCodebook[extendedSortedCodebookIdx]; *iNode++ = startNode; *pCntSign++ = 0; *iResultPointer++ = quantizedSpectralCoefficientsIdx; *pEscapeSequenceInfo++ = 0; quantizedSpectralCoefficientsIdx += codebookDim; /* update pointer by codebookDim --> point to next starting value for writing out */ if (quantizedSpectralCoefficientsIdx >= 1024) { return; } } numExtendedSortedCodewordInSectionIdx++; /* inc ptr for next ext sort sec in current set */ extendedSortedCodebookIdx++; /* inc ptr for next ext sort sec in current set */ if (numExtendedSortedCodewordInSectionIdx >= (MAX_SFB_HCR + MAX_HCR_SETS) || extendedSortedCodebookIdx >= (MAX_SFB_HCR + MAX_HCR_SETS)) { return; } } numExtendedSortedSectionsInSetsIdx++; /* inc ptr for next set of non-PCWs */ if (numExtendedSortedCodewordInSectionIdx >= (MAX_SFB_HCR + MAX_HCR_SETS)) { return; } /* Write back indexes */ pHcr->sectionInfo.numExtendedSortedCodewordInSectionIdx = numExtendedSortedCodewordInSectionIdx; pHcr->sectionInfo.extendedSortedCodebookIdx = extendedSortedCodebookIdx; pHcr->sectionInfo.numExtendedSortedSectionsInSetsIdx = numExtendedSortedSectionsInSetsIdx; pHcr->sectionInfo.numExtendedSortedCodewordInSectionIdx = numExtendedSortedCodewordInSectionIdx; pHcr->decInOut.quantizedSpectralCoefficientsIdx = quantizedSpectralCoefficientsIdx; } /*--------------------------------------------------------------------------------------------- description: This function returns the input value if the value is in the range of bufferlength. If is smaller, one bufferlength is added, if is bigger one bufferlength is subtracted. ----------------------------------------------------------------------------------------------- return: modulo result -------------------------------------------------------------------------------------------- */ static INT ModuloValue(INT input, INT bufferlength) { if (input > (bufferlength - 1)) { return (input - bufferlength); } if (input < 0) { return (input + bufferlength); } return input; } /*--------------------------------------------------------------------------------------------- description: This function clears a bit from current bitfield and switches off the statemachine. A bit is cleared in two cases: a) a codeword is decoded, then a bit is cleared in codeword bitfield b) a segment is decoded empty, then a bit is cleared in segment bitfield -------------------------------------------------------------------------------------------- */ static void ClearBitFromBitfield(STATEFUNC *ptrState, UINT offset, UINT *pBitfield) { UINT numBitfieldWord; UINT numBitfieldBit; /* get both values needed for clearing the bit */ numBitfieldWord = offset >> THIRTYTWO_LOG_DIV_TWO_LOG; /* int = wordNr */ numBitfieldBit = offset - (numBitfieldWord << THIRTYTWO_LOG_DIV_TWO_LOG); /* fract = bitNr */ /* clear a bit in bitfield */ pBitfield[numBitfieldWord] = pBitfield[numBitfieldWord] & ~(1 << (NUMBER_OF_BIT_IN_WORD - 1 - numBitfieldBit)); /* switch off state machine because codeword is decoded and/or because segment * is empty */ *ptrState = NULL; } /* ========================================================================================= the states of the statemachine ========================================================================================= */ /*--------------------------------------------------------------------------------------------- description: Decodes the body of a codeword. This State is used for codebooks 1,2,5 and 6. No sign bits are decoded, because the table of the quantized spectral values has got a valid sign at the quantized spectral lines. ----------------------------------------------------------------------------------------------- output: Two or four quantizes spectral values written at position where pResultPointr points to ----------------------------------------------------------------------------------------------- return: 0 -------------------------------------------------------------------------------------------- */ UINT Hcr_State_BODY_ONLY(HANDLE_FDK_BITSTREAM bs, void *ptr) { H_HCR_INFO pHcr = (H_HCR_INFO)ptr; UINT *pSegmentBitfield; UINT *pCodewordBitfield; UINT segmentOffset; FIXP_DBL *pResultBase; UINT *iNode; USHORT *iResultPointer; UINT codewordOffset; UINT branchNode; UINT branchValue; UINT iQSC; UINT treeNode; UCHAR carryBit; INT *pLeftStartOfSegment; INT *pRightStartOfSegment; SCHAR *pRemainingBitsInSegment; UCHAR readDirection; UCHAR *pCodebook; UCHAR dimCntr; const UINT *pCurrentTree; const UCHAR *pCbDimension; const SCHAR *pQuantVal; const SCHAR *pQuantValBase; pRemainingBitsInSegment = pHcr->segmentInfo.pRemainingBitsInSegment; pLeftStartOfSegment = pHcr->segmentInfo.pLeftStartOfSegment; pRightStartOfSegment = pHcr->segmentInfo.pRightStartOfSegment; readDirection = pHcr->segmentInfo.readDirection; pSegmentBitfield = pHcr->segmentInfo.pSegmentBitfield; pCodewordBitfield = pHcr->segmentInfo.pCodewordBitfield; segmentOffset = pHcr->segmentInfo.segmentOffset; pCodebook = pHcr->nonPcwSideinfo.pCodebook; iNode = pHcr->nonPcwSideinfo.iNode; pResultBase = pHcr->nonPcwSideinfo.pResultBase; iResultPointer = pHcr->nonPcwSideinfo.iResultPointer; codewordOffset = pHcr->nonPcwSideinfo.codewordOffset; pCbDimension = aDimCb; treeNode = iNode[codewordOffset]; pCurrentTree = aHuffTable[pCodebook[codewordOffset]]; for (; pRemainingBitsInSegment[segmentOffset] > 0; pRemainingBitsInSegment[segmentOffset] -= 1) { carryBit = HcrGetABitFromBitstream( bs, pHcr->decInOut.bitstreamAnchor, &pLeftStartOfSegment[segmentOffset], &pRightStartOfSegment[segmentOffset], readDirection); CarryBitToBranchValue(carryBit, /* make a step in decoding tree */ treeNode, &branchValue, &branchNode); /* if end of branch reached write out lines and count bits needed for sign, * otherwise store node in codeword sideinfo */ if ((branchNode & TEST_BIT_10) == TEST_BIT_10) { /* test bit 10 ; ==> body is complete */ pQuantValBase = aQuantTable[pCodebook[codewordOffset]]; /* get base address of quantized values belonging to current codebook */ pQuantVal = pQuantValBase + branchValue; /* set pointer to first valid line [of 2 or 4 quantized values] */ iQSC = iResultPointer[codewordOffset]; /* get position of first line for writing out result */ for (dimCntr = pCbDimension[pCodebook[codewordOffset]]; dimCntr != 0; dimCntr--) { pResultBase[iQSC++] = (FIXP_DBL)*pQuantVal++; /* write out 2 or 4 lines into spectrum; no Sign bits available in this state */ } ClearBitFromBitfield(&(pHcr->nonPcwSideinfo.pState), segmentOffset, pCodewordBitfield); /* clear a bit in bitfield and switch off statemachine */ pRemainingBitsInSegment[segmentOffset] -= 1; /* last reinitialzation of for loop counter (see above) is done here */ break; /* end of branch in tree reached i.e. a whole nonPCW-Body is decoded */ } else { /* body is not decoded completely: */ treeNode = *( pCurrentTree + branchValue); /* update treeNode for further step in decoding tree */ } } iNode[codewordOffset] = treeNode; /* store updated treeNode because maybe decoding of codeword body not finished yet */ if (pRemainingBitsInSegment[segmentOffset] <= 0) { ClearBitFromBitfield(&(pHcr->nonPcwSideinfo.pState), segmentOffset, pSegmentBitfield); /* clear a bit in bitfield and switch off statemachine */ if (pRemainingBitsInSegment[segmentOffset] < 0) { pHcr->decInOut.errorLog |= STATE_ERROR_BODY_ONLY; return BODY_ONLY; } } return STOP_THIS_STATE; } /*--------------------------------------------------------------------------------------------- description: Decodes the codeword body, writes out result and counts the number of quantized spectral values, which are different form zero. For those values sign bits are needed. If sign bit counter cntSign is different from zero, switch to next state to decode sign Bits there. If sign bit counter cntSign is zero, no sign bits are needed and codeword is decoded. ----------------------------------------------------------------------------------------------- output: Two or four written quantizes spectral values written at position where pResultPointr points to. The signs of those lines may be wrong. If the signs [on just one signle sign] is wrong, the next state will correct it. ----------------------------------------------------------------------------------------------- return: 0 -------------------------------------------------------------------------------------------- */ UINT Hcr_State_BODY_SIGN__BODY(HANDLE_FDK_BITSTREAM bs, void *ptr) { H_HCR_INFO pHcr = (H_HCR_INFO)ptr; SCHAR *pRemainingBitsInSegment; INT *pLeftStartOfSegment; INT *pRightStartOfSegment; UCHAR readDirection; UINT *pSegmentBitfield; UINT *pCodewordBitfield; UINT segmentOffset; UCHAR *pCodebook; UINT *iNode; UCHAR *pCntSign; FIXP_DBL *pResultBase; USHORT *iResultPointer; UINT codewordOffset; UINT iQSC; UINT cntSign; UCHAR dimCntr; UCHAR carryBit; SCHAR *pSta; UINT treeNode; UINT branchValue; UINT branchNode; const UCHAR *pCbDimension; const UINT *pCurrentTree; const SCHAR *pQuantValBase; const SCHAR *pQuantVal; pRemainingBitsInSegment = pHcr->segmentInfo.pRemainingBitsInSegment; pLeftStartOfSegment = pHcr->segmentInfo.pLeftStartOfSegment; pRightStartOfSegment = pHcr->segmentInfo.pRightStartOfSegment; readDirection = pHcr->segmentInfo.readDirection; pSegmentBitfield = pHcr->segmentInfo.pSegmentBitfield; pCodewordBitfield = pHcr->segmentInfo.pCodewordBitfield; segmentOffset = pHcr->segmentInfo.segmentOffset; pCodebook = pHcr->nonPcwSideinfo.pCodebook; iNode = pHcr->nonPcwSideinfo.iNode; pCntSign = pHcr->nonPcwSideinfo.pCntSign; pResultBase = pHcr->nonPcwSideinfo.pResultBase; iResultPointer = pHcr->nonPcwSideinfo.iResultPointer; codewordOffset = pHcr->nonPcwSideinfo.codewordOffset; pSta = pHcr->nonPcwSideinfo.pSta; pCbDimension = aDimCb; treeNode = iNode[codewordOffset]; pCurrentTree = aHuffTable[pCodebook[codewordOffset]]; for (; pRemainingBitsInSegment[segmentOffset] > 0; pRemainingBitsInSegment[segmentOffset] -= 1) { carryBit = HcrGetABitFromBitstream( bs, pHcr->decInOut.bitstreamAnchor, &pLeftStartOfSegment[segmentOffset], &pRightStartOfSegment[segmentOffset], readDirection); CarryBitToBranchValue(carryBit, /* make a step in decoding tree */ treeNode, &branchValue, &branchNode); /* if end of branch reached write out lines and count bits needed for sign, * otherwise store node in codeword sideinfo */ if ((branchNode & TEST_BIT_10) == TEST_BIT_10) { /* test bit 10 ; if set body complete */ /* body completely decoded; branchValue is valid, set pQuantVal to first * (of two or four) quantized spectral coefficients */ pQuantValBase = aQuantTable[pCodebook[codewordOffset]]; /* get base address of quantized values belonging to current codebook */ pQuantVal = pQuantValBase + branchValue; /* set pointer to first valid line [of 2 or 4 quantized values] */ iQSC = iResultPointer[codewordOffset]; /* get position of first line for writing result */ /* codeword decoding result is written out here: Write out 2 or 4 * quantized spectral values with probably */ /* wrong sign and count number of values which are different from zero for * sign bit decoding [which happens in next state] */ cntSign = 0; for (dimCntr = pCbDimension[pCodebook[codewordOffset]]; dimCntr != 0; dimCntr--) { pResultBase[iQSC++] = (FIXP_DBL)*pQuantVal; /* write quant. spec. coef. into spectrum */ if (*pQuantVal++ != 0) { cntSign += 1; } } if (cntSign == 0) { ClearBitFromBitfield( &(pHcr->nonPcwSideinfo.pState), segmentOffset, pCodewordBitfield); /* clear a bit in bitfield and switch off statemachine */ } else { pCntSign[codewordOffset] = cntSign; /* write sign count result into codewordsideinfo of current codeword */ pSta[codewordOffset] = BODY_SIGN__SIGN; /* change state */ pHcr->nonPcwSideinfo.pState = aStateConstant2State[pSta[codewordOffset]]; /* get state from separate array of cw-sideinfo */ } pRemainingBitsInSegment[segmentOffset] -= 1; /* last reinitialzation of for loop counter (see above) is done here */ break; /* end of branch in tree reached i.e. a whole nonPCW-Body is decoded */ } else { /* body is not decoded completely: */ treeNode = *( pCurrentTree + branchValue); /* update treeNode for further step in decoding tree */ } } iNode[codewordOffset] = treeNode; /* store updated treeNode because maybe decoding of codeword body not finished yet */ if (pRemainingBitsInSegment[segmentOffset] <= 0) { ClearBitFromBitfield(&(pHcr->nonPcwSideinfo.pState), segmentOffset, pSegmentBitfield); /* clear a bit in bitfield and switch off statemachine */ if (pRemainingBitsInSegment[segmentOffset] < 0) { pHcr->decInOut.errorLog |= STATE_ERROR_BODY_SIGN__BODY; return BODY_SIGN__BODY; } } return STOP_THIS_STATE; } /*--------------------------------------------------------------------------------------------- description: This state decodes the sign bits belonging to a codeword. The state is called as often in different "trials" until pCntSgn[codewordOffset] is zero. ----------------------------------------------------------------------------------------------- output: The two or four quantizes spectral values (written in previous state) have now the correct sign. ----------------------------------------------------------------------------------------------- return: 0 -------------------------------------------------------------------------------------------- */ UINT Hcr_State_BODY_SIGN__SIGN(HANDLE_FDK_BITSTREAM bs, void *ptr) { H_HCR_INFO pHcr = (H_HCR_INFO)ptr; SCHAR *pRemainingBitsInSegment; INT *pLeftStartOfSegment; INT *pRightStartOfSegment; UCHAR readDirection; UINT *pSegmentBitfield; UINT *pCodewordBitfield; UINT segmentOffset; UCHAR *pCntSign; FIXP_DBL *pResultBase; USHORT *iResultPointer; UINT codewordOffset; UCHAR carryBit; UINT iQSC; UCHAR cntSign; pRemainingBitsInSegment = pHcr->segmentInfo.pRemainingBitsInSegment; pLeftStartOfSegment = pHcr->segmentInfo.pLeftStartOfSegment; pRightStartOfSegment = pHcr->segmentInfo.pRightStartOfSegment; readDirection = pHcr->segmentInfo.readDirection; pSegmentBitfield = pHcr->segmentInfo.pSegmentBitfield; pCodewordBitfield = pHcr->segmentInfo.pCodewordBitfield; segmentOffset = pHcr->segmentInfo.segmentOffset; /*pCodebook = */ pCntSign = pHcr->nonPcwSideinfo.pCntSign; pResultBase = pHcr->nonPcwSideinfo.pResultBase; iResultPointer = pHcr->nonPcwSideinfo.iResultPointer; codewordOffset = pHcr->nonPcwSideinfo.codewordOffset; iQSC = iResultPointer[codewordOffset]; cntSign = pCntSign[codewordOffset]; /* loop for sign bit decoding */ for (; pRemainingBitsInSegment[segmentOffset] > 0; pRemainingBitsInSegment[segmentOffset] -= 1) { carryBit = HcrGetABitFromBitstream( bs, pHcr->decInOut.bitstreamAnchor, &pLeftStartOfSegment[segmentOffset], &pRightStartOfSegment[segmentOffset], readDirection); cntSign -= 1; /* decrement sign counter because one sign bit has been read */ /* search for a line (which was decoded in previous state) which is not * zero. [This value will get a sign] */ while (pResultBase[iQSC] == (FIXP_DBL)0) { if (++iQSC >= 1024) { /* points to current value different from zero */ return BODY_SIGN__SIGN; } } /* put sign together with line; if carryBit is zero, the sign is ok already; * no write operation necessary in this case */ if (carryBit != 0) { pResultBase[iQSC] = -pResultBase[iQSC]; /* carryBit = 1 --> minus */ } iQSC++; /* update pointer to next (maybe valid) value */ if (cntSign == 0) { /* if (cntSign==0) ==> set state CODEWORD_DECODED */ ClearBitFromBitfield(&(pHcr->nonPcwSideinfo.pState), segmentOffset, pCodewordBitfield); /* clear a bit in bitfield and switch off statemachine */ pRemainingBitsInSegment[segmentOffset] -= 1; /* last reinitialzation of for loop counter (see above) is done here */ break; /* whole nonPCW-Body and according sign bits are decoded */ } } pCntSign[codewordOffset] = cntSign; iResultPointer[codewordOffset] = iQSC; /* store updated pResultPointer */ if (pRemainingBitsInSegment[segmentOffset] <= 0) { ClearBitFromBitfield(&(pHcr->nonPcwSideinfo.pState), segmentOffset, pSegmentBitfield); /* clear a bit in bitfield and switch off statemachine */ if (pRemainingBitsInSegment[segmentOffset] < 0) { pHcr->decInOut.errorLog |= STATE_ERROR_BODY_SIGN__SIGN; return BODY_SIGN__SIGN; } } return STOP_THIS_STATE; } /*--------------------------------------------------------------------------------------------- description: Decodes the codeword body in case of codebook is 11. Writes out resulting two or four lines [with probably wrong sign] and counts the number of lines, which are different form zero. This information is needed in next state where sign bits will be decoded, if necessary. If sign bit counter cntSign is zero, no sign bits are needed and codeword is decoded completely. ----------------------------------------------------------------------------------------------- output: Two lines (quantizes spectral coefficients) which are probably wrong. The sign may be wrong and if one or two values is/are 16, the following states will decode the escape sequence to correct the values which are wirtten here. ----------------------------------------------------------------------------------------------- return: 0 -------------------------------------------------------------------------------------------- */ UINT Hcr_State_BODY_SIGN_ESC__BODY(HANDLE_FDK_BITSTREAM bs, void *ptr) { H_HCR_INFO pHcr = (H_HCR_INFO)ptr; SCHAR *pRemainingBitsInSegment; INT *pLeftStartOfSegment; INT *pRightStartOfSegment; UCHAR readDirection; UINT *pSegmentBitfield; UINT *pCodewordBitfield; UINT segmentOffset; UINT *iNode; UCHAR *pCntSign; FIXP_DBL *pResultBase; USHORT *iResultPointer; UINT codewordOffset; UCHAR carryBit; UINT iQSC; UINT cntSign; UINT dimCntr; UINT treeNode; SCHAR *pSta; UINT branchNode; UINT branchValue; const UINT *pCurrentTree; const SCHAR *pQuantValBase; const SCHAR *pQuantVal; pRemainingBitsInSegment = pHcr->segmentInfo.pRemainingBitsInSegment; pLeftStartOfSegment = pHcr->segmentInfo.pLeftStartOfSegment; pRightStartOfSegment = pHcr->segmentInfo.pRightStartOfSegment; readDirection = pHcr->segmentInfo.readDirection; pSegmentBitfield = pHcr->segmentInfo.pSegmentBitfield; pCodewordBitfield = pHcr->segmentInfo.pCodewordBitfield; segmentOffset = pHcr->segmentInfo.segmentOffset; iNode = pHcr->nonPcwSideinfo.iNode; pCntSign = pHcr->nonPcwSideinfo.pCntSign; pResultBase = pHcr->nonPcwSideinfo.pResultBase; iResultPointer = pHcr->nonPcwSideinfo.iResultPointer; codewordOffset = pHcr->nonPcwSideinfo.codewordOffset; pSta = pHcr->nonPcwSideinfo.pSta; treeNode = iNode[codewordOffset]; pCurrentTree = aHuffTable[ESCAPE_CODEBOOK]; for (; pRemainingBitsInSegment[segmentOffset] > 0; pRemainingBitsInSegment[segmentOffset] -= 1) { carryBit = HcrGetABitFromBitstream( bs, pHcr->decInOut.bitstreamAnchor, &pLeftStartOfSegment[segmentOffset], &pRightStartOfSegment[segmentOffset], readDirection); /* make a step in tree */ CarryBitToBranchValue(carryBit, treeNode, &branchValue, &branchNode); /* if end of branch reached write out lines and count bits needed for sign, * otherwise store node in codeword sideinfo */ if ((branchNode & TEST_BIT_10) == TEST_BIT_10) { /* test bit 10 ; if set body complete */ /* body completely decoded; branchValue is valid */ /* set pQuantVol to first (of two or four) quantized spectral coefficients */ pQuantValBase = aQuantTable[ESCAPE_CODEBOOK]; /* get base address of quantized values belonging to current codebook */ pQuantVal = pQuantValBase + branchValue; /* set pointer to first valid line [of 2 or 4 quantized values] */ /* make backup from original resultPointer in node storage for state * BODY_SIGN_ESC__SIGN */ iNode[codewordOffset] = iResultPointer[codewordOffset]; /* get position of first line for writing result */ iQSC = iResultPointer[codewordOffset]; /* codeword decoding result is written out here: Write out 2 or 4 * quantized spectral values with probably */ /* wrong sign and count number of values which are different from zero for * sign bit decoding [which happens in next state] */ cntSign = 0; for (dimCntr = DIMENSION_OF_ESCAPE_CODEBOOK; dimCntr != 0; dimCntr--) { pResultBase[iQSC++] = (FIXP_DBL)*pQuantVal; /* write quant. spec. coef. into spectrum */ if (*pQuantVal++ != 0) { cntSign += 1; } } if (cntSign == 0) { ClearBitFromBitfield( &(pHcr->nonPcwSideinfo.pState), segmentOffset, pCodewordBitfield); /* clear a bit in bitfield and switch off statemachine */ /* codeword decoded */ } else { /* write sign count result into codewordsideinfo of current codeword */ pCntSign[codewordOffset] = cntSign; pSta[codewordOffset] = BODY_SIGN_ESC__SIGN; /* change state */ pHcr->nonPcwSideinfo.pState = aStateConstant2State[pSta[codewordOffset]]; /* get state from separate array of cw-sideinfo */ } pRemainingBitsInSegment[segmentOffset] -= 1; /* the last reinitialzation of for loop counter (see above) is done here */ break; /* end of branch in tree reached i.e. a whole nonPCW-Body is decoded */ } else { /* body is not decoded completely: */ /* update treeNode for further step in decoding tree and store updated * treeNode because maybe no more bits left in segment */ treeNode = *(pCurrentTree + branchValue); iNode[codewordOffset] = treeNode; } } if (pRemainingBitsInSegment[segmentOffset] <= 0) { ClearBitFromBitfield(&(pHcr->nonPcwSideinfo.pState), segmentOffset, pSegmentBitfield); /* clear a bit in bitfield and switch off statemachine */ if (pRemainingBitsInSegment[segmentOffset] < 0) { pHcr->decInOut.errorLog |= STATE_ERROR_BODY_SIGN_ESC__BODY; return BODY_SIGN_ESC__BODY; } } return STOP_THIS_STATE; } /*--------------------------------------------------------------------------------------------- description: This state decodes the sign bits, if a codeword of codebook 11 needs some. A flag named 'flagB' in codeword sideinfo is set, if the second line of quantized spectral values is 16. The 'flagB' is used in case of decoding of a escape sequence is necessary as far as the second line is concerned. If only the first line needs an escape sequence, the flagB is cleared. If only the second line needs an escape sequence, the flagB is not used. For storing sideinfo in case of escape sequence decoding one single word can be used for both escape sequences because they are decoded not at the same time: bit 23 22 21 20 19 18 17 16 15 14 13 12 11 10 9 8 7 6 5 4 3 2 1 0 ===== == == =========== =========== =================================== ^ ^ ^ ^ ^ ^ | | | | | | res. flagA flagB escapePrefixUp escapePrefixDown escapeWord ----------------------------------------------------------------------------------------------- output: Two lines with correct sign. If one or two values is/are 16, the lines are not valid, otherwise they are. ----------------------------------------------------------------------------------------------- return: 0 -------------------------------------------------------------------------------------------- */ UINT Hcr_State_BODY_SIGN_ESC__SIGN(HANDLE_FDK_BITSTREAM bs, void *ptr) { H_HCR_INFO pHcr = (H_HCR_INFO)ptr; SCHAR *pRemainingBitsInSegment; INT *pLeftStartOfSegment; INT *pRightStartOfSegment; UCHAR readDirection; UINT *pSegmentBitfield; UINT *pCodewordBitfield; UINT segmentOffset; UINT *iNode; UCHAR *pCntSign; FIXP_DBL *pResultBase; USHORT *iResultPointer; UINT *pEscapeSequenceInfo; UINT codewordOffset; UINT iQSC; UCHAR cntSign; UINT flagA; UINT flagB; UINT flags; UCHAR carryBit; SCHAR *pSta; pRemainingBitsInSegment = pHcr->segmentInfo.pRemainingBitsInSegment; pLeftStartOfSegment = pHcr->segmentInfo.pLeftStartOfSegment; pRightStartOfSegment = pHcr->segmentInfo.pRightStartOfSegment; readDirection = pHcr->segmentInfo.readDirection; pSegmentBitfield = pHcr->segmentInfo.pSegmentBitfield; pCodewordBitfield = pHcr->segmentInfo.pCodewordBitfield; segmentOffset = pHcr->segmentInfo.segmentOffset; iNode = pHcr->nonPcwSideinfo.iNode; pCntSign = pHcr->nonPcwSideinfo.pCntSign; pResultBase = pHcr->nonPcwSideinfo.pResultBase; iResultPointer = pHcr->nonPcwSideinfo.iResultPointer; pEscapeSequenceInfo = pHcr->nonPcwSideinfo.pEscapeSequenceInfo; codewordOffset = pHcr->nonPcwSideinfo.codewordOffset; pSta = pHcr->nonPcwSideinfo.pSta; iQSC = iResultPointer[codewordOffset]; cntSign = pCntSign[codewordOffset]; /* loop for sign bit decoding */ for (; pRemainingBitsInSegment[segmentOffset] > 0; pRemainingBitsInSegment[segmentOffset] -= 1) { carryBit = HcrGetABitFromBitstream( bs, pHcr->decInOut.bitstreamAnchor, &pLeftStartOfSegment[segmentOffset], &pRightStartOfSegment[segmentOffset], readDirection); /* decrement sign counter because one sign bit has been read */ cntSign -= 1; pCntSign[codewordOffset] = cntSign; /* get a quantized spectral value (which was decoded in previous state) * which is not zero. [This value will get a sign] */ while (pResultBase[iQSC] == (FIXP_DBL)0) { if (++iQSC >= 1024) { return BODY_SIGN_ESC__SIGN; } } iResultPointer[codewordOffset] = iQSC; /* put negative sign together with quantized spectral value; if carryBit is * zero, the sign is ok already; no write operation necessary in this case */ if (carryBit != 0) { pResultBase[iQSC] = -pResultBase[iQSC]; /* carryBit = 1 --> minus */ } iQSC++; /* update index to next (maybe valid) value */ iResultPointer[codewordOffset] = iQSC; if (cntSign == 0) { /* all sign bits are decoded now */ pRemainingBitsInSegment[segmentOffset] -= 1; /* last reinitialzation of for loop counter (see above) is done here */ /* check decoded values if codeword is decoded: Check if one or two escape * sequences 16 follow */ /* step 0 */ /* restore pointer to first decoded quantized value [ = original * pResultPointr] from index iNode prepared in State_BODY_SIGN_ESC__BODY */ iQSC = iNode[codewordOffset]; /* step 1 */ /* test first value if escape sequence follows */ flagA = 0; /* for first possible escape sequence */ if (fixp_abs(pResultBase[iQSC++]) == (FIXP_DBL)ESCAPE_VALUE) { flagA = 1; } /* step 2 */ /* test second value if escape sequence follows */ flagB = 0; /* for second possible escape sequence */ if (fixp_abs(pResultBase[iQSC]) == (FIXP_DBL)ESCAPE_VALUE) { flagB = 1; } /* step 3 */ /* evaluate flag result and go on if necessary */ if (!flagA && !flagB) { ClearBitFromBitfield( &(pHcr->nonPcwSideinfo.pState), segmentOffset, pCodewordBitfield); /* clear a bit in bitfield and switch off statemachine */ } else { /* at least one of two lines is 16 */ /* store both flags at correct positions in non PCW codeword sideinfo * pEscapeSequenceInfo[codewordOffset] */ flags = flagA << POSITION_OF_FLAG_A; flags |= (flagB << POSITION_OF_FLAG_B); pEscapeSequenceInfo[codewordOffset] = flags; /* set next state */ pSta[codewordOffset] = BODY_SIGN_ESC__ESC_PREFIX; pHcr->nonPcwSideinfo.pState = aStateConstant2State[pSta[codewordOffset]]; /* get state from separate array of cw-sideinfo */ /* set result pointer to the first line of the two decoded lines */ iResultPointer[codewordOffset] = iNode[codewordOffset]; if (!flagA && flagB) { /* update pResultPointr ==> state Stat_BODY_SIGN_ESC__ESC_WORD writes * to correct position. Second value is the one and only escape value */ iQSC = iResultPointer[codewordOffset]; iQSC++; iResultPointer[codewordOffset] = iQSC; } } /* at least one of two lines is 16 */ break; /* nonPCW-Body at cb 11 and according sign bits are decoded */ } /* if ( cntSign == 0 ) */ } /* loop over remaining Bits in segment */ if (pRemainingBitsInSegment[segmentOffset] <= 0) { ClearBitFromBitfield(&(pHcr->nonPcwSideinfo.pState), segmentOffset, pSegmentBitfield); /* clear a bit in bitfield and switch off statemachine */ if (pRemainingBitsInSegment[segmentOffset] < 0) { pHcr->decInOut.errorLog |= STATE_ERROR_BODY_SIGN_ESC__SIGN; return BODY_SIGN_ESC__SIGN; } } return STOP_THIS_STATE; } /*--------------------------------------------------------------------------------------------- description: Decode escape prefix of first or second escape sequence. The escape prefix consists of ones. The following zero is also decoded here. ----------------------------------------------------------------------------------------------- output: If the single separator-zero which follows the escape-prefix-ones is not yet decoded: The value 'escapePrefixUp' in word pEscapeSequenceInfo[codewordOffset] is updated. If the single separator-zero which follows the escape-prefix-ones is decoded: Two updated values 'escapePrefixUp' and 'escapePrefixDown' in word pEscapeSequenceInfo[codewordOffset]. This State is finished. Switch to next state. ----------------------------------------------------------------------------------------------- return: 0 -------------------------------------------------------------------------------------------- */ UINT Hcr_State_BODY_SIGN_ESC__ESC_PREFIX(HANDLE_FDK_BITSTREAM bs, void *ptr) { H_HCR_INFO pHcr = (H_HCR_INFO)ptr; SCHAR *pRemainingBitsInSegment; INT *pLeftStartOfSegment; INT *pRightStartOfSegment; UCHAR readDirection; UINT *pSegmentBitfield; UINT segmentOffset; UINT *pEscapeSequenceInfo; UINT codewordOffset; UCHAR carryBit; UINT escapePrefixUp; SCHAR *pSta; pRemainingBitsInSegment = pHcr->segmentInfo.pRemainingBitsInSegment; pLeftStartOfSegment = pHcr->segmentInfo.pLeftStartOfSegment; pRightStartOfSegment = pHcr->segmentInfo.pRightStartOfSegment; readDirection = pHcr->segmentInfo.readDirection; pSegmentBitfield = pHcr->segmentInfo.pSegmentBitfield; segmentOffset = pHcr->segmentInfo.segmentOffset; pEscapeSequenceInfo = pHcr->nonPcwSideinfo.pEscapeSequenceInfo; codewordOffset = pHcr->nonPcwSideinfo.codewordOffset; pSta = pHcr->nonPcwSideinfo.pSta; escapePrefixUp = (pEscapeSequenceInfo[codewordOffset] & MASK_ESCAPE_PREFIX_UP) >> LSB_ESCAPE_PREFIX_UP; /* decode escape prefix */ for (; pRemainingBitsInSegment[segmentOffset] > 0; pRemainingBitsInSegment[segmentOffset] -= 1) { carryBit = HcrGetABitFromBitstream( bs, pHcr->decInOut.bitstreamAnchor, &pLeftStartOfSegment[segmentOffset], &pRightStartOfSegment[segmentOffset], readDirection); /* count ones and store sum in escapePrefixUp */ if (carryBit == 1) { escapePrefixUp += 1; /* update conter for ones */ if (escapePrefixUp > 8) { pHcr->decInOut.errorLog |= STATE_ERROR_BODY_SIGN_ESC__ESC_PREFIX; return BODY_SIGN_ESC__ESC_PREFIX; } /* store updated counter in sideinfo of current codeword */ pEscapeSequenceInfo[codewordOffset] &= ~MASK_ESCAPE_PREFIX_UP; /* delete old escapePrefixUp */ escapePrefixUp <<= LSB_ESCAPE_PREFIX_UP; /* shift to correct position */ pEscapeSequenceInfo[codewordOffset] |= escapePrefixUp; /* insert new escapePrefixUp */ escapePrefixUp >>= LSB_ESCAPE_PREFIX_UP; /* shift back down */ } else { /* separator [zero] reached */ pRemainingBitsInSegment[segmentOffset] -= 1; /* last reinitialzation of for loop counter (see above) is done here */ escapePrefixUp += 4; /* if escape_separator '0' appears, add 4 and ==> break */ /* store escapePrefixUp in pEscapeSequenceInfo[codewordOffset] at bit * position escapePrefixUp */ pEscapeSequenceInfo[codewordOffset] &= ~MASK_ESCAPE_PREFIX_UP; /* delete old escapePrefixUp */ escapePrefixUp <<= LSB_ESCAPE_PREFIX_UP; /* shift to correct position */ pEscapeSequenceInfo[codewordOffset] |= escapePrefixUp; /* insert new escapePrefixUp */ escapePrefixUp >>= LSB_ESCAPE_PREFIX_UP; /* shift back down */ /* store escapePrefixUp in pEscapeSequenceInfo[codewordOffset] at bit * position escapePrefixDown */ pEscapeSequenceInfo[codewordOffset] &= ~MASK_ESCAPE_PREFIX_DOWN; /* delete old escapePrefixDown */ escapePrefixUp <<= LSB_ESCAPE_PREFIX_DOWN; /* shift to correct position */ pEscapeSequenceInfo[codewordOffset] |= escapePrefixUp; /* insert new escapePrefixDown */ pSta[codewordOffset] = BODY_SIGN_ESC__ESC_WORD; /* set next state */ pHcr->nonPcwSideinfo.pState = aStateConstant2State[pSta[codewordOffset]]; /* get state from separate array of cw-sideinfo */ break; } } if (pRemainingBitsInSegment[segmentOffset] <= 0) { ClearBitFromBitfield(&(pHcr->nonPcwSideinfo.pState), segmentOffset, pSegmentBitfield); /* clear a bit in bitfield and switch off statemachine */ if (pRemainingBitsInSegment[segmentOffset] < 0) { pHcr->decInOut.errorLog |= STATE_ERROR_BODY_SIGN_ESC__ESC_PREFIX; return BODY_SIGN_ESC__ESC_PREFIX; } } return STOP_THIS_STATE; } /*--------------------------------------------------------------------------------------------- description: Decode escapeWord of escape sequence. If the escape sequence is decoded completely, assemble quantized-spectral-escape-coefficient and replace the previous decoded 16 by the new value. Test flagB. If flagB is set, the second escape sequence must be decoded. If flagB is not set, the codeword is decoded and the state machine is switched off. ----------------------------------------------------------------------------------------------- output: Two lines with valid sign. At least one of both lines has got the correct value. ----------------------------------------------------------------------------------------------- return: 0 -------------------------------------------------------------------------------------------- */ UINT Hcr_State_BODY_SIGN_ESC__ESC_WORD(HANDLE_FDK_BITSTREAM bs, void *ptr) { H_HCR_INFO pHcr = (H_HCR_INFO)ptr; SCHAR *pRemainingBitsInSegment; INT *pLeftStartOfSegment; INT *pRightStartOfSegment; UCHAR readDirection; UINT *pSegmentBitfield; UINT *pCodewordBitfield; UINT segmentOffset; FIXP_DBL *pResultBase; USHORT *iResultPointer; UINT *pEscapeSequenceInfo; UINT codewordOffset; UINT escapeWord; UINT escapePrefixDown; UINT escapePrefixUp; UCHAR carryBit; UINT iQSC; INT sign; UINT flagA; UINT flagB; SCHAR *pSta; pRemainingBitsInSegment = pHcr->segmentInfo.pRemainingBitsInSegment; pLeftStartOfSegment = pHcr->segmentInfo.pLeftStartOfSegment; pRightStartOfSegment = pHcr->segmentInfo.pRightStartOfSegment; readDirection = pHcr->segmentInfo.readDirection; pSegmentBitfield = pHcr->segmentInfo.pSegmentBitfield; pCodewordBitfield = pHcr->segmentInfo.pCodewordBitfield; segmentOffset = pHcr->segmentInfo.segmentOffset; pResultBase = pHcr->nonPcwSideinfo.pResultBase; iResultPointer = pHcr->nonPcwSideinfo.iResultPointer; pEscapeSequenceInfo = pHcr->nonPcwSideinfo.pEscapeSequenceInfo; codewordOffset = pHcr->nonPcwSideinfo.codewordOffset; pSta = pHcr->nonPcwSideinfo.pSta; escapeWord = pEscapeSequenceInfo[codewordOffset] & MASK_ESCAPE_WORD; escapePrefixDown = (pEscapeSequenceInfo[codewordOffset] & MASK_ESCAPE_PREFIX_DOWN) >> LSB_ESCAPE_PREFIX_DOWN; /* decode escape word */ for (; pRemainingBitsInSegment[segmentOffset] > 0; pRemainingBitsInSegment[segmentOffset] -= 1) { carryBit = HcrGetABitFromBitstream( bs, pHcr->decInOut.bitstreamAnchor, &pLeftStartOfSegment[segmentOffset], &pRightStartOfSegment[segmentOffset], readDirection); /* build escape word */ escapeWord <<= 1; /* left shift previous decoded part of escapeWord by on bit */ escapeWord = escapeWord | carryBit; /* assemble escape word by bitwise or */ /* decrement counter for length of escape word because one more bit was * decoded */ escapePrefixDown -= 1; /* store updated escapePrefixDown */ pEscapeSequenceInfo[codewordOffset] &= ~MASK_ESCAPE_PREFIX_DOWN; /* delete old escapePrefixDown */ escapePrefixDown <<= LSB_ESCAPE_PREFIX_DOWN; /* shift to correct position */ pEscapeSequenceInfo[codewordOffset] |= escapePrefixDown; /* insert new escapePrefixDown */ escapePrefixDown >>= LSB_ESCAPE_PREFIX_DOWN; /* shift back */ /* store updated escapeWord */ pEscapeSequenceInfo[codewordOffset] &= ~MASK_ESCAPE_WORD; /* delete old escapeWord */ pEscapeSequenceInfo[codewordOffset] |= escapeWord; /* insert new escapeWord */ if (escapePrefixDown == 0) { pRemainingBitsInSegment[segmentOffset] -= 1; /* last reinitialzation of for loop counter (see above) is done here */ /* escape sequence decoded. Assemble escape-line and replace original line */ /* step 0 */ /* derive sign */ iQSC = iResultPointer[codewordOffset]; sign = (pResultBase[iQSC] >= (FIXP_DBL)0) ? 1 : -1; /* get sign of escape value 16 */ /* step 1 */ /* get escapePrefixUp */ escapePrefixUp = (pEscapeSequenceInfo[codewordOffset] & MASK_ESCAPE_PREFIX_UP) >> LSB_ESCAPE_PREFIX_UP; /* step 2 */ /* calculate escape value */ pResultBase[iQSC] = (FIXP_DBL)(sign * (((INT)1 << escapePrefixUp) + (INT)escapeWord)); /* get both flags from sideinfo (flags are not shifted to the * lsb-position) */ flagA = pEscapeSequenceInfo[codewordOffset] & MASK_FLAG_A; flagB = pEscapeSequenceInfo[codewordOffset] & MASK_FLAG_B; /* step 3 */ /* clear the whole escape sideinfo word */ pEscapeSequenceInfo[codewordOffset] = 0; /* change state in dependence of flag flagB */ if (flagA != 0) { /* first escape sequence decoded; previous decoded 16 has been replaced * by valid line */ /* clear flagA in sideinfo word because this escape sequence has already * beed decoded */ pEscapeSequenceInfo[codewordOffset] &= ~MASK_FLAG_A; if (flagB == 0) { ClearBitFromBitfield(&(pHcr->nonPcwSideinfo.pState), segmentOffset, pCodewordBitfield); /* clear a bit in bitfield and switch off statemachine */ } else { /* updated pointer to next and last 16 */ iQSC++; iResultPointer[codewordOffset] = iQSC; /* change state */ pSta[codewordOffset] = BODY_SIGN_ESC__ESC_PREFIX; pHcr->nonPcwSideinfo.pState = aStateConstant2State[pSta[codewordOffset]]; /* get state from separate array of cw-sideinfo */ } } else { ClearBitFromBitfield( &(pHcr->nonPcwSideinfo.pState), segmentOffset, pCodewordBitfield); /* clear a bit in bitfield and switch off statemachine */ } break; } } if (pRemainingBitsInSegment[segmentOffset] <= 0) { ClearBitFromBitfield(&(pHcr->nonPcwSideinfo.pState), segmentOffset, pSegmentBitfield); /* clear a bit in bitfield and switch off statemachine */ if (pRemainingBitsInSegment[segmentOffset] < 0) { pHcr->decInOut.errorLog |= STATE_ERROR_BODY_SIGN_ESC__ESC_WORD; return BODY_SIGN_ESC__ESC_WORD; } } return STOP_THIS_STATE; } fdk-aac-2.0.2/libAACdec/src/aacdec_hcrs.h000066400000000000000000000202321404225310700177040ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Robert Weidner (DSP Solutions) Description: HCR Decoder: Defines of state-constants, masks and state-prototypes *******************************************************************************/ #ifndef AACDEC_HCRS_H #define AACDEC_HCRS_H #include "FDK_bitstream.h" #include "aacdec_hcr_types.h" /* The four different kinds of types of states are: */ /* different states are defined as constants */ /* start middle=self next stop */ #define STOP_THIS_STATE \ 0 /* */ #define BODY_ONLY \ 1 /* X X X */ #define BODY_SIGN__BODY \ 2 /* X X X X [stop if no sign] */ #define BODY_SIGN__SIGN \ 3 /* X X [stop if sign bits decoded] */ #define BODY_SIGN_ESC__BODY \ 4 /* X X X X [stop if no sign] */ #define BODY_SIGN_ESC__SIGN \ 5 /* X X X [stop if no escape sequence] */ #define BODY_SIGN_ESC__ESC_PREFIX \ 6 /* X X */ #define BODY_SIGN_ESC__ESC_WORD \ 7 /* X X X [stop if abs(second qsc) != 16] */ /* examples: */ /* BODY_ONLY means only the codeword body will be decoded; no * sign bits will follow and no escapesequence will follow */ /* BODY_SIGN__BODY means that the codeword consists of two parts; * body and sign part. The part '__BODY' after the two underscores shows */ /* that the bits which are currently decoded belong * to the '__BODY' of the codeword and not to the sign part. */ /* BODY_SIGN_ESC__ESC_PB means that the codeword consists of three parts; * body, sign and (here: two) escape sequences; */ /* P = Prefix = ones */ /* W = Escape Word */ /* A = first possible (of two) Escape sequeces */ /* B = second possible (of two) Escape sequeces */ /* The part after the two underscores shows that * the current bits which are decoded belong to the '__ESC_PB' - part of the */ /* codeword. That means the body and the sign bits * are decoded completely and the bits which are decoded now belong to */ /* the escape sequence [P = prefix; B=second * possible escape sequence] */ #define MSB_31_MASK 0x80000000 /* masks MSB (= Bit 31) in a 32 bit word */ #define DIMENSION_OF_ESCAPE_CODEBOOK 2 /* for cb >= 11 is dimension 2 */ #define ESCAPE_CODEBOOK 11 #define MASK_ESCAPE_PREFIX_UP 0x000F0000 #define LSB_ESCAPE_PREFIX_UP 16 #define MASK_ESCAPE_PREFIX_DOWN 0x0000F000 #define LSB_ESCAPE_PREFIX_DOWN 12 #define MASK_ESCAPE_WORD 0x00000FFF #define MASK_FLAG_A 0x00200000 #define MASK_FLAG_B 0x00100000 extern void DecodeNonPCWs(HANDLE_FDK_BITSTREAM bs, H_HCR_INFO hHcr); UINT Hcr_State_BODY_ONLY(HANDLE_FDK_BITSTREAM, void*); UINT Hcr_State_BODY_SIGN__BODY(HANDLE_FDK_BITSTREAM, void*); UINT Hcr_State_BODY_SIGN__SIGN(HANDLE_FDK_BITSTREAM, void*); UINT Hcr_State_BODY_SIGN_ESC__BODY(HANDLE_FDK_BITSTREAM, void*); UINT Hcr_State_BODY_SIGN_ESC__SIGN(HANDLE_FDK_BITSTREAM, void*); UINT Hcr_State_BODY_SIGN_ESC__ESC_PREFIX(HANDLE_FDK_BITSTREAM, void*); UINT Hcr_State_BODY_SIGN_ESC__ESC_WORD(HANDLE_FDK_BITSTREAM, void*); #endif /* AACDEC_HCRS_H */ fdk-aac-2.0.2/libAACdec/src/aacdec_pns.cpp000066400000000000000000000312741404225310700201100ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Josef Hoepfl Description: perceptual noise substitution tool *******************************************************************************/ #include "aacdec_pns.h" #include "aac_ram.h" #include "aac_rom.h" #include "channelinfo.h" #include "block.h" #include "FDK_bitstream.h" #include "genericStds.h" #define NOISE_OFFSET 90 /* cf. ISO 14496-3 p. 175 */ /*! \brief Reset InterChannel and PNS data The function resets the InterChannel and PNS data */ void CPns_ResetData(CPnsData *pPnsData, CPnsInterChannelData *pPnsInterChannelData) { FDK_ASSERT(pPnsData != NULL); FDK_ASSERT(pPnsInterChannelData != NULL); /* Assign pointer always, since pPnsData is not persistent data */ pPnsData->pPnsInterChannelData = pPnsInterChannelData; pPnsData->PnsActive = 0; pPnsData->CurrentEnergy = 0; FDKmemclear(pPnsData->pnsUsed, (8 * 16) * sizeof(UCHAR)); FDKmemclear(pPnsInterChannelData->correlated, (8 * 16) * sizeof(UCHAR)); } /*! \brief Update PNS noise generator state. The function sets the seed for PNS noise generation. It can be used to link two or more channels in terms of PNS. */ void CPns_UpdateNoiseState(CPnsData *pPnsData, INT *currentSeed, INT *randomSeed) { /* use pointer because seed has to be same, left and right channel ! */ pPnsData->currentSeed = currentSeed; pPnsData->randomSeed = randomSeed; } /*! \brief Indicates if PNS is used The function returns a value indicating whether PNS is used or not acordding to the noise energy \return PNS used */ int CPns_IsPnsUsed(const CPnsData *pPnsData, const int group, const int band) { unsigned pns_band = group * 16 + band; return pPnsData->pnsUsed[pns_band] & (UCHAR)1; } /*! \brief Set correlation The function activates the noise correlation between the channel pair */ void CPns_SetCorrelation(CPnsData *pPnsData, const int group, const int band, const int outofphase) { CPnsInterChannelData *pInterChannelData = pPnsData->pPnsInterChannelData; unsigned pns_band = group * 16 + band; pInterChannelData->correlated[pns_band] = (outofphase) ? 3 : 1; } /*! \brief Indicates if correlation is used The function indicates if the noise correlation between the channel pair is activated \return PNS is correlated */ static int CPns_IsCorrelated(const CPnsData *pPnsData, const int group, const int band) { CPnsInterChannelData *pInterChannelData = pPnsData->pPnsInterChannelData; unsigned pns_band = group * 16 + band; return (pInterChannelData->correlated[pns_band] & 0x01) ? 1 : 0; } /*! \brief Indicates if correlated out of phase mode is used. The function indicates if the noise correlation between the channel pair is activated in out-of-phase mode. \return PNS is out-of-phase */ static int CPns_IsOutOfPhase(const CPnsData *pPnsData, const int group, const int band) { CPnsInterChannelData *pInterChannelData = pPnsData->pPnsInterChannelData; unsigned pns_band = group * 16 + band; return (pInterChannelData->correlated[pns_band] & 0x02) ? 1 : 0; } /*! \brief Read PNS information The function reads the PNS information from the bitstream */ void CPns_Read(CPnsData *pPnsData, HANDLE_FDK_BITSTREAM bs, const CodeBookDescription *hcb, SHORT *pScaleFactor, UCHAR global_gain, int band, int group /* = 0 */) { int delta; UINT pns_band = group * 16 + band; if (pPnsData->PnsActive) { /* Next PNS band case */ delta = CBlock_DecodeHuffmanWord(bs, hcb) - 60; } else { /* First PNS band case */ int noiseStartValue = FDKreadBits(bs, 9); delta = noiseStartValue - 256; pPnsData->PnsActive = 1; pPnsData->CurrentEnergy = global_gain - NOISE_OFFSET; } pPnsData->CurrentEnergy += delta; pScaleFactor[pns_band] = pPnsData->CurrentEnergy; pPnsData->pnsUsed[pns_band] = 1; } /** * \brief Generate a vector of noise of given length. The noise values are * scaled in order to yield a noise energy of 1.0 * \param spec pointer to were the noise values will be written to. * \param size amount of noise values to be generated. * \param pRandomState pointer to the state of the random generator being used. * \return exponent of generated noise vector. */ static int GenerateRandomVector(FIXP_DBL *RESTRICT spec, int size, int *pRandomState) { int i, invNrg_e = 0, nrg_e = 0; FIXP_DBL invNrg_m, nrg_m = FL2FXCONST_DBL(0.0f); FIXP_DBL *RESTRICT ptr = spec; int randomState = *pRandomState; #define GEN_NOISE_NRG_SCALE 7 /* Generate noise and calculate energy. */ for (i = 0; i < size; i++) { randomState = (((INT64)1664525 * randomState) + (INT64)1013904223) & 0xFFFFFFFF; nrg_m = fPow2AddDiv2(nrg_m, (FIXP_DBL)randomState >> GEN_NOISE_NRG_SCALE); *ptr++ = (FIXP_DBL)randomState; } nrg_e = GEN_NOISE_NRG_SCALE * 2 + 1; /* weight noise with = 1 / sqrt_nrg; */ invNrg_m = invSqrtNorm2(nrg_m << 1, &invNrg_e); invNrg_e += -((nrg_e - 1) >> 1); for (i = size; i--;) { spec[i] = fMult(spec[i], invNrg_m); } /* Store random state */ *pRandomState = randomState; return invNrg_e; } static void ScaleBand(FIXP_DBL *RESTRICT spec, int size, int scaleFactor, int specScale, int noise_e, int out_of_phase) { int i, shift, sfExponent; FIXP_DBL sfMatissa; /* Get gain from scale factor value = 2^(scaleFactor * 0.25) */ sfMatissa = MantissaTable[scaleFactor & 0x03][0]; /* sfExponent = (scaleFactor >> 2) + ExponentTable[scaleFactor & 0x03][0]; */ /* Note: ExponentTable[scaleFactor & 0x03][0] is always 1. */ sfExponent = (scaleFactor >> 2) + 1; if (out_of_phase != 0) { sfMatissa = -sfMatissa; } /* +1 because of fMultDiv2 below. */ shift = sfExponent - specScale + 1 + noise_e; /* Apply gain to noise values */ if (shift >= 0) { shift = fixMin(shift, DFRACT_BITS - 1); for (i = size; i-- != 0;) { spec[i] = fMultDiv2(spec[i], sfMatissa) << shift; } } else { shift = fixMin(-shift, DFRACT_BITS - 1); for (i = size; i-- != 0;) { spec[i] = fMultDiv2(spec[i], sfMatissa) >> shift; } } } /*! \brief Apply PNS The function applies PNS (i.e. it generates noise) on the bands flagged as noisy bands */ void CPns_Apply(const CPnsData *pPnsData, const CIcsInfo *pIcsInfo, SPECTRAL_PTR pSpectrum, const SHORT *pSpecScale, const SHORT *pScaleFactor, const SamplingRateInfo *pSamplingRateInfo, const INT granuleLength, const int channel) { if (pPnsData->PnsActive) { const short *BandOffsets = GetScaleFactorBandOffsets(pIcsInfo, pSamplingRateInfo); int ScaleFactorBandsTransmitted = GetScaleFactorBandsTransmitted(pIcsInfo); for (int window = 0, group = 0; group < GetWindowGroups(pIcsInfo); group++) { for (int groupwin = 0; groupwin < GetWindowGroupLength(pIcsInfo, group); groupwin++, window++) { FIXP_DBL *spectrum = SPEC(pSpectrum, window, granuleLength); for (int band = 0; band < ScaleFactorBandsTransmitted; band++) { if (CPns_IsPnsUsed(pPnsData, group, band)) { UINT pns_band = window * 16 + band; int bandWidth = BandOffsets[band + 1] - BandOffsets[band]; int noise_e; FDK_ASSERT(bandWidth >= 0); if (channel > 0 && CPns_IsCorrelated(pPnsData, group, band)) { noise_e = GenerateRandomVector(spectrum + BandOffsets[band], bandWidth, &pPnsData->randomSeed[pns_band]); } else { pPnsData->randomSeed[pns_band] = *pPnsData->currentSeed; noise_e = GenerateRandomVector(spectrum + BandOffsets[band], bandWidth, pPnsData->currentSeed); } int outOfPhase = CPns_IsOutOfPhase(pPnsData, group, band); ScaleBand(spectrum + BandOffsets[band], bandWidth, pScaleFactor[group * 16 + band], pSpecScale[window], noise_e, outOfPhase); } } } } } } fdk-aac-2.0.2/libAACdec/src/aacdec_pns.h000066400000000000000000000125271404225310700175550ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Josef Hoepfl Description: perceptual noise substitution tool *******************************************************************************/ #ifndef AACDEC_PNS_H #define AACDEC_PNS_H #include "common_fix.h" #define NO_OFBANDS ((8 * 16)) typedef struct { UCHAR correlated[NO_OFBANDS]; } CPnsInterChannelData; typedef struct { CPnsInterChannelData *pPnsInterChannelData; UCHAR pnsUsed[NO_OFBANDS]; int CurrentEnergy; UCHAR PnsActive; INT *currentSeed; INT *randomSeed; } CPnsData; void CPns_UpdateNoiseState(CPnsData *pPnsData, INT *currentSeed, INT *randomSeed); void CPns_ResetData(CPnsData *pPnsData, CPnsInterChannelData *pPnsInterChannelData); #endif /* #ifndef AACDEC_PNS_H */ fdk-aac-2.0.2/libAACdec/src/aacdec_tns.cpp000066400000000000000000000303311404225310700201050ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Josef Hoepfl Description: temporal noise shaping tool *******************************************************************************/ #include "aacdec_tns.h" #include "aac_rom.h" #include "FDK_bitstream.h" #include "channelinfo.h" #include "FDK_lpc.h" #define TNS_MAXIMUM_ORDER_AAC 12 /*! \brief Reset tns data The function resets the tns data \return none */ void CTns_Reset(CTnsData *pTnsData) { /* Note: the following FDKmemclear should not be required. */ FDKmemclear(pTnsData->Filter, TNS_MAX_WINDOWS * TNS_MAXIMUM_FILTERS * sizeof(CFilter)); FDKmemclear(pTnsData->NumberOfFilters, TNS_MAX_WINDOWS * sizeof(UCHAR)); pTnsData->DataPresent = 0; pTnsData->Active = 0; } void CTns_ReadDataPresentFlag( HANDLE_FDK_BITSTREAM bs, /*!< pointer to bitstream */ CTnsData *pTnsData) /*!< pointer to aac decoder channel info */ { pTnsData->DataPresent = (UCHAR)FDKreadBits(bs, 1); } /*! \brief Read tns data from bitstream The function reads the elements for tns from the bitstream. \return none */ AAC_DECODER_ERROR CTns_Read(HANDLE_FDK_BITSTREAM bs, CTnsData *pTnsData, const CIcsInfo *pIcsInfo, const UINT flags) { UCHAR n_filt, order; UCHAR length, coef_res, coef_compress; UCHAR window; UCHAR wins_per_frame; UCHAR isLongFlag; UCHAR start_window; AAC_DECODER_ERROR ErrorStatus = AAC_DEC_OK; if (!pTnsData->DataPresent) { return ErrorStatus; } { start_window = 0; wins_per_frame = GetWindowsPerFrame(pIcsInfo); isLongFlag = IsLongBlock(pIcsInfo); } pTnsData->GainLd = 0; for (window = start_window; window < wins_per_frame; window++) { pTnsData->NumberOfFilters[window] = n_filt = (UCHAR)FDKreadBits(bs, isLongFlag ? 2 : 1); if (n_filt) { int index; UCHAR nextstopband; coef_res = (UCHAR)FDKreadBits(bs, 1); nextstopband = GetScaleFactorBandsTotal(pIcsInfo); for (index = 0; index < n_filt; index++) { CFilter *filter = &pTnsData->Filter[window][index]; length = (UCHAR)FDKreadBits(bs, isLongFlag ? 6 : 4); if (length > nextstopband) { length = nextstopband; } filter->StartBand = nextstopband - length; filter->StopBand = nextstopband; nextstopband = filter->StartBand; if (flags & (AC_USAC | AC_RSVD50 | AC_RSV603DA)) { /* max(Order) = 15 (long), 7 (short) */ filter->Order = order = (UCHAR)FDKreadBits(bs, isLongFlag ? 4 : 3); } else { filter->Order = order = (UCHAR)FDKreadBits(bs, isLongFlag ? 5 : 3); if (filter->Order > TNS_MAXIMUM_ORDER) { ErrorStatus = AAC_DEC_TNS_READ_ERROR; return ErrorStatus; } } FDK_ASSERT(order <= TNS_MAXIMUM_ORDER); /* avoid illegal memory access */ if (order) { UCHAR coef, s_mask; UCHAR i; SCHAR n_mask; static const UCHAR sgn_mask[] = {0x2, 0x4, 0x8}; static const SCHAR neg_mask[] = {~0x3, ~0x7, ~0xF}; filter->Direction = FDKreadBits(bs, 1) ? -1 : 1; coef_compress = (UCHAR)FDKreadBits(bs, 1); filter->Resolution = coef_res + 3; s_mask = sgn_mask[coef_res + 1 - coef_compress]; n_mask = neg_mask[coef_res + 1 - coef_compress]; for (i = 0; i < order; i++) { coef = (UCHAR)FDKreadBits(bs, filter->Resolution - coef_compress); filter->Coeff[i] = (coef & s_mask) ? (coef | n_mask) : coef; } pTnsData->GainLd = 4; } } } } pTnsData->Active = 1; return ErrorStatus; } void CTns_ReadDataPresentUsac(HANDLE_FDK_BITSTREAM hBs, CTnsData *pTnsData0, CTnsData *pTnsData1, UCHAR *ptns_on_lr, const CIcsInfo *pIcsInfo, const UINT flags, const UINT elFlags, const int fCommonWindow) { int common_tns = 0; if (fCommonWindow) { common_tns = FDKreadBit(hBs); } { *ptns_on_lr = FDKreadBit(hBs); } if (common_tns) { pTnsData0->DataPresent = 1; CTns_Read(hBs, pTnsData0, pIcsInfo, flags); pTnsData0->DataPresent = 0; pTnsData0->Active = 1; *pTnsData1 = *pTnsData0; } else { int tns_present_both; tns_present_both = FDKreadBit(hBs); if (tns_present_both) { pTnsData0->DataPresent = 1; pTnsData1->DataPresent = 1; } else { pTnsData1->DataPresent = FDKreadBit(hBs); pTnsData0->DataPresent = !pTnsData1->DataPresent; } } } /*! \brief Apply tns to spectral lines The function applies the tns to the spectrum, \return none */ void CTns_Apply(CTnsData *RESTRICT pTnsData, /*!< pointer to aac decoder info */ const CIcsInfo *pIcsInfo, SPECTRAL_PTR pSpectralCoefficient, const SamplingRateInfo *pSamplingRateInfo, const INT granuleLength, const UCHAR nbands, const UCHAR igf_active, const UINT flags) { int window, index, start, stop, size, start_window, wins_per_frame; if (pTnsData->Active) { C_AALLOC_SCRATCH_START(coeff, FIXP_TCC, TNS_MAXIMUM_ORDER) { start_window = 0; wins_per_frame = GetWindowsPerFrame(pIcsInfo); } for (window = start_window; window < wins_per_frame; window++) { FIXP_DBL *pSpectrum; { pSpectrum = SPEC(pSpectralCoefficient, window, granuleLength); } for (index = 0; index < pTnsData->NumberOfFilters[window]; index++) { CFilter *filter = &pTnsData->Filter[window][index]; if (filter->Order > 0) { FIXP_TCC *pCoeff; UCHAR tns_max_bands; pCoeff = coeff; if (filter->Resolution == 3) { int i; for (i = 0; i < filter->Order; i++) *pCoeff++ = FDKaacDec_tnsCoeff3[filter->Coeff[i] + 4]; } else { int i; for (i = 0; i < filter->Order; i++) *pCoeff++ = FDKaacDec_tnsCoeff4[filter->Coeff[i] + 8]; } switch (granuleLength) { case 480: tns_max_bands = tns_max_bands_tbl_480[pSamplingRateInfo->samplingRateIndex]; break; case 512: tns_max_bands = tns_max_bands_tbl_512[pSamplingRateInfo->samplingRateIndex]; break; default: tns_max_bands = GetMaximumTnsBands( pIcsInfo, pSamplingRateInfo->samplingRateIndex); /* See redefinition of TNS_MAX_BANDS table */ if ((flags & (AC_USAC | AC_RSVD50 | AC_RSV603DA)) && (pSamplingRateInfo->samplingRateIndex > 5)) { tns_max_bands += 1; } break; } start = fixMin(fixMin(filter->StartBand, tns_max_bands), nbands); start = GetScaleFactorBandOffsets(pIcsInfo, pSamplingRateInfo)[start]; if (igf_active) { stop = fixMin(filter->StopBand, nbands); } else { stop = fixMin(fixMin(filter->StopBand, tns_max_bands), nbands); } stop = GetScaleFactorBandOffsets(pIcsInfo, pSamplingRateInfo)[stop]; size = stop - start; if (size) { C_ALLOC_SCRATCH_START(state, FIXP_DBL, TNS_MAXIMUM_ORDER) FDKmemclear(state, TNS_MAXIMUM_ORDER * sizeof(FIXP_DBL)); CLpc_SynthesisLattice(pSpectrum + start, size, 0, 0, filter->Direction, coeff, filter->Order, state); C_ALLOC_SCRATCH_END(state, FIXP_DBL, TNS_MAXIMUM_ORDER) } } } } C_AALLOC_SCRATCH_END(coeff, FIXP_TCC, TNS_MAXIMUM_ORDER) } } fdk-aac-2.0.2/libAACdec/src/aacdec_tns.h000066400000000000000000000134031404225310700175530ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Josef Hoepfl Description: temporal noise shaping tool *******************************************************************************/ #ifndef AACDEC_TNS_H #define AACDEC_TNS_H #include "common_fix.h" enum { TNS_MAX_WINDOWS = 8, /* 8 */ TNS_MAXIMUM_FILTERS = 3 }; /* TNS_MAXIMUM_ORDER (for memory allocation) 12 for AAC-LC and AAC-SSR. Set to 20 for AAC-Main (AOT 1). Some broken encoders also do order 20 for AAC-LC :( 15 for USAC (AOT 42) */ #define TNS_MAXIMUM_ORDER (20) #if (TNS_MAXIMUM_ORDER < 15) #error USAC: TNS filter order up 15 can be signaled! #endif typedef struct { SCHAR Coeff[TNS_MAXIMUM_ORDER]; UCHAR StartBand; UCHAR StopBand; SCHAR Direction; SCHAR Resolution; UCHAR Order; } CFilter; typedef struct { CFilter Filter[TNS_MAX_WINDOWS][TNS_MAXIMUM_FILTERS]; UCHAR NumberOfFilters[TNS_MAX_WINDOWS]; UCHAR DataPresent; UCHAR Active; /* log2 of the maximum total filter gains. The value is required to keep necessary mantissa headroom so that while applying the TNS predictor the mantissas do not overflow. */ UCHAR GainLd; } CTnsData; void CTns_Reset(CTnsData *pTnsData); #endif /* #ifndef AACDEC_TNS_H */ fdk-aac-2.0.2/libAACdec/src/aacdecoder.cpp000066400000000000000000003552271404225310700201110ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Josef Hoepfl Description: *******************************************************************************/ /*! \page default General Overview of the AAC Decoder Implementation The main entry point to decode a AAC frame is CAacDecoder_DecodeFrame(). It handles the different transport multiplexes and bitstream formats supported by this implementation. It extracts the AAC_raw_data_blocks from these bitstreams to further process then in the actual decoding stages. Note: Click on a function of file in the above image to see details about the function. Also note, that this is just an overview of the most important functions and not a complete call graph.

1 Bitstream deformatter

The basic bit stream parser function CChannelElement_Read() is called. It uses other subcalls in order to parse and unpack the bitstreams. Note, that this includes huffmann decoding of the coded spectral data. This operation can be computational significant specifically at higher bitrates. Optimization is likely in CBlock_ReadSpectralData(). The bitstream deformatter also includes many bitfield operations. Profiling on the target will determine required optimizations.

2 Actual decoding to retain the time domain output

The basic bitstream deformatter function CChannelElement_Decode() for CPE elements and SCE elements are called. Except for the stereo processing (2.1) which is only used for CPE elements, the function calls for CPE or SCE are similar, except that CPE always processes to independent channels while SCE only processes one channel. Often there is the distinction between long blocks and short blocks. However, computational expensive functions that ususally require optimization are being shared by these two groups,

2.1 Stereo processing for CPE elements

CChannelPairElement_Decode() first calles the joint stereo tools in stereo.cpp when required.

2.2 Scaling of spectral data

CBlock_ScaleSpectralData().

2.3 Apply additional coding tools

ApplyTools() calles the PNS tools in case of MPEG-4 bitstreams, and TNS filtering CTns_Apply() for MPEG-2 and MPEG-4 bitstreams. The function TnsFilterIIR() which is called by CTns_Apply() (2.3.1) might require some optimization.

3 Frequency-To-Time conversion

The filterbank is called using CBlock_FrequencyToTime() using the MDCT module from the FDK Tools */ #include "aacdecoder.h" #include "aac_rom.h" #include "aac_ram.h" #include "channel.h" #include "FDK_audio.h" #include "aacdec_pns.h" #include "sbrdecoder.h" #include "sac_dec_lib.h" #include "aacdec_hcr.h" #include "rvlc.h" #include "usacdec_lpd.h" #include "ac_arith_coder.h" #include "tpdec_lib.h" #include "conceal.h" #include "FDK_crc.h" #define PS_IS_EXPLICITLY_DISABLED(aot, flags) \ (((aot) == AOT_DRM_AAC) && !(flags & AC_PS_PRESENT)) #define IS_STEREO_SBR(el_id, stereoConfigIndex) \ (((el_id) == ID_USAC_CPE && (stereoConfigIndex) == 0) || \ ((el_id) == ID_USAC_CPE && (stereoConfigIndex) == 3)) void CAacDecoder_SyncQmfMode(HANDLE_AACDECODER self) { FDK_ASSERT( !((self->flags[0] & AC_MPS_PRESENT) && (self->flags[0] & AC_PS_PRESENT))); /* Assign user requested mode */ self->qmfModeCurr = self->qmfModeUser; if (IS_USAC(self->streamInfo.aot)) { self->qmfModeCurr = MODE_HQ; } if (self->qmfModeCurr == NOT_DEFINED) { if ((IS_LOWDELAY(self->streamInfo.aot) && (self->flags[0] & AC_MPS_PRESENT)) || ((self->streamInfo.aacNumChannels == 1) && ((CAN_DO_PS(self->streamInfo.aot) && !(self->flags[0] & AC_MPS_PRESENT)) || (IS_USAC(self->streamInfo.aot))))) { self->qmfModeCurr = MODE_HQ; } else { self->qmfModeCurr = MODE_LP; } } if (self->mpsEnableCurr) { if (IS_LOWDELAY(self->streamInfo.aot) && (self->qmfModeCurr == MODE_LP)) { /* Overrule user requested QMF mode */ self->qmfModeCurr = MODE_HQ; } /* Set and check if MPS decoder allows the current mode */ switch (mpegSurroundDecoder_SetParam( (CMpegSurroundDecoder *)self->pMpegSurroundDecoder, SACDEC_PARTIALLY_COMPLEX, self->qmfModeCurr == MODE_LP)) { case MPS_OK: break; case MPS_INVALID_PARAMETER: { /* Only one mode supported. Find out which one: */ LIB_INFO libInfo[FDK_MODULE_LAST]; UINT mpsCaps; FDKinitLibInfo(libInfo); mpegSurroundDecoder_GetLibInfo(libInfo); mpsCaps = FDKlibInfo_getCapabilities(libInfo, FDK_MPSDEC); if (((mpsCaps & CAPF_MPS_LP) && (self->qmfModeCurr == MODE_LP)) || ((mpsCaps & CAPF_MPS_HQ) && (self->qmfModeCurr == MODE_HQ))) { /* MPS decoder does support the requested mode. */ break; } } FDK_FALLTHROUGH; default: if (self->qmfModeUser == NOT_DEFINED) { /* Revert in case mpegSurroundDecoder_SetParam() fails. */ self->qmfModeCurr = (self->qmfModeCurr == MODE_LP) ? MODE_HQ : MODE_LP; } else { /* in case specific mode was requested we disable MPS and playout the * downmix */ self->mpsEnableCurr = 0; } } } /* Set SBR to current QMF mode. Error does not matter. */ sbrDecoder_SetParam(self->hSbrDecoder, SBR_QMF_MODE, (self->qmfModeCurr == MODE_LP)); self->psPossible = ((CAN_DO_PS(self->streamInfo.aot) && !PS_IS_EXPLICITLY_DISABLED(self->streamInfo.aot, self->flags[0]) && self->streamInfo.aacNumChannels == 1 && !(self->flags[0] & AC_MPS_PRESENT))) && self->qmfModeCurr == MODE_HQ; FDK_ASSERT(!((self->flags[0] & AC_MPS_PRESENT) && self->psPossible)); } void CAacDecoder_SignalInterruption(HANDLE_AACDECODER self) { if (self->flags[0] & (AC_USAC | AC_RSVD50 | AC_RSV603DA)) { int i; for (i = 0; i < fMin(self->aacChannels, (8)); i++) { if (self->pAacDecoderStaticChannelInfo [i]) { /* number of active channels can be smaller */ self->pAacDecoderStaticChannelInfo[i]->hArCo->m_numberLinesPrev = 0; } } } } /*! \brief Calculates the number of element channels \type channel type \usacStereoConfigIndex usac stereo config index \return element channels */ static int CAacDecoder_GetELChannels(MP4_ELEMENT_ID type, UCHAR usacStereoConfigIndex) { int el_channels = 0; switch (type) { case ID_USAC_CPE: if (usacStereoConfigIndex == 1) { el_channels = 1; } else { el_channels = 2; } break; case ID_CPE: el_channels = 2; break; case ID_USAC_SCE: case ID_USAC_LFE: case ID_SCE: case ID_LFE: el_channels = 1; break; default: el_channels = 0; break; } return el_channels; } /*! \brief Reset ancillary data struct. Call before parsing a new frame. \ancData Pointer to ancillary data structure \return Error code */ static AAC_DECODER_ERROR CAacDecoder_AncDataReset(CAncData *ancData) { int i; for (i = 0; i < 8; i++) { ancData->offset[i] = 0; } ancData->nrElements = 0; return AAC_DEC_OK; } /*! \brief Initialize ancillary buffer \ancData Pointer to ancillary data structure \buffer Pointer to (external) anc data buffer \size Size of the buffer pointed on by buffer in bytes \return Error code */ AAC_DECODER_ERROR CAacDecoder_AncDataInit(CAncData *ancData, unsigned char *buffer, int size) { if (size >= 0) { ancData->buffer = buffer; ancData->bufferSize = size; CAacDecoder_AncDataReset(ancData); return AAC_DEC_OK; } return AAC_DEC_ANC_DATA_ERROR; } /*! \brief Get one ancillary data element \ancData Pointer to ancillary data structure \index Index of the anc data element to get \ptr Pointer to a buffer receiving a pointer to the requested anc data element \size Pointer to a buffer receiving the length of the requested anc data element in bytes \return Error code */ AAC_DECODER_ERROR CAacDecoder_AncDataGet(CAncData *ancData, int index, unsigned char **ptr, int *size) { AAC_DECODER_ERROR error = AAC_DEC_OK; *ptr = NULL; *size = 0; if (index >= 0 && index < 8 - 1 && index < ancData->nrElements) { *ptr = &ancData->buffer[ancData->offset[index]]; *size = ancData->offset[index + 1] - ancData->offset[index]; } return error; } /*! \brief Parse ancillary data \ancData Pointer to ancillary data structure \hBs Handle to FDK bitstream \ancBytes Length of ancillary data to read from the bitstream \return Error code */ static AAC_DECODER_ERROR CAacDecoder_AncDataParse(CAncData *ancData, HANDLE_FDK_BITSTREAM hBs, const int ancBytes) { AAC_DECODER_ERROR error = AAC_DEC_OK; int readBytes = 0; if (ancData->buffer != NULL) { if (ancBytes > 0) { /* write ancillary data to external buffer */ int offset = ancData->offset[ancData->nrElements]; if ((offset + ancBytes) > ancData->bufferSize) { error = AAC_DEC_TOO_SMALL_ANC_BUFFER; } else if (ancData->nrElements >= 8 - 1) { error = AAC_DEC_TOO_MANY_ANC_ELEMENTS; } else { int i; for (i = 0; i < ancBytes; i++) { ancData->buffer[i + offset] = FDKreadBits(hBs, 8); readBytes++; } ancData->nrElements++; ancData->offset[ancData->nrElements] = ancBytes + ancData->offset[ancData->nrElements - 1]; } } } readBytes = ancBytes - readBytes; if (readBytes > 0) { /* skip data */ FDKpushFor(hBs, readBytes << 3); } return error; } /*! \brief Read Stream Data Element \bs Bitstream Handle \return Error code */ static AAC_DECODER_ERROR CDataStreamElement_Read(HANDLE_AACDECODER self, HANDLE_FDK_BITSTREAM bs, UCHAR *elementInstanceTag, UINT alignmentAnchor) { AAC_DECODER_ERROR error = AAC_DEC_OK; UINT dseBits; INT dataStart; int dataByteAlignFlag, count; FDK_ASSERT(self != NULL); int crcReg = transportDec_CrcStartReg(self->hInput, 0); /* Element Instance Tag */ *elementInstanceTag = FDKreadBits(bs, 4); /* Data Byte Align Flag */ dataByteAlignFlag = FDKreadBits(bs, 1); count = FDKreadBits(bs, 8); if (count == 255) { count += FDKreadBits(bs, 8); /* EscCount */ } dseBits = count * 8; if (dataByteAlignFlag) { FDKbyteAlign(bs, alignmentAnchor); } dataStart = (INT)FDKgetValidBits(bs); error = CAacDecoder_AncDataParse(&self->ancData, bs, count); transportDec_CrcEndReg(self->hInput, crcReg); { /* Move to the beginning of the data chunk */ FDKpushBack(bs, dataStart - (INT)FDKgetValidBits(bs)); /* Read Anc data if available */ aacDecoder_drcMarkPayload(self->hDrcInfo, bs, DVB_DRC_ANC_DATA); } { PCMDMX_ERROR dmxErr = PCMDMX_OK; /* Move to the beginning of the data chunk */ FDKpushBack(bs, dataStart - (INT)FDKgetValidBits(bs)); /* Read DMX meta-data */ dmxErr = pcmDmx_Parse(self->hPcmUtils, bs, dseBits, 0 /* not mpeg2 */); if (error == AAC_DEC_OK && dmxErr != PCMDMX_OK) { error = AAC_DEC_UNKNOWN; } } /* Move to the very end of the element. */ FDKpushBiDirectional(bs, (INT)FDKgetValidBits(bs) - dataStart + (INT)dseBits); return error; } /*! \brief Read Program Config Element \bs Bitstream Handle \pTp Transport decoder handle for CRC handling \pce Pointer to PCE buffer \channelConfig Current channel configuration \alignAnchor Anchor for byte alignment \return PCE status (-1: fail, 0: no new PCE, 1: PCE updated, 2: PCE updated need re-config). */ static int CProgramConfigElement_Read(HANDLE_FDK_BITSTREAM bs, HANDLE_TRANSPORTDEC pTp, CProgramConfig *pce, const UINT channelConfig, const UINT alignAnchor) { int pceStatus = 0; int crcReg; /* read PCE to temporal buffer first */ C_ALLOC_SCRATCH_START(tmpPce, CProgramConfig, 1); CProgramConfig_Init(tmpPce); crcReg = transportDec_CrcStartReg(pTp, 0); CProgramConfig_Read(tmpPce, bs, alignAnchor); transportDec_CrcEndReg(pTp, crcReg); if (CProgramConfig_IsValid(tmpPce) && (tmpPce->Profile == 1)) { if (!CProgramConfig_IsValid(pce) && (channelConfig > 0)) { /* Create a standard channel config PCE to compare with */ CProgramConfig_GetDefault(pce, channelConfig); } if (CProgramConfig_IsValid(pce)) { /* Compare the new and the old PCE (tags ignored) */ switch (CProgramConfig_Compare(pce, tmpPce)) { case 1: /* Channel configuration not changed. Just new metadata. */ FDKmemcpy(pce, tmpPce, sizeof(CProgramConfig)); /* Store the complete PCE */ pceStatus = 1; /* New PCE but no change of config */ break; case 2: /* The number of channels are identical but not the config */ case -1: /* The channel configuration is completely different */ pceStatus = -1; /* Not supported! */ break; case 0: /* Nothing to do because PCE matches the old one exactly. */ default: /* pceStatus = 0; */ break; } } } C_ALLOC_SCRATCH_END(tmpPce, CProgramConfig, 1); return pceStatus; } /*! \brief Prepares crossfade for USAC DASH IPF config change \pTimeData Pointer to time data \pTimeDataFlush Pointer to flushed time data \numChannels Number of channels \frameSize Size of frame \interleaved Indicates if time data is interleaved \return Error code */ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_PrepareCrossFade( const INT_PCM *pTimeData, INT_PCM **pTimeDataFlush, const INT numChannels, const INT frameSize, const INT interleaved) { int i, ch, s1, s2; AAC_DECODER_ERROR ErrorStatus; ErrorStatus = AAC_DEC_OK; if (interleaved) { s1 = 1; s2 = numChannels; } else { s1 = frameSize; s2 = 1; } for (ch = 0; ch < numChannels; ch++) { const INT_PCM *pIn = &pTimeData[ch * s1]; for (i = 0; i < TIME_DATA_FLUSH_SIZE; i++) { pTimeDataFlush[ch][i] = *pIn; pIn += s2; } } return ErrorStatus; } /*! \brief Applies crossfade for USAC DASH IPF config change \pTimeData Pointer to time data \pTimeDataFlush Pointer to flushed time data \numChannels Number of channels \frameSize Size of frame \interleaved Indicates if time data is interleaved \return Error code */ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_ApplyCrossFade( INT_PCM *pTimeData, INT_PCM **pTimeDataFlush, const INT numChannels, const INT frameSize, const INT interleaved) { int i, ch, s1, s2; AAC_DECODER_ERROR ErrorStatus; ErrorStatus = AAC_DEC_OK; if (interleaved) { s1 = 1; s2 = numChannels; } else { s1 = frameSize; s2 = 1; } for (ch = 0; ch < numChannels; ch++) { INT_PCM *pIn = &pTimeData[ch * s1]; for (i = 0; i < TIME_DATA_FLUSH_SIZE; i++) { FIXP_SGL alpha = (FIXP_SGL)i << (FRACT_BITS - 1 - TIME_DATA_FLUSH_SIZE_SF); FIXP_DBL time = FX_PCM2FX_DBL(*pIn); FIXP_DBL timeFlush = FX_PCM2FX_DBL(pTimeDataFlush[ch][i]); *pIn = (INT_PCM)(FIXP_PCM)FX_DBL2FX_PCM( timeFlush - fMult(timeFlush, alpha) + fMult(time, alpha)); pIn += s2; } } return ErrorStatus; } /*! \brief Parse PreRoll Extension Payload \self Handle of AAC decoder \numPrerollAU Number of preRoll AUs \prerollAUOffset Offset to each preRoll AU \prerollAULength Length of each preRoll AU \return Error code */ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_PreRollExtensionPayloadParse( HANDLE_AACDECODER self, UINT *numPrerollAU, UINT *prerollAUOffset, UINT *prerollAULength) { FDK_BITSTREAM bs; HANDLE_FDK_BITSTREAM hBs; AAC_DECODER_ERROR ErrorStatus; INT auStartAnchor; UINT independencyFlag; UINT extPayloadPresentFlag; UINT useDefaultLengthFlag; UINT configLength = 0; UINT preRollPossible = 1; UINT i; UCHAR configChanged = 0; UCHAR config[TP_USAC_MAX_CONFIG_LEN] = {0}; UCHAR implicitExplicitCfgDiff = 0; /* in case implicit and explicit config is equal preroll AU's should be processed after decoder reset */ ErrorStatus = AAC_DEC_OK; hBs = transportDec_GetBitstream(self->hInput, 0); bs = *hBs; auStartAnchor = (INT)FDKgetValidBits(hBs); if (auStartAnchor <= 0) { ErrorStatus = AAC_DEC_NOT_ENOUGH_BITS; goto bail; } /* Independency flag */ FDKreadBit(hBs); /* Payload present flag of extension ID_EXT_ELE_AUDIOPREROLL must be one */ extPayloadPresentFlag = FDKreadBits(hBs, 1); if (!extPayloadPresentFlag) { preRollPossible = 0; } /* Default length flag of extension ID_EXT_ELE_AUDIOPREROLL must be zero */ useDefaultLengthFlag = FDKreadBits(hBs, 1); if (useDefaultLengthFlag) { preRollPossible = 0; } if (preRollPossible) { /* extPayloadPresentFlag && !useDefaultLengthFlag */ /* Read overall ext payload length, useDefaultLengthFlag must be zero. */ escapedValue(hBs, 8, 16, 0); /* Read RSVD60 Config size */ configLength = escapedValue(hBs, 4, 4, 8); /* Avoid decoding pre roll frames if there was no config change and no * config is included in the pre roll ext payload. */ } /* If pre roll not possible then exit. */ if (preRollPossible == 0) { /* Sanity check: if flushing is switched on, preRollPossible must be 1 */ if (self->flushStatus != AACDEC_FLUSH_OFF) { /* Mismatch of current payload and flushing status */ self->flushStatus = AACDEC_FLUSH_OFF; ErrorStatus = AAC_DEC_PARSE_ERROR; } goto bail; } if (self->flags[0] & AC_USAC) { if (configLength > 0) { /* DASH IPF USAC Config Change: Read new config and compare with current * config. Apply reconfiguration if config's are different. */ for (i = 0; i < configLength; i++) { config[i] = FDKreadBits(hBs, 8); } TRANSPORTDEC_ERROR terr; terr = transportDec_InBandConfig(self->hInput, config, configLength, self->buildUpStatus, &configChanged, 0, &implicitExplicitCfgDiff); if (terr != TRANSPORTDEC_OK) { ErrorStatus = AAC_DEC_PARSE_ERROR; goto bail; } } } /* For the first frame buildUpStatus is not set and no flushing is performed * but preroll AU's should processed. */ /* For USAC there is no idle state. */ if ((self->streamInfo.numChannels == 0) && !implicitExplicitCfgDiff && (self->flags[0] & AC_USAC)) { self->buildUpStatus = AACDEC_USAC_BUILD_UP_ON; /* sanity check: if buildUp status on -> flushing must be off */ if (self->flushStatus != AACDEC_FLUSH_OFF) { self->flushStatus = AACDEC_FLUSH_OFF; ErrorStatus = AAC_DEC_PARSE_ERROR; goto bail; } } if (self->flags[0] & AC_USAC) { /* We are interested in preroll AUs if an explicit or an implicit config * change is signalized in other words if the build up status is set. */ if (self->buildUpStatus == AACDEC_USAC_BUILD_UP_ON) { self->applyCrossfade |= FDKreadBit(hBs); FDKreadBit(hBs); /* reserved */ /* Read num preroll AU's */ *numPrerollAU = escapedValue(hBs, 2, 4, 0); /* check limits for USAC */ if (*numPrerollAU > AACDEC_MAX_NUM_PREROLL_AU_USAC) { *numPrerollAU = 0; ErrorStatus = AAC_DEC_PARSE_ERROR; goto bail; } } } for (i = 0; i < *numPrerollAU; i++) { /* For every AU get length and offset in the bitstream */ prerollAULength[i] = escapedValue(hBs, 16, 16, 0); if (prerollAULength[i] > 0) { prerollAUOffset[i] = auStartAnchor - (INT)FDKgetValidBits(hBs); independencyFlag = FDKreadBit(hBs); if (i == 0 && !independencyFlag) { *numPrerollAU = 0; ErrorStatus = AAC_DEC_PARSE_ERROR; goto bail; } FDKpushFor(hBs, prerollAULength[i] * 8 - 1); self->prerollAULength[i] = (prerollAULength[i] * 8) + prerollAUOffset[i]; } else { *numPrerollAU = 0; ErrorStatus = AAC_DEC_PARSE_ERROR; /* Something is wrong */ goto bail; } } bail: *hBs = bs; return ErrorStatus; } /*! \brief Parse Extension Payload \self Handle of AAC decoder \count Pointer to bit counter. \previous_element ID of previous element (required by some extension payloads) \return Error code */ static AAC_DECODER_ERROR CAacDecoder_ExtPayloadParse( HANDLE_AACDECODER self, HANDLE_FDK_BITSTREAM hBs, int *count, MP4_ELEMENT_ID previous_element, int elIndex, int fIsFillElement) { AAC_DECODER_ERROR error = AAC_DEC_OK; EXT_PAYLOAD_TYPE extension_type; int bytes = (*count) >> 3; int crcFlag = 0; if (*count < 4) { return AAC_DEC_PARSE_ERROR; } else if ((INT)FDKgetValidBits(hBs) < *count) { return AAC_DEC_DECODE_FRAME_ERROR; } extension_type = (EXT_PAYLOAD_TYPE)FDKreadBits(hBs, 4); /* bs_extension_type */ *count -= 4; /* For ELD, the SBR signaling is explicit and parsed in aacDecoder_ParseExplicitMpsAndSbr(), therefore skip SBR if implicit present. */ if ((self->flags[0] & AC_ELD) && ((extension_type == EXT_SBR_DATA_CRC) || (extension_type == EXT_SBR_DATA))) { extension_type = EXT_FIL; /* skip sbr data */ } switch (extension_type) { case EXT_DYNAMIC_RANGE: { INT readBits = aacDecoder_drcMarkPayload(self->hDrcInfo, hBs, MPEG_DRC_EXT_DATA); if (readBits > *count) { /* Read too much. Something went wrong! */ error = AAC_DEC_PARSE_ERROR; } *count -= readBits; } break; case EXT_UNI_DRC: { DRC_DEC_ERROR drcErr = DRC_DEC_OK; DRC_DEC_CODEC_MODE drcDecCodecMode = DRC_DEC_CODEC_MODE_UNDEFINED; INT nBitsRemaining = FDKgetValidBits(hBs); INT readBits; switch (self->streamInfo.aot) { case AOT_AAC_LC: case AOT_SBR: case AOT_PS: drcDecCodecMode = DRC_DEC_MPEG_4_AAC; break; default: error = AAC_DEC_PARSE_ERROR; goto bail; } drcErr = FDK_drcDec_SetCodecMode(self->hUniDrcDecoder, drcDecCodecMode); if (drcErr) { error = AAC_DEC_PARSE_ERROR; goto bail; } drcErr = FDK_drcDec_ReadUniDrc(self->hUniDrcDecoder, hBs); if (drcErr) { error = AAC_DEC_PARSE_ERROR; goto bail; } readBits = (INT)nBitsRemaining - (INT)FDKgetValidBits(hBs); if (readBits > *count) { /* Read too much. Something went wrong! */ error = AAC_DEC_PARSE_ERROR; } *count -= readBits; /* Skip any trailing bits */ FDKpushFor(hBs, *count); *count = 0; } break; case EXT_LDSAC_DATA: case EXT_SAC_DATA: /* Read MPEG Surround Extension payload */ { int err, mpsSampleRate, mpsFrameSize; if (self->flags[0] & AC_PS_PRESENT) { error = AAC_DEC_PARSE_ERROR; goto bail; } /* Handle SBR dual rate case */ if (self->streamInfo.extSamplingRate != 0) { mpsSampleRate = self->streamInfo.extSamplingRate; mpsFrameSize = self->streamInfo.aacSamplesPerFrame * (self->streamInfo.extSamplingRate / self->streamInfo.aacSampleRate); } else { mpsSampleRate = self->streamInfo.aacSampleRate; mpsFrameSize = self->streamInfo.aacSamplesPerFrame; } /* Setting of internal MPS state; may be reset in CAacDecoder_SyncQmfMode if decoder is unable to decode with user defined qmfMode */ if (!(self->flags[0] & (AC_USAC | AC_RSVD50 | AC_ELD))) { self->mpsEnableCurr = self->mpsEnableUser; } if (self->mpsEnableCurr) { if (!self->qmfDomain.globalConf.qmfDomainExplicitConfig) { /* if not done yet, allocate full MPEG Surround decoder instance */ if (mpegSurroundDecoder_IsFullMpegSurroundDecoderInstanceAvailable( (CMpegSurroundDecoder *)self->pMpegSurroundDecoder) == SAC_INSTANCE_NOT_FULL_AVAILABLE) { if (mpegSurroundDecoder_Open( (CMpegSurroundDecoder **)&self->pMpegSurroundDecoder, -1, &self->qmfDomain)) { return AAC_DEC_OUT_OF_MEMORY; } } } err = mpegSurroundDecoder_Parse( (CMpegSurroundDecoder *)self->pMpegSurroundDecoder, hBs, count, self->streamInfo.aot, mpsSampleRate, mpsFrameSize, self->flags[0] & AC_INDEP); if (err == MPS_OK) { self->flags[0] |= AC_MPS_PRESENT; } else { error = AAC_DEC_PARSE_ERROR; } } /* Skip any trailing bytes */ FDKpushFor(hBs, *count); *count = 0; } break; case EXT_SBR_DATA_CRC: crcFlag = 1; FDK_FALLTHROUGH; case EXT_SBR_DATA: if (IS_CHANNEL_ELEMENT(previous_element)) { SBR_ERROR sbrError; UCHAR configMode = 0; UCHAR configChanged = 0; CAacDecoder_SyncQmfMode(self); configMode |= AC_CM_ALLOC_MEM; sbrError = sbrDecoder_InitElement( self->hSbrDecoder, self->streamInfo.aacSampleRate, self->streamInfo.extSamplingRate, self->streamInfo.aacSamplesPerFrame, self->streamInfo.aot, previous_element, elIndex, 2, /* Signalize that harmonicSBR shall be ignored in the config change detection */ 0, configMode, &configChanged, self->downscaleFactor); if (sbrError == SBRDEC_OK) { sbrError = sbrDecoder_Parse(self->hSbrDecoder, hBs, self->pDrmBsBuffer, self->drmBsBufferSize, count, *count, crcFlag, previous_element, elIndex, self->flags[0], self->elFlags); /* Enable SBR for implicit SBR signalling but only if no severe error * happend. */ if ((sbrError == SBRDEC_OK) || (sbrError == SBRDEC_PARSE_ERROR)) { self->sbrEnabled = 1; } } else { /* Do not try to apply SBR because initializing the element failed. */ self->sbrEnabled = 0; } /* Citation from ISO/IEC 14496-3 chapter 4.5.2.1.5.2 Fill elements containing an extension_payload() with an extension_type of EXT_SBR_DATA or EXT_SBR_DATA_CRC shall not contain any other extension_payload of any other extension_type. */ if (fIsFillElement) { FDKpushBiDirectional(hBs, *count); *count = 0; } else { /* If this is not a fill element with a known length, we are screwed * and further parsing makes no sense. */ if (sbrError != SBRDEC_OK) { self->frameOK = 0; } } } else { error = AAC_DEC_PARSE_ERROR; } break; case EXT_FILL_DATA: { int temp; temp = FDKreadBits(hBs, 4); bytes--; if (temp != 0) { error = AAC_DEC_PARSE_ERROR; break; } while (bytes > 0) { temp = FDKreadBits(hBs, 8); bytes--; if (temp != 0xa5) { error = AAC_DEC_PARSE_ERROR; break; } } *count = bytes << 3; } break; case EXT_DATA_ELEMENT: { int dataElementVersion; dataElementVersion = FDKreadBits(hBs, 4); *count -= 4; if (dataElementVersion == 0) /* ANC_DATA */ { int temp, dataElementLength = 0; do { temp = FDKreadBits(hBs, 8); *count -= 8; dataElementLength += temp; } while (temp == 255); CAacDecoder_AncDataParse(&self->ancData, hBs, dataElementLength); *count -= (dataElementLength << 3); } else { /* align = 0 */ error = AAC_DEC_PARSE_ERROR; goto bail; } } break; case EXT_DATA_LENGTH: if (!fIsFillElement /* Makes no sens to have an additional length in a fill ... */ && (self->flags[0] & AC_ER)) /* ... element because this extension payload type was ... */ { /* ... created to circumvent the missing length in ER-Syntax. */ int bitCnt, len = FDKreadBits(hBs, 4); *count -= 4; if (len == 15) { int add_len = FDKreadBits(hBs, 8); *count -= 8; len += add_len; if (add_len == 255) { len += FDKreadBits(hBs, 16); *count -= 16; } } len <<= 3; bitCnt = len; if ((EXT_PAYLOAD_TYPE)FDKreadBits(hBs, 4) == EXT_DATA_LENGTH) { /* Check NOTE 2: The extension_payload() included here must not have extension_type == EXT_DATA_LENGTH. */ error = AAC_DEC_PARSE_ERROR; goto bail; } else { /* rewind and call myself again. */ FDKpushBack(hBs, 4); error = CAacDecoder_ExtPayloadParse( self, hBs, &bitCnt, previous_element, elIndex, 1); /* Treat same as fill element */ *count -= len - bitCnt; } /* Note: the fall through in case the if statement above is not taken is * intentional. */ break; } FDK_FALLTHROUGH; case EXT_FIL: default: /* align = 4 */ FDKpushFor(hBs, *count); *count = 0; break; } bail: if ((error != AAC_DEC_OK) && fIsFillElement) { /* Skip the remaining extension bytes */ FDKpushBiDirectional(hBs, *count); *count = 0; /* Patch error code because decoding can go on. */ error = AAC_DEC_OK; /* Be sure that parsing errors have been stored. */ } return error; } static AAC_DECODER_ERROR aacDecoder_ParseExplicitMpsAndSbr( HANDLE_AACDECODER self, HANDLE_FDK_BITSTREAM bs, const MP4_ELEMENT_ID previous_element, const int previous_element_index, const int element_index, const int el_cnt[]) { AAC_DECODER_ERROR ErrorStatus = AAC_DEC_OK; INT bitCnt = 0; /* get the remaining bits of this frame */ bitCnt = transportDec_GetAuBitsRemaining(self->hInput, 0); if ((self->flags[0] & AC_SBR_PRESENT) && (self->flags[0] & (AC_USAC | AC_RSVD50 | AC_ELD | AC_DRM))) { SBR_ERROR err = SBRDEC_OK; int chElIdx, numChElements = el_cnt[ID_SCE] + el_cnt[ID_CPE] + el_cnt[ID_LFE] + el_cnt[ID_USAC_SCE] + el_cnt[ID_USAC_CPE] + el_cnt[ID_USAC_LFE]; INT bitCntTmp = bitCnt; if (self->flags[0] & AC_USAC) { chElIdx = numChElements - 1; } else { chElIdx = 0; /* ELD case */ } for (; chElIdx < numChElements; chElIdx += 1) { MP4_ELEMENT_ID sbrType; SBR_ERROR errTmp; if (self->flags[0] & (AC_USAC)) { FDK_ASSERT((self->elements[element_index] == ID_USAC_SCE) || (self->elements[element_index] == ID_USAC_CPE)); sbrType = IS_STEREO_SBR(self->elements[element_index], self->usacStereoConfigIndex[element_index]) ? ID_CPE : ID_SCE; } else sbrType = self->elements[chElIdx]; errTmp = sbrDecoder_Parse(self->hSbrDecoder, bs, self->pDrmBsBuffer, self->drmBsBufferSize, &bitCnt, -1, self->flags[0] & AC_SBRCRC, sbrType, chElIdx, self->flags[0], self->elFlags); if (errTmp != SBRDEC_OK) { err = errTmp; bitCntTmp = bitCnt; bitCnt = 0; } } switch (err) { case SBRDEC_PARSE_ERROR: /* Can not go on parsing because we do not know the length of the SBR extension data. */ FDKpushFor(bs, bitCntTmp); bitCnt = 0; break; case SBRDEC_OK: self->sbrEnabled = 1; break; default: self->frameOK = 0; break; } } if ((bitCnt > 0) && (self->flags[0] & (AC_USAC | AC_RSVD50))) { if ((self->flags[0] & AC_MPS_PRESENT) || (self->elFlags[element_index] & AC_EL_USAC_MPS212)) { int err; err = mpegSurroundDecoder_ParseNoHeader( (CMpegSurroundDecoder *)self->pMpegSurroundDecoder, bs, &bitCnt, self->flags[0] & AC_INDEP); if (err != MPS_OK) { self->frameOK = 0; ErrorStatus = AAC_DEC_PARSE_ERROR; } } } if (self->flags[0] & AC_DRM) { if ((bitCnt = (INT)FDKgetValidBits(bs)) != 0) { FDKpushBiDirectional(bs, bitCnt); } } if (!(self->flags[0] & (AC_USAC | AC_RSVD50 | AC_DRM))) { while (bitCnt > 7) { ErrorStatus = CAacDecoder_ExtPayloadParse( self, bs, &bitCnt, previous_element, previous_element_index, 0); if (ErrorStatus != AAC_DEC_OK) { self->frameOK = 0; ErrorStatus = AAC_DEC_PARSE_ERROR; break; } } } return ErrorStatus; } /* Stream Configuration and Information. This class holds configuration and information data for a stream to be decoded. It provides the calling application as well as the decoder with substantial information, e.g. profile, sampling rate, number of channels found in the bitstream etc. */ static void CStreamInfoInit(CStreamInfo *pStreamInfo) { pStreamInfo->aacSampleRate = 0; pStreamInfo->profile = -1; pStreamInfo->aot = AOT_NONE; pStreamInfo->channelConfig = -1; pStreamInfo->bitRate = 0; pStreamInfo->aacSamplesPerFrame = 0; pStreamInfo->extAot = AOT_NONE; pStreamInfo->extSamplingRate = 0; pStreamInfo->flags = 0; pStreamInfo->epConfig = -1; /* default: no ER */ pStreamInfo->numChannels = 0; pStreamInfo->sampleRate = 0; pStreamInfo->frameSize = 0; pStreamInfo->outputDelay = 0; /* DRC */ pStreamInfo->drcProgRefLev = -1; /* set program reference level to not indicated */ pStreamInfo->drcPresMode = -1; /* default: presentation mode not indicated */ pStreamInfo->outputLoudness = -1; /* default: no loudness metadata present */ } /*! \brief Initialization of AacDecoderChannelInfo The function initializes the pointers to AacDecoderChannelInfo for each channel, set the start values for window shape and window sequence of overlap&add to zero, set the overlap buffer to zero and initializes the pointers to the window coefficients. \param bsFormat is the format of the AAC bitstream \return AACDECODER instance */ LINKSPEC_CPP HANDLE_AACDECODER CAacDecoder_Open( TRANSPORT_TYPE bsFormat) /*!< bitstream format (adif,adts,loas,...). */ { HANDLE_AACDECODER self; self = GetAacDecoder(); if (self == NULL) { goto bail; } FDK_QmfDomain_ClearRequested(&self->qmfDomain.globalConf); /* Assign channel mapping info arrays (doing so removes dependency of settings * header in API header). */ self->streamInfo.pChannelIndices = self->channelIndices; self->streamInfo.pChannelType = self->channelType; self->downscaleFactor = 1; self->downscaleFactorInBS = 1; /* initialize anc data */ CAacDecoder_AncDataInit(&self->ancData, NULL, 0); /* initialize stream info */ CStreamInfoInit(&self->streamInfo); /* initialize progam config */ CProgramConfig_Init(&self->pce); /* initialize error concealment common data */ CConcealment_InitCommonData(&self->concealCommonData); self->concealMethodUser = ConcealMethodNone; /* undefined -> auto mode */ self->hDrcInfo = GetDrcInfo(); if (self->hDrcInfo == NULL) { goto bail; } /* Init common DRC structure */ aacDecoder_drcInit(self->hDrcInfo); /* Set default frame delay */ aacDecoder_drcSetParam(self->hDrcInfo, DRC_BS_DELAY, CConcealment_GetDelay(&self->concealCommonData)); self->workBufferCore1 = (FIXP_DBL *)GetWorkBufferCore1(); self->workBufferCore2 = GetWorkBufferCore2(); if (self->workBufferCore2 == NULL) goto bail; /* When RSVD60 is active use dedicated memory for core decoding */ self->pTimeData2 = GetWorkBufferCore5(); self->timeData2Size = GetRequiredMemWorkBufferCore5(); if (self->pTimeData2 == NULL) { goto bail; } return self; bail: CAacDecoder_Close(self); return NULL; } /* Revert CAacDecoder_Init() */ static void CAacDecoder_DeInit(HANDLE_AACDECODER self, const int subStreamIndex) { int ch; int aacChannelOffset = 0, aacChannels = (8); int numElements = (3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + 1), elementOffset = 0; if (self == NULL) return; { self->ascChannels[0] = 0; self->elements[0] = ID_END; } for (ch = aacChannelOffset; ch < aacChannelOffset + aacChannels; ch++) { if (self->pAacDecoderChannelInfo[ch] != NULL) { if (self->pAacDecoderChannelInfo[ch]->pComStaticData != NULL) { if (self->pAacDecoderChannelInfo[ch] ->pComStaticData->pWorkBufferCore1 != NULL) { if (ch == aacChannelOffset) { FreeWorkBufferCore1(&self->pAacDecoderChannelInfo[ch] ->pComStaticData->pWorkBufferCore1); } } if (self->pAacDecoderChannelInfo[ch] ->pComStaticData->cplxPredictionData != NULL) { FreeCplxPredictionData(&self->pAacDecoderChannelInfo[ch] ->pComStaticData->cplxPredictionData); } /* Avoid double free of linked pComStaticData in case of CPE by settings * pointer to NULL. */ if (ch < (8) - 1) { if ((self->pAacDecoderChannelInfo[ch + 1] != NULL) && (self->pAacDecoderChannelInfo[ch + 1]->pComStaticData == self->pAacDecoderChannelInfo[ch]->pComStaticData)) { self->pAacDecoderChannelInfo[ch + 1]->pComStaticData = NULL; } } FDKfree(self->pAacDecoderChannelInfo[ch]->pComStaticData); self->pAacDecoderChannelInfo[ch]->pComStaticData = NULL; } if (self->pAacDecoderChannelInfo[ch]->pComData != NULL) { /* Avoid double free of linked pComData in case of CPE by settings * pointer to NULL. */ if (ch < (8) - 1) { if ((self->pAacDecoderChannelInfo[ch + 1] != NULL) && (self->pAacDecoderChannelInfo[ch + 1]->pComData == self->pAacDecoderChannelInfo[ch]->pComData)) { self->pAacDecoderChannelInfo[ch + 1]->pComData = NULL; } } if (ch == aacChannelOffset) { FreeWorkBufferCore6( (SCHAR **)&self->pAacDecoderChannelInfo[ch]->pComData); } else { FDKafree(self->pAacDecoderChannelInfo[ch]->pComData); } self->pAacDecoderChannelInfo[ch]->pComData = NULL; } } if (self->pAacDecoderStaticChannelInfo[ch] != NULL) { if (self->pAacDecoderStaticChannelInfo[ch]->pOverlapBuffer != NULL) { FreeOverlapBuffer( &self->pAacDecoderStaticChannelInfo[ch]->pOverlapBuffer); } if (self->pAacDecoderStaticChannelInfo[ch]->hArCo != NULL) { CArco_Destroy(self->pAacDecoderStaticChannelInfo[ch]->hArCo); } FreeAacDecoderStaticChannelInfo(&self->pAacDecoderStaticChannelInfo[ch]); } if (self->pAacDecoderChannelInfo[ch] != NULL) { FreeAacDecoderChannelInfo(&self->pAacDecoderChannelInfo[ch]); } } { int el; for (el = elementOffset; el < elementOffset + numElements; el++) { if (self->cpeStaticData[el] != NULL) { FreeCpePersistentData(&self->cpeStaticData[el]); } } } FDK_Delay_Destroy(&self->usacResidualDelay); self->aacChannels = 0; self->streamInfo.aacSampleRate = 0; self->streamInfo.sampleRate = 0; /* This samplerate value is checked for configuration change, not the others * above. */ self->samplingRateInfo[subStreamIndex].samplingRate = 0; } /*! * \brief CAacDecoder_CtrlCFGChange Set config change parameters. * * \param self [i] handle to AACDECODER structure * \param flushStatus [i] flush status: on|off * \param flushCnt [i] flush frame counter * \param buildUpStatus [i] build up status: on|off * \param buildUpCnt [i] build up frame counter * * \return error */ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_CtrlCFGChange(HANDLE_AACDECODER self, UCHAR flushStatus, SCHAR flushCnt, UCHAR buildUpStatus, SCHAR buildUpCnt) { AAC_DECODER_ERROR err = AAC_DEC_OK; self->flushStatus = flushStatus; self->flushCnt = flushCnt; self->buildUpStatus = buildUpStatus; self->buildUpCnt = buildUpCnt; return (err); } /*! * \brief CAacDecoder_FreeMem Free config dependent AAC memory. * * \param self [i] handle to AACDECODER structure * * \return error */ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_FreeMem(HANDLE_AACDECODER self, const int subStreamIndex) { AAC_DECODER_ERROR err = AAC_DEC_OK; CAacDecoder_DeInit(self, subStreamIndex); return (err); } /* Destroy aac decoder */ LINKSPEC_CPP void CAacDecoder_Close(HANDLE_AACDECODER self) { if (self == NULL) return; CAacDecoder_DeInit(self, 0); { int ch; for (ch = 0; ch < (8); ch++) { if (self->pTimeDataFlush[ch] != NULL) { FreeTimeDataFlush(&self->pTimeDataFlush[ch]); } } } if (self->hDrcInfo) { FreeDrcInfo(&self->hDrcInfo); } if (self->workBufferCore1 != NULL) { FreeWorkBufferCore1((CWorkBufferCore1 **)&self->workBufferCore1); } /* Free WorkBufferCore2 */ if (self->workBufferCore2 != NULL) { FreeWorkBufferCore2(&self->workBufferCore2); } if (self->pTimeData2 != NULL) { FreeWorkBufferCore5(&self->pTimeData2); } FDK_QmfDomain_Close(&self->qmfDomain); FreeAacDecoder(&self); } /*! \brief Initialization of decoder instance The function initializes the decoder. \return error status: 0 for success, <>0 for unsupported configurations */ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, UCHAR configMode, UCHAR *configChanged) { AAC_DECODER_ERROR err = AAC_DEC_OK; INT ascChannels, ascChanged = 0; AACDEC_RENDER_MODE initRenderMode = AACDEC_RENDER_INVALID; SCHAR usacStereoConfigIndex = -1; int usacResidualDelayCompSamples = 0; int elementOffset, aacChannelsOffset, aacChannelsOffsetIdx; const int streamIndex = 0; INT flushChannels = 0; if (!self) return AAC_DEC_INVALID_HANDLE; UCHAR downscaleFactor = self->downscaleFactor; UCHAR downscaleFactorInBS = self->downscaleFactorInBS; self->aacOutDataHeadroom = (3); // set profile and check for supported aot // leave profile on default (=-1) for all other supported MPEG-4 aot's except // aot=2 (=AAC-LC) switch (asc->m_aot) { case AOT_AAC_LC: self->streamInfo.profile = 1; FDK_FALLTHROUGH; case AOT_ER_AAC_SCAL: if (asc->m_sc.m_gaSpecificConfig.m_layer > 0) { /* aac_scalable_extension_element() currently not supported. */ return AAC_DEC_UNSUPPORTED_FORMAT; } FDK_FALLTHROUGH; case AOT_SBR: case AOT_PS: case AOT_ER_AAC_LC: case AOT_ER_AAC_LD: case AOT_DRM_AAC: case AOT_DRM_SURROUND: initRenderMode = AACDEC_RENDER_IMDCT; break; case AOT_ER_AAC_ELD: initRenderMode = AACDEC_RENDER_ELDFB; break; case AOT_USAC: initRenderMode = AACDEC_RENDER_IMDCT; break; default: return AAC_DEC_UNSUPPORTED_AOT; } if (CProgramConfig_IsValid(&self->pce) && (asc->m_channelConfiguration > 0)) { /* Compare the stored (old) PCE with a default PCE created from the (new) channel_config (on a temporal buffer) to find out wheter we can keep it (and its metadata) or not. */ int pceCmpResult; C_ALLOC_SCRATCH_START(tmpPce, CProgramConfig, 1); CProgramConfig_GetDefault(tmpPce, asc->m_channelConfiguration); pceCmpResult = CProgramConfig_Compare(&self->pce, tmpPce); if ((pceCmpResult < 0) /* Reset if PCEs are completely different ... */ || (pceCmpResult > 1)) { /* ... or have a different layout. */ CProgramConfig_Init(&self->pce); } /* Otherwise keep the PCE (and its metadata). */ C_ALLOC_SCRATCH_END(tmpPce, CProgramConfig, 1); } else { CProgramConfig_Init(&self->pce); } /* set channels */ switch (asc->m_channelConfiguration) { case 0: switch (asc->m_aot) { case AOT_USAC: self->chMapIndex = 0; ascChannels = asc->m_sc.m_usacConfig.m_nUsacChannels; break; default: /* get channels from program config (ASC) */ if (CProgramConfig_IsValid(&asc->m_progrConfigElement)) { ascChannels = asc->m_progrConfigElement.NumChannels; if (ascChannels > 0) { int el_tmp; /* valid number of channels -> copy program config element (PCE) * from ASC */ FDKmemcpy(&self->pce, &asc->m_progrConfigElement, sizeof(CProgramConfig)); /* Built element table */ el_tmp = CProgramConfig_GetElementTable( &asc->m_progrConfigElement, self->elements, (((8)) + (8)), &self->chMapIndex); for (; el_tmp < (3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + 1); el_tmp++) { self->elements[el_tmp] = ID_NONE; } } else { return AAC_DEC_UNSUPPORTED_CHANNELCONFIG; } } else { self->chMapIndex = 0; return AAC_DEC_UNSUPPORTED_CHANNELCONFIG; } break; } break; case 1: case 2: case 3: case 4: case 5: case 6: ascChannels = asc->m_channelConfiguration; break; case 11: ascChannels = 7; break; case 7: case 12: case 14: ascChannels = 8; break; default: return AAC_DEC_UNSUPPORTED_CHANNELCONFIG; } if (asc->m_aot == AOT_USAC) { flushChannels = fMin(ascChannels, (8)); INT numChannel; pcmDmx_GetParam(self->hPcmUtils, MIN_NUMBER_OF_OUTPUT_CHANNELS, &numChannel); flushChannels = fMin(fMax(numChannel, flushChannels), (8)); } if (IS_USAC(asc->m_aot)) { for (int el = 0; el < (INT)asc->m_sc.m_usacConfig.m_usacNumElements; el++) { /* fix number of core channels aka ascChannels for stereoConfigIndex = 1 * cases */ if (asc->m_sc.m_usacConfig.element[el].m_stereoConfigIndex == 1) { ascChannels--; /* stereoConfigIndex == 1 stereo cases do actually contain only a mono core channel. */ } else if (asc->m_sc.m_usacConfig.element[el].m_stereoConfigIndex == 2) { /* In this case it is necessary to follow up the DMX signal delay caused by HBE also with the residual signal (2nd core channel). The SBR overlap delay is not regarded here, this is handled by the MPS212 implementation. */ if (asc->m_sc.m_usacConfig.element[el].m_harmonicSBR) { usacResidualDelayCompSamples += asc->m_samplesPerFrame; } if (asc->m_sc.m_usacConfig.m_coreSbrFrameLengthIndex == 4) { usacResidualDelayCompSamples += 6 * 16; /* difference between 12 SBR overlap slots from SBR and 6 slots delayed in MPS212 */ } } } } aacChannelsOffset = 0; aacChannelsOffsetIdx = 0; elementOffset = 0; if ((ascChannels <= 0) || (ascChannels > (8)) || (asc->m_channelConfiguration > AACDEC_MAX_CH_CONF)) { return AAC_DEC_UNSUPPORTED_CHANNELCONFIG; } /* Set syntax flags */ self->flags[streamIndex] = 0; { FDKmemclear(self->elFlags, sizeof(self->elFlags)); } if ((asc->m_channelConfiguration > 0) || IS_USAC(asc->m_aot)) { if (IS_USAC(asc->m_aot)) { /* copy pointer to usac config (this is preliminary since there's an ongoing discussion about storing the config-part of the bitstream rather than the complete decoded configuration) */ self->pUsacConfig[streamIndex] = &asc->m_sc.m_usacConfig; /* copy list of elements */ if (self->pUsacConfig[streamIndex]->m_usacNumElements > (3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + 1)) { goto bail; } if (self->numUsacElements[streamIndex] != asc->m_sc.m_usacConfig.m_usacNumElements) { ascChanged = 1; } if (configMode & AC_CM_ALLOC_MEM) { self->numUsacElements[streamIndex] = asc->m_sc.m_usacConfig.m_usacNumElements; } self->mpsEnableCurr = 0; for (int _el = 0; _el < (int)self->pUsacConfig[streamIndex]->m_usacNumElements; _el++) { int el = _el + elementOffset; if (self->elements[el] != self->pUsacConfig[streamIndex]->element[_el].usacElementType) { ascChanged = 1; } if (self->usacStereoConfigIndex[el] != asc->m_sc.m_usacConfig.element[_el].m_stereoConfigIndex) { ascChanged = 1; } if (configMode & AC_CM_ALLOC_MEM) { self->elements[el] = self->pUsacConfig[streamIndex]->element[_el].usacElementType; /* for Unified Stereo Coding */ self->usacStereoConfigIndex[el] = asc->m_sc.m_usacConfig.element[_el].m_stereoConfigIndex; if (self->elements[el] == ID_USAC_CPE) { self->mpsEnableCurr |= self->usacStereoConfigIndex[el] ? 1 : 0; } } self->elFlags[el] |= (asc->m_sc.m_usacConfig.element[_el].m_noiseFilling) ? AC_EL_USAC_NOISE : 0; self->elFlags[el] |= (asc->m_sc.m_usacConfig.element[_el].m_stereoConfigIndex > 0) ? AC_EL_USAC_MPS212 : 0; self->elFlags[el] |= (asc->m_sc.m_usacConfig.element[_el].m_interTes) ? AC_EL_USAC_ITES : 0; self->elFlags[el] |= (asc->m_sc.m_usacConfig.element[_el].m_pvc) ? AC_EL_USAC_PVC : 0; self->elFlags[el] |= (asc->m_sc.m_usacConfig.element[_el].usacElementType == ID_USAC_LFE) ? AC_EL_USAC_LFE : 0; self->elFlags[el] |= (asc->m_sc.m_usacConfig.element[_el].usacElementType == ID_USAC_LFE) ? AC_EL_LFE : 0; if ((asc->m_sc.m_usacConfig.element[_el].usacElementType == ID_USAC_CPE) && ((self->usacStereoConfigIndex[el] == 0))) { self->elFlags[el] |= AC_EL_USAC_CP_POSSIBLE; } } self->hasAudioPreRoll = 0; if (self->pUsacConfig[streamIndex]->m_usacNumElements) { self->hasAudioPreRoll = asc->m_sc.m_usacConfig.element[0] .extElement.usacExtElementHasAudioPreRoll; } if (configMode & AC_CM_ALLOC_MEM) { self->elements[elementOffset + self->pUsacConfig[streamIndex]->m_usacNumElements] = ID_END; } } else { /* Initialize constant mappings for channel config 1-7 */ int i; for (i = 0; i < AACDEC_CH_ELEMENTS_TAB_SIZE; i++) { self->elements[i] = elementsTab[asc->m_channelConfiguration - 1][i]; } for (; i < (3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + 1); i++) { self->elements[i] = ID_NONE; } } { int ch; for (ch = 0; ch < ascChannels; ch++) { self->chMapping[ch] = ch; } for (; ch < (8); ch++) { self->chMapping[ch] = 255; } } self->chMapIndex = asc->m_channelConfiguration; } else { if (CProgramConfig_IsValid(&asc->m_progrConfigElement)) { /* Set matrix mixdown infos if available from PCE. */ pcmDmx_SetMatrixMixdownFromPce( self->hPcmUtils, asc->m_progrConfigElement.MatrixMixdownIndexPresent, asc->m_progrConfigElement.MatrixMixdownIndex, asc->m_progrConfigElement.PseudoSurroundEnable); } } self->streamInfo.channelConfig = asc->m_channelConfiguration; if (self->streamInfo.aot != asc->m_aot) { if (configMode & AC_CM_ALLOC_MEM) { self->streamInfo.aot = asc->m_aot; } ascChanged = 1; } if (asc->m_aot == AOT_ER_AAC_ELD && asc->m_sc.m_eldSpecificConfig.m_downscaledSamplingFrequency != 0) { if (self->samplingRateInfo[0].samplingRate != asc->m_sc.m_eldSpecificConfig.m_downscaledSamplingFrequency || self->samplingRateInfo[0].samplingRate * self->downscaleFactor != asc->m_samplingFrequency) { /* get downscaledSamplingFrequency from ESC and compute the downscale * factor */ downscaleFactorInBS = asc->m_samplingFrequency / asc->m_sc.m_eldSpecificConfig.m_downscaledSamplingFrequency; if (downscaleFactorInBS == 1 || downscaleFactorInBS == 2 || downscaleFactorInBS == 3 || downscaleFactorInBS == 4) { downscaleFactor = downscaleFactorInBS; } } } else { downscaleFactorInBS = 1; downscaleFactor = 1; } if (self->downscaleFactorInBS != downscaleFactorInBS) { if (configMode & AC_CM_ALLOC_MEM) { self->downscaleFactorInBS = downscaleFactorInBS; self->downscaleFactor = downscaleFactor; } ascChanged = 1; } if ((INT)asc->m_samplesPerFrame % downscaleFactor != 0) { return AAC_DEC_UNSUPPORTED_SAMPLINGRATE; /* frameSize/dsf must be an integer number */ } self->streamInfo.bitRate = 0; if (asc->m_aot == AOT_ER_AAC_ELD) { if (self->useLdQmfTimeAlign != asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign) { ascChanged = 1; } if (configMode & AC_CM_ALLOC_MEM) { self->useLdQmfTimeAlign = asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign; } if (self->sbrEnabled != asc->m_sbrPresentFlag) { ascChanged = 1; } } self->streamInfo.extAot = asc->m_extensionAudioObjectType; if (self->streamInfo.extSamplingRate != (INT)asc->m_extensionSamplingFrequency) { ascChanged = 1; } if (configMode & AC_CM_ALLOC_MEM) { self->streamInfo.extSamplingRate = asc->m_extensionSamplingFrequency; } self->flags[streamIndex] |= (asc->m_sbrPresentFlag) ? AC_SBR_PRESENT : 0; self->flags[streamIndex] |= (asc->m_psPresentFlag) ? AC_PS_PRESENT : 0; if (asc->m_sbrPresentFlag) { self->sbrEnabled = 1; self->sbrEnabledPrev = 1; } else { self->sbrEnabled = 0; self->sbrEnabledPrev = 0; } if (self->sbrEnabled && asc->m_extensionSamplingFrequency) { if (downscaleFactor != 1 && (downscaleFactor)&1) { return AAC_DEC_UNSUPPORTED_SAMPLINGRATE; /* SBR needs an even downscale factor */ } if (configMode & AC_CM_ALLOC_MEM) { self->streamInfo.extSamplingRate = self->streamInfo.extSamplingRate / self->downscaleFactor; } } if ((asc->m_aot == AOT_AAC_LC) && (asc->m_sbrPresentFlag == 1) && (asc->m_extensionSamplingFrequency > (2 * asc->m_samplingFrequency))) { return AAC_DEC_UNSUPPORTED_SAMPLINGRATE; /* Core decoder supports at most a 1:2 upsampling for HE-AAC and HE-AACv2 */ } /* --------- vcb11 ------------ */ self->flags[streamIndex] |= (asc->m_vcb11Flag) ? AC_ER_VCB11 : 0; /* ---------- rvlc ------------ */ self->flags[streamIndex] |= (asc->m_rvlcFlag) ? AC_ER_RVLC : 0; /* ----------- hcr ------------ */ self->flags[streamIndex] |= (asc->m_hcrFlag) ? AC_ER_HCR : 0; if (asc->m_aot == AOT_ER_AAC_ELD) { self->mpsEnableCurr = 0; self->flags[streamIndex] |= AC_ELD; self->flags[streamIndex] |= (asc->m_sbrPresentFlag) ? AC_SBR_PRESENT : 0; /* Need to set the SBR flag for backward-compatibility reasons. Even if SBR is not supported. */ self->flags[streamIndex] |= (asc->m_sc.m_eldSpecificConfig.m_sbrCrcFlag) ? AC_SBRCRC : 0; self->flags[streamIndex] |= (asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign) ? AC_MPS_PRESENT : 0; if (self->mpsApplicable) { self->mpsEnableCurr = asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign; } } self->flags[streamIndex] |= (asc->m_aot == AOT_ER_AAC_LD) ? AC_LD : 0; self->flags[streamIndex] |= (asc->m_epConfig >= 0) ? AC_ER : 0; if (asc->m_aot == AOT_USAC) { self->flags[streamIndex] |= AC_USAC; self->flags[streamIndex] |= (asc->m_sc.m_usacConfig.element[0].m_stereoConfigIndex > 0) ? AC_MPS_PRESENT : 0; } if (asc->m_aot == AOT_DRM_AAC) { self->flags[streamIndex] |= AC_DRM | AC_SBRCRC | AC_SCALABLE; } if (asc->m_aot == AOT_DRM_SURROUND) { self->flags[streamIndex] |= AC_DRM | AC_SBRCRC | AC_SCALABLE | AC_MPS_PRESENT; FDK_ASSERT(!asc->m_psPresentFlag); } if ((asc->m_aot == AOT_AAC_SCAL) || (asc->m_aot == AOT_ER_AAC_SCAL)) { self->flags[streamIndex] |= AC_SCALABLE; } if ((asc->m_epConfig >= 0) && (asc->m_channelConfiguration <= 0)) { /* we have to know the number of channels otherwise no decoding is possible */ return AAC_DEC_UNSUPPORTED_ER_FORMAT; } self->streamInfo.epConfig = asc->m_epConfig; /* self->hInput->asc.m_epConfig = asc->m_epConfig; */ if (asc->m_epConfig > 1) return AAC_DEC_UNSUPPORTED_ER_FORMAT; /* Check if samplerate changed. */ if ((self->samplingRateInfo[streamIndex].samplingRate != asc->m_samplingFrequency) || (self->streamInfo.aacSamplesPerFrame != (INT)asc->m_samplesPerFrame / downscaleFactor)) { AAC_DECODER_ERROR error; ascChanged = 1; if (configMode & AC_CM_ALLOC_MEM) { /* Update samplerate info. */ error = getSamplingRateInfo( &self->samplingRateInfo[streamIndex], asc->m_samplesPerFrame, asc->m_samplingFrequencyIndex, asc->m_samplingFrequency); if (error != AAC_DEC_OK) { return error; } self->streamInfo.aacSampleRate = self->samplingRateInfo[0].samplingRate / self->downscaleFactor; self->streamInfo.aacSamplesPerFrame = asc->m_samplesPerFrame / self->downscaleFactor; if (self->streamInfo.aacSampleRate <= 0) { return AAC_DEC_UNSUPPORTED_SAMPLINGRATE; } } } /* Check if amount of channels has changed. */ if (self->ascChannels[streamIndex] != ascChannels) { ascChanged = 1; } /* detect config change */ if (configMode & AC_CM_DET_CFG_CHANGE) { if (ascChanged != 0) { *configChanged = 1; } return err; } /* set AC_USAC_SCFGI3 globally if any usac element uses */ switch (asc->m_aot) { case AOT_USAC: if (self->sbrEnabled) { for (int _el = 0; _el < (int)self->pUsacConfig[streamIndex]->m_usacNumElements; _el++) { int el = elementOffset + _el; if (IS_USAC_CHANNEL_ELEMENT(self->elements[el])) { if (usacStereoConfigIndex < 0) { usacStereoConfigIndex = self->usacStereoConfigIndex[el]; } else { if ((usacStereoConfigIndex != self->usacStereoConfigIndex[el]) || (self->usacStereoConfigIndex[el] > 0)) { goto bail; } } } } if (usacStereoConfigIndex < 0) { goto bail; } if (usacStereoConfigIndex == 3) { self->flags[streamIndex] |= AC_USAC_SCFGI3; } } break; default: break; } if (*configChanged) { /* Set up QMF domain for AOTs with explicit signalling of SBR and or MPS. This is to be able to play out the first frame alway with the correct frame size and sampling rate even in case of concealment. */ switch (asc->m_aot) { case AOT_USAC: if (self->sbrEnabled) { const UCHAR map_sbrRatio_2_nAnaBands[] = {16, 24, 32}; FDK_ASSERT(asc->m_sc.m_usacConfig.m_sbrRatioIndex > 0); FDK_ASSERT(streamIndex == 0); self->qmfDomain.globalConf.nInputChannels_requested = ascChannels; self->qmfDomain.globalConf.nOutputChannels_requested = (usacStereoConfigIndex == 1) ? 2 : ascChannels; self->qmfDomain.globalConf.flags_requested = 0; self->qmfDomain.globalConf.nBandsAnalysis_requested = map_sbrRatio_2_nAnaBands[asc->m_sc.m_usacConfig.m_sbrRatioIndex - 1]; self->qmfDomain.globalConf.nBandsSynthesis_requested = 64; self->qmfDomain.globalConf.nQmfTimeSlots_requested = (asc->m_sc.m_usacConfig.m_sbrRatioIndex == 1) ? 64 : 32; self->qmfDomain.globalConf.nQmfOvTimeSlots_requested = (asc->m_sc.m_usacConfig.m_sbrRatioIndex == 1) ? 12 : 6; self->qmfDomain.globalConf.nQmfProcBands_requested = 64; self->qmfDomain.globalConf.nQmfProcChannels_requested = 1; self->qmfDomain.globalConf.parkChannel = (usacStereoConfigIndex == 3) ? 1 : 0; self->qmfDomain.globalConf.parkChannel_requested = (usacStereoConfigIndex == 3) ? 1 : 0; self->qmfDomain.globalConf.qmfDomainExplicitConfig = 1; } break; case AOT_ER_AAC_ELD: if (self->mpsEnableCurr && asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign) { SAC_INPUT_CONFIG sac_interface = (self->sbrEnabled && self->hSbrDecoder) ? SAC_INTERFACE_QMF : SAC_INTERFACE_TIME; mpegSurroundDecoder_ConfigureQmfDomain( (CMpegSurroundDecoder *)self->pMpegSurroundDecoder, sac_interface, (UINT)self->streamInfo.aacSampleRate, asc->m_aot); self->qmfDomain.globalConf.qmfDomainExplicitConfig = 1; } break; default: self->qmfDomain.globalConf.qmfDomainExplicitConfig = 0; /* qmfDomain is initialized by SBR and MPS init functions if required */ break; } /* Allocate all memory structures for each channel */ { int ch = aacChannelsOffset; for (int _ch = 0; _ch < ascChannels; _ch++) { if (ch >= (8)) { goto bail; } self->pAacDecoderChannelInfo[ch] = GetAacDecoderChannelInfo(ch); /* This is temporary until the DynamicData is split into two or more regions! The memory could be reused after completed core decoding. */ if (self->pAacDecoderChannelInfo[ch] == NULL) { goto bail; } ch++; } int chIdx = aacChannelsOffsetIdx; ch = aacChannelsOffset; int _numElements; _numElements = (((8)) + (8)); if (self->flags[streamIndex] & (AC_RSV603DA | AC_USAC)) { _numElements = (int)asc->m_sc.m_usacConfig.m_usacNumElements; } for (int _el = 0; _el < _numElements; _el++) { int el_channels = 0; int el = elementOffset + _el; if (self->flags[streamIndex] & (AC_ER | AC_LD | AC_ELD | AC_RSV603DA | AC_USAC | AC_RSVD50)) { if (ch >= ascChannels) { break; } } switch (self->elements[el]) { case ID_SCE: case ID_CPE: case ID_LFE: case ID_USAC_SCE: case ID_USAC_CPE: case ID_USAC_LFE: el_channels = CAacDecoder_GetELChannels( self->elements[el], self->usacStereoConfigIndex[el]); { self->pAacDecoderChannelInfo[ch]->pComStaticData = (CAacDecoderCommonStaticData *)FDKcalloc( 1, sizeof(CAacDecoderCommonStaticData)); if (self->pAacDecoderChannelInfo[ch]->pComStaticData == NULL) { goto bail; } if (ch == aacChannelsOffset) { self->pAacDecoderChannelInfo[ch]->pComData = (CAacDecoderCommonData *)GetWorkBufferCore6(); self->pAacDecoderChannelInfo[ch] ->pComStaticData->pWorkBufferCore1 = GetWorkBufferCore1(); } else { self->pAacDecoderChannelInfo[ch]->pComData = (CAacDecoderCommonData *)FDKaalloc( sizeof(CAacDecoderCommonData), ALIGNMENT_DEFAULT); self->pAacDecoderChannelInfo[ch] ->pComStaticData->pWorkBufferCore1 = self->pAacDecoderChannelInfo[aacChannelsOffset] ->pComStaticData->pWorkBufferCore1; } if ((self->pAacDecoderChannelInfo[ch]->pComData == NULL) || (self->pAacDecoderChannelInfo[ch] ->pComStaticData->pWorkBufferCore1 == NULL)) { goto bail; } self->pAacDecoderChannelInfo[ch]->pDynData = &(self->pAacDecoderChannelInfo[ch] ->pComData->pAacDecoderDynamicData[0]); self->pAacDecoderChannelInfo[ch]->pSpectralCoefficient = (SPECTRAL_PTR)&self->workBufferCore2[ch * 1024]; if (el_channels == 2) { if (ch >= (8) - 1) { return AAC_DEC_UNSUPPORTED_CHANNELCONFIG; } self->pAacDecoderChannelInfo[ch + 1]->pComData = self->pAacDecoderChannelInfo[ch]->pComData; self->pAacDecoderChannelInfo[ch + 1]->pComStaticData = self->pAacDecoderChannelInfo[ch]->pComStaticData; self->pAacDecoderChannelInfo[ch + 1] ->pComStaticData->pWorkBufferCore1 = self->pAacDecoderChannelInfo[ch] ->pComStaticData->pWorkBufferCore1; self->pAacDecoderChannelInfo[ch + 1]->pDynData = &(self->pAacDecoderChannelInfo[ch] ->pComData->pAacDecoderDynamicData[1]); self->pAacDecoderChannelInfo[ch + 1]->pSpectralCoefficient = (SPECTRAL_PTR)&self->workBufferCore2[(ch + 1) * 1024]; } ch += el_channels; } chIdx += el_channels; break; default: break; } if (self->elements[el] == ID_END) { break; } el++; } chIdx = aacChannelsOffsetIdx; ch = aacChannelsOffset; for (int _ch = 0; _ch < ascChannels; _ch++) { /* Allocate persistent channel memory */ { self->pAacDecoderStaticChannelInfo[ch] = GetAacDecoderStaticChannelInfo(ch); if (self->pAacDecoderStaticChannelInfo[ch] == NULL) { goto bail; } self->pAacDecoderStaticChannelInfo[ch]->pOverlapBuffer = GetOverlapBuffer(ch); /* This area size depends on the AOT */ if (self->pAacDecoderStaticChannelInfo[ch]->pOverlapBuffer == NULL) { goto bail; } if (self->flags[streamIndex] & (AC_USAC | AC_RSVD50 | AC_RSV603DA /*|AC_BSAC*/)) { self->pAacDecoderStaticChannelInfo[ch]->hArCo = CArco_Create(); if (self->pAacDecoderStaticChannelInfo[ch]->hArCo == NULL) { goto bail; } } if (!(self->flags[streamIndex] & (AC_USAC | AC_RSV603DA))) { CPns_UpdateNoiseState( &self->pAacDecoderChannelInfo[ch]->data.aac.PnsData, &self->pAacDecoderStaticChannelInfo[ch]->pnsCurrentSeed, self->pAacDecoderChannelInfo[ch]->pComData->pnsRandomSeed); } ch++; } chIdx++; } if (self->flags[streamIndex] & AC_USAC) { for (int _ch = 0; _ch < flushChannels; _ch++) { ch = aacChannelsOffset + _ch; if (self->pTimeDataFlush[ch] == NULL) { self->pTimeDataFlush[ch] = GetTimeDataFlush(ch); if (self->pTimeDataFlush[ch] == NULL) { goto bail; } } } } if (self->flags[streamIndex] & (AC_USAC | AC_RSV603DA)) { int complexStereoPredPossible = 0; ch = aacChannelsOffset; chIdx = aacChannelsOffsetIdx; for (int _el2 = 0; _el2 < (int)asc->m_sc.m_usacConfig.m_usacNumElements; _el2++) { int el2 = elementOffset + _el2; int elCh = 0, ch2; if ((self->elements[el2] == ID_USAC_CPE) && !(self->usacStereoConfigIndex[el2] == 1)) { elCh = 2; } else if (IS_CHANNEL_ELEMENT(self->elements[el2])) { elCh = 1; } if (self->elFlags[el2] & AC_EL_USAC_CP_POSSIBLE) { complexStereoPredPossible = 1; if (self->cpeStaticData[el2] == NULL) { self->cpeStaticData[el2] = GetCpePersistentData(); if (self->cpeStaticData[el2] == NULL) { goto bail; } } } for (ch2 = 0; ch2 < elCh; ch2++) { /* Hook element specific cpeStaticData into channel specific * aacDecoderStaticChannelInfo */ self->pAacDecoderStaticChannelInfo[ch]->pCpeStaticData = self->cpeStaticData[el2]; if (self->pAacDecoderStaticChannelInfo[ch]->pCpeStaticData != NULL) { self->pAacDecoderStaticChannelInfo[ch] ->pCpeStaticData->jointStereoPersistentData .spectralCoeffs[ch2] = self->pAacDecoderStaticChannelInfo[ch] ->concealmentInfo.spectralCoefficient; self->pAacDecoderStaticChannelInfo[ch] ->pCpeStaticData->jointStereoPersistentData.specScale[ch2] = self->pAacDecoderStaticChannelInfo[ch] ->concealmentInfo.specScale; self->pAacDecoderStaticChannelInfo[ch] ->pCpeStaticData->jointStereoPersistentData.scratchBuffer = (FIXP_DBL *)self->pTimeData2; } chIdx++; ch++; } /* for each channel in current element */ if (complexStereoPredPossible && (elCh == 2)) { /* needed once for all channels */ if (self->pAacDecoderChannelInfo[ch - 1] ->pComStaticData->cplxPredictionData == NULL) { self->pAacDecoderChannelInfo[ch - 1] ->pComStaticData->cplxPredictionData = GetCplxPredictionData(); } if (self->pAacDecoderChannelInfo[ch - 1] ->pComStaticData->cplxPredictionData == NULL) { goto bail; } } if (elCh > 0) { self->pAacDecoderStaticChannelInfo[ch - elCh]->nfRandomSeed = (ULONG)0x3039; if (self->elements[el2] == ID_USAC_CPE) { if (asc->m_sc.m_usacConfig.element[el2].m_stereoConfigIndex != 1) { self->pAacDecoderStaticChannelInfo[ch - elCh + 1] ->nfRandomSeed = (ULONG)0x10932; } } } } /* for each element */ } if (ascChannels != self->aacChannels) { /* Make allocated channel count persistent in decoder context. */ self->aacChannels = aacChannelsOffset + ch; } } if (usacResidualDelayCompSamples) { INT delayErr = FDK_Delay_Create(&self->usacResidualDelay, (USHORT)usacResidualDelayCompSamples, 1); if (delayErr) { goto bail; } } /* Make amount of signalled channels persistent in decoder context. */ self->ascChannels[streamIndex] = ascChannels; /* Init the previous channel count values. This is required to avoid a mismatch of memory accesses in the error concealment module and the allocated channel structures in this function. */ self->aacChannelsPrev = 0; } if (self->pAacDecoderChannelInfo[0] != NULL) { self->pDrmBsBuffer = self->pAacDecoderChannelInfo[0] ->pComStaticData->pWorkBufferCore1->DrmBsBuffer; self->drmBsBufferSize = DRM_BS_BUFFER_SIZE; } /* Update structures */ if (*configChanged) { /* Things to be done for each channel, which do not involve allocating memory. Doing these things only on the channels needed for the current configuration (ascChannels) could lead to memory access violation later (error concealment). */ int ch = 0; int chIdx = 0; for (int _ch = 0; _ch < self->ascChannels[streamIndex]; _ch++) { switch (self->streamInfo.aot) { case AOT_ER_AAC_ELD: case AOT_ER_AAC_LD: self->pAacDecoderChannelInfo[ch]->granuleLength = self->streamInfo.aacSamplesPerFrame; break; default: self->pAacDecoderChannelInfo[ch]->granuleLength = self->streamInfo.aacSamplesPerFrame / 8; break; } self->pAacDecoderChannelInfo[ch]->renderMode = initRenderMode; mdct_init(&self->pAacDecoderStaticChannelInfo[ch]->IMdct, self->pAacDecoderStaticChannelInfo[ch]->pOverlapBuffer, OverlapBufferSize); self->pAacDecoderStaticChannelInfo[ch]->last_core_mode = FD_LONG; self->pAacDecoderStaticChannelInfo[ch]->last_lpd_mode = 255; self->pAacDecoderStaticChannelInfo[ch]->last_tcx_pitch = L_DIV; /* Reset DRC control data for this channel */ aacDecoder_drcInitChannelData( &self->pAacDecoderStaticChannelInfo[ch]->drcData); /* Delete mixdown metadata from the past */ pcmDmx_Reset(self->hPcmUtils, PCMDMX_RESET_BS_DATA); /* Reset concealment only if ASC changed. Otherwise it will be done with any config callback. E.g. every time the LATM SMC is present. */ CConcealment_InitChannelData( &self->pAacDecoderStaticChannelInfo[ch]->concealmentInfo, &self->concealCommonData, initRenderMode, self->streamInfo.aacSamplesPerFrame); ch++; chIdx++; } } /* Update externally visible copy of flags */ self->streamInfo.flags = self->flags[0]; if (*configChanged) { int drcDecSampleRate, drcDecFrameSize; if (self->streamInfo.extSamplingRate != 0) { drcDecSampleRate = self->streamInfo.extSamplingRate; drcDecFrameSize = (self->streamInfo.aacSamplesPerFrame * self->streamInfo.extSamplingRate) / self->streamInfo.aacSampleRate; } else { drcDecSampleRate = self->streamInfo.aacSampleRate; drcDecFrameSize = self->streamInfo.aacSamplesPerFrame; } if (FDK_drcDec_Init(self->hUniDrcDecoder, drcDecFrameSize, drcDecSampleRate, self->aacChannels) != 0) goto bail; } if (*configChanged) { if (asc->m_aot == AOT_USAC) { self->hDrcInfo->enable = 0; self->hDrcInfo->progRefLevelPresent = 0; } } if (asc->m_aot == AOT_USAC) { pcmLimiter_SetAttack(self->hLimiter, (5)); pcmLimiter_SetThreshold(self->hLimiter, FL2FXCONST_DBL(0.89125094f)); } return err; bail: CAacDecoder_DeInit(self, 0); return AAC_DEC_OUT_OF_MEMORY; } LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame( HANDLE_AACDECODER self, const UINT flags, PCM_DEC *pTimeData, const INT timeDataSize, const int timeDataChannelOffset) { AAC_DECODER_ERROR ErrorStatus = AAC_DEC_OK; CProgramConfig *pce; HANDLE_FDK_BITSTREAM bs = transportDec_GetBitstream(self->hInput, 0); MP4_ELEMENT_ID type = ID_NONE; /* Current element type */ INT aacChannels = 0; /* Channel counter for channels found in the bitstream */ const int streamIndex = 0; /* index of the current substream */ INT auStartAnchor = (INT)FDKgetValidBits( bs); /* AU start bit buffer position for AU byte alignment */ INT checkSampleRate = self->streamInfo.aacSampleRate; INT CConceal_TDFading_Applied[(8)] = { 0}; /* Initialize status of Time Domain fading */ if (self->aacChannels <= 0) { return AAC_DEC_UNSUPPORTED_CHANNELCONFIG; } /* Any supported base layer valid AU will require more than 16 bits. */ if ((transportDec_GetAuBitsRemaining(self->hInput, 0) < 15) && (flags & (AACDEC_CONCEAL | AACDEC_FLUSH)) == 0) { self->frameOK = 0; ErrorStatus = AAC_DEC_DECODE_FRAME_ERROR; } /* Reset Program Config structure */ pce = &self->pce; CProgramConfig_Reset(pce); CAacDecoder_AncDataReset(&self->ancData); if (!(flags & (AACDEC_CONCEAL | AACDEC_FLUSH)) && !(self->flags[0] & (AC_USAC | AC_RSV603DA))) { int ch; if (self->streamInfo.channelConfig == 0) { /* Init Channel/Element mapping table */ for (ch = 0; ch < (8); ch++) { self->chMapping[ch] = 255; } if (!CProgramConfig_IsValid(pce)) { int el; for (el = 0; el < (3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + 1); el++) { self->elements[el] = ID_NONE; } } } } if (self->downscaleFactor > 1 && (self->flags[0] & AC_ELD)) { self->flags[0] |= AC_ELD_DOWNSCALE; } else { self->flags[0] &= ~AC_ELD_DOWNSCALE; } /* unsupported dsf (aacSampleRate has not yet been divided by dsf) -> divide */ if (self->downscaleFactorInBS > 1 && (self->flags[0] & AC_ELD_DOWNSCALE) == 0) { checkSampleRate = self->streamInfo.aacSampleRate / self->downscaleFactorInBS; } /* Check sampling frequency */ if (self->streamInfo.aacSampleRate <= 0) { /* Instance maybe uninitialized! */ return AAC_DEC_UNSUPPORTED_SAMPLINGRATE; } switch (checkSampleRate) { case 96000: case 88200: case 64000: case 16000: case 12000: case 11025: case 8000: case 7350: case 48000: case 44100: case 32000: case 24000: case 22050: break; default: if (!(self->flags[0] & (AC_USAC | AC_RSVD50 | AC_RSV603DA))) { return AAC_DEC_UNSUPPORTED_SAMPLINGRATE; } break; } if (flags & AACDEC_CLRHIST) { if (!(self->flags[0] & AC_USAC)) { int ch; /* Clear history */ for (ch = 0; ch < self->aacChannels; ch++) { /* Reset concealment */ CConcealment_InitChannelData( &self->pAacDecoderStaticChannelInfo[ch]->concealmentInfo, &self->concealCommonData, self->pAacDecoderChannelInfo[0]->renderMode, self->streamInfo.aacSamplesPerFrame); /* Clear overlap-add buffers to avoid clicks. */ FDKmemclear(self->pAacDecoderStaticChannelInfo[ch]->pOverlapBuffer, OverlapBufferSize * sizeof(FIXP_DBL)); } if (self->streamInfo.channelConfig > 0) { /* Declare the possibly adopted old PCE (with outdated metadata) * invalid. */ CProgramConfig_Init(pce); } } } int pceRead = 0; /* Flag indicating a PCE in the current raw_data_block() */ INT hdaacDecoded = 0; MP4_ELEMENT_ID previous_element = ID_END; /* Last element ID (required for extension payload mapping */ UCHAR previous_element_index = 0; /* Canonical index of last element */ int element_count = 0; /* Element counter for elements found in the bitstream */ int channel_element_count = 0; /* Channel element counter */ MP4_ELEMENT_ID channel_elements[(3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + 1)]; /* Channel elements in bit stream order. */ int el_cnt[ID_LAST] = {0}; /* element counter ( robustness ) */ int element_count_prev_streams = 0; /* Element count of all previous sub streams. */ while ((type != ID_END) && (!(flags & (AACDEC_CONCEAL | AACDEC_FLUSH))) && self->frameOK) { int el_channels; if (!(self->flags[0] & (AC_USAC | AC_RSVD50 | AC_RSV603DA | AC_ELD | AC_SCALABLE | AC_ER))) type = (MP4_ELEMENT_ID)FDKreadBits(bs, 3); else { if (element_count >= (3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + 1)) { self->frameOK = 0; ErrorStatus = AAC_DEC_PARSE_ERROR; break; } type = self->elements[element_count]; } if ((self->flags[streamIndex] & (AC_USAC | AC_RSVD50) && element_count == 0) || (self->flags[streamIndex] & AC_RSV603DA)) { self->flags[streamIndex] &= ~AC_INDEP; if (FDKreadBit(bs)) { self->flags[streamIndex] |= AC_INDEP; } int ch = aacChannels; for (int chIdx = aacChannels; chIdx < self->ascChannels[streamIndex]; chIdx++) { { /* Robustness check */ if (ch >= self->aacChannels) { return AAC_DEC_UNKNOWN; } /* if last frame was broken and this frame is no independent frame, * correct decoding is impossible we need to trigger concealment */ if ((CConcealment_GetLastFrameOk( &self->pAacDecoderStaticChannelInfo[ch]->concealmentInfo, 1) == 0) && !(self->flags[streamIndex] & AC_INDEP)) { self->frameOK = 0; } ch++; } } } if ((INT)FDKgetValidBits(bs) < 0) { self->frameOK = 0; } switch (type) { case ID_SCE: case ID_CPE: case ID_LFE: case ID_USAC_SCE: case ID_USAC_CPE: case ID_USAC_LFE: if (element_count >= (3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + 1)) { self->frameOK = 0; ErrorStatus = AAC_DEC_PARSE_ERROR; break; } el_channels = CAacDecoder_GetELChannels( type, self->usacStereoConfigIndex[element_count]); /* Consistency check */ { int totalAscChannels = 0; for (int i = 0; i < (1 * 1); i++) { totalAscChannels += self->ascChannels[i]; } if ((el_cnt[type] >= (totalAscChannels >> (el_channels - 1))) || (aacChannels > (totalAscChannels - el_channels))) { ErrorStatus = AAC_DEC_DECODE_FRAME_ERROR; self->frameOK = 0; break; } } if (!(self->flags[streamIndex] & (AC_USAC | AC_RSVD50 | AC_RSV603DA))) { int ch; for (ch = 0; ch < el_channels; ch += 1) { CPns_ResetData(&self->pAacDecoderChannelInfo[aacChannels + ch] ->data.aac.PnsData, &self->pAacDecoderChannelInfo[aacChannels + ch] ->pComData->pnsInterChannelData); } } if (self->frameOK) { ErrorStatus = CChannelElement_Read( bs, &self->pAacDecoderChannelInfo[aacChannels], &self->pAacDecoderStaticChannelInfo[aacChannels], self->streamInfo.aot, &self->samplingRateInfo[streamIndex], self->flags[streamIndex], self->elFlags[element_count], self->streamInfo.aacSamplesPerFrame, el_channels, self->streamInfo.epConfig, self->hInput); if (ErrorStatus != AAC_DEC_OK) { self->frameOK = 0; } } if (self->frameOK) { /* Lookup the element and decode it only if it belongs to the current * program */ if (CProgramConfig_LookupElement( pce, self->streamInfo.channelConfig, self->pAacDecoderChannelInfo[aacChannels]->ElementInstanceTag, aacChannels, self->chMapping, self->channelType, self->channelIndices, (8), &previous_element_index, self->elements, type)) { channel_elements[channel_element_count++] = type; aacChannels += el_channels; } else { self->frameOK = 0; } /* Create SBR element for SBR for upsampling for LFE elements, and if SBR was implicitly signaled, because the first frame(s) may not contain SBR payload (broken encoder, bit errors). */ if (self->frameOK && ((self->flags[streamIndex] & AC_SBR_PRESENT) || (self->sbrEnabled == 1)) && !(self->flags[streamIndex] & AC_USAC) /* Is done during explicit config set up */ ) { SBR_ERROR sbrError; UCHAR configMode = 0; UCHAR configChanged = 0; configMode |= AC_CM_ALLOC_MEM; sbrError = sbrDecoder_InitElement( self->hSbrDecoder, self->streamInfo.aacSampleRate, self->streamInfo.extSamplingRate, self->streamInfo.aacSamplesPerFrame, self->streamInfo.aot, type, previous_element_index, 2, /* Signalize that harmonicSBR shall be ignored in the config change detection */ 0, configMode, &configChanged, self->downscaleFactor); if (sbrError != SBRDEC_OK) { /* Do not try to apply SBR because initializing the element * failed. */ self->sbrEnabled = 0; } } } el_cnt[type]++; if (self->frameOK && (self->flags[streamIndex] & AC_USAC) && (type == ID_USAC_CPE || type == ID_USAC_SCE)) { ErrorStatus = aacDecoder_ParseExplicitMpsAndSbr( self, bs, previous_element, previous_element_index, element_count, el_cnt); if (ErrorStatus != AAC_DEC_OK) { self->frameOK = 0; } } break; case ID_CCE: /* Consistency check */ if (el_cnt[type] > self->ascChannels[streamIndex]) { ErrorStatus = AAC_DEC_DECODE_FRAME_ERROR; self->frameOK = 0; break; } if (self->frameOK) { CAacDecoderCommonData commonData; CAacDecoderCommonStaticData commonStaticData; CWorkBufferCore1 workBufferCore1; commonStaticData.pWorkBufferCore1 = &workBufferCore1; /* memory for spectral lines temporal on scratch */ C_AALLOC_SCRATCH_START(mdctSpec, FIXP_DBL, 1024); /* create dummy channel for CCE parsing on stack */ CAacDecoderChannelInfo tmpAacDecoderChannelInfo, *pTmpAacDecoderChannelInfo; FDKmemclear(mdctSpec, 1024 * sizeof(FIXP_DBL)); tmpAacDecoderChannelInfo.pDynData = commonData.pAacDecoderDynamicData; tmpAacDecoderChannelInfo.pComData = &commonData; tmpAacDecoderChannelInfo.pComStaticData = &commonStaticData; tmpAacDecoderChannelInfo.pSpectralCoefficient = (SPECTRAL_PTR)mdctSpec; /* Assume AAC-LC */ tmpAacDecoderChannelInfo.granuleLength = self->streamInfo.aacSamplesPerFrame / 8; /* Reset PNS data. */ CPns_ResetData( &tmpAacDecoderChannelInfo.data.aac.PnsData, &tmpAacDecoderChannelInfo.pComData->pnsInterChannelData); pTmpAacDecoderChannelInfo = &tmpAacDecoderChannelInfo; /* do CCE parsing */ ErrorStatus = CChannelElement_Read( bs, &pTmpAacDecoderChannelInfo, NULL, self->streamInfo.aot, &self->samplingRateInfo[streamIndex], self->flags[streamIndex], AC_EL_GA_CCE, self->streamInfo.aacSamplesPerFrame, 1, self->streamInfo.epConfig, self->hInput); C_AALLOC_SCRATCH_END(mdctSpec, FIXP_DBL, 1024); if (ErrorStatus) { self->frameOK = 0; } if (self->frameOK) { /* Lookup the element and decode it only if it belongs to the * current program */ if (CProgramConfig_LookupElement( pce, self->streamInfo.channelConfig, pTmpAacDecoderChannelInfo->ElementInstanceTag, 0, self->chMapping, self->channelType, self->channelIndices, (8), &previous_element_index, self->elements, type)) { /* decoding of CCE not supported */ } else { self->frameOK = 0; } } } el_cnt[type]++; break; case ID_DSE: { UCHAR element_instance_tag; CDataStreamElement_Read(self, bs, &element_instance_tag, auStartAnchor); if (!CProgramConfig_LookupElement( pce, self->streamInfo.channelConfig, element_instance_tag, 0, self->chMapping, self->channelType, self->channelIndices, (8), &previous_element_index, self->elements, type)) { /* most likely an error in bitstream occured */ // self->frameOK = 0; } } break; case ID_PCE: { int result = CProgramConfigElement_Read(bs, self->hInput, pce, self->streamInfo.channelConfig, auStartAnchor); if (result < 0) { /* Something went wrong */ ErrorStatus = AAC_DEC_PARSE_ERROR; self->frameOK = 0; } else if (result > 1) { /* Built element table */ int elIdx = CProgramConfig_GetElementTable( pce, self->elements, (((8)) + (8)), &self->chMapIndex); /* Reset the remaining tabs */ for (; elIdx < (3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + 1); elIdx++) { self->elements[elIdx] = ID_NONE; } /* Make new number of channel persistent */ self->ascChannels[streamIndex] = pce->NumChannels; /* If PCE is not first element conceal this frame to avoid * inconsistencies */ if (element_count != 0) { self->frameOK = 0; } } pceRead = (result >= 0) ? 1 : 0; } break; case ID_FIL: { int bitCnt = FDKreadBits(bs, 4); /* bs_count */ if (bitCnt == 15) { int esc_count = FDKreadBits(bs, 8); /* bs_esc_count */ bitCnt = esc_count + 14; } /* Convert to bits */ bitCnt <<= 3; while (bitCnt > 0) { ErrorStatus = CAacDecoder_ExtPayloadParse( self, bs, &bitCnt, previous_element, previous_element_index, 1); if (ErrorStatus != AAC_DEC_OK) { self->frameOK = 0; break; } } } break; case ID_EXT: if (element_count >= (3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + 1)) { self->frameOK = 0; ErrorStatus = AAC_DEC_PARSE_ERROR; break; } ErrorStatus = aacDecoder_ParseExplicitMpsAndSbr( self, bs, previous_element, previous_element_index, element_count, el_cnt); break; case ID_USAC_EXT: { if ((element_count - element_count_prev_streams) >= TP_USAC_MAX_ELEMENTS) { self->frameOK = 0; ErrorStatus = AAC_DEC_PARSE_ERROR; break; } /* parse extension element payload q.v. rsv603daExtElement() ISO/IEC DIS 23008-3 Table 30 or UsacExElement() ISO/IEC FDIS 23003-3:2011(E) Table 21 */ int usacExtElementPayloadLength; /* int usacExtElementStart, usacExtElementStop; */ if (FDKreadBit(bs)) { /* usacExtElementPresent */ if (FDKreadBit(bs)) { /* usacExtElementUseDefaultLength */ usacExtElementPayloadLength = self->pUsacConfig[streamIndex] ->element[element_count - element_count_prev_streams] .extElement.usacExtElementDefaultLength; } else { usacExtElementPayloadLength = FDKreadBits(bs, 8); if (usacExtElementPayloadLength == (UINT)(1 << 8) - 1) { UINT valueAdd = FDKreadBits(bs, 16); usacExtElementPayloadLength += (INT)valueAdd - 2; } } if (usacExtElementPayloadLength > 0) { int usacExtBitPos; if (self->pUsacConfig[streamIndex] ->element[element_count - element_count_prev_streams] .extElement.usacExtElementPayloadFrag) { /* usacExtElementStart = */ FDKreadBit(bs); /* usacExtElementStop = */ FDKreadBit(bs); } else { /* usacExtElementStart = 1; */ /* usacExtElementStop = 1; */ } usacExtBitPos = (INT)FDKgetValidBits(bs); USAC_EXT_ELEMENT_TYPE usacExtElementType = self->pUsacConfig[streamIndex] ->element[element_count - element_count_prev_streams] .extElement.usacExtElementType; switch (usacExtElementType) { case ID_EXT_ELE_UNI_DRC: /* uniDrcGain() */ if (streamIndex == 0) { int drcErr; drcErr = FDK_drcDec_ReadUniDrcGain(self->hUniDrcDecoder, bs); if (drcErr != 0) { ErrorStatus = AAC_DEC_PARSE_ERROR; } } break; default: break; } /* Skip any remaining bits of extension payload */ usacExtBitPos = (usacExtElementPayloadLength * 8) - (usacExtBitPos - (INT)FDKgetValidBits(bs)); if (usacExtBitPos < 0) { self->frameOK = 0; ErrorStatus = AAC_DEC_PARSE_ERROR; } FDKpushBiDirectional(bs, usacExtBitPos); } } } break; case ID_END: case ID_USAC_END: break; default: ErrorStatus = AAC_DEC_DECODE_FRAME_ERROR; self->frameOK = 0; break; } previous_element = type; element_count++; } /* while ( (type != ID_END) ... ) */ if (!(flags & (AACDEC_CONCEAL | AACDEC_FLUSH))) { /* float decoder checks if bitsLeft is in range 0-7; only prerollAUs are * byteAligned with respect to the first bit */ /* Byte alignment with respect to the first bit of the raw_data_block(). */ if (!(self->flags[streamIndex] & (AC_RSVD50 | AC_USAC)) || (self->prerollAULength[self->accessUnit]) /* indicates preroll */ ) { FDKbyteAlign(bs, auStartAnchor); } /* Check if all bits of the raw_data_block() have been read. */ if (transportDec_GetAuBitsTotal(self->hInput, 0) > 0) { INT unreadBits = transportDec_GetAuBitsRemaining(self->hInput, 0); /* for pre-roll frames pre-roll length has to be used instead of total AU * lenght */ /* unreadBits regarding preroll bounds */ if (self->prerollAULength[self->accessUnit]) { unreadBits = unreadBits - transportDec_GetAuBitsTotal(self->hInput, 0) + (INT)self->prerollAULength[self->accessUnit]; } if (((self->flags[streamIndex] & (AC_RSVD50 | AC_USAC)) && ((unreadBits < 0) || (unreadBits > 7)) && !(self->prerollAULength[self->accessUnit])) || ((!(self->flags[streamIndex] & (AC_RSVD50 | AC_USAC)) || (self->prerollAULength[self->accessUnit])) && (unreadBits != 0))) { if ((((unreadBits < 0) || (unreadBits > 7)) && self->frameOK) && ((transportDec_GetFormat(self->hInput) == TT_DRM) && (self->flags[streamIndex] & AC_USAC))) { /* Set frame OK because of fill bits. */ self->frameOK = 1; } else { self->frameOK = 0; } /* Do not overwrite current error */ if (ErrorStatus == AAC_DEC_OK && self->frameOK == 0) { ErrorStatus = AAC_DEC_PARSE_ERROR; } /* Always put the bitbuffer at the right position after the current * Access Unit. */ FDKpushBiDirectional(bs, unreadBits); } } /* Check the last element. The terminator (ID_END) has to be the last one * (even if ER syntax is used). */ if (self->frameOK && type != ID_END) { /* Do not overwrite current error */ if (ErrorStatus == AAC_DEC_OK) { ErrorStatus = AAC_DEC_PARSE_ERROR; } self->frameOK = 0; } } if (!(flags & (AACDEC_CONCEAL | AACDEC_FLUSH)) && self->frameOK) { channel_elements[channel_element_count++] = ID_END; } element_count = 0; aacChannels = 0; type = ID_NONE; previous_element_index = 0; while (type != ID_END && element_count < (3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + 1)) { int el_channels; if ((flags & (AACDEC_CONCEAL | AACDEC_FLUSH)) || !self->frameOK) { channel_elements[element_count] = self->elements[element_count]; if (channel_elements[element_count] == ID_NONE) { channel_elements[element_count] = ID_END; } } if (self->flags[streamIndex] & (AC_USAC | AC_RSV603DA | AC_BSAC)) { type = self->elements[element_count]; } else { type = channel_elements[element_count]; } if (!(flags & (AACDEC_CONCEAL | AACDEC_FLUSH)) && self->frameOK) { switch (type) { case ID_SCE: case ID_CPE: case ID_LFE: case ID_USAC_SCE: case ID_USAC_CPE: case ID_USAC_LFE: el_channels = CAacDecoder_GetELChannels( type, self->usacStereoConfigIndex[element_count]); if (!hdaacDecoded) { if (self->pAacDecoderStaticChannelInfo[aacChannels] ->pCpeStaticData != NULL) { self->pAacDecoderStaticChannelInfo[aacChannels] ->pCpeStaticData->jointStereoPersistentData.scratchBuffer = (FIXP_DBL *)pTimeData; } CChannelElement_Decode( &self->pAacDecoderChannelInfo[aacChannels], &self->pAacDecoderStaticChannelInfo[aacChannels], &self->samplingRateInfo[streamIndex], self->flags[streamIndex], self->elFlags[element_count], el_channels); } aacChannels += el_channels; break; case ID_NONE: type = ID_END; break; default: break; } } element_count++; } /* More AAC channels than specified by the ASC not allowed. */ if ((aacChannels == 0 || aacChannels > self->aacChannels) && !(flags & (AACDEC_CONCEAL | AACDEC_FLUSH))) { /* Do not overwrite current error */ if (ErrorStatus == AAC_DEC_OK) { ErrorStatus = AAC_DEC_DECODE_FRAME_ERROR; } self->frameOK = 0; aacChannels = 0; } if (!(flags & (AACDEC_CONCEAL | AACDEC_FLUSH))) { if (TRANSPORTDEC_OK != transportDec_CrcCheck(self->hInput)) { ErrorStatus = AAC_DEC_CRC_ERROR; self->frameOK = 0; } } /* Ensure that in case of concealment a proper error status is set. */ if ((self->frameOK == 0) && (ErrorStatus == AAC_DEC_OK)) { ErrorStatus = AAC_DEC_DECODE_FRAME_ERROR; } if (self->frameOK && (flags & AACDEC_FLUSH)) { aacChannels = self->aacChannelsPrev; /* Because the downmix could be active, its necessary to restore the channel * type and indices. */ FDKmemcpy(self->channelType, self->channelTypePrev, (8) * sizeof(AUDIO_CHANNEL_TYPE)); /* restore */ FDKmemcpy(self->channelIndices, self->channelIndicesPrev, (8) * sizeof(UCHAR)); /* restore */ self->sbrEnabled = self->sbrEnabledPrev; } else { /* store or restore the number of channels and the corresponding info */ if (self->frameOK && !(flags & AACDEC_CONCEAL)) { self->aacChannelsPrev = aacChannels; /* store */ FDKmemcpy(self->channelTypePrev, self->channelType, (8) * sizeof(AUDIO_CHANNEL_TYPE)); /* store */ FDKmemcpy(self->channelIndicesPrev, self->channelIndices, (8) * sizeof(UCHAR)); /* store */ self->sbrEnabledPrev = self->sbrEnabled; } else { if (self->aacChannels > 0) { if ((self->buildUpStatus == AACDEC_RSV60_BUILD_UP_ON) || (self->buildUpStatus == AACDEC_RSV60_BUILD_UP_ON_IN_BAND) || (self->buildUpStatus == AACDEC_USAC_BUILD_UP_ON)) { aacChannels = self->aacChannels; self->aacChannelsPrev = aacChannels; /* store */ } else { aacChannels = self->aacChannelsPrev; /* restore */ } FDKmemcpy(self->channelType, self->channelTypePrev, (8) * sizeof(AUDIO_CHANNEL_TYPE)); /* restore */ FDKmemcpy(self->channelIndices, self->channelIndicesPrev, (8) * sizeof(UCHAR)); /* restore */ self->sbrEnabled = self->sbrEnabledPrev; } } } /* Update number of output channels */ self->streamInfo.aacNumChannels = aacChannels; /* Ensure consistency of IS_OUTPUT_VALID() macro. */ if (aacChannels == 0) { ErrorStatus = AAC_DEC_UNKNOWN; } if (pceRead == 1 && CProgramConfig_IsValid(pce)) { /* Set matrix mixdown infos if available from PCE. */ pcmDmx_SetMatrixMixdownFromPce( self->hPcmUtils, pce->MatrixMixdownIndexPresent, pce->MatrixMixdownIndex, pce->PseudoSurroundEnable); ; } /* If there is no valid data to transfrom into time domain, return. */ if (!IS_OUTPUT_VALID(ErrorStatus)) { return ErrorStatus; } /* Setup the output channel mapping. The table below shows the three * possibilities: # | chCfg | PCE | chMapIndex * ---+-------+-----+------------------ * 1 | > 0 | no | chCfg * 2 | 0 | yes | cChCfg * 3 | 0 | no | aacChannels || 0 * ---+-------+-----+--------+------------------ * Where chCfg is the channel configuration index from ASC and cChCfg is a * corresponding chCfg derived from a given PCE. The variable aacChannels * represents the number of channel found during bitstream decoding. Due to * the structure of the mapping table it can only be used for mapping if its * value is smaller than 7. Otherwise we use the fallback (0) which is a * simple pass-through. The possibility #3 should appear only with MPEG-2 * (ADTS) streams. This is mode is called "implicit channel mapping". */ if ((self->streamInfo.channelConfig == 0) && !pce->isValid) { self->chMapIndex = (aacChannels < 7) ? aacChannels : 0; } /* Inverse transform */ { int c, cIdx; int mapped, fCopyChMap = 1; UCHAR drcChMap[(8)]; if ((self->streamInfo.channelConfig == 0) && CProgramConfig_IsValid(pce)) { /* ISO/IEC 14496-3 says: If a PCE is present, the exclude_mask bits correspond to the audio channels in the SCE, CPE, CCE and LFE syntax elements in the order of their appearance in the PCE. In the case of a CPE, the first transmitted mask bit corresponds to the first channel in the CPE, the second transmitted mask bit to the second channel. In the case of a CCE, a mask bit is transmitted only if the coupling channel is specified to be an independently switched coupling channel. Thus we have to convert the internal channel mapping from "canonical" MPEG to PCE order: */ UCHAR tmpChMap[(8)]; if (CProgramConfig_GetPceChMap(pce, tmpChMap, (8)) == 0) { for (c = 0; c < aacChannels; c += 1) { drcChMap[c] = (self->chMapping[c] == 255) ? 255 : tmpChMap[self->chMapping[c]]; } fCopyChMap = 0; } } if (fCopyChMap != 0) { FDKmemcpy(drcChMap, self->chMapping, (8) * sizeof(UCHAR)); } /* deactivate legacy DRC in case uniDrc is active, i.e. uniDrc payload is * present and one of DRC or Loudness Normalization is switched on */ aacDecoder_drcSetParam( self->hDrcInfo, UNIDRC_PRECEDENCE, FDK_drcDec_GetParam(self->hUniDrcDecoder, DRC_DEC_IS_ACTIVE)); /* Extract DRC control data and map it to channels (without bitstream delay) */ mapped = aacDecoder_drcProlog( self->hDrcInfo, bs, self->pAacDecoderStaticChannelInfo, pce->ElementInstanceTag, drcChMap, aacChannels); if (mapped > 0) { if (!(self->flags[streamIndex] & (AC_USAC | AC_RSV603DA))) { /* If at least one DRC thread has been mapped to a channel there was DRC * data in the bitstream. */ self->flags[streamIndex] |= AC_DRC_PRESENT; } else { self->hDrcInfo->enable = 0; self->hDrcInfo->progRefLevelPresent = 0; ErrorStatus = AAC_DEC_UNSUPPORTED_FORMAT; } } /* Create a reverse mapping table */ UCHAR Reverse_chMapping[((8) * 2)]; for (c = 0; c < aacChannels; c++) { int d; for (d = 0; d < aacChannels - 1; d++) { if (self->chMapping[d] == c) { break; } } Reverse_chMapping[c] = d; } int el; int el_channels; c = 0; cIdx = 0; el_channels = 0; for (el = 0; el < element_count; el++) { int frameOk_butConceal = 0; /* Force frame concealment during mute release active state. */ int concealApplyReturnCode; if (self->flags[streamIndex] & (AC_USAC | AC_RSV603DA | AC_BSAC)) { type = self->elements[el]; } else { type = channel_elements[el]; } { int nElementChannels; nElementChannels = CAacDecoder_GetELChannels(type, self->usacStereoConfigIndex[el]); el_channels += nElementChannels; if (nElementChannels == 0) { continue; } } int offset; int elCh = 0; /* "c" iterates in canonical MPEG channel order */ for (; cIdx < el_channels; c++, cIdx++, elCh++) { /* Robustness check */ if (c >= aacChannels) { return AAC_DEC_UNKNOWN; } CAacDecoderChannelInfo *pAacDecoderChannelInfo = self->pAacDecoderChannelInfo[c]; CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo = self->pAacDecoderStaticChannelInfo[c]; /* Setup offset for time buffer traversal. */ { pAacDecoderStaticChannelInfo = self->pAacDecoderStaticChannelInfo[Reverse_chMapping[c]]; offset = FDK_chMapDescr_getMapValue( &self->mapDescr, Reverse_chMapping[cIdx], self->chMapIndex) * timeDataChannelOffset; } if (flags & AACDEC_FLUSH) { /* Clear pAacDecoderChannelInfo->pSpectralCoefficient because with * AACDEC_FLUSH set it contains undefined data. */ FDKmemclear(pAacDecoderChannelInfo->pSpectralCoefficient, sizeof(FIXP_DBL) * self->streamInfo.aacSamplesPerFrame); } /* if The ics info is not valid and it will be stored and used in the * following concealment method, mark the frame as erroneous */ { CIcsInfo *pIcsInfo = &pAacDecoderChannelInfo->icsInfo; CConcealmentInfo *hConcealmentInfo = &pAacDecoderStaticChannelInfo->concealmentInfo; const int mute_release_active = (self->frameOK && !(flags & AACDEC_CONCEAL)) && ((hConcealmentInfo->concealState >= ConcealState_Mute) && (hConcealmentInfo->cntValidFrames + 1 <= hConcealmentInfo->pConcealParams->numMuteReleaseFrames)); const int icsIsInvalid = (GetScaleFactorBandsTransmitted(pIcsInfo) > GetScaleFactorBandsTotal(pIcsInfo)); const int icsInfoUsedinFadeOut = !(pAacDecoderChannelInfo->renderMode == AACDEC_RENDER_LPD && pAacDecoderStaticChannelInfo->last_lpd_mode == 0); if (icsInfoUsedinFadeOut && icsIsInvalid && !mute_release_active) { self->frameOK = 0; } } /* Conceal defective spectral data */ { CAacDecoderChannelInfo **ppAacDecoderChannelInfo = &pAacDecoderChannelInfo; CAacDecoderStaticChannelInfo **ppAacDecoderStaticChannelInfo = &pAacDecoderStaticChannelInfo; { concealApplyReturnCode = CConcealment_Apply( &(*ppAacDecoderStaticChannelInfo)->concealmentInfo, *ppAacDecoderChannelInfo, *ppAacDecoderStaticChannelInfo, &self->samplingRateInfo[streamIndex], self->streamInfo.aacSamplesPerFrame, pAacDecoderStaticChannelInfo->last_lpd_mode, (self->frameOK && !(flags & AACDEC_CONCEAL)), self->flags[streamIndex]); } } if (concealApplyReturnCode == -1) { frameOk_butConceal = 1; } if (flags & (AACDEC_INTR)) { /* Reset DRC control data for this channel */ aacDecoder_drcInitChannelData(&pAacDecoderStaticChannelInfo->drcData); } if (flags & (AACDEC_CLRHIST)) { if (!(self->flags[0] & AC_USAC)) { /* Reset DRC control data for this channel */ aacDecoder_drcInitChannelData( &pAacDecoderStaticChannelInfo->drcData); } } /* The DRC module demands to be called with the gain field holding the * gain scale. */ self->extGain[0] = (FIXP_DBL)AACDEC_DRC_GAIN_SCALING; /* DRC processing */ aacDecoder_drcApply( self->hDrcInfo, self->hSbrDecoder, pAacDecoderChannelInfo, &pAacDecoderStaticChannelInfo->drcData, self->extGain, c, self->streamInfo.aacSamplesPerFrame, self->sbrEnabled ); if (timeDataSize < timeDataChannelOffset * self->aacChannels) { ErrorStatus = AAC_DEC_OUTPUT_BUFFER_TOO_SMALL; break; } if (self->flushStatus && (self->flushCnt > 0) && !(flags & AACDEC_CONCEAL)) { FDKmemclear(pTimeData + offset, sizeof(PCM_DEC) * self->streamInfo.aacSamplesPerFrame); } else switch (pAacDecoderChannelInfo->renderMode) { case AACDEC_RENDER_IMDCT: CBlock_FrequencyToTime( pAacDecoderStaticChannelInfo, pAacDecoderChannelInfo, pTimeData + offset, self->streamInfo.aacSamplesPerFrame, (self->frameOK && !(flags & AACDEC_CONCEAL) && !frameOk_butConceal), pAacDecoderChannelInfo->pComStaticData->pWorkBufferCore1 ->mdctOutTemp, self->aacOutDataHeadroom, self->elFlags[el], elCh); self->extGainDelay = self->streamInfo.aacSamplesPerFrame; break; case AACDEC_RENDER_ELDFB: { CBlock_FrequencyToTimeLowDelay( pAacDecoderStaticChannelInfo, pAacDecoderChannelInfo, pTimeData + offset, self->streamInfo.aacSamplesPerFrame); self->extGainDelay = (self->streamInfo.aacSamplesPerFrame * 2 - self->streamInfo.aacSamplesPerFrame / 2 - 1) / 2; } break; case AACDEC_RENDER_LPD: CLpd_RenderTimeSignal( pAacDecoderStaticChannelInfo, pAacDecoderChannelInfo, pTimeData + offset, self->streamInfo.aacSamplesPerFrame, &self->samplingRateInfo[streamIndex], (self->frameOK && !(flags & AACDEC_CONCEAL) && !frameOk_butConceal), self->aacOutDataHeadroom, flags, self->flags[streamIndex]); self->extGainDelay = self->streamInfo.aacSamplesPerFrame; break; default: ErrorStatus = AAC_DEC_UNKNOWN; break; } /* TimeDomainFading */ if (!CConceal_TDFading_Applied[c]) { CConceal_TDFading_Applied[c] = CConcealment_TDFading( self->streamInfo.aacSamplesPerFrame, &self->pAacDecoderStaticChannelInfo[c], self->aacOutDataHeadroom, pTimeData + offset, 0); if (c + 1 < (8) && c < aacChannels - 1) { /* update next TDNoise Seed to avoid muting in case of Parametric * Stereo */ self->pAacDecoderStaticChannelInfo[c + 1] ->concealmentInfo.TDNoiseSeed = self->pAacDecoderStaticChannelInfo[c] ->concealmentInfo.TDNoiseSeed; } } } } if (self->flags[streamIndex] & AC_USAC) { int bsPseudoLr = 0; mpegSurroundDecoder_IsPseudoLR( (CMpegSurroundDecoder *)self->pMpegSurroundDecoder, &bsPseudoLr); /* ISO/IEC 23003-3, 7.11.2.6 Modification of core decoder output (pseudo * LR) */ if ((aacChannels == 2) && bsPseudoLr) { int i, offset2; const FIXP_SGL invSqrt2 = FL2FXCONST_SGL(0.707106781186547f); PCM_DEC *pTD = pTimeData; offset2 = timeDataChannelOffset; for (i = 0; i < self->streamInfo.aacSamplesPerFrame; i++) { FIXP_DBL L = PCM_DEC2FIXP_DBL(pTD[0]); FIXP_DBL R = PCM_DEC2FIXP_DBL(pTD[offset2]); L = fMult(L, invSqrt2); R = fMult(R, invSqrt2); pTD[0] = L + R; pTD[offset2] = L - R; pTD++; } } } /* Extract DRC control data and map it to channels (with bitstream delay) */ mapped = aacDecoder_drcEpilog( self->hDrcInfo, bs, self->pAacDecoderStaticChannelInfo, pce->ElementInstanceTag, drcChMap, aacChannels); if (mapped > 0) { if (!(self->flags[streamIndex] & (AC_USAC | AC_RSV603DA))) { /* If at least one DRC thread has been mapped to a channel there was DRC * data in the bitstream. */ self->flags[streamIndex] |= AC_DRC_PRESENT; } else { self->hDrcInfo->enable = 0; self->hDrcInfo->progRefLevelPresent = 0; ErrorStatus = AAC_DEC_UNSUPPORTED_FORMAT; } } } /* Add additional concealment delay */ self->streamInfo.outputDelay += CConcealment_GetDelay(&self->concealCommonData) * self->streamInfo.aacSamplesPerFrame; /* Map DRC data to StreamInfo structure */ aacDecoder_drcGetInfo(self->hDrcInfo, &self->streamInfo.drcPresMode, &self->streamInfo.drcProgRefLev); /* Reorder channel type information tables. */ if (!(self->flags[0] & AC_RSV603DA)) { AUDIO_CHANNEL_TYPE types[(8)]; UCHAR idx[(8)]; int c; int mapValue; FDK_ASSERT(sizeof(self->channelType) == sizeof(types)); FDK_ASSERT(sizeof(self->channelIndices) == sizeof(idx)); FDKmemcpy(types, self->channelType, sizeof(types)); FDKmemcpy(idx, self->channelIndices, sizeof(idx)); for (c = 0; c < aacChannels; c++) { mapValue = FDK_chMapDescr_getMapValue(&self->mapDescr, c, self->chMapIndex); self->channelType[mapValue] = types[c]; self->channelIndices[mapValue] = idx[c]; } } self->blockNumber++; return ErrorStatus; } /*! \brief returns the streaminfo pointer The function hands back a pointer to the streaminfo structure \return pointer to the struct */ LINKSPEC_CPP CStreamInfo *CAacDecoder_GetStreamInfo(HANDLE_AACDECODER self) { if (!self) { return NULL; } return &self->streamInfo; } fdk-aac-2.0.2/libAACdec/src/aacdecoder.h000066400000000000000000000451151404225310700175460ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Josef Hoepfl Description: *******************************************************************************/ #ifndef AACDECODER_H #define AACDECODER_H #include "common_fix.h" #include "FDK_bitstream.h" #include "channel.h" #include "tpdec_lib.h" #include "FDK_audio.h" #include "block.h" #include "genericStds.h" #include "FDK_qmf_domain.h" #include "sbrdecoder.h" #include "aacdec_drc.h" #include "pcmdmx_lib.h" #include "FDK_drcDecLib.h" #include "limiter.h" #include "FDK_delay.h" #define TIME_DATA_FLUSH_SIZE (128) #define TIME_DATA_FLUSH_SIZE_SF (7) #define AACDEC_MAX_NUM_PREROLL_AU_USAC (3) #if (AACDEC_MAX_NUM_PREROLL_AU < 3) #undef AACDEC_MAX_NUM_PREROLL_AU #define AACDEC_MAX_NUM_PREROLL_AU AACDEC_MAX_NUM_PREROLL_AU_USAC #endif typedef struct AAC_DECODER_INSTANCE *HANDLE_AACDECODER; enum { L = 0, R = 1 }; typedef struct { unsigned char *buffer; int bufferSize; int offset[8]; int nrElements; } CAncData; typedef enum { NOT_DEFINED = -1, MODE_HQ = 0, MODE_LP = 1 } QMF_MODE; typedef struct { int bsDelay; } SBR_PARAMS; enum { AACDEC_FLUSH_OFF = 0, AACDEC_RSV60_CFG_CHANGE_ATSC_FLUSH_ON = 1, AACDEC_RSV60_DASH_IPF_ATSC_FLUSH_ON = 2, AACDEC_USAC_DASH_IPF_FLUSH_ON = 3 }; enum { AACDEC_BUILD_UP_OFF = 0, AACDEC_RSV60_BUILD_UP_ON = 1, AACDEC_RSV60_BUILD_UP_ON_IN_BAND = 2, AACDEC_USAC_BUILD_UP_ON = 3, AACDEC_RSV60_BUILD_UP_IDLE = 4, AACDEC_RSV60_BUILD_UP_IDLE_IN_BAND = 5 }; typedef struct { /* Usac Extension Elements */ USAC_EXT_ELEMENT_TYPE usacExtElementType[(3)]; UINT usacExtElementDefaultLength[(3)]; UCHAR usacExtElementPayloadFrag[(3)]; } CUsacCoreExtensions; /* AAC decoder (opaque toward userland) struct declaration */ struct AAC_DECODER_INSTANCE { INT aacChannels; /*!< Amount of AAC decoder channels allocated. */ INT ascChannels[(1 * 1)]; /*!< Amount of AAC decoder channels signalled in ASC. */ INT blockNumber; /*!< frame counter */ INT nrOfLayers; INT outputInterleaved; /*!< PCM output format (interleaved/none interleaved). */ INT aacOutDataHeadroom; /*!< Headroom of the output time signal to prevent clipping */ HANDLE_TRANSPORTDEC hInput; /*!< Transport layer handle. */ SamplingRateInfo samplingRateInfo[(1 * 1)]; /*!< Sampling Rate information table */ UCHAR frameOK; /*!< Will be unset if a consistency check, e.g. CRC etc. fails */ UINT flags[(1 * 1)]; /*!< Flags for internal decoder use. DO NOT USE self::streaminfo::flags ! */ UINT elFlags[(3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + 1)]; /*!< Flags for internal decoder use (element specific). DO NOT USE self::streaminfo::flags ! */ MP4_ELEMENT_ID elements[(3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + 1)]; /*!< Table where the element Id's are listed */ UCHAR elTags[(3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + 1)]; /*!< Table where the elements id Tags are listed */ UCHAR chMapping[((8) * 2)]; /*!< Table of MPEG canonical order to bitstream channel order mapping. */ AUDIO_CHANNEL_TYPE channelType[(8)]; /*!< Audio channel type of each output audio channel (from 0 upto numChannels). */ UCHAR channelIndices[(8)]; /*!< Audio channel index for each output audio channel (from 0 upto numChannels). */ /* See ISO/IEC 13818-7:2005(E), 8.5.3.2 Explicit channel mapping using a * program_config_element() */ FDK_channelMapDescr mapDescr; /*!< Describes the output channel mapping. */ UCHAR chMapIndex; /*!< Index to access one line of the channelOutputMapping table. This is required because not all 8 channel configurations have the same output mapping. */ INT sbrDataLen; /*!< Expected length of the SBR remaining in bitbuffer after the AAC payload has been pared. */ CProgramConfig pce; CStreamInfo streamInfo; /*!< Pointer to StreamInfo data (read from the bitstream) */ CAacDecoderChannelInfo *pAacDecoderChannelInfo[(8)]; /*!< Temporal channel memory */ CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo[(8)]; /*!< Persistent channel memory */ FIXP_DBL *workBufferCore1; FIXP_DBL *workBufferCore2; PCM_DEC *pTimeData2; INT timeData2Size; CpePersistentData *cpeStaticData[( 3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + 1)]; /*!< Pointer to persistent data shared by both channels of a CPE. This structure is allocated once for each CPE. */ CConcealParams concealCommonData; CConcealmentMethod concealMethodUser; CUsacCoreExtensions usacCoreExt; /*!< Data and handles to extend USAC FD/LPD core decoder (SBR, MPS, ...) */ UINT numUsacElements[(1 * 1)]; UCHAR usacStereoConfigIndex[(3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + 1)]; const CSUsacConfig *pUsacConfig[(1 * 1)]; INT nbDiv; /*!< number of frame divisions in LPD-domain */ UCHAR useLdQmfTimeAlign; INT aacChannelsPrev; /*!< The amount of AAC core channels of the last successful decode call. */ AUDIO_CHANNEL_TYPE channelTypePrev[(8)]; /*!< Array holding the channelType values of the last successful decode call. */ UCHAR channelIndicesPrev[(8)]; /*!< Array holding the channelIndices values of the last successful decode call. */ UCHAR downscaleFactor; /*!< Variable to store a supported ELD downscale factor of 1, 2, 3 or 4 */ UCHAR downscaleFactorInBS; /*!< Variable to store the (not necessarily supported) ELD downscale factor discovered in the bitstream */ HANDLE_SBRDECODER hSbrDecoder; /*!< SBR decoder handle. */ UCHAR sbrEnabled; /*!< flag to store if SBR has been detected */ UCHAR sbrEnabledPrev; /*!< flag to store if SBR has been detected from previous frame */ UCHAR psPossible; /*!< flag to store if PS is possible */ SBR_PARAMS sbrParams; /*!< struct to store all sbr parameters */ UCHAR *pDrmBsBuffer; /*!< Pointer to dynamic buffer which is used to reverse the bits of the DRM SBR payload */ USHORT drmBsBufferSize; /*!< Size of the dynamic buffer which is used to reverse the bits of the DRM SBR payload */ FDK_QMF_DOMAIN qmfDomain; /*!< Instance of module for QMF domain data handling */ QMF_MODE qmfModeCurr; /*!< The current QMF mode */ QMF_MODE qmfModeUser; /*!< The QMF mode requested by the library user */ HANDLE_AAC_DRC hDrcInfo; /*!< handle to DRC data structure */ INT metadataExpiry; /*!< Metadata expiry time in milli-seconds. */ void *pMpegSurroundDecoder; /*!< pointer to mpeg surround decoder structure */ UCHAR mpsEnableUser; /*!< MPS enable user flag */ UCHAR mpsEnableCurr; /*!< MPS enable decoder state */ UCHAR mpsApplicable; /*!< MPS applicable */ SCHAR mpsOutputMode; /*!< setting: normal = 0, binaural = 1, stereo = 2, 5.1ch = 3 */ INT mpsOutChannelsLast; /*!< The amount of channels returned by the last successful MPS decoder call. */ INT mpsFrameSizeLast; /*!< The frame length returned by the last successful MPS decoder call. */ CAncData ancData; /*!< structure to handle ancillary data */ HANDLE_PCM_DOWNMIX hPcmUtils; /*!< privat data for the PCM utils. */ TDLimiterPtr hLimiter; /*!< Handle of time domain limiter. */ UCHAR limiterEnableUser; /*!< The limiter configuration requested by the library user */ UCHAR limiterEnableCurr; /*!< The current limiter configuration. */ FIXP_DBL extGain[1]; /*!< Gain that must be applied to the output signal. */ UINT extGainDelay; /*!< Delay that must be accounted for extGain. */ HANDLE_DRC_DECODER hUniDrcDecoder; UCHAR multibandDrcPresent; UCHAR numTimeSlots; UINT loudnessInfoSetPosition[3]; SCHAR defaultTargetLoudness; INT_PCM *pTimeDataFlush[((8) * 2)]; /*!< Pointer to the flushed time data which will be used for the crossfade in case of an USAC DASH IPF config change */ UCHAR flushStatus; /*!< Indicates flush status: on|off */ SCHAR flushCnt; /*!< Flush frame counter */ UCHAR buildUpStatus; /*!< Indicates build up status: on|off */ SCHAR buildUpCnt; /*!< Build up frame counter */ UCHAR hasAudioPreRoll; /*!< Indicates preRoll status: on|off */ UINT prerollAULength[AACDEC_MAX_NUM_PREROLL_AU + 1]; /*!< Relative offset of the prerollAU end position to the AU start position in the bitstream */ INT accessUnit; /*!< Number of the actual processed preroll accessUnit */ UCHAR applyCrossfade; /*!< if set crossfade for seamless stream switching is applied */ FDK_SignalDelay usacResidualDelay; /*!< Delay residual signal to compensate for eSBR delay of DMX signal in case of stereoConfigIndex==2. */ }; #define AAC_DEBUG_EXTHLP \ "\ --- AAC-Core ---\n\ 0x00010000 Header data\n\ 0x00020000 CRC data\n\ 0x00040000 Channel info\n\ 0x00080000 Section data\n\ 0x00100000 Scalefactor data\n\ 0x00200000 Pulse data\n\ 0x00400000 Tns data\n\ 0x00800000 Quantized spectrum\n\ 0x01000000 Requantized spectrum\n\ 0x02000000 Time output\n\ 0x04000000 Fatal errors\n\ 0x08000000 Buffer fullness\n\ 0x10000000 Average bitrate\n\ 0x20000000 Synchronization\n\ 0x40000000 Concealment\n\ 0x7FFF0000 all AAC-Core-Info\n\ " /** * \brief Synchronise QMF mode for all modules using QMF data. * \param self decoder handle */ void CAacDecoder_SyncQmfMode(HANDLE_AACDECODER self); /** * \brief Signal a bit stream interruption to the decoder * \param self decoder handle */ void CAacDecoder_SignalInterruption(HANDLE_AACDECODER self); /*! \brief Initialize ancillary buffer \ancData Pointer to ancillary data structure \buffer Pointer to (external) anc data buffer \size Size of the buffer pointed on by buffer \return Error code */ AAC_DECODER_ERROR CAacDecoder_AncDataInit(CAncData *ancData, unsigned char *buffer, int size); /*! \brief Get one ancillary data element \ancData Pointer to ancillary data structure \index Index of the anc data element to get \ptr Pointer to a buffer receiving a pointer to the requested anc data element \size Pointer to a buffer receiving the length of the requested anc data element \return Error code */ AAC_DECODER_ERROR CAacDecoder_AncDataGet(CAncData *ancData, int index, unsigned char **ptr, int *size); /* initialization of aac decoder */ LINKSPEC_H HANDLE_AACDECODER CAacDecoder_Open(TRANSPORT_TYPE bsFormat); /* Initialization of channel elements */ LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, UCHAR configMode, UCHAR *configChanged); /*! \brief Decodes one aac frame The function decodes one aac frame. The decoding of coupling channel elements are not supported. The transport layer might signal, that the data of the current frame is invalid, e.g. as a result of a packet loss in streaming mode. The bitstream position of transportDec_GetBitstream(self->hInput) must be exactly the end of the access unit, including all byte alignment bits. For this purpose, the variable auStartAnchor is used. \return error status */ LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_DecodeFrame( HANDLE_AACDECODER self, const UINT flags, PCM_DEC *pTimeData, const INT timeDataSize, const int timeDataChannelOffset); /* Free config dependent AAC memory */ LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_FreeMem(HANDLE_AACDECODER self, const int subStreamIndex); /* Prepare crossfade for USAC DASH IPF config change */ LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_PrepareCrossFade( const INT_PCM *pTimeData, INT_PCM **pTimeDataFlush, const INT numChannels, const INT frameSize, const INT interleaved); /* Apply crossfade for USAC DASH IPF config change */ LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_ApplyCrossFade( INT_PCM *pTimeData, INT_PCM **pTimeDataFlush, const INT numChannels, const INT frameSize, const INT interleaved); /* Set flush and build up mode */ LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_CtrlCFGChange(HANDLE_AACDECODER self, UCHAR flushStatus, SCHAR flushCnt, UCHAR buildUpStatus, SCHAR buildUpCnt); /* Parse preRoll Extension Payload */ LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_PreRollExtensionPayloadParse( HANDLE_AACDECODER self, UINT *numPrerollAU, UINT *prerollAUOffset, UINT *prerollAULength); /* Destroy aac decoder */ LINKSPEC_H void CAacDecoder_Close(HANDLE_AACDECODER self); /* get streaminfo handle from decoder */ LINKSPEC_H CStreamInfo *CAacDecoder_GetStreamInfo(HANDLE_AACDECODER self); #endif /* #ifndef AACDECODER_H */ fdk-aac-2.0.2/libAACdec/src/aacdecoder_lib.cpp000066400000000000000000002300441404225310700207240ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Manuel Jander Description: *******************************************************************************/ #include "aacdecoder_lib.h" #include "aac_ram.h" #include "aacdecoder.h" #include "tpdec_lib.h" #include "FDK_core.h" /* FDK_tools version info */ #include "sbrdecoder.h" #include "conceal.h" #include "aacdec_drc.h" #include "sac_dec_lib.h" #include "pcm_utils.h" /* Decoder library info */ #define AACDECODER_LIB_VL0 3 #define AACDECODER_LIB_VL1 2 #define AACDECODER_LIB_VL2 0 #define AACDECODER_LIB_TITLE "AAC Decoder Lib" #ifdef SUPPRESS_BUILD_DATE_INFO #define AACDECODER_LIB_BUILD_DATE "" #define AACDECODER_LIB_BUILD_TIME "" #else #define AACDECODER_LIB_BUILD_DATE __DATE__ #define AACDECODER_LIB_BUILD_TIME __TIME__ #endif static AAC_DECODER_ERROR setConcealMethod(const HANDLE_AACDECODER self, const INT method); static void aacDecoder_setMetadataExpiry(const HANDLE_AACDECODER self, const INT value) { /* check decoder handle */ if (self != NULL) { INT mdExpFrame = 0; /* default: disable */ if ((value > 0) && (self->streamInfo.aacSamplesPerFrame > 0)) { /* Determine the corresponding number of frames: */ FIXP_DBL frameTime = fDivNorm(self->streamInfo.aacSampleRate, self->streamInfo.aacSamplesPerFrame * 1000); mdExpFrame = fMultIceil(frameTime, value); } /* Configure DRC module */ aacDecoder_drcSetParam(self->hDrcInfo, DRC_DATA_EXPIRY_FRAME, mdExpFrame); /* Configure PCM downmix module */ pcmDmx_SetParam(self->hPcmUtils, DMX_BS_DATA_EXPIRY_FRAME, mdExpFrame); } } LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_GetFreeBytes(const HANDLE_AACDECODER self, UINT *pFreeBytes) { /* reset free bytes */ *pFreeBytes = 0; /* check handle */ if (!self) return AAC_DEC_INVALID_HANDLE; /* return nr of free bytes */ HANDLE_FDK_BITSTREAM hBs = transportDec_GetBitstream(self->hInput, 0); *pFreeBytes = FDKgetFreeBits(hBs) >> 3; /* success */ return AAC_DEC_OK; } /** * Config Decoder using a CSAudioSpecificConfig struct. */ static LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_Config( HANDLE_AACDECODER self, const CSAudioSpecificConfig *pAscStruct, UCHAR configMode, UCHAR *configChanged) { AAC_DECODER_ERROR err; /* Initialize AAC core decoder, and update self->streaminfo */ err = CAacDecoder_Init(self, pAscStruct, configMode, configChanged); if (!FDK_chMapDescr_isValid(&self->mapDescr)) { return AAC_DEC_UNSUPPORTED_CHANNELCONFIG; } return err; } LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_ConfigRaw(HANDLE_AACDECODER self, UCHAR *conf[], const UINT length[]) { AAC_DECODER_ERROR err = AAC_DEC_OK; TRANSPORTDEC_ERROR errTp; UINT layer, nrOfLayers = self->nrOfLayers; for (layer = 0; layer < nrOfLayers; layer++) { if (length[layer] > 0) { errTp = transportDec_OutOfBandConfig(self->hInput, conf[layer], length[layer], layer); if (errTp != TRANSPORTDEC_OK) { switch (errTp) { case TRANSPORTDEC_NEED_TO_RESTART: err = AAC_DEC_NEED_TO_RESTART; break; case TRANSPORTDEC_UNSUPPORTED_FORMAT: err = AAC_DEC_UNSUPPORTED_FORMAT; break; default: err = AAC_DEC_UNKNOWN; break; } /* if baselayer is OK we continue decoding */ if (layer >= 1) { self->nrOfLayers = layer; err = AAC_DEC_OK; } break; } } } return err; } LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_RawISOBMFFData(HANDLE_AACDECODER self, UCHAR *buffer, UINT length) { FDK_BITSTREAM bs; HANDLE_FDK_BITSTREAM hBs = &bs; AAC_DECODER_ERROR err = AAC_DEC_OK; if (length < 8) return AAC_DEC_UNKNOWN; while (length >= 8) { UINT size = (buffer[0] << 24) | (buffer[1] << 16) | (buffer[2] << 8) | buffer[3]; DRC_DEC_ERROR uniDrcErr = DRC_DEC_OK; if (length < size) return AAC_DEC_UNKNOWN; if (size <= 8) return AAC_DEC_UNKNOWN; FDKinitBitStream(hBs, buffer + 8, 0x10000000, (size - 8) * 8); if ((buffer[4] == 'l') && (buffer[5] == 'u') && (buffer[6] == 'd') && (buffer[7] == 't')) { uniDrcErr = FDK_drcDec_ReadLoudnessBox(self->hUniDrcDecoder, hBs); } else if ((buffer[4] == 'd') && (buffer[5] == 'm') && (buffer[6] == 'i') && (buffer[7] == 'x')) { uniDrcErr = FDK_drcDec_ReadDownmixInstructions_Box(self->hUniDrcDecoder, hBs); } else if ((buffer[4] == 'u') && (buffer[5] == 'd') && (buffer[6] == 'i') && (buffer[7] == '2')) { uniDrcErr = FDK_drcDec_ReadUniDrcInstructions_Box(self->hUniDrcDecoder, hBs); } else if ((buffer[4] == 'u') && (buffer[5] == 'd') && (buffer[6] == 'c') && (buffer[7] == '2')) { uniDrcErr = FDK_drcDec_ReadUniDrcCoefficients_Box(self->hUniDrcDecoder, hBs); } if (uniDrcErr != DRC_DEC_OK) err = AAC_DEC_UNKNOWN; buffer += size; length -= size; } return err; } static INT aacDecoder_ConfigCallback(void *handle, const CSAudioSpecificConfig *pAscStruct, UCHAR configMode, UCHAR *configChanged) { HANDLE_AACDECODER self = (HANDLE_AACDECODER)handle; AAC_DECODER_ERROR err = AAC_DEC_OK; TRANSPORTDEC_ERROR errTp; FDK_ASSERT(self != NULL); { { err = aacDecoder_Config(self, pAscStruct, configMode, configChanged); } } if (err == AAC_DEC_OK) { /* revert concealment method if either - Interpolation concealment might not be meaningful - Interpolation concealment is not implemented */ if ((self->flags[0] & (AC_LD | AC_ELD) && (self->concealMethodUser == ConcealMethodNone) && CConcealment_GetDelay(&self->concealCommonData) > 0) /* might not be meaningful but allow if user has set it expicitly */ || (self->flags[0] & (AC_USAC | AC_RSVD50 | AC_RSV603DA) && CConcealment_GetDelay(&self->concealCommonData) > 0) /* not implemented */ ) { /* Revert to error concealment method Noise Substitution. Because interpolation is not implemented for USAC or the additional delay is unwanted for low delay codecs. */ setConcealMethod(self, 1); } aacDecoder_setMetadataExpiry(self, self->metadataExpiry); errTp = TRANSPORTDEC_OK; } else { if (err == AAC_DEC_NEED_TO_RESTART) { errTp = TRANSPORTDEC_NEED_TO_RESTART; } else if (IS_INIT_ERROR(err)) { errTp = TRANSPORTDEC_UNSUPPORTED_FORMAT; } /* Fatal errors */ else { errTp = TRANSPORTDEC_UNKOWN_ERROR; } } return errTp; } static INT aacDecoder_FreeMemCallback(void *handle, const CSAudioSpecificConfig *pAscStruct) { TRANSPORTDEC_ERROR errTp = TRANSPORTDEC_OK; HANDLE_AACDECODER self = (HANDLE_AACDECODER)handle; const int subStreamIndex = 0; FDK_ASSERT(self != NULL); if (CAacDecoder_FreeMem(self, subStreamIndex) != AAC_DEC_OK) { errTp = TRANSPORTDEC_UNKOWN_ERROR; } /* free Ram_SbrDecoder and Ram_SbrDecChannel */ if (self->hSbrDecoder != NULL) { if (sbrDecoder_FreeMem(&self->hSbrDecoder) != SBRDEC_OK) { errTp = TRANSPORTDEC_UNKOWN_ERROR; } } /* free pSpatialDec and mpsData */ if (self->pMpegSurroundDecoder != NULL) { if (mpegSurroundDecoder_FreeMem( (CMpegSurroundDecoder *)self->pMpegSurroundDecoder) != MPS_OK) { errTp = TRANSPORTDEC_UNKOWN_ERROR; } } /* free persistent qmf domain buffer, QmfWorkBufferCore3, QmfWorkBufferCore4, * QmfWorkBufferCore5 and configuration variables */ FDK_QmfDomain_FreeMem(&self->qmfDomain); return errTp; } static INT aacDecoder_CtrlCFGChangeCallback( void *handle, const CCtrlCFGChange *pCtrlCFGChangeStruct) { TRANSPORTDEC_ERROR errTp = TRANSPORTDEC_OK; HANDLE_AACDECODER self = (HANDLE_AACDECODER)handle; if (self != NULL) { CAacDecoder_CtrlCFGChange( self, pCtrlCFGChangeStruct->flushStatus, pCtrlCFGChangeStruct->flushCnt, pCtrlCFGChangeStruct->buildUpStatus, pCtrlCFGChangeStruct->buildUpCnt); } else { errTp = TRANSPORTDEC_UNKOWN_ERROR; } return errTp; } static INT aacDecoder_SbrCallback( void *handle, HANDLE_FDK_BITSTREAM hBs, const INT sampleRateIn, const INT sampleRateOut, const INT samplesPerFrame, const AUDIO_OBJECT_TYPE coreCodec, const MP4_ELEMENT_ID elementID, const INT elementIndex, const UCHAR harmonicSBR, const UCHAR stereoConfigIndex, const UCHAR configMode, UCHAR *configChanged, const INT downscaleFactor) { HANDLE_SBRDECODER self = (HANDLE_SBRDECODER)handle; INT errTp = sbrDecoder_Header(self, hBs, sampleRateIn, sampleRateOut, samplesPerFrame, coreCodec, elementID, elementIndex, harmonicSBR, stereoConfigIndex, configMode, configChanged, downscaleFactor); return errTp; } static INT aacDecoder_SscCallback(void *handle, HANDLE_FDK_BITSTREAM hBs, const AUDIO_OBJECT_TYPE coreCodec, const INT samplingRate, const INT frameSize, const INT stereoConfigIndex, const INT coreSbrFrameLengthIndex, const INT configBytes, const UCHAR configMode, UCHAR *configChanged) { SACDEC_ERROR err; TRANSPORTDEC_ERROR errTp; HANDLE_AACDECODER hAacDecoder = (HANDLE_AACDECODER)handle; err = mpegSurroundDecoder_Config( (CMpegSurroundDecoder *)hAacDecoder->pMpegSurroundDecoder, hBs, coreCodec, samplingRate, frameSize, stereoConfigIndex, coreSbrFrameLengthIndex, configBytes, configMode, configChanged); switch (err) { case MPS_UNSUPPORTED_CONFIG: /* MPS found but invalid or not decodable by this instance */ /* We switch off MPS and keep going */ hAacDecoder->mpsEnableCurr = 0; hAacDecoder->mpsApplicable = 0; errTp = TRANSPORTDEC_OK; break; case MPS_PARSE_ERROR: /* MPS found but invalid or not decodable by this instance */ hAacDecoder->mpsEnableCurr = 0; hAacDecoder->mpsApplicable = 0; if ((coreCodec == AOT_USAC) || (coreCodec == AOT_DRM_USAC) || IS_LOWDELAY(coreCodec)) { errTp = TRANSPORTDEC_PARSE_ERROR; } else { errTp = TRANSPORTDEC_OK; } break; case MPS_OK: hAacDecoder->mpsApplicable = 1; errTp = TRANSPORTDEC_OK; break; default: /* especially Parsing error is critical for transport layer */ hAacDecoder->mpsApplicable = 0; errTp = TRANSPORTDEC_UNKOWN_ERROR; } return (INT)errTp; } static INT aacDecoder_UniDrcCallback(void *handle, HANDLE_FDK_BITSTREAM hBs, const INT fullPayloadLength, const INT payloadType, const INT subStreamIndex, const INT payloadStart, const AUDIO_OBJECT_TYPE aot) { DRC_DEC_ERROR err = DRC_DEC_OK; TRANSPORTDEC_ERROR errTp; HANDLE_AACDECODER hAacDecoder = (HANDLE_AACDECODER)handle; DRC_DEC_CODEC_MODE drcDecCodecMode = DRC_DEC_CODEC_MODE_UNDEFINED; if (subStreamIndex != 0) { return TRANSPORTDEC_OK; } else if (aot == AOT_USAC) { drcDecCodecMode = DRC_DEC_MPEG_D_USAC; } err = FDK_drcDec_SetCodecMode(hAacDecoder->hUniDrcDecoder, drcDecCodecMode); if (err) return (INT)TRANSPORTDEC_UNKOWN_ERROR; if (payloadType == 0) /* uniDrcConfig */ { err = FDK_drcDec_ReadUniDrcConfig(hAacDecoder->hUniDrcDecoder, hBs); } else /* loudnessInfoSet */ { err = FDK_drcDec_ReadLoudnessInfoSet(hAacDecoder->hUniDrcDecoder, hBs); hAacDecoder->loudnessInfoSetPosition[1] = payloadStart; hAacDecoder->loudnessInfoSetPosition[2] = fullPayloadLength; } if (err == DRC_DEC_OK) errTp = TRANSPORTDEC_OK; else errTp = TRANSPORTDEC_UNKOWN_ERROR; return (INT)errTp; } LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_AncDataInit(HANDLE_AACDECODER self, UCHAR *buffer, int size) { CAncData *ancData = &self->ancData; return CAacDecoder_AncDataInit(ancData, buffer, size); } LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_AncDataGet(HANDLE_AACDECODER self, int index, UCHAR **ptr, int *size) { CAncData *ancData = &self->ancData; return CAacDecoder_AncDataGet(ancData, index, ptr, size); } /* If MPS is present in stream, but not supported by this instance, we'll have to switch off MPS and use QMF synthesis in the SBR module if required */ static int isSupportedMpsConfig(AUDIO_OBJECT_TYPE aot, unsigned int numInChannels, unsigned int fMpsPresent) { LIB_INFO libInfo[FDK_MODULE_LAST]; UINT mpsCaps; int isSupportedCfg = 1; FDKinitLibInfo(libInfo); mpegSurroundDecoder_GetLibInfo(libInfo); mpsCaps = FDKlibInfo_getCapabilities(libInfo, FDK_MPSDEC); if (!(mpsCaps & CAPF_MPS_LD) && IS_LOWDELAY(aot)) { /* We got an LD AOT but MPS decoder does not support LD. */ isSupportedCfg = 0; } if ((mpsCaps & CAPF_MPS_LD) && IS_LOWDELAY(aot) && !fMpsPresent) { /* We got an LD AOT and the MPS decoder supports it. * But LD-MPS is not explicitly signaled. */ isSupportedCfg = 0; } if (!(mpsCaps & CAPF_MPS_USAC) && IS_USAC(aot)) { /* We got an USAC AOT but MPS decoder does not support USAC. */ isSupportedCfg = 0; } if (!(mpsCaps & CAPF_MPS_STD) && !IS_LOWDELAY(aot) && !IS_USAC(aot)) { /* We got an GA AOT but MPS decoder does not support it. */ isSupportedCfg = 0; } /* Check whether the MPS modul supports the given number of input channels: */ switch (numInChannels) { case 1: if (!(mpsCaps & CAPF_MPS_1CH_IN)) { /* We got a one channel input to MPS decoder but it does not support it. */ isSupportedCfg = 0; } break; case 2: if (!(mpsCaps & CAPF_MPS_2CH_IN)) { /* We got a two channel input to MPS decoder but it does not support it. */ isSupportedCfg = 0; } break; case 5: case 6: if (!(mpsCaps & CAPF_MPS_6CH_IN)) { /* We got a six channel input to MPS decoder but it does not support it. */ isSupportedCfg = 0; } break; default: isSupportedCfg = 0; } return (isSupportedCfg); } static AAC_DECODER_ERROR setConcealMethod( const HANDLE_AACDECODER self, /*!< Handle of the decoder instance */ const INT method) { AAC_DECODER_ERROR errorStatus = AAC_DEC_OK; CConcealParams *pConcealData = NULL; int method_revert = 0; HANDLE_SBRDECODER hSbrDec = NULL; HANDLE_AAC_DRC hDrcInfo = NULL; HANDLE_PCM_DOWNMIX hPcmDmx = NULL; CConcealmentMethod backupMethod = ConcealMethodNone; int backupDelay = 0; int bsDelay = 0; /* check decoder handle */ if (self != NULL) { pConcealData = &self->concealCommonData; hSbrDec = self->hSbrDecoder; hDrcInfo = self->hDrcInfo; hPcmDmx = self->hPcmUtils; if (self->flags[0] & (AC_USAC | AC_RSVD50 | AC_RSV603DA) && method >= 2) { /* Interpolation concealment is not implemented for USAC/RSVD50 */ /* errorStatus = AAC_DEC_SET_PARAM_FAIL; goto bail; */ method_revert = 1; } if (self->flags[0] & (AC_USAC | AC_RSVD50 | AC_RSV603DA) && method >= 2) { /* Interpolation concealment is not implemented for USAC/RSVD50 */ errorStatus = AAC_DEC_SET_PARAM_FAIL; goto bail; } } /* Get current method/delay */ backupMethod = CConcealment_GetMethod(pConcealData); backupDelay = CConcealment_GetDelay(pConcealData); /* Be sure to set AAC and SBR concealment method simultaneously! */ errorStatus = CConcealment_SetParams( pConcealData, (method_revert == 0) ? (int)method : (int)1, // concealMethod AACDEC_CONCEAL_PARAM_NOT_SPECIFIED, // concealFadeOutSlope AACDEC_CONCEAL_PARAM_NOT_SPECIFIED, // concealFadeInSlope AACDEC_CONCEAL_PARAM_NOT_SPECIFIED, // concealMuteRelease AACDEC_CONCEAL_PARAM_NOT_SPECIFIED // concealComfNoiseLevel ); if ((errorStatus != AAC_DEC_OK) && (errorStatus != AAC_DEC_INVALID_HANDLE)) { goto bail; } /* Get new delay */ bsDelay = CConcealment_GetDelay(pConcealData); { SBR_ERROR sbrErr = SBRDEC_OK; /* set SBR bitstream delay */ sbrErr = sbrDecoder_SetParam(hSbrDec, SBR_SYSTEM_BITSTREAM_DELAY, bsDelay); switch (sbrErr) { case SBRDEC_OK: case SBRDEC_NOT_INITIALIZED: if (self != NULL) { /* save the param value and set later (when SBR has been initialized) */ self->sbrParams.bsDelay = bsDelay; } break; default: errorStatus = AAC_DEC_SET_PARAM_FAIL; goto bail; } } errorStatus = aacDecoder_drcSetParam(hDrcInfo, DRC_BS_DELAY, bsDelay); if ((errorStatus != AAC_DEC_OK) && (errorStatus != AAC_DEC_INVALID_HANDLE)) { goto bail; } if (errorStatus == AAC_DEC_OK) { PCMDMX_ERROR err = pcmDmx_SetParam(hPcmDmx, DMX_BS_DATA_DELAY, bsDelay); switch (err) { case PCMDMX_INVALID_HANDLE: errorStatus = AAC_DEC_INVALID_HANDLE; break; case PCMDMX_OK: break; default: errorStatus = AAC_DEC_SET_PARAM_FAIL; goto bail; } } bail: if ((errorStatus != AAC_DEC_OK) && (errorStatus != AAC_DEC_INVALID_HANDLE)) { /* Revert to the initial state */ CConcealment_SetParams( pConcealData, (int)backupMethod, AACDEC_CONCEAL_PARAM_NOT_SPECIFIED, AACDEC_CONCEAL_PARAM_NOT_SPECIFIED, AACDEC_CONCEAL_PARAM_NOT_SPECIFIED, AACDEC_CONCEAL_PARAM_NOT_SPECIFIED); /* Revert SBR bitstream delay */ sbrDecoder_SetParam(hSbrDec, SBR_SYSTEM_BITSTREAM_DELAY, backupDelay); /* Revert DRC bitstream delay */ aacDecoder_drcSetParam(hDrcInfo, DRC_BS_DELAY, backupDelay); /* Revert PCM mixdown bitstream delay */ pcmDmx_SetParam(hPcmDmx, DMX_BS_DATA_DELAY, backupDelay); } return errorStatus; } LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_SetParam( const HANDLE_AACDECODER self, /*!< Handle of the decoder instance */ const AACDEC_PARAM param, /*!< Parameter to set */ const INT value) /*!< Parameter valued */ { AAC_DECODER_ERROR errorStatus = AAC_DEC_OK; HANDLE_TRANSPORTDEC hTpDec = NULL; TRANSPORTDEC_ERROR errTp = TRANSPORTDEC_OK; HANDLE_AAC_DRC hDrcInfo = NULL; HANDLE_PCM_DOWNMIX hPcmDmx = NULL; PCMDMX_ERROR dmxErr = PCMDMX_OK; TDLimiterPtr hPcmTdl = NULL; DRC_DEC_ERROR uniDrcErr = DRC_DEC_OK; /* check decoder handle */ if (self != NULL) { hTpDec = self->hInput; hDrcInfo = self->hDrcInfo; hPcmDmx = self->hPcmUtils; hPcmTdl = self->hLimiter; } else { errorStatus = AAC_DEC_INVALID_HANDLE; goto bail; } /* configure the subsystems */ switch (param) { case AAC_PCM_MIN_OUTPUT_CHANNELS: if (value < -1 || value > (8)) { return AAC_DEC_SET_PARAM_FAIL; } dmxErr = pcmDmx_SetParam(hPcmDmx, MIN_NUMBER_OF_OUTPUT_CHANNELS, value); break; case AAC_PCM_MAX_OUTPUT_CHANNELS: if (value < -1 || value > (8)) { return AAC_DEC_SET_PARAM_FAIL; } dmxErr = pcmDmx_SetParam(hPcmDmx, MAX_NUMBER_OF_OUTPUT_CHANNELS, value); if (dmxErr != PCMDMX_OK) { goto bail; } errorStatus = aacDecoder_drcSetParam(hDrcInfo, MAX_OUTPUT_CHANNELS, value); if (value > 0) { uniDrcErr = FDK_drcDec_SetParam(self->hUniDrcDecoder, DRC_DEC_TARGET_CHANNEL_COUNT_REQUESTED, (FIXP_DBL)value); } break; case AAC_PCM_DUAL_CHANNEL_OUTPUT_MODE: dmxErr = pcmDmx_SetParam(hPcmDmx, DMX_DUAL_CHANNEL_MODE, value); break; case AAC_PCM_LIMITER_ENABLE: if (value < -2 || value > 1) { return AAC_DEC_SET_PARAM_FAIL; } self->limiterEnableUser = value; break; case AAC_PCM_LIMITER_ATTACK_TIME: if (value <= 0) { /* module function converts value to unsigned */ return AAC_DEC_SET_PARAM_FAIL; } switch (pcmLimiter_SetAttack(hPcmTdl, value)) { case TDLIMIT_OK: break; case TDLIMIT_INVALID_HANDLE: return AAC_DEC_INVALID_HANDLE; case TDLIMIT_INVALID_PARAMETER: default: return AAC_DEC_SET_PARAM_FAIL; } break; case AAC_PCM_LIMITER_RELEAS_TIME: if (value <= 0) { /* module function converts value to unsigned */ return AAC_DEC_SET_PARAM_FAIL; } switch (pcmLimiter_SetRelease(hPcmTdl, value)) { case TDLIMIT_OK: break; case TDLIMIT_INVALID_HANDLE: return AAC_DEC_INVALID_HANDLE; case TDLIMIT_INVALID_PARAMETER: default: return AAC_DEC_SET_PARAM_FAIL; } break; case AAC_METADATA_PROFILE: { DMX_PROFILE_TYPE dmxProfile; INT mdExpiry = -1; /* in ms (-1: don't change) */ switch ((AAC_MD_PROFILE)value) { case AAC_MD_PROFILE_MPEG_STANDARD: dmxProfile = DMX_PRFL_STANDARD; break; case AAC_MD_PROFILE_MPEG_LEGACY: dmxProfile = DMX_PRFL_MATRIX_MIX; break; case AAC_MD_PROFILE_MPEG_LEGACY_PRIO: dmxProfile = DMX_PRFL_FORCE_MATRIX_MIX; break; case AAC_MD_PROFILE_ARIB_JAPAN: dmxProfile = DMX_PRFL_ARIB_JAPAN; mdExpiry = 550; /* ms */ break; default: return AAC_DEC_SET_PARAM_FAIL; } dmxErr = pcmDmx_SetParam(hPcmDmx, DMX_PROFILE_SETTING, (INT)dmxProfile); if (dmxErr != PCMDMX_OK) { goto bail; } if ((self != NULL) && (mdExpiry >= 0)) { self->metadataExpiry = mdExpiry; /* Determine the corresponding number of frames and configure all * related modules. */ aacDecoder_setMetadataExpiry(self, mdExpiry); } } break; case AAC_METADATA_EXPIRY_TIME: if (value < 0) { return AAC_DEC_SET_PARAM_FAIL; } if (self != NULL) { self->metadataExpiry = value; /* Determine the corresponding number of frames and configure all * related modules. */ aacDecoder_setMetadataExpiry(self, value); } break; case AAC_PCM_OUTPUT_CHANNEL_MAPPING: if (value < 0 || value > 1) { return AAC_DEC_SET_PARAM_FAIL; } /* CAUTION: The given value must be inverted to match the logic! */ FDK_chMapDescr_setPassThrough(&self->mapDescr, !value); break; case AAC_QMF_LOWPOWER: if (value < -1 || value > 1) { return AAC_DEC_SET_PARAM_FAIL; } /** * Set QMF mode (might be overriden) * 0:HQ (complex) * 1:LP (partially complex) */ self->qmfModeUser = (QMF_MODE)value; break; case AAC_DRC_ATTENUATION_FACTOR: /* DRC compression factor (where 0 is no and 127 is max compression) */ errorStatus = aacDecoder_drcSetParam(hDrcInfo, DRC_CUT_SCALE, value); uniDrcErr = FDK_drcDec_SetParam(self->hUniDrcDecoder, DRC_DEC_COMPRESS, value * (FL2FXCONST_DBL(0.5f / 127.0f))); break; case AAC_DRC_BOOST_FACTOR: /* DRC boost factor (where 0 is no and 127 is max boost) */ errorStatus = aacDecoder_drcSetParam(hDrcInfo, DRC_BOOST_SCALE, value); uniDrcErr = FDK_drcDec_SetParam(self->hUniDrcDecoder, DRC_DEC_BOOST, value * (FL2FXCONST_DBL(0.5f / 127.0f))); break; case AAC_DRC_REFERENCE_LEVEL: if ((value >= 0) && ((value < 40) || (value > 127))) /* allowed range: -10 to -31.75 dB */ return AAC_DEC_SET_PARAM_FAIL; /* DRC target reference level quantized in 0.25dB steps using values [40..127]. Negative values switch off loudness normalisation. Negative values also switch off MPEG-4 DRC, while MPEG-D DRC can be separately switched on/off with AAC_UNIDRC_SET_EFFECT */ errorStatus = aacDecoder_drcSetParam(hDrcInfo, TARGET_REF_LEVEL, value); uniDrcErr = FDK_drcDec_SetParam(self->hUniDrcDecoder, DRC_DEC_LOUDNESS_NORMALIZATION_ON, (FIXP_DBL)(value >= 0)); /* set target loudness also for MPEG-D DRC */ self->defaultTargetLoudness = (SCHAR)value; break; case AAC_DRC_HEAVY_COMPRESSION: /* Don't need to overwrite cut/boost values */ errorStatus = aacDecoder_drcSetParam(hDrcInfo, APPLY_HEAVY_COMPRESSION, value); break; case AAC_DRC_DEFAULT_PRESENTATION_MODE: /* DRC default presentation mode */ errorStatus = aacDecoder_drcSetParam(hDrcInfo, DEFAULT_PRESENTATION_MODE, value); break; case AAC_DRC_ENC_TARGET_LEVEL: /* Encoder target level for light (i.e. not heavy) compression: Target reference level assumed at encoder for deriving limiting gains */ errorStatus = aacDecoder_drcSetParam(hDrcInfo, ENCODER_TARGET_LEVEL, value); break; case AAC_UNIDRC_SET_EFFECT: if ((value < -1) || (value > 6)) return AAC_DEC_SET_PARAM_FAIL; uniDrcErr = FDK_drcDec_SetParam(self->hUniDrcDecoder, DRC_DEC_EFFECT_TYPE, (FIXP_DBL)value); break; case AAC_UNIDRC_ALBUM_MODE: uniDrcErr = FDK_drcDec_SetParam(self->hUniDrcDecoder, DRC_DEC_ALBUM_MODE, (FIXP_DBL)value); break; case AAC_TPDEC_CLEAR_BUFFER: errTp = transportDec_SetParam(hTpDec, TPDEC_PARAM_RESET, 1); self->streamInfo.numLostAccessUnits = 0; self->streamInfo.numBadBytes = 0; self->streamInfo.numTotalBytes = 0; /* aacDecoder_SignalInterruption(self); */ break; case AAC_CONCEAL_METHOD: /* Changing the concealment method can introduce additional bitstream delay. And that in turn affects sub libraries and modules which makes the whole thing quite complex. So the complete changing routine is packed into a helper function which keeps all modules and libs in a consistent state even in the case an error occures. */ errorStatus = setConcealMethod(self, value); if (errorStatus == AAC_DEC_OK) { self->concealMethodUser = (CConcealmentMethod)value; } break; default: return AAC_DEC_SET_PARAM_FAIL; } /* switch(param) */ bail: if (errorStatus == AAC_DEC_OK) { /* Check error code returned by DMX module library: */ switch (dmxErr) { case PCMDMX_OK: break; case PCMDMX_INVALID_HANDLE: errorStatus = AAC_DEC_INVALID_HANDLE; break; default: errorStatus = AAC_DEC_SET_PARAM_FAIL; } } if (errTp != TRANSPORTDEC_OK && errorStatus == AAC_DEC_OK) { errorStatus = AAC_DEC_SET_PARAM_FAIL; } if (errorStatus == AAC_DEC_OK) { /* Check error code returned by MPEG-D DRC decoder library: */ switch (uniDrcErr) { case 0: break; case -9998: errorStatus = AAC_DEC_INVALID_HANDLE; break; default: errorStatus = AAC_DEC_SET_PARAM_FAIL; break; } } return (errorStatus); } LINKSPEC_CPP HANDLE_AACDECODER aacDecoder_Open(TRANSPORT_TYPE transportFmt, UINT nrOfLayers) { AAC_DECODER_INSTANCE *aacDec = NULL; HANDLE_TRANSPORTDEC pIn; int err = 0; int stereoConfigIndex = -1; UINT nrOfLayers_min = fMin(nrOfLayers, (UINT)1); /* Allocate transport layer struct. */ pIn = transportDec_Open(transportFmt, TP_FLAG_MPEG4, nrOfLayers_min); if (pIn == NULL) { return NULL; } transportDec_SetParam(pIn, TPDEC_PARAM_IGNORE_BUFFERFULLNESS, 1); /* Allocate AAC decoder core struct. */ aacDec = CAacDecoder_Open(transportFmt); if (aacDec == NULL) { transportDec_Close(&pIn); goto bail; } aacDec->hInput = pIn; aacDec->nrOfLayers = nrOfLayers_min; /* Setup channel mapping descriptor. */ FDK_chMapDescr_init(&aacDec->mapDescr, NULL, 0, 0); /* Register Config Update callback. */ transportDec_RegisterAscCallback(pIn, aacDecoder_ConfigCallback, (void *)aacDec); /* Register Free Memory callback. */ transportDec_RegisterFreeMemCallback(pIn, aacDecoder_FreeMemCallback, (void *)aacDec); /* Register config switch control callback. */ transportDec_RegisterCtrlCFGChangeCallback( pIn, aacDecoder_CtrlCFGChangeCallback, (void *)aacDec); FDKmemclear(&aacDec->qmfDomain, sizeof(FDK_QMF_DOMAIN)); /* open SBR decoder */ if (SBRDEC_OK != sbrDecoder_Open(&aacDec->hSbrDecoder, &aacDec->qmfDomain)) { err = -1; goto bail; } aacDec->qmfModeUser = NOT_DEFINED; transportDec_RegisterSbrCallback(aacDec->hInput, aacDecoder_SbrCallback, (void *)aacDec->hSbrDecoder); if (mpegSurroundDecoder_Open( (CMpegSurroundDecoder **)&aacDec->pMpegSurroundDecoder, stereoConfigIndex, &aacDec->qmfDomain)) { err = -1; goto bail; } /* Set MPEG Surround defaults */ aacDec->mpsEnableUser = 0; aacDec->mpsEnableCurr = 0; aacDec->mpsApplicable = 0; aacDec->mpsOutputMode = (SCHAR)SACDEC_OUT_MODE_NORMAL; transportDec_RegisterSscCallback(pIn, aacDecoder_SscCallback, (void *)aacDec); { if (FDK_drcDec_Open(&(aacDec->hUniDrcDecoder), DRC_DEC_ALL) != 0) { err = -1; goto bail; } } transportDec_RegisterUniDrcConfigCallback(pIn, aacDecoder_UniDrcCallback, (void *)aacDec, aacDec->loudnessInfoSetPosition); aacDec->defaultTargetLoudness = (SCHAR)96; pcmDmx_Open(&aacDec->hPcmUtils); if (aacDec->hPcmUtils == NULL) { err = -1; goto bail; } aacDec->hLimiter = pcmLimiter_Create(TDL_ATTACK_DEFAULT_MS, TDL_RELEASE_DEFAULT_MS, (FIXP_DBL)MAXVAL_DBL, (8), 96000); if (NULL == aacDec->hLimiter) { err = -1; goto bail; } aacDec->limiterEnableUser = (UCHAR)-1; aacDec->limiterEnableCurr = 0; /* Assure that all modules have same delay */ if (setConcealMethod(aacDec, CConcealment_GetMethod(&aacDec->concealCommonData))) { err = -1; goto bail; } bail: if (err == -1) { aacDecoder_Close(aacDec); aacDec = NULL; } return aacDec; } LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_Fill(HANDLE_AACDECODER self, UCHAR *pBuffer[], const UINT bufferSize[], UINT *pBytesValid) { TRANSPORTDEC_ERROR tpErr; /* loop counter for layers; if not TT_MP4_RAWPACKETS used as index for only available layer */ INT layer = 0; INT nrOfLayers = self->nrOfLayers; { for (layer = 0; layer < nrOfLayers; layer++) { { tpErr = transportDec_FillData(self->hInput, pBuffer[layer], bufferSize[layer], &pBytesValid[layer], layer); if (tpErr != TRANSPORTDEC_OK) { return AAC_DEC_UNKNOWN; /* Must be an internal error */ } } } } return AAC_DEC_OK; } static void aacDecoder_SignalInterruption(HANDLE_AACDECODER self) { CAacDecoder_SignalInterruption(self); if (self->hSbrDecoder != NULL) { sbrDecoder_SetParam(self->hSbrDecoder, SBR_BS_INTERRUPTION, 1); } if (self->mpsEnableUser) { mpegSurroundDecoder_SetParam( (CMpegSurroundDecoder *)self->pMpegSurroundDecoder, SACDEC_BS_INTERRUPTION, 1); } } static void aacDecoder_UpdateBitStreamCounters(CStreamInfo *pSi, HANDLE_FDK_BITSTREAM hBs, INT nBits, AAC_DECODER_ERROR ErrorStatus) { /* calculate bit difference (amount of bits moved forward) */ nBits = nBits - (INT)FDKgetValidBits(hBs); /* Note: The amount of bits consumed might become negative when parsing a bit stream with several sub frames, and we find out at the last sub frame that the total frame length does not match the sum of sub frame length. If this happens, the transport decoder might want to rewind to the supposed ending of the transport frame, and this position might be before the last access unit beginning. */ /* Calc bitrate. */ if (pSi->frameSize > 0) { /* bitRate = nBits * sampleRate / frameSize */ int ratio_e = 0; FIXP_DBL ratio_m = fDivNorm(pSi->sampleRate, pSi->frameSize, &ratio_e); pSi->bitRate = (INT)fMultNorm(nBits, DFRACT_BITS - 1, ratio_m, ratio_e, DFRACT_BITS - 1); } /* bit/byte counters */ { INT nBytes; nBytes = nBits >> 3; pSi->numTotalBytes += nBytes; if (IS_OUTPUT_VALID(ErrorStatus)) { pSi->numTotalAccessUnits++; } if (IS_DECODE_ERROR(ErrorStatus)) { pSi->numBadBytes += nBytes; pSi->numBadAccessUnits++; } } } static INT aacDecoder_EstimateNumberOfLostFrames(HANDLE_AACDECODER self) { INT n; transportDec_GetMissingAccessUnitCount(&n, self->hInput); return n; } LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData, const INT timeDataSize, const UINT flags) { AAC_DECODER_ERROR ErrorStatus; INT layer; INT nBits; INT timeData2Size; INT timeData3Size; INT timeDataHeadroom; HANDLE_FDK_BITSTREAM hBs; int fTpInterruption = 0; /* Transport originated interruption detection. */ int fTpConceal = 0; /* Transport originated concealment. */ UINT accessUnit = 0; UINT numAccessUnits = 1; UINT numPrerollAU = 0; int fEndAuNotAdjusted = 0; /* The end of the access unit was not adjusted */ int applyCrossfade = 1; /* flag indicates if flushing was possible */ PCM_DEC *pTimeData2; PCM_AAC *pTimeData3; if (self == NULL) { return AAC_DEC_INVALID_HANDLE; } if (flags & AACDEC_INTR) { self->streamInfo.numLostAccessUnits = 0; } hBs = transportDec_GetBitstream(self->hInput, 0); /* Get current bits position for bitrate calculation. */ nBits = FDKgetValidBits(hBs); if (flags & AACDEC_CLRHIST) { if (self->flags[0] & AC_USAC) { /* 1) store AudioSpecificConfig always in AudioSpecificConfig_Parse() */ /* 2) free memory of dynamic allocated data */ CSAudioSpecificConfig asc; transportDec_GetAsc(self->hInput, 0, &asc); aacDecoder_FreeMemCallback(self, &asc); self->streamInfo.numChannels = 0; /* 3) restore AudioSpecificConfig */ transportDec_OutOfBandConfig(self->hInput, asc.config, (asc.configBits + 7) >> 3, 0); } } if (!((flags & (AACDEC_CONCEAL | AACDEC_FLUSH)) || (self->flushStatus == AACDEC_RSV60_DASH_IPF_ATSC_FLUSH_ON) || (self->flushStatus == AACDEC_USAC_DASH_IPF_FLUSH_ON) || (self->buildUpStatus == AACDEC_RSV60_BUILD_UP_IDLE_IN_BAND))) { TRANSPORTDEC_ERROR err; for (layer = 0; layer < self->nrOfLayers; layer++) { err = transportDec_ReadAccessUnit(self->hInput, layer); if (err != TRANSPORTDEC_OK) { switch (err) { case TRANSPORTDEC_NOT_ENOUGH_BITS: ErrorStatus = AAC_DEC_NOT_ENOUGH_BITS; goto bail; case TRANSPORTDEC_SYNC_ERROR: self->streamInfo.numLostAccessUnits = aacDecoder_EstimateNumberOfLostFrames(self); fTpInterruption = 1; break; case TRANSPORTDEC_NEED_TO_RESTART: ErrorStatus = AAC_DEC_NEED_TO_RESTART; goto bail; case TRANSPORTDEC_CRC_ERROR: fTpConceal = 1; break; case TRANSPORTDEC_UNSUPPORTED_FORMAT: ErrorStatus = AAC_DEC_UNSUPPORTED_FORMAT; goto bail; default: ErrorStatus = AAC_DEC_UNKNOWN; goto bail; } } } } else { if (self->streamInfo.numLostAccessUnits > 0) { self->streamInfo.numLostAccessUnits--; } } self->frameOK = 1; UINT prerollAUOffset[AACDEC_MAX_NUM_PREROLL_AU]; UINT prerollAULength[AACDEC_MAX_NUM_PREROLL_AU]; for (int i = 0; i < AACDEC_MAX_NUM_PREROLL_AU + 1; i++) self->prerollAULength[i] = 0; INT auStartAnchor; HANDLE_FDK_BITSTREAM hBsAu; /* Process preroll frames and current frame */ do { if (!(flags & (AACDEC_CONCEAL | AACDEC_FLUSH)) && (self->flushStatus != AACDEC_RSV60_CFG_CHANGE_ATSC_FLUSH_ON) && (accessUnit == 0) && (self->hasAudioPreRoll || (self->buildUpStatus == AACDEC_RSV60_BUILD_UP_IDLE_IN_BAND)) && !fTpInterruption && !fTpConceal /* Bit stream pointer needs to be at the beginning of a (valid) AU. */ ) { ErrorStatus = CAacDecoder_PreRollExtensionPayloadParse( self, &numPrerollAU, prerollAUOffset, prerollAULength); if (ErrorStatus != AAC_DEC_OK) { switch (ErrorStatus) { case AAC_DEC_NOT_ENOUGH_BITS: goto bail; case AAC_DEC_PARSE_ERROR: self->frameOK = 0; break; default: break; } } numAccessUnits += numPrerollAU; } hBsAu = transportDec_GetBitstream(self->hInput, 0); auStartAnchor = (INT)FDKgetValidBits(hBsAu); self->accessUnit = accessUnit; if (accessUnit < numPrerollAU) { FDKpushFor(hBsAu, prerollAUOffset[accessUnit]); } /* Signal bit stream interruption to other modules if required. */ if (fTpInterruption || ((flags & AACDEC_INTR) && (accessUnit == 0))) { aacDecoder_SignalInterruption(self); if (!((flags & AACDEC_INTR) && (accessUnit == 0))) { ErrorStatus = AAC_DEC_TRANSPORT_SYNC_ERROR; goto bail; } } /* Clearing core data will be done in CAacDecoder_DecodeFrame() below. Tell other modules to clear states if required. */ if (flags & AACDEC_CLRHIST) { if (!(self->flags[0] & AC_USAC)) { sbrDecoder_SetParam(self->hSbrDecoder, SBR_CLEAR_HISTORY, 1); mpegSurroundDecoder_SetParam( (CMpegSurroundDecoder *)self->pMpegSurroundDecoder, SACDEC_CLEAR_HISTORY, 1); if (FDK_QmfDomain_ClearPersistentMemory(&self->qmfDomain) != 0) { ErrorStatus = AAC_DEC_UNKNOWN; goto bail; } } } /* Empty bit buffer in case of flush request. */ if (flags & AACDEC_FLUSH && !(flags & AACDEC_CONCEAL)) { if (!self->flushStatus) { transportDec_SetParam(self->hInput, TPDEC_PARAM_RESET, 1); self->streamInfo.numLostAccessUnits = 0; self->streamInfo.numBadBytes = 0; self->streamInfo.numTotalBytes = 0; } } /* Reset the output delay field. The modules will add their figures one * after another. */ self->streamInfo.outputDelay = 0; if (self->limiterEnableUser == (UCHAR)-2) { /* Enable limiter only for RSVD60. */ self->limiterEnableCurr = (self->flags[0] & AC_RSV603DA) ? 1 : 0; } else if (self->limiterEnableUser == (UCHAR)-1) { /* Enable limiter for all non-lowdelay AOT's. */ self->limiterEnableCurr = (self->flags[0] & (AC_LD | AC_ELD)) ? 0 : 1; } else { /* Use limiter configuration as requested. */ self->limiterEnableCurr = self->limiterEnableUser; } /* reset DRC level normalization gain on a per frame basis */ self->extGain[0] = AACDEC_DRC_GAIN_INIT_VALUE; pTimeData2 = self->pTimeData2; timeData2Size = self->timeData2Size / sizeof(PCM_DEC); pTimeData3 = (PCM_AAC *)self->pTimeData2; timeData3Size = self->timeData2Size / sizeof(PCM_AAC); ErrorStatus = CAacDecoder_DecodeFrame( self, flags | (fTpConceal ? AACDEC_CONCEAL : 0) | ((self->flushStatus && !(flags & AACDEC_CONCEAL)) ? AACDEC_FLUSH : 0), pTimeData2 + 0, timeData2Size, self->streamInfo.aacSamplesPerFrame + 0); timeDataHeadroom = self->aacOutDataHeadroom; /* if flushing for USAC DASH IPF was not possible go on with decoding * preroll */ if ((self->flags[0] & AC_USAC) && (self->flushStatus == AACDEC_USAC_DASH_IPF_FLUSH_ON) && !(flags & AACDEC_CONCEAL) && (ErrorStatus != AAC_DEC_OK)) { applyCrossfade = 0; } else /* USAC DASH IPF flushing possible begin */ { if (!((flags & (AACDEC_CONCEAL | AACDEC_FLUSH)) || fTpConceal || self->flushStatus) && (!(IS_OUTPUT_VALID(ErrorStatus)) || !(accessUnit < numPrerollAU))) { TRANSPORTDEC_ERROR tpErr; tpErr = transportDec_EndAccessUnit(self->hInput); if (tpErr != TRANSPORTDEC_OK) { self->frameOK = 0; } } else { /* while preroll processing later possibly an error in the renderer part occurrs */ if (IS_OUTPUT_VALID(ErrorStatus)) { fEndAuNotAdjusted = 1; } } /* If the current pTimeData2 does not contain a valid signal, there * nothing else we can do, so bail. */ if (!IS_OUTPUT_VALID(ErrorStatus)) { goto bail; } { self->streamInfo.sampleRate = self->streamInfo.aacSampleRate; self->streamInfo.frameSize = self->streamInfo.aacSamplesPerFrame; } self->streamInfo.numChannels = self->streamInfo.aacNumChannels; { FDK_Delay_Apply( &self->usacResidualDelay, pTimeData2 + 1 * (self->streamInfo.aacSamplesPerFrame + 0) + 0, self->streamInfo.frameSize, 0); } /* Setting of internal MPS state; may be reset in CAacDecoder_SyncQmfMode if decoder is unable to decode with user defined qmfMode */ if (!(self->flags[0] & (AC_USAC | AC_RSVD50 | AC_RSV603DA | AC_ELD))) { self->mpsEnableCurr = (self->mpsEnableUser && isSupportedMpsConfig(self->streamInfo.aot, self->streamInfo.numChannels, (self->flags[0] & AC_MPS_PRESENT) ? 1 : 0)); } if (!self->qmfDomain.globalConf.qmfDomainExplicitConfig && self->mpsEnableCurr) { /* if not done yet, allocate full MPEG Surround decoder instance */ if (mpegSurroundDecoder_IsFullMpegSurroundDecoderInstanceAvailable( (CMpegSurroundDecoder *)self->pMpegSurroundDecoder) == SAC_INSTANCE_NOT_FULL_AVAILABLE) { if (mpegSurroundDecoder_Open( (CMpegSurroundDecoder **)&self->pMpegSurroundDecoder, -1, &self->qmfDomain)) { return AAC_DEC_OUT_OF_MEMORY; } } } CAacDecoder_SyncQmfMode(self); if (!self->qmfDomain.globalConf.qmfDomainExplicitConfig && self->mpsEnableCurr) { SAC_INPUT_CONFIG sac_interface = (self->sbrEnabled && self->hSbrDecoder) ? SAC_INTERFACE_QMF : SAC_INTERFACE_TIME; /* needs to be done before first SBR apply. */ mpegSurroundDecoder_ConfigureQmfDomain( (CMpegSurroundDecoder *)self->pMpegSurroundDecoder, sac_interface, (UINT)self->streamInfo.aacSampleRate, self->streamInfo.aot); if (self->qmfDomain.globalConf.nBandsAnalysis_requested > 0) { self->qmfDomain.globalConf.nQmfTimeSlots_requested = self->streamInfo.aacSamplesPerFrame / self->qmfDomain.globalConf.nBandsAnalysis_requested; } else { self->qmfDomain.globalConf.nQmfTimeSlots_requested = 0; } } switch (FDK_QmfDomain_Configure(&self->qmfDomain)) { default: case QMF_DOMAIN_INIT_ERROR: ErrorStatus = AAC_DEC_UNKNOWN; goto bail; case QMF_DOMAIN_OUT_OF_MEMORY: ErrorStatus = AAC_DEC_OUT_OF_MEMORY; goto bail; case QMF_DOMAIN_OK: break; } /* sbr decoder */ if ((ErrorStatus != AAC_DEC_OK) || (flags & AACDEC_CONCEAL) || self->pAacDecoderStaticChannelInfo[0]->concealmentInfo.concealState > ConcealState_FadeIn) { self->frameOK = 0; /* if an error has occured do concealment in the SBR decoder too */ } if (self->sbrEnabled && (!(self->flags[0] & AC_USAC_SCFGI3))) { SBR_ERROR sbrError = SBRDEC_OK; int chIdx, numCoreChannel = self->streamInfo.numChannels; /* set params */ sbrDecoder_SetParam(self->hSbrDecoder, SBR_SYSTEM_BITSTREAM_DELAY, self->sbrParams.bsDelay); sbrDecoder_SetParam( self->hSbrDecoder, SBR_FLUSH_DATA, (flags & AACDEC_FLUSH) | ((self->flushStatus && !(flags & AACDEC_CONCEAL)) ? AACDEC_FLUSH : 0)); if (self->streamInfo.aot == AOT_ER_AAC_ELD) { /* Configure QMF */ sbrDecoder_SetParam(self->hSbrDecoder, SBR_LD_QMF_TIME_ALIGN, (self->flags[0] & AC_MPS_PRESENT) ? 1 : 0); } { PCMDMX_ERROR dmxErr; INT maxOutCh = 0; dmxErr = pcmDmx_GetParam(self->hPcmUtils, MAX_NUMBER_OF_OUTPUT_CHANNELS, &maxOutCh); if ((dmxErr == PCMDMX_OK) && (maxOutCh == 1)) { /* Disable PS processing if we have to create a mono output signal. */ self->psPossible = 0; } } sbrDecoder_SetParam(self->hSbrDecoder, SBR_SKIP_QMF, (self->mpsEnableCurr) ? 2 : 0); PCM_AAC *input; input = (PCM_AAC *)self->workBufferCore2; FDKmemcpy(input, pTimeData3, sizeof(PCM_AAC) * (self->streamInfo.numChannels) * (self->streamInfo.frameSize)); /* apply SBR processing */ sbrError = sbrDecoder_Apply( self->hSbrDecoder, input, pTimeData3, timeData3Size, &self->streamInfo.numChannels, &self->streamInfo.sampleRate, &self->mapDescr, self->chMapIndex, self->frameOK, &self->psPossible, self->aacOutDataHeadroom, &timeDataHeadroom); if (sbrError == SBRDEC_OK) { /* Update data in streaminfo structure. Assume that the SBR upsampling factor is either 1, 2, 8/3 or 4. Maximum upsampling factor is 4 (CELP+SBR or USAC 4:1 SBR) */ self->flags[0] |= AC_SBR_PRESENT; if (self->streamInfo.aacSampleRate != self->streamInfo.sampleRate) { if (self->streamInfo.aacSampleRate >> 2 == self->streamInfo.sampleRate) { self->streamInfo.frameSize = self->streamInfo.aacSamplesPerFrame >> 2; self->streamInfo.outputDelay = self->streamInfo.outputDelay >> 2; } else if (self->streamInfo.aacSampleRate >> 1 == self->streamInfo.sampleRate) { self->streamInfo.frameSize = self->streamInfo.aacSamplesPerFrame >> 1; self->streamInfo.outputDelay = self->streamInfo.outputDelay >> 1; } else if (self->streamInfo.aacSampleRate << 1 == self->streamInfo.sampleRate) { self->streamInfo.frameSize = self->streamInfo.aacSamplesPerFrame << 1; self->streamInfo.outputDelay = self->streamInfo.outputDelay << 1; } else if (self->streamInfo.aacSampleRate << 2 == self->streamInfo.sampleRate) { self->streamInfo.frameSize = self->streamInfo.aacSamplesPerFrame << 2; self->streamInfo.outputDelay = self->streamInfo.outputDelay << 2; } else if (self->streamInfo.frameSize == 768) { self->streamInfo.frameSize = (self->streamInfo.aacSamplesPerFrame << 3) / 3; self->streamInfo.outputDelay = (self->streamInfo.outputDelay << 3) / 3; } else { ErrorStatus = AAC_DEC_SET_PARAM_FAIL; goto bail; } } else { self->streamInfo.frameSize = self->streamInfo.aacSamplesPerFrame; } self->streamInfo.outputDelay += sbrDecoder_GetDelay(self->hSbrDecoder); if (self->psPossible) { self->flags[0] |= AC_PS_PRESENT; } for (chIdx = numCoreChannel; chIdx < self->streamInfo.numChannels; chIdx += 1) { self->channelType[chIdx] = ACT_FRONT; self->channelIndices[chIdx] = chIdx; } } if (sbrError == SBRDEC_OUTPUT_BUFFER_TOO_SMALL) { ErrorStatus = AAC_DEC_OUTPUT_BUFFER_TOO_SMALL; goto bail; } } if (self->mpsEnableCurr) { int err, sac_interface, nChannels, frameSize; nChannels = self->streamInfo.numChannels; frameSize = self->streamInfo.frameSize; sac_interface = SAC_INTERFACE_TIME; if (self->sbrEnabled && self->hSbrDecoder) sac_interface = SAC_INTERFACE_QMF; if (self->streamInfo.aot == AOT_USAC) { if (self->flags[0] & AC_USAC_SCFGI3) { sac_interface = SAC_INTERFACE_TIME; } } err = mpegSurroundDecoder_SetParam( (CMpegSurroundDecoder *)self->pMpegSurroundDecoder, SACDEC_INTERFACE, sac_interface); if (err == 0) { err = mpegSurroundDecoder_Apply( (CMpegSurroundDecoder *)self->pMpegSurroundDecoder, (PCM_AAC *)self->workBufferCore2, pTimeData3, timeData3Size, self->streamInfo.aacSamplesPerFrame, &nChannels, &frameSize, self->streamInfo.sampleRate, self->streamInfo.aot, self->channelType, self->channelIndices, &self->mapDescr, self->aacOutDataHeadroom, &timeDataHeadroom); } if (err == MPS_OUTPUT_BUFFER_TOO_SMALL) { ErrorStatus = AAC_DEC_OUTPUT_BUFFER_TOO_SMALL; goto bail; } if (err == 0) { /* Update output parameter */ self->streamInfo.numChannels = nChannels; self->streamInfo.frameSize = frameSize; self->streamInfo.outputDelay += mpegSurroundDecoder_GetDelay( (CMpegSurroundDecoder *)self->pMpegSurroundDecoder); /* Save current parameter for possible concealment of next frame */ self->mpsOutChannelsLast = nChannels; self->mpsFrameSizeLast = frameSize; } else if ((self->mpsOutChannelsLast > 0) && (self->mpsFrameSizeLast > 0)) { /* Restore parameters of last frame ... */ self->streamInfo.numChannels = self->mpsOutChannelsLast; self->streamInfo.frameSize = self->mpsFrameSizeLast; /* ... and clear output buffer so that potentially corrupted data does * not reach the framework. */ FDKmemclear(pTimeData3, self->mpsOutChannelsLast * self->mpsFrameSizeLast * sizeof(PCM_AAC)); /* Additionally proclaim that this frame had errors during decoding. */ ErrorStatus = AAC_DEC_DECODE_FRAME_ERROR; } else { ErrorStatus = AAC_DEC_UNKNOWN; /* no output */ } } /* SBR decoder for Unified Stereo Config (stereoConfigIndex == 3) */ if (self->sbrEnabled && (self->flags[0] & AC_USAC_SCFGI3)) { SBR_ERROR sbrError = SBRDEC_OK; /* set params */ sbrDecoder_SetParam(self->hSbrDecoder, SBR_SYSTEM_BITSTREAM_DELAY, self->sbrParams.bsDelay); sbrDecoder_SetParam(self->hSbrDecoder, SBR_SKIP_QMF, 1); /* apply SBR processing */ sbrError = sbrDecoder_Apply( self->hSbrDecoder, pTimeData3, pTimeData3, timeData3Size, &self->streamInfo.numChannels, &self->streamInfo.sampleRate, &self->mapDescr, self->chMapIndex, self->frameOK, &self->psPossible, self->aacOutDataHeadroom, &timeDataHeadroom); if (sbrError == SBRDEC_OK) { /* Update data in streaminfo structure. Assume that the SBR upsampling * factor is either 1,2 or 4 */ self->flags[0] |= AC_SBR_PRESENT; if (self->streamInfo.aacSampleRate != self->streamInfo.sampleRate) { if (self->streamInfo.frameSize == 768) { self->streamInfo.frameSize = (self->streamInfo.aacSamplesPerFrame * 8) / 3; } else if (self->streamInfo.aacSampleRate << 2 == self->streamInfo.sampleRate) { self->streamInfo.frameSize = self->streamInfo.aacSamplesPerFrame << 2; } else { self->streamInfo.frameSize = self->streamInfo.aacSamplesPerFrame << 1; } } self->flags[0] &= ~AC_PS_PRESENT; } if (sbrError == SBRDEC_OUTPUT_BUFFER_TOO_SMALL) { ErrorStatus = AAC_DEC_OUTPUT_BUFFER_TOO_SMALL; goto bail; } } { if ((INT)PCM_OUT_HEADROOM != timeDataHeadroom) { for (int i = ((self->streamInfo.frameSize * self->streamInfo.numChannels) - 1); i >= 0; i--) { pTimeData2[i] = (PCM_DEC)pTimeData3[i] >> (PCM_OUT_HEADROOM - timeDataHeadroom); } } } { if ((FDK_drcDec_GetParam(self->hUniDrcDecoder, DRC_DEC_IS_ACTIVE)) && !(self->flags[0] & AC_RSV603DA)) { /* Apply DRC gains*/ int ch, drcDelay = 0; int needsDeinterleaving = 0; FIXP_DBL *drcWorkBuffer = NULL; FIXP_DBL channelGain[(8)]; int reverseInChannelMap[(8)]; int reverseOutChannelMap[(8)]; int numDrcOutChannels = FDK_drcDec_GetParam( self->hUniDrcDecoder, DRC_DEC_TARGET_CHANNEL_COUNT_SELECTED); FDKmemclear(channelGain, sizeof(channelGain)); for (ch = 0; ch < (8); ch++) { reverseInChannelMap[ch] = ch; reverseOutChannelMap[ch] = ch; } /* Update sampleRate and frameSize. This may be necessary in case of * implicit SBR signaling */ FDK_drcDec_SetParam(self->hUniDrcDecoder, DRC_DEC_SAMPLE_RATE, self->streamInfo.sampleRate); FDK_drcDec_SetParam(self->hUniDrcDecoder, DRC_DEC_FRAME_SIZE, self->streamInfo.frameSize); /* If SBR and/or MPS is active, the DRC gains are aligned to the QMF domain signal before the QMF synthesis. Therefore the DRC gains need to be delayed by the QMF synthesis delay. */ if (self->sbrEnabled) drcDelay = 257; if (self->mpsEnableCurr) drcDelay = 257; /* Take into account concealment delay */ drcDelay += CConcealment_GetDelay(&self->concealCommonData) * self->streamInfo.frameSize; for (ch = 0; ch < self->streamInfo.numChannels; ch++) { UCHAR mapValue = FDK_chMapDescr_getMapValue( &self->mapDescr, (UCHAR)ch, self->chMapIndex); if (mapValue < (8)) reverseInChannelMap[mapValue] = ch; } for (ch = 0; ch < (int)numDrcOutChannels; ch++) { UCHAR mapValue = FDK_chMapDescr_getMapValue( &self->mapDescr, (UCHAR)ch, numDrcOutChannels); if (mapValue < (8)) reverseOutChannelMap[mapValue] = ch; } /* The output of SBR and MPS is interleaved. Deinterleaving may be * necessary for FDK_drcDec_ProcessTime, which accepts deinterleaved * audio only. */ if ((self->streamInfo.numChannels > 1) && (0 || (self->sbrEnabled) || (self->mpsEnableCurr))) { /* interleaving/deinterleaving is performed on upper part of * pTimeData2. Check if this buffer is large enough. */ if (timeData2Size < (INT)(2 * self->streamInfo.numChannels * self->streamInfo.frameSize)) { ErrorStatus = AAC_DEC_UNKNOWN; goto bail; } needsDeinterleaving = 1; drcWorkBuffer = (FIXP_DBL *)pTimeData2 + self->streamInfo.numChannels * self->streamInfo.frameSize; FDK_deinterleave( pTimeData2, drcWorkBuffer, self->streamInfo.numChannels, self->streamInfo.frameSize, self->streamInfo.frameSize); } else { drcWorkBuffer = pTimeData2; } /* prepare Loudness Normalisation gain */ FDK_drcDec_SetParam(self->hUniDrcDecoder, DRC_DEC_TARGET_LOUDNESS, (INT)-self->defaultTargetLoudness * FL2FXCONST_DBL(1.0f / (float)(1 << 9))); FDK_drcDec_SetChannelGains(self->hUniDrcDecoder, self->streamInfo.numChannels, self->streamInfo.frameSize, channelGain, drcWorkBuffer, self->streamInfo.frameSize); FDK_drcDec_Preprocess(self->hUniDrcDecoder); /* apply DRC1 gain sequence */ for (ch = 0; ch < self->streamInfo.numChannels; ch++) { FDK_drcDec_ProcessTime(self->hUniDrcDecoder, drcDelay, DRC_DEC_DRC1, ch, reverseInChannelMap[ch] - ch, 1, drcWorkBuffer, self->streamInfo.frameSize); } /* apply downmix */ FDK_drcDec_ApplyDownmix( self->hUniDrcDecoder, reverseInChannelMap, reverseOutChannelMap, drcWorkBuffer, &self->streamInfo.numChannels); /* self->streamInfo.numChannels may change here */ /* apply DRC2/3 gain sequence */ for (ch = 0; ch < self->streamInfo.numChannels; ch++) { FDK_drcDec_ProcessTime(self->hUniDrcDecoder, drcDelay, DRC_DEC_DRC2_DRC3, ch, reverseOutChannelMap[ch] - ch, 1, drcWorkBuffer, self->streamInfo.frameSize); } if (needsDeinterleaving) { FDK_interleave( drcWorkBuffer, pTimeData2, self->streamInfo.numChannels, self->streamInfo.frameSize, self->streamInfo.frameSize); } } } if (FDK_drcDec_GetParam(self->hUniDrcDecoder, DRC_DEC_IS_ACTIVE)) { /* return output loudness information for MPEG-D DRC */ LONG outputLoudness = FDK_drcDec_GetParam(self->hUniDrcDecoder, DRC_DEC_OUTPUT_LOUDNESS); if (outputLoudness == DRC_DEC_LOUDNESS_NOT_PRESENT) { /* no valid MPEG-D DRC loudness value contained */ self->streamInfo.outputLoudness = -1; } else { if (outputLoudness > 0) { /* positive output loudness values (very unusual) are limited to 0 * dB */ self->streamInfo.outputLoudness = 0; } else { self->streamInfo.outputLoudness = -(INT)outputLoudness >> 22; /* negate and scale from e = 7 to e = (31-2) */ } } } else { /* return output loudness information for MPEG-4 DRC */ if (self->streamInfo.drcProgRefLev < 0) { /* no MPEG-4 DRC loudness metadata contained */ self->streamInfo.outputLoudness = -1; } else { if (self->defaultTargetLoudness < 0) { /* loudness normalization is off */ self->streamInfo.outputLoudness = self->streamInfo.drcProgRefLev; } else { self->streamInfo.outputLoudness = self->defaultTargetLoudness; } } } if (self->streamInfo.extAot != AOT_AAC_SLS) { INT pcmLimiterScale = 0; INT interleaved = 0; interleaved |= (self->sbrEnabled) ? 1 : 0; interleaved |= (self->mpsEnableCurr) ? 1 : 0; PCMDMX_ERROR dmxErr = PCMDMX_OK; if ((flags & AACDEC_INTR) && (accessUnit == 0)) { /* delete data from the past (e.g. mixdown coeficients) */ pcmDmx_Reset(self->hPcmUtils, PCMDMX_RESET_BS_DATA); } if (flags & (AACDEC_CLRHIST)) { if (!(self->flags[0] & AC_USAC)) { /* delete data from the past (e.g. mixdown coeficients) */ pcmDmx_Reset(self->hPcmUtils, PCMDMX_RESET_BS_DATA); } } /* do PCM post processing */ dmxErr = pcmDmx_ApplyFrame(self->hPcmUtils, pTimeData2, timeData2Size, self->streamInfo.frameSize, &self->streamInfo.numChannels, interleaved, self->channelType, self->channelIndices, &self->mapDescr, &pcmLimiterScale); if (dmxErr == PCMDMX_OUTPUT_BUFFER_TOO_SMALL) { ErrorStatus = AAC_DEC_OUTPUT_BUFFER_TOO_SMALL; goto bail; } if ((ErrorStatus == AAC_DEC_OK) && (dmxErr == PCMDMX_INVALID_MODE)) { /* Announce the framework that the current combination of channel * configuration and downmix settings are not know to produce a * predictable behavior and thus maybe produce strange output. */ ErrorStatus = AAC_DEC_DECODE_FRAME_ERROR; } pcmLimiterScale += PCM_OUT_HEADROOM; if (flags & AACDEC_CLRHIST) { if (!(self->flags[0] & AC_USAC)) { /* Reset DRC data */ aacDecoder_drcReset(self->hDrcInfo); /* Delete the delayed signal. */ pcmLimiter_Reset(self->hLimiter); } } /* Set applyExtGain if DRC processing is enabled and if progRefLevelPresent is present for the first time. Consequences: The headroom of the output signal can be set to AACDEC_DRC_GAIN_SCALING only for audio formats which support legacy DRC Level Normalization. For all other audio formats the headroom of the output signal is set to PCM_OUT_HEADROOM. */ if (self->hDrcInfo->enable && (self->hDrcInfo->progRefLevelPresent == 1)) { self->hDrcInfo->applyExtGain |= 1; } /* Check whether time data buffer is large enough. */ if (timeDataSize < (self->streamInfo.numChannels * self->streamInfo.frameSize)) { ErrorStatus = AAC_DEC_OUTPUT_BUFFER_TOO_SMALL; goto bail; } if (self->limiterEnableCurr) { /* use workBufferCore2 buffer for interleaving */ PCM_LIM *pInterleaveBuffer; int blockLength = self->streamInfo.frameSize; /* Set actual signal parameters */ pcmLimiter_SetNChannels(self->hLimiter, self->streamInfo.numChannels); pcmLimiter_SetSampleRate(self->hLimiter, self->streamInfo.sampleRate); if ((self->streamInfo.numChannels == 1) || (self->sbrEnabled) || (self->mpsEnableCurr)) { pInterleaveBuffer = (PCM_LIM *)pTimeData2; } else { pInterleaveBuffer = (PCM_LIM *)self->workBufferCore2; /* applyLimiter requests for interleaved data */ /* Interleave ouput buffer */ FDK_interleave(pTimeData2, pInterleaveBuffer, self->streamInfo.numChannels, blockLength, self->streamInfo.frameSize); } FIXP_DBL *pGainPerSample = NULL; if (self->hDrcInfo->enable && self->hDrcInfo->applyExtGain) { pGainPerSample = self->workBufferCore1; if ((INT)GetRequiredMemWorkBufferCore1() < (INT)(self->streamInfo.frameSize * sizeof(FIXP_DBL))) { ErrorStatus = AAC_DEC_UNKNOWN; goto bail; } pcmLimiterScale = applyDrcLevelNormalization( self->hDrcInfo, (PCM_DEC *)pInterleaveBuffer, self->extGain, pGainPerSample, pcmLimiterScale, self->extGainDelay, self->streamInfo.frameSize, self->streamInfo.numChannels, 1, 1); } pcmLimiter_Apply(self->hLimiter, pInterleaveBuffer, pTimeData, pGainPerSample, pcmLimiterScale, self->streamInfo.frameSize); { /* Announce the additional limiter output delay */ self->streamInfo.outputDelay += pcmLimiter_GetDelay(self->hLimiter); } } else { if (self->hDrcInfo->enable && self->hDrcInfo->applyExtGain) { pcmLimiterScale = applyDrcLevelNormalization( self->hDrcInfo, pTimeData2, self->extGain, NULL, pcmLimiterScale, self->extGainDelay, self->streamInfo.frameSize, self->streamInfo.numChannels, (interleaved || (self->streamInfo.numChannels == 1)) ? 1 : self->streamInfo.frameSize, 0); } /* If numChannels = 1 we do not need interleaving. The same applies if SBR or MPS are used, since their output is interleaved already (resampled or not) */ if ((self->streamInfo.numChannels == 1) || (self->sbrEnabled) || (self->mpsEnableCurr)) { scaleValuesSaturate( pTimeData, pTimeData2, self->streamInfo.frameSize * self->streamInfo.numChannels, pcmLimiterScale); } else { scaleValuesSaturate( (INT_PCM *)self->workBufferCore2, pTimeData2, self->streamInfo.frameSize * self->streamInfo.numChannels, pcmLimiterScale); /* Interleave ouput buffer */ FDK_interleave((INT_PCM *)self->workBufferCore2, pTimeData, self->streamInfo.numChannels, self->streamInfo.frameSize, self->streamInfo.frameSize); } } } /* if (self->streamInfo.extAot != AOT_AAC_SLS)*/ if (self->flags[0] & AC_USAC) { if (self->flushStatus == AACDEC_USAC_DASH_IPF_FLUSH_ON && !(flags & AACDEC_CONCEAL)) { CAacDecoder_PrepareCrossFade(pTimeData, self->pTimeDataFlush, self->streamInfo.numChannels, self->streamInfo.frameSize, 1); } /* prepare crossfade buffer for fade in */ if (!applyCrossfade && self->applyCrossfade && !(flags & AACDEC_CONCEAL)) { for (int ch = 0; ch < self->streamInfo.numChannels; ch++) { for (int i = 0; i < TIME_DATA_FLUSH_SIZE; i++) { self->pTimeDataFlush[ch][i] = 0; } } applyCrossfade = 1; } if (applyCrossfade && self->applyCrossfade && !(accessUnit < numPrerollAU) && (self->buildUpStatus == AACDEC_USAC_BUILD_UP_ON)) { CAacDecoder_ApplyCrossFade(pTimeData, self->pTimeDataFlush, self->streamInfo.numChannels, self->streamInfo.frameSize, 1); self->applyCrossfade = 0; } } /* Signal interruption to take effect in next frame. */ if ((flags & AACDEC_FLUSH || self->flushStatus) && !(flags & AACDEC_CONCEAL)) { aacDecoder_SignalInterruption(self); } /* Update externally visible copy of flags */ self->streamInfo.flags = self->flags[0]; } /* USAC DASH IPF flushing possible end */ if (accessUnit < numPrerollAU) { FDKpushBack(hBsAu, auStartAnchor - (INT)FDKgetValidBits(hBsAu)); } else { if ((self->buildUpStatus == AACDEC_RSV60_BUILD_UP_ON) || (self->buildUpStatus == AACDEC_RSV60_BUILD_UP_ON_IN_BAND) || (self->buildUpStatus == AACDEC_USAC_BUILD_UP_ON)) { self->buildUpCnt--; if (self->buildUpCnt < 0) { self->buildUpStatus = 0; } } if (self->flags[0] & AC_USAC) { if (self->flushStatus == AACDEC_USAC_DASH_IPF_FLUSH_ON && !(flags & AACDEC_CONCEAL)) { self->streamInfo.frameSize = 0; } } } if (self->flushStatus != AACDEC_USAC_DASH_IPF_FLUSH_ON) { accessUnit++; } } while ((accessUnit < numAccessUnits) || ((self->flushStatus == AACDEC_USAC_DASH_IPF_FLUSH_ON) && !(flags & AACDEC_CONCEAL))); bail: /* error in renderer part occurred, ErrorStatus was set to invalid output */ if (fEndAuNotAdjusted && !IS_OUTPUT_VALID(ErrorStatus) && (accessUnit < numPrerollAU)) { transportDec_EndAccessUnit(self->hInput); } /* Update Statistics */ aacDecoder_UpdateBitStreamCounters(&self->streamInfo, hBs, nBits, ErrorStatus); if (((self->streamInfo.numChannels <= 0) || (self->streamInfo.frameSize <= 0) || (self->streamInfo.sampleRate <= 0)) && IS_OUTPUT_VALID(ErrorStatus)) { /* Ensure consistency of IS_OUTPUT_VALID() macro. */ ErrorStatus = AAC_DEC_UNKNOWN; } if (!IS_OUTPUT_VALID(ErrorStatus)) { FDKmemclear(pTimeData, timeDataSize * sizeof(*pTimeData)); } return ErrorStatus; } LINKSPEC_CPP void aacDecoder_Close(HANDLE_AACDECODER self) { if (self == NULL) return; if (self->hLimiter != NULL) { pcmLimiter_Destroy(self->hLimiter); } if (self->hPcmUtils != NULL) { pcmDmx_Close(&self->hPcmUtils); } FDK_drcDec_Close(&self->hUniDrcDecoder); if (self->pMpegSurroundDecoder != NULL) { mpegSurroundDecoder_Close( (CMpegSurroundDecoder *)self->pMpegSurroundDecoder); } if (self->hSbrDecoder != NULL) { sbrDecoder_Close(&self->hSbrDecoder); } if (self->hInput != NULL) { transportDec_Close(&self->hInput); } CAacDecoder_Close(self); } LINKSPEC_CPP CStreamInfo *aacDecoder_GetStreamInfo(HANDLE_AACDECODER self) { return CAacDecoder_GetStreamInfo(self); } LINKSPEC_CPP INT aacDecoder_GetLibInfo(LIB_INFO *info) { int i; if (info == NULL) { return -1; } sbrDecoder_GetLibInfo(info); mpegSurroundDecoder_GetLibInfo(info); transportDec_GetLibInfo(info); FDK_toolsGetLibInfo(info); pcmDmx_GetLibInfo(info); pcmLimiter_GetLibInfo(info); FDK_drcDec_GetLibInfo(info); /* search for next free tab */ for (i = 0; i < FDK_MODULE_LAST; i++) { if (info[i].module_id == FDK_NONE) break; } if (i == FDK_MODULE_LAST) { return -1; } info += i; info->module_id = FDK_AACDEC; /* build own library info */ info->version = LIB_VERSION(AACDECODER_LIB_VL0, AACDECODER_LIB_VL1, AACDECODER_LIB_VL2); LIB_VERSION_STRING(info); info->build_date = AACDECODER_LIB_BUILD_DATE; info->build_time = AACDECODER_LIB_BUILD_TIME; info->title = AACDECODER_LIB_TITLE; /* Set flags */ info->flags = 0 | CAPF_AAC_LC | CAPF_ER_AAC_LC | CAPF_ER_AAC_SCAL | CAPF_AAC_VCB11 | CAPF_AAC_HCR | CAPF_AAC_RVLC | CAPF_ER_AAC_LD | CAPF_ER_AAC_ELD | CAPF_AAC_CONCEALMENT | CAPF_AAC_DRC | CAPF_AAC_MPEG4 | CAPF_AAC_DRM_BSFORMAT | CAPF_AAC_1024 | CAPF_AAC_960 | CAPF_AAC_512 | CAPF_AAC_480 | CAPF_AAC_ELD_DOWNSCALE | CAPF_AAC_USAC | CAPF_ER_AAC_ELDV2 | CAPF_AAC_UNIDRC; /* End of flags */ return 0; } fdk-aac-2.0.2/libAACdec/src/arm/000077500000000000000000000000001404225310700160745ustar00rootroot00000000000000fdk-aac-2.0.2/libAACdec/src/arm/block_arm.cpp000066400000000000000000000141001404225310700205250ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Arthur Tritthart Description: (ARM optimised) Scaling of spectral data *******************************************************************************/ #define FUNCTION_CBlock_ScaleSpectralData_func1 /* Note: This loop is only separated for ARM in order to save cycles by loop unrolling. The ARM core provides by default a 5-cycle loop overhead per sample, that goes down to 1-cycle per sample with an optimal 4x-loop construct (do - 4x - while). */ static inline void CBlock_ScaleSpectralData_func1( FIXP_DBL *pSpectrum, int maxSfbs, const SHORT *RESTRICT BandOffsets, int SpecScale_window, const SHORT *RESTRICT pSfbScale, int window) { int band_offset = 0; for (int band = 0; band < maxSfbs; band++) { int runs = band_offset; band_offset = BandOffsets[band + 1]; runs = band_offset - runs; /* is always a multiple of 4 */ FDK_ASSERT((runs & 3) == 0); int scale = fMin(DFRACT_BITS - 1, SpecScale_window - pSfbScale[window * 16 + band]); if (scale) { do { FIXP_DBL tmp0, tmp1, tmp2, tmp3; tmp0 = pSpectrum[0]; tmp1 = pSpectrum[1]; tmp2 = pSpectrum[2]; tmp3 = pSpectrum[3]; tmp0 >>= scale; tmp1 >>= scale; tmp2 >>= scale; tmp3 >>= scale; *pSpectrum++ = tmp0; *pSpectrum++ = tmp1; *pSpectrum++ = tmp2; *pSpectrum++ = tmp3; } while ((runs = runs - 4) != 0); } else { pSpectrum += runs; } } } fdk-aac-2.0.2/libAACdec/src/block.cpp000066400000000000000000001341011404225310700171130ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Josef Hoepfl Description: long/short-block decoding *******************************************************************************/ #include "block.h" #include "aac_rom.h" #include "FDK_bitstream.h" #include "scale.h" #include "FDK_tools_rom.h" #include "usacdec_fac.h" #include "usacdec_lpd.h" #include "usacdec_lpc.h" #include "FDK_trigFcts.h" #include "ac_arith_coder.h" #include "aacdec_hcr.h" #include "rvlc.h" #if defined(__arm__) #include "arm/block_arm.cpp" #endif /*! \brief Read escape sequence of codeword The function reads the escape sequence from the bitstream, if the absolute value of the quantized coefficient has the value 16. A limitation is implemented to maximal 21 bits according to ISO/IEC 14496-3:2009(E) 4.6.3.3. This limits the escape prefix to a maximum of eight 1's. If more than eight 1's are read, MAX_QUANTIZED_VALUE + 1 is returned, independent of the sign of parameter q. \return quantized coefficient */ LONG CBlock_GetEscape(HANDLE_FDK_BITSTREAM bs, /*!< pointer to bitstream */ const LONG q) /*!< quantized coefficient */ { if (fAbs(q) != 16) return (q); LONG i, off; for (i = 4; i < 13; i++) { if (FDKreadBit(bs) == 0) break; } if (i == 13) return (MAX_QUANTIZED_VALUE + 1); off = FDKreadBits(bs, i); i = off + (1 << i); if (q < 0) i = -i; return i; } AAC_DECODER_ERROR CBlock_ReadScaleFactorData( CAacDecoderChannelInfo *pAacDecoderChannelInfo, HANDLE_FDK_BITSTREAM bs, UINT flags) { int temp; int band; int group; int position = 0; /* accu for intensity delta coding */ int factor = pAacDecoderChannelInfo->pDynData->RawDataInfo .GlobalGain; /* accu for scale factor delta coding */ UCHAR *pCodeBook = pAacDecoderChannelInfo->pDynData->aCodeBook; SHORT *pScaleFactor = pAacDecoderChannelInfo->pDynData->aScaleFactor; const CodeBookDescription *hcb = &AACcodeBookDescriptionTable[BOOKSCL]; const USHORT(*CodeBook)[HuffmanEntries] = hcb->CodeBook; int ScaleFactorBandsTransmitted = GetScaleFactorBandsTransmitted(&pAacDecoderChannelInfo->icsInfo); for (group = 0; group < GetWindowGroups(&pAacDecoderChannelInfo->icsInfo); group++) { for (band = 0; band < ScaleFactorBandsTransmitted; band++) { switch (pCodeBook[band]) { case ZERO_HCB: /* zero book */ pScaleFactor[band] = 0; break; default: /* decode scale factor */ if (!((flags & (AC_USAC | AC_RSVD50 | AC_RSV603DA)) && band == 0 && group == 0)) { temp = CBlock_DecodeHuffmanWordCB(bs, CodeBook); factor += temp - 60; /* MIDFAC 1.5 dB */ } pScaleFactor[band] = factor - 100; break; case INTENSITY_HCB: /* intensity steering */ case INTENSITY_HCB2: temp = CBlock_DecodeHuffmanWordCB(bs, CodeBook); position += temp - 60; pScaleFactor[band] = position - 100; break; case NOISE_HCB: /* PNS */ if (flags & (AC_MPEGD_RES | AC_USAC | AC_RSVD50 | AC_RSV603DA)) { return AAC_DEC_PARSE_ERROR; } CPns_Read(&pAacDecoderChannelInfo->data.aac.PnsData, bs, hcb, pAacDecoderChannelInfo->pDynData->aScaleFactor, pAacDecoderChannelInfo->pDynData->RawDataInfo.GlobalGain, band, group); break; } } pCodeBook += 16; pScaleFactor += 16; } return AAC_DEC_OK; } void CBlock_ScaleSpectralData(CAacDecoderChannelInfo *pAacDecoderChannelInfo, UCHAR maxSfbs, SamplingRateInfo *pSamplingRateInfo) { int band; int window; const SHORT *RESTRICT pSfbScale = pAacDecoderChannelInfo->pDynData->aSfbScale; SHORT *RESTRICT pSpecScale = pAacDecoderChannelInfo->specScale; int groupwin, group; const SHORT *RESTRICT BandOffsets = GetScaleFactorBandOffsets( &pAacDecoderChannelInfo->icsInfo, pSamplingRateInfo); SPECTRAL_PTR RESTRICT pSpectralCoefficient = pAacDecoderChannelInfo->pSpectralCoefficient; FDKmemclear(pSpecScale, 8 * sizeof(SHORT)); for (window = 0, group = 0; group < GetWindowGroups(&pAacDecoderChannelInfo->icsInfo); group++) { for (groupwin = 0; groupwin < GetWindowGroupLength( &pAacDecoderChannelInfo->icsInfo, group); groupwin++, window++) { int SpecScale_window = pSpecScale[window]; FIXP_DBL *pSpectrum = SPEC(pSpectralCoefficient, window, pAacDecoderChannelInfo->granuleLength); /* find scaling for current window */ for (band = 0; band < maxSfbs; band++) { SpecScale_window = fMax(SpecScale_window, (int)pSfbScale[window * 16 + band]); } if (pAacDecoderChannelInfo->pDynData->TnsData.Active && pAacDecoderChannelInfo->pDynData->TnsData.NumberOfFilters[window] > 0) { int filter_index, SpecScale_window_tns; int tns_start, tns_stop; /* Find max scale of TNS bands */ SpecScale_window_tns = 0; tns_start = GetMaximumTnsBands(&pAacDecoderChannelInfo->icsInfo, pSamplingRateInfo->samplingRateIndex); tns_stop = 0; for (filter_index = 0; filter_index < (int)pAacDecoderChannelInfo->pDynData->TnsData .NumberOfFilters[window]; filter_index++) { for (band = pAacDecoderChannelInfo->pDynData->TnsData .Filter[window][filter_index] .StartBand; band < pAacDecoderChannelInfo->pDynData->TnsData .Filter[window][filter_index] .StopBand; band++) { SpecScale_window_tns = fMax(SpecScale_window_tns, (int)pSfbScale[window * 16 + band]); } /* Find TNS line boundaries for all TNS filters */ tns_start = fMin(tns_start, (int)pAacDecoderChannelInfo->pDynData->TnsData .Filter[window][filter_index] .StartBand); tns_stop = fMax(tns_stop, (int)pAacDecoderChannelInfo->pDynData->TnsData .Filter[window][filter_index] .StopBand); } SpecScale_window_tns = SpecScale_window_tns + pAacDecoderChannelInfo->pDynData->TnsData.GainLd; FDK_ASSERT(tns_stop >= tns_start); /* Consider existing headroom of all MDCT lines inside the TNS bands. */ SpecScale_window_tns -= getScalefactor(pSpectrum + BandOffsets[tns_start], BandOffsets[tns_stop] - BandOffsets[tns_start]); if (SpecScale_window <= 17) { SpecScale_window_tns++; } /* Add enough mantissa head room such that the spectrum is still representable after applying TNS. */ SpecScale_window = fMax(SpecScale_window, SpecScale_window_tns); } /* store scaling of current window */ pSpecScale[window] = SpecScale_window; #ifdef FUNCTION_CBlock_ScaleSpectralData_func1 CBlock_ScaleSpectralData_func1(pSpectrum, maxSfbs, BandOffsets, SpecScale_window, pSfbScale, window); #else /* FUNCTION_CBlock_ScaleSpectralData_func1 */ for (band = 0; band < maxSfbs; band++) { int scale = fMin(DFRACT_BITS - 1, SpecScale_window - pSfbScale[window * 16 + band]); if (scale) { FDK_ASSERT(scale > 0); /* following relation can be used for optimizations: * (BandOffsets[i]%4) == 0 for all i */ int max_index = BandOffsets[band + 1]; DWORD_ALIGNED(pSpectrum); for (int index = BandOffsets[band]; index < max_index; index++) { pSpectrum[index] >>= scale; } } } #endif /* FUNCTION_CBlock_ScaleSpectralData_func1 */ } } } AAC_DECODER_ERROR CBlock_ReadSectionData( HANDLE_FDK_BITSTREAM bs, CAacDecoderChannelInfo *pAacDecoderChannelInfo, const SamplingRateInfo *pSamplingRateInfo, const UINT flags) { int top, band; int sect_len, sect_len_incr; int group; UCHAR sect_cb; UCHAR *pCodeBook = pAacDecoderChannelInfo->pDynData->aCodeBook; /* HCR input (long) */ SHORT *pNumLinesInSec = pAacDecoderChannelInfo->pDynData->specificTo.aac.aNumLineInSec4Hcr; int numLinesInSecIdx = 0; UCHAR *pHcrCodeBook = pAacDecoderChannelInfo->pDynData->specificTo.aac.aCodeBooks4Hcr; const SHORT *BandOffsets = GetScaleFactorBandOffsets( &pAacDecoderChannelInfo->icsInfo, pSamplingRateInfo); pAacDecoderChannelInfo->pDynData->specificTo.aac.numberSection = 0; AAC_DECODER_ERROR ErrorStatus = AAC_DEC_OK; FDKmemclear(pCodeBook, sizeof(UCHAR) * (8 * 16)); const int nbits = (IsLongBlock(&pAacDecoderChannelInfo->icsInfo) == 1) ? 5 : 3; int sect_esc_val = (1 << nbits) - 1; UCHAR ScaleFactorBandsTransmitted = GetScaleFactorBandsTransmitted(&pAacDecoderChannelInfo->icsInfo); for (group = 0; group < GetWindowGroups(&pAacDecoderChannelInfo->icsInfo); group++) { for (band = 0; band < ScaleFactorBandsTransmitted;) { sect_len = 0; if (flags & AC_ER_VCB11) { sect_cb = (UCHAR)FDKreadBits(bs, 5); } else sect_cb = (UCHAR)FDKreadBits(bs, 4); if (((flags & AC_ER_VCB11) == 0) || (sect_cb < 11) || ((sect_cb > 11) && (sect_cb < 16))) { sect_len_incr = FDKreadBits(bs, nbits); while (sect_len_incr == sect_esc_val) { sect_len += sect_esc_val; sect_len_incr = FDKreadBits(bs, nbits); } } else { sect_len_incr = 1; } sect_len += sect_len_incr; top = band + sect_len; if (flags & AC_ER_HCR) { /* HCR input (long) -- collecting sideinfo (for HCR-_long_ only) */ if (numLinesInSecIdx >= MAX_SFB_HCR) { return AAC_DEC_PARSE_ERROR; } if (top > (int)GetNumberOfScaleFactorBands( &pAacDecoderChannelInfo->icsInfo, pSamplingRateInfo)) { return AAC_DEC_PARSE_ERROR; } pNumLinesInSec[numLinesInSecIdx] = BandOffsets[top] - BandOffsets[band]; numLinesInSecIdx++; if (sect_cb == BOOKSCL) { return AAC_DEC_INVALID_CODE_BOOK; } else { *pHcrCodeBook++ = sect_cb; } pAacDecoderChannelInfo->pDynData->specificTo.aac.numberSection++; } /* Check spectral line limits */ if (IsLongBlock(&(pAacDecoderChannelInfo->icsInfo))) { if (top > 64) { return AAC_DEC_DECODE_FRAME_ERROR; } } else { /* short block */ if (top + group * 16 > (8 * 16)) { return AAC_DEC_DECODE_FRAME_ERROR; } } /* Check if decoded codebook index is feasible */ if ((sect_cb == BOOKSCL) || ((sect_cb == INTENSITY_HCB || sect_cb == INTENSITY_HCB2) && pAacDecoderChannelInfo->pDynData->RawDataInfo.CommonWindow == 0)) { return AAC_DEC_INVALID_CODE_BOOK; } /* Store codebook index */ for (; band < top; band++) { pCodeBook[group * 16 + band] = sect_cb; } } } return ErrorStatus; } /* mso: provides a faster way to i-quantize a whole band in one go */ /** * \brief inverse quantize one sfb. Each value of the sfb is processed according * to the formula: spectrum[i] = Sign(spectrum[i]) * Matissa(spectrum[i])^(4/3) * * 2^(lsb/4). * \param spectrum pointer to first line of the sfb to be inverse quantized. * \param noLines number of lines belonging to the sfb. * \param lsb last 2 bits of the scale factor of the sfb. * \param scale max allowed shift scale for the sfb. */ static inline void InverseQuantizeBand( FIXP_DBL *RESTRICT spectrum, const FIXP_DBL *RESTRICT InverseQuantTabler, const FIXP_DBL *RESTRICT MantissaTabler, const SCHAR *RESTRICT ExponentTabler, INT noLines, INT scale) { scale = scale + 1; /* +1 to compensate fMultDiv2 shift-right in loop */ FIXP_DBL *RESTRICT ptr = spectrum; FIXP_DBL signedValue; for (INT i = noLines; i--;) { if ((signedValue = *ptr++) != FL2FXCONST_DBL(0)) { FIXP_DBL value = fAbs(signedValue); UINT freeBits = CntLeadingZeros(value); UINT exponent = 32 - freeBits; UINT x = (UINT)(LONG)value << (INT)freeBits; x <<= 1; /* shift out sign bit to avoid masking later on */ UINT tableIndex = x >> 24; x = (x >> 20) & 0x0F; UINT r0 = (UINT)(LONG)InverseQuantTabler[tableIndex + 0]; UINT r1 = (UINT)(LONG)InverseQuantTabler[tableIndex + 1]; UINT temp = (r1 - r0) * x + (r0 << 4); value = fMultDiv2((FIXP_DBL)temp, MantissaTabler[exponent]); /* + 1 compensates fMultDiv2() */ scaleValueInPlace(&value, scale + ExponentTabler[exponent]); signedValue = (signedValue < (FIXP_DBL)0) ? -value : value; ptr[-1] = signedValue; } } } static inline FIXP_DBL maxabs_D(const FIXP_DBL *pSpectralCoefficient, const int noLines) { /* Find max spectral line value of the current sfb */ FIXP_DBL locMax = (FIXP_DBL)0; int i; DWORD_ALIGNED(pSpectralCoefficient); for (i = noLines; i-- > 0;) { /* Expensive memory access */ locMax = fMax(fixp_abs(pSpectralCoefficient[i]), locMax); } return locMax; } AAC_DECODER_ERROR CBlock_InverseQuantizeSpectralData( CAacDecoderChannelInfo *pAacDecoderChannelInfo, SamplingRateInfo *pSamplingRateInfo, UCHAR *band_is_noise, UCHAR active_band_search) { int window, group, groupwin, band; int ScaleFactorBandsTransmitted = GetScaleFactorBandsTransmitted(&pAacDecoderChannelInfo->icsInfo); UCHAR *RESTRICT pCodeBook = pAacDecoderChannelInfo->pDynData->aCodeBook; SHORT *RESTRICT pSfbScale = pAacDecoderChannelInfo->pDynData->aSfbScale; SHORT *RESTRICT pScaleFactor = pAacDecoderChannelInfo->pDynData->aScaleFactor; const SHORT *RESTRICT BandOffsets = GetScaleFactorBandOffsets( &pAacDecoderChannelInfo->icsInfo, pSamplingRateInfo); const SHORT total_bands = GetScaleFactorBandsTotal(&pAacDecoderChannelInfo->icsInfo); FDKmemclear(pAacDecoderChannelInfo->pDynData->aSfbScale, (8 * 16) * sizeof(SHORT)); for (window = 0, group = 0; group < GetWindowGroups(&pAacDecoderChannelInfo->icsInfo); group++) { for (groupwin = 0; groupwin < GetWindowGroupLength( &pAacDecoderChannelInfo->icsInfo, group); groupwin++, window++) { /* inverse quantization */ for (band = 0; band < ScaleFactorBandsTransmitted; band++) { FIXP_DBL *pSpectralCoefficient = SPEC(pAacDecoderChannelInfo->pSpectralCoefficient, window, pAacDecoderChannelInfo->granuleLength) + BandOffsets[band]; FIXP_DBL locMax; const int noLines = BandOffsets[band + 1] - BandOffsets[band]; const int bnds = group * 16 + band; if ((pCodeBook[bnds] == ZERO_HCB) || (pCodeBook[bnds] == INTENSITY_HCB) || (pCodeBook[bnds] == INTENSITY_HCB2)) continue; if (pCodeBook[bnds] == NOISE_HCB) { /* Leave headroom for PNS values. + 1 because ceil(log2(2^(0.25*3))) = 1, worst case of additional headroom required because of the scalefactor. */ pSfbScale[window * 16 + band] = (pScaleFactor[bnds] >> 2) + 1; continue; } locMax = maxabs_D(pSpectralCoefficient, noLines); if (active_band_search) { if (locMax != FIXP_DBL(0)) { band_is_noise[group * 16 + band] = 0; } } /* Cheap robustness improvement - Do not remove!!! */ if (fixp_abs(locMax) > (FIXP_DBL)MAX_QUANTIZED_VALUE) { return AAC_DEC_PARSE_ERROR; } /* Added by Youliy Ninov: The inverse quantization operation is given by (ISO/IEC 14496-3:2009(E)) by: x_invquant=Sign(x_quant). abs(x_quant)^(4/3) We apply a gain, derived from the scale factor for the particular sfb, according to the following function: gain=2^(0.25*ScaleFactor) So, after scaling we have: x_rescale=gain*x_invquant=Sign(x_quant)*2^(0.25*ScaleFactor)*abs(s_quant)^(4/3) We could represent the ScaleFactor as: ScaleFactor= (ScaleFactor >> 2)*4 + ScaleFactor %4 When we substitute it we get: x_rescale=Sign(x_quant)*2^(ScaleFactor>>2)* ( 2^(0.25*(ScaleFactor%4))*abs(s_quant)^(4/3)) When we set: msb=(ScaleFactor>>2) and lsb=(ScaleFactor%4), we obtain: x_rescale=Sign(x_quant)*(2^msb)* ( 2^(lsb/4)*abs(s_quant)^(4/3)) The rescaled output can be represented by: mantissa : Sign(x_quant)*( 2^(lsb/4)*abs(s_quant)^(4/3)) exponent :(2^msb) */ int msb = pScaleFactor[bnds] >> 2; /* Inverse quantize band only if it is not empty */ if (locMax != FIXP_DBL(0)) { int lsb = pScaleFactor[bnds] & 0x03; int scale = EvaluatePower43(&locMax, lsb); scale = CntLeadingZeros(locMax) - scale - 2; pSfbScale[window * 16 + band] = msb - scale; InverseQuantizeBand(pSpectralCoefficient, InverseQuantTable, MantissaTable[lsb], ExponentTable[lsb], noLines, scale); } else { pSfbScale[window * 16 + band] = msb; } } /* for (band=0; band < ScaleFactorBandsTransmitted; band++) */ /* Make sure the array is cleared to the end */ SHORT start_clear = BandOffsets[ScaleFactorBandsTransmitted]; SHORT end_clear = BandOffsets[total_bands]; int diff_clear = (int)(end_clear - start_clear); FIXP_DBL *pSpectralCoefficient = SPEC(pAacDecoderChannelInfo->pSpectralCoefficient, window, pAacDecoderChannelInfo->granuleLength) + start_clear; FDKmemclear(pSpectralCoefficient, diff_clear * sizeof(FIXP_DBL)); } /* for (groupwin=0; groupwin < GetWindowGroupLength(&pAacDecoderChannelInfo->icsInfo,group); groupwin++, window++) */ } /* for (window=0, group=0; group < GetWindowGroups(&pAacDecoderChannelInfo->icsInfo); group++)*/ return AAC_DEC_OK; } AAC_DECODER_ERROR CBlock_ReadSpectralData( HANDLE_FDK_BITSTREAM bs, CAacDecoderChannelInfo *pAacDecoderChannelInfo, const SamplingRateInfo *pSamplingRateInfo, const UINT flags) { int index, i; const SHORT *RESTRICT BandOffsets = GetScaleFactorBandOffsets( &pAacDecoderChannelInfo->icsInfo, pSamplingRateInfo); SPECTRAL_PTR pSpectralCoefficient = pAacDecoderChannelInfo->pSpectralCoefficient; FDK_ASSERT(BandOffsets != NULL); FDKmemclear(pSpectralCoefficient, sizeof(SPECTRUM)); if ((flags & AC_ER_HCR) == 0) { int group; int groupoffset; UCHAR *pCodeBook = pAacDecoderChannelInfo->pDynData->aCodeBook; int ScaleFactorBandsTransmitted = GetScaleFactorBandsTransmitted(&pAacDecoderChannelInfo->icsInfo); int granuleLength = pAacDecoderChannelInfo->granuleLength; groupoffset = 0; /* plain huffman decoder short */ int max_group = GetWindowGroups(&pAacDecoderChannelInfo->icsInfo); for (group = 0; group < max_group; group++) { int max_groupwin = GetWindowGroupLength(&pAacDecoderChannelInfo->icsInfo, group); int band; int bnds = group * 16; int bandOffset1 = BandOffsets[0]; for (band = 0; band < ScaleFactorBandsTransmitted; band++, bnds++) { UCHAR currentCB = pCodeBook[bnds]; int bandOffset0 = bandOffset1; bandOffset1 = BandOffsets[band + 1]; /* patch to run plain-huffman-decoder with vcb11 input codebooks * (LAV-checking might be possible below using the virtual cb and a * LAV-table) */ if ((currentCB >= 16) && (currentCB <= 31)) { pCodeBook[bnds] = currentCB = 11; } if (((currentCB != ZERO_HCB) && (currentCB != NOISE_HCB) && (currentCB != INTENSITY_HCB) && (currentCB != INTENSITY_HCB2))) { const CodeBookDescription *hcb = &AACcodeBookDescriptionTable[currentCB]; int step = hcb->Dimension; int offset = hcb->Offset; int bits = hcb->numBits; int mask = (1 << bits) - 1; const USHORT(*CodeBook)[HuffmanEntries] = hcb->CodeBook; int groupwin; FIXP_DBL *mdctSpectrum = &pSpectralCoefficient[groupoffset * granuleLength]; if (offset == 0) { for (groupwin = 0; groupwin < max_groupwin; groupwin++) { for (index = bandOffset0; index < bandOffset1; index += step) { int idx = CBlock_DecodeHuffmanWordCB(bs, CodeBook); for (i = 0; i < step; i++, idx >>= bits) { FIXP_DBL tmp = (FIXP_DBL)((idx & mask) - offset); if (tmp != FIXP_DBL(0)) tmp = (FDKreadBit(bs)) ? -tmp : tmp; mdctSpectrum[index + i] = tmp; } if (currentCB == ESCBOOK) { for (int j = 0; j < 2; j++) mdctSpectrum[index + j] = (FIXP_DBL)CBlock_GetEscape( bs, (LONG)mdctSpectrum[index + j]); } } mdctSpectrum += granuleLength; } } else { for (groupwin = 0; groupwin < max_groupwin; groupwin++) { for (index = bandOffset0; index < bandOffset1; index += step) { int idx = CBlock_DecodeHuffmanWordCB(bs, CodeBook); for (i = 0; i < step; i++, idx >>= bits) { mdctSpectrum[index + i] = (FIXP_DBL)((idx & mask) - offset); } if (currentCB == ESCBOOK) { for (int j = 0; j < 2; j++) mdctSpectrum[index + j] = (FIXP_DBL)CBlock_GetEscape( bs, (LONG)mdctSpectrum[index + j]); } } mdctSpectrum += granuleLength; } } } } groupoffset += max_groupwin; } /* plain huffman decoding (short) finished */ } /* HCR - Huffman Codeword Reordering short */ else /* if ( flags & AC_ER_HCR ) */ { H_HCR_INFO hHcr = &pAacDecoderChannelInfo->pComData->overlay.aac.erHcrInfo; int hcrStatus = 0; /* advanced Huffman decoding starts here (HCR decoding :) */ if (pAacDecoderChannelInfo->pDynData->specificTo.aac .lenOfReorderedSpectralData != 0) { /* HCR initialization short */ hcrStatus = HcrInit(hHcr, pAacDecoderChannelInfo, pSamplingRateInfo, bs); if (hcrStatus != 0) { return AAC_DEC_DECODE_FRAME_ERROR; } /* HCR decoding short */ hcrStatus = HcrDecoder(hHcr, pAacDecoderChannelInfo, pSamplingRateInfo, bs); if (hcrStatus != 0) { #if HCR_ERROR_CONCEALMENT HcrMuteErroneousLines(hHcr); #else return AAC_DEC_DECODE_FRAME_ERROR; #endif /* HCR_ERROR_CONCEALMENT */ } FDKpushFor(bs, pAacDecoderChannelInfo->pDynData->specificTo.aac .lenOfReorderedSpectralData); } } /* HCR - Huffman Codeword Reordering short finished */ if (IsLongBlock(&pAacDecoderChannelInfo->icsInfo) && !(flags & (AC_ELD | AC_SCALABLE))) { /* apply pulse data */ CPulseData_Apply( &pAacDecoderChannelInfo->pDynData->specificTo.aac.PulseData, GetScaleFactorBandOffsets(&pAacDecoderChannelInfo->icsInfo, pSamplingRateInfo), SPEC_LONG(pSpectralCoefficient)); } return AAC_DEC_OK; } static const FIXP_SGL noise_level_tab[8] = { /* FDKpow(2, (float)(noise_level-14)/3.0f) * 2; (*2 to compensate for fMultDiv2) noise_level_tab(noise_level==0) == 0 by definition */ FX_DBL2FXCONST_SGL(0x00000000 /*0x0a145173*/), FX_DBL2FXCONST_SGL(0x0cb2ff5e), FX_DBL2FXCONST_SGL(0x10000000), FX_DBL2FXCONST_SGL(0x1428a2e7), FX_DBL2FXCONST_SGL(0x1965febd), FX_DBL2FXCONST_SGL(0x20000000), FX_DBL2FXCONST_SGL(0x28514606), FX_DBL2FXCONST_SGL(0x32cbfd33)}; void CBlock_ApplyNoise(CAacDecoderChannelInfo *pAacDecoderChannelInfo, SamplingRateInfo *pSamplingRateInfo, ULONG *nfRandomSeed, UCHAR *band_is_noise) { const SHORT *swb_offset = GetScaleFactorBandOffsets( &pAacDecoderChannelInfo->icsInfo, pSamplingRateInfo); int g, win, gwin, sfb, noiseFillingStartOffset, nfStartOffset_sfb; /* Obtain noise level and scale factor offset. */ int noise_level = pAacDecoderChannelInfo->pDynData->specificTo.usac .fd_noise_level_and_offset >> 5; const FIXP_SGL noiseVal_pos = noise_level_tab[noise_level]; /* noise_offset can change even when noise_level=0. Neccesary for IGF stereo * filling */ const int noise_offset = (pAacDecoderChannelInfo->pDynData->specificTo.usac .fd_noise_level_and_offset & 0x1f) - 16; int max_sfb = GetScaleFactorBandsTransmitted(&pAacDecoderChannelInfo->icsInfo); noiseFillingStartOffset = (GetWindowSequence(&pAacDecoderChannelInfo->icsInfo) == BLOCK_SHORT) ? 20 : 160; if (pAacDecoderChannelInfo->granuleLength == 96) { noiseFillingStartOffset = (3 * noiseFillingStartOffset) / 4; /* scale offset with 3/4 for coreCoderFrameLength == 768 */ } /* determine sfb from where on noise filling is applied */ for (sfb = 0; swb_offset[sfb] < noiseFillingStartOffset; sfb++) ; nfStartOffset_sfb = sfb; /* if (noise_level!=0) */ { for (g = 0, win = 0; g < GetWindowGroups(&pAacDecoderChannelInfo->icsInfo); g++) { int windowGroupLength = GetWindowGroupLength(&pAacDecoderChannelInfo->icsInfo, g); for (sfb = nfStartOffset_sfb; sfb < max_sfb; sfb++) { int bin_start = swb_offset[sfb]; int bin_stop = swb_offset[sfb + 1]; int flagN = band_is_noise[g * 16 + sfb]; /* if all bins of one sfb in one window group are zero modify the scale * factor by noise_offset */ if (flagN) { /* Change scaling factors for empty signal bands */ pAacDecoderChannelInfo->pDynData->aScaleFactor[g * 16 + sfb] += noise_offset; /* scale factor "sf" implied gain "g" is g = 2^(sf/4) */ for (gwin = 0; gwin < windowGroupLength; gwin++) { pAacDecoderChannelInfo->pDynData ->aSfbScale[(win + gwin) * 16 + sfb] += (noise_offset >> 2); } } ULONG seed = *nfRandomSeed; /* + 1 because exponent of MantissaTable[lsb][0] is always 1. */ int scale = (pAacDecoderChannelInfo->pDynData->aScaleFactor[g * 16 + sfb] >> 2) + 1; int lsb = pAacDecoderChannelInfo->pDynData->aScaleFactor[g * 16 + sfb] & 3; FIXP_DBL mantissa = MantissaTable[lsb][0]; for (gwin = 0; gwin < windowGroupLength; gwin++) { FIXP_DBL *pSpec = SPEC(pAacDecoderChannelInfo->pSpectralCoefficient, win + gwin, pAacDecoderChannelInfo->granuleLength); int scale1 = scale - pAacDecoderChannelInfo->pDynData ->aSfbScale[(win + gwin) * 16 + sfb]; FIXP_DBL scaled_noiseVal_pos = scaleValue(fMultDiv2(noiseVal_pos, mantissa), scale1); FIXP_DBL scaled_noiseVal_neg = -scaled_noiseVal_pos; /* If the whole band is zero, just fill without checking */ if (flagN) { for (int bin = bin_start; bin < bin_stop; bin++) { seed = (ULONG)( (UINT64)seed * 69069 + 5); /* Inlined: UsacRandomSign - origin in usacdec_lpd.h */ pSpec[bin] = (seed & 0x10000) ? scaled_noiseVal_neg : scaled_noiseVal_pos; } /* for (bin...) */ } /*If band is sparsely filled, check for 0 and fill */ else { for (int bin = bin_start; bin < bin_stop; bin++) { if (pSpec[bin] == (FIXP_DBL)0) { seed = (ULONG)( (UINT64)seed * 69069 + 5); /* Inlined: UsacRandomSign - origin in usacdec_lpd.h */ pSpec[bin] = (seed & 0x10000) ? scaled_noiseVal_neg : scaled_noiseVal_pos; } } /* for (bin...) */ } } /* for (gwin...) */ *nfRandomSeed = seed; } /* for (sfb...) */ win += windowGroupLength; } /* for (g...) */ } /* ... */ } AAC_DECODER_ERROR CBlock_ReadAcSpectralData( HANDLE_FDK_BITSTREAM hBs, CAacDecoderChannelInfo *pAacDecoderChannelInfo, CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, const SamplingRateInfo *pSamplingRateInfo, const UINT frame_length, const UINT flags) { AAC_DECODER_ERROR errorAAC = AAC_DEC_OK; ARITH_CODING_ERROR error = ARITH_CODER_OK; int arith_reset_flag, lg, numWin, win, winLen; const SHORT *RESTRICT BandOffsets; /* number of transmitted spectral coefficients */ BandOffsets = GetScaleFactorBandOffsets(&pAacDecoderChannelInfo->icsInfo, pSamplingRateInfo); lg = BandOffsets[GetScaleFactorBandsTransmitted( &pAacDecoderChannelInfo->icsInfo)]; numWin = GetWindowsPerFrame(&pAacDecoderChannelInfo->icsInfo); winLen = (IsLongBlock(&pAacDecoderChannelInfo->icsInfo)) ? (int)frame_length : (int)frame_length / numWin; if (flags & AC_INDEP) { arith_reset_flag = 1; } else { arith_reset_flag = (USHORT)FDKreadBits(hBs, 1); } for (win = 0; win < numWin; win++) { error = CArco_DecodeArithData(pAacDecoderStaticChannelInfo->hArCo, hBs, SPEC(pAacDecoderChannelInfo->pSpectralCoefficient, win, pAacDecoderChannelInfo->granuleLength), lg, winLen, arith_reset_flag && (win == 0)); if (error != ARITH_CODER_OK) { goto bail; } } bail: if (error == ARITH_CODER_ERROR) { errorAAC = AAC_DEC_PARSE_ERROR; } return errorAAC; } void ApplyTools(CAacDecoderChannelInfo *pAacDecoderChannelInfo[], const SamplingRateInfo *pSamplingRateInfo, const UINT flags, const UINT elFlags, const int channel, const int common_window) { if (!(flags & (AC_USAC | AC_RSVD50 | AC_MPEGD_RES | AC_RSV603DA))) { CPns_Apply(&pAacDecoderChannelInfo[channel]->data.aac.PnsData, &pAacDecoderChannelInfo[channel]->icsInfo, pAacDecoderChannelInfo[channel]->pSpectralCoefficient, pAacDecoderChannelInfo[channel]->specScale, pAacDecoderChannelInfo[channel]->pDynData->aScaleFactor, pSamplingRateInfo, pAacDecoderChannelInfo[channel]->granuleLength, channel); } UCHAR nbands = GetScaleFactorBandsTransmitted(&pAacDecoderChannelInfo[channel]->icsInfo); CTns_Apply(&pAacDecoderChannelInfo[channel]->pDynData->TnsData, &pAacDecoderChannelInfo[channel]->icsInfo, pAacDecoderChannelInfo[channel]->pSpectralCoefficient, pSamplingRateInfo, pAacDecoderChannelInfo[channel]->granuleLength, nbands, (elFlags & AC_EL_ENHANCED_NOISE) ? 1 : 0, flags); } static int getWindow2Nr(int length, int shape) { int nr = 0; if (shape == 2) { /* Low Overlap, 3/4 zeroed */ nr = (length * 3) >> 2; } return nr; } FIXP_DBL get_gain(const FIXP_DBL *x, const FIXP_DBL *y, int n) { FIXP_DBL corr = (FIXP_DBL)0; FIXP_DBL ener = (FIXP_DBL)1; int headroom_x = getScalefactor(x, n); int headroom_y = getScalefactor(y, n); /*Calculate the normalization necessary due to addition*/ /* Check for power of two /special case */ INT width_shift = (INT)(fNormz((FIXP_DBL)n)); /* Get the number of bits necessary minus one, because we need one sign bit * only */ width_shift = 31 - width_shift; for (int i = 0; i < n; i++) { corr += fMultDiv2((x[i] << headroom_x), (y[i] << headroom_y)) >> width_shift; ener += fPow2Div2((y[i] << headroom_y)) >> width_shift; } int exp_corr = (17 - headroom_x) + (17 - headroom_y) + width_shift + 1; int exp_ener = ((17 - headroom_y) << 1) + width_shift + 1; int temp_exp = 0; FIXP_DBL output = fDivNormSigned(corr, ener, &temp_exp); int output_exp = (exp_corr - exp_ener) + temp_exp; INT output_shift = 17 - output_exp; output_shift = fMin(output_shift, 31); output = scaleValue(output, -output_shift); return output; } void CBlock_FrequencyToTime( CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, CAacDecoderChannelInfo *pAacDecoderChannelInfo, PCM_DEC outSamples[], const SHORT frameLen, const int frameOk, FIXP_DBL *pWorkBuffer1, const INT aacOutDataHeadroom, UINT elFlags, INT elCh) { int fr, fl, tl, nSpec; #if defined(FDK_ASSERT_ENABLE) LONG nSamples; #endif /* Determine left slope length (fl), right slope length (fr) and transform length (tl). USAC: The slope length may mismatch with the previous frame in case of LPD / FD transitions. The adjustment is handled by the imdct implementation. */ tl = frameLen; nSpec = 1; switch (pAacDecoderChannelInfo->icsInfo.WindowSequence) { default: case BLOCK_LONG: fl = frameLen; fr = frameLen - getWindow2Nr(frameLen, GetWindowShape(&pAacDecoderChannelInfo->icsInfo)); /* New startup needs differentiation between sine shape and low overlap shape. This is a special case for the LD-AAC transformation windows, because the slope length can be different while using the same window sequence. */ if (pAacDecoderStaticChannelInfo->IMdct.prev_tl == 0) { fl = fr; } break; case BLOCK_STOP: fl = frameLen >> 3; fr = frameLen; break; case BLOCK_START: /* or StopStartSequence */ fl = frameLen; fr = frameLen >> 3; break; case BLOCK_SHORT: fl = fr = frameLen >> 3; tl >>= 3; nSpec = 8; break; } { int last_frame_lost = pAacDecoderStaticChannelInfo->last_lpc_lost; if (pAacDecoderStaticChannelInfo->last_core_mode == LPD) { INT fac_FB = 1; if (elFlags & AC_EL_FULLBANDLPD) { fac_FB = 2; } FIXP_DBL *synth; /* Keep some free space at the beginning of the buffer. To be used for * past data */ if (!(elFlags & AC_EL_LPDSTEREOIDX)) { synth = pWorkBuffer1 + ((PIT_MAX_MAX - (1 * L_SUBFR)) * fac_FB); } else { synth = pWorkBuffer1 + PIT_MAX_MAX * fac_FB; } int fac_length = (pAacDecoderChannelInfo->icsInfo.WindowSequence == BLOCK_SHORT) ? (frameLen >> 4) : (frameLen >> 3); INT pitch[NB_SUBFR_SUPERFR + SYN_SFD]; FIXP_DBL pit_gain[NB_SUBFR_SUPERFR + SYN_SFD]; int nbDiv = (elFlags & AC_EL_FULLBANDLPD) ? 2 : 4; int lFrame = (elFlags & AC_EL_FULLBANDLPD) ? frameLen / 2 : frameLen; int nbSubfr = lFrame / (nbDiv * L_SUBFR); /* number of subframes per division */ int LpdSfd = (nbDiv * nbSubfr) >> 1; int SynSfd = LpdSfd - BPF_SFD; FDKmemclear( pitch, sizeof( pitch)); // added to prevent ferret errors in bass_pf_1sf_delay FDKmemclear(pit_gain, sizeof(pit_gain)); /* FAC case */ if (pAacDecoderStaticChannelInfo->last_lpd_mode == 0 || pAacDecoderStaticChannelInfo->last_lpd_mode == 4) { FIXP_DBL fac_buf[LFAC]; FIXP_LPC *A = pAacDecoderChannelInfo->data.usac.lp_coeff[0]; if (!frameOk || last_frame_lost || (pAacDecoderChannelInfo->data.usac.fac_data[0] == NULL)) { FDKmemclear(fac_buf, pAacDecoderChannelInfo->granuleLength * sizeof(FIXP_DBL)); pAacDecoderChannelInfo->data.usac.fac_data[0] = fac_buf; pAacDecoderChannelInfo->data.usac.fac_data_e[0] = 0; } INT A_exp; /* linear prediction coefficients exponent */ { for (int i = 0; i < M_LP_FILTER_ORDER; i++) { A[i] = FX_DBL2FX_LPC(fixp_cos( fMult(pAacDecoderStaticChannelInfo->lpc4_lsf[i], FL2FXCONST_SGL((1 << LSPARG_SCALE) * M_PI / 6400.0)), LSF_SCALE - LSPARG_SCALE)); } E_LPC_f_lsp_a_conversion(A, A, &A_exp); } #if defined(FDK_ASSERT_ENABLE) nSamples = #endif CLpd_FAC_Acelp2Mdct( &pAacDecoderStaticChannelInfo->IMdct, synth, SPEC_LONG(pAacDecoderChannelInfo->pSpectralCoefficient), pAacDecoderChannelInfo->specScale, nSpec, pAacDecoderChannelInfo->data.usac.fac_data[0], pAacDecoderChannelInfo->data.usac.fac_data_e[0], fac_length, frameLen, tl, FDKgetWindowSlope( fr, GetWindowShape(&pAacDecoderChannelInfo->icsInfo)), fr, A, A_exp, &pAacDecoderStaticChannelInfo->acelp, (FIXP_DBL)0, /* FAC gain has already been applied. */ (last_frame_lost || !frameOk), 1, pAacDecoderStaticChannelInfo->last_lpd_mode, 0, pAacDecoderChannelInfo->currAliasingSymmetry); } else { #if defined(FDK_ASSERT_ENABLE) nSamples = #endif imlt_block( &pAacDecoderStaticChannelInfo->IMdct, synth, SPEC_LONG(pAacDecoderChannelInfo->pSpectralCoefficient), pAacDecoderChannelInfo->specScale, nSpec, frameLen, tl, FDKgetWindowSlope( fl, GetWindowShape(&pAacDecoderChannelInfo->icsInfo)), fl, FDKgetWindowSlope( fr, GetWindowShape(&pAacDecoderChannelInfo->icsInfo)), fr, (FIXP_DBL)0, pAacDecoderChannelInfo->currAliasingSymmetry ? MLT_FLAG_CURR_ALIAS_SYMMETRY : 0); } FDK_ASSERT(nSamples == frameLen); /* The "if" clause is entered both for fullbandLpd mono and * non-fullbandLpd*. The "else"-> just for fullbandLpd stereo*/ if (!(elFlags & AC_EL_LPDSTEREOIDX)) { FDKmemcpy(pitch, pAacDecoderStaticChannelInfo->old_T_pf, SynSfd * sizeof(INT)); FDKmemcpy(pit_gain, pAacDecoderStaticChannelInfo->old_gain_pf, SynSfd * sizeof(FIXP_DBL)); for (int i = SynSfd; i < LpdSfd + 3; i++) { pitch[i] = L_SUBFR; pit_gain[i] = (FIXP_DBL)0; } if (pAacDecoderStaticChannelInfo->last_lpd_mode == 0) { pitch[SynSfd] = pitch[SynSfd - 1]; pit_gain[SynSfd] = pit_gain[SynSfd - 1]; if (IsLongBlock(&pAacDecoderChannelInfo->icsInfo)) { pitch[SynSfd + 1] = pitch[SynSfd]; pit_gain[SynSfd + 1] = pit_gain[SynSfd]; } } /* Copy old data to the beginning of the buffer */ { FDKmemcpy( pWorkBuffer1, pAacDecoderStaticChannelInfo->old_synth, ((PIT_MAX_MAX - (1 * L_SUBFR)) * fac_FB) * sizeof(FIXP_DBL)); } FIXP_DBL *p2_synth = pWorkBuffer1 + (PIT_MAX_MAX * fac_FB); /* recalculate pitch gain to allow postfilering on FAC area */ for (int i = 0; i < SynSfd + 2; i++) { int T = pitch[i]; FIXP_DBL gain = pit_gain[i]; if (gain > (FIXP_DBL)0) { gain = get_gain(&p2_synth[i * L_SUBFR * fac_FB], &p2_synth[(i * L_SUBFR * fac_FB) - fac_FB * T], L_SUBFR * fac_FB); pit_gain[i] = gain; } } bass_pf_1sf_delay(p2_synth, pitch, pit_gain, frameLen, (LpdSfd + 2) * L_SUBFR + BPF_SFD * L_SUBFR, frameLen - (LpdSfd + 4) * L_SUBFR, outSamples, aacOutDataHeadroom, pAacDecoderStaticChannelInfo->mem_bpf); } } else /* last_core_mode was not LPD */ { FIXP_DBL *tmp = pAacDecoderChannelInfo->pComStaticData->pWorkBufferCore1->mdctOutTemp; #if defined(FDK_ASSERT_ENABLE) nSamples = #endif imlt_block(&pAacDecoderStaticChannelInfo->IMdct, tmp, SPEC_LONG(pAacDecoderChannelInfo->pSpectralCoefficient), pAacDecoderChannelInfo->specScale, nSpec, frameLen, tl, FDKgetWindowSlope( fl, GetWindowShape(&pAacDecoderChannelInfo->icsInfo)), fl, FDKgetWindowSlope( fr, GetWindowShape(&pAacDecoderChannelInfo->icsInfo)), fr, (FIXP_DBL)0, pAacDecoderChannelInfo->currAliasingSymmetry ? MLT_FLAG_CURR_ALIAS_SYMMETRY : 0); scaleValuesSaturate(outSamples, tmp, frameLen, MDCT_OUT_HEADROOM - aacOutDataHeadroom); } } FDK_ASSERT(nSamples == frameLen); pAacDecoderStaticChannelInfo->last_core_mode = (pAacDecoderChannelInfo->icsInfo.WindowSequence == BLOCK_SHORT) ? FD_SHORT : FD_LONG; pAacDecoderStaticChannelInfo->last_lpd_mode = 255; } #include "ldfiltbank.h" void CBlock_FrequencyToTimeLowDelay( CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, CAacDecoderChannelInfo *pAacDecoderChannelInfo, PCM_DEC outSamples[], const short frameLen) { InvMdctTransformLowDelay_fdk( SPEC_LONG(pAacDecoderChannelInfo->pSpectralCoefficient), pAacDecoderChannelInfo->specScale[0], outSamples, pAacDecoderStaticChannelInfo->pOverlapBuffer, frameLen); } fdk-aac-2.0.2/libAACdec/src/block.h000066400000000000000000000313541404225310700165660ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Josef Hoepfl Description: long/short-block decoding *******************************************************************************/ #ifndef BLOCK_H #define BLOCK_H #include "common_fix.h" #include "channelinfo.h" #include "FDK_bitstream.h" /* PNS (of block) */ void CPns_Read(CPnsData *pPnsData, HANDLE_FDK_BITSTREAM bs, const CodeBookDescription *hcb, SHORT *pScaleFactor, UCHAR global_gain, int band, int group); void CPns_Apply(const CPnsData *pPnsData, const CIcsInfo *pIcsInfo, SPECTRAL_PTR pSpectrum, const SHORT *pSpecScale, const SHORT *pScaleFactor, const SamplingRateInfo *pSamplingRateInfo, const INT granuleLength, const int channel); void CBlock_ApplyNoise(CAacDecoderChannelInfo *pAacDecoderChannelInfo, SamplingRateInfo *pSamplingRateInfo, ULONG *nfRandomSeed, UCHAR *band_is_noise); /* TNS (of block) */ /*! \brief Read tns data-present flag from bitstream The function reads the data-present flag for tns from the bitstream. \return none */ void CTns_ReadDataPresentFlag(HANDLE_FDK_BITSTREAM bs, CTnsData *pTnsData); void CTns_ReadDataPresentUsac(HANDLE_FDK_BITSTREAM hBs, CTnsData *pTnsData0, CTnsData *pTnsData1, UCHAR *ptns_on_lr, const CIcsInfo *pIcsInfo, const UINT flags, const UINT elFlags, const int fCommonWindow); AAC_DECODER_ERROR CTns_Read(HANDLE_FDK_BITSTREAM bs, CTnsData *pTnsData, const CIcsInfo *pIcsInfo, const UINT flags); void CTns_Apply(CTnsData *RESTRICT pTnsData, /*!< pointer to aac decoder info */ const CIcsInfo *pIcsInfo, SPECTRAL_PTR pSpectralCoefficient, const SamplingRateInfo *pSamplingRateInfo, const INT granuleLength, const UCHAR nbands, const UCHAR igf_active, const UINT flags); /* Block */ LONG CBlock_GetEscape(HANDLE_FDK_BITSTREAM bs, const LONG q); /** * \brief Read scale factor data. See chapter 4.6.2.3.2 of ISO/IEC 14496-3. * The SF_OFFSET = 100 value referenced in chapter 4.6.2.3.3 is already * substracted from the scale factor values. Also includes PNS data reading. * \param bs bit stream handle data source * \param pAacDecoderChannelInfo channel context info were decoded data is * stored into. * \param flags the decoder flags. */ AAC_DECODER_ERROR CBlock_ReadScaleFactorData( CAacDecoderChannelInfo *pAacDecoderChannelInfo, HANDLE_FDK_BITSTREAM bs, const UINT flags); /** * \brief Read Huffman encoded spectral data. * \param pAacDecoderChannelInfo channel context info. * \param pSamplingRateInfo sampling rate info (sfb offsets). * \param flags syntax flags. */ AAC_DECODER_ERROR CBlock_ReadSpectralData( HANDLE_FDK_BITSTREAM bs, CAacDecoderChannelInfo *pAacDecoderChannelInfo, const SamplingRateInfo *pSamplingRateInfo, const UINT flags); /** * \brief Read Arithmetic encoded spectral data. * \param pAacDecoderChannelInfo channel context info. * \param pAacDecoderStaticChannelInfo static channel context info. * \param pSamplingRateInfo sampling rate info (sfb offsets). * \param frame_length spectral window length. * \param flags syntax flags. * \return error code. */ AAC_DECODER_ERROR CBlock_ReadAcSpectralData( HANDLE_FDK_BITSTREAM hBs, CAacDecoderChannelInfo *pAacDecoderChannelInfo, CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, const SamplingRateInfo *pSamplingRateInfo, const UINT frame_length, const UINT flags); AAC_DECODER_ERROR CBlock_ReadSectionData( HANDLE_FDK_BITSTREAM bs, CAacDecoderChannelInfo *pAacDecoderChannelInfo, const SamplingRateInfo *pSamplingRateInfo, const UINT flags); /** * \brief find a common exponent (shift factor) for all sfb in each Spectral * window, and store them into CAacDecoderChannelInfo::specScale. * \param pAacDecoderChannelInfo channel context info. * \param UCHAR maxSfbs maximum number of SFBs to be processed (might differ * from pAacDecoderChannelInfo->icsInfo.MaxSfBands) * \param pSamplingRateInfo sampling rate info (sfb offsets). */ void CBlock_ScaleSpectralData(CAacDecoderChannelInfo *pAacDecoderChannelInfo, UCHAR maxSfbs, SamplingRateInfo *pSamplingRateInfo); /** * \brief Apply TNS and PNS tools. */ void ApplyTools(CAacDecoderChannelInfo *pAacDecoderChannelInfo[], const SamplingRateInfo *pSamplingRateInfo, const UINT flags, const UINT elFlags, const int channel, const int maybe_jstereo); /** * \brief Transform MDCT spectral data into time domain */ void CBlock_FrequencyToTime( CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, CAacDecoderChannelInfo *pAacDecoderChannelInfo, PCM_DEC outSamples[], const SHORT frameLen, const int frameOk, FIXP_DBL *pWorkBuffer1, const INT aacOutDataHeadroom, UINT elFlags, INT elCh); /** * \brief Transform double lapped MDCT (AAC-ELD) spectral data into time domain. */ void CBlock_FrequencyToTimeLowDelay( CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, CAacDecoderChannelInfo *pAacDecoderChannelInfo, PCM_DEC outSamples[], const short frameLen); AAC_DECODER_ERROR CBlock_InverseQuantizeSpectralData( CAacDecoderChannelInfo *pAacDecoderChannelInfo, SamplingRateInfo *pSamplingRateInfo, UCHAR *band_is_noise, UCHAR active_band_search); /** * \brief Calculate 2^(lsb/4) * value^(4/3) * \param pValue pointer to quantized value. The inverse quantized result is * stored back here. * \param lsb 2 LSBs of the scale factor (scaleFactor % 4) applied as power 2 * factor to the resulting inverse quantized value. * \return the exponent of the result (mantissa) stored into *pValue. */ FDK_INLINE int EvaluatePower43(FIXP_DBL *pValue, UINT lsb) { FIXP_DBL value; UINT freeBits; UINT exponent; value = *pValue; freeBits = fNormz(value); exponent = DFRACT_BITS - freeBits; FDK_ASSERT(exponent < 14); UINT x = (((int)value << freeBits) >> 19); UINT tableIndex = (x & 0x0FFF) >> 4; FIXP_DBL invQVal; x = x & 0x0F; UINT r0 = (LONG)InverseQuantTable[tableIndex + 0]; UINT r1 = (LONG)InverseQuantTable[tableIndex + 1]; USHORT nx = 16 - x; UINT temp = (r0)*nx + (r1)*x; invQVal = (FIXP_DBL)temp; FDK_ASSERT(lsb < 4); *pValue = fMultDiv2(invQVal, MantissaTable[lsb][exponent]); /* + 1 compensates fMultDiv2(). */ return ExponentTable[lsb][exponent] + 1; } /* Recalculate gain */ FIXP_DBL get_gain(const FIXP_DBL *x, const FIXP_DBL *y, int n); /** * \brief determine the required shift scale for the given quantized value and * scale (factor % 4) value. */ FDK_INLINE int GetScaleFromValue(FIXP_DBL value, unsigned int lsb) { if (value != (FIXP_DBL)0) { int scale = EvaluatePower43(&value, lsb); return CntLeadingZeros(value) - scale - 2; } else return 0; /* Return zero, because its useless to scale a zero value, saves workload and avoids scaling overshifts. */ } /*! \brief Read huffman codeword The function reads the huffman codeword from the bitstream and returns the index value. \return index value */ inline int CBlock_DecodeHuffmanWord( HANDLE_FDK_BITSTREAM bs, /*!< pointer to bitstream */ const CodeBookDescription *hcb) /*!< pointer to codebook description */ { UINT val; UINT index = 0; const USHORT(*CodeBook)[HuffmanEntries] = hcb->CodeBook; while (1) { val = CodeBook[index] [FDKreadBits(bs, HuffmanBits)]; /* Expensive memory access */ if ((val & 1) == 0) { index = val >> 2; continue; } else { if (val & 2) { FDKpushBackCache(bs, 1); } val >>= 2; break; } } return val; } inline int CBlock_DecodeHuffmanWordCB( HANDLE_FDK_BITSTREAM bs, /*!< pointer to bitstream */ const USHORT ( *CodeBook)[HuffmanEntries]) /*!< pointer to codebook description */ { UINT index = 0; while (1) { index = CodeBook[index][FDKread2Bits(bs)]; /* Expensive memory access */ if (index & 1) break; index >>= 2; } if (index & 2) { FDKpushBackCache(bs, 1); } return index >> 2; } #endif /* #ifndef BLOCK_H */ fdk-aac-2.0.2/libAACdec/src/channel.cpp000066400000000000000000001046511404225310700174400ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Josef Hoepfl Description: *******************************************************************************/ #include "channel.h" #include "aacdecoder.h" #include "block.h" #include "aacdec_tns.h" #include "FDK_bitstream.h" #include "conceal.h" #include "rvlc.h" #include "aacdec_hcr.h" #include "usacdec_lpd.h" #include "usacdec_fac.h" static void MapMidSideMaskToPnsCorrelation( CAacDecoderChannelInfo *pAacDecoderChannelInfo[2]) { int group; for (group = 0; group < pAacDecoderChannelInfo[L]->icsInfo.WindowGroups; group++) { UCHAR groupMask = 1 << group; for (UCHAR band = 0; band < pAacDecoderChannelInfo[L]->icsInfo.MaxSfBands; band++) { if (pAacDecoderChannelInfo[L]->pComData->jointStereoData.MsUsed[band] & groupMask) { /* channels are correlated */ CPns_SetCorrelation(&pAacDecoderChannelInfo[L]->data.aac.PnsData, group, band, 0); if (CPns_IsPnsUsed(&pAacDecoderChannelInfo[L]->data.aac.PnsData, group, band) && CPns_IsPnsUsed(&pAacDecoderChannelInfo[R]->data.aac.PnsData, group, band)) pAacDecoderChannelInfo[L]->pComData->jointStereoData.MsUsed[band] ^= groupMask; /* clear the groupMask-bit */ } } } } static void Clean_Complex_Prediction_coefficients( CJointStereoPersistentData *pJointStereoPersistentData, int windowGroups, const int low_limit, const int high_limit) { for (int group = 0; group < windowGroups; group++) { for (int sfb = low_limit; sfb < high_limit; sfb++) { pJointStereoPersistentData->alpha_q_re_prev[group][sfb] = 0; pJointStereoPersistentData->alpha_q_im_prev[group][sfb] = 0; } } } /*! \brief Decode channel pair element The function decodes a channel pair element. \return none */ void CChannelElement_Decode( CAacDecoderChannelInfo *pAacDecoderChannelInfo[2], /*!< pointer to aac decoder channel info */ CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo[2], SamplingRateInfo *pSamplingRateInfo, UINT flags, UINT elFlags, int el_channels) { int ch = 0; int maxSfBandsL = 0, maxSfBandsR = 0; int maybe_jstereo = (el_channels > 1); if (flags & (AC_USAC | AC_RSVD50 | AC_RSV603DA) && el_channels == 2) { if (pAacDecoderChannelInfo[L]->data.usac.core_mode || pAacDecoderChannelInfo[R]->data.usac.core_mode) { maybe_jstereo = 0; } } if (maybe_jstereo) { maxSfBandsL = GetScaleFactorBandsTransmitted(&pAacDecoderChannelInfo[L]->icsInfo); maxSfBandsR = GetScaleFactorBandsTransmitted(&pAacDecoderChannelInfo[R]->icsInfo); /* apply ms */ if (pAacDecoderChannelInfo[L]->pDynData->RawDataInfo.CommonWindow) { if (!(flags & (AC_USAC | AC_RSVD50 | AC_RSV603DA))) { if (pAacDecoderChannelInfo[L]->data.aac.PnsData.PnsActive || pAacDecoderChannelInfo[R]->data.aac.PnsData.PnsActive) { MapMidSideMaskToPnsCorrelation(pAacDecoderChannelInfo); } } /* if tns_on_lr == 1 run MS */ /* && (pAacDecoderChannelInfo[L]->pDynData->specificTo.usac.tns_active == 1) */ if (((flags & (AC_USAC | AC_RSVD50 | AC_RSV603DA)) && (pAacDecoderChannelInfo[L]->pDynData->specificTo.usac.tns_on_lr == 1)) || ((flags & (AC_USAC | AC_RSVD50 | AC_RSV603DA)) == 0)) { int max_sfb_ste = (INT)(pAacDecoderChannelInfo[L]->icsInfo.max_sfb_ste); CJointStereo_ApplyMS( pAacDecoderChannelInfo, pAacDecoderStaticChannelInfo, pAacDecoderChannelInfo[L]->pSpectralCoefficient, pAacDecoderChannelInfo[R]->pSpectralCoefficient, pAacDecoderChannelInfo[L]->pDynData->aSfbScale, pAacDecoderChannelInfo[R]->pDynData->aSfbScale, pAacDecoderChannelInfo[L]->specScale, pAacDecoderChannelInfo[R]->specScale, GetScaleFactorBandOffsets(&pAacDecoderChannelInfo[L]->icsInfo, pSamplingRateInfo), GetWindowGroupLengthTable(&pAacDecoderChannelInfo[L]->icsInfo), GetWindowGroups(&pAacDecoderChannelInfo[L]->icsInfo), max_sfb_ste, maxSfBandsL, maxSfBandsR, pAacDecoderChannelInfo[L] ->pComData->jointStereoData.store_dmx_re_prev, &(pAacDecoderChannelInfo[L] ->pComData->jointStereoData.store_dmx_re_prev_e), 1); } /* if ( ((elFlags & AC_EL_USAC_CP_POSSIBLE).... */ } /* if (pAacDecoderChannelInfo[L]->pDynData->RawDataInfo.CommonWindow)*/ /* apply intensity stereo */ /* modifies pAacDecoderChannelInfo[]->aSpecSfb */ if (!(flags & (AC_USAC | AC_RSVD50 | AC_RSV603DA))) { if ((pAacDecoderChannelInfo[L]->pDynData->RawDataInfo.CommonWindow == 1) && (el_channels == 2)) { CJointStereo_ApplyIS( pAacDecoderChannelInfo, GetScaleFactorBandOffsets(&pAacDecoderChannelInfo[L]->icsInfo, pSamplingRateInfo), GetWindowGroupLengthTable(&pAacDecoderChannelInfo[L]->icsInfo), GetWindowGroups(&pAacDecoderChannelInfo[L]->icsInfo), GetScaleFactorBandsTransmitted( &pAacDecoderChannelInfo[L]->icsInfo)); } } } /* maybe_stereo */ for (ch = 0; ch < el_channels; ch++) { if (pAacDecoderChannelInfo[ch]->renderMode == AACDEC_RENDER_LPD) { /* Decode LPD data */ CLpdChannelStream_Decode(pAacDecoderChannelInfo[ch], pAacDecoderStaticChannelInfo[ch], flags); } else { UCHAR noSfbs = GetScaleFactorBandsTransmitted(&pAacDecoderChannelInfo[ch]->icsInfo); /* For USAC common window: max_sfb of both channels may differ * (common_max_sfb == 0). */ if ((maybe_jstereo == 1) && (pAacDecoderChannelInfo[L]->pDynData->RawDataInfo.CommonWindow == 1)) { noSfbs = fMax(maxSfBandsL, maxSfBandsR); } int CP_active = 0; if (elFlags & AC_EL_USAC_CP_POSSIBLE) { CP_active = pAacDecoderChannelInfo[ch] ->pComData->jointStereoData.cplx_pred_flag; } /* Omit writing of pAacDecoderChannelInfo[ch]->specScale for complex stereo prediction since scaling has already been carried out. */ int max_sfb_ste = (INT)(pAacDecoderChannelInfo[L]->icsInfo.max_sfb_ste); if ((!CP_active) || (CP_active && (max_sfb_ste < noSfbs)) || ((flags & (AC_USAC | AC_RSVD50 | AC_RSV603DA)) && (pAacDecoderChannelInfo[L]->pDynData->specificTo.usac.tns_on_lr == 0))) { CBlock_ScaleSpectralData(pAacDecoderChannelInfo[ch], noSfbs, pSamplingRateInfo); /*Active for the case of TNS applied before MS/CP*/ if ((flags & (AC_USAC | AC_RSVD50 | AC_RSV603DA)) && (pAacDecoderChannelInfo[L]->pDynData->specificTo.usac.tns_on_lr == 0)) { if (IsLongBlock(&pAacDecoderChannelInfo[ch]->icsInfo)) { for (int i = 0; i < noSfbs; i++) { pAacDecoderChannelInfo[ch]->pDynData->aSfbScale[i] = pAacDecoderChannelInfo[ch]->specScale[0]; } } else { for (int i = 0; i < 8; i++) { for (int j = 0; j < noSfbs; j++) { pAacDecoderChannelInfo[ch]->pDynData->aSfbScale[i * 16 + j] = pAacDecoderChannelInfo[ch]->specScale[i]; } } } } } } } /* End "for (ch = 0; ch < el_channels; ch++)" */ if (maybe_jstereo) { /* apply ms */ if (pAacDecoderChannelInfo[L]->pDynData->RawDataInfo.CommonWindow) { } /* CommonWindow */ else { if (elFlags & AC_EL_USAC_CP_POSSIBLE) { FDKmemclear( pAacDecoderStaticChannelInfo[L] ->pCpeStaticData->jointStereoPersistentData.alpha_q_re_prev, JointStereoMaximumGroups * JointStereoMaximumBands * sizeof(SHORT)); FDKmemclear( pAacDecoderStaticChannelInfo[L] ->pCpeStaticData->jointStereoPersistentData.alpha_q_im_prev, JointStereoMaximumGroups * JointStereoMaximumBands * sizeof(SHORT)); } } } /* if (maybe_jstereo) */ for (ch = 0; ch < el_channels; ch++) { if (pAacDecoderChannelInfo[ch]->renderMode == AACDEC_RENDER_LPD) { } else { if (!(flags & (AC_USAC | AC_RSVD50 | AC_RSV603DA))) { /* Use same seed for coupled channels (CPE) */ int pnsCh = (ch > 0) ? L : ch; CPns_UpdateNoiseState( &pAacDecoderChannelInfo[ch]->data.aac.PnsData, pAacDecoderChannelInfo[pnsCh]->data.aac.PnsData.currentSeed, pAacDecoderChannelInfo[ch]->pComData->pnsRandomSeed); } if ((!(flags & (AC_USAC))) || ((flags & (AC_USAC)) && (pAacDecoderChannelInfo[L]->pDynData->specificTo.usac.tns_active == 1)) || (maybe_jstereo == 0)) { ApplyTools( pAacDecoderChannelInfo, pSamplingRateInfo, flags, elFlags, ch, pAacDecoderChannelInfo[L]->pDynData->RawDataInfo.CommonWindow); } } /* End "} else" */ } /* End "for (ch = 0; ch < el_channels; ch++)" */ if (maybe_jstereo) { /* apply ms */ if (pAacDecoderChannelInfo[L]->pDynData->RawDataInfo.CommonWindow) { /* if tns_on_lr == 0 run MS */ if ((flags & (AC_USAC | AC_RSVD50 | AC_RSV603DA)) && (pAacDecoderChannelInfo[L]->pDynData->specificTo.usac.tns_on_lr == 0)) { int max_sfb_ste = (INT)(pAacDecoderChannelInfo[L]->icsInfo.max_sfb_ste); CJointStereo_ApplyMS( pAacDecoderChannelInfo, pAacDecoderStaticChannelInfo, pAacDecoderChannelInfo[L]->pSpectralCoefficient, pAacDecoderChannelInfo[R]->pSpectralCoefficient, pAacDecoderChannelInfo[L]->pDynData->aSfbScale, pAacDecoderChannelInfo[R]->pDynData->aSfbScale, pAacDecoderChannelInfo[L]->specScale, pAacDecoderChannelInfo[R]->specScale, GetScaleFactorBandOffsets(&pAacDecoderChannelInfo[L]->icsInfo, pSamplingRateInfo), GetWindowGroupLengthTable(&pAacDecoderChannelInfo[L]->icsInfo), GetWindowGroups(&pAacDecoderChannelInfo[L]->icsInfo), max_sfb_ste, maxSfBandsL, maxSfBandsR, pAacDecoderChannelInfo[L] ->pComData->jointStereoData.store_dmx_re_prev, &(pAacDecoderChannelInfo[L] ->pComData->jointStereoData.store_dmx_re_prev_e), 1); } } /* if (pAacDecoderChannelInfo[L]->pDynData->RawDataInfo.CommonWindow) */ } /* if (maybe_jstereo) */ for (ch = 0; ch < el_channels; ch++) { if (elFlags & AC_EL_USAC_CP_POSSIBLE) { pAacDecoderStaticChannelInfo[L] ->pCpeStaticData->jointStereoPersistentData.clearSpectralCoeffs = 0; } } CRvlc_ElementCheck(pAacDecoderChannelInfo, pAacDecoderStaticChannelInfo, flags, el_channels); } void CChannel_CodebookTableInit( CAacDecoderChannelInfo *pAacDecoderChannelInfo) { int b, w, maxBands, maxWindows; int maxSfb = GetScaleFactorBandsTransmitted(&pAacDecoderChannelInfo->icsInfo); UCHAR *pCodeBook = pAacDecoderChannelInfo->pDynData->aCodeBook; if (IsLongBlock(&pAacDecoderChannelInfo->icsInfo)) { maxBands = 64; maxWindows = 1; } else { maxBands = 16; maxWindows = 8; } for (w = 0; w < maxWindows; w++) { for (b = 0; b < maxSfb; b++) { pCodeBook[b] = ESCBOOK; } for (; b < maxBands; b++) { pCodeBook[b] = ZERO_HCB; } pCodeBook += maxBands; } } /* * Arbitrary order bitstream parser */ AAC_DECODER_ERROR CChannelElement_Read( HANDLE_FDK_BITSTREAM hBs, CAacDecoderChannelInfo *pAacDecoderChannelInfo[], CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo[], const AUDIO_OBJECT_TYPE aot, SamplingRateInfo *pSamplingRateInfo, const UINT flags, const UINT elFlags, const UINT frame_length, const UCHAR numberOfChannels, const SCHAR epConfig, HANDLE_TRANSPORTDEC pTpDec) { AAC_DECODER_ERROR error = AAC_DEC_OK; const element_list_t *list; int i, ch, decision_bit; int crcReg1 = -1, crcReg2 = -1; int cplxPred; int ind_sw_cce_flag = 0, num_gain_element_lists = 0; FDK_ASSERT((numberOfChannels == 1) || (numberOfChannels == 2)); /* Get channel element sequence table */ list = getBitstreamElementList(aot, epConfig, numberOfChannels, 0, elFlags); if (list == NULL) { error = AAC_DEC_UNSUPPORTED_FORMAT; goto bail; } CTns_Reset(&pAacDecoderChannelInfo[0]->pDynData->TnsData); /* Set common window to 0 by default. If signalized in the bit stream it will * be overwritten later explicitely */ pAacDecoderChannelInfo[0]->pDynData->RawDataInfo.CommonWindow = 0; if (flags & (AC_USAC | AC_RSVD50 | AC_RSV603DA)) { pAacDecoderChannelInfo[0]->pDynData->specificTo.usac.tns_active = 0; pAacDecoderChannelInfo[0]->pDynData->specificTo.usac.tns_on_lr = 0; } if (numberOfChannels == 2) { CTns_Reset(&pAacDecoderChannelInfo[1]->pDynData->TnsData); pAacDecoderChannelInfo[1]->pDynData->RawDataInfo.CommonWindow = 0; } cplxPred = 0; if (pAacDecoderStaticChannelInfo != NULL) { if (elFlags & AC_EL_USAC_CP_POSSIBLE) { pAacDecoderChannelInfo[0]->pComData->jointStereoData.cplx_pred_flag = 0; cplxPred = 1; } } if (0 || (flags & (AC_ELD | AC_SCALABLE))) { pAacDecoderChannelInfo[0]->pDynData->RawDataInfo.CommonWindow = 1; if (numberOfChannels == 2) { pAacDecoderChannelInfo[1]->pDynData->RawDataInfo.CommonWindow = pAacDecoderChannelInfo[0]->pDynData->RawDataInfo.CommonWindow; } } /* Iterate through sequence table */ i = 0; ch = 0; decision_bit = 0; do { switch (list->id[i]) { case element_instance_tag: pAacDecoderChannelInfo[0]->ElementInstanceTag = FDKreadBits(hBs, 4); if (numberOfChannels == 2) { pAacDecoderChannelInfo[1]->ElementInstanceTag = pAacDecoderChannelInfo[0]->ElementInstanceTag; } break; case common_window: decision_bit = pAacDecoderChannelInfo[ch]->pDynData->RawDataInfo.CommonWindow = FDKreadBits(hBs, 1); if (numberOfChannels == 2) { pAacDecoderChannelInfo[1]->pDynData->RawDataInfo.CommonWindow = pAacDecoderChannelInfo[0]->pDynData->RawDataInfo.CommonWindow; } break; case ics_info: /* store last window sequence (utilized in complex stereo prediction) * before reading new channel-info */ if (cplxPred) { if (pAacDecoderChannelInfo[0]->pDynData->RawDataInfo.CommonWindow) { pAacDecoderStaticChannelInfo[0] ->pCpeStaticData->jointStereoPersistentData.winSeqPrev = pAacDecoderChannelInfo[0]->icsInfo.WindowSequence; pAacDecoderStaticChannelInfo[0] ->pCpeStaticData->jointStereoPersistentData.winShapePrev = pAacDecoderChannelInfo[0]->icsInfo.WindowShape; } } /* Read individual channel info */ error = IcsRead(hBs, &pAacDecoderChannelInfo[ch]->icsInfo, pSamplingRateInfo, flags); if (elFlags & AC_EL_LFE && GetWindowSequence(&pAacDecoderChannelInfo[ch]->icsInfo) != BLOCK_LONG) { error = AAC_DEC_PARSE_ERROR; break; } if (numberOfChannels == 2 && pAacDecoderChannelInfo[0]->pDynData->RawDataInfo.CommonWindow) { pAacDecoderChannelInfo[1]->icsInfo = pAacDecoderChannelInfo[0]->icsInfo; } break; case common_max_sfb: if (FDKreadBit(hBs) == 0) { error = IcsReadMaxSfb(hBs, &pAacDecoderChannelInfo[1]->icsInfo, pSamplingRateInfo); } break; case ltp_data_present: if (FDKreadBits(hBs, 1) != 0) { error = AAC_DEC_UNSUPPORTED_PREDICTION; } break; case ms: INT max_sfb_ste; INT max_sfb_ste_clear; max_sfb_ste = GetScaleMaxFactorBandsTransmitted( &pAacDecoderChannelInfo[0]->icsInfo, &pAacDecoderChannelInfo[1]->icsInfo); max_sfb_ste_clear = 64; pAacDecoderChannelInfo[0]->icsInfo.max_sfb_ste = (UCHAR)max_sfb_ste; pAacDecoderChannelInfo[1]->icsInfo.max_sfb_ste = (UCHAR)max_sfb_ste; if (flags & (AC_USAC | AC_RSV603DA) && pAacDecoderChannelInfo[ch]->pDynData->RawDataInfo.CommonWindow == 0) { Clean_Complex_Prediction_coefficients( &pAacDecoderStaticChannelInfo[0] ->pCpeStaticData->jointStereoPersistentData, GetWindowGroups(&pAacDecoderChannelInfo[0]->icsInfo), 0, 64); } if (CJointStereo_Read( hBs, &pAacDecoderChannelInfo[0]->pComData->jointStereoData, GetWindowGroups(&pAacDecoderChannelInfo[0]->icsInfo), max_sfb_ste, max_sfb_ste_clear, /* jointStereoPersistentData and cplxPredictionData are only available/allocated if cplxPred is active. */ ((cplxPred == 0) || (pAacDecoderStaticChannelInfo == NULL)) ? NULL : &pAacDecoderStaticChannelInfo[0] ->pCpeStaticData->jointStereoPersistentData, ((cplxPred == 0) || (pAacDecoderChannelInfo[0] == NULL)) ? NULL : pAacDecoderChannelInfo[0] ->pComStaticData->cplxPredictionData, cplxPred, GetScaleFactorBandsTotal(&pAacDecoderChannelInfo[0]->icsInfo), GetWindowSequence(&pAacDecoderChannelInfo[0]->icsInfo), flags)) { error = AAC_DEC_PARSE_ERROR; } break; case global_gain: pAacDecoderChannelInfo[ch]->pDynData->RawDataInfo.GlobalGain = (UCHAR)FDKreadBits(hBs, 8); break; case section_data: error = CBlock_ReadSectionData(hBs, pAacDecoderChannelInfo[ch], pSamplingRateInfo, flags); break; case scale_factor_data_usac: pAacDecoderChannelInfo[ch]->currAliasingSymmetry = 0; /* Set active sfb codebook indexes to HCB_ESC to make them "active" */ CChannel_CodebookTableInit( pAacDecoderChannelInfo[ch]); /* equals ReadSectionData(self, bs) in float soft. block.c line: ~599 */ /* Note: The missing "break" is intentional here, since we need to call * CBlock_ReadScaleFactorData(). */ FDK_FALLTHROUGH; case scale_factor_data: if (flags & AC_ER_RVLC) { /* read RVLC data from bitstream (error sens. cat. 1) */ CRvlc_Read(pAacDecoderChannelInfo[ch], hBs); } else { error = CBlock_ReadScaleFactorData(pAacDecoderChannelInfo[ch], hBs, flags); } break; case pulse: if (CPulseData_Read( hBs, &pAacDecoderChannelInfo[ch]->pDynData->specificTo.aac.PulseData, pSamplingRateInfo->ScaleFactorBands_Long, /* pulse data is only allowed to be present in long blocks! */ (void *)&pAacDecoderChannelInfo[ch]->icsInfo, frame_length) != 0) { error = AAC_DEC_DECODE_FRAME_ERROR; } break; case tns_data_present: CTns_ReadDataPresentFlag( hBs, &pAacDecoderChannelInfo[ch]->pDynData->TnsData); if (elFlags & AC_EL_LFE && pAacDecoderChannelInfo[ch]->pDynData->TnsData.DataPresent) { error = AAC_DEC_PARSE_ERROR; } break; case tns_data: /* tns_data_present is checked inside CTns_Read(). */ error = CTns_Read(hBs, &pAacDecoderChannelInfo[ch]->pDynData->TnsData, &pAacDecoderChannelInfo[ch]->icsInfo, flags); break; case gain_control_data: break; case gain_control_data_present: if (FDKreadBits(hBs, 1)) { error = AAC_DEC_UNSUPPORTED_GAIN_CONTROL_DATA; } break; case tw_data: break; case common_tw: break; case tns_data_present_usac: if (pAacDecoderChannelInfo[0]->pDynData->specificTo.usac.tns_active) { CTns_ReadDataPresentUsac( hBs, &pAacDecoderChannelInfo[0]->pDynData->TnsData, &pAacDecoderChannelInfo[1]->pDynData->TnsData, &pAacDecoderChannelInfo[0]->pDynData->specificTo.usac.tns_on_lr, &pAacDecoderChannelInfo[0]->icsInfo, flags, elFlags, pAacDecoderChannelInfo[0]->pDynData->RawDataInfo.CommonWindow); } else { pAacDecoderChannelInfo[0]->pDynData->specificTo.usac.tns_on_lr = (UCHAR)1; } break; case core_mode: decision_bit = FDKreadBits(hBs, 1); pAacDecoderChannelInfo[ch]->data.usac.core_mode = decision_bit; if ((ch == 1) && (pAacDecoderChannelInfo[0]->data.usac.core_mode != pAacDecoderChannelInfo[1]->data.usac.core_mode)) { /* StereoCoreToolInfo(core_mode[ch] ) */ pAacDecoderChannelInfo[0]->pDynData->RawDataInfo.CommonWindow = 0; pAacDecoderChannelInfo[1]->pDynData->RawDataInfo.CommonWindow = 0; } break; case tns_active: pAacDecoderChannelInfo[0]->pDynData->specificTo.usac.tns_active = FDKreadBit(hBs); break; case noise: if (elFlags & AC_EL_USAC_NOISE) { pAacDecoderChannelInfo[ch] ->pDynData->specificTo.usac.fd_noise_level_and_offset = FDKreadBits(hBs, 3 + 5); /* Noise level */ } break; case lpd_channel_stream: { error = CLpdChannelStream_Read(/* = lpd_channel_stream() */ hBs, pAacDecoderChannelInfo[ch], pAacDecoderStaticChannelInfo[ch], pSamplingRateInfo, flags); } pAacDecoderChannelInfo[ch]->renderMode = AACDEC_RENDER_LPD; break; case fac_data: { int fFacDatPresent = FDKreadBit(hBs); /* Wee need a valid fac_data[0] even if no FAC data is present (as * temporal buffer) */ pAacDecoderChannelInfo[ch]->data.usac.fac_data[0] = pAacDecoderChannelInfo[ch]->data.usac.fac_data0; if (fFacDatPresent) { if (elFlags & AC_EL_LFE) { error = AAC_DEC_PARSE_ERROR; break; } /* FAC data present, this frame is FD, so the last mode had to be * ACELP. */ if (pAacDecoderStaticChannelInfo[ch]->last_core_mode != LPD || pAacDecoderStaticChannelInfo[ch]->last_lpd_mode != 0) { pAacDecoderChannelInfo[ch]->data.usac.core_mode_last = LPD; pAacDecoderChannelInfo[ch]->data.usac.lpd_mode_last = 0; /* We can't change the past! So look to the future and go ahead! */ } CLpd_FAC_Read(hBs, pAacDecoderChannelInfo[ch]->data.usac.fac_data[0], pAacDecoderChannelInfo[ch]->data.usac.fac_data_e, CLpd_FAC_getLength( IsLongBlock(&pAacDecoderChannelInfo[ch]->icsInfo), pAacDecoderChannelInfo[ch]->granuleLength), 1, 0); } else { if (pAacDecoderStaticChannelInfo[ch]->last_core_mode == LPD && pAacDecoderStaticChannelInfo[ch]->last_lpd_mode == 0) { /* ACELP to FD transitons without FAC are possible. That is why we zero it out (i.e FAC will not be considered in the subsequent calculations */ FDKmemclear(pAacDecoderChannelInfo[ch]->data.usac.fac_data0, LFAC * sizeof(FIXP_DBL)); } } } break; case esc2_rvlc: if (flags & AC_ER_RVLC) { CRvlc_Decode(pAacDecoderChannelInfo[ch], pAacDecoderStaticChannelInfo[ch], hBs); } break; case esc1_hcr: if (flags & AC_ER_HCR) { CHcr_Read(hBs, pAacDecoderChannelInfo[ch], numberOfChannels == 2 ? ID_CPE : ID_SCE); } break; case spectral_data: error = CBlock_ReadSpectralData(hBs, pAacDecoderChannelInfo[ch], pSamplingRateInfo, flags); if (flags & AC_ELD) { pAacDecoderChannelInfo[ch]->renderMode = AACDEC_RENDER_ELDFB; } else { if (flags & AC_HDAAC) { pAacDecoderChannelInfo[ch]->renderMode = AACDEC_RENDER_INTIMDCT; } else { pAacDecoderChannelInfo[ch]->renderMode = AACDEC_RENDER_IMDCT; } } break; case ac_spectral_data: error = CBlock_ReadAcSpectralData( hBs, pAacDecoderChannelInfo[ch], pAacDecoderStaticChannelInfo[ch], pSamplingRateInfo, frame_length, flags); pAacDecoderChannelInfo[ch]->renderMode = AACDEC_RENDER_IMDCT; break; case coupled_elements: { int num_coupled_elements, c; ind_sw_cce_flag = FDKreadBit(hBs); num_coupled_elements = FDKreadBits(hBs, 3); for (c = 0; c < (num_coupled_elements + 1); c++) { int cc_target_is_cpe; num_gain_element_lists++; cc_target_is_cpe = FDKreadBit(hBs); /* cc_target_is_cpe[c] */ FDKreadBits(hBs, 4); /* cc_target_tag_select[c] */ if (cc_target_is_cpe) { int cc_l, cc_r; cc_l = FDKreadBit(hBs); /* cc_l[c] */ cc_r = FDKreadBit(hBs); /* cc_r[c] */ if (cc_l && cc_r) { num_gain_element_lists++; } } } FDKreadBit(hBs); /* cc_domain */ FDKreadBit(hBs); /* gain_element_sign */ FDKreadBits(hBs, 2); /* gain_element_scale */ } break; case gain_element_lists: { const CodeBookDescription *hcb; UCHAR *pCodeBook; int c; hcb = &AACcodeBookDescriptionTable[BOOKSCL]; pCodeBook = pAacDecoderChannelInfo[ch]->pDynData->aCodeBook; for (c = 1; c < num_gain_element_lists; c++) { int cge; if (ind_sw_cce_flag) { cge = 1; } else { cge = FDKreadBits(hBs, 1); /* common_gain_element_present[c] */ } if (cge) { /* Huffman */ CBlock_DecodeHuffmanWord( hBs, hcb); /* hcod_sf[common_gain_element[c]] 1..19 */ } else { int g, sfb; for (g = 0; g < GetWindowGroups(&pAacDecoderChannelInfo[ch]->icsInfo); g++) { for (sfb = 0; sfb < GetScaleFactorBandsTransmitted( &pAacDecoderChannelInfo[ch]->icsInfo); sfb++) { if (pCodeBook[sfb] != ZERO_HCB) { /* Huffman */ CBlock_DecodeHuffmanWord( hBs, hcb); /* hcod_sf[dpcm_gain_element[c][g][sfb]] 1..19 */ } } } } } } break; /* CRC handling */ case adtscrc_start_reg1: if (pTpDec != NULL) { crcReg1 = transportDec_CrcStartReg(pTpDec, 192); } break; case adtscrc_start_reg2: if (pTpDec != NULL) { crcReg2 = transportDec_CrcStartReg(pTpDec, 128); } break; case adtscrc_end_reg1: case drmcrc_end_reg: if (pTpDec != NULL) { transportDec_CrcEndReg(pTpDec, crcReg1); crcReg1 = -1; } break; case adtscrc_end_reg2: if (crcReg1 != -1) { error = AAC_DEC_DECODE_FRAME_ERROR; } else if (pTpDec != NULL) { transportDec_CrcEndReg(pTpDec, crcReg2); crcReg2 = -1; } break; case drmcrc_start_reg: if (pTpDec != NULL) { crcReg1 = transportDec_CrcStartReg(pTpDec, 0); } break; /* Non data cases */ case next_channel: ch = (ch + 1) % numberOfChannels; break; case link_sequence: list = list->next[decision_bit]; i = -1; break; default: error = AAC_DEC_UNSUPPORTED_FORMAT; break; } if (error != AAC_DEC_OK) { goto bail; } i++; } while (list->id[i] != end_of_sequence); for (ch = 0; ch < numberOfChannels; ch++) { if (pAacDecoderChannelInfo[ch]->renderMode == AACDEC_RENDER_IMDCT || pAacDecoderChannelInfo[ch]->renderMode == AACDEC_RENDER_ELDFB) { /* Shows which bands are empty. */ UCHAR *band_is_noise = pAacDecoderChannelInfo[ch]->pDynData->band_is_noise; FDKmemset(band_is_noise, (UCHAR)1, sizeof(UCHAR) * (8 * 16)); error = CBlock_InverseQuantizeSpectralData( pAacDecoderChannelInfo[ch], pSamplingRateInfo, band_is_noise, 1); if (error != AAC_DEC_OK) { return error; } if (elFlags & AC_EL_USAC_NOISE) { CBlock_ApplyNoise(pAacDecoderChannelInfo[ch], pSamplingRateInfo, &pAacDecoderStaticChannelInfo[ch]->nfRandomSeed, band_is_noise); } /* if (elFlags & AC_EL_USAC_NOISE) */ } } bail: if (crcReg1 != -1 || crcReg2 != -1) { if (error == AAC_DEC_OK) { error = AAC_DEC_DECODE_FRAME_ERROR; } if (crcReg1 != -1) { transportDec_CrcEndReg(pTpDec, crcReg1); } if (crcReg2 != -1) { transportDec_CrcEndReg(pTpDec, crcReg2); } } return error; } fdk-aac-2.0.2/libAACdec/src/channel.h000066400000000000000000000160071404225310700171020ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Josef Hoepfl Description: *******************************************************************************/ #ifndef CHANNEL_H #define CHANNEL_H #include "common_fix.h" #include "FDK_tools_rom.h" #include "channelinfo.h" #include "tpdec_lib.h" /** * \brief Init codeBook SFB indices (section data) with HCB_ESC. Useful for * bitstreams which do not have any section data, but still SFB's (scale factor * bands). This has the effect that upto the amount of transmitted SFB are * treated as non-zero. * \param pAacDecoderChannelInfo channel info structure containing a valid * icsInfo struct. */ void CChannel_CodebookTableInit(CAacDecoderChannelInfo *pAacDecoderChannelInfo); /** * \brief decode a channel element. To be called after CChannelElement_Read() * \param pAacDecoderChannelInfo pointer to channel data struct. Depending on * el_channels either one or two. * \param pSamplingRateInfo pointer to sample rate information structure * \param el_channels amount of channels of the element to be decoded. * \param output pointer to time domain output buffer (ACELP) */ void CChannelElement_Decode( CAacDecoderChannelInfo *pAacDecoderChannelInfo[2], CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo[2], SamplingRateInfo *pSamplingRateInfo, UINT flags, UINT elFlags, int el_channels); /** * \brief Read channel element of given type from bitstream. * \param hBs bitstream handle to access bitstream data. * \param pAacDecoderChannelInfo pointer array to store channel information. * \param aot Audio Object Type * \param pSamplingRateInfo sampling rate info table. * \param flags common parser guidance flags * \param elFlags element specific parser guidance flags * \param numberOfChannels amoun of channels contained in the object to be * parsed. * \param epConfig the current epConfig value obtained from the Audio Specific * Config. * \param pTp transport decoder handle required for ADTS CRC checking. * ... * \return an AAC_DECODER_ERROR error code. */ AAC_DECODER_ERROR CChannelElement_Read( HANDLE_FDK_BITSTREAM hBs, CAacDecoderChannelInfo *pAacDecoderChannelInfo[], CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo[], const AUDIO_OBJECT_TYPE aot, SamplingRateInfo *pSamplingRateInfo, const UINT flags, const UINT elFlags, const UINT frame_length, const UCHAR numberOfChannels, const SCHAR epConfig, HANDLE_TRANSPORTDEC pTpDec); #endif /* #ifndef CHANNEL_H */ fdk-aac-2.0.2/libAACdec/src/channelinfo.cpp000066400000000000000000000237051404225310700203140ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Josef Hoepfl Description: individual channel stream info *******************************************************************************/ #include "channelinfo.h" #include "aac_rom.h" #include "aac_ram.h" #include "FDK_bitstream.h" AAC_DECODER_ERROR IcsReadMaxSfb(HANDLE_FDK_BITSTREAM bs, CIcsInfo *pIcsInfo, const SamplingRateInfo *pSamplingRateInfo) { AAC_DECODER_ERROR ErrorStatus = AAC_DEC_OK; int nbits; if (IsLongBlock(pIcsInfo)) { nbits = 6; pIcsInfo->TotalSfBands = pSamplingRateInfo->NumberOfScaleFactorBands_Long; } else { nbits = 4; pIcsInfo->TotalSfBands = pSamplingRateInfo->NumberOfScaleFactorBands_Short; } pIcsInfo->MaxSfBands = (UCHAR)FDKreadBits(bs, nbits); if (pIcsInfo->MaxSfBands > pIcsInfo->TotalSfBands) { ErrorStatus = AAC_DEC_PARSE_ERROR; } return ErrorStatus; } AAC_DECODER_ERROR IcsRead(HANDLE_FDK_BITSTREAM bs, CIcsInfo *pIcsInfo, const SamplingRateInfo *pSamplingRateInfo, const UINT flags) { AAC_DECODER_ERROR ErrorStatus = AAC_DEC_OK; pIcsInfo->Valid = 0; if (flags & AC_ELD) { pIcsInfo->WindowSequence = BLOCK_LONG; pIcsInfo->WindowShape = 0; } else { if (!(flags & (AC_USAC | AC_RSVD50 | AC_RSV603DA))) { FDKreadBits(bs, 1); } pIcsInfo->WindowSequence = (BLOCK_TYPE)FDKreadBits(bs, 2); pIcsInfo->WindowShape = (UCHAR)FDKreadBits(bs, 1); if (flags & AC_LD) { if (pIcsInfo->WindowShape) { pIcsInfo->WindowShape = 2; /* select low overlap instead of KBD */ } } } /* Sanity check */ if ((flags & (AC_ELD | AC_LD)) && pIcsInfo->WindowSequence != BLOCK_LONG) { pIcsInfo->WindowSequence = BLOCK_LONG; ErrorStatus = AAC_DEC_PARSE_ERROR; goto bail; } ErrorStatus = IcsReadMaxSfb(bs, pIcsInfo, pSamplingRateInfo); if (ErrorStatus != AAC_DEC_OK) { goto bail; } if (IsLongBlock(pIcsInfo)) { if (!(flags & (AC_ELD | AC_SCALABLE | AC_BSAC | AC_USAC | AC_RSVD50 | AC_RSV603DA))) /* If not ELD nor Scalable nor BSAC nor USAC syntax then ... */ { if ((UCHAR)FDKreadBits(bs, 1) != 0) /* UCHAR PredictorDataPresent */ { ErrorStatus = AAC_DEC_UNSUPPORTED_PREDICTION; goto bail; } } pIcsInfo->WindowGroups = 1; pIcsInfo->WindowGroupLength[0] = 1; } else { INT i; UINT mask; pIcsInfo->ScaleFactorGrouping = (UCHAR)FDKreadBits(bs, 7); pIcsInfo->WindowGroups = 0; for (i = 0; i < (8 - 1); i++) { mask = 1 << (6 - i); pIcsInfo->WindowGroupLength[i] = 1; if (pIcsInfo->ScaleFactorGrouping & mask) { pIcsInfo->WindowGroupLength[pIcsInfo->WindowGroups]++; } else { pIcsInfo->WindowGroups++; } } /* loop runs to i < 7 only */ pIcsInfo->WindowGroupLength[8 - 1] = 1; pIcsInfo->WindowGroups++; } bail: if (ErrorStatus == AAC_DEC_OK) pIcsInfo->Valid = 1; return ErrorStatus; } /* interleave codebooks the following way 9 (84w) | 1 (51w) 10 (82w) | 2 (39w) SCL (65w) | 4 (38w) 3 (39w) | 5 (41w) | 6 (40w) | 7 (31w) | 8 (31w) (270w) (271w) */ /* Table entries are sorted as following: | num_swb_long_window | sfbands_long | num_swb_short_window | sfbands_short | */ AAC_DECODER_ERROR getSamplingRateInfo(SamplingRateInfo *t, UINT samplesPerFrame, UINT samplingRateIndex, UINT samplingRate) { int index = 0; /* Search closest samplerate according to ISO/IEC 13818-7:2005(E) 8.2.4 (Table * 38): */ if ((samplingRateIndex >= 15) || (samplesPerFrame == 768)) { const UINT borders[] = {(UINT)-1, 92017, 75132, 55426, 46009, 37566, 27713, 23004, 18783, 13856, 11502, 9391}; UINT i, samplingRateSearch = samplingRate; if (samplesPerFrame == 768) { samplingRateSearch = (samplingRate * 4) / 3; } for (i = 0; i < 11; i++) { if (borders[i] > samplingRateSearch && samplingRateSearch >= borders[i + 1]) { break; } } samplingRateIndex = i; } t->samplingRateIndex = samplingRateIndex; t->samplingRate = samplingRate; switch (samplesPerFrame) { case 1024: index = 0; break; case 960: index = 1; break; case 768: index = 2; break; case 512: index = 3; break; case 480: index = 4; break; default: return AAC_DEC_UNSUPPORTED_FORMAT; } t->ScaleFactorBands_Long = sfbOffsetTables[index][samplingRateIndex].sfbOffsetLong; t->ScaleFactorBands_Short = sfbOffsetTables[index][samplingRateIndex].sfbOffsetShort; t->NumberOfScaleFactorBands_Long = sfbOffsetTables[index][samplingRateIndex].numberOfSfbLong; t->NumberOfScaleFactorBands_Short = sfbOffsetTables[index][samplingRateIndex].numberOfSfbShort; if (t->ScaleFactorBands_Long == NULL || t->NumberOfScaleFactorBands_Long == 0) { t->samplingRate = 0; return AAC_DEC_UNSUPPORTED_FORMAT; } FDK_ASSERT((UINT)t->ScaleFactorBands_Long[t->NumberOfScaleFactorBands_Long] == samplesPerFrame); FDK_ASSERT( t->ScaleFactorBands_Short == NULL || (UINT)t->ScaleFactorBands_Short[t->NumberOfScaleFactorBands_Short] * 8 == samplesPerFrame); return AAC_DEC_OK; } fdk-aac-2.0.2/libAACdec/src/channelinfo.h000066400000000000000000000471671404225310700177710ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Josef Hoepfl Description: individual channel stream info *******************************************************************************/ #ifndef CHANNELINFO_H #define CHANNELINFO_H #include "common_fix.h" #include "aac_rom.h" #include "aacdecoder_lib.h" #include "FDK_bitstream.h" #include "overlapadd.h" #include "mdct.h" #include "stereo.h" #include "pulsedata.h" #include "aacdec_tns.h" #include "aacdec_pns.h" #include "aacdec_hcr_types.h" #include "rvlc_info.h" #include "usacdec_acelp.h" #include "usacdec_const.h" #include "usacdec_rom.h" #include "ac_arith_coder.h" #include "conceal_types.h" #include "aacdec_drc_types.h" #define WB_SECTION_SIZE (1024 * 2) #define DRM_BS_BUFFER_SIZE \ (512) /* size of the dynamic buffer which is used to reverse the bits of \ the DRM SBR payload */ /* Output rendering mode */ typedef enum { AACDEC_RENDER_INVALID = 0, AACDEC_RENDER_IMDCT, AACDEC_RENDER_ELDFB, AACDEC_RENDER_LPD, AACDEC_RENDER_INTIMDCT } AACDEC_RENDER_MODE; enum { MAX_QUANTIZED_VALUE = 8191 }; typedef enum { FD_LONG, FD_SHORT, LPD } USAC_COREMODE; typedef struct { const SHORT *ScaleFactorBands_Long; const SHORT *ScaleFactorBands_Short; UCHAR NumberOfScaleFactorBands_Long; UCHAR NumberOfScaleFactorBands_Short; UINT samplingRateIndex; UINT samplingRate; } SamplingRateInfo; typedef struct { UCHAR CommonWindow; UCHAR GlobalGain; } CRawDataInfo; typedef struct { UCHAR WindowGroupLength[8]; UCHAR WindowGroups; UCHAR Valid; UCHAR WindowShape; /* 0: sine window, 1: KBD, 2: low overlap */ BLOCK_TYPE WindowSequence; /* mdct.h; 0: long, 1: start, 2: short, 3: stop */ UCHAR MaxSfBands; UCHAR max_sfb_ste; UCHAR ScaleFactorGrouping; UCHAR TotalSfBands; } CIcsInfo; enum { ZERO_HCB = 0, ESCBOOK = 11, NSPECBOOKS = ESCBOOK + 1, BOOKSCL = NSPECBOOKS, NOISE_HCB = 13, INTENSITY_HCB2 = 14, INTENSITY_HCB = 15, LAST_HCB }; /* This struct holds the persistent data shared by both channels of a CPE. It needs to be allocated for each CPE. */ typedef struct { CJointStereoPersistentData jointStereoPersistentData; } CpePersistentData; /* * This struct must be allocated one for every channel and must be persistent. */ typedef struct { FIXP_DBL *pOverlapBuffer; mdct_t IMdct; CArcoData *hArCo; INT pnsCurrentSeed; /* LPD memory */ FIXP_DBL old_synth[PIT_MAX_MAX - L_SUBFR]; INT old_T_pf[SYN_SFD]; FIXP_DBL old_gain_pf[SYN_SFD]; FIXP_DBL mem_bpf[L_FILT + L_SUBFR]; UCHAR old_bpf_control_info; /* (1: enable, 0: disable) bpf for past superframe */ USAC_COREMODE last_core_mode; /* core mode used by the decoder in previous frame. (not signalled by the bitstream, see CAacDecoderChannelInfo::core_mode_last !! ) */ UCHAR last_lpd_mode; /* LPD mode used by the decoder in last LPD subframe (not signalled by the bitstream, see CAacDecoderChannelInfo::lpd_mode_last !! ) */ UCHAR last_last_lpd_mode; /* LPD mode used in second last LPD subframe (not signalled by the bitstream) */ UCHAR last_lpc_lost; /* Flag indicating that the previous LPC is lost */ FIXP_LPC lpc4_lsf[M_LP_FILTER_ORDER]; /* Last LPC4 coefficients in LSF domain. */ FIXP_LPC lsf_adaptive_mean[M_LP_FILTER_ORDER]; /* Adaptive mean of LPC coefficients in LSF domain for concealment. */ FIXP_LPC lp_coeff_old[2][M_LP_FILTER_ORDER]; /* Last LPC coefficients in LP domain. lp_coeff_old[0] is lpc4 (coeffs for right folding point of last tcx frame), lp_coeff_old[1] are coeffs for left folding point of last tcx frame */ INT lp_coeff_old_exp[2]; FIXP_SGL oldStability; /* LPC coeff stability value from last frame (required for TCX concealment). */ UINT numLostLpdFrames; /* Number of consecutive lost subframes. */ /* TCX memory */ FIXP_DBL last_tcx_gain; INT last_tcx_gain_e; FIXP_DBL last_alfd_gains[32]; /* Scaled by one bit. */ SHORT last_tcx_pitch; UCHAR last_tcx_noise_factor; /* ACELP memory */ CAcelpStaticMem acelp; ULONG nfRandomSeed; /* seed value for USAC noise filling random generator */ CDrcChannelData drcData; CConcealmentInfo concealmentInfo; CpePersistentData *pCpeStaticData; } CAacDecoderStaticChannelInfo; /* * This union must be allocated for every element (up to 2 channels). */ typedef struct { /* Common bit stream data */ SHORT aScaleFactor[( 8 * 16)]; /* Spectral scale factors for each sfb in each window. */ SHORT aSfbScale[(8 * 16)]; /* could be free after ApplyTools() */ UCHAR aCodeBook[(8 * 16)]; /* section data: codebook for each window and sfb. */ UCHAR band_is_noise[(8 * 16)]; CTnsData TnsData; CRawDataInfo RawDataInfo; shouldBeUnion { struct { CPulseData PulseData; SHORT aNumLineInSec4Hcr[MAX_SFB_HCR]; /* needed once for all channels except for Drm syntax */ UCHAR aCodeBooks4Hcr[MAX_SFB_HCR]; /* needed once for all channels except for Drm syntax. Same as "aCodeBook" ? */ SHORT lenOfReorderedSpectralData; SCHAR lenOfLongestCodeword; SCHAR numberSection; SCHAR rvlcCurrentScaleFactorOK; SCHAR rvlcIntensityUsed; } aac; struct { UCHAR fd_noise_level_and_offset; UCHAR tns_active; UCHAR tns_on_lr; UCHAR tcx_noise_factor[4]; UCHAR tcx_global_gain[4]; } usac; } specificTo; } CAacDecoderDynamicData; typedef shouldBeUnion { UCHAR DrmBsBuffer[DRM_BS_BUFFER_SIZE]; /* Common signal data, can be used once the bit stream data from above is not * used anymore. */ FIXP_DBL mdctOutTemp[1024]; FIXP_DBL synth_buf[(PIT_MAX_MAX + SYN_DELAY + L_FRAME_PLUS)]; FIXP_DBL workBuffer[WB_SECTION_SIZE]; } CWorkBufferCore1; /* Common data referenced by all channels */ typedef struct { CAacDecoderDynamicData pAacDecoderDynamicData[2]; CPnsInterChannelData pnsInterChannelData; INT pnsRandomSeed[(8 * 16)]; CJointStereoData jointStereoData; /* One for one element */ shouldBeUnion { struct { CErHcrInfo erHcrInfo; CErRvlcInfo erRvlcInfo; SHORT aRvlcScfEsc[RVLC_MAX_SFB]; /* needed once for all channels */ SHORT aRvlcScfFwd[RVLC_MAX_SFB]; /* needed once for all channels */ SHORT aRvlcScfBwd[RVLC_MAX_SFB]; /* needed once for all channels */ } aac; } overlay; } CAacDecoderCommonData; typedef struct { CWorkBufferCore1 *pWorkBufferCore1; CCplxPredictionData *cplxPredictionData; } CAacDecoderCommonStaticData; /* * This struct must be allocated one for every channel of every element and must * be persistent. Among its members, the following memory areas can be * overwritten under the given conditions: * - pSpectralCoefficient The memory pointed to can be overwritten after time * signal rendering. * - data can be overwritten after time signal rendering. * - pDynData memory pointed to can be overwritten after each * CChannelElement_Decode() call. * - pComData->overlay memory pointed to can be overwritten after each * CChannelElement_Decode() call.. */ typedef struct { shouldBeUnion { struct { FIXP_DBL fac_data0[LFAC]; SCHAR fac_data_e[4]; FIXP_DBL *fac_data[4]; /* Pointers to unused parts of pSpectralCoefficient */ UCHAR core_mode; /* current core mode */ USAC_COREMODE core_mode_last; /* previous core mode, signalled in the bitstream (not done by the decoder, see CAacDecoderStaticChannelInfo::last_core_mode !!)*/ UCHAR lpd_mode_last; /* previous LPD mode, signalled in the bitstream (not done by the decoder, see CAacDecoderStaticChannelInfo::last_core_mode !!)*/ UCHAR mod[4]; UCHAR bpf_control_info; /* (1: enable, 0: disable) bpf for current superframe */ FIXP_LPC lsp_coeff[5][M_LP_FILTER_ORDER]; /* linear prediction coefficients in LSP domain */ FIXP_LPC lp_coeff[5][M_LP_FILTER_ORDER]; /* linear prediction coefficients in LP domain */ INT lp_coeff_exp[5]; FIXP_LPC lsf_adaptive_mean_cand [M_LP_FILTER_ORDER]; /* concealment: is copied to CAacDecoderStaticChannelInfo->lsf_adaptive_mean once frame is assumed to be correct*/ FIXP_SGL aStability[4]; /* LPC coeff stability values required for ACELP and TCX (concealment) */ CAcelpChannelData acelp[4]; FIXP_DBL tcx_gain[4]; SCHAR tcx_gain_e[4]; } usac; struct { CPnsData PnsData; /* Not required for USAC */ } aac; } data; SPECTRAL_PTR pSpectralCoefficient; /* Spectral coefficients of each window */ SHORT specScale[8]; /* Scale shift values of each spectrum window */ CIcsInfo icsInfo; INT granuleLength; /* Size of smallest spectrum piece */ UCHAR ElementInstanceTag; AACDEC_RENDER_MODE renderMode; /* Output signal rendering mode */ CAacDecoderDynamicData * pDynData; /* Data required for one element and discarded after decoding */ CAacDecoderCommonData *pComData; /* Data required for one channel at a time during decode */ CAacDecoderCommonStaticData *pComStaticData; /* Persistent data required for one channel at a time during decode */ int currAliasingSymmetry; /* required for RSVD60 MCT */ } CAacDecoderChannelInfo; /* channelinfo.cpp */ AAC_DECODER_ERROR getSamplingRateInfo(SamplingRateInfo *t, UINT samplesPerFrame, UINT samplingRateIndex, UINT samplingRate); /** * \brief Read max SFB from bit stream and assign TotalSfBands according * to the window sequence and sample rate. * \param hBs bit stream handle as data source * \param pIcsInfo IcsInfo structure to read the window sequence and store * MaxSfBands and TotalSfBands * \param pSamplingRateInfo read only */ AAC_DECODER_ERROR IcsReadMaxSfb(HANDLE_FDK_BITSTREAM hBs, CIcsInfo *pIcsInfo, const SamplingRateInfo *pSamplingRateInfo); AAC_DECODER_ERROR IcsRead(HANDLE_FDK_BITSTREAM bs, CIcsInfo *pIcsInfo, const SamplingRateInfo *SamplingRateInfoTable, const UINT flags); /* stereo.cpp, only called from this file */ /*! \brief Applies MS stereo. The function applies MS stereo. \param pAacDecoderChannelInfo aac channel info. \param pScaleFactorBandOffsets pointer to scalefactor band offsets. \param pWindowGroupLength pointer to window group length array. \param windowGroups number of window groups. \param scaleFactorBandsTransmittedL number of transmitted scalefactor bands in left channel. \param scaleFactorBandsTransmittedR number of transmitted scalefactor bands in right channel. May differ from scaleFactorBandsTransmittedL only for USAC. \return none */ void CJointStereo_ApplyMS( CAacDecoderChannelInfo *pAacDecoderChannelInfo[2], CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo[2], FIXP_DBL *spectrumL, FIXP_DBL *spectrumR, SHORT *SFBleftScale, SHORT *SFBrightScale, SHORT *specScaleL, SHORT *specScaleR, const SHORT *pScaleFactorBandOffsets, const UCHAR *pWindowGroupLength, const int windowGroups, const int max_sfb_ste_outside, const int scaleFactorBandsTransmittedL, const int scaleFactorBandsTransmittedR, FIXP_DBL *store_dmx_re_prev, SHORT *store_dmx_re_prev_e, const int mainband_flag); /*! \brief Applies intensity stereo The function applies intensity stereo. \param pAacDecoderChannelInfo aac channel info. \param pScaleFactorBandOffsets pointer to scalefactor band offsets. \param pWindowGroupLength pointer to window group length array. \param windowGroups number of window groups. \param scaleFactorBandsTransmitted number of transmitted scalefactor bands. \return none */ void CJointStereo_ApplyIS(CAacDecoderChannelInfo *pAacDecoderChannelInfo[2], const short *pScaleFactorBandOffsets, const UCHAR *pWindowGroupLength, const int windowGroups, const int scaleFactorBandsTransmitted); /* aacdec_pns.cpp */ int CPns_IsPnsUsed(const CPnsData *pPnsData, const int group, const int band); void CPns_SetCorrelation(CPnsData *pPnsData, const int group, const int band, const int outofphase); /****************** inline functions ******************/ inline UCHAR IsValid(const CIcsInfo *pIcsInfo) { return pIcsInfo->Valid; } inline UCHAR IsLongBlock(const CIcsInfo *pIcsInfo) { return (pIcsInfo->WindowSequence != BLOCK_SHORT); } inline UCHAR GetWindowShape(const CIcsInfo *pIcsInfo) { return pIcsInfo->WindowShape; } inline BLOCK_TYPE GetWindowSequence(const CIcsInfo *pIcsInfo) { return pIcsInfo->WindowSequence; } inline const SHORT *GetScaleFactorBandOffsets( const CIcsInfo *pIcsInfo, const SamplingRateInfo *samplingRateInfo) { if (IsLongBlock(pIcsInfo)) { return samplingRateInfo->ScaleFactorBands_Long; } else { return samplingRateInfo->ScaleFactorBands_Short; } } inline UCHAR GetNumberOfScaleFactorBands( const CIcsInfo *pIcsInfo, const SamplingRateInfo *samplingRateInfo) { if (IsLongBlock(pIcsInfo)) { return samplingRateInfo->NumberOfScaleFactorBands_Long; } else { return samplingRateInfo->NumberOfScaleFactorBands_Short; } } inline int GetWindowsPerFrame(const CIcsInfo *pIcsInfo) { return (pIcsInfo->WindowSequence == BLOCK_SHORT) ? 8 : 1; } inline UCHAR GetWindowGroups(const CIcsInfo *pIcsInfo) { return pIcsInfo->WindowGroups; } inline UCHAR GetWindowGroupLength(const CIcsInfo *pIcsInfo, const INT index) { return pIcsInfo->WindowGroupLength[index]; } inline const UCHAR *GetWindowGroupLengthTable(const CIcsInfo *pIcsInfo) { return pIcsInfo->WindowGroupLength; } inline UCHAR GetScaleFactorBandsTransmitted(const CIcsInfo *pIcsInfo) { return pIcsInfo->MaxSfBands; } inline UCHAR GetScaleMaxFactorBandsTransmitted(const CIcsInfo *pIcsInfo0, const CIcsInfo *pIcsInfo1) { return fMax(pIcsInfo0->MaxSfBands, pIcsInfo1->MaxSfBands); } inline UCHAR GetScaleFactorBandsTotal(const CIcsInfo *pIcsInfo) { return pIcsInfo->TotalSfBands; } /* Note: This function applies to AAC-LC only ! */ inline UCHAR GetMaximumTnsBands(const CIcsInfo *pIcsInfo, const int samplingRateIndex) { return tns_max_bands_tbl[samplingRateIndex][!IsLongBlock(pIcsInfo)]; } #endif /* #ifndef CHANNELINFO_H */ fdk-aac-2.0.2/libAACdec/src/conceal.cpp000066400000000000000000002311571404225310700174360ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Josef Hoepfl Description: independent channel concealment *******************************************************************************/ /*! \page concealment AAC core concealment This AAC core implementation includes a concealment function, which can be enabled using the several defines during compilation. There are various tests inside the core, starting with simple CRC tests and ending in a variety of plausibility checks. If such a check indicates an invalid bitstream, then concealment is applied. Concealment is also applied when the calling main program indicates a distorted or missing data frame using the frameOK flag. This is used for error detection on the transport layer. (See below) There are three concealment-modes: 1) Muting: The spectral data is simply set to zero in case of an detected error. 2) Noise substitution: In case of an detected error, concealment copies the last frame and adds attenuates the spectral data. For this mode you have to set the #CONCEAL_NOISE define. Noise substitution adds no additional delay. 3) Interpolation: The interpolation routine swaps the spectral data from the previous and the current frame just before the final frequency to time conversion. In case a single frame is corrupted, concealmant interpolates between the last good and the first good frame to create the spectral data for the missing frame. If multiple frames are corrupted, concealment implements first a fade out based on slightly modified spectral values from the last good frame. As soon as good frames are available, concealmant fades in the new spectral data. For this mode you have to set the #CONCEAL_INTER define. Note that in this case, you also need to set #SBR_BS_DELAY_ENABLE, which basically adds approriate delay in the SBR decoder. Note that the Interpolating-Concealment increases the delay of your decoder by one frame and that it does require additional resources such as memory and computational complexity.

How concealment can be used with errors on the transport layer

Many errors can or have to be detected on the transport layer. For example in IP based systems packet loss can occur. The transport protocol used should indicate such packet loss by inserting an empty frame with frameOK=0. */ #include "conceal.h" #include "aac_rom.h" #include "genericStds.h" /* PNS (of block) */ #include "aacdec_pns.h" #include "block.h" #define CONCEAL_DFLT_COMF_NOISE_LEVEL (0x100000) #define CONCEAL_NOT_DEFINED ((UCHAR)-1) /* default settings */ #define CONCEAL_DFLT_FADEOUT_FRAMES (6) #define CONCEAL_DFLT_FADEIN_FRAMES (5) #define CONCEAL_DFLT_MUTE_RELEASE_FRAMES (0) #define CONCEAL_DFLT_FADE_FACTOR (0.707106781186548f) /* 1/sqrt(2) */ /* some often used constants: */ #define FIXP_ZERO FL2FXCONST_DBL(0.0f) #define FIXP_ONE FL2FXCONST_DBL(1.0f) #define FIXP_FL_CORRECTION FL2FXCONST_DBL(0.53333333333333333f) /* For parameter conversion */ #define CONCEAL_PARAMETER_BITS (8) #define CONCEAL_MAX_QUANT_FACTOR ((1 << CONCEAL_PARAMETER_BITS) - 1) /*#define CONCEAL_MIN_ATTENUATION_FACTOR_025 ( FL2FXCONST_DBL(0.971627951577106174) )*/ /* -0.25 dB */ #define CONCEAL_MIN_ATTENUATION_FACTOR_025_LD \ FL2FXCONST_DBL(-0.041524101186092029596853445212299) /*#define CONCEAL_MIN_ATTENUATION_FACTOR_050 ( FL2FXCONST_DBL(0.944060876285923380) )*/ /* -0.50 dB */ #define CONCEAL_MIN_ATTENUATION_FACTOR_050_LD \ FL2FXCONST_DBL(-0.083048202372184059253597008145293) typedef enum { CConcealment_NoExpand, CConcealment_Expand, CConcealment_Compress } CConcealmentExpandType; static const FIXP_SGL facMod4Table[4] = { FL2FXCONST_SGL(0.500000000f), /* FIXP_SGL(0x4000), 2^-(1-0,00) */ FL2FXCONST_SGL(0.594603558f), /* FIXP_SGL(0x4c1b), 2^-(1-0,25) */ FL2FXCONST_SGL(0.707106781f), /* FIXP_SGL(0x5a82), 2^-(1-0,50) */ FL2FXCONST_SGL(0.840896415f) /* FIXP_SGL(0x6ba2) 2^-(1-0,75) */ }; static void CConcealment_CalcBandEnergy( FIXP_DBL *spectrum, const SamplingRateInfo *pSamplingRateInfo, const int blockType, CConcealmentExpandType ex, int *sfbEnergy); static void CConcealment_InterpolateBuffer(FIXP_DBL *spectrum, SHORT *pSpecScalePrev, SHORT *pSpecScaleAct, SHORT *pSpecScaleOut, int *enPrv, int *enAct, int sfbCnt, const SHORT *pSfbOffset); static int CConcealment_ApplyInter( CConcealmentInfo *pConcealmentInfo, CAacDecoderChannelInfo *pAacDecoderChannelInfo, const SamplingRateInfo *pSamplingRateInfo, const int samplesPerFrame, const int improveTonal, const int frameOk, const int mute_release_active); static int CConcealment_ApplyNoise( CConcealmentInfo *pConcealmentInfo, CAacDecoderChannelInfo *pAacDecoderChannelInfo, CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, const SamplingRateInfo *pSamplingRateInfo, const int samplesPerFrame, const UINT flags); static void CConcealment_UpdateState( CConcealmentInfo *pConcealmentInfo, int frameOk, CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, const int samplesPerFrame, CAacDecoderChannelInfo *pAacDecoderChannelInfo); static void CConcealment_ApplyRandomSign(int iRandomPhase, FIXP_DBL *spec, int samplesPerFrame); /* TimeDomainFading */ static void CConcealment_TDFadePcmAtt(int start, int len, FIXP_DBL fadeStart, FIXP_DBL fadeStop, PCM_DEC *pcmdata); static void CConcealment_TDFadeFillFadingStations(FIXP_DBL *fadingStations, int *fadingSteps, FIXP_DBL fadeStop, FIXP_DBL fadeStart, TDfadingType fadingType); static void CConcealment_TDFading_doLinearFadingSteps(int *fadingSteps); /* Streamline the state machine */ static int CConcealment_ApplyFadeOut( int mode, CConcealmentInfo *pConcealmentInfo, CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, const int samplesPerFrame, CAacDecoderChannelInfo *pAacDecoderChannelInfo); static int CConcealment_TDNoise_Random(ULONG *seed); static void CConcealment_TDNoise_Apply(CConcealmentInfo *const pConcealmentInfo, const int len, const INT aacOutDataHeadroom, PCM_DEC *const pcmdata); static BLOCK_TYPE CConcealment_GetWinSeq(int prevWinSeq) { BLOCK_TYPE newWinSeq = BLOCK_LONG; /* Try to have only long blocks */ if (prevWinSeq == BLOCK_START || prevWinSeq == BLOCK_SHORT) { newWinSeq = BLOCK_STOP; } return (newWinSeq); } /*! \brief Init common concealment information data \param pConcealCommonData Pointer to the concealment common data structure. */ void CConcealment_InitCommonData(CConcealParams *pConcealCommonData) { if (pConcealCommonData != NULL) { int i; /* Set default error concealment technique */ pConcealCommonData->method = ConcealMethodInter; pConcealCommonData->numFadeOutFrames = CONCEAL_DFLT_FADEOUT_FRAMES; pConcealCommonData->numFadeInFrames = CONCEAL_DFLT_FADEIN_FRAMES; pConcealCommonData->numMuteReleaseFrames = CONCEAL_DFLT_MUTE_RELEASE_FRAMES; pConcealCommonData->comfortNoiseLevel = (FIXP_DBL)CONCEAL_DFLT_COMF_NOISE_LEVEL; /* Init fade factors (symetric) */ pConcealCommonData->fadeOutFactor[0] = FL2FXCONST_SGL(CONCEAL_DFLT_FADE_FACTOR); pConcealCommonData->fadeInFactor[0] = pConcealCommonData->fadeOutFactor[0]; for (i = 1; i < CONCEAL_MAX_NUM_FADE_FACTORS; i++) { pConcealCommonData->fadeOutFactor[i] = FX_DBL2FX_SGL(fMult(pConcealCommonData->fadeOutFactor[i - 1], FL2FXCONST_SGL(CONCEAL_DFLT_FADE_FACTOR))); pConcealCommonData->fadeInFactor[i] = pConcealCommonData->fadeOutFactor[i]; } } } /*! \brief Get current concealment method. \param pConcealCommonData Pointer to common concealment data (for all channels) */ CConcealmentMethod CConcealment_GetMethod(CConcealParams *pConcealCommonData) { CConcealmentMethod method = ConcealMethodNone; if (pConcealCommonData != NULL) { method = pConcealCommonData->method; } return (method); } /*! \brief Init concealment information for each channel \param pConcealChannelInfo Pointer to the channel related concealment info structure to be initialized. \param pConcealCommonData Pointer to common concealment data (for all channels) \param initRenderMode Initial render mode to be set for the current channel. \param samplesPerFrame The number of samples per frame. */ void CConcealment_InitChannelData(CConcealmentInfo *pConcealChannelInfo, CConcealParams *pConcealCommonData, AACDEC_RENDER_MODE initRenderMode, int samplesPerFrame) { int i; pConcealChannelInfo->TDNoiseSeed = 0; FDKmemclear(pConcealChannelInfo->TDNoiseStates, sizeof(pConcealChannelInfo->TDNoiseStates)); pConcealChannelInfo->TDNoiseCoef[0] = FL2FXCONST_SGL(0.05f); pConcealChannelInfo->TDNoiseCoef[1] = FL2FXCONST_SGL(0.5f); pConcealChannelInfo->TDNoiseCoef[2] = FL2FXCONST_SGL(0.45f); pConcealChannelInfo->pConcealParams = pConcealCommonData; pConcealChannelInfo->lastRenderMode = initRenderMode; pConcealChannelInfo->windowShape = CONCEAL_NOT_DEFINED; pConcealChannelInfo->windowSequence = BLOCK_LONG; /* default type */ pConcealChannelInfo->lastWinGrpLen = 1; pConcealChannelInfo->concealState = ConcealState_Ok; FDKmemclear(pConcealChannelInfo->spectralCoefficient, 1024 * sizeof(FIXP_CNCL)); for (i = 0; i < 8; i++) { pConcealChannelInfo->specScale[i] = 0; } pConcealChannelInfo->iRandomPhase = 0; pConcealChannelInfo->prevFrameOk[0] = 1; pConcealChannelInfo->prevFrameOk[1] = 1; pConcealChannelInfo->cntFadeFrames = 0; pConcealChannelInfo->cntValidFrames = 0; pConcealChannelInfo->fade_old = (FIXP_DBL)MAXVAL_DBL; pConcealChannelInfo->winGrpOffset[0] = 0; pConcealChannelInfo->winGrpOffset[1] = 0; pConcealChannelInfo->attGrpOffset[0] = 0; pConcealChannelInfo->attGrpOffset[1] = 0; } /*! \brief Set error concealment parameters \param concealParams \param method \param fadeOutSlope \param fadeInSlope \param muteRelease \param comfNoiseLevel */ AAC_DECODER_ERROR CConcealment_SetParams(CConcealParams *concealParams, int method, int fadeOutSlope, int fadeInSlope, int muteRelease, FIXP_DBL comfNoiseLevel) { /* set concealment technique */ if (method != AACDEC_CONCEAL_PARAM_NOT_SPECIFIED) { switch ((CConcealmentMethod)method) { case ConcealMethodMute: case ConcealMethodNoise: case ConcealMethodInter: /* Be sure to enable delay adjustment of SBR decoder! */ if (concealParams == NULL) { return AAC_DEC_INVALID_HANDLE; } else { /* set param */ concealParams->method = (CConcealmentMethod)method; } break; default: return AAC_DEC_SET_PARAM_FAIL; } } /* set number of frames for fade-out slope */ if (fadeOutSlope != AACDEC_CONCEAL_PARAM_NOT_SPECIFIED) { if ((fadeOutSlope < CONCEAL_MAX_NUM_FADE_FACTORS) && (fadeOutSlope >= 0)) { if (concealParams == NULL) { return AAC_DEC_INVALID_HANDLE; } else { /* set param */ concealParams->numFadeOutFrames = fadeOutSlope; } } else { return AAC_DEC_SET_PARAM_FAIL; } } /* set number of frames for fade-in slope */ if (fadeInSlope != AACDEC_CONCEAL_PARAM_NOT_SPECIFIED) { if ((fadeInSlope < CONCEAL_MAX_NUM_FADE_FACTORS) && (fadeInSlope >= 0)) { if (concealParams == NULL) { return AAC_DEC_INVALID_HANDLE; } else { /* set param */ concealParams->numFadeInFrames = fadeInSlope; } } else { return AAC_DEC_SET_PARAM_FAIL; } } /* set number of error-free frames after which the muting will be released */ if (muteRelease != AACDEC_CONCEAL_PARAM_NOT_SPECIFIED) { if ((muteRelease < (CONCEAL_MAX_NUM_FADE_FACTORS << 1)) && (muteRelease >= 0)) { if (concealParams == NULL) { return AAC_DEC_INVALID_HANDLE; } else { /* set param */ concealParams->numMuteReleaseFrames = muteRelease; } } else { return AAC_DEC_SET_PARAM_FAIL; } } /* set confort noise level which will be inserted while in state 'muting' */ if (comfNoiseLevel != (FIXP_DBL)AACDEC_CONCEAL_PARAM_NOT_SPECIFIED) { if ((comfNoiseLevel < (FIXP_DBL)0) || (comfNoiseLevel > (FIXP_DBL)MAXVAL_DBL)) { return AAC_DEC_SET_PARAM_FAIL; } if (concealParams == NULL) { return AAC_DEC_INVALID_HANDLE; } else { concealParams->comfortNoiseLevel = (FIXP_DBL)comfNoiseLevel; } } return (AAC_DEC_OK); } /*! \brief Set fade-out/in attenuation factor vectors \param concealParams \param fadeOutAttenuationVector \param fadeInAttenuationVector \return 0 if OK all other values indicate errors */ AAC_DECODER_ERROR CConcealment_SetAttenuation(CConcealParams *concealParams, const SHORT *fadeOutAttenuationVector, const SHORT *fadeInAttenuationVector) { if ((fadeOutAttenuationVector == NULL) && (fadeInAttenuationVector == NULL)) { return AAC_DEC_SET_PARAM_FAIL; } /* Fade-out factors */ if (fadeOutAttenuationVector != NULL) { int i; /* check quantized factors first */ for (i = 0; i < CONCEAL_MAX_NUM_FADE_FACTORS; i++) { if ((fadeOutAttenuationVector[i] < 0) || (fadeOutAttenuationVector[i] > CONCEAL_MAX_QUANT_FACTOR)) { return AAC_DEC_SET_PARAM_FAIL; } } if (concealParams == NULL) { return AAC_DEC_INVALID_HANDLE; } /* now dequantize factors */ for (i = 0; i < CONCEAL_MAX_NUM_FADE_FACTORS; i++) { concealParams->fadeOutFactor[i] = FX_DBL2FX_SGL(fLdPow(CONCEAL_MIN_ATTENUATION_FACTOR_025_LD, 0, (FIXP_DBL)((INT)(FL2FXCONST_DBL(1.0 / 2.0) >> (CONCEAL_PARAMETER_BITS - 1)) * (INT)fadeOutAttenuationVector[i]), CONCEAL_PARAMETER_BITS)); } } /* Fade-in factors */ if (fadeInAttenuationVector != NULL) { int i; /* check quantized factors first */ for (i = 0; i < CONCEAL_MAX_NUM_FADE_FACTORS; i++) { if ((fadeInAttenuationVector[i] < 0) || (fadeInAttenuationVector[i] > CONCEAL_MAX_QUANT_FACTOR)) { return AAC_DEC_SET_PARAM_FAIL; } } if (concealParams == NULL) { return AAC_DEC_INVALID_HANDLE; } /* now dequantize factors */ for (i = 0; i < CONCEAL_MAX_NUM_FADE_FACTORS; i++) { concealParams->fadeInFactor[i] = FX_DBL2FX_SGL( fLdPow(CONCEAL_MIN_ATTENUATION_FACTOR_025_LD, 0, (FIXP_DBL)((INT)(FIXP_ONE >> CONCEAL_PARAMETER_BITS) * (INT)fadeInAttenuationVector[i]), CONCEAL_PARAMETER_BITS)); } } return (AAC_DEC_OK); } /*! \brief Get state of concealment module. \param pConcealChannelInfo \return Concealment state. */ CConcealmentState CConcealment_GetState(CConcealmentInfo *pConcealChannelInfo) { CConcealmentState state = ConcealState_Ok; if (pConcealChannelInfo != NULL) { state = pConcealChannelInfo->concealState; } return (state); } /*! \brief Store data for concealment techniques applied later Interface function to store data for different concealment strategies */ void CConcealment_Store( CConcealmentInfo *hConcealmentInfo, CAacDecoderChannelInfo *pAacDecoderChannelInfo, CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo) { UCHAR nbDiv = NB_DIV; if (!(pAacDecoderChannelInfo->renderMode == AACDEC_RENDER_LPD && pAacDecoderChannelInfo->data.usac.mod[nbDiv - 1] == 0)) { FIXP_DBL *pSpectralCoefficient = SPEC_LONG(pAacDecoderChannelInfo->pSpectralCoefficient); SHORT *pSpecScale = pAacDecoderChannelInfo->specScale; CIcsInfo *pIcsInfo = &pAacDecoderChannelInfo->icsInfo; SHORT tSpecScale[8]; UCHAR tWindowShape; BLOCK_TYPE tWindowSequence; /* store old window infos for swapping */ tWindowSequence = hConcealmentInfo->windowSequence; tWindowShape = hConcealmentInfo->windowShape; /* store old scale factors for swapping */ FDKmemcpy(tSpecScale, hConcealmentInfo->specScale, 8 * sizeof(SHORT)); /* store new window infos */ hConcealmentInfo->windowSequence = GetWindowSequence(pIcsInfo); hConcealmentInfo->windowShape = GetWindowShape(pIcsInfo); hConcealmentInfo->lastWinGrpLen = *(GetWindowGroupLengthTable(pIcsInfo) + GetWindowGroups(pIcsInfo) - 1); /* store new scale factors */ FDKmemcpy(hConcealmentInfo->specScale, pSpecScale, 8 * sizeof(SHORT)); if (hConcealmentInfo->pConcealParams->method < ConcealMethodInter) { /* store new spectral bins */ #if (CNCL_FRACT_BITS == DFRACT_BITS) FDKmemcpy(hConcealmentInfo->spectralCoefficient, pSpectralCoefficient, 1024 * sizeof(FIXP_CNCL)); #else FIXP_CNCL *RESTRICT pCncl = &hConcealmentInfo->spectralCoefficient[1024 - 1]; FIXP_DBL *RESTRICT pSpec = &pSpectralCoefficient[1024 - 1]; int i; for (i = 1024; i != 0; i--) { *pCncl-- = FX_DBL2FX_CNCL(*pSpec--); } #endif } else { /* swap spectral data */ #if (FIXP_CNCL == FIXP_DBL) C_ALLOC_SCRATCH_START(pSpecTmp, FIXP_DBL, 1024); FDKmemcpy(pSpecTmp, pSpectralCoefficient, 1024 * sizeof(FIXP_DBL)); FDKmemcpy(pSpectralCoefficient, hConcealmentInfo->spectralCoefficient, 1024 * sizeof(FIXP_DBL)); FDKmemcpy(hConcealmentInfo->spectralCoefficient, pSpecTmp, 1024 * sizeof(FIXP_DBL)); C_ALLOC_SCRATCH_END(pSpecTmp, FIXP_DBL, 1024); #else FIXP_CNCL *RESTRICT pCncl = &hConcealmentInfo->spectralCoefficient[1024 - 1]; FIXP_DBL *RESTRICT pSpec = &pSpectralCoefficient[1024 - 1]; FIXP_DBL tSpec; for (int i = 1024; i != 0; i--) { tSpec = *pSpec; *pSpec-- = FX_CNCL2FX_DBL(*pCncl); *pCncl-- = FX_DBL2FX_CNCL(tSpec); } #endif /* complete swapping of window infos */ pIcsInfo->WindowSequence = tWindowSequence; pIcsInfo->WindowShape = tWindowShape; /* complete swapping of scale factors */ FDKmemcpy(pSpecScale, tSpecScale, 8 * sizeof(SHORT)); } } if (pAacDecoderChannelInfo->renderMode == AACDEC_RENDER_LPD) { /* Store LSF4 */ FDKmemcpy(hConcealmentInfo->lsf4, pAacDecoderStaticChannelInfo->lpc4_lsf, sizeof(hConcealmentInfo->lsf4)); /* Store TCX gain */ hConcealmentInfo->last_tcx_gain = pAacDecoderStaticChannelInfo->last_tcx_gain; hConcealmentInfo->last_tcx_gain_e = pAacDecoderStaticChannelInfo->last_tcx_gain_e; } } /*! \brief Apply concealment Interface function to different concealment strategies */ int CConcealment_Apply( CConcealmentInfo *hConcealmentInfo, CAacDecoderChannelInfo *pAacDecoderChannelInfo, CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, const SamplingRateInfo *pSamplingRateInfo, const int samplesPerFrame, const UCHAR lastLpdMode, const int frameOk, const UINT flags) { int appliedProcessing = 0; const int mute_release_active = frameOk && (hConcealmentInfo->concealState >= ConcealState_Mute) && (hConcealmentInfo->cntValidFrames + 1 <= hConcealmentInfo->pConcealParams->numMuteReleaseFrames); if (hConcealmentInfo->windowShape == CONCEAL_NOT_DEFINED) { /* Initialize window_shape with same value as in the current (parsed) frame. Because section 4.6.11.3.2 (Windowing and block switching) of ISO/IEC 14496-3:2009 says: For the first raw_data_block() to be decoded the window_shape of the left and right half of the window are identical. */ hConcealmentInfo->windowShape = pAacDecoderChannelInfo->icsInfo.WindowShape; } if (frameOk && !mute_release_active) { /* Update render mode if frameOk except for ongoing mute release state. */ hConcealmentInfo->lastRenderMode = (SCHAR)pAacDecoderChannelInfo->renderMode; /* Rescue current data for concealment in future frames */ CConcealment_Store(hConcealmentInfo, pAacDecoderChannelInfo, pAacDecoderStaticChannelInfo); /* Reset index to random sign vector to make sign calculation frame agnostic (only depends on number of subsequently concealed spectral blocks) */ hConcealmentInfo->iRandomPhase = 0; } else { if (hConcealmentInfo->lastRenderMode == AACDEC_RENDER_INVALID) { hConcealmentInfo->lastRenderMode = AACDEC_RENDER_IMDCT; } pAacDecoderChannelInfo->renderMode = (AACDEC_RENDER_MODE)hConcealmentInfo->lastRenderMode; } /* hand current frame status to the state machine */ CConcealment_UpdateState(hConcealmentInfo, frameOk, pAacDecoderStaticChannelInfo, samplesPerFrame, pAacDecoderChannelInfo); { if (!frameOk && pAacDecoderChannelInfo->renderMode == AACDEC_RENDER_IMDCT) { /* LPC extrapolation */ CLpc_Conceal(pAacDecoderChannelInfo->data.usac.lsp_coeff, pAacDecoderStaticChannelInfo->lpc4_lsf, pAacDecoderStaticChannelInfo->lsf_adaptive_mean, hConcealmentInfo->lastRenderMode == AACDEC_RENDER_IMDCT); FDKmemcpy(hConcealmentInfo->lsf4, pAacDecoderStaticChannelInfo->lpc4_lsf, sizeof(pAacDecoderStaticChannelInfo->lpc4_lsf)); } /* Create data for signal rendering according to the selected concealment * method and decoder operating mode. */ if ((!frameOk || mute_release_active) && (pAacDecoderChannelInfo->renderMode == AACDEC_RENDER_LPD)) { /* Restore old LSF4 */ FDKmemcpy(pAacDecoderStaticChannelInfo->lpc4_lsf, hConcealmentInfo->lsf4, sizeof(pAacDecoderStaticChannelInfo->lpc4_lsf)); /* Restore old TCX gain */ pAacDecoderStaticChannelInfo->last_tcx_gain = hConcealmentInfo->last_tcx_gain; pAacDecoderStaticChannelInfo->last_tcx_gain_e = hConcealmentInfo->last_tcx_gain_e; } if (!(pAacDecoderChannelInfo->renderMode == AACDEC_RENDER_LPD && pAacDecoderStaticChannelInfo->last_lpd_mode == 0)) { switch (hConcealmentInfo->pConcealParams->method) { default: case ConcealMethodMute: if (!frameOk) { /* Mute spectral data in case of errors */ FDKmemclear(pAacDecoderChannelInfo->pSpectralCoefficient, samplesPerFrame * sizeof(FIXP_DBL)); /* Set last window shape */ pAacDecoderChannelInfo->icsInfo.WindowShape = hConcealmentInfo->windowShape; appliedProcessing = 1; } break; case ConcealMethodNoise: /* Noise substitution error concealment technique */ appliedProcessing = CConcealment_ApplyNoise( hConcealmentInfo, pAacDecoderChannelInfo, pAacDecoderStaticChannelInfo, pSamplingRateInfo, samplesPerFrame, flags); break; case ConcealMethodInter: /* Energy interpolation concealment based on 3GPP */ appliedProcessing = CConcealment_ApplyInter( hConcealmentInfo, pAacDecoderChannelInfo, pSamplingRateInfo, samplesPerFrame, 0, /* don't use tonal improvement */ frameOk, mute_release_active); break; } } else if (!frameOk || mute_release_active) { /* simply restore the buffer */ FIXP_DBL *pSpectralCoefficient = SPEC_LONG(pAacDecoderChannelInfo->pSpectralCoefficient); SHORT *pSpecScale = pAacDecoderChannelInfo->specScale; CIcsInfo *pIcsInfo = &pAacDecoderChannelInfo->icsInfo; #if (CNCL_FRACT_BITS != DFRACT_BITS) FIXP_CNCL *RESTRICT pCncl = &hConcealmentInfo->spectralCoefficient[1024 - 1]; FIXP_DBL *RESTRICT pSpec = &pSpectralCoefficient[1024 - 1]; int i; #endif /* restore window infos (gri) do we need that? */ pIcsInfo->WindowSequence = hConcealmentInfo->windowSequence; pIcsInfo->WindowShape = hConcealmentInfo->windowShape; if (hConcealmentInfo->concealState != ConcealState_Mute) { /* restore scale factors */ FDKmemcpy(pSpecScale, hConcealmentInfo->specScale, 8 * sizeof(SHORT)); /* restore spectral bins */ #if (CNCL_FRACT_BITS == DFRACT_BITS) FDKmemcpy(pSpectralCoefficient, hConcealmentInfo->spectralCoefficient, 1024 * sizeof(FIXP_DBL)); #else for (i = 1024; i != 0; i--) { *pSpec-- = FX_CNCL2FX_DBL(*pCncl--); } #endif } else { /* clear scale factors */ FDKmemclear(pSpecScale, 8 * sizeof(SHORT)); /* clear buffer */ FDKmemclear(pSpectralCoefficient, 1024 * sizeof(FIXP_CNCL)); } } } /* update history */ hConcealmentInfo->prevFrameOk[0] = hConcealmentInfo->prevFrameOk[1]; hConcealmentInfo->prevFrameOk[1] = frameOk; return mute_release_active ? -1 : appliedProcessing; } /*! \brief Apply concealment noise substitution In case of frame lost this function produces a noisy frame with respect to the energies values of past frame. */ static int CConcealment_ApplyNoise( CConcealmentInfo *pConcealmentInfo, CAacDecoderChannelInfo *pAacDecoderChannelInfo, CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, const SamplingRateInfo *pSamplingRateInfo, const int samplesPerFrame, const UINT flags) { FIXP_DBL *pSpectralCoefficient = SPEC_LONG(pAacDecoderChannelInfo->pSpectralCoefficient); CIcsInfo *pIcsInfo = &pAacDecoderChannelInfo->icsInfo; int appliedProcessing = 0; FDK_ASSERT(pConcealmentInfo != NULL); FDK_ASSERT((samplesPerFrame >= 120) && (samplesPerFrame <= 1024)); switch (pConcealmentInfo->concealState) { case ConcealState_Ok: /* Nothing to do here! */ break; case ConcealState_Single: case ConcealState_FadeOut: appliedProcessing = CConcealment_ApplyFadeOut( /*mode =*/1, pConcealmentInfo, pAacDecoderStaticChannelInfo, samplesPerFrame, pAacDecoderChannelInfo); break; case ConcealState_Mute: { /* set dummy window parameters */ pIcsInfo->Valid = 0; /* Trigger the generation of a consitent IcsInfo */ pIcsInfo->WindowShape = pConcealmentInfo->windowShape; /* Prevent an invalid WindowShape (required for F/T transform) */ pIcsInfo->WindowSequence = CConcealment_GetWinSeq(pConcealmentInfo->windowSequence); pConcealmentInfo->windowSequence = pIcsInfo->WindowSequence; /* Store for next frame (spectrum in concealment buffer can't be used at all) */ /* mute spectral data */ FDKmemclear(pSpectralCoefficient, samplesPerFrame * sizeof(FIXP_DBL)); FDKmemclear(pConcealmentInfo->spectralCoefficient, samplesPerFrame * sizeof(FIXP_DBL)); appliedProcessing = 1; } break; case ConcealState_FadeIn: { /* TimeDomainFading: */ /* Attenuation of signal is done in CConcealment_TDFading() */ appliedProcessing = 1; } break; default: /* we shouldn't come here anyway */ FDK_ASSERT(0); break; } return appliedProcessing; } /*! \brief Apply concealment interpolation The function swaps the data from the current and the previous frame. If an error has occured, frame interpolation is performed to restore the missing frame. In case of multiple faulty frames, fade-in and fade-out is applied. */ static int CConcealment_ApplyInter( CConcealmentInfo *pConcealmentInfo, CAacDecoderChannelInfo *pAacDecoderChannelInfo, const SamplingRateInfo *pSamplingRateInfo, const int samplesPerFrame, const int improveTonal, const int frameOk, const int mute_release_active) { #if defined(FDK_ASSERT_ENABLE) CConcealParams *pConcealCommonData = pConcealmentInfo->pConcealParams; #endif FIXP_DBL *pSpectralCoefficient = SPEC_LONG(pAacDecoderChannelInfo->pSpectralCoefficient); CIcsInfo *pIcsInfo = &pAacDecoderChannelInfo->icsInfo; SHORT *pSpecScale = pAacDecoderChannelInfo->specScale; int sfbEnergyPrev[64]; int sfbEnergyAct[64]; int i, appliedProcessing = 0; /* clear/init */ FDKmemclear(sfbEnergyPrev, 64 * sizeof(int)); FDKmemclear(sfbEnergyAct, 64 * sizeof(int)); if (!frameOk || mute_release_active) { /* Restore last frame from concealment buffer */ pIcsInfo->WindowShape = pConcealmentInfo->windowShape; pIcsInfo->WindowSequence = pConcealmentInfo->windowSequence; /* Restore spectral data */ for (i = 0; i < samplesPerFrame; i++) { pSpectralCoefficient[i] = FX_CNCL2FX_DBL(pConcealmentInfo->spectralCoefficient[i]); } /* Restore scale factors */ FDKmemcpy(pSpecScale, pConcealmentInfo->specScale, 8 * sizeof(SHORT)); } /* if previous frame was not ok */ if (!pConcealmentInfo->prevFrameOk[1] || mute_release_active) { /* if current frame (f_n) is ok and the last but one frame (f_(n-2)) was ok, too, then interpolate both frames in order to generate the current output frame (f_(n-1)). Otherwise, use the last stored frame (f_(n-2) or f_(n-3) or ...). */ if (frameOk && pConcealmentInfo->prevFrameOk[0] && !mute_release_active) { appliedProcessing = 1; /* Interpolate both frames in order to generate the current output frame * (f_(n-1)). */ if (pIcsInfo->WindowSequence == BLOCK_SHORT) { /* f_(n-2) == BLOCK_SHORT */ /* short--??????--short, short--??????--long interpolation */ /* short--short---short, short---long---long interpolation */ int wnd; if (pConcealmentInfo->windowSequence == BLOCK_SHORT) { /* f_n == BLOCK_SHORT */ /* short--short---short interpolation */ int scaleFactorBandsTotal = pSamplingRateInfo->NumberOfScaleFactorBands_Short; const SHORT *pSfbOffset = pSamplingRateInfo->ScaleFactorBands_Short; pIcsInfo->WindowShape = (samplesPerFrame <= 512) ? 2 : 1; pIcsInfo->WindowSequence = BLOCK_SHORT; for (wnd = 0; wnd < 8; wnd++) { CConcealment_CalcBandEnergy( &pSpectralCoefficient[wnd * (samplesPerFrame / 8)], /* spec_(n-2) */ pSamplingRateInfo, BLOCK_SHORT, CConcealment_NoExpand, sfbEnergyPrev); CConcealment_CalcBandEnergy( &pConcealmentInfo->spectralCoefficient[wnd * (samplesPerFrame / 8)], /* spec_n */ pSamplingRateInfo, BLOCK_SHORT, CConcealment_NoExpand, sfbEnergyAct); CConcealment_InterpolateBuffer( &pSpectralCoefficient[wnd * (samplesPerFrame / 8)], /* spec_(n-1) */ &pSpecScale[wnd], &pConcealmentInfo->specScale[wnd], &pSpecScale[wnd], sfbEnergyPrev, sfbEnergyAct, scaleFactorBandsTotal, pSfbOffset); } } else { /* f_n != BLOCK_SHORT */ /* short---long---long interpolation */ int scaleFactorBandsTotal = pSamplingRateInfo->NumberOfScaleFactorBands_Long; const SHORT *pSfbOffset = pSamplingRateInfo->ScaleFactorBands_Long; SHORT specScaleOut; CConcealment_CalcBandEnergy( &pSpectralCoefficient[samplesPerFrame - (samplesPerFrame / 8)], /* [wnd] spec_(n-2) */ pSamplingRateInfo, BLOCK_SHORT, CConcealment_Expand, sfbEnergyAct); CConcealment_CalcBandEnergy( pConcealmentInfo->spectralCoefficient, /* spec_n */ pSamplingRateInfo, BLOCK_LONG, CConcealment_NoExpand, sfbEnergyPrev); pIcsInfo->WindowShape = 0; pIcsInfo->WindowSequence = BLOCK_STOP; for (i = 0; i < samplesPerFrame; i++) { pSpectralCoefficient[i] = pConcealmentInfo->spectralCoefficient[i]; /* spec_n */ } for (i = 0; i < 8; i++) { /* search for max(specScale) */ if (pSpecScale[i] > pSpecScale[0]) { pSpecScale[0] = pSpecScale[i]; } } CConcealment_InterpolateBuffer( pSpectralCoefficient, /* spec_(n-1) */ &pConcealmentInfo->specScale[0], &pSpecScale[0], &specScaleOut, sfbEnergyPrev, sfbEnergyAct, scaleFactorBandsTotal, pSfbOffset); pSpecScale[0] = specScaleOut; } } else { /* long--??????--short, long--??????--long interpolation */ /* long---long---short, long---long---long interpolation */ int scaleFactorBandsTotal = pSamplingRateInfo->NumberOfScaleFactorBands_Long; const SHORT *pSfbOffset = pSamplingRateInfo->ScaleFactorBands_Long; SHORT specScaleAct = pConcealmentInfo->specScale[0]; CConcealment_CalcBandEnergy(pSpectralCoefficient, /* spec_(n-2) */ pSamplingRateInfo, BLOCK_LONG, CConcealment_NoExpand, sfbEnergyPrev); if (pConcealmentInfo->windowSequence == BLOCK_SHORT) { /* f_n == BLOCK_SHORT */ /* long---long---short interpolation */ pIcsInfo->WindowShape = (samplesPerFrame <= 512) ? 2 : 1; pIcsInfo->WindowSequence = BLOCK_START; for (i = 1; i < 8; i++) { /* search for max(specScale) */ if (pConcealmentInfo->specScale[i] > specScaleAct) { specScaleAct = pConcealmentInfo->specScale[i]; } } /* Expand first short spectrum */ CConcealment_CalcBandEnergy( pConcealmentInfo->spectralCoefficient, /* spec_n */ pSamplingRateInfo, BLOCK_SHORT, CConcealment_Expand, /* !!! */ sfbEnergyAct); } else { /* long---long---long interpolation */ pIcsInfo->WindowShape = 0; pIcsInfo->WindowSequence = BLOCK_LONG; CConcealment_CalcBandEnergy( pConcealmentInfo->spectralCoefficient, /* spec_n */ pSamplingRateInfo, BLOCK_LONG, CConcealment_NoExpand, sfbEnergyAct); } CConcealment_InterpolateBuffer( pSpectralCoefficient, /* spec_(n-1) */ &pSpecScale[0], &specScaleAct, &pSpecScale[0], sfbEnergyPrev, sfbEnergyAct, scaleFactorBandsTotal, pSfbOffset); } } /* Noise substitution of sign of the output spectral coefficients */ CConcealment_ApplyRandomSign(pConcealmentInfo->iRandomPhase, pSpectralCoefficient, samplesPerFrame); /* Increment random phase index to avoid repetition artifacts. */ pConcealmentInfo->iRandomPhase = (pConcealmentInfo->iRandomPhase + 1) & (AAC_NF_NO_RANDOM_VAL - 1); } /* scale spectrum according to concealment state */ switch (pConcealmentInfo->concealState) { case ConcealState_Single: appliedProcessing = 1; break; case ConcealState_FadeOut: { FDK_ASSERT(pConcealmentInfo->cntFadeFrames >= 0); FDK_ASSERT(pConcealmentInfo->cntFadeFrames < CONCEAL_MAX_NUM_FADE_FACTORS); FDK_ASSERT(pConcealmentInfo->cntFadeFrames < pConcealCommonData->numFadeOutFrames); /* TimeDomainFading: */ /* Attenuation of signal is done in CConcealment_TDFading() */ appliedProcessing = 1; } break; case ConcealState_FadeIn: { FDK_ASSERT(pConcealmentInfo->cntFadeFrames >= 0); FDK_ASSERT(pConcealmentInfo->cntFadeFrames < CONCEAL_MAX_NUM_FADE_FACTORS); FDK_ASSERT(pConcealmentInfo->cntFadeFrames < pConcealCommonData->numFadeInFrames); /* TimeDomainFading: */ /* Attenuation of signal is done in CConcealment_TDFading() */ appliedProcessing = 1; } break; case ConcealState_Mute: { /* set dummy window parameters */ pIcsInfo->Valid = 0; /* Trigger the generation of a consitent IcsInfo */ pIcsInfo->WindowShape = pConcealmentInfo->windowShape; /* Prevent an invalid WindowShape (required for F/T transform) */ pIcsInfo->WindowSequence = CConcealment_GetWinSeq(pConcealmentInfo->windowSequence); pConcealmentInfo->windowSequence = pIcsInfo->WindowSequence; /* Store for next frame (spectrum in concealment buffer can't be used at all) */ /* mute spectral data */ FDKmemclear(pSpectralCoefficient, samplesPerFrame * sizeof(FIXP_DBL)); appliedProcessing = 1; } break; default: /* nothing to do here */ break; } return appliedProcessing; } /*! \brief Calculate the spectral energy The function calculates band-wise the spectral energy. This is used for frame interpolation. */ static void CConcealment_CalcBandEnergy( FIXP_DBL *spectrum, const SamplingRateInfo *pSamplingRateInfo, const int blockType, CConcealmentExpandType expandType, int *sfbEnergy) { const SHORT *pSfbOffset; int line, sfb, scaleFactorBandsTotal = 0; /* In the following calculations, enAccu is initialized with LSB-value in * order to avoid zero energy-level */ line = 0; switch (blockType) { case BLOCK_LONG: case BLOCK_START: case BLOCK_STOP: if (expandType == CConcealment_NoExpand) { /* standard long calculation */ scaleFactorBandsTotal = pSamplingRateInfo->NumberOfScaleFactorBands_Long; pSfbOffset = pSamplingRateInfo->ScaleFactorBands_Long; for (sfb = 0; sfb < scaleFactorBandsTotal; sfb++) { FIXP_DBL enAccu = (FIXP_DBL)(LONG)1; int sfbScale = (sizeof(LONG) << 3) - CntLeadingZeros(pSfbOffset[sfb + 1] - pSfbOffset[sfb]) - 1; /* scaling depends on sfb width. */ for (; line < pSfbOffset[sfb + 1]; line++) { enAccu += fPow2Div2(*(spectrum + line)) >> sfbScale; } *(sfbEnergy + sfb) = CntLeadingZeros(enAccu) - 1; } } else { /* compress long to short */ scaleFactorBandsTotal = pSamplingRateInfo->NumberOfScaleFactorBands_Short; pSfbOffset = pSamplingRateInfo->ScaleFactorBands_Short; for (sfb = 0; sfb < scaleFactorBandsTotal; sfb++) { FIXP_DBL enAccu = (FIXP_DBL)(LONG)1; int sfbScale = (sizeof(LONG) << 3) - CntLeadingZeros(pSfbOffset[sfb + 1] - pSfbOffset[sfb]) - 1; /* scaling depends on sfb width. */ for (; line < pSfbOffset[sfb + 1] << 3; line++) { enAccu += (enAccu + (fPow2Div2(*(spectrum + line)) >> sfbScale)) >> 3; } *(sfbEnergy + sfb) = CntLeadingZeros(enAccu) - 1; } } break; case BLOCK_SHORT: if (expandType == CConcealment_NoExpand) { /* standard short calculation */ scaleFactorBandsTotal = pSamplingRateInfo->NumberOfScaleFactorBands_Short; pSfbOffset = pSamplingRateInfo->ScaleFactorBands_Short; for (sfb = 0; sfb < scaleFactorBandsTotal; sfb++) { FIXP_DBL enAccu = (FIXP_DBL)(LONG)1; int sfbScale = (sizeof(LONG) << 3) - CntLeadingZeros(pSfbOffset[sfb + 1] - pSfbOffset[sfb]) - 1; /* scaling depends on sfb width. */ for (; line < pSfbOffset[sfb + 1]; line++) { enAccu += fPow2Div2(*(spectrum + line)) >> sfbScale; } *(sfbEnergy + sfb) = CntLeadingZeros(enAccu) - 1; } } else { /* expand short to long spectrum */ scaleFactorBandsTotal = pSamplingRateInfo->NumberOfScaleFactorBands_Long; pSfbOffset = pSamplingRateInfo->ScaleFactorBands_Long; for (sfb = 0; sfb < scaleFactorBandsTotal; sfb++) { FIXP_DBL enAccu = (FIXP_DBL)(LONG)1; int sfbScale = (sizeof(LONG) << 3) - CntLeadingZeros(pSfbOffset[sfb + 1] - pSfbOffset[sfb]) - 1; /* scaling depends on sfb width. */ for (; line < pSfbOffset[sfb + 1]; line++) { enAccu += fPow2Div2(*(spectrum + (line >> 3))) >> sfbScale; } *(sfbEnergy + sfb) = CntLeadingZeros(enAccu) - 1; } } break; } } /*! \brief Interpolate buffer The function creates the interpolated spectral data according to the energy of the last good frame and the current (good) frame. */ static void CConcealment_InterpolateBuffer(FIXP_DBL *spectrum, SHORT *pSpecScalePrv, SHORT *pSpecScaleAct, SHORT *pSpecScaleOut, int *enPrv, int *enAct, int sfbCnt, const SHORT *pSfbOffset) { int sfb, line = 0; int fac_shift; int fac_mod; for (sfb = 0; sfb < sfbCnt; sfb++) { fac_shift = enPrv[sfb] - enAct[sfb] + ((*pSpecScaleAct - *pSpecScalePrv) << 1); fac_mod = fac_shift & 3; fac_shift = (fac_shift >> 2) + 1; fac_shift += *pSpecScalePrv - fixMax(*pSpecScalePrv, *pSpecScaleAct); fac_shift = fMax(fMin(fac_shift, DFRACT_BITS - 1), -(DFRACT_BITS - 1)); for (; line < pSfbOffset[sfb + 1]; line++) { FIXP_DBL accu = fMult(*(spectrum + line), facMod4Table[fac_mod]); *(spectrum + line) = scaleValue(accu, fac_shift); } } *pSpecScaleOut = fixMax(*pSpecScalePrv, *pSpecScaleAct); } /*! \brief Find next fading frame in case of changing fading direction \param pConcealCommonData Pointer to the concealment common data structure. \param actFadeIndex Last index used for fading \param direction Direction of change: 0 : change from FADE-OUT to FADE-IN, 1 : change from FADE-IN to FADE-OUT This function determines the next fading index to be used for the fading direction to be changed to. */ static INT findEquiFadeFrame(CConcealParams *pConcealCommonData, INT actFadeIndex, int direction) { FIXP_SGL *pFactor; FIXP_SGL referenceVal; FIXP_SGL minDiff = (FIXP_SGL)MAXVAL_SGL; INT nextFadeIndex = 0; int i; /* init depending on direction */ if (direction == 0) { /* FADE-OUT => FADE-IN */ if (actFadeIndex < 0) { referenceVal = (FIXP_SGL)MAXVAL_SGL; } else { referenceVal = pConcealCommonData->fadeOutFactor[actFadeIndex] >> 1; } pFactor = pConcealCommonData->fadeInFactor; } else { /* FADE-IN => FADE-OUT */ if (actFadeIndex < 0) { referenceVal = (FIXP_SGL)MAXVAL_SGL; } else { referenceVal = pConcealCommonData->fadeInFactor[actFadeIndex] >> 1; } pFactor = pConcealCommonData->fadeOutFactor; } /* search for minimum difference */ for (i = 0; i < CONCEAL_MAX_NUM_FADE_FACTORS; i++) { FIXP_SGL diff = fixp_abs((pFactor[i] >> 1) - referenceVal); if (diff < minDiff) { minDiff = diff; nextFadeIndex = i; } } /* check and adjust depending on direction */ if (direction == 0) { /* FADE-OUT => FADE-IN */ if (nextFadeIndex > pConcealCommonData->numFadeInFrames) { nextFadeIndex = fMax(pConcealCommonData->numFadeInFrames - 1, 0); } if (((pFactor[nextFadeIndex] >> 1) <= referenceVal) && (nextFadeIndex > 0)) { nextFadeIndex -= 1; } } else { /* FADE-IN => FADE-OUT */ if (((pFactor[nextFadeIndex] >> 1) >= referenceVal) && (nextFadeIndex < CONCEAL_MAX_NUM_FADE_FACTORS - 1)) { nextFadeIndex += 1; } } return (nextFadeIndex); } /*! \brief Update the concealment state The function updates the state of the concealment state-machine. The states are: mute, fade-in, fade-out, interpolate and frame-ok. */ static void CConcealment_UpdateState( CConcealmentInfo *pConcealmentInfo, int frameOk, CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, const int samplesPerFrame, CAacDecoderChannelInfo *pAacDecoderChannelInfo) { CConcealParams *pConcealCommonData = pConcealmentInfo->pConcealParams; switch (pConcealCommonData->method) { case ConcealMethodNoise: { if (pConcealmentInfo->concealState != ConcealState_Ok) { /* count the valid frames during concealment process */ if (frameOk) { pConcealmentInfo->cntValidFrames += 1; } else { pConcealmentInfo->cntValidFrames = 0; } } /* -- STATE MACHINE for Noise Substitution -- */ switch (pConcealmentInfo->concealState) { case ConcealState_Ok: if (!frameOk) { pConcealmentInfo->cntFadeFrames = 0; pConcealmentInfo->cntValidFrames = 0; pConcealmentInfo->attGrpOffset[0] = 0; pConcealmentInfo->attGrpOffset[1] = 0; pConcealmentInfo->winGrpOffset[0] = 0; pConcealmentInfo->winGrpOffset[1] = 0; if (pConcealCommonData->numFadeOutFrames > 0) { /* change to state SINGLE-FRAME-LOSS */ pConcealmentInfo->concealState = ConcealState_Single; /* mode 0 just updates the Fading counter */ CConcealment_ApplyFadeOut( /*mode =*/0, pConcealmentInfo, pAacDecoderStaticChannelInfo, samplesPerFrame, pAacDecoderChannelInfo); } else { /* change to state MUTE */ pConcealmentInfo->concealState = ConcealState_Mute; } } break; case ConcealState_Single: /* Just a pre-stage before fade-out begins. Stay here only one frame! */ if (frameOk) { /* change to state OK */ pConcealmentInfo->concealState = ConcealState_Ok; } else { if (pConcealmentInfo->cntFadeFrames >= pConcealCommonData->numFadeOutFrames) { /* change to state MUTE */ pConcealmentInfo->concealState = ConcealState_Mute; } else { /* change to state FADE-OUT */ pConcealmentInfo->concealState = ConcealState_FadeOut; /* mode 0 just updates the Fading counter */ CConcealment_ApplyFadeOut( /*mode =*/0, pConcealmentInfo, pAacDecoderStaticChannelInfo, samplesPerFrame, pAacDecoderChannelInfo); } } break; case ConcealState_FadeOut: if (pConcealmentInfo->cntValidFrames > pConcealCommonData->numMuteReleaseFrames) { if (pConcealCommonData->numFadeInFrames > 0) { /* change to state FADE-IN */ pConcealmentInfo->concealState = ConcealState_FadeIn; pConcealmentInfo->cntFadeFrames = findEquiFadeFrame( pConcealCommonData, pConcealmentInfo->cntFadeFrames, 0 /* FadeOut -> FadeIn */); } else { /* change to state OK */ pConcealmentInfo->concealState = ConcealState_Ok; } } else { if (frameOk) { /* we have good frame information but stay fully in concealment - * reset winGrpOffset/attGrpOffset */ pConcealmentInfo->winGrpOffset[0] = 0; pConcealmentInfo->winGrpOffset[1] = 0; pConcealmentInfo->attGrpOffset[0] = 0; pConcealmentInfo->attGrpOffset[1] = 0; } if (pConcealmentInfo->cntFadeFrames >= pConcealCommonData->numFadeOutFrames) { /* change to state MUTE */ pConcealmentInfo->concealState = ConcealState_Mute; } else /* Stay in FADE-OUT */ { /* mode 0 just updates the Fading counter */ CConcealment_ApplyFadeOut( /*mode =*/0, pConcealmentInfo, pAacDecoderStaticChannelInfo, samplesPerFrame, pAacDecoderChannelInfo); } } break; case ConcealState_Mute: if (pConcealmentInfo->cntValidFrames > pConcealCommonData->numMuteReleaseFrames) { if (pConcealCommonData->numFadeInFrames > 0) { /* change to state FADE-IN */ pConcealmentInfo->concealState = ConcealState_FadeIn; pConcealmentInfo->cntFadeFrames = pConcealCommonData->numFadeInFrames - 1; } else { /* change to state OK */ pConcealmentInfo->concealState = ConcealState_Ok; } } else { if (frameOk) { /* we have good frame information but stay fully in concealment - * reset winGrpOffset/attGrpOffset */ pConcealmentInfo->winGrpOffset[0] = 0; pConcealmentInfo->winGrpOffset[1] = 0; pConcealmentInfo->attGrpOffset[0] = 0; pConcealmentInfo->attGrpOffset[1] = 0; } } break; case ConcealState_FadeIn: pConcealmentInfo->cntFadeFrames -= 1; if (frameOk) { if (pConcealmentInfo->cntFadeFrames < 0) { /* change to state OK */ pConcealmentInfo->concealState = ConcealState_Ok; } } else { if (pConcealCommonData->numFadeOutFrames > 0) { /* change to state FADE-OUT */ pConcealmentInfo->concealState = ConcealState_FadeOut; pConcealmentInfo->cntFadeFrames = findEquiFadeFrame( pConcealCommonData, pConcealmentInfo->cntFadeFrames + 1, 1 /* FadeIn -> FadeOut */); pConcealmentInfo->winGrpOffset[0] = 0; pConcealmentInfo->winGrpOffset[1] = 0; pConcealmentInfo->attGrpOffset[0] = 0; pConcealmentInfo->attGrpOffset[1] = 0; pConcealmentInfo ->cntFadeFrames--; /* decrease because CConcealment_ApplyFadeOut() will increase, accordingly */ /* mode 0 just updates the Fading counter */ CConcealment_ApplyFadeOut( /*mode =*/0, pConcealmentInfo, pAacDecoderStaticChannelInfo, samplesPerFrame, pAacDecoderChannelInfo); } else { /* change to state MUTE */ pConcealmentInfo->concealState = ConcealState_Mute; } } break; default: FDK_ASSERT(0); break; } } break; case ConcealMethodInter: case ConcealMethodTonal: { if (pConcealmentInfo->concealState != ConcealState_Ok) { /* count the valid frames during concealment process */ if (pConcealmentInfo->prevFrameOk[1] || (pConcealmentInfo->prevFrameOk[0] && !pConcealmentInfo->prevFrameOk[1] && frameOk)) { /* The frame is OK even if it can be estimated by the energy * interpolation algorithm */ pConcealmentInfo->cntValidFrames += 1; } else { pConcealmentInfo->cntValidFrames = 0; } } /* -- STATE MACHINE for energy interpolation -- */ switch (pConcealmentInfo->concealState) { case ConcealState_Ok: if (!(pConcealmentInfo->prevFrameOk[1] || (pConcealmentInfo->prevFrameOk[0] && !pConcealmentInfo->prevFrameOk[1] && frameOk))) { if (pConcealCommonData->numFadeOutFrames > 0) { /* Fade out only if the energy interpolation algorithm can not be * applied! */ pConcealmentInfo->concealState = ConcealState_FadeOut; } else { /* change to state MUTE */ pConcealmentInfo->concealState = ConcealState_Mute; } pConcealmentInfo->cntFadeFrames = 0; pConcealmentInfo->cntValidFrames = 0; } break; case ConcealState_Single: pConcealmentInfo->concealState = ConcealState_Ok; break; case ConcealState_FadeOut: pConcealmentInfo->cntFadeFrames += 1; if (pConcealmentInfo->cntValidFrames > pConcealCommonData->numMuteReleaseFrames) { if (pConcealCommonData->numFadeInFrames > 0) { /* change to state FADE-IN */ pConcealmentInfo->concealState = ConcealState_FadeIn; pConcealmentInfo->cntFadeFrames = findEquiFadeFrame( pConcealCommonData, pConcealmentInfo->cntFadeFrames - 1, 0 /* FadeOut -> FadeIn */); } else { /* change to state OK */ pConcealmentInfo->concealState = ConcealState_Ok; } } else { if (pConcealmentInfo->cntFadeFrames >= pConcealCommonData->numFadeOutFrames) { /* change to state MUTE */ pConcealmentInfo->concealState = ConcealState_Mute; } } break; case ConcealState_Mute: if (pConcealmentInfo->cntValidFrames > pConcealCommonData->numMuteReleaseFrames) { if (pConcealCommonData->numFadeInFrames > 0) { /* change to state FADE-IN */ pConcealmentInfo->concealState = ConcealState_FadeIn; pConcealmentInfo->cntFadeFrames = pConcealCommonData->numFadeInFrames - 1; } else { /* change to state OK */ pConcealmentInfo->concealState = ConcealState_Ok; } } break; case ConcealState_FadeIn: pConcealmentInfo->cntFadeFrames -= 1; /* used to address the fade-in factors */ if (frameOk || pConcealmentInfo->prevFrameOk[1]) { if (pConcealmentInfo->cntFadeFrames < 0) { /* change to state OK */ pConcealmentInfo->concealState = ConcealState_Ok; } } else { if (pConcealCommonData->numFadeOutFrames > 0) { /* change to state FADE-OUT */ pConcealmentInfo->concealState = ConcealState_FadeOut; pConcealmentInfo->cntFadeFrames = findEquiFadeFrame( pConcealCommonData, pConcealmentInfo->cntFadeFrames + 1, 1 /* FadeIn -> FadeOut */); } else { /* change to state MUTE */ pConcealmentInfo->concealState = ConcealState_Mute; } } break; } /* End switch(pConcealmentInfo->concealState) */ } break; default: /* Don't need a state machine for other concealment methods. */ break; } } /*! \brief Randomizes the sign of the spectral data The function toggles the sign of the spectral data randomly. This is useful to ensure the quality of the concealed frames. */ static void CConcealment_ApplyRandomSign(int randomPhase, FIXP_DBL *spec, int samplesPerFrame) { int i; USHORT packedSign = 0; /* random table 512x16bit has been reduced to 512 packed sign bits = 32x16 bit */ /* read current packed sign word */ packedSign = AacDec_randomSign[randomPhase >> 4]; packedSign >>= (randomPhase & 0xf); for (i = 0; i < samplesPerFrame; i++) { if ((randomPhase & 0xf) == 0) { packedSign = AacDec_randomSign[randomPhase >> 4]; } if (packedSign & 0x1) { spec[i] = -fMax(spec[i], (FIXP_DBL)(MINVAL_DBL + 1)); } packedSign >>= 1; randomPhase = (randomPhase + 1) & (AAC_NF_NO_RANDOM_VAL - 1); } } /*! \brief Get fadeing factor for current concealment state. The function returns the state (ok or not) of the previous frame. If called before the function CConcealment_Apply() set the fBeforeApply flag to get the correct value. \return Frame OK flag of previous frame. */ int CConcealment_GetLastFrameOk(CConcealmentInfo *hConcealmentInfo, const int fBeforeApply) { int prevFrameOk = 1; if (hConcealmentInfo != NULL) { prevFrameOk = hConcealmentInfo->prevFrameOk[fBeforeApply & 0x1]; } return prevFrameOk; } /*! \brief Get the number of delay frames introduced by concealment technique. \return Number of delay frames. */ UINT CConcealment_GetDelay(CConcealParams *pConcealCommonData) { UINT frameDelay = 0; if (pConcealCommonData != NULL) { switch (pConcealCommonData->method) { case ConcealMethodTonal: case ConcealMethodInter: frameDelay = 1; break; default: break; } } return frameDelay; } static int CConcealment_ApplyFadeOut( int mode, CConcealmentInfo *pConcealmentInfo, CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, const int samplesPerFrame, CAacDecoderChannelInfo *pAacDecoderChannelInfo) { /* mode 1 = apply RandomSign and mute spectral coefficients if necessary, * * mode 0 = Update cntFadeFrames */ /* restore frequency coefficients from buffer with a specific muting */ int srcWin, dstWin, numWindows = 1; int windowLen = samplesPerFrame; int srcGrpStart = 0; int winIdxStride = 1; int numWinGrpPerFac, attIdx, attIdxStride; int i; int appliedProcessing = 0; CIcsInfo *pIcsInfo = &pAacDecoderChannelInfo->icsInfo; FIXP_DBL *pSpectralCoefficient = SPEC_LONG(pAacDecoderChannelInfo->pSpectralCoefficient); SHORT *pSpecScale = pAacDecoderChannelInfo->specScale; /* set old window parameters */ if (pConcealmentInfo->lastRenderMode == AACDEC_RENDER_LPD) { switch (pAacDecoderStaticChannelInfo->last_lpd_mode) { case 1: numWindows = 4; srcGrpStart = 3; windowLen = samplesPerFrame >> 2; break; case 2: numWindows = 2; srcGrpStart = 1; windowLen = samplesPerFrame >> 1; winIdxStride = 2; break; case 3: numWindows = 1; srcGrpStart = 0; windowLen = samplesPerFrame; winIdxStride = 4; break; } pConcealmentInfo->lastWinGrpLen = 1; } else { pIcsInfo->WindowShape = pConcealmentInfo->windowShape; pIcsInfo->WindowSequence = pConcealmentInfo->windowSequence; if (pConcealmentInfo->windowSequence == BLOCK_SHORT) { /* short block handling */ numWindows = 8; windowLen = samplesPerFrame >> 3; srcGrpStart = numWindows - pConcealmentInfo->lastWinGrpLen; } } attIdxStride = fMax(1, (int)(numWindows / (pConcealmentInfo->lastWinGrpLen + 1))); /* load last state */ attIdx = pConcealmentInfo->cntFadeFrames; numWinGrpPerFac = pConcealmentInfo->attGrpOffset[mode]; srcWin = srcGrpStart + pConcealmentInfo->winGrpOffset[mode]; FDK_ASSERT((srcGrpStart * windowLen + windowLen) <= samplesPerFrame); FDK_ASSERT((srcWin * windowLen + windowLen) <= 1024); for (dstWin = 0; dstWin < numWindows; dstWin += 1) { FIXP_CNCL *pCncl = pConcealmentInfo->spectralCoefficient + (srcWin * windowLen); FIXP_DBL *pOut = pSpectralCoefficient + (dstWin * windowLen); if (mode == 1) { /* mute if attIdx gets large enaugh */ if (attIdx > pConcealmentInfo->pConcealParams->numFadeOutFrames) { FDKmemclear(pCncl, sizeof(FIXP_DBL) * windowLen); } /* restore frequency coefficients from buffer - attenuation is done later */ for (i = 0; i < windowLen; i++) { pOut[i] = pCncl[i]; } /* apply random change of sign for spectral coefficients */ CConcealment_ApplyRandomSign(pConcealmentInfo->iRandomPhase, pOut, windowLen); /* Increment random phase index to avoid repetition artifacts. */ pConcealmentInfo->iRandomPhase = (pConcealmentInfo->iRandomPhase + 1) & (AAC_NF_NO_RANDOM_VAL - 1); /* set old scale factors */ pSpecScale[dstWin * winIdxStride] = pConcealmentInfo->specScale[srcWin * winIdxStride]; } srcWin += 1; if (srcWin >= numWindows) { /* end of sequence -> rewind to first window of group */ srcWin = srcGrpStart; numWinGrpPerFac += 1; if (numWinGrpPerFac >= attIdxStride) { numWinGrpPerFac = 0; attIdx += 1; } } } /* store current state */ pConcealmentInfo->winGrpOffset[mode] = srcWin - srcGrpStart; FDK_ASSERT((pConcealmentInfo->winGrpOffset[mode] >= 0) && (pConcealmentInfo->winGrpOffset[mode] < 8)); pConcealmentInfo->attGrpOffset[mode] = numWinGrpPerFac; FDK_ASSERT((pConcealmentInfo->attGrpOffset[mode] >= 0) && (pConcealmentInfo->attGrpOffset[mode] < attIdxStride)); if (mode == 0) { pConcealmentInfo->cntFadeFrames = attIdx; } appliedProcessing = 1; return appliedProcessing; } /*! \brief Do Time domain fading (TDFading) in concealment case In case of concealment, this function takes care of the fading, after time domain signal has been rendered by the respective signal rendering functions. The fading out in case of ACELP decoding is not done by this function but by the ACELP decoder for the first concealed frame if CONCEAL_CORE_IGNORANT_FADE is not set. TimeDomain fading never creates jumps in energy / discontinuities, it always does a continuous fading. To achieve this, fading is always done from a starting point to a target point, while the starting point is always determined to be the last target point. By varying the target point of a fading, the fading slope can be controlled. This principle is applied to the fading within a frame and the fading from frame to frame. One frame is divided into 8 subframes to obtain 8 parts of fading slopes within a frame, each maybe with its own gradient. Workflow: 1.) Determine Fading behavior and end-of-frame target fading level, based on concealmentState (determined by CConcealment_UpdateState()) and the core mode. - By _DEFAULT_, The target fading level is determined by fadeOutFactor[cntFadeFrames] in case of fadeOut, or fadeInFactor[cntFadeFrames] in case of fadeIn. --> fading type is FADE_TIMEDOMAIN in this case. Target fading level is determined by fading index cntFadeFrames. - If concealmentState is signalling a _MUTED SIGNAL_, TDFading decays to 0 within 1/8th of a frame if numFadeOutFrames == 0. --> fading type is FADE_TIMEDOMAIN_TOSPECTRALMUTE in this case. - If concealmentState is signalling the _END OF MUTING_, TDFading fades to target fading level within 1/8th of a frame if numFadeInFrames == 0. --> fading type is FADE_TIMEDOMAIN_FROMSPECTRALMUTE in this case. Target fading level is determined by fading index cntFadeFrames. #ifndef CONCEAL_CORE_IGNORANT_FADE - In case of an _ACELP FADEOUT_, TDFading leaves fading control to ACELP decoder for 1/2 frame. --> fading type is FADE_ACELPDOMAIN in this case. #endif 2.) Render fading levels within current frame and do the final fading: Map Fading slopes to fading levels and apply to time domain signal. */ INT CConcealment_TDFading( int len, CAacDecoderStaticChannelInfo **ppAacDecoderStaticChannelInfo, const INT aacOutDataHeadroom, PCM_DEC *pcmdata, PCM_DEC *pcmdata_1) { /* Do the fading in Time domain based on concealment states and core mode */ FIXP_DBL fadeStop, attMute = (FIXP_DBL)0; int idx = 0, ii; CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo = *ppAacDecoderStaticChannelInfo; CConcealmentInfo *pConcealmentInfo = &pAacDecoderStaticChannelInfo->concealmentInfo; CConcealParams *pConcealParams = pConcealmentInfo->pConcealParams; const CConcealmentState concealState = pConcealmentInfo->concealState; TDfadingType fadingType; FIXP_DBL fadingStations[9] = {0}; int fadingSteps[8] = {0}; const FIXP_DBL fadeStart = pConcealmentInfo ->fade_old; /* start fading at last end-of-frame attenuation */ FIXP_SGL *fadeFactor = pConcealParams->fadeOutFactor; const INT cntFadeFrames = pConcealmentInfo->cntFadeFrames; int TDFadeOutStopBeforeMute = 1; int TDFadeInStopBeforeFullLevel = 1; /* determine Fading behaviour (end-of-frame attenuation and fading type) (1.) */ switch (concealState) { case ConcealState_Single: case ConcealState_Mute: case ConcealState_FadeOut: idx = (pConcealParams->method == ConcealMethodNoise) ? cntFadeFrames - 1 : cntFadeFrames; fadingType = FADE_TIMEDOMAIN; if (concealState == ConcealState_Mute || (cntFadeFrames + TDFadeOutStopBeforeMute) > pConcealmentInfo->pConcealParams->numFadeOutFrames) { fadingType = FADE_TIMEDOMAIN_TOSPECTRALMUTE; } break; case ConcealState_FadeIn: idx = cntFadeFrames; idx -= TDFadeInStopBeforeFullLevel; FDK_FALLTHROUGH; case ConcealState_Ok: fadeFactor = pConcealParams->fadeInFactor; idx = (concealState == ConcealState_Ok) ? -1 : idx; fadingType = (pConcealmentInfo->concealState_old == ConcealState_Mute) ? FADE_TIMEDOMAIN_FROMSPECTRALMUTE : FADE_TIMEDOMAIN; break; default: FDK_ASSERT(0); fadingType = FADE_TIMEDOMAIN_TOSPECTRALMUTE; break; } /* determine Target end-of-frame fading level and fading slope */ switch (fadingType) { case FADE_TIMEDOMAIN_FROMSPECTRALMUTE: fadeStop = (idx < 0) ? (FIXP_DBL)MAXVAL_DBL : FX_SGL2FX_DBL(fadeFactor[idx]); if (pConcealmentInfo->pConcealParams->numFadeInFrames == 0) { /* do step as fast as possible */ fadingSteps[0] = 1; break; } CConcealment_TDFading_doLinearFadingSteps(&fadingSteps[0]); break; case FADE_TIMEDOMAIN: fadeStop = (idx < 0) ? (FIXP_DBL)MAXVAL_DBL : FX_SGL2FX_DBL(fadeFactor[idx]); CConcealment_TDFading_doLinearFadingSteps(&fadingSteps[0]); break; case FADE_TIMEDOMAIN_TOSPECTRALMUTE: fadeStop = attMute; if (pConcealmentInfo->pConcealParams->numFadeOutFrames == 0) { /* do step as fast as possible */ fadingSteps[0] = 1; break; } CConcealment_TDFading_doLinearFadingSteps(&fadingSteps[0]); break; } /* Render fading levels within current frame and do the final fading (2.) */ len >>= 3; CConcealment_TDFadeFillFadingStations(fadingStations, fadingSteps, fadeStop, fadeStart, fadingType); if ((fadingStations[8] != (FIXP_DBL)MAXVAL_DBL) || (fadingStations[7] != (FIXP_DBL)MAXVAL_DBL) || (fadingStations[6] != (FIXP_DBL)MAXVAL_DBL) || (fadingStations[5] != (FIXP_DBL)MAXVAL_DBL) || (fadingStations[4] != (FIXP_DBL)MAXVAL_DBL) || (fadingStations[3] != (FIXP_DBL)MAXVAL_DBL) || (fadingStations[2] != (FIXP_DBL)MAXVAL_DBL) || (fadingStations[1] != (FIXP_DBL)MAXVAL_DBL) || (fadingStations[0] != (FIXP_DBL)MAXVAL_DBL)) /* if there's something to fade */ { int start = 0; for (ii = 0; ii < 8; ii++) { CConcealment_TDFadePcmAtt(start, len, fadingStations[ii], fadingStations[ii + 1], pcmdata); start += len; } } CConcealment_TDNoise_Apply(pConcealmentInfo, len, aacOutDataHeadroom, pcmdata); /* Save end-of-frame attenuation and fading type */ pConcealmentInfo->lastFadingType = fadingType; pConcealmentInfo->fade_old = fadeStop; pConcealmentInfo->concealState_old = concealState; return 1; } /* attenuate pcmdata in Time Domain Fading process */ static void CConcealment_TDFadePcmAtt(int start, int len, FIXP_DBL fadeStart, FIXP_DBL fadeStop, PCM_DEC *pcmdata) { int i; FIXP_DBL dStep; FIXP_DBL dGain; FIXP_DBL dGain_apply; /* set start energy */ dGain = fadeStart; /* determine energy steps from sample to sample */ dStep = (FIXP_DBL)((int)((fadeStart >> 1) - (fadeStop >> 1)) / len) << 1; for (i = start; i < (start + len); i++) { dGain -= dStep; /* prevent gain from getting negative due to possible fixpoint inaccuracies */ dGain_apply = fMax((FIXP_DBL)0, dGain); /* finally, attenuate samples */ pcmdata[i] = FIXP_DBL2PCM_DEC(fMult(pcmdata[i], dGain_apply)); } } /* \brief Fill FadingStations The fadingstations are the attenuation factors, being applied to its dedicated portions of pcm data. They are calculated using the fadingsteps. One fadingstep is the weighted contribution to the fading slope within its dedicated portion of pcm data. *Fadingsteps : 0 0 0 1 0 1 2 0 |<- 1 Frame pcm data ->| fadeStart-->|__________ | ^ ^ ^ ^ \____ | Attenuation : | | | | ^ ^\__ | | | | | | | ^\ | | | | | | | | \___|<-- fadeStop | | | | | | | ^ ^ | | | | | | | | | Fadingstations: [0][1][2][3][4][5][6][7][8] (Fadingstations "[0]" is "[8] from previous frame", therefore its not meaningful to be edited) */ static void CConcealment_TDFadeFillFadingStations(FIXP_DBL *fadingStations, int *fadingSteps, FIXP_DBL fadeStop, FIXP_DBL fadeStart, TDfadingType fadingType) { int i; INT fadingSteps_sum = 0; INT fadeDiff; fadingSteps_sum = fadingSteps[0] + fadingSteps[1] + fadingSteps[2] + fadingSteps[3] + fadingSteps[4] + fadingSteps[5] + fadingSteps[6] + fadingSteps[7]; fadeDiff = ((INT)(fadeStop - fadeStart) / fMax(fadingSteps_sum, (INT)1)); fadingStations[0] = fadeStart; for (i = 1; i < 8; i++) { fadingStations[i] = fadingStations[i - 1] + (FIXP_DBL)(fadeDiff * fadingSteps[i - 1]); } fadingStations[8] = fadeStop; } static void CConcealment_TDFading_doLinearFadingSteps(int *fadingSteps) { fadingSteps[0] = fadingSteps[1] = fadingSteps[2] = fadingSteps[3] = fadingSteps[4] = fadingSteps[5] = fadingSteps[6] = fadingSteps[7] = 1; } /* end of TimeDomainFading functions */ /* derived from int UsacRandomSign() */ static int CConcealment_TDNoise_Random(ULONG *seed) { *seed = (ULONG)(((UINT64)(*seed) * 69069) + 5); return (int)(*seed); } static void CConcealment_TDNoise_Apply(CConcealmentInfo *const pConcealmentInfo, const int len, const INT aacOutDataHeadroom, PCM_DEC *const pcmdata) { PCM_DEC *states = pConcealmentInfo->TDNoiseStates; PCM_DEC noiseVal; FIXP_DBL noiseValLong; FIXP_SGL *coef = pConcealmentInfo->TDNoiseCoef; FIXP_DBL TDNoiseAtt; ULONG seed = pConcealmentInfo->TDNoiseSeed = (ULONG)CConcealment_TDNoise_Random(&pConcealmentInfo->TDNoiseSeed) + 1; TDNoiseAtt = pConcealmentInfo->pConcealParams->comfortNoiseLevel; int ii; if ((pConcealmentInfo->concealState != ConcealState_Ok || pConcealmentInfo->concealState_old != ConcealState_Ok) && TDNoiseAtt != (FIXP_DBL)0) { for (ii = 0; ii < (len << 3); ii++) { /* create filtered noise */ states[2] = states[1]; states[1] = states[0]; states[0] = FIXP_DBL2PCM_DEC((FIXP_DBL)CConcealment_TDNoise_Random(&seed)); noiseValLong = fMult(states[0], coef[0]) + fMult(states[1], coef[1]) + fMult(states[2], coef[2]); noiseVal = FIXP_DBL2PCM_DEC(fMult(noiseValLong, TDNoiseAtt) >> aacOutDataHeadroom); /* add filtered noise - check for clipping, before */ if (noiseVal > (PCM_DEC)0 && pcmdata[ii] > (PCM_DEC)MAXVAL_PCM_DEC - noiseVal) { noiseVal = noiseVal * (PCM_DEC)-1; } else if (noiseVal < (PCM_DEC)0 && pcmdata[ii] < (PCM_DEC)MINVAL_PCM_DEC - noiseVal) { noiseVal = noiseVal * (PCM_DEC)-1; } pcmdata[ii] += noiseVal; } } } fdk-aac-2.0.2/libAACdec/src/conceal.h000066400000000000000000000152031404225310700170730ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Josef Hoepfl Description: independent channel concealment *******************************************************************************/ #ifndef CONCEAL_H #define CONCEAL_H #include "channelinfo.h" #define AACDEC_CONCEAL_PARAM_NOT_SPECIFIED (0xFFFE) void CConcealment_InitCommonData(CConcealParams *pConcealCommonData); void CConcealment_InitChannelData(CConcealmentInfo *hConcealmentInfo, CConcealParams *pConcealCommonData, AACDEC_RENDER_MODE initRenderMode, int samplesPerFrame); CConcealmentMethod CConcealment_GetMethod(CConcealParams *pConcealCommonData); UINT CConcealment_GetDelay(CConcealParams *pConcealCommonData); AAC_DECODER_ERROR CConcealment_SetParams(CConcealParams *concealParams, int method, int fadeOutSlope, int fadeInSlope, int muteRelease, FIXP_DBL comfNoiseLevel); CConcealmentState CConcealment_GetState(CConcealmentInfo *hConcealmentInfo); AAC_DECODER_ERROR CConcealment_SetAttenuation(CConcealParams *concealParams, const SHORT *fadeOutAttenuationVector, const SHORT *fadeInAttenuationVector); void CConcealment_Store( CConcealmentInfo *hConcealmentInfo, CAacDecoderChannelInfo *pAacDecoderChannelInfo, CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo); int CConcealment_Apply( CConcealmentInfo *hConcealmentInfo, CAacDecoderChannelInfo *pAacDecoderChannelInfo, CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, const SamplingRateInfo *pSamplingRateInfo, const int samplesPerFrame, const UCHAR lastLpdMode, const int FrameOk, const UINT flags); int CConcealment_GetLastFrameOk(CConcealmentInfo *hConcealmentInfo, const int fBeforeApply); INT CConcealment_TDFading( int len, CAacDecoderStaticChannelInfo **ppAacDecoderStaticChannelInfo, const INT aacOutDataHeadroom, PCM_DEC *pcmdata, PCM_DEC *pcmdata_1); #endif /* #ifndef CONCEAL_H */ fdk-aac-2.0.2/libAACdec/src/conceal_types.h000066400000000000000000000163371404225310700203300ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Christian Griebel Description: Error concealment structs and types *******************************************************************************/ #ifndef CONCEAL_TYPES_H #define CONCEAL_TYPES_H #include "machine_type.h" #include "common_fix.h" #include "rvlc_info.h" #include "usacdec_lpc.h" #define CONCEAL_MAX_NUM_FADE_FACTORS (32) #define FIXP_CNCL FIXP_DBL #define FL2FXCONST_CNCL FL2FXCONST_DBL #define FX_DBL2FX_CNCL #define FX_CNCL2FX_DBL #define CNCL_FRACT_BITS DFRACT_BITS /* Warning: Do not ever change these values. */ typedef enum { ConcealMethodNone = -1, ConcealMethodMute = 0, ConcealMethodNoise = 1, ConcealMethodInter = 2, ConcealMethodTonal = 3 } CConcealmentMethod; typedef enum { ConcealState_Ok, ConcealState_Single, ConcealState_FadeIn, ConcealState_Mute, ConcealState_FadeOut } CConcealmentState; typedef struct { FIXP_SGL fadeOutFactor[CONCEAL_MAX_NUM_FADE_FACTORS]; FIXP_SGL fadeInFactor[CONCEAL_MAX_NUM_FADE_FACTORS]; CConcealmentMethod method; int numFadeOutFrames; int numFadeInFrames; int numMuteReleaseFrames; FIXP_DBL comfortNoiseLevel; } CConcealParams; typedef enum { FADE_TIMEDOMAIN_TOSPECTRALMUTE = 1, FADE_TIMEDOMAIN_FROMSPECTRALMUTE, FADE_TIMEDOMAIN } TDfadingType; typedef struct { CConcealParams *pConcealParams; FIXP_CNCL spectralCoefficient[1024]; SHORT specScale[8]; INT iRandomPhase; INT prevFrameOk[2]; INT cntValidFrames; INT cntFadeFrames; /* State for signal fade-in/out */ /* States for signal fade-out of frames with more than one window/subframe - [0] used by Update CntFadeFrames mode of CConcealment_ApplyFadeOut, [1] used by FadeOut mode */ int winGrpOffset[2]; /* State for signal fade-out of frames with more than one window/subframe */ int attGrpOffset[2]; /* State for faster signal fade-out of frames with transient signal parts */ SCHAR lastRenderMode; UCHAR windowShape; BLOCK_TYPE windowSequence; UCHAR lastWinGrpLen; CConcealmentState concealState; CConcealmentState concealState_old; FIXP_DBL fade_old; /* last fading factor */ TDfadingType lastFadingType; /* last fading type */ SHORT aRvlcPreviousScaleFactor[RVLC_MAX_SFB]; /* needed once per channel */ UCHAR aRvlcPreviousCodebook[RVLC_MAX_SFB]; /* needed once per channel */ SCHAR rvlcPreviousScaleFactorOK; SCHAR rvlcPreviousBlockType; FIXP_LPC lsf4[M_LP_FILTER_ORDER]; FIXP_DBL last_tcx_gain; INT last_tcx_gain_e; ULONG TDNoiseSeed; PCM_DEC TDNoiseStates[3]; FIXP_SGL TDNoiseCoef[3]; FIXP_SGL TDNoiseAtt; } CConcealmentInfo; #endif /* #ifndef CONCEAL_TYPES_H */ fdk-aac-2.0.2/libAACdec/src/ldfiltbank.cpp000066400000000000000000000244421404225310700201410ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Description: low delay filterbank *******************************************************************************/ #include "ldfiltbank.h" #include "aac_rom.h" #include "dct.h" #include "FDK_tools_rom.h" #include "mdct.h" #define LDFB_HEADROOM 2 #if defined(__arm__) #endif static void multE2_DinvF_fdk(PCM_DEC *output, FIXP_DBL *x, const FIXP_WTB *fb, FIXP_DBL *z, const int N) { int i; /* scale for FIXP_DBL -> PCM_DEC conversion: */ const int scale = (DFRACT_BITS - PCM_OUT_BITS) - LDFB_HEADROOM + (3); #if ((DFRACT_BITS - PCM_OUT_BITS - LDFB_HEADROOM + (3) - 1) > 0) FIXP_DBL rnd_val_wts0 = (FIXP_DBL)0; FIXP_DBL rnd_val_wts1 = (FIXP_DBL)0; #if ((DFRACT_BITS - PCM_OUT_BITS - LDFB_HEADROOM + (3) - WTS0 - 1) > 0) if (-WTS0 - 1 + scale) rnd_val_wts0 = (FIXP_DBL)(1 << (-WTS0 - 1 + scale - 1)); #endif if (-WTS1 - 1 + scale) rnd_val_wts1 = (FIXP_DBL)(1 << (-WTS1 - 1 + scale - 1)); #endif for (i = 0; i < N / 4; i++) { FIXP_DBL z0, z2, tmp; z2 = x[N / 2 + i]; z0 = fAddSaturate(z2, (fMultDiv2(z[N / 2 + i], fb[2 * N + i]) >> (-WTS2 - 1))); z[N / 2 + i] = fAddSaturate( x[N / 2 - 1 - i], (fMultDiv2(z[N + i], fb[2 * N + N / 2 + i]) >> (-WTS2 - 1))); tmp = (fMultDiv2(z[N / 2 + i], fb[N + N / 2 - 1 - i]) + fMultDiv2(z[i], fb[N + N / 2 + i])); #if ((DFRACT_BITS - PCM_OUT_BITS - LDFB_HEADROOM + (3) - 1) > 0) FDK_ASSERT((-WTS1 - 1 + scale) >= 0); FDK_ASSERT(tmp <= ((FIXP_DBL)0x7FFFFFFF - rnd_val_wts1)); /* rounding must not cause overflow */ output[(N * 3 / 4 - 1 - i)] = (PCM_DEC)SATURATE_RIGHT_SHIFT( tmp + rnd_val_wts1, -WTS1 - 1 + scale, PCM_OUT_BITS); #else FDK_ASSERT((WTS1 + 1 - scale) >= 0); output[(N * 3 / 4 - 1 - i)] = (PCM_DEC)SATURATE_LEFT_SHIFT(tmp, WTS1 + 1 - scale, PCM_OUT_BITS); #endif z[i] = z0; z[N + i] = z2; } for (i = N / 4; i < N / 2; i++) { FIXP_DBL z0, z2, tmp0, tmp1; z2 = x[N / 2 + i]; z0 = fAddSaturate(z2, (fMultDiv2(z[N / 2 + i], fb[2 * N + i]) >> (-WTS2 - 1))); z[N / 2 + i] = fAddSaturate( x[N / 2 - 1 - i], (fMultDiv2(z[N + i], fb[2 * N + N / 2 + i]) >> (-WTS2 - 1))); tmp0 = (fMultDiv2(z[N / 2 + i], fb[N / 2 - 1 - i]) + fMultDiv2(z[i], fb[N / 2 + i])); tmp1 = (fMultDiv2(z[N / 2 + i], fb[N + N / 2 - 1 - i]) + fMultDiv2(z[i], fb[N + N / 2 + i])); #if ((DFRACT_BITS - PCM_OUT_BITS - LDFB_HEADROOM + (3) - 1) > 0) FDK_ASSERT((-WTS0 - 1 + scale) >= 0); FDK_ASSERT(tmp0 <= ((FIXP_DBL)0x7FFFFFFF - rnd_val_wts0)); /* rounding must not cause overflow */ FDK_ASSERT(tmp1 <= ((FIXP_DBL)0x7FFFFFFF - rnd_val_wts1)); /* rounding must not cause overflow */ output[(i - N / 4)] = (PCM_DEC)SATURATE_RIGHT_SHIFT( tmp0 + rnd_val_wts0, -WTS0 - 1 + scale, PCM_OUT_BITS); output[(N * 3 / 4 - 1 - i)] = (PCM_DEC)SATURATE_RIGHT_SHIFT( tmp1 + rnd_val_wts1, -WTS1 - 1 + scale, PCM_OUT_BITS); #else FDK_ASSERT((WTS0 + 1 - scale) >= 0); output[(i - N / 4)] = (PCM_DEC)SATURATE_LEFT_SHIFT(tmp0, WTS0 + 1 - scale, PCM_OUT_BITS); output[(N * 3 / 4 - 1 - i)] = (PCM_DEC)SATURATE_LEFT_SHIFT(tmp1, WTS1 + 1 - scale, PCM_OUT_BITS); #endif z[i] = z0; z[N + i] = z2; } /* Exchange quarter parts of x to bring them in the "right" order */ for (i = 0; i < N / 4; i++) { FIXP_DBL tmp0 = fMultDiv2(z[i], fb[N / 2 + i]); #if ((DFRACT_BITS - PCM_OUT_BITS - LDFB_HEADROOM + (3) - 1) > 0) FDK_ASSERT((-WTS0 - 1 + scale) >= 0); FDK_ASSERT(tmp0 <= ((FIXP_DBL)0x7FFFFFFF - rnd_val_wts0)); /* rounding must not cause overflow */ output[(N * 3 / 4 + i)] = (PCM_DEC)SATURATE_RIGHT_SHIFT( tmp0 + rnd_val_wts0, -WTS0 - 1 + scale, PCM_OUT_BITS); #else FDK_ASSERT((WTS0 + 1 - scale) >= 0); output[(N * 3 / 4 + i)] = (PCM_DEC)SATURATE_LEFT_SHIFT(tmp0, WTS0 + 1 - scale, PCM_OUT_BITS); #endif } } int InvMdctTransformLowDelay_fdk(FIXP_DBL *mdctData, const int mdctData_e, PCM_DEC *output, FIXP_DBL *fs_buffer, const int N) { const FIXP_WTB *coef; FIXP_DBL gain = (FIXP_DBL)0; int scale = mdctData_e + MDCT_OUT_HEADROOM - LDFB_HEADROOM; /* The LDFB_HEADROOM is compensated inside multE2_DinvF_fdk() below */ int i; /* Select LD window slope */ switch (N) { case 256: coef = LowDelaySynthesis256; break; case 240: coef = LowDelaySynthesis240; break; case 160: coef = LowDelaySynthesis160; break; case 128: coef = LowDelaySynthesis128; break; case 120: coef = LowDelaySynthesis120; break; case 512: coef = LowDelaySynthesis512; break; case 480: default: coef = LowDelaySynthesis480; break; } /* Apply exponent and 1/N factor. Note: "scale" is off by one because for LD_MDCT the window length is twice the window length of a regular MDCT. This is corrected inside multE2_DinvF_fdk(). Refer to ISO/IEC 14496-3:2009 page 277, chapter 4.6.20.2 "Low Delay Window". */ imdct_gain(&gain, &scale, N); dct_IV(mdctData, N, &scale); if (N == 256 || N == 240 || N == 160) { scale -= 1; } else if (N == 128 || N == 120) { scale -= 2; } if (gain != (FIXP_DBL)0) { for (i = 0; i < N; i++) { mdctData[i] = fMult(mdctData[i], gain); } } scaleValuesSaturate(mdctData, N, scale); /* Since all exponent and factors have been applied, current exponent is zero. */ multE2_DinvF_fdk(output, mdctData, coef, fs_buffer, N); return (1); } fdk-aac-2.0.2/libAACdec/src/ldfiltbank.h000066400000000000000000000120231404225310700175760ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Description: low delay filterbank interface *******************************************************************************/ #ifndef LDFILTBANK_H #define LDFILTBANK_H #include "common_fix.h" #include "aac_rom.h" int InvMdctTransformLowDelay_fdk(FIXP_DBL *mdctdata_m, const int mdctdata_e, PCM_DEC *mdctOut, FIXP_DBL *fs_buffer, const int frameLength); #endif fdk-aac-2.0.2/libAACdec/src/overlapadd.h000066400000000000000000000122041404225310700176060ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Josef Hoepfl Description: *******************************************************************************/ #ifndef OVERLAPADD_H #define OVERLAPADD_H #include "common_fix.h" /* ELD uses different overlap which is twice the frame size: */ #define OverlapBufferSize (768) typedef FIXP_DBL SPECTRUM[1024]; typedef FIXP_DBL* SPECTRAL_PTR; #define SPEC_LONG(ptr) (ptr) #define SPEC(ptr, w, gl) ((ptr) + ((w) * (gl))) #define SPEC_TCX(ptr, f, gl, fb) \ ((ptr) + ((f) * (gl * 2) * (((fb) == 0) ? 1 : 2))) #endif /* #ifndef OVERLAPADD_H */ fdk-aac-2.0.2/libAACdec/src/pulsedata.cpp000066400000000000000000000147361404225310700200160ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Josef Hoepfl Description: pulse data tool *******************************************************************************/ #include "pulsedata.h" #include "channelinfo.h" INT CPulseData_Read(HANDLE_FDK_BITSTREAM bs, CPulseData *const PulseData, const SHORT *sfb_startlines, const void *pIcsInfo, const SHORT frame_length) { int i, k = 0; const UINT MaxSfBands = GetScaleFactorBandsTransmitted((const CIcsInfo *)pIcsInfo); /* reset pulse data flag */ PulseData->PulseDataPresent = 0; if ((PulseData->PulseDataPresent = (UCHAR)FDKreadBit(bs)) != 0) { if (!IsLongBlock((const CIcsInfo *)pIcsInfo)) { return AAC_DEC_DECODE_FRAME_ERROR; } PulseData->NumberPulse = (UCHAR)FDKreadBits(bs, 2); PulseData->PulseStartBand = (UCHAR)FDKreadBits(bs, 6); if (PulseData->PulseStartBand >= MaxSfBands) { return AAC_DEC_DECODE_FRAME_ERROR; } k = sfb_startlines[PulseData->PulseStartBand]; for (i = 0; i <= PulseData->NumberPulse; i++) { PulseData->PulseOffset[i] = (UCHAR)FDKreadBits(bs, 5); PulseData->PulseAmp[i] = (UCHAR)FDKreadBits(bs, 4); k += PulseData->PulseOffset[i]; } if (k >= frame_length) { return AAC_DEC_DECODE_FRAME_ERROR; } } return 0; } void CPulseData_Apply( CPulseData *PulseData, /*!< pointer to pulse data side info */ const short *pScaleFactorBandOffsets, /*!< pointer to scalefactor band offsets */ FIXP_DBL *coef) /*!< pointer to spectrum */ { int i, k; if (PulseData->PulseDataPresent) { k = pScaleFactorBandOffsets[PulseData->PulseStartBand]; for (i = 0; i <= PulseData->NumberPulse; i++) { k += PulseData->PulseOffset[i]; if (coef[k] > (FIXP_DBL)0) coef[k] += (FIXP_DBL)(int)PulseData->PulseAmp[i]; else coef[k] -= (FIXP_DBL)(int)PulseData->PulseAmp[i]; } } } fdk-aac-2.0.2/libAACdec/src/pulsedata.h000066400000000000000000000140761404225310700174600ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Josef Hoepfl Description: pulse data tool *******************************************************************************/ #ifndef PULSEDATA_H #define PULSEDATA_H #include "common_fix.h" #include "FDK_bitstream.h" #define N_MAX_LINES 4 typedef struct { UCHAR PulseDataPresent; UCHAR NumberPulse; UCHAR PulseStartBand; UCHAR PulseOffset[N_MAX_LINES]; UCHAR PulseAmp[N_MAX_LINES]; } CPulseData; /** * \brief Read pulse data from bitstream * * The function reads the elements for pulse data from * the bitstream. * * \param bs bit stream handle data source. * \param PulseData pointer to a CPulseData were the decoded data is stored * into. * \param MaxSfBands max number of scale factor bands. * \return 0 on success, != 0 on parse error. */ INT CPulseData_Read(const HANDLE_FDK_BITSTREAM bs, CPulseData *const PulseData, const SHORT *sfb_startlines, const void *pIcsInfo, const SHORT frame_length); /** * \brief Apply pulse data to spectral lines * * The function applies the pulse data to the * specified spectral lines. * * \param PulseData pointer to the previously decoded pulse data. * \param pScaleFactorBandOffsets scale factor band line offset table. * \param coef pointer to the spectral data were pulse data should be applied * to. * \return none */ void CPulseData_Apply(CPulseData *PulseData, const short *pScaleFactorBandOffsets, FIXP_DBL *coef); #endif /* #ifndef PULSEDATA_H */ fdk-aac-2.0.2/libAACdec/src/rvlc.cpp000066400000000000000000001323331404225310700167740ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief RVLC Decoder \author Robert Weidner */ #include "rvlc.h" #include "block.h" #include "aac_rom.h" #include "rvlcbit.h" #include "rvlcconceal.h" #include "aacdec_hcr.h" /*--------------------------------------------------------------------------------------------- function: rvlcInit description: init RVLC by data from channelinfo, which was decoded previously and set up pointers ----------------------------------------------------------------------------------------------- input: - pointer rvlc structure - pointer channel info structure - pointer bitstream structure ----------------------------------------------------------------------------------------------- return: - -------------------------------------------------------------------------------------------- */ static void rvlcInit(CErRvlcInfo *pRvlc, CAacDecoderChannelInfo *pAacDecoderChannelInfo, HANDLE_FDK_BITSTREAM bs) { /* RVLC common initialization part 2 of 2 */ SHORT *pScfEsc = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfEsc; SHORT *pScfFwd = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd; SHORT *pScfBwd = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd; SHORT *pScaleFactor = pAacDecoderChannelInfo->pDynData->aScaleFactor; int bnds; pAacDecoderChannelInfo->pDynData->specificTo.aac.rvlcIntensityUsed = 0; pRvlc->numDecodedEscapeWordsEsc = 0; pRvlc->numDecodedEscapeWordsFwd = 0; pRvlc->numDecodedEscapeWordsBwd = 0; pRvlc->intensity_used = 0; pRvlc->errorLogRvlc = 0; pRvlc->conceal_max = CONCEAL_MAX_INIT; pRvlc->conceal_min = CONCEAL_MIN_INIT; pRvlc->conceal_max_esc = CONCEAL_MAX_INIT; pRvlc->conceal_min_esc = CONCEAL_MIN_INIT; pRvlc->pHuffTreeRvlcEscape = aHuffTreeRvlcEscape; pRvlc->pHuffTreeRvlCodewds = aHuffTreeRvlCodewds; /* init scf arrays (for savety (in case of there are only zero codebooks)) */ for (bnds = 0; bnds < RVLC_MAX_SFB; bnds++) { pScfFwd[bnds] = 0; pScfBwd[bnds] = 0; pScfEsc[bnds] = 0; pScaleFactor[bnds] = 0; } /* set base bitstream ptr to the RVL-coded part (start of RVLC data (ESC 2)) */ FDKsyncCache(bs); pRvlc->bsAnchor = (INT)FDKgetValidBits(bs); pRvlc->bitstreamIndexRvlFwd = 0; /* first bit within RVL coded block as start address for forward decoding */ pRvlc->bitstreamIndexRvlBwd = pRvlc->length_of_rvlc_sf - 1; /* last bit within RVL coded block as start address for backward decoding */ /* skip RVLC-bitstream-part -- pointing now to escapes (if present) or to TNS * data (if present) */ FDKpushFor(bs, pRvlc->length_of_rvlc_sf); if (pRvlc->sf_escapes_present != 0) { /* locate internal bitstream ptr at escapes (which is the second part) */ FDKsyncCache(bs); pRvlc->bitstreamIndexEsc = pRvlc->bsAnchor - (INT)FDKgetValidBits(bs); /* skip escapeRVLC-bitstream-part -- pointing to TNS data (if present) to * make decoder continue */ /* decoding of RVLC should work despite this second pushFor during * initialization because */ /* bitstream initialization is valid for both ESC2 data parts (RVL-coded * values and ESC-coded values) */ FDKpushFor(bs, pRvlc->length_of_rvlc_escapes); } } /*--------------------------------------------------------------------------------------------- function: rvlcCheckIntensityCb description: Check if a intensity codebook is used in the current channel. ----------------------------------------------------------------------------------------------- input: - pointer rvlc structure - pointer channel info structure ----------------------------------------------------------------------------------------------- output: - intensity_used: 0 no intensity codebook is used 1 intensity codebook is used ----------------------------------------------------------------------------------------------- return: - -------------------------------------------------------------------------------------------- */ static void rvlcCheckIntensityCb( CErRvlcInfo *pRvlc, CAacDecoderChannelInfo *pAacDecoderChannelInfo) { int group, band, bnds; pRvlc->intensity_used = 0; for (group = 0; group < pRvlc->numWindowGroups; group++) { for (band = 0; band < pRvlc->maxSfbTransmitted; band++) { bnds = 16 * group + band; if ((pAacDecoderChannelInfo->pDynData->aCodeBook[bnds] == INTENSITY_HCB) || (pAacDecoderChannelInfo->pDynData->aCodeBook[bnds] == INTENSITY_HCB2)) { pRvlc->intensity_used = 1; break; } } } } /*--------------------------------------------------------------------------------------------- function: rvlcDecodeEscapeWord description: Decode a huffman coded RVLC Escape-word. This value is part of a DPCM coded scalefactor. ----------------------------------------------------------------------------------------------- input: - pointer rvlc structure ----------------------------------------------------------------------------------------------- return: - a single RVLC-Escape value which had to be applied to a DPCM value (which has a absolute value of 7) -------------------------------------------------------------------------------------------- */ static SCHAR rvlcDecodeEscapeWord(CErRvlcInfo *pRvlc, HANDLE_FDK_BITSTREAM bs) { int i; SCHAR value; UCHAR carryBit; UINT treeNode; UINT branchValue; UINT branchNode; INT *pBitstreamIndexEsc; const UINT *pEscTree; pEscTree = pRvlc->pHuffTreeRvlcEscape; pBitstreamIndexEsc = &(pRvlc->bitstreamIndexEsc); treeNode = *pEscTree; /* init at starting node */ for (i = MAX_LEN_RVLC_ESCAPE_WORD - 1; i >= 0; i--) { carryBit = rvlcReadBitFromBitstream(bs, /* get next bit */ pRvlc->bsAnchor, pBitstreamIndexEsc, FWD); CarryBitToBranchValue(carryBit, /* huffman decoding, do a single step in huffman decoding tree */ treeNode, &branchValue, &branchNode); if ((branchNode & TEST_BIT_10) == TEST_BIT_10) { /* test bit 10 ; if set --> a RVLC-escape-word is completely decoded */ value = (SCHAR)branchNode & CLR_BIT_10; pRvlc->length_of_rvlc_escapes -= (MAX_LEN_RVLC_ESCAPE_WORD - i); if (pRvlc->length_of_rvlc_escapes < 0) { pRvlc->errorLogRvlc |= RVLC_ERROR_ALL_ESCAPE_WORDS_INVALID; value = -1; } return value; } else { treeNode = *( pEscTree + branchValue); /* update treeNode for further step in decoding tree */ } } pRvlc->errorLogRvlc |= RVLC_ERROR_ALL_ESCAPE_WORDS_INVALID; return -1; /* should not be reached */ } /*--------------------------------------------------------------------------------------------- function: rvlcDecodeEscapes description: Decodes all huffman coded RVLC Escape Words. Here a difference to the pseudo-code-implementation from standard can be found. A while loop (and not two nested for loops) is used for two reasons: 1. The plain huffman encoded escapes are decoded before the RVL-coded scalefactors. Therefore the escapes are present in the second step when decoding the RVL-coded-scalefactor values in forward and backward direction. When the RVL-coded scalefactors are decoded and there a escape is needed, then it is just taken out of the array in ascending order. 2. It's faster. ----------------------------------------------------------------------------------------------- input: - pointer rvlc structure - handle to FDK bitstream ----------------------------------------------------------------------------------------------- return: - 0 ok the decoded escapes seem to be valid - 1 error there was a error detected during decoding escapes --> all escapes are invalid -------------------------------------------------------------------------------------------- */ static void rvlcDecodeEscapes(CErRvlcInfo *pRvlc, SHORT *pEsc, HANDLE_FDK_BITSTREAM bs) { SCHAR escWord; SCHAR escCnt = 0; SHORT *pEscBitCntSum; pEscBitCntSum = &(pRvlc->length_of_rvlc_escapes); /* Decode all RVLC-Escape words with a plain Huffman-Decoder */ while (*pEscBitCntSum > 0) { escWord = rvlcDecodeEscapeWord(pRvlc, bs); if (escWord >= 0) { pEsc[escCnt] = escWord; escCnt++; } else { pRvlc->errorLogRvlc |= RVLC_ERROR_ALL_ESCAPE_WORDS_INVALID; pRvlc->numDecodedEscapeWordsEsc = escCnt; return; } } /* all RVLC escapes decoded */ pRvlc->numDecodedEscapeWordsEsc = escCnt; } /*--------------------------------------------------------------------------------------------- function: decodeRVLCodeword description: Decodes a RVL-coded dpcm-word (-part). ----------------------------------------------------------------------------------------------- input: - FDK bitstream handle - pointer rvlc structure ----------------------------------------------------------------------------------------------- return: - a dpcm value which is within range [0,1,..,14] in case of no errors. The offset of 7 must be subtracted to get a valid dpcm scalefactor value. In case of errors a forbidden codeword is detected --> returning -1 -------------------------------------------------------------------------------------------- */ SCHAR decodeRVLCodeword(HANDLE_FDK_BITSTREAM bs, CErRvlcInfo *pRvlc) { int i; SCHAR value; UCHAR carryBit; UINT branchValue; UINT branchNode; const UINT *pRvlCodeTree = pRvlc->pHuffTreeRvlCodewds; UCHAR direction = pRvlc->direction; INT *pBitstrIndxRvl = pRvlc->pBitstrIndxRvl_RVL; UINT treeNode = *pRvlCodeTree; for (i = MAX_LEN_RVLC_CODE_WORD - 1; i >= 0; i--) { carryBit = rvlcReadBitFromBitstream(bs, /* get next bit */ pRvlc->bsAnchor, pBitstrIndxRvl, direction); CarryBitToBranchValue(carryBit, /* huffman decoding, do a single step in huffman decoding tree */ treeNode, &branchValue, &branchNode); if ((branchNode & TEST_BIT_10) == TEST_BIT_10) { /* test bit 10 ; if set --> a RVLC-codeword is completely decoded */ value = (SCHAR)(branchNode & CLR_BIT_10); *pRvlc->pRvlBitCnt_RVL -= (MAX_LEN_RVLC_CODE_WORD - i); /* check available bits for decoding */ if (*pRvlc->pRvlBitCnt_RVL < 0) { if (direction == FWD) { pRvlc->errorLogRvlc |= RVLC_ERROR_RVL_SUM_BIT_COUNTER_BELOW_ZERO_FWD; } else { pRvlc->errorLogRvlc |= RVLC_ERROR_RVL_SUM_BIT_COUNTER_BELOW_ZERO_BWD; } value = -1; /* signalize an error in return value, because too many bits was decoded */ } /* check max value of dpcm value */ if (value > MAX_ALLOWED_DPCM_INDEX) { if (direction == FWD) { pRvlc->errorLogRvlc |= RVLC_ERROR_FORBIDDEN_CW_DETECTED_FWD; } else { pRvlc->errorLogRvlc |= RVLC_ERROR_FORBIDDEN_CW_DETECTED_BWD; } value = -1; /* signalize an error in return value, because a forbidden cw was detected*/ } return value; /* return a dpcm value with offset +7 or an error status */ } else { treeNode = *( pRvlCodeTree + branchValue); /* update treeNode for further step in decoding tree */ } } return -1; } /*--------------------------------------------------------------------------------------------- function: rvlcDecodeForward description: Decode RVL-coded codewords in forward direction. ----------------------------------------------------------------------------------------------- input: - pointer rvlc structure - pointer channel info structure - handle to FDK bitstream ----------------------------------------------------------------------------------------------- return: - -------------------------------------------------------------------------------------------- */ static void rvlcDecodeForward(CErRvlcInfo *pRvlc, CAacDecoderChannelInfo *pAacDecoderChannelInfo, HANDLE_FDK_BITSTREAM bs) { int band = 0; int group = 0; int bnds = 0; SHORT dpcm; SHORT factor = pAacDecoderChannelInfo->pDynData->RawDataInfo.GlobalGain - SF_OFFSET; SHORT position = -SF_OFFSET; SHORT noisenrg = pAacDecoderChannelInfo->pDynData->RawDataInfo.GlobalGain - SF_OFFSET - 90 - 256; SHORT *pScfFwd = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd; SHORT *pScfEsc = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfEsc; UCHAR *pEscFwdCnt = &(pRvlc->numDecodedEscapeWordsFwd); pRvlc->pRvlBitCnt_RVL = &(pRvlc->length_of_rvlc_sf_fwd); pRvlc->pBitstrIndxRvl_RVL = &(pRvlc->bitstreamIndexRvlFwd); *pEscFwdCnt = 0; pRvlc->direction = FWD; pRvlc->noise_used = 0; pRvlc->sf_used = 0; pRvlc->lastScf = 0; pRvlc->lastNrg = 0; pRvlc->lastIs = 0; rvlcCheckIntensityCb(pRvlc, pAacDecoderChannelInfo); /* main loop fwd long */ for (group = 0; group < pRvlc->numWindowGroups; group++) { for (band = 0; band < pRvlc->maxSfbTransmitted; band++) { bnds = 16 * group + band; switch (pAacDecoderChannelInfo->pDynData->aCodeBook[bnds]) { case ZERO_HCB: pScfFwd[bnds] = 0; break; case INTENSITY_HCB2: case INTENSITY_HCB: /* store dpcm_is_position */ dpcm = decodeRVLCodeword(bs, pRvlc); if (dpcm < 0) { pRvlc->conceal_max = bnds; return; } dpcm -= TABLE_OFFSET; if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) { if (pRvlc->length_of_rvlc_escapes) { pRvlc->conceal_max = bnds; return; } else { if (dpcm == MIN_RVL) { dpcm -= *pScfEsc++; } else { dpcm += *pScfEsc++; } (*pEscFwdCnt)++; if (pRvlc->conceal_max_esc == CONCEAL_MAX_INIT) { pRvlc->conceal_max_esc = bnds; } } } position += dpcm; pScfFwd[bnds] = position; pRvlc->lastIs = position; break; case NOISE_HCB: if (pRvlc->noise_used == 0) { pRvlc->noise_used = 1; pRvlc->first_noise_band = bnds; noisenrg += pRvlc->dpcm_noise_nrg; pScfFwd[bnds] = 100 + noisenrg; pRvlc->lastNrg = noisenrg; } else { dpcm = decodeRVLCodeword(bs, pRvlc); if (dpcm < 0) { pRvlc->conceal_max = bnds; return; } dpcm -= TABLE_OFFSET; if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) { if (pRvlc->length_of_rvlc_escapes) { pRvlc->conceal_max = bnds; return; } else { if (dpcm == MIN_RVL) { dpcm -= *pScfEsc++; } else { dpcm += *pScfEsc++; } (*pEscFwdCnt)++; if (pRvlc->conceal_max_esc == CONCEAL_MAX_INIT) { pRvlc->conceal_max_esc = bnds; } } } noisenrg += dpcm; pScfFwd[bnds] = 100 + noisenrg; pRvlc->lastNrg = noisenrg; } pAacDecoderChannelInfo->data.aac.PnsData.pnsUsed[bnds] = 1; break; default: pRvlc->sf_used = 1; dpcm = decodeRVLCodeword(bs, pRvlc); if (dpcm < 0) { pRvlc->conceal_max = bnds; return; } dpcm -= TABLE_OFFSET; if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) { if (pRvlc->length_of_rvlc_escapes) { pRvlc->conceal_max = bnds; return; } else { if (dpcm == MIN_RVL) { dpcm -= *pScfEsc++; } else { dpcm += *pScfEsc++; } (*pEscFwdCnt)++; if (pRvlc->conceal_max_esc == CONCEAL_MAX_INIT) { pRvlc->conceal_max_esc = bnds; } } } factor += dpcm; pScfFwd[bnds] = factor; pRvlc->lastScf = factor; break; } } } /* postfetch fwd long */ if (pRvlc->intensity_used) { dpcm = decodeRVLCodeword(bs, pRvlc); /* dpcm_is_last_position */ if (dpcm < 0) { pRvlc->conceal_max = bnds; return; } dpcm -= TABLE_OFFSET; if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) { if (pRvlc->length_of_rvlc_escapes) { pRvlc->conceal_max = bnds; return; } else { if (dpcm == MIN_RVL) { dpcm -= *pScfEsc++; } else { dpcm += *pScfEsc++; } (*pEscFwdCnt)++; if (pRvlc->conceal_max_esc == CONCEAL_MAX_INIT) { pRvlc->conceal_max_esc = bnds; } } } pRvlc->dpcm_is_last_position = dpcm; } } /*--------------------------------------------------------------------------------------------- function: rvlcDecodeBackward description: Decode RVL-coded codewords in backward direction. ----------------------------------------------------------------------------------------------- input: - pointer rvlc structure - pointer channel info structure - handle FDK bitstream ----------------------------------------------------------------------------------------------- return: - -------------------------------------------------------------------------------------------- */ static void rvlcDecodeBackward(CErRvlcInfo *pRvlc, CAacDecoderChannelInfo *pAacDecoderChannelInfo, HANDLE_FDK_BITSTREAM bs) { SHORT band, group, dpcm, offset; SHORT bnds = pRvlc->maxSfbTransmitted - 1; SHORT factor = pRvlc->rev_global_gain - SF_OFFSET; SHORT position = pRvlc->dpcm_is_last_position - SF_OFFSET; SHORT noisenrg = pRvlc->rev_global_gain + pRvlc->dpcm_noise_last_position - SF_OFFSET - 90 - 256; SHORT *pScfBwd = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd; SHORT *pScfEsc = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfEsc; UCHAR *pEscEscCnt = &(pRvlc->numDecodedEscapeWordsEsc); UCHAR *pEscBwdCnt = &(pRvlc->numDecodedEscapeWordsBwd); pRvlc->pRvlBitCnt_RVL = &(pRvlc->length_of_rvlc_sf_bwd); pRvlc->pBitstrIndxRvl_RVL = &(pRvlc->bitstreamIndexRvlBwd); *pEscBwdCnt = 0; pRvlc->direction = BWD; pScfEsc += *pEscEscCnt - 1; /* set pScfEsc to last entry */ pRvlc->firstScf = 0; pRvlc->firstNrg = 0; pRvlc->firstIs = 0; /* prefetch long BWD */ if (pRvlc->intensity_used) { dpcm = decodeRVLCodeword(bs, pRvlc); /* dpcm_is_last_position */ if (dpcm < 0) { pRvlc->dpcm_is_last_position = 0; pRvlc->conceal_min = bnds; return; } dpcm -= TABLE_OFFSET; if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) { if (pRvlc->length_of_rvlc_escapes) { pRvlc->conceal_min = bnds; return; } else { if (dpcm == MIN_RVL) { dpcm -= *pScfEsc--; } else { dpcm += *pScfEsc--; } (*pEscBwdCnt)++; if (pRvlc->conceal_min_esc == CONCEAL_MIN_INIT) { pRvlc->conceal_min_esc = bnds; } } } pRvlc->dpcm_is_last_position = dpcm; } /* main loop long BWD */ for (group = pRvlc->numWindowGroups - 1; group >= 0; group--) { for (band = pRvlc->maxSfbTransmitted - 1; band >= 0; band--) { bnds = 16 * group + band; if ((band == 0) && (pRvlc->numWindowGroups != 1)) offset = 16 - pRvlc->maxSfbTransmitted + 1; else offset = 1; switch (pAacDecoderChannelInfo->pDynData->aCodeBook[bnds]) { case ZERO_HCB: pScfBwd[bnds] = 0; break; case INTENSITY_HCB2: case INTENSITY_HCB: /* store dpcm_is_position */ dpcm = decodeRVLCodeword(bs, pRvlc); if (dpcm < 0) { pScfBwd[bnds] = position; pRvlc->conceal_min = fMax(0, bnds - offset); return; } dpcm -= TABLE_OFFSET; if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) { if (pRvlc->length_of_rvlc_escapes) { pScfBwd[bnds] = position; pRvlc->conceal_min = fMax(0, bnds - offset); return; } else { if (dpcm == MIN_RVL) { dpcm -= *pScfEsc--; } else { dpcm += *pScfEsc--; } (*pEscBwdCnt)++; if (pRvlc->conceal_min_esc == CONCEAL_MIN_INIT) { pRvlc->conceal_min_esc = fMax(0, bnds - offset); } } } pScfBwd[bnds] = position; position -= dpcm; pRvlc->firstIs = position; break; case NOISE_HCB: if (bnds == pRvlc->first_noise_band) { pScfBwd[bnds] = pRvlc->dpcm_noise_nrg + pAacDecoderChannelInfo->pDynData->RawDataInfo.GlobalGain - SF_OFFSET - 90 - 256; pRvlc->firstNrg = pScfBwd[bnds]; } else { dpcm = decodeRVLCodeword(bs, pRvlc); if (dpcm < 0) { pScfBwd[bnds] = noisenrg; pRvlc->conceal_min = fMax(0, bnds - offset); return; } dpcm -= TABLE_OFFSET; if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) { if (pRvlc->length_of_rvlc_escapes) { pScfBwd[bnds] = noisenrg; pRvlc->conceal_min = fMax(0, bnds - offset); return; } else { if (dpcm == MIN_RVL) { dpcm -= *pScfEsc--; } else { dpcm += *pScfEsc--; } (*pEscBwdCnt)++; if (pRvlc->conceal_min_esc == CONCEAL_MIN_INIT) { pRvlc->conceal_min_esc = fMax(0, bnds - offset); } } } pScfBwd[bnds] = noisenrg; noisenrg -= dpcm; pRvlc->firstNrg = noisenrg; } break; default: dpcm = decodeRVLCodeword(bs, pRvlc); if (dpcm < 0) { pScfBwd[bnds] = factor; pRvlc->conceal_min = fMax(0, bnds - offset); return; } dpcm -= TABLE_OFFSET; if ((dpcm == MIN_RVL) || (dpcm == MAX_RVL)) { if (pRvlc->length_of_rvlc_escapes) { pScfBwd[bnds] = factor; pRvlc->conceal_min = fMax(0, bnds - offset); return; } else { if (dpcm == MIN_RVL) { dpcm -= *pScfEsc--; } else { dpcm += *pScfEsc--; } (*pEscBwdCnt)++; if (pRvlc->conceal_min_esc == CONCEAL_MIN_INIT) { pRvlc->conceal_min_esc = fMax(0, bnds - offset); } } } pScfBwd[bnds] = factor; factor -= dpcm; pRvlc->firstScf = factor; break; } } } } /*--------------------------------------------------------------------------------------------- function: rvlcFinalErrorDetection description: Call RVLC concealment if error was detected in decoding process ----------------------------------------------------------------------------------------------- input: - pointer rvlc structure - pointer channel info structure ----------------------------------------------------------------------------------------------- return: - -------------------------------------------------------------------------------------------- */ static void rvlcFinalErrorDetection( CAacDecoderChannelInfo *pAacDecoderChannelInfo, CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo) { CErRvlcInfo *pRvlc = &pAacDecoderChannelInfo->pComData->overlay.aac.erRvlcInfo; UCHAR ErrorStatusComplete = 0; UCHAR ErrorStatusLengthFwd = 0; UCHAR ErrorStatusLengthBwd = 0; UCHAR ErrorStatusLengthEscapes = 0; UCHAR ErrorStatusFirstScf = 0; UCHAR ErrorStatusLastScf = 0; UCHAR ErrorStatusFirstNrg = 0; UCHAR ErrorStatusLastNrg = 0; UCHAR ErrorStatusFirstIs = 0; UCHAR ErrorStatusLastIs = 0; UCHAR ErrorStatusForbiddenCwFwd = 0; UCHAR ErrorStatusForbiddenCwBwd = 0; UCHAR ErrorStatusNumEscapesFwd = 0; UCHAR ErrorStatusNumEscapesBwd = 0; UCHAR ConcealStatus = 1; UCHAR currentBlockType; /* short: 0, not short: 1*/ pAacDecoderChannelInfo->pDynData->specificTo.aac.rvlcCurrentScaleFactorOK = 1; /* invalid escape words, bit counter unequal zero, forbidden codeword detected */ if (pRvlc->errorLogRvlc & RVLC_ERROR_FORBIDDEN_CW_DETECTED_FWD) ErrorStatusForbiddenCwFwd = 1; if (pRvlc->errorLogRvlc & RVLC_ERROR_FORBIDDEN_CW_DETECTED_BWD) ErrorStatusForbiddenCwBwd = 1; /* bit counter forward unequal zero */ if (pRvlc->length_of_rvlc_sf_fwd) ErrorStatusLengthFwd = 1; /* bit counter backward unequal zero */ if (pRvlc->length_of_rvlc_sf_bwd) ErrorStatusLengthBwd = 1; /* bit counter escape sequences unequal zero */ if (pRvlc->sf_escapes_present) if (pRvlc->length_of_rvlc_escapes) ErrorStatusLengthEscapes = 1; if (pRvlc->sf_used) { /* first decoded scf does not match to global gain in backward direction */ if (pRvlc->firstScf != (pAacDecoderChannelInfo->pDynData->RawDataInfo.GlobalGain - SF_OFFSET)) ErrorStatusFirstScf = 1; /* last decoded scf does not match to rev global gain in forward direction */ if (pRvlc->lastScf != (pRvlc->rev_global_gain - SF_OFFSET)) ErrorStatusLastScf = 1; } if (pRvlc->noise_used) { /* first decoded nrg does not match to dpcm_noise_nrg in backward direction */ if (pRvlc->firstNrg != (pAacDecoderChannelInfo->pDynData->RawDataInfo.GlobalGain + pRvlc->dpcm_noise_nrg - SF_OFFSET - 90 - 256)) ErrorStatusFirstNrg = 1; /* last decoded nrg does not match to dpcm_noise_last_position in forward * direction */ if (pRvlc->lastNrg != (pRvlc->rev_global_gain + pRvlc->dpcm_noise_last_position - SF_OFFSET - 90 - 256)) ErrorStatusLastNrg = 1; } if (pRvlc->intensity_used) { /* first decoded is position does not match in backward direction */ if (pRvlc->firstIs != (-SF_OFFSET)) ErrorStatusFirstIs = 1; /* last decoded is position does not match in forward direction */ if (pRvlc->lastIs != (pRvlc->dpcm_is_last_position - SF_OFFSET)) ErrorStatusLastIs = 1; } /* decoded escapes and used escapes in forward direction do not fit */ if ((pRvlc->numDecodedEscapeWordsFwd != pRvlc->numDecodedEscapeWordsEsc) && (pRvlc->conceal_max == CONCEAL_MAX_INIT)) { ErrorStatusNumEscapesFwd = 1; } /* decoded escapes and used escapes in backward direction do not fit */ if ((pRvlc->numDecodedEscapeWordsBwd != pRvlc->numDecodedEscapeWordsEsc) && (pRvlc->conceal_min == CONCEAL_MIN_INIT)) { ErrorStatusNumEscapesBwd = 1; } if (ErrorStatusLengthEscapes || (((pRvlc->conceal_max == CONCEAL_MAX_INIT) && (pRvlc->numDecodedEscapeWordsFwd != pRvlc->numDecodedEscapeWordsEsc) && (ErrorStatusLastScf || ErrorStatusLastNrg || ErrorStatusLastIs)) && ((pRvlc->conceal_min == CONCEAL_MIN_INIT) && (pRvlc->numDecodedEscapeWordsBwd != pRvlc->numDecodedEscapeWordsEsc) && (ErrorStatusFirstScf || ErrorStatusFirstNrg || ErrorStatusFirstIs))) || ((pRvlc->conceal_max == CONCEAL_MAX_INIT) && ((pRvlc->rev_global_gain - SF_OFFSET - pRvlc->lastScf) < -15)) || ((pRvlc->conceal_min == CONCEAL_MIN_INIT) && ((pAacDecoderChannelInfo->pDynData->RawDataInfo.GlobalGain - SF_OFFSET - pRvlc->firstScf) < -15))) { if ((pRvlc->conceal_max == CONCEAL_MAX_INIT) || (pRvlc->conceal_min == CONCEAL_MIN_INIT)) { pRvlc->conceal_max = 0; pRvlc->conceal_min = fMax( 0, (pRvlc->numWindowGroups - 1) * 16 + pRvlc->maxSfbTransmitted - 1); } else { pRvlc->conceal_max = fMin(pRvlc->conceal_max, pRvlc->conceal_max_esc); pRvlc->conceal_min = fMax(pRvlc->conceal_min, pRvlc->conceal_min_esc); } } ErrorStatusComplete = ErrorStatusLastScf || ErrorStatusFirstScf || ErrorStatusLastNrg || ErrorStatusFirstNrg || ErrorStatusLastIs || ErrorStatusFirstIs || ErrorStatusForbiddenCwFwd || ErrorStatusForbiddenCwBwd || ErrorStatusLengthFwd || ErrorStatusLengthBwd || ErrorStatusLengthEscapes || ErrorStatusNumEscapesFwd || ErrorStatusNumEscapesBwd; currentBlockType = (GetWindowSequence(&pAacDecoderChannelInfo->icsInfo) == BLOCK_SHORT) ? 0 : 1; if (!ErrorStatusComplete) { int band; int group; int bnds; int lastSfbIndex; lastSfbIndex = (pRvlc->numWindowGroups > 1) ? 16 : 64; for (group = 0; group < pRvlc->numWindowGroups; group++) { for (band = 0; band < pRvlc->maxSfbTransmitted; band++) { bnds = 16 * group + band; pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = pAacDecoderStaticChannelInfo->concealmentInfo .aRvlcPreviousScaleFactor[bnds] = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds]; } } for (group = 0; group < pRvlc->numWindowGroups; group++) { for (band = 0; band < pRvlc->maxSfbTransmitted; band++) { bnds = 16 * group + band; pAacDecoderStaticChannelInfo->concealmentInfo .aRvlcPreviousCodebook[bnds] = pAacDecoderChannelInfo->pDynData->aCodeBook[bnds]; } for (; band < lastSfbIndex; band++) { bnds = 16 * group + band; FDK_ASSERT(bnds >= 0 && bnds < RVLC_MAX_SFB); pAacDecoderStaticChannelInfo->concealmentInfo .aRvlcPreviousCodebook[bnds] = ZERO_HCB; } } } else { int band; int group; /* A single bit error was detected in decoding of dpcm values. It also could be an error with more bits in decoding of escapes and dpcm values whereby an illegal codeword followed not directly after the corrupted bits but just after decoding some more (wrong) scalefactors. Use the smaller scalefactor from forward decoding, backward decoding and previous frame. */ if (((pRvlc->conceal_min != CONCEAL_MIN_INIT) || (pRvlc->conceal_max != CONCEAL_MAX_INIT)) && (pRvlc->conceal_min <= pRvlc->conceal_max) && (pAacDecoderStaticChannelInfo->concealmentInfo.rvlcPreviousBlockType == currentBlockType) && pAacDecoderStaticChannelInfo->concealmentInfo .rvlcPreviousScaleFactorOK && pRvlc->sf_concealment && ConcealStatus) { BidirectionalEstimation_UseScfOfPrevFrameAsReference( pAacDecoderChannelInfo, pAacDecoderStaticChannelInfo); ConcealStatus = 0; } /* A single bit error was detected in decoding of dpcm values. It also could be an error with more bits in decoding of escapes and dpcm values whereby an illegal codeword followed not directly after the corrupted bits but just after decoding some more (wrong) scalefactors. Use the smaller scalefactor from forward and backward decoding. */ if ((pRvlc->conceal_min <= pRvlc->conceal_max) && ((pRvlc->conceal_min != CONCEAL_MIN_INIT) || (pRvlc->conceal_max != CONCEAL_MAX_INIT)) && !(pAacDecoderStaticChannelInfo->concealmentInfo .rvlcPreviousScaleFactorOK && pRvlc->sf_concealment && (pAacDecoderStaticChannelInfo->concealmentInfo .rvlcPreviousBlockType == currentBlockType)) && ConcealStatus) { BidirectionalEstimation_UseLowerScfOfCurrentFrame(pAacDecoderChannelInfo); ConcealStatus = 0; } /* No errors were detected in decoding of escapes and dpcm values however the first and last value of a group (is,nrg,sf) is incorrect */ if ((pRvlc->conceal_min <= pRvlc->conceal_max) && ((ErrorStatusLastScf && ErrorStatusFirstScf) || (ErrorStatusLastNrg && ErrorStatusFirstNrg) || (ErrorStatusLastIs && ErrorStatusFirstIs)) && !(ErrorStatusForbiddenCwFwd || ErrorStatusForbiddenCwBwd || ErrorStatusLengthEscapes) && ConcealStatus) { StatisticalEstimation(pAacDecoderChannelInfo); ConcealStatus = 0; } /* A error with more bits in decoding of escapes and dpcm values was detected. Use the smaller scalefactor from forward decoding, backward decoding and previous frame. */ if ((pRvlc->conceal_min <= pRvlc->conceal_max) && pAacDecoderStaticChannelInfo->concealmentInfo .rvlcPreviousScaleFactorOK && pRvlc->sf_concealment && (pAacDecoderStaticChannelInfo->concealmentInfo.rvlcPreviousBlockType == currentBlockType) && ConcealStatus) { PredictiveInterpolation(pAacDecoderChannelInfo, pAacDecoderStaticChannelInfo); ConcealStatus = 0; } /* Call frame concealment, because no better strategy was found. Setting the scalefactors to zero is done for debugging purposes */ if (ConcealStatus) { for (group = 0; group < pRvlc->numWindowGroups; group++) { for (band = 0; band < pRvlc->maxSfbTransmitted; band++) { pAacDecoderChannelInfo->pDynData->aScaleFactor[16 * group + band] = 0; } } pAacDecoderChannelInfo->pDynData->specificTo.aac .rvlcCurrentScaleFactorOK = 0; } } } /*--------------------------------------------------------------------------------------------- function: CRvlc_Read description: Read RVLC ESC1 data (side info) from bitstream. ----------------------------------------------------------------------------------------------- input: - pointer rvlc structure - pointer channel info structure - pointer bitstream structure ----------------------------------------------------------------------------------------------- return: - -------------------------------------------------------------------------------------------- */ void CRvlc_Read(CAacDecoderChannelInfo *pAacDecoderChannelInfo, HANDLE_FDK_BITSTREAM bs) { CErRvlcInfo *pRvlc = &pAacDecoderChannelInfo->pComData->overlay.aac.erRvlcInfo; int group, band; /* RVLC long specific initialization Init part 1 of 2 */ pRvlc->numWindowGroups = GetWindowGroups(&pAacDecoderChannelInfo->icsInfo); pRvlc->maxSfbTransmitted = GetScaleFactorBandsTransmitted(&pAacDecoderChannelInfo->icsInfo); pRvlc->noise_used = 0; /* noise detection */ pRvlc->dpcm_noise_nrg = 0; /* only for debugging */ pRvlc->dpcm_noise_last_position = 0; /* only for debugging */ pRvlc->length_of_rvlc_escapes = -1; /* default value is used for error detection and concealment */ /* read only error sensitivity class 1 data (ESC 1 - data) */ pRvlc->sf_concealment = FDKreadBits(bs, 1); /* #1 */ pRvlc->rev_global_gain = FDKreadBits(bs, 8); /* #2 */ if (GetWindowSequence(&pAacDecoderChannelInfo->icsInfo) == BLOCK_SHORT) { pRvlc->length_of_rvlc_sf = FDKreadBits(bs, 11); /* #3 */ } else { pRvlc->length_of_rvlc_sf = FDKreadBits(bs, 9); /* #3 */ } /* check if noise codebook is used */ for (group = 0; group < pRvlc->numWindowGroups; group++) { for (band = 0; band < pRvlc->maxSfbTransmitted; band++) { if (pAacDecoderChannelInfo->pDynData->aCodeBook[16 * group + band] == NOISE_HCB) { pRvlc->noise_used = 1; break; } } } if (pRvlc->noise_used) pRvlc->dpcm_noise_nrg = FDKreadBits(bs, 9); /* #4 PNS */ pRvlc->sf_escapes_present = FDKreadBits(bs, 1); /* #5 */ if (pRvlc->sf_escapes_present) { pRvlc->length_of_rvlc_escapes = FDKreadBits(bs, 8); /* #6 */ } if (pRvlc->noise_used) { pRvlc->dpcm_noise_last_position = FDKreadBits(bs, 9); /* #7 PNS */ pRvlc->length_of_rvlc_sf -= 9; } pRvlc->length_of_rvlc_sf_fwd = pRvlc->length_of_rvlc_sf; pRvlc->length_of_rvlc_sf_bwd = pRvlc->length_of_rvlc_sf; } /*--------------------------------------------------------------------------------------------- function: CRvlc_Decode description: Decode rvlc data The function reads both the escape sequences and the scalefactors in forward and backward direction. If an error occured during decoding process which can not be concealed with the rvlc concealment frame concealment will be initiated. Then the element "rvlcCurrentScaleFactorOK" in the decoder channel info is set to 0 otherwise it is set to 1. ----------------------------------------------------------------------------------------------- input: - pointer rvlc structure - pointer channel info structure - pointer to persistent channel info structure - pointer bitstream structure ----------------------------------------------------------------------------------------------- return: ErrorStatus = AAC_DEC_OK -------------------------------------------------------------------------------------------- */ void CRvlc_Decode(CAacDecoderChannelInfo *pAacDecoderChannelInfo, CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, HANDLE_FDK_BITSTREAM bs) { CErRvlcInfo *pRvlc = &pAacDecoderChannelInfo->pComData->overlay.aac.erRvlcInfo; INT bitCntOffst; INT saveBitCnt; rvlcInit(pRvlc, pAacDecoderChannelInfo, bs); /* save bitstream position */ saveBitCnt = (INT)FDKgetValidBits(bs); if (pRvlc->sf_escapes_present) rvlcDecodeEscapes( pRvlc, pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfEsc, bs); rvlcDecodeForward(pRvlc, pAacDecoderChannelInfo, bs); rvlcDecodeBackward(pRvlc, pAacDecoderChannelInfo, bs); rvlcFinalErrorDetection(pAacDecoderChannelInfo, pAacDecoderStaticChannelInfo); pAacDecoderChannelInfo->pDynData->specificTo.aac.rvlcIntensityUsed = pRvlc->intensity_used; pAacDecoderChannelInfo->data.aac.PnsData.PnsActive = pRvlc->noise_used; /* restore bitstream position */ bitCntOffst = (INT)FDKgetValidBits(bs) - saveBitCnt; if (bitCntOffst) { FDKpushBiDirectional(bs, bitCntOffst); } } void CRvlc_ElementCheck( CAacDecoderChannelInfo *pAacDecoderChannelInfo[], CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo[], const UINT flags, const INT elChannels) { int ch; /* Required for MPS residuals. */ if (pAacDecoderStaticChannelInfo == NULL) { return; } /* RVLC specific sanity checks */ if ((flags & AC_ER_RVLC) && (elChannels == 2)) { /* to be reviewed */ if (((pAacDecoderChannelInfo[0] ->pDynData->specificTo.aac.rvlcCurrentScaleFactorOK == 0) || (pAacDecoderChannelInfo[1] ->pDynData->specificTo.aac.rvlcCurrentScaleFactorOK == 0)) && pAacDecoderChannelInfo[0]->pComData->jointStereoData.MsMaskPresent) { pAacDecoderChannelInfo[0] ->pDynData->specificTo.aac.rvlcCurrentScaleFactorOK = 0; pAacDecoderChannelInfo[1] ->pDynData->specificTo.aac.rvlcCurrentScaleFactorOK = 0; } if ((pAacDecoderChannelInfo[0] ->pDynData->specificTo.aac.rvlcCurrentScaleFactorOK == 0) && (pAacDecoderChannelInfo[1] ->pDynData->specificTo.aac.rvlcCurrentScaleFactorOK == 1) && (pAacDecoderChannelInfo[1] ->pDynData->specificTo.aac.rvlcIntensityUsed == 1)) { pAacDecoderChannelInfo[1] ->pDynData->specificTo.aac.rvlcCurrentScaleFactorOK = 0; } } for (ch = 0; ch < elChannels; ch++) { pAacDecoderStaticChannelInfo[ch]->concealmentInfo.rvlcPreviousBlockType = (GetWindowSequence(&pAacDecoderChannelInfo[ch]->icsInfo) == BLOCK_SHORT) ? 0 : 1; if (flags & AC_ER_RVLC) { pAacDecoderStaticChannelInfo[ch] ->concealmentInfo.rvlcPreviousScaleFactorOK = pAacDecoderChannelInfo[ch] ->pDynData->specificTo.aac.rvlcCurrentScaleFactorOK; } else { pAacDecoderStaticChannelInfo[ch] ->concealmentInfo.rvlcPreviousScaleFactorOK = 0; } } } fdk-aac-2.0.2/libAACdec/src/rvlc.h000066400000000000000000000153101404225310700164340ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Defines structures and prototypes for RVLC \author Robert Weidner */ #ifndef RVLC_H #define RVLC_H #include "aacdecoder.h" #include "channel.h" #include "rvlc_info.h" /* ------------------------------------------------------------------- */ /* errorLogRvlc: A word of 32 bits used for logging possible errors */ /* within RVLC in case of distorted bitstreams. */ /* ------------------------------------------------------------------- */ #define RVLC_ERROR_ALL_ESCAPE_WORDS_INVALID \ 0x80000000 /* ESC-Dec During RVLC-Escape-decoding there have been more \ bits decoded as there are available */ #define RVLC_ERROR_RVL_SUM_BIT_COUNTER_BELOW_ZERO_FWD \ 0x40000000 /* RVL-Dec negative sum-bitcounter during RVL-fwd-decoding \ (long+shrt) */ #define RVLC_ERROR_RVL_SUM_BIT_COUNTER_BELOW_ZERO_BWD \ 0x20000000 /* RVL-Dec negative sum-bitcounter during RVL-fwd-decoding \ (long+shrt) */ #define RVLC_ERROR_FORBIDDEN_CW_DETECTED_FWD \ 0x08000000 /* RVL-Dec forbidden codeword detected fwd (long+shrt) */ #define RVLC_ERROR_FORBIDDEN_CW_DETECTED_BWD \ 0x04000000 /* RVL-Dec forbidden codeword detected bwd (long+shrt) */ void CRvlc_Read(CAacDecoderChannelInfo *pAacDecoderChannelInfo, HANDLE_FDK_BITSTREAM bs); void CRvlc_Decode(CAacDecoderChannelInfo *pAacDecoderChannelInfo, CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, HANDLE_FDK_BITSTREAM bs); /** * \brief performe sanity checks to the channel data corresponding to one * channel element. * \param pAacDecoderChannelInfo * \param pAacDecoderStaticChannelInfo * \param elChannels amount of channels of the channel element. */ void CRvlc_ElementCheck( CAacDecoderChannelInfo *pAacDecoderChannelInfo[], CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo[], const UINT flags, const INT elChannels); #endif /* RVLC_H */ fdk-aac-2.0.2/libAACdec/src/rvlc_info.h000066400000000000000000000210201404225310700174420ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Defines structures for RVLC \author Robert Weidner */ #ifndef RVLC_INFO_H #define RVLC_INFO_H #define FWD 0 /* bitstream decoding direction forward (RVL coded part) */ #define BWD 1 /* bitstream decoding direction backward (RVL coded part) */ #define MAX_RVL 7 /* positive RVLC escape */ #define MIN_RVL -7 /* negative RVLC escape */ #define MAX_ALLOWED_DPCM_INDEX \ 14 /* the maximum allowed index of a decoded dpcm value (offset \ 'TABLE_OFFSET' incl --> must be subtracted) */ #define TABLE_OFFSET \ 7 /* dpcm offset of valid output values of rvl table decoding, the rvl table \ ouly returns positive values, therefore the offset */ #define MAX_LEN_RVLC_CODE_WORD 9 /* max length of a RVL codeword in bits */ #define MAX_LEN_RVLC_ESCAPE_WORD \ 20 /* max length of huffman coded RVLC escape word in bits */ #define DPCM_NOISE_NRG_BITS 9 #define SF_OFFSET 100 /* offset for correcting scf value */ #define CONCEAL_MAX_INIT 1311 /* arbitrary value */ #define CONCEAL_MIN_INIT -1311 /* arbitrary value */ #define RVLC_MAX_SFB ((8) * (16)) /* sideinfo of RVLC */ typedef struct { /* ------- ESC 1 Data: --------- */ /* order of RVLC-bitstream components in bitstream (RVLC-initialization), every component appears only once in bitstream */ INT sf_concealment; /* 1 */ INT rev_global_gain; /* 2 */ SHORT length_of_rvlc_sf; /* 3 */ /* original value, gets modified (subtract 9) in case of noise (PNS); is kept for later use */ INT dpcm_noise_nrg; /* 4 optional */ INT sf_escapes_present; /* 5 */ SHORT length_of_rvlc_escapes; /* 6 optional */ INT dpcm_noise_last_position; /* 7 optional */ INT dpcm_is_last_position; SHORT length_of_rvlc_sf_fwd; /* length_of_rvlc_sf used for forward decoding */ SHORT length_of_rvlc_sf_bwd; /* length_of_rvlc_sf used for backward decoding */ /* for RVL-Codeword decoder to distinguish between fwd and bwd decoding */ SHORT *pRvlBitCnt_RVL; INT *pBitstrIndxRvl_RVL; UCHAR numWindowGroups; UCHAR maxSfbTransmitted; UCHAR first_noise_group; UCHAR first_noise_band; UCHAR direction; /* bitstream indices */ INT bsAnchor; /* hcr bit buffer reference index */ INT bitstreamIndexRvlFwd; /* base address of RVL-coded-scalefactor data (ESC 2) for forward decoding */ INT bitstreamIndexRvlBwd; /* base address of RVL-coded-scalefactor data (ESC 2) for backward decoding */ INT bitstreamIndexEsc; /* base address where RVLC-escapes start (ESC 2) */ /* decoding trees */ const UINT *pHuffTreeRvlCodewds; const UINT *pHuffTreeRvlcEscape; /* escape counters */ UCHAR numDecodedEscapeWordsFwd; /* when decoding RVL-codes forward */ UCHAR numDecodedEscapeWordsBwd; /* when decoding RVL-codes backward */ UCHAR numDecodedEscapeWordsEsc; /* when decoding the escape-Words */ SCHAR noise_used; SCHAR intensity_used; SCHAR sf_used; SHORT firstScf; SHORT lastScf; SHORT firstNrg; SHORT lastNrg; SHORT firstIs; SHORT lastIs; /* ------ RVLC error detection ------ */ UINT errorLogRvlc; /* store RVLC errors */ SHORT conceal_min; /* is set at backward decoding */ SHORT conceal_max; /* is set at forward decoding */ SHORT conceal_min_esc; /* is set at backward decoding */ SHORT conceal_max_esc; /* is set at forward decoding */ } CErRvlcInfo; typedef CErRvlcInfo RVLC_INFO; /* temp */ #endif /* RVLC_INFO_H */ fdk-aac-2.0.2/libAACdec/src/rvlcbit.cpp000066400000000000000000000140671404225310700174760ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief RVLC bitstream reading \author Robert Weidner */ #include "rvlcbit.h" /*--------------------------------------------------------------------------------------------- function: rvlcReadBitFromBitstream description: This function returns a bit from the bitstream according to read direction. It is called very often, therefore it makes sense to inline it (runtime). ----------------------------------------------------------------------------------------------- input: - bitstream - pPosition - readDirection ----------------------------------------------------------------------------------------------- return: - bit from bitstream -------------------------------------------------------------------------------------------- */ UCHAR rvlcReadBitFromBitstream(HANDLE_FDK_BITSTREAM bs, const INT bsAnchor, INT *pPosition, UCHAR readDirection) { UINT bit; INT readBitOffset = (INT)FDKgetValidBits(bs) - bsAnchor + *pPosition; if (readBitOffset) { FDKpushBiDirectional(bs, readBitOffset); } if (readDirection == FWD) { bit = FDKreadBits(bs, 1); *pPosition += 1; } else { /* to be replaced with a brother function of FDKreadBits() */ bit = FDKreadBits(bs, 1); FDKpushBack(bs, 2); *pPosition -= 1; } return (bit); } fdk-aac-2.0.2/libAACdec/src/rvlcbit.h000066400000000000000000000117471404225310700171450ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Robert Weidner (DSP Solutions) Description: RVLC Decoder: Bitstream reading *******************************************************************************/ #ifndef RVLCBIT_H #define RVLCBIT_H #include "rvlc.h" UCHAR rvlcReadBitFromBitstream(HANDLE_FDK_BITSTREAM bs, const INT bsAnchor, INT *pPosition, UCHAR readDirection); #endif /* RVLCBIT_H */ fdk-aac-2.0.2/libAACdec/src/rvlcconceal.cpp000066400000000000000000001021361404225310700203170ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief rvlc concealment \author Josef Hoepfl */ #include "rvlcconceal.h" #include "block.h" #include "rvlc.h" /*--------------------------------------------------------------------------------------------- function: calcRefValFwd description: The function determines the scalefactor which is closed to the scalefactorband conceal_min. The same is done for intensity data and noise energies. ----------------------------------------------------------------------------------------------- output: - reference value scf - reference value internsity data - reference value noise energy ----------------------------------------------------------------------------------------------- return: - -------------------------------------------------------------------------------------------- */ static void calcRefValFwd(CErRvlcInfo *pRvlc, CAacDecoderChannelInfo *pAacDecoderChannelInfo, int *refIsFwd, int *refNrgFwd, int *refScfFwd) { int band, bnds, group, startBand; int idIs, idNrg, idScf; int conceal_min, conceal_group_min; int MaximumScaleFactorBands; if (GetWindowSequence(&pAacDecoderChannelInfo->icsInfo) == BLOCK_SHORT) MaximumScaleFactorBands = 16; else MaximumScaleFactorBands = 64; conceal_min = pRvlc->conceal_min % MaximumScaleFactorBands; conceal_group_min = pRvlc->conceal_min / MaximumScaleFactorBands; /* calculate first reference value for approach in forward direction */ idIs = idNrg = idScf = 1; /* set reference values */ *refIsFwd = -SF_OFFSET; *refNrgFwd = pAacDecoderChannelInfo->pDynData->RawDataInfo.GlobalGain - SF_OFFSET - 90 - 256; *refScfFwd = pAacDecoderChannelInfo->pDynData->RawDataInfo.GlobalGain - SF_OFFSET; startBand = conceal_min - 1; for (group = conceal_group_min; group >= 0; group--) { for (band = startBand; band >= 0; band--) { bnds = 16 * group + band; switch (pAacDecoderChannelInfo->pDynData->aCodeBook[bnds]) { case ZERO_HCB: break; case INTENSITY_HCB: case INTENSITY_HCB2: if (idIs) { *refIsFwd = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds]; idIs = 0; /* reference value has been set */ } break; case NOISE_HCB: if (idNrg) { *refNrgFwd = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds]; idNrg = 0; /* reference value has been set */ } break; default: if (idScf) { *refScfFwd = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds]; idScf = 0; /* reference value has been set */ } break; } } startBand = pRvlc->maxSfbTransmitted - 1; } } /*--------------------------------------------------------------------------------------------- function: calcRefValBwd description: The function determines the scalefactor which is closed to the scalefactorband conceal_max. The same is done for intensity data and noise energies. ----------------------------------------------------------------------------------------------- output: - reference value scf - reference value internsity data - reference value noise energy ----------------------------------------------------------------------------------------------- return: - -------------------------------------------------------------------------------------------- */ static void calcRefValBwd(CErRvlcInfo *pRvlc, CAacDecoderChannelInfo *pAacDecoderChannelInfo, int *refIsBwd, int *refNrgBwd, int *refScfBwd) { int band, bnds, group, startBand; int idIs, idNrg, idScf; int conceal_max, conceal_group_max; int MaximumScaleFactorBands; if (GetWindowSequence(&pAacDecoderChannelInfo->icsInfo) == BLOCK_SHORT) MaximumScaleFactorBands = 16; else MaximumScaleFactorBands = 64; conceal_max = pRvlc->conceal_max % MaximumScaleFactorBands; conceal_group_max = pRvlc->conceal_max / MaximumScaleFactorBands; /* calculate first reference value for approach in backward direction */ idIs = idNrg = idScf = 1; /* set reference values */ *refIsBwd = pRvlc->dpcm_is_last_position - SF_OFFSET; *refNrgBwd = pRvlc->rev_global_gain + pRvlc->dpcm_noise_last_position - SF_OFFSET - 90 - 256 + pRvlc->dpcm_noise_nrg; *refScfBwd = pRvlc->rev_global_gain - SF_OFFSET; startBand = conceal_max + 1; /* if needed, re-set reference values */ for (group = conceal_group_max; group < pRvlc->numWindowGroups; group++) { for (band = startBand; band < pRvlc->maxSfbTransmitted; band++) { bnds = 16 * group + band; switch (pAacDecoderChannelInfo->pDynData->aCodeBook[bnds]) { case ZERO_HCB: break; case INTENSITY_HCB: case INTENSITY_HCB2: if (idIs) { *refIsBwd = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]; idIs = 0; /* reference value has been set */ } break; case NOISE_HCB: if (idNrg) { *refNrgBwd = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]; idNrg = 0; /* reference value has been set */ } break; default: if (idScf) { *refScfBwd = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]; idScf = 0; /* reference value has been set */ } break; } } startBand = 0; } } /*--------------------------------------------------------------------------------------------- function: BidirectionalEstimation_UseLowerScfOfCurrentFrame description: This approach by means of bidirectional estimation is generally performed when a single bit error has been detected, the bit error can be isolated between 'conceal_min' and 'conceal_max' and the 'sf_concealment' flag is not set. The sets of scalefactors decoded in forward and backward direction are compared with each other. The smaller scalefactor will be considered as the correct one respectively. The reconstruction of the scalefactors with this approach archieve good results in audio quality. The strategy must be applied to scalefactors, intensity data and noise energy seperately. ----------------------------------------------------------------------------------------------- output: Concealed scalefactor, noise energy and intensity data between conceal_min and conceal_max ----------------------------------------------------------------------------------------------- return: - -------------------------------------------------------------------------------------------- */ void BidirectionalEstimation_UseLowerScfOfCurrentFrame( CAacDecoderChannelInfo *pAacDecoderChannelInfo) { CErRvlcInfo *pRvlc = &pAacDecoderChannelInfo->pComData->overlay.aac.erRvlcInfo; int band, bnds, startBand, endBand, group; int conceal_min, conceal_max; int conceal_group_min, conceal_group_max; int MaximumScaleFactorBands; if (GetWindowSequence(&pAacDecoderChannelInfo->icsInfo) == BLOCK_SHORT) { MaximumScaleFactorBands = 16; } else { MaximumScaleFactorBands = 64; } /* If an error was detected just in forward or backward direction, set the corresponding border for concealment to a appropriate scalefactor band. The border is set to first or last sfb respectively, because the error will possibly not follow directly after the corrupt bit but just after decoding some more (wrong) scalefactors. */ if (pRvlc->conceal_min == CONCEAL_MIN_INIT) pRvlc->conceal_min = 0; if (pRvlc->conceal_max == CONCEAL_MAX_INIT) pRvlc->conceal_max = (pRvlc->numWindowGroups - 1) * 16 + pRvlc->maxSfbTransmitted - 1; conceal_min = pRvlc->conceal_min % MaximumScaleFactorBands; conceal_group_min = pRvlc->conceal_min / MaximumScaleFactorBands; conceal_max = pRvlc->conceal_max % MaximumScaleFactorBands; conceal_group_max = pRvlc->conceal_max / MaximumScaleFactorBands; if (pRvlc->conceal_min == pRvlc->conceal_max) { int refIsFwd, refNrgFwd, refScfFwd; int refIsBwd, refNrgBwd, refScfBwd; bnds = pRvlc->conceal_min; calcRefValFwd(pRvlc, pAacDecoderChannelInfo, &refIsFwd, &refNrgFwd, &refScfFwd); calcRefValBwd(pRvlc, pAacDecoderChannelInfo, &refIsBwd, &refNrgBwd, &refScfBwd); switch (pAacDecoderChannelInfo->pDynData->aCodeBook[bnds]) { case ZERO_HCB: break; case INTENSITY_HCB: case INTENSITY_HCB2: if (refIsFwd < refIsBwd) pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = refIsFwd; else pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = refIsBwd; break; case NOISE_HCB: if (refNrgFwd < refNrgBwd) pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = refNrgFwd; else pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = refNrgBwd; break; default: if (refScfFwd < refScfBwd) pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = refScfFwd; else pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = refScfBwd; break; } } else { pAacDecoderChannelInfo->pComData->overlay.aac .aRvlcScfFwd[pRvlc->conceal_max] = pAacDecoderChannelInfo->pComData->overlay.aac .aRvlcScfBwd[pRvlc->conceal_max]; pAacDecoderChannelInfo->pComData->overlay.aac .aRvlcScfBwd[pRvlc->conceal_min] = pAacDecoderChannelInfo->pComData->overlay.aac .aRvlcScfFwd[pRvlc->conceal_min]; /* consider the smaller of the forward and backward decoded value as the * correct one */ startBand = conceal_min; if (conceal_group_min == conceal_group_max) endBand = conceal_max; else endBand = pRvlc->maxSfbTransmitted - 1; for (group = conceal_group_min; group <= conceal_group_max; group++) { for (band = startBand; band <= endBand; band++) { bnds = 16 * group + band; if (pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds] < pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]) pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds]; else pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]; } startBand = 0; if ((group + 1) == conceal_group_max) endBand = conceal_max; } } /* now copy all data to the output buffer which needs not to be concealed */ if (conceal_group_min == 0) endBand = conceal_min; else endBand = pRvlc->maxSfbTransmitted; for (group = 0; group <= conceal_group_min; group++) { for (band = 0; band < endBand; band++) { bnds = 16 * group + band; pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds]; } if ((group + 1) == conceal_group_min) endBand = conceal_min; } startBand = conceal_max + 1; for (group = conceal_group_max; group < pRvlc->numWindowGroups; group++) { for (band = startBand; band < pRvlc->maxSfbTransmitted; band++) { bnds = 16 * group + band; pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]; } startBand = 0; } } /*--------------------------------------------------------------------------------------------- function: BidirectionalEstimation_UseScfOfPrevFrameAsReference description: This approach by means of bidirectional estimation is generally performed when a single bit error has been detected, the bit error can be isolated between 'conceal_min' and 'conceal_max', the 'sf_concealment' flag is set and the previous frame has the same block type as the current frame. The scalefactor decoded in forward and backward direction and the scalefactor of the previous frame are compared with each other. The smaller scalefactor will be considered as the correct one. At this the codebook of the previous and current frame must be of the same set (scf, nrg, is) in each scalefactorband. Otherwise the scalefactor of the previous frame is not considered in the minimum calculation. The reconstruction of the scalefactors with this approach archieve good results in audio quality. The strategy must be applied to scalefactors, intensity data and noise energy seperately. ----------------------------------------------------------------------------------------------- output: Concealed scalefactor, noise energy and intensity data between conceal_min and conceal_max ----------------------------------------------------------------------------------------------- return: - -------------------------------------------------------------------------------------------- */ void BidirectionalEstimation_UseScfOfPrevFrameAsReference( CAacDecoderChannelInfo *pAacDecoderChannelInfo, CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo) { CErRvlcInfo *pRvlc = &pAacDecoderChannelInfo->pComData->overlay.aac.erRvlcInfo; int band, bnds, startBand, endBand, group; int conceal_min, conceal_max; int conceal_group_min, conceal_group_max; int MaximumScaleFactorBands; SHORT commonMin; if (GetWindowSequence(&pAacDecoderChannelInfo->icsInfo) == BLOCK_SHORT) { MaximumScaleFactorBands = 16; } else { MaximumScaleFactorBands = 64; } /* If an error was detected just in forward or backward direction, set the corresponding border for concealment to a appropriate scalefactor band. The border is set to first or last sfb respectively, because the error will possibly not follow directly after the corrupt bit but just after decoding some more (wrong) scalefactors. */ if (pRvlc->conceal_min == CONCEAL_MIN_INIT) pRvlc->conceal_min = 0; if (pRvlc->conceal_max == CONCEAL_MAX_INIT) pRvlc->conceal_max = (pRvlc->numWindowGroups - 1) * 16 + pRvlc->maxSfbTransmitted - 1; conceal_min = pRvlc->conceal_min % MaximumScaleFactorBands; conceal_group_min = pRvlc->conceal_min / MaximumScaleFactorBands; conceal_max = pRvlc->conceal_max % MaximumScaleFactorBands; conceal_group_max = pRvlc->conceal_max / MaximumScaleFactorBands; pAacDecoderChannelInfo->pComData->overlay.aac .aRvlcScfFwd[pRvlc->conceal_max] = pAacDecoderChannelInfo->pComData->overlay.aac .aRvlcScfBwd[pRvlc->conceal_max]; pAacDecoderChannelInfo->pComData->overlay.aac .aRvlcScfBwd[pRvlc->conceal_min] = pAacDecoderChannelInfo->pComData->overlay.aac .aRvlcScfFwd[pRvlc->conceal_min]; /* consider the smaller of the forward and backward decoded value as the * correct one */ startBand = conceal_min; if (conceal_group_min == conceal_group_max) endBand = conceal_max; else endBand = pRvlc->maxSfbTransmitted - 1; for (group = conceal_group_min; group <= conceal_group_max; group++) { for (band = startBand; band <= endBand; band++) { bnds = 16 * group + band; switch (pAacDecoderChannelInfo->pDynData->aCodeBook[bnds]) { case ZERO_HCB: pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = 0; break; case INTENSITY_HCB: case INTENSITY_HCB2: if ((pAacDecoderStaticChannelInfo->concealmentInfo .aRvlcPreviousCodebook[bnds] == INTENSITY_HCB) || (pAacDecoderStaticChannelInfo->concealmentInfo .aRvlcPreviousCodebook[bnds] == INTENSITY_HCB2)) { commonMin = fMin( pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds], pAacDecoderChannelInfo->pComData->overlay.aac .aRvlcScfBwd[bnds]); pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = fMin(commonMin, pAacDecoderStaticChannelInfo->concealmentInfo .aRvlcPreviousScaleFactor[bnds]); } else { pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = fMin( pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds], pAacDecoderChannelInfo->pComData->overlay.aac .aRvlcScfBwd[bnds]); } break; case NOISE_HCB: if (pAacDecoderStaticChannelInfo->concealmentInfo .aRvlcPreviousCodebook[bnds] == NOISE_HCB) { commonMin = fMin( pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds], pAacDecoderChannelInfo->pComData->overlay.aac .aRvlcScfBwd[bnds]); pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = fMin(commonMin, pAacDecoderStaticChannelInfo->concealmentInfo .aRvlcPreviousScaleFactor[bnds]); } else { pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = fMin( pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds], pAacDecoderChannelInfo->pComData->overlay.aac .aRvlcScfBwd[bnds]); } break; default: if ((pAacDecoderStaticChannelInfo->concealmentInfo .aRvlcPreviousCodebook[bnds] != ZERO_HCB) && (pAacDecoderStaticChannelInfo->concealmentInfo .aRvlcPreviousCodebook[bnds] != NOISE_HCB) && (pAacDecoderStaticChannelInfo->concealmentInfo .aRvlcPreviousCodebook[bnds] != INTENSITY_HCB) && (pAacDecoderStaticChannelInfo->concealmentInfo .aRvlcPreviousCodebook[bnds] != INTENSITY_HCB2)) { commonMin = fMin( pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds], pAacDecoderChannelInfo->pComData->overlay.aac .aRvlcScfBwd[bnds]); pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = fMin(commonMin, pAacDecoderStaticChannelInfo->concealmentInfo .aRvlcPreviousScaleFactor[bnds]); } else { pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = fMin( pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds], pAacDecoderChannelInfo->pComData->overlay.aac .aRvlcScfBwd[bnds]); } break; } } startBand = 0; if ((group + 1) == conceal_group_max) endBand = conceal_max; } /* now copy all data to the output buffer which needs not to be concealed */ if (conceal_group_min == 0) endBand = conceal_min; else endBand = pRvlc->maxSfbTransmitted; for (group = 0; group <= conceal_group_min; group++) { for (band = 0; band < endBand; band++) { bnds = 16 * group + band; pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds]; } if ((group + 1) == conceal_group_min) endBand = conceal_min; } startBand = conceal_max + 1; for (group = conceal_group_max; group < pRvlc->numWindowGroups; group++) { for (band = startBand; band < pRvlc->maxSfbTransmitted; band++) { bnds = 16 * group + band; pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]; } startBand = 0; } } /*--------------------------------------------------------------------------------------------- function: StatisticalEstimation description: This approach by means of statistical estimation is generally performed when both the start value and the end value are different and no further errors have been detected. Considering the forward and backward decoded scalefactors, the set with the lower scalefactors in sum will be considered as the correct one. The scalefactors are differentially encoded. Normally it would reach to compare one pair of the forward and backward decoded scalefactors to specify the lower set. But having detected no further errors does not necessarily mean the absence of errors. Therefore all scalefactors decoded in forward and backward direction are summed up seperately. The set with the lower sum will be used. The strategy must be applied to scalefactors, intensity data and noise energy seperately. ----------------------------------------------------------------------------------------------- output: Concealed scalefactor, noise energy and intensity data ----------------------------------------------------------------------------------------------- return: - -------------------------------------------------------------------------------------------- */ void StatisticalEstimation(CAacDecoderChannelInfo *pAacDecoderChannelInfo) { CErRvlcInfo *pRvlc = &pAacDecoderChannelInfo->pComData->overlay.aac.erRvlcInfo; int band, bnds, group; int sumIsFwd, sumIsBwd; /* sum of intensity data forward/backward */ int sumNrgFwd, sumNrgBwd; /* sum of noise energy data forward/backward */ int sumScfFwd, sumScfBwd; /* sum of scalefactor data forward/backward */ int useIsFwd, useNrgFwd, useScfFwd; /* the flags signals the elements which are used for the final result */ sumIsFwd = sumIsBwd = sumNrgFwd = sumNrgBwd = sumScfFwd = sumScfBwd = 0; useIsFwd = useNrgFwd = useScfFwd = 0; /* calculate sum of each group (scf,nrg,is) of forward and backward direction */ for (group = 0; group < pRvlc->numWindowGroups; group++) { for (band = 0; band < pRvlc->maxSfbTransmitted; band++) { bnds = 16 * group + band; switch (pAacDecoderChannelInfo->pDynData->aCodeBook[bnds]) { case ZERO_HCB: break; case INTENSITY_HCB: case INTENSITY_HCB2: sumIsFwd += pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds]; sumIsBwd += pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]; break; case NOISE_HCB: sumNrgFwd += pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds]; sumNrgBwd += pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]; break; default: sumScfFwd += pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds]; sumScfBwd += pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]; break; } } } /* find for each group (scf,nrg,is) the correct direction */ if (sumIsFwd < sumIsBwd) useIsFwd = 1; if (sumNrgFwd < sumNrgBwd) useNrgFwd = 1; if (sumScfFwd < sumScfBwd) useScfFwd = 1; /* conceal each group (scf,nrg,is) */ for (group = 0; group < pRvlc->numWindowGroups; group++) { for (band = 0; band < pRvlc->maxSfbTransmitted; band++) { bnds = 16 * group + band; switch (pAacDecoderChannelInfo->pDynData->aCodeBook[bnds]) { case ZERO_HCB: break; case INTENSITY_HCB: case INTENSITY_HCB2: if (useIsFwd) pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds]; else pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]; break; case NOISE_HCB: if (useNrgFwd) pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds]; else pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]; break; default: if (useScfFwd) pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds]; else pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]; break; } } } } /*--------------------------------------------------------------------------------------------- description: Approach by means of predictive interpolation This approach by means of predictive estimation is generally performed when the error cannot be isolated between 'conceal_min' and 'conceal_max', the 'sf_concealment' flag is set and the previous frame has the same block type as the current frame. Check for each scalefactorband if the same type of data (scalefactor, internsity data, noise energies) is transmitted. If so use the scalefactor (intensity data, noise energy) in the current frame. Otherwise set the scalefactor (intensity data, noise energy) for this scalefactorband to zero. ----------------------------------------------------------------------------------------------- output: Concealed scalefactor, noise energy and intensity data ----------------------------------------------------------------------------------------------- return: - -------------------------------------------------------------------------------------------- */ void PredictiveInterpolation( CAacDecoderChannelInfo *pAacDecoderChannelInfo, CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo) { CErRvlcInfo *pRvlc = &pAacDecoderChannelInfo->pComData->overlay.aac.erRvlcInfo; int band, bnds, group; SHORT commonMin; for (group = 0; group < pRvlc->numWindowGroups; group++) { for (band = 0; band < pRvlc->maxSfbTransmitted; band++) { bnds = 16 * group + band; switch (pAacDecoderChannelInfo->pDynData->aCodeBook[bnds]) { case ZERO_HCB: pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = 0; break; case INTENSITY_HCB: case INTENSITY_HCB2: if ((pAacDecoderStaticChannelInfo->concealmentInfo .aRvlcPreviousCodebook[bnds] == INTENSITY_HCB) || (pAacDecoderStaticChannelInfo->concealmentInfo .aRvlcPreviousCodebook[bnds] == INTENSITY_HCB2)) { commonMin = fMin( pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds], pAacDecoderChannelInfo->pComData->overlay.aac .aRvlcScfBwd[bnds]); pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = fMin(commonMin, pAacDecoderStaticChannelInfo->concealmentInfo .aRvlcPreviousScaleFactor[bnds]); } else { pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = -110; } break; case NOISE_HCB: if (pAacDecoderStaticChannelInfo->concealmentInfo .aRvlcPreviousCodebook[bnds] == NOISE_HCB) { commonMin = fMin( pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds], pAacDecoderChannelInfo->pComData->overlay.aac .aRvlcScfBwd[bnds]); pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = fMin(commonMin, pAacDecoderStaticChannelInfo->concealmentInfo .aRvlcPreviousScaleFactor[bnds]); } else { pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = -110; } break; default: if ((pAacDecoderStaticChannelInfo->concealmentInfo .aRvlcPreviousCodebook[bnds] != ZERO_HCB) && (pAacDecoderStaticChannelInfo->concealmentInfo .aRvlcPreviousCodebook[bnds] != NOISE_HCB) && (pAacDecoderStaticChannelInfo->concealmentInfo .aRvlcPreviousCodebook[bnds] != INTENSITY_HCB) && (pAacDecoderStaticChannelInfo->concealmentInfo .aRvlcPreviousCodebook[bnds] != INTENSITY_HCB2)) { commonMin = fMin( pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds], pAacDecoderChannelInfo->pComData->overlay.aac .aRvlcScfBwd[bnds]); pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = fMin(commonMin, pAacDecoderStaticChannelInfo->concealmentInfo .aRvlcPreviousScaleFactor[bnds]); } else { pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = 0; } break; } } } } fdk-aac-2.0.2/libAACdec/src/rvlcconceal.h000066400000000000000000000125421404225310700177650ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief rvlc concealment \author Josef Hoepfl */ #ifndef RVLCCONCEAL_H #define RVLCCONCEAL_H #include "rvlc.h" void BidirectionalEstimation_UseLowerScfOfCurrentFrame( CAacDecoderChannelInfo *pAacDecoderChannelInfo); void BidirectionalEstimation_UseScfOfPrevFrameAsReference( CAacDecoderChannelInfo *pAacDecoderChannelInfo, CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo); void StatisticalEstimation(CAacDecoderChannelInfo *pAacDecoderChannelInfo); void PredictiveInterpolation( CAacDecoderChannelInfo *pAacDecoderChannelInfo, CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo); #endif /* RVLCCONCEAL_H */ fdk-aac-2.0.2/libAACdec/src/stereo.cpp000066400000000000000000001433541404225310700173340ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Josef Hoepfl Description: joint stereo processing *******************************************************************************/ #include "stereo.h" #include "aac_rom.h" #include "FDK_bitstream.h" #include "channelinfo.h" #include "FDK_audio.h" enum { L = 0, R = 1 }; #include "block.h" int CJointStereo_Read(HANDLE_FDK_BITSTREAM bs, CJointStereoData *pJointStereoData, const int windowGroups, const int scaleFactorBandsTransmitted, const int max_sfb_ste_clear, CJointStereoPersistentData *pJointStereoPersistentData, CCplxPredictionData *cplxPredictionData, int cplxPredictionActiv, int scaleFactorBandsTotal, int windowSequence, const UINT flags) { int group, band; pJointStereoData->MsMaskPresent = (UCHAR)FDKreadBits(bs, 2); FDKmemclear(pJointStereoData->MsUsed, scaleFactorBandsTransmitted * sizeof(UCHAR)); pJointStereoData->cplx_pred_flag = 0; if (cplxPredictionActiv) { cplxPredictionData->pred_dir = 0; cplxPredictionData->complex_coef = 0; cplxPredictionData->use_prev_frame = 0; cplxPredictionData->igf_pred_dir = 0; } switch (pJointStereoData->MsMaskPresent) { case 0: /* no M/S */ /* all flags are already cleared */ break; case 1: /* read ms_used */ for (group = 0; group < windowGroups; group++) { for (band = 0; band < scaleFactorBandsTransmitted; band++) { pJointStereoData->MsUsed[band] |= (FDKreadBits(bs, 1) << group); } } break; case 2: /* full spectrum M/S */ for (band = 0; band < scaleFactorBandsTransmitted; band++) { pJointStereoData->MsUsed[band] = 255; /* set all flags to 1 */ } break; case 3: /* M/S coding is disabled, complex stereo prediction is enabled */ if (flags & (AC_USAC | AC_RSVD50 | AC_RSV603DA)) { if (cplxPredictionActiv) { /* 'if (stereoConfigIndex == 0)' */ pJointStereoData->cplx_pred_flag = 1; /* cplx_pred_data() cp. ISO/IEC FDIS 23003-3:2011(E) Table 26 */ int cplx_pred_all = 0; /* local use only */ cplx_pred_all = FDKreadBits(bs, 1); if (cplx_pred_all) { for (group = 0; group < windowGroups; group++) { UCHAR groupmask = ((UCHAR)1 << group); for (band = 0; band < scaleFactorBandsTransmitted; band++) { pJointStereoData->MsUsed[band] |= groupmask; } } } else { for (group = 0; group < windowGroups; group++) { for (band = 0; band < scaleFactorBandsTransmitted; band += SFB_PER_PRED_BAND) { pJointStereoData->MsUsed[band] |= (FDKreadBits(bs, 1) << group); if ((band + 1) < scaleFactorBandsTotal) { pJointStereoData->MsUsed[band + 1] |= (pJointStereoData->MsUsed[band] & ((UCHAR)1 << group)); } } } } } else { return -1; } } break; } if (cplxPredictionActiv) { /* If all sfb are MS-ed then no complex prediction */ if (pJointStereoData->MsMaskPresent == 3) { if (pJointStereoData->cplx_pred_flag) { int delta_code_time = 0; /* set pointer to Huffman codebooks */ const CodeBookDescription *hcb = &AACcodeBookDescriptionTable[BOOKSCL]; /* set predictors to zero in case of a transition from long to short * window sequences and vice versa */ if (((windowSequence == BLOCK_SHORT) && (pJointStereoPersistentData->winSeqPrev != BLOCK_SHORT)) || ((pJointStereoPersistentData->winSeqPrev == BLOCK_SHORT) && (windowSequence != BLOCK_SHORT))) { FDKmemclear(pJointStereoPersistentData->alpha_q_re_prev, JointStereoMaximumGroups * JointStereoMaximumBands * sizeof(SHORT)); FDKmemclear(pJointStereoPersistentData->alpha_q_im_prev, JointStereoMaximumGroups * JointStereoMaximumBands * sizeof(SHORT)); } { FDKmemclear(cplxPredictionData->alpha_q_re, JointStereoMaximumGroups * JointStereoMaximumBands * sizeof(SHORT)); FDKmemclear(cplxPredictionData->alpha_q_im, JointStereoMaximumGroups * JointStereoMaximumBands * sizeof(SHORT)); } /* 0 = mid->side prediction, 1 = side->mid prediction */ cplxPredictionData->pred_dir = FDKreadBits(bs, 1); cplxPredictionData->complex_coef = FDKreadBits(bs, 1); if (cplxPredictionData->complex_coef) { if (flags & AC_INDEP) { cplxPredictionData->use_prev_frame = 0; } else { cplxPredictionData->use_prev_frame = FDKreadBits(bs, 1); } } if (flags & AC_INDEP) { delta_code_time = 0; } else { delta_code_time = FDKreadBits(bs, 1); } { int last_alpha_q_re = 0, last_alpha_q_im = 0; for (group = 0; group < windowGroups; group++) { for (band = 0; band < scaleFactorBandsTransmitted; band += SFB_PER_PRED_BAND) { if (delta_code_time == 1) { if (group > 0) { last_alpha_q_re = cplxPredictionData->alpha_q_re[group - 1][band]; last_alpha_q_im = cplxPredictionData->alpha_q_im[group - 1][band]; } else if ((windowSequence == BLOCK_SHORT) && (pJointStereoPersistentData->winSeqPrev == BLOCK_SHORT)) { /* Included for error-robustness */ if (pJointStereoPersistentData->winGroupsPrev == 0) return -1; last_alpha_q_re = pJointStereoPersistentData->alpha_q_re_prev [pJointStereoPersistentData->winGroupsPrev - 1][band]; last_alpha_q_im = pJointStereoPersistentData->alpha_q_im_prev [pJointStereoPersistentData->winGroupsPrev - 1][band]; } else { last_alpha_q_re = pJointStereoPersistentData->alpha_q_re_prev[group][band]; last_alpha_q_im = pJointStereoPersistentData->alpha_q_im_prev[group][band]; } } else { if (band > 0) { last_alpha_q_re = cplxPredictionData->alpha_q_re[group][band - 1]; last_alpha_q_im = cplxPredictionData->alpha_q_im[group][band - 1]; } else { last_alpha_q_re = 0; last_alpha_q_im = 0; } } /* if (delta_code_time == 1) */ if (pJointStereoData->MsUsed[band] & ((UCHAR)1 << group)) { int dpcm_alpha_re, dpcm_alpha_im; dpcm_alpha_re = CBlock_DecodeHuffmanWord(bs, hcb); dpcm_alpha_re -= 60; dpcm_alpha_re *= -1; cplxPredictionData->alpha_q_re[group][band] = dpcm_alpha_re + last_alpha_q_re; if (cplxPredictionData->complex_coef) { dpcm_alpha_im = CBlock_DecodeHuffmanWord(bs, hcb); dpcm_alpha_im -= 60; dpcm_alpha_im *= -1; cplxPredictionData->alpha_q_im[group][band] = dpcm_alpha_im + last_alpha_q_im; } else { cplxPredictionData->alpha_q_im[group][band] = 0; } } else { cplxPredictionData->alpha_q_re[group][band] = 0; cplxPredictionData->alpha_q_im[group][band] = 0; } /* if (pJointStereoData->MsUsed[band] & ((UCHAR)1 << group)) */ if ((band + 1) < scaleFactorBandsTransmitted) { /* <= this should be the correct way (cp. ISO_IEC_FDIS_23003-0(E) */ /* 7.7.2.3.2 Decoding of prediction coefficients) */ cplxPredictionData->alpha_q_re[group][band + 1] = cplxPredictionData->alpha_q_re[group][band]; cplxPredictionData->alpha_q_im[group][band + 1] = cplxPredictionData->alpha_q_im[group][band]; } /* if ((band+1)alpha_q_re_prev[group][band] = cplxPredictionData->alpha_q_re[group][band]; pJointStereoPersistentData->alpha_q_im_prev[group][band] = cplxPredictionData->alpha_q_im[group][band]; } for (band = scaleFactorBandsTransmitted; band < max_sfb_ste_clear; band++) { cplxPredictionData->alpha_q_re[group][band] = 0; cplxPredictionData->alpha_q_im[group][band] = 0; pJointStereoPersistentData->alpha_q_re_prev[group][band] = 0; pJointStereoPersistentData->alpha_q_im_prev[group][band] = 0; } } } } } else { for (group = 0; group < windowGroups; group++) { for (band = 0; band < max_sfb_ste_clear; band++) { pJointStereoPersistentData->alpha_q_re_prev[group][band] = 0; pJointStereoPersistentData->alpha_q_im_prev[group][band] = 0; } } } pJointStereoPersistentData->winGroupsPrev = windowGroups; } return 0; } static void CJointStereo_filterAndAdd( FIXP_DBL *in, int len, int windowLen, const FIXP_FILT *coeff, FIXP_DBL *out, UCHAR isCurrent /* output values with even index get a positve addon (=1) or a negative addon (=0) */ ) { int i, j; int indices_1[] = {2, 1, 0, 1, 2, 3}; int indices_2[] = {1, 0, 0, 2, 3, 4}; int indices_3[] = {0, 0, 1, 3, 4, 5}; int subtr_1[] = {6, 5, 4, 2, 1, 1}; int subtr_2[] = {5, 4, 3, 1, 1, 2}; int subtr_3[] = {4, 3, 2, 1, 2, 3}; if (isCurrent == 1) { /* exploit the symmetry of the table: coeff[6] = - coeff[0], coeff[5] = - coeff[1], coeff[4] = - coeff[2], coeff[3] = 0 */ for (i = 0; i < 3; i++) { out[0] -= (FIXP_DBL)fMultDiv2(coeff[i], in[indices_1[i]]) >> SR_FNA_OUT; out[0] += (FIXP_DBL)fMultDiv2(coeff[i], in[indices_1[5 - i]]) >> SR_FNA_OUT; } for (i = 0; i < 3; i++) { out[1] -= (FIXP_DBL)fMultDiv2(coeff[i], in[indices_2[i]]) >> SR_FNA_OUT; out[1] += (FIXP_DBL)fMultDiv2(coeff[i], in[indices_2[5 - i]]) >> SR_FNA_OUT; } for (i = 0; i < 3; i++) { out[2] -= (FIXP_DBL)fMultDiv2(coeff[i], in[indices_3[i]]) >> SR_FNA_OUT; out[2] += (FIXP_DBL)fMultDiv2(coeff[i], in[indices_3[5 - i]]) >> SR_FNA_OUT; } for (j = 3; j < (len - 3); j++) { for (i = 0; i < 3; i++) { out[j] -= (FIXP_DBL)fMultDiv2(coeff[i], in[j - 3 + i]) >> SR_FNA_OUT; out[j] += (FIXP_DBL)fMultDiv2(coeff[i], in[j + 3 - i]) >> SR_FNA_OUT; } } for (i = 0; i < 3; i++) { out[len - 3] -= (FIXP_DBL)fMultDiv2(coeff[i], in[len - subtr_1[i]]) >> SR_FNA_OUT; out[len - 3] += (FIXP_DBL)fMultDiv2(coeff[i], in[len - subtr_1[5 - i]]) >> SR_FNA_OUT; } for (i = 0; i < 3; i++) { out[len - 2] -= (FIXP_DBL)fMultDiv2(coeff[i], in[len - subtr_2[i]]) >> SR_FNA_OUT; out[len - 2] += (FIXP_DBL)fMultDiv2(coeff[i], in[len - subtr_2[5 - i]]) >> SR_FNA_OUT; } for (i = 0; i < 3; i++) { out[len - 1] -= (FIXP_DBL)fMultDiv2(coeff[i], in[len - subtr_3[i]]) >> SR_FNA_OUT; out[len - 1] += (FIXP_DBL)fMultDiv2(coeff[i], in[len - subtr_3[5 - i]]) >> SR_FNA_OUT; } } else { /* exploit the symmetry of the table: coeff[6] = coeff[0], coeff[5] = coeff[1], coeff[4] = coeff[2] */ for (i = 0; i < 3; i++) { out[0] -= (FIXP_DBL)fMultDiv2(coeff[i], in[indices_1[i]] >> SR_FNA_OUT); out[0] -= (FIXP_DBL)fMultDiv2(coeff[i], in[indices_1[5 - i]] >> SR_FNA_OUT); } out[0] -= (FIXP_DBL)fMultDiv2(coeff[3], in[0] >> SR_FNA_OUT); for (i = 0; i < 3; i++) { out[1] += (FIXP_DBL)fMultDiv2(coeff[i], in[indices_2[i]] >> SR_FNA_OUT); out[1] += (FIXP_DBL)fMultDiv2(coeff[i], in[indices_2[5 - i]] >> SR_FNA_OUT); } out[1] += (FIXP_DBL)fMultDiv2(coeff[3], in[1] >> SR_FNA_OUT); for (i = 0; i < 3; i++) { out[2] -= (FIXP_DBL)fMultDiv2(coeff[i], in[indices_3[i]] >> SR_FNA_OUT); out[2] -= (FIXP_DBL)fMultDiv2(coeff[i], in[indices_3[5 - i]] >> SR_FNA_OUT); } out[2] -= (FIXP_DBL)fMultDiv2(coeff[3], in[2] >> SR_FNA_OUT); for (j = 3; j < (len - 4); j++) { for (i = 0; i < 3; i++) { out[j] += (FIXP_DBL)fMultDiv2(coeff[i], in[j - 3 + i] >> SR_FNA_OUT); out[j] += (FIXP_DBL)fMultDiv2(coeff[i], in[j + 3 - i] >> SR_FNA_OUT); } out[j] += (FIXP_DBL)fMultDiv2(coeff[3], in[j] >> SR_FNA_OUT); j++; for (i = 0; i < 3; i++) { out[j] -= (FIXP_DBL)fMultDiv2(coeff[i], in[j - 3 + i] >> SR_FNA_OUT); out[j] -= (FIXP_DBL)fMultDiv2(coeff[i], in[j + 3 - i] >> SR_FNA_OUT); } out[j] -= (FIXP_DBL)fMultDiv2(coeff[3], in[j] >> SR_FNA_OUT); } for (i = 0; i < 3; i++) { out[len - 3] += (FIXP_DBL)fMultDiv2(coeff[i], in[len - subtr_1[i]] >> SR_FNA_OUT); out[len - 3] += (FIXP_DBL)fMultDiv2(coeff[i], in[len - subtr_1[5 - i]] >> SR_FNA_OUT); } out[len - 3] += (FIXP_DBL)fMultDiv2(coeff[3], in[len - 3] >> SR_FNA_OUT); for (i = 0; i < 3; i++) { out[len - 2] -= (FIXP_DBL)fMultDiv2(coeff[i], in[len - subtr_2[i]] >> SR_FNA_OUT); out[len - 2] -= (FIXP_DBL)fMultDiv2(coeff[i], in[len - subtr_2[5 - i]] >> SR_FNA_OUT); } out[len - 2] -= (FIXP_DBL)fMultDiv2(coeff[3], in[len - 2] >> SR_FNA_OUT); for (i = 0; i < 3; i++) { out[len - 1] += (FIXP_DBL)fMultDiv2(coeff[i], in[len - subtr_3[i]] >> SR_FNA_OUT); out[len - 1] += (FIXP_DBL)fMultDiv2(coeff[i], in[len - subtr_3[5 - i]] >> SR_FNA_OUT); } out[len - 1] += (FIXP_DBL)fMultDiv2(coeff[3], in[len - 1] >> SR_FNA_OUT); } } static inline void CJointStereo_GenerateMSOutput(FIXP_DBL *pSpecLCurrBand, FIXP_DBL *pSpecRCurrBand, UINT leftScale, UINT rightScale, UINT nSfbBands) { unsigned int i; FIXP_DBL leftCoefficient0; FIXP_DBL leftCoefficient1; FIXP_DBL leftCoefficient2; FIXP_DBL leftCoefficient3; FIXP_DBL rightCoefficient0; FIXP_DBL rightCoefficient1; FIXP_DBL rightCoefficient2; FIXP_DBL rightCoefficient3; for (i = nSfbBands; i > 0; i -= 4) { leftCoefficient0 = pSpecLCurrBand[i - 4]; leftCoefficient1 = pSpecLCurrBand[i - 3]; leftCoefficient2 = pSpecLCurrBand[i - 2]; leftCoefficient3 = pSpecLCurrBand[i - 1]; rightCoefficient0 = pSpecRCurrBand[i - 4]; rightCoefficient1 = pSpecRCurrBand[i - 3]; rightCoefficient2 = pSpecRCurrBand[i - 2]; rightCoefficient3 = pSpecRCurrBand[i - 1]; /* MS output generation */ leftCoefficient0 >>= leftScale; leftCoefficient1 >>= leftScale; leftCoefficient2 >>= leftScale; leftCoefficient3 >>= leftScale; rightCoefficient0 >>= rightScale; rightCoefficient1 >>= rightScale; rightCoefficient2 >>= rightScale; rightCoefficient3 >>= rightScale; pSpecLCurrBand[i - 4] = leftCoefficient0 + rightCoefficient0; pSpecLCurrBand[i - 3] = leftCoefficient1 + rightCoefficient1; pSpecLCurrBand[i - 2] = leftCoefficient2 + rightCoefficient2; pSpecLCurrBand[i - 1] = leftCoefficient3 + rightCoefficient3; pSpecRCurrBand[i - 4] = leftCoefficient0 - rightCoefficient0; pSpecRCurrBand[i - 3] = leftCoefficient1 - rightCoefficient1; pSpecRCurrBand[i - 2] = leftCoefficient2 - rightCoefficient2; pSpecRCurrBand[i - 1] = leftCoefficient3 - rightCoefficient3; } } void CJointStereo_ApplyMS( CAacDecoderChannelInfo *pAacDecoderChannelInfo[2], CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo[2], FIXP_DBL *spectrumL, FIXP_DBL *spectrumR, SHORT *SFBleftScale, SHORT *SFBrightScale, SHORT *specScaleL, SHORT *specScaleR, const SHORT *pScaleFactorBandOffsets, const UCHAR *pWindowGroupLength, const int windowGroups, const int max_sfb_ste_outside, const int scaleFactorBandsTransmittedL, const int scaleFactorBandsTransmittedR, FIXP_DBL *store_dmx_re_prev, SHORT *store_dmx_re_prev_e, const int mainband_flag) { int window, group, band; UCHAR groupMask; CJointStereoData *pJointStereoData = &pAacDecoderChannelInfo[L]->pComData->jointStereoData; CCplxPredictionData *cplxPredictionData = pAacDecoderChannelInfo[L]->pComStaticData->cplxPredictionData; int max_sfb_ste = fMax(scaleFactorBandsTransmittedL, scaleFactorBandsTransmittedR); int min_sfb_ste = fMin(scaleFactorBandsTransmittedL, scaleFactorBandsTransmittedR); int scaleFactorBandsTransmitted = min_sfb_ste; if (pJointStereoData->cplx_pred_flag) { int windowLen, groupwin, frameMaxScale; CJointStereoPersistentData *pJointStereoPersistentData = &pAacDecoderStaticChannelInfo[L] ->pCpeStaticData->jointStereoPersistentData; FIXP_DBL *const staticSpectralCoeffsL = pAacDecoderStaticChannelInfo[L] ->pCpeStaticData->jointStereoPersistentData.spectralCoeffs[L]; FIXP_DBL *const staticSpectralCoeffsR = pAacDecoderStaticChannelInfo[L] ->pCpeStaticData->jointStereoPersistentData.spectralCoeffs[R]; SHORT *const staticSpecScaleL = pAacDecoderStaticChannelInfo[L] ->pCpeStaticData->jointStereoPersistentData.specScale[L]; SHORT *const staticSpecScaleR = pAacDecoderStaticChannelInfo[L] ->pCpeStaticData->jointStereoPersistentData.specScale[R]; FIXP_DBL *dmx_re = pAacDecoderStaticChannelInfo[L] ->pCpeStaticData->jointStereoPersistentData.scratchBuffer; FIXP_DBL *dmx_re_prev = pAacDecoderStaticChannelInfo[L] ->pCpeStaticData->jointStereoPersistentData.scratchBuffer + 1024; /* When MS is applied over the main band this value gets computed. Otherwise * (for the tiles) it uses the assigned value */ SHORT dmx_re_prev_e = *store_dmx_re_prev_e; const FIXP_FILT *pCoeff; const FIXP_FILT *pCoeffPrev; int coeffPointerOffset; int previousShape = (int)pJointStereoPersistentData->winShapePrev; int currentShape = (int)pAacDecoderChannelInfo[L]->icsInfo.WindowShape; /* complex stereo prediction */ /* 0. preparations */ /* 0.0. get scratch buffer for downmix MDST */ C_AALLOC_SCRATCH_START(dmx_im, FIXP_DBL, 1024); /* 0.1. window lengths */ /* get length of short window for current configuration */ windowLen = pAacDecoderChannelInfo[L]->granuleLength; /* framelength 768 => 96, framelength 1024 => 128 */ /* if this is no short-block set length for long-block */ if (pAacDecoderChannelInfo[L]->icsInfo.WindowSequence != BLOCK_SHORT) { windowLen *= 8; } /* 0.2. set pointer to filter-coefficients for MDST excitation including * previous frame portions */ /* cp. ISO/IEC FDIS 23003-3:2011(E) table 125 */ /* set pointer to default-position */ pCoeffPrev = mdst_filt_coef_prev[previousShape]; if (cplxPredictionData->complex_coef == 1) { switch (pAacDecoderChannelInfo[L] ->icsInfo.WindowSequence) { /* current window sequence */ case BLOCK_SHORT: case BLOCK_LONG: pCoeffPrev = mdst_filt_coef_prev[previousShape]; break; case BLOCK_START: if ((pJointStereoPersistentData->winSeqPrev == BLOCK_SHORT) || (pJointStereoPersistentData->winSeqPrev == BLOCK_START)) { /* a stop-start-sequence can only follow on an eight-short-sequence * or a start-sequence */ pCoeffPrev = mdst_filt_coef_prev[2 + previousShape]; } else { pCoeffPrev = mdst_filt_coef_prev[previousShape]; } break; case BLOCK_STOP: pCoeffPrev = mdst_filt_coef_prev[2 + previousShape]; break; default: pCoeffPrev = mdst_filt_coef_prev[previousShape]; break; } } /* 0.3. set pointer to filter-coefficients for MDST excitation */ /* define offset of pointer to filter-coefficients for MDST exitation * employing only the current frame */ if ((previousShape == SHAPE_SINE) && (currentShape == SHAPE_SINE)) { coeffPointerOffset = 0; } else if ((previousShape == SHAPE_SINE) && (currentShape == SHAPE_KBD)) { coeffPointerOffset = 2; } else if ((previousShape == SHAPE_KBD) && (currentShape == SHAPE_KBD)) { coeffPointerOffset = 1; } else /* if ( (previousShape == SHAPE_KBD) && (currentShape == SHAPE_SINE) ) */ { coeffPointerOffset = 3; } /* set pointer to filter-coefficient table cp. ISO/IEC FDIS 23003-3:2011(E) * table 124 */ switch (pAacDecoderChannelInfo[L] ->icsInfo.WindowSequence) { /* current window sequence */ case BLOCK_SHORT: case BLOCK_LONG: pCoeff = mdst_filt_coef_curr[coeffPointerOffset]; break; case BLOCK_START: if ((pJointStereoPersistentData->winSeqPrev == BLOCK_SHORT) || (pJointStereoPersistentData->winSeqPrev == BLOCK_START)) { /* a stop-start-sequence can only follow on an eight-short-sequence or * a start-sequence */ pCoeff = mdst_filt_coef_curr[12 + coeffPointerOffset]; } else { pCoeff = mdst_filt_coef_curr[4 + coeffPointerOffset]; } break; case BLOCK_STOP: pCoeff = mdst_filt_coef_curr[8 + coeffPointerOffset]; break; default: pCoeff = mdst_filt_coef_curr[coeffPointerOffset]; } /* 0.4. find maximum common (l/r) band-scaling-factor for whole sequence * (all windows) */ frameMaxScale = 0; for (window = 0, group = 0; group < windowGroups; group++) { for (groupwin = 0; groupwin < pWindowGroupLength[group]; groupwin++, window++) { SHORT *leftScale = &SFBleftScale[window * 16]; SHORT *rightScale = &SFBrightScale[window * 16]; int windowMaxScale = 0; /* find maximum scaling factor of all bands in this window */ for (band = 0; band < min_sfb_ste; band++) { int lScale = leftScale[band]; int rScale = rightScale[band]; int commonScale = ((lScale > rScale) ? lScale : rScale); windowMaxScale = (windowMaxScale < commonScale) ? commonScale : windowMaxScale; } if (scaleFactorBandsTransmittedL > min_sfb_ste) { /* i.e. scaleFactorBandsTransmittedL == max_sfb_ste */ for (; band < max_sfb_ste; band++) { int lScale = leftScale[band]; windowMaxScale = (windowMaxScale < lScale) ? lScale : windowMaxScale; } } else { if (scaleFactorBandsTransmittedR > min_sfb_ste) { /* i.e. scaleFactorBandsTransmittedR == max_sfb_ste */ for (; band < max_sfb_ste; band++) { int rScale = rightScale[band]; windowMaxScale = (windowMaxScale < rScale) ? rScale : windowMaxScale; } } } /* find maximum common SF of all windows */ frameMaxScale = (frameMaxScale < windowMaxScale) ? windowMaxScale : frameMaxScale; } } /* add some headroom for overflow protection during filter and add operation */ frameMaxScale += 2; /* process on window-basis (i.e. iterate over all groups and corresponding * windows) */ for (window = 0, group = 0; group < windowGroups; group++) { groupMask = 1 << group; for (groupwin = 0; groupwin < pWindowGroupLength[group]; groupwin++, window++) { /* initialize the MDST with zeros */ FDKmemclear(&dmx_im[windowLen * window], windowLen * sizeof(FIXP_DBL)); /* 1. calculate the previous downmix MDCT. We do this once just for the * Main band. */ if (cplxPredictionData->complex_coef == 1) { if ((cplxPredictionData->use_prev_frame == 1) && (mainband_flag)) { /* if this is a long-block or the first window of a short-block calculate the downmix MDCT of the previous frame. use_prev_frame is assumed not to change during a frame! */ /* first determine shiftfactors to scale left and right channel */ if ((pAacDecoderChannelInfo[L]->icsInfo.WindowSequence != BLOCK_SHORT) || (window == 0)) { int index_offset = 0; int srLeftChan = 0; int srRightChan = 0; if (pAacDecoderChannelInfo[L]->icsInfo.WindowSequence == BLOCK_SHORT) { /* use the last window of the previous frame for MDCT * calculation if this is a short-block. */ index_offset = windowLen * 7; if (staticSpecScaleL[7] > staticSpecScaleR[7]) { srRightChan = staticSpecScaleL[7] - staticSpecScaleR[7]; dmx_re_prev_e = staticSpecScaleL[7]; } else { srLeftChan = staticSpecScaleR[7] - staticSpecScaleL[7]; dmx_re_prev_e = staticSpecScaleR[7]; } } else { if (staticSpecScaleL[0] > staticSpecScaleR[0]) { srRightChan = staticSpecScaleL[0] - staticSpecScaleR[0]; dmx_re_prev_e = staticSpecScaleL[0]; } else { srLeftChan = staticSpecScaleR[0] - staticSpecScaleL[0]; dmx_re_prev_e = staticSpecScaleR[0]; } } /* now scale channels and determine downmix MDCT of previous frame */ if (pAacDecoderStaticChannelInfo[L] ->pCpeStaticData->jointStereoPersistentData .clearSpectralCoeffs == 1) { FDKmemclear(dmx_re_prev, windowLen * sizeof(FIXP_DBL)); dmx_re_prev_e = 0; } else { if (cplxPredictionData->pred_dir == 0) { for (int i = 0; i < windowLen; i++) { dmx_re_prev[i] = ((staticSpectralCoeffsL[index_offset + i] >> fMin(DFRACT_BITS - 1, srLeftChan + 1)) + (staticSpectralCoeffsR[index_offset + i] >> fMin(DFRACT_BITS - 1, srRightChan + 1))); } } else { for (int i = 0; i < windowLen; i++) { dmx_re_prev[i] = ((staticSpectralCoeffsL[index_offset + i] >> fMin(DFRACT_BITS - 1, srLeftChan + 1)) - (staticSpectralCoeffsR[index_offset + i] >> fMin(DFRACT_BITS - 1, srRightChan + 1))); } } } /* In case that we use INF we have to preserve the state of the "dmx_re_prev" (original or computed). This is necessary because we have to apply MS over the separate IGF tiles. */ FDKmemcpy(store_dmx_re_prev, &dmx_re_prev[0], windowLen * sizeof(FIXP_DBL)); /* Particular exponent of the computed/original "dmx_re_prev" must * be kept for the tile MS calculations if necessary.*/ *store_dmx_re_prev_e = dmx_re_prev_e; } /* if ( (pAacDecoderChannelInfo[L]->icsInfo.WindowSequence != BLOCK_SHORT) || (window == 0) ) */ } /* if ( pJointStereoData->use_prev_frame == 1 ) */ } /* if ( pJointStereoData->complex_coef == 1 ) */ /* 2. calculate downmix MDCT of current frame */ /* set pointer to scale-factor-bands of current window */ SHORT *leftScale = &SFBleftScale[window * 16]; SHORT *rightScale = &SFBrightScale[window * 16]; specScaleL[window] = specScaleR[window] = frameMaxScale; /* adapt scaling-factors to previous frame */ if (cplxPredictionData->use_prev_frame == 1) { if (window == 0) { if (dmx_re_prev_e < frameMaxScale) { if (mainband_flag == 0) { scaleValues( dmx_re_prev, store_dmx_re_prev, windowLen, -fMin(DFRACT_BITS - 1, (frameMaxScale - dmx_re_prev_e))); } else { scaleValues( dmx_re_prev, windowLen, -fMin(DFRACT_BITS - 1, (frameMaxScale - dmx_re_prev_e))); } } else { if (mainband_flag == 0) { FDKmemcpy(dmx_re_prev, store_dmx_re_prev, windowLen * sizeof(FIXP_DBL)); } specScaleL[0] = dmx_re_prev_e; specScaleR[0] = dmx_re_prev_e; } } else { /* window != 0 */ FDK_ASSERT(pAacDecoderChannelInfo[L]->icsInfo.WindowSequence == BLOCK_SHORT); if (specScaleL[window - 1] < frameMaxScale) { scaleValues(&dmx_re[windowLen * (window - 1)], windowLen, -fMin(DFRACT_BITS - 1, (frameMaxScale - specScaleL[window - 1]))); } else { specScaleL[window] = specScaleL[window - 1]; specScaleR[window] = specScaleR[window - 1]; } } } /* if ( pJointStereoData->use_prev_frame == 1 ) */ /* scaling factors of both channels ought to be equal now */ FDK_ASSERT(specScaleL[window] == specScaleR[window]); /* rescale signal and calculate downmix MDCT */ for (band = 0; band < max_sfb_ste; band++) { /* first adapt scaling of current band to scaling of current window => * shift signal right */ int lScale = leftScale[band]; int rScale = rightScale[band]; lScale = fMin(DFRACT_BITS - 1, specScaleL[window] - lScale); rScale = fMin(DFRACT_BITS - 1, specScaleL[window] - rScale); /* L or R doesn't matter, specScales are equal at this point */ /* Write back to sfb scale to cover the case when max_sfb_ste < * max_sfb */ leftScale[band] = rightScale[band] = specScaleL[window]; for (int i = pScaleFactorBandOffsets[band]; i < pScaleFactorBandOffsets[band + 1]; i++) { spectrumL[windowLen * window + i] >>= lScale; spectrumR[windowLen * window + i] >>= rScale; } /* now calculate downmix MDCT */ if (pJointStereoData->MsUsed[band] & groupMask) { for (int i = pScaleFactorBandOffsets[band]; i < pScaleFactorBandOffsets[band + 1]; i++) { dmx_re[windowLen * window + i] = spectrumL[windowLen * window + i]; } } else { if (cplxPredictionData->pred_dir == 0) { for (int i = pScaleFactorBandOffsets[band]; i < pScaleFactorBandOffsets[band + 1]; i++) { dmx_re[windowLen * window + i] = (spectrumL[windowLen * window + i] + spectrumR[windowLen * window + i]) >> 1; } } else { for (int i = pScaleFactorBandOffsets[band]; i < pScaleFactorBandOffsets[band + 1]; i++) { dmx_re[windowLen * window + i] = (spectrumL[windowLen * window + i] - spectrumR[windowLen * window + i]) >> 1; } } } } /* for ( band=0; bandcomplex_coef == 1) { { /* 3.1 move pointer in filter-coefficient table in case of short * window sequence */ /* (other coefficients are utilized for the last 7 short * windows) */ if ((pAacDecoderChannelInfo[L]->icsInfo.WindowSequence == BLOCK_SHORT) && (window != 0)) { pCoeff = mdst_filt_coef_curr[currentShape]; pCoeffPrev = mdst_filt_coef_prev[currentShape]; } /* The length of the filter processing must be extended because of * filter boundary problems */ int extended_band = fMin( pScaleFactorBandOffsets[max_sfb_ste_outside] + 7, windowLen); /* 3.2. estimate downmix MDST from current frame downmix MDCT */ if ((pAacDecoderChannelInfo[L]->icsInfo.WindowSequence == BLOCK_SHORT) && (window != 0)) { CJointStereo_filterAndAdd(&dmx_re[windowLen * window], extended_band, windowLen, pCoeff, &dmx_im[windowLen * window], 1); CJointStereo_filterAndAdd(&dmx_re[windowLen * (window - 1)], extended_band, windowLen, pCoeffPrev, &dmx_im[windowLen * window], 0); } else { CJointStereo_filterAndAdd(dmx_re, extended_band, windowLen, pCoeff, dmx_im, 1); if (cplxPredictionData->use_prev_frame == 1) { CJointStereo_filterAndAdd(dmx_re_prev, extended_band, windowLen, pCoeffPrev, &dmx_im[windowLen * window], 0); } } } /* if(pAacDecoderChannelInfo[L]->transform_splitting_active) */ } /* if ( pJointStereoData->complex_coef == 1 ) */ /* 4. upmix process */ LONG pred_dir = cplxPredictionData->pred_dir ? -1 : 1; /* 0.1 in Q-3.34 */ const FIXP_DBL pointOne = 0x66666666; /* 0.8 */ /* Shift value for the downmix */ const INT shift_dmx = SF_FNA_COEFFS + 1; for (band = 0; band < max_sfb_ste_outside; band++) { if (pJointStereoData->MsUsed[band] & groupMask) { FIXP_SGL tempRe = (FIXP_SGL)cplxPredictionData->alpha_q_re[group][band]; FIXP_SGL tempIm = (FIXP_SGL)cplxPredictionData->alpha_q_im[group][band]; /* Find the minimum common headroom for alpha_re and alpha_im */ int alpha_re_headroom = CountLeadingBits((INT)tempRe) - 16; if (tempRe == (FIXP_SGL)0) alpha_re_headroom = 15; int alpha_im_headroom = CountLeadingBits((INT)tempIm) - 16; if (tempIm == (FIXP_SGL)0) alpha_im_headroom = 15; int val = fMin(alpha_re_headroom, alpha_im_headroom); /* Multiply alpha by 0.1 with maximum precision */ FDK_ASSERT(val >= 0); FIXP_DBL alpha_re_tmp = fMult((FIXP_SGL)(tempRe << val), pointOne); FIXP_DBL alpha_im_tmp = fMult((FIXP_SGL)(tempIm << val), pointOne); /* Calculate alpha exponent */ /* (Q-3.34 * Q15.0) shifted left by "val" */ int alpha_re_exp = -3 + 15 - val; int help3_shift = alpha_re_exp + 1; FIXP_DBL *p2CoeffL = &( spectrumL[windowLen * window + pScaleFactorBandOffsets[band]]); FIXP_DBL *p2CoeffR = &( spectrumR[windowLen * window + pScaleFactorBandOffsets[band]]); FIXP_DBL *p2dmxIm = &(dmx_im[windowLen * window + pScaleFactorBandOffsets[band]]); FIXP_DBL *p2dmxRe = &(dmx_re[windowLen * window + pScaleFactorBandOffsets[band]]); for (int i = pScaleFactorBandOffsets[band]; i < pScaleFactorBandOffsets[band + 1]; i++) { /* Calculating helper term: side = specR[i] - alpha_re[i] * dmx_re[i] - alpha_im[i] * dmx_im[i]; Here "dmx_re" may be the same as "specL" or alternatively keep the downmix. "dmx_re" and "specL" are two different pointers pointing to separate arrays, which may or may not contain the same data (with different scaling). specL[i] = + (specL[i] + side); specR[i] = -/+ (specL[i] - side); */ FIXP_DBL side, left, right; side = fMultAddDiv2(fMultDiv2(alpha_re_tmp, *p2dmxRe++), alpha_im_tmp, (*p2dmxIm++) << shift_dmx); side = ((*p2CoeffR) >> 2) - (FIXP_DBL)SATURATE_SHIFT(side, -(help3_shift - 2), DFRACT_BITS - 2); left = ((*p2CoeffL) >> 2) + side; right = ((*p2CoeffL) >> 2) - side; right = (FIXP_DBL)((LONG)right * pred_dir); *p2CoeffL++ = SATURATE_LEFT_SHIFT_ALT(left, 2, DFRACT_BITS); *p2CoeffR++ = SATURATE_LEFT_SHIFT_ALT(right, 2, DFRACT_BITS); } } } /* for ( band=0; band < max_sfb_ste; band++ ) */ } /* for ( groupwin=0; groupwingranuleLength); rightSpectrum = SPEC(spectrumR, window, pAacDecoderChannelInfo[R]->granuleLength); for (band = 0; band < max_sfb_ste_outside; band++) { if (pJointStereoData->MsUsed[band] & groupMask) { int lScale = leftScale[band]; int rScale = rightScale[band]; int commonScale = lScale > rScale ? lScale : rScale; unsigned int offsetCurrBand, offsetNextBand; /* ISO/IEC 14496-3 Chapter 4.6.8.1.1 : M/S joint channel coding can only be used if common_window is 1. */ FDK_ASSERT(GetWindowSequence(&pAacDecoderChannelInfo[L]->icsInfo) == GetWindowSequence(&pAacDecoderChannelInfo[R]->icsInfo)); FDK_ASSERT(GetWindowShape(&pAacDecoderChannelInfo[L]->icsInfo) == GetWindowShape(&pAacDecoderChannelInfo[R]->icsInfo)); commonScale++; leftScale[band] = commonScale; rightScale[band] = commonScale; lScale = fMin(DFRACT_BITS - 1, commonScale - lScale); rScale = fMin(DFRACT_BITS - 1, commonScale - rScale); FDK_ASSERT(lScale >= 0 && rScale >= 0); offsetCurrBand = pScaleFactorBandOffsets[band]; offsetNextBand = pScaleFactorBandOffsets[band + 1]; CJointStereo_GenerateMSOutput(&(leftSpectrum[offsetCurrBand]), &(rightSpectrum[offsetCurrBand]), lScale, rScale, offsetNextBand - offsetCurrBand); } } if (scaleFactorBandsTransmittedL > scaleFactorBandsTransmitted) { for (; band < scaleFactorBandsTransmittedL; band++) { if (pJointStereoData->MsUsed[band] & groupMask) { rightScale[band] = leftScale[band]; for (int index = pScaleFactorBandOffsets[band]; index < pScaleFactorBandOffsets[band + 1]; index++) { FIXP_DBL leftCoefficient = leftSpectrum[index]; /* FIXP_DBL rightCoefficient = (FIXP_DBL)0; */ rightSpectrum[index] = leftCoefficient; } } } } else if (scaleFactorBandsTransmittedR > scaleFactorBandsTransmitted) { for (; band < scaleFactorBandsTransmittedR; band++) { if (pJointStereoData->MsUsed[band] & groupMask) { leftScale[band] = rightScale[band]; for (int index = pScaleFactorBandOffsets[band]; index < pScaleFactorBandOffsets[band + 1]; index++) { /* FIXP_DBL leftCoefficient = (FIXP_DBL)0; */ FIXP_DBL rightCoefficient = rightSpectrum[index]; leftSpectrum[index] = rightCoefficient; rightSpectrum[index] = -rightCoefficient; } } } } } } /* Reset MsUsed flags if no explicit signalling was transmitted. Necessary for intensity coding. PNS correlation signalling was mapped before calling CJointStereo_ApplyMS(). */ if (pJointStereoData->MsMaskPresent == 2) { FDKmemclear(pJointStereoData->MsUsed, JointStereoMaximumBands * sizeof(UCHAR)); } } } void CJointStereo_ApplyIS(CAacDecoderChannelInfo *pAacDecoderChannelInfo[2], const SHORT *pScaleFactorBandOffsets, const UCHAR *pWindowGroupLength, const int windowGroups, const int scaleFactorBandsTransmitted) { CJointStereoData *pJointStereoData = &pAacDecoderChannelInfo[L]->pComData->jointStereoData; for (int window = 0, group = 0; group < windowGroups; group++) { UCHAR *CodeBook; SHORT *ScaleFactor; UCHAR groupMask = 1 << group; CodeBook = &pAacDecoderChannelInfo[R]->pDynData->aCodeBook[group * 16]; ScaleFactor = &pAacDecoderChannelInfo[R]->pDynData->aScaleFactor[group * 16]; for (int groupwin = 0; groupwin < pWindowGroupLength[group]; groupwin++, window++) { FIXP_DBL *leftSpectrum, *rightSpectrum; SHORT *leftScale = &pAacDecoderChannelInfo[L]->pDynData->aSfbScale[window * 16]; SHORT *rightScale = &pAacDecoderChannelInfo[R]->pDynData->aSfbScale[window * 16]; int band; leftSpectrum = SPEC(pAacDecoderChannelInfo[L]->pSpectralCoefficient, window, pAacDecoderChannelInfo[L]->granuleLength); rightSpectrum = SPEC(pAacDecoderChannelInfo[R]->pSpectralCoefficient, window, pAacDecoderChannelInfo[R]->granuleLength); for (band = 0; band < scaleFactorBandsTransmitted; band++) { if ((CodeBook[band] == INTENSITY_HCB) || (CodeBook[band] == INTENSITY_HCB2)) { int bandScale = -(ScaleFactor[band] + 100); int msb = bandScale >> 2; int lsb = bandScale & 0x03; /* exponent of MantissaTable[lsb][0] is 1, thus msb+1 below. */ FIXP_DBL scale = MantissaTable[lsb][0]; /* ISO/IEC 14496-3 Chapter 4.6.8.2.3 : The use of intensity stereo coding is signaled by the use of the pseudo codebooks INTENSITY_HCB and INTENSITY_HCB2 (15 and 14) only in the right channel of a channel_pair_element() having a common ics_info() (common_window == 1). */ FDK_ASSERT(GetWindowSequence(&pAacDecoderChannelInfo[L]->icsInfo) == GetWindowSequence(&pAacDecoderChannelInfo[R]->icsInfo)); FDK_ASSERT(GetWindowShape(&pAacDecoderChannelInfo[L]->icsInfo) == GetWindowShape(&pAacDecoderChannelInfo[R]->icsInfo)); rightScale[band] = leftScale[band] + msb + 1; if (pJointStereoData->MsUsed[band] & groupMask) { if (CodeBook[band] == INTENSITY_HCB) /* _NOT_ in-phase */ { scale = -scale; } } else { if (CodeBook[band] == INTENSITY_HCB2) /* out-of-phase */ { scale = -scale; } } for (int index = pScaleFactorBandOffsets[band]; index < pScaleFactorBandOffsets[band + 1]; index++) { rightSpectrum[index] = fMult(leftSpectrum[index], scale); } } } } } } fdk-aac-2.0.2/libAACdec/src/stereo.h000066400000000000000000000216541404225310700167770ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Josef Hoepfl Description: joint stereo processing *******************************************************************************/ #ifndef STEREO_H #define STEREO_H #include "machine_type.h" #include "FDK_bitstream.h" #include "common_fix.h" #define SFB_PER_PRED_BAND 2 #define SR_FNA_OUT \ 0 /* Additional scaling of the CJointStereo_filterAndAdd()-output to avoid \ overflows. */ /* The scaling factor can be set to 0 if the coefficients are prescaled * appropriately. */ /* Prescaling via factor SF_FNA_COEFFS is done at compile-time but should only * be */ /* utilized if the coefficients are stored as FIXP_DBL. (cp. aac_rom.cpp/.h) */ /* The NO_CPLX_PRED_BUGFIX-switch was introduced to enable decoding of conformance-streams in way that they are comparable to buggy reference-streams. This is established by storing the prediction direction for computation of the "downmix MDCT of previous frame". This is not standard compliant. Once correct reference-streams for complex-stereo-prediction are available this switch becomes obsolete. */ /*#define NO_CPLX_PRED_BUGFIX*/ enum { JointStereoMaximumGroups = 8, JointStereoMaximumBands = 64 }; typedef struct { UCHAR pred_dir; // 0 = prediction from mid to side channel, 1 = vice versa UCHAR igf_pred_dir; // 0 = prediction from mid to side channel, 1 = vice versa UCHAR complex_coef; // 0 = alpha_q_im[x] is 0 for all prediction bands, 1 = // alpha_q_im[x] is transmitted via bitstream UCHAR use_prev_frame; // 0 = use current frame for MDST estimation, 1 = use // current and previous frame SHORT alpha_q_re[JointStereoMaximumGroups][JointStereoMaximumBands]; SHORT alpha_q_im[JointStereoMaximumGroups][JointStereoMaximumBands]; } CCplxPredictionData; /* joint stereo scratch memory (valid for this frame) */ typedef struct { UCHAR MsMaskPresent; UCHAR MsUsed[JointStereoMaximumBands]; /*!< every arry element contains flags for up to 8 groups. this array is also utilized for complex stereo prediction. */ UCHAR IGF_MsMaskPresent; UCHAR cplx_pred_flag; /* stereo complex prediction was signalled for this frame */ UCHAR igf_cplx_pred_flag; /* The following array and variable are needed for the case when INF is * active */ FIXP_DBL store_dmx_re_prev[1024]; SHORT store_dmx_re_prev_e; } CJointStereoData; /* joint stereo persistent memory */ typedef struct { UCHAR clearSpectralCoeffs; /* indicates that the spectral coeffs must be cleared because the transform splitting active flag of the left and right channel was different */ FIXP_DBL *scratchBuffer; /* pointer to scratch buffer */ FIXP_DBL *spectralCoeffs[2]; /* spectral coefficients of this channel utilized by complex stereo prediction */ SHORT *specScale[2]; SHORT alpha_q_re_prev[JointStereoMaximumGroups][JointStereoMaximumBands]; SHORT alpha_q_im_prev[JointStereoMaximumGroups][JointStereoMaximumBands]; UCHAR winSeqPrev; UCHAR winShapePrev; UCHAR winGroupsPrev; } CJointStereoPersistentData; /*! \brief Read joint stereo data from bitstream The function reads joint stereo data from bitstream. \param bs bit stream handle data source. \param pJointStereoData pointer to stereo data structure to receive decoded data. \param windowGroups number of window groups. \param scaleFactorBandsTransmitted number of transmitted scalefactor bands. \param flags decoder flags \return 0 on success, -1 on error. */ int CJointStereo_Read(HANDLE_FDK_BITSTREAM bs, CJointStereoData *pJointStereoData, const int windowGroups, const int scaleFactorBandsTransmitted, const int max_sfb_ste_clear, CJointStereoPersistentData *pJointStereoPersistentData, CCplxPredictionData *cplxPredictionData, int cplxPredictionActiv, int scaleFactorBandsTotal, int windowSequence, const UINT flags); #endif /* #ifndef STEREO_H */ fdk-aac-2.0.2/libAACdec/src/usacdec_ace_d4t64.cpp000066400000000000000000000317641404225310700212000ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Description: ACELP *******************************************************************************/ #include "usacdec_ace_d4t64.h" #define L_SUBFR 64 /* Subframe size */ /* * D_ACELP_add_pulse * * Parameters: * pos I: position of pulse * nb_pulse I: number of pulses * track I: track * code O: fixed codebook * * Function: * Add pulses to fixed codebook * * Returns: * void */ static void D_ACELP_add_pulse(SHORT pos[], SHORT nb_pulse, SHORT track, FIXP_COD code[]) { SHORT i, k; for (k = 0; k < nb_pulse; k++) { /* i = ((pos[k] & (16-1))*NB_TRACK) + track; */ i = ((pos[k] & (16 - 1)) << 2) + track; if ((pos[k] & 16) == 0) { code[i] = code[i] + (FIXP_COD)(512 << (COD_BITS - FRACT_BITS)); } else { code[i] = code[i] - (FIXP_COD)(512 << (COD_BITS - FRACT_BITS)); } } return; } /* * D_ACELP_decode_1p_N1 * * Parameters: * index I: pulse index * N I: number of bits for position * offset I: offset * pos O: position of the pulse * * Function: * Decode 1 pulse with N+1 bits * * Returns: * void */ static void D_ACELP_decode_1p_N1(LONG index, SHORT N, SHORT offset, SHORT pos[]) { SHORT pos1; LONG i, mask; mask = ((1 << N) - 1); /* * Decode 1 pulse with N+1 bits */ pos1 = (SHORT)((index & mask) + offset); i = ((index >> N) & 1); if (i == 1) { pos1 += 16; } pos[0] = pos1; return; } /* * D_ACELP_decode_2p_2N1 * * Parameters: * index I: pulse index * N I: number of bits for position * offset I: offset * pos O: position of the pulse * * Function: * Decode 2 pulses with 2*N+1 bits * * Returns: * void */ static void D_ACELP_decode_2p_2N1(LONG index, SHORT N, SHORT offset, SHORT pos[]) { SHORT pos1, pos2; LONG mask, i; mask = ((1 << N) - 1); /* * Decode 2 pulses with 2*N+1 bits */ pos1 = (SHORT)(((index >> N) & mask) + offset); i = (index >> (2 * N)) & 1; pos2 = (SHORT)((index & mask) + offset); if ((pos2 - pos1) < 0) { if (i == 1) { pos1 += 16; } else { pos2 += 16; } } else { if (i == 1) { pos1 += 16; pos2 += 16; } } pos[0] = pos1; pos[1] = pos2; return; } /* * D_ACELP_decode_3p_3N1 * * Parameters: * index I: pulse index * N I: number of bits for position * offset I: offset * pos O: position of the pulse * * Function: * Decode 3 pulses with 3*N+1 bits * * Returns: * void */ static void D_ACELP_decode_3p_3N1(LONG index, SHORT N, SHORT offset, SHORT pos[]) { SHORT j; LONG mask, idx; /* * Decode 3 pulses with 3*N+1 bits */ mask = ((1 << ((2 * N) - 1)) - 1); idx = index & mask; j = offset; if (((index >> ((2 * N) - 1)) & 1) == 1) { j += (1 << (N - 1)); } D_ACELP_decode_2p_2N1(idx, N - 1, j, pos); mask = ((1 << (N + 1)) - 1); idx = (index >> (2 * N)) & mask; D_ACELP_decode_1p_N1(idx, N, offset, pos + 2); return; } /* * D_ACELP_decode_4p_4N1 * * Parameters: * index I: pulse index * N I: number of bits for position * offset I: offset * pos O: position of the pulse * * Function: * Decode 4 pulses with 4*N+1 bits * * Returns: * void */ static void D_ACELP_decode_4p_4N1(LONG index, SHORT N, SHORT offset, SHORT pos[]) { SHORT j; LONG mask, idx; /* * Decode 4 pulses with 4*N+1 bits */ mask = ((1 << ((2 * N) - 1)) - 1); idx = index & mask; j = offset; if (((index >> ((2 * N) - 1)) & 1) == 1) { j += (1 << (N - 1)); } D_ACELP_decode_2p_2N1(idx, N - 1, j, pos); mask = ((1 << ((2 * N) + 1)) - 1); idx = (index >> (2 * N)) & mask; D_ACELP_decode_2p_2N1(idx, N, offset, pos + 2); return; } /* * D_ACELP_decode_4p_4N * * Parameters: * index I: pulse index * N I: number of bits for position * offset I: offset * pos O: position of the pulse * * Function: * Decode 4 pulses with 4*N bits * * Returns: * void */ static void D_ACELP_decode_4p_4N(LONG index, SHORT N, SHORT offset, SHORT pos[]) { SHORT j, n_1; /* * Decode 4 pulses with 4*N bits */ n_1 = N - 1; j = offset + (1 << n_1); switch ((index >> ((4 * N) - 2)) & 3) { case 0: if (((index >> ((4 * n_1) + 1)) & 1) == 0) { D_ACELP_decode_4p_4N1(index, n_1, offset, pos); } else { D_ACELP_decode_4p_4N1(index, n_1, j, pos); } break; case 1: D_ACELP_decode_1p_N1((index >> ((3 * n_1) + 1)), n_1, offset, pos); D_ACELP_decode_3p_3N1(index, n_1, j, pos + 1); break; case 2: D_ACELP_decode_2p_2N1((index >> ((2 * n_1) + 1)), n_1, offset, pos); D_ACELP_decode_2p_2N1(index, n_1, j, pos + 2); break; case 3: D_ACELP_decode_3p_3N1((index >> (n_1 + 1)), n_1, offset, pos); D_ACELP_decode_1p_N1(index, n_1, j, pos + 3); break; } return; } /* * D_ACELP_decode_4t * * Parameters: * index I: index * mode I: speech mode * code I: (Q9) algebraic (fixed) codebook excitation * * Function: * 20, 36, 44, 52, 64, 72, 88 bits algebraic codebook. * 4 tracks x 16 positions per track = 64 samples. * * 20 bits 5+5+5+5 --> 4 pulses in a frame of 64 samples. * 36 bits 9+9+9+9 --> 8 pulses in a frame of 64 samples. * 44 bits 13+9+13+9 --> 10 pulses in a frame of 64 samples. * 52 bits 13+13+13+13 --> 12 pulses in a frame of 64 samples. * 64 bits 2+2+2+2+14+14+14+14 --> 16 pulses in a frame of 64 samples. * 72 bits 10+2+10+2+10+14+10+14 --> 18 pulses in a frame of 64 samples. * 88 bits 11+11+11+11+11+11+11+11 --> 24 pulses in a frame of 64 samples. * * All pulses can have two (2) possible amplitudes: +1 or -1. * Each pulse can sixteen (16) possible positions. * * codevector length 64 * number of track 4 * number of position 16 * * Returns: * void */ void D_ACELP_decode_4t64(SHORT index[], int nbits, FIXP_COD code[]) { LONG L_index; SHORT k, pos[6]; FDKmemclear(code, L_SUBFR * sizeof(FIXP_COD)); /* decode the positions and signs of pulses and build the codeword */ switch (nbits) { case 12: for (k = 0; k < 4; k += 2) { L_index = index[2 * (k / 2) + 1]; D_ACELP_decode_1p_N1(L_index, 4, 0, pos); D_ACELP_add_pulse(pos, 1, 2 * (index[2 * (k / 2)]) + k / 2, code); } break; case 16: { int i = 0; int offset = index[i++]; offset = (offset == 0) ? 1 : 3; for (k = 0; k < 4; k++) { if (k != offset) { L_index = index[i++]; D_ACELP_decode_1p_N1(L_index, 4, 0, pos); D_ACELP_add_pulse(pos, 1, k, code); } } } break; case 20: for (k = 0; k < 4; k++) { L_index = (LONG)index[k]; D_ACELP_decode_1p_N1(L_index, 4, 0, pos); D_ACELP_add_pulse(pos, 1, k, code); } break; case 28: for (k = 0; k < 4 - 2; k++) { L_index = (LONG)index[k]; D_ACELP_decode_2p_2N1(L_index, 4, 0, pos); D_ACELP_add_pulse(pos, 2, k, code); } for (k = 2; k < 4; k++) { L_index = (LONG)index[k]; D_ACELP_decode_1p_N1(L_index, 4, 0, pos); D_ACELP_add_pulse(pos, 1, k, code); } break; case 36: for (k = 0; k < 4; k++) { L_index = (LONG)index[k]; D_ACELP_decode_2p_2N1(L_index, 4, 0, pos); D_ACELP_add_pulse(pos, 2, k, code); } break; case 44: for (k = 0; k < 4 - 2; k++) { L_index = (LONG)index[k]; D_ACELP_decode_3p_3N1(L_index, 4, 0, pos); D_ACELP_add_pulse(pos, 3, k, code); } for (k = 2; k < 4; k++) { L_index = (LONG)index[k]; D_ACELP_decode_2p_2N1(L_index, 4, 0, pos); D_ACELP_add_pulse(pos, 2, k, code); } break; case 52: for (k = 0; k < 4; k++) { L_index = (LONG)index[k]; D_ACELP_decode_3p_3N1(L_index, 4, 0, pos); D_ACELP_add_pulse(pos, 3, k, code); } break; case 64: for (k = 0; k < 4; k++) { L_index = (((LONG)index[k] << 14) + (LONG)index[k + 4]); D_ACELP_decode_4p_4N(L_index, 4, 0, pos); D_ACELP_add_pulse(pos, 4, k, code); } break; default: FDK_ASSERT(0); } return; } fdk-aac-2.0.2/libAACdec/src/usacdec_ace_d4t64.h000066400000000000000000000121741404225310700206370ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Description: ACELP *******************************************************************************/ #ifndef USACDEC_ACE_D4T64_H #define USACDEC_ACE_D4T64_H #include "common_fix.h" /* Data type definition for the fixed codebook vector */ #define FIXP_COD FIXP_SGL #define FX_COD2FX_DBL(x) (FX_SGL2FX_DBL(x)) #define FX_DBL2FX_COD(x) FX_DBL2FX_SGL((x) + (FIXP_DBL)0x8000) #define FX_SGL2FX_COD(x) (x) #define COD_BITS FRACT_BITS void D_ACELP_decode_4t64(SHORT index[], int nbits, FIXP_COD code[]); #endif /* USACDEC_ACE_D4T64_H */ fdk-aac-2.0.2/libAACdec/src/usacdec_ace_ltp.cpp000066400000000000000000000214501404225310700211210ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Matthias Hildenbrand Description: USAC ACELP LTP filter *******************************************************************************/ #include "usacdec_ace_ltp.h" #include "genericStds.h" #include "common_fix.h" #define UP_SAMP 4 #define L_INTERPOL2 16 #define L_SUBFR 64 #define A2 FL2FX_SGL(2 * 0.18f) #define B FL2FX_SGL(0.64f) static const LONG Pred_lt4_inter4_2[UP_SAMP][L_INTERPOL2] = { {(LONG)0x0000FFFC, (LONG)0x0008FFFC, (LONG)0xFFEB004C, (LONG)0xFF50014A, (LONG)0xFDD90351, (LONG)0xFB2A06CD, (LONG)0xF6920D46, (LONG)0xEBB42B35, (LONG)0x6D9EEF39, (LONG)0x0618FE0F, (LONG)0xFFE00131, (LONG)0xFE5501C5, (LONG)0xFE5E015D, (LONG)0xFEF700B6, (LONG)0xFF920037, (LONG)0xFFEC0003}, {(LONG)0x0002FFF2, (LONG)0x0026FFBD, (LONG)0x005DFF98, (LONG)0x0055FFEF, (LONG)0xFF89015F, (LONG)0xFD3A04E5, (LONG)0xF7D90DAA, (LONG)0xE67A50EE, (LONG)0x50EEE67A, (LONG)0x0DAAF7D9, (LONG)0x04E5FD3A, (LONG)0x015FFF89, (LONG)0xFFEF0055, (LONG)0xFF98005D, (LONG)0xFFBD0026, (LONG)0xFFF20002}, {(LONG)0x0003FFEC, (LONG)0x0037FF92, (LONG)0x00B6FEF7, (LONG)0x015DFE5E, (LONG)0x01C5FE55, (LONG)0x0131FFE0, (LONG)0xFE0F0618, (LONG)0xEF396D9E, (LONG)0x2B35EBB4, (LONG)0x0D46F692, (LONG)0x06CDFB2A, (LONG)0x0351FDD9, (LONG)0x014AFF50, (LONG)0x004CFFEB, (LONG)0xFFFC0008, (LONG)0xFFFC0000}, {(LONG)0x0002FFF2, (LONG)0x002BFF9E, (LONG)0x00B9FECE, (LONG)0x01CFFD75, (LONG)0x035EFBC1, (LONG)0x0521FA0C, (LONG)0x06AAF8C9, (LONG)0x07907852, (LONG)0x0790F8C9, (LONG)0x06AAFA0C, (LONG)0x0521FBC1, (LONG)0x035EFD75, (LONG)0x01CFFECE, (LONG)0x00B9FF9E, (LONG)0x002BFFF2, (LONG)0x00020000}}; void Pred_lt4(FIXP_DBL exc[], /* in/out: excitation buffer */ int T0, /* input : integer pitch lag */ int frac /* input : fraction of lag in range 0..3 */ ) { int j; FIXP_DBL *x; const LONG *interpol; FIXP_DBL L_sumb, L_sumt; x = &exc[-T0 - L_INTERPOL2 + 1]; /* remap frac and x: 0 -> 3 x (unchanged) 1 -> 0 x-- 2 -> 1 x-- 3 -> 2 x-- */ if (--frac < 0) frac += UP_SAMP; else x--; j = L_SUBFR + 1; do { LONG filt; FIXP_DBL x0, x1; FIXP_DBL *xi = x++; interpol = Pred_lt4_inter4_2[frac]; int i = 3; filt = *interpol++; x0 = *xi++; x1 = *xi++; L_sumt = fMultDiv2(x0, (FIXP_SGL)((SHORT)(filt >> 16))); L_sumb = fMultDiv2(x1, (FIXP_SGL)((SHORT)filt)); do { filt = *interpol++; x0 = *xi++; x1 = *xi++; L_sumt = fMultAddDiv2(L_sumt, x0, (FIXP_SGL)((SHORT)(filt >> 16))); L_sumb = fMultAddDiv2(L_sumb, x1, (FIXP_SGL)((SHORT)filt)); filt = *interpol++; x0 = *xi++; x1 = *xi++; L_sumt = fMultAddDiv2(L_sumt, x0, (FIXP_SGL)((SHORT)(filt >> 16))); L_sumb = fMultAddDiv2(L_sumb, x1, (FIXP_SGL)((SHORT)filt)); filt = *interpol++; x0 = *xi++; x1 = *xi++; L_sumt = fMultAddDiv2(L_sumt, x0, (FIXP_SGL)((SHORT)(filt >> 16))); L_sumb = fMultAddDiv2(L_sumb, x1, (FIXP_SGL)((SHORT)filt)); filt = *interpol++; x0 = *xi++; x1 = *xi++; L_sumt = fMultAddDiv2(L_sumt, x0, (FIXP_SGL)((SHORT)(filt >> 16))); L_sumb = fMultAddDiv2(L_sumb, x1, (FIXP_SGL)((SHORT)filt)); filt = *interpol++; x0 = *xi++; x1 = *xi++; L_sumt = fMultAddDiv2(L_sumt, x0, (FIXP_SGL)((SHORT)(filt >> 16))); L_sumb = fMultAddDiv2(L_sumb, x1, (FIXP_SGL)((SHORT)filt)); } while (--i != 0); L_sumb <<= 1; L_sumb = fAddSaturate(L_sumt << 1, L_sumb); *exc++ = L_sumb; } while (--j != 0); return; } void Pred_lt4_postfilter(FIXP_DBL exc[] /* in/out: excitation buffer */ ) { /* exc[i] = A*exc[i-1] + B*exc[i] + A*exc[i+1] exc[i+1] = A*exc[i] + B*exc[i+1] + A*exc[i+2] ; i = 0:2:62 */ int i; FIXP_DBL sum0, sum1, a_exc0, a_exc1; a_exc0 = fMultDiv2(A2, exc[-1]); a_exc1 = fMultDiv2(A2, exc[0]); /* ARM926: 22 cycles/iteration */ for (i = 0; i < L_SUBFR; i += 2) { sum0 = a_exc0 + fMult(B, exc[i]); sum1 = a_exc1 + fMult(B, exc[i + 1]); a_exc0 = fMultDiv2(A2, exc[i + 1]); a_exc1 = fMultDiv2(A2, exc[i + 2]); exc[i] = sum0 + a_exc0; exc[i + 1] = sum1 + a_exc1; } return; } fdk-aac-2.0.2/libAACdec/src/usacdec_ace_ltp.h000066400000000000000000000131431404225310700205660ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Matthias Hildenbrand Description: USAC ACELP LTP filter *******************************************************************************/ #ifndef USACDEC_ACE_LTP_H #define USACDEC_ACE_LTP_H #include "common_fix.h" /** * \brief Compute the initial adaptive codebook excitation v'(n) by * interpolating the past excitation vector u'(n). * \param exc points to adaptive codebook of current subframe (input/output) * \param T0 integer part of decoded pitch lag (input) * \param frac fractional part of decoded pitch lag (0..3) (input) */ void Pred_lt4(FIXP_DBL exc[], /* in/out: excitation buffer */ int T0, /* input : integer pitch lag */ int frac /* input : fraction of lag */ ); /** * \brief Compute the adaptive codebook excitation v(n) in case of * ltp_filtering_flag == 0. * \param exc points to adaptive codebook of current subframe (input/output) */ void Pred_lt4_postfilter(FIXP_DBL exc[] /* in/out: excitation buffer */ ); #endif /* USACDEC_ACE_LTP_H */ fdk-aac-2.0.2/libAACdec/src/usacdec_acelp.cpp000066400000000000000000001356351404225310700206110ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Matthias Hildenbrand Description: USAC ACELP frame decoder *******************************************************************************/ #include "usacdec_acelp.h" #include "usacdec_ace_d4t64.h" #include "usacdec_ace_ltp.h" #include "usacdec_rom.h" #include "usacdec_lpc.h" #include "genericStds.h" #define PIT_FR2_12k8 128 /* Minimum pitch lag with resolution 1/2 */ #define PIT_FR1_12k8 160 /* Minimum pitch lag with resolution 1 */ #define TILT_CODE2 \ FL2FXCONST_SGL(0.3f * 2.0f) /* ACELP code pre-emphasis factor ( *2 ) */ #define PIT_SHARP \ FL2FXCONST_SGL(0.85f) /* pitch sharpening factor */ #define PREEMPH_FAC \ FL2FXCONST_SGL(0.68f) /* ACELP synth pre-emphasis factor */ #define ACELP_HEADROOM 1 #define ACELP_OUTSCALE (MDCT_OUT_HEADROOM - ACELP_HEADROOM) /** * \brief Calculate pre-emphasis (1 - mu z^-1) on input signal. * \param[in] in pointer to input signal; in[-1] is also needed. * \param[out] out pointer to output signal. * \param[in] L length of filtering. */ /* static */ void E_UTIL_preemph(const FIXP_DBL *in, FIXP_DBL *out, INT L) { int i; for (i = 0; i < L; i++) { out[i] = fAddSaturate(in[i], -fMult(PREEMPH_FAC, in[i - 1])); } return; } /** * \brief Calculate de-emphasis 1/(1 - TILT_CODE z^-1) on innovative codebook * vector. * \param[in,out] x innovative codebook vector. */ static void Preemph_code( FIXP_COD x[] /* (i/o) : input signal overwritten by the output */ ) { int i; FIXP_DBL L_tmp; /* ARM926: 12 cycles per sample */ for (i = L_SUBFR - 1; i > 0; i--) { L_tmp = FX_COD2FX_DBL(x[i]); L_tmp -= fMultDiv2(x[i - 1], TILT_CODE2); x[i] = FX_DBL2FX_COD(L_tmp); } } /** * \brief Apply pitch sharpener to the innovative codebook vector. * \param[in,out] x innovative codebook vector. * \param[in] pit_lag decoded pitch lag. */ static void Pit_shrp( FIXP_COD x[], /* in/out: impulse response (or algebraic code) */ int pit_lag /* input : pitch lag */ ) { int i; FIXP_DBL L_tmp; for (i = pit_lag; i < L_SUBFR; i++) { L_tmp = FX_COD2FX_DBL(x[i]); L_tmp += fMult(x[i - pit_lag], PIT_SHARP); x[i] = FX_DBL2FX_COD(L_tmp); } return; } /** * \brief Calculate Quantized codebook gain, Quantized pitch gain and unbiased * Innovative code vector energy. * \param[in] index index of quantizer. * \param[in] code innovative code vector with exponent = SF_CODE. * \param[out] gain_pit Quantized pitch gain g_p with exponent = SF_GAIN_P. * \param[out] gain_code Quantized codebook gain g_c. * \param[in] mean_ener mean_ener defined in open-loop (2 bits), exponent = 7. * \param[out] E_code unbiased innovative code vector energy. * \param[out] E_code_e exponent of unbiased innovative code vector energy. */ #define SF_MEAN_ENER_LG10 9 /* pow(10.0, {18, 30, 42, 54}/20.0) /(float)(1<>= 1; } else { ener_code_e -= 6; } gcode_inov = invSqrtNorm2(ener_code, &gcode0_e); gcode_inov_e = gcode0_e - (ener_code_e >> 1); if (bfi) { FIXP_DBL tgcode; FIXP_SGL tgpit; tgpit = *past_gpit; if (tgpit > FL2FXCONST_SGL(0.95f / (1 << SF_GAIN_P))) { tgpit = FL2FXCONST_SGL(0.95f / (1 << SF_GAIN_P)); } else if (tgpit < FL2FXCONST_SGL(0.5f / (1 << SF_GAIN_P))) { tgpit = FL2FXCONST_SGL(0.5f / (1 << SF_GAIN_P)); } *gain_pit = tgpit; tgpit = FX_DBL2FX_SGL(fMult(tgpit, FL2FXCONST_DBL(0.95f))); *past_gpit = tgpit; tgpit = FL2FXCONST_SGL(1.4f / (1 << SF_GAIN_P)) - tgpit; tgcode = fMult(*past_gcode, tgpit) << SF_GAIN_P; *gain_code = scaleValue(fMult(tgcode, gcode_inov), gcode_inov_e); *past_gcode = tgcode; return; } /*-------------- Decode gains ---------------*/ /* gcode0 = pow(10.0, (float)mean_ener/20.0); gcode0 = gcode0 / sqrt(ener_code/L_SUBFR); */ gcode0 = pow_10_mean_energy[mean_ener_bits]; gcode0 = fMultDiv2(gcode0, gcode_inov); gcode0_e = gcode0_e + SF_MEAN_ENER_LG10 - (ener_code_e >> 1) + 1; i = index << 1; *gain_pit = fdk_t_qua_gain7b[i]; /* adaptive codebook gain */ /* t_qua_gain[ind2p1] : fixed codebook gain correction factor */ Ltmp = fMult(fdk_t_qua_gain7b[i + 1], gcode0); *gain_code = scaleValue(Ltmp, gcode0_e - SF_GAIN_C + SF_QUA_GAIN7B); /* update bad frame handler */ *past_gpit = *gain_pit; /*-------------------------------------------------------- past_gcode = gain_code/gcode_inov --------------------------------------------------------*/ { FIXP_DBL gcode_m; INT gcode_e; gcode_m = fDivNormHighPrec(Ltmp, gcode_inov, &gcode_e); gcode_e += (gcode0_e - SF_GAIN_C + SF_QUA_GAIN7B) - (gcode_inov_e); *past_gcode = scaleValue(gcode_m, gcode_e); } } /** * \brief Calculate period/voicing factor r_v * \param[in] exc pitch excitation. * \param[in] gain_pit gain of pitch g_p. * \param[in] gain_code gain of code g_c. * \param[in] gain_code_e exponent of gain of code. * \param[in] ener_code unbiased innovative code vector energy. * \param[in] ener_code_e exponent of unbiased innovative code vector energy. * \return period/voice factor r_v (-1=unvoiced to 1=voiced), exponent SF_PFAC. */ static FIXP_DBL calc_period_factor(FIXP_DBL exc[], FIXP_SGL gain_pit, FIXP_DBL gain_code, FIXP_DBL ener_code, int ener_code_e) { int ener_exc_e, L_tmp_e, s = 0; FIXP_DBL ener_exc, L_tmp; FIXP_DBL period_fac; /* energy of pitch excitation */ ener_exc = (FIXP_DBL)0; for (int i = 0; i < L_SUBFR; i++) { ener_exc += fPow2Div2(exc[i]) >> s; if (ener_exc >= FL2FXCONST_DBL(0.5f)) { ener_exc >>= 1; s++; } } ener_exc_e = fNorm(ener_exc); ener_exc = fMult(ener_exc << ener_exc_e, fPow2(gain_pit)); if (ener_exc != (FIXP_DBL)0) { ener_exc_e = 2 * SF_EXC + 1 + 2 * SF_GAIN_P - ener_exc_e + s; } else { ener_exc_e = 0; } /* energy of innovative code excitation */ /* L_tmp = ener_code * gain_code*gain_code; */ L_tmp_e = fNorm(gain_code); L_tmp = fPow2(gain_code << L_tmp_e); L_tmp = fMult(ener_code, L_tmp); L_tmp_e = 2 * SF_GAIN_C + ener_code_e - 2 * L_tmp_e; /* Find common exponent */ { FIXP_DBL num, den; int exp_diff; exp_diff = ener_exc_e - L_tmp_e; if (exp_diff >= 0) { ener_exc >>= 1; if (exp_diff <= DFRACT_BITS - 2) { L_tmp >>= exp_diff + 1; } else { L_tmp = (FIXP_DBL)0; } den = ener_exc + L_tmp; if (ener_exc_e < DFRACT_BITS - 1) { den += scaleValue(FL2FXCONST_DBL(0.01f), -ener_exc_e - 1); } } else { if (exp_diff >= -(DFRACT_BITS - 2)) { ener_exc >>= 1 - exp_diff; } else { ener_exc = (FIXP_DBL)0; } L_tmp >>= 1; den = ener_exc + L_tmp; if (L_tmp_e < DFRACT_BITS - 1) { den += scaleValue(FL2FXCONST_DBL(0.01f), -L_tmp_e - 1); } } num = (ener_exc - L_tmp); num >>= SF_PFAC; if (den > (FIXP_DBL)0) { if (ener_exc > L_tmp) { period_fac = schur_div(num, den, 16); } else { period_fac = -schur_div(-num, den, 16); } } else { period_fac = (FIXP_DBL)MAXVAL_DBL; } } /* exponent = SF_PFAC */ return period_fac; } /*------------------------------------------------------------* * noise enhancer * * ~~~~~~~~~~~~~~ * * - Enhance excitation on noise. (modify gain of code) * * If signal is noisy and LPC filter is stable, move gain * * of code 1.5 dB toward gain of code threshold. * * This decrease by 3 dB noise energy variation. * *------------------------------------------------------------*/ /** * \brief Enhance excitation on noise. (modify gain of code) * \param[in] gain_code Quantized codebook gain g_c, exponent = SF_GAIN_C. * \param[in] period_fac periodicity factor, exponent = SF_PFAC. * \param[in] stab_fac stability factor, exponent = SF_STAB. * \param[in,out] p_gc_threshold modified gain of previous subframe. * \return gain_code smoothed gain of code g_sc, exponent = SF_GAIN_C. */ static FIXP_DBL noise_enhancer(/* (o) : smoothed gain g_sc SF_GAIN_C */ FIXP_DBL gain_code, /* (i) : Quantized codebook gain SF_GAIN_C */ FIXP_DBL period_fac, /* (i) : periodicity factor (-1=unvoiced to 1=voiced), SF_PFAC */ FIXP_SGL stab_fac, /* (i) : stability factor (0 <= ... < 1.0) SF_STAB */ FIXP_DBL *p_gc_threshold) /* (io): gain of code threshold SF_GAIN_C */ { FIXP_DBL fac, L_tmp, gc_thres; gc_thres = *p_gc_threshold; L_tmp = gain_code; if (L_tmp < gc_thres) { L_tmp += fMultDiv2(gain_code, FL2FXCONST_SGL(2.0 * 0.19f)); /* +1.5dB => *(1.0+0.19) */ if (L_tmp > gc_thres) { L_tmp = gc_thres; } } else { L_tmp = fMult(gain_code, FL2FXCONST_SGL(1.0f / 1.19f)); /* -1.5dB => *10^(-1.5/20) */ if (L_tmp < gc_thres) { L_tmp = gc_thres; } } *p_gc_threshold = L_tmp; /* voicing factor lambda = 0.5*(1-period_fac) */ /* gain smoothing factor S_m = lambda*stab_fac (=fac) = 0.5(stab_fac - stab_fac * period_fac) */ fac = (FX_SGL2FX_DBL(stab_fac) >> (SF_PFAC + 1)) - fMultDiv2(stab_fac, period_fac); /* fac_e = SF_PFAC + SF_STAB */ FDK_ASSERT(fac >= (FIXP_DBL)0); /* gain_code = (float)((fac*tmp) + ((1.0-fac)*gain_code)); */ gain_code = fMult(fac, L_tmp) - fMult(FL2FXCONST_DBL(-1.0f / (1 << (SF_PFAC + SF_STAB))) + fac, gain_code); gain_code <<= (SF_PFAC + SF_STAB); return gain_code; } /** * \brief Update adaptive codebook u'(n) (exc) * Enhance pitch of c(n) and build post-processed excitation u(n) (exc2) * \param[in] code innovative codevector c(n), exponent = SF_CODE. * \param[in,out] exc filtered adaptive codebook v(n), exponent = SF_EXC. * \param[in] gain_pit adaptive codebook gain, exponent = SF_GAIN_P. * \param[in] gain_code innovative codebook gain g_c, exponent = SF_GAIN_C. * \param[in] gain_code_smoothed smoothed innov. codebook gain g_sc, exponent = * SF_GAIN_C. * \param[in] period_fac periodicity factor r_v, exponent = SF_PFAC. * \param[out] exc2 post-processed excitation u(n), exponent = SF_EXC. */ void BuildAdaptiveExcitation( FIXP_COD code[], /* (i) : algebraic codevector c(n) Q9 */ FIXP_DBL exc[], /* (io): filtered adaptive codebook v(n) Q15 */ FIXP_SGL gain_pit, /* (i) : adaptive codebook gain g_p Q14 */ FIXP_DBL gain_code, /* (i) : innovative codebook gain g_c Q16 */ FIXP_DBL gain_code_smoothed, /* (i) : smoothed innov. codebook gain g_sc Q16 */ FIXP_DBL period_fac, /* (i) : periodicity factor r_v Q15 */ FIXP_DBL exc2[] /* (o) : post-processed excitation u(n) Q15 */ ) { /* Note: code[L_SUBFR] and exc2[L_SUBFR] share the same memory! If exc2[i] is written, code[i] will be destroyed! */ #define SF_HEADROOM (1) #define SF (SF_CODE + SF_GAIN_C + 1 - SF_EXC - SF_HEADROOM) #define SF_GAIN_P2 (SF_GAIN_P - SF_HEADROOM) int i; FIXP_DBL tmp, cpe, code_smooth_prev, code_smooth; FIXP_COD code_i; FIXP_DBL cpe_code_smooth, cpe_code_smooth_prev; /* cpe = (1+r_v)/8 * 2 ; ( SF = -1) */ cpe = (period_fac >> (2 - SF_PFAC)) + FL2FXCONST_DBL(0.25f); /* u'(n) */ tmp = fMultDiv2(*exc, gain_pit) << (SF_GAIN_P2 + 1); /* v(0)*g_p */ *exc++ = (tmp + (fMultDiv2(code[0], gain_code) << SF)) << SF_HEADROOM; /* u(n) */ code_smooth_prev = fMultDiv2(*code++, gain_code_smoothed) << SF; /* c(0) * g_sc */ code_i = *code++; code_smooth = fMultDiv2(code_i, gain_code_smoothed) << SF; /* c(1) * g_sc */ tmp += code_smooth_prev; /* tmp = v(0)*g_p + c(0)*g_sc */ cpe_code_smooth = fMultDiv2(cpe, code_smooth); *exc2++ = (tmp - cpe_code_smooth) << SF_HEADROOM; cpe_code_smooth_prev = fMultDiv2(cpe, code_smooth_prev); i = L_SUBFR - 2; do /* ARM926: 22 cycles per iteration */ { /* u'(n) */ tmp = fMultDiv2(*exc, gain_pit) << (SF_GAIN_P2 + 1); *exc++ = (tmp + (fMultDiv2(code_i, gain_code) << SF)) << SF_HEADROOM; /* u(n) */ tmp += code_smooth; /* += g_sc * c(i) */ tmp -= cpe_code_smooth_prev; cpe_code_smooth_prev = cpe_code_smooth; code_i = *code++; code_smooth = fMultDiv2(code_i, gain_code_smoothed) << SF; cpe_code_smooth = fMultDiv2(cpe, code_smooth); *exc2++ = (tmp - cpe_code_smooth) << SF_HEADROOM; /* tmp - c_pe * g_sc * c(i+1) */ } while (--i != 0); /* u'(n) */ tmp = fMultDiv2(*exc, gain_pit) << (SF_GAIN_P2 + 1); *exc = (tmp + (fMultDiv2(code_i, gain_code) << SF)) << SF_HEADROOM; /* u(n) */ tmp += code_smooth; tmp -= cpe_code_smooth_prev; *exc2++ = tmp << SF_HEADROOM; return; } /** * \brief Interpolate LPC vector in LSP domain for current subframe and convert * to LP domain * \param[in] lsp_old LPC vector (LSP domain) corresponding to the beginning of * current ACELP frame. * \param[in] lsp_new LPC vector (LSP domain) corresponding to the end of * current ACELP frame. * \param[in] subfr_nr number of current ACELP subframe 0..3. * \param[in] nb_subfr total number of ACELP subframes in this frame. * \param[out] A LP filter coefficients for current ACELP subframe, exponent = * SF_A_COEFFS. */ /* static */ void int_lpc_acelp( const FIXP_LPC lsp_old[], /* input : LSPs from past frame */ const FIXP_LPC lsp_new[], /* input : LSPs from present frame */ int subfr_nr, int nb_subfr, FIXP_LPC A[], /* output: interpolated LP coefficients for current subframe */ INT *A_exp) { int i; FIXP_LPC lsp_interpol[M_LP_FILTER_ORDER]; FIXP_SGL fac_old, fac_new; FDK_ASSERT((nb_subfr == 3) || (nb_subfr == 4)); fac_old = lsp_interpol_factor[nb_subfr & 0x1][(nb_subfr - 1) - subfr_nr]; fac_new = lsp_interpol_factor[nb_subfr & 0x1][subfr_nr]; for (i = 0; i < M_LP_FILTER_ORDER; i++) { lsp_interpol[i] = FX_DBL2FX_LPC( (fMultDiv2(lsp_old[i], fac_old) + fMultDiv2(lsp_new[i], fac_new)) << 1); } E_LPC_f_lsp_a_conversion(lsp_interpol, A, A_exp); return; } /** * \brief Perform LP synthesis by filtering the post-processed excitation u(n) * through the LP synthesis filter 1/A(z) * \param[in] a LP filter coefficients, exponent = SF_A_COEFFS. * \param[in] length length of input/output signal. * \param[in] x post-processed excitation u(n). * \param[in,out] y LP synthesis signal and filter memory * y[-M_LP_FILTER_ORDER..-1]. */ /* static */ void Syn_filt(const FIXP_LPC a[], /* (i) : a[m] prediction coefficients Q12 */ const INT a_exp, INT length, /* (i) : length of input/output signal (64|128) */ FIXP_DBL x[], /* (i) : input signal Qx */ FIXP_DBL y[] /* (i/o) : filter states / output signal Qx-s*/ ) { int i, j; FIXP_DBL L_tmp; for (i = 0; i < length; i++) { L_tmp = (FIXP_DBL)0; for (j = 0; j < M_LP_FILTER_ORDER; j++) { L_tmp -= fMultDiv2(a[j], y[i - (j + 1)]) >> (LP_FILTER_SCALE - 1); } L_tmp = scaleValue(L_tmp, a_exp + LP_FILTER_SCALE); y[i] = fAddSaturate(L_tmp, x[i]); } return; } /** * \brief Calculate de-emphasis 1/(1 - mu z^-1) on input signal. * \param[in] x input signal. * \param[out] y output signal. * \param[in] L length of signal. * \param[in,out] mem memory (signal[-1]). */ /* static */ void Deemph(FIXP_DBL *x, FIXP_DBL *y, int L, FIXP_DBL *mem) { int i; FIXP_DBL yi = *mem; for (i = 0; i < L; i++) { FIXP_DBL xi = x[i] >> 1; xi = fMultAddDiv2(xi, PREEMPH_FAC, yi); yi = SATURATE_LEFT_SHIFT(xi, 1, 32); y[i] = yi; } *mem = yi; return; } /** * \brief Compute the LP residual by filtering the input speech through the * analysis filter A(z). * \param[in] a LP filter coefficients, exponent = SF_A_COEFFS * \param[in] x input signal (note that values x[-m..-1] are needed), exponent = * SF_SYNTH * \param[out] y output signal (residual), exponent = SF_EXC * \param[in] l length of filtering */ /* static */ void E_UTIL_residu(const FIXP_LPC *a, const INT a_exp, FIXP_DBL *x, FIXP_DBL *y, INT l) { FIXP_DBL s; INT i, j; /* (note that values x[-m..-1] are needed) */ for (i = 0; i < l; i++) { s = (FIXP_DBL)0; for (j = 0; j < M_LP_FILTER_ORDER; j++) { s += fMultDiv2(a[j], x[i - j - 1]) >> (LP_FILTER_SCALE - 1); } s = scaleValue(s, a_exp + LP_FILTER_SCALE); y[i] = fAddSaturate(s, x[i]); } return; } /* use to map subfr number to number of bits used for acb_index */ static const UCHAR num_acb_idx_bits_table[2][NB_SUBFR] = { {9, 6, 9, 6}, /* coreCoderFrameLength == 1024 */ {9, 6, 6, 0} /* coreCoderFrameLength == 768 */ }; static int DecodePitchLag(HANDLE_FDK_BITSTREAM hBs, const UCHAR num_acb_idx_bits, const int PIT_MIN, /* TMIN */ const int PIT_FR2, /* TFR2 */ const int PIT_FR1, /* TFR1 */ const int PIT_MAX, /* TMAX */ int *pT0, int *pT0_frac, int *pT0_min, int *pT0_max) { int acb_idx; int error = 0; int T0, T0_frac; FDK_ASSERT((num_acb_idx_bits == 9) || (num_acb_idx_bits == 6)); acb_idx = FDKreadBits(hBs, num_acb_idx_bits); if (num_acb_idx_bits == 6) { /* When the pitch value is encoded on 6 bits, a pitch resolution of 1/4 is always used in the range [T1-8, T1+7.75], where T1 is nearest integer to the fractional pitch lag of the previous subframe. */ T0 = *pT0_min + acb_idx / 4; T0_frac = acb_idx & 0x3; } else { /* num_acb_idx_bits == 9 */ /* When the pitch value is encoded on 9 bits, a fractional pitch delay is used with resolutions 0.25 in the range [TMIN, TFR2-0.25], resolutions 0.5 in the range [TFR2, TFR1-0.5], and integers only in the range [TFR1, TMAX]. NOTE: for small sampling rates TMAX can get smaller than TFR1. */ int T0_min, T0_max; if (acb_idx < (PIT_FR2 - PIT_MIN) * 4) { /* first interval with 0.25 pitch resolution */ T0 = PIT_MIN + (acb_idx / 4); T0_frac = acb_idx & 0x3; } else if (acb_idx < ((PIT_FR2 - PIT_MIN) * 4 + (PIT_FR1 - PIT_FR2) * 2)) { /* second interval with 0.5 pitch resolution */ acb_idx -= (PIT_FR2 - PIT_MIN) * 4; T0 = PIT_FR2 + (acb_idx / 2); T0_frac = (acb_idx & 0x1) * 2; } else { /* third interval with 1.0 pitch resolution */ T0 = acb_idx + PIT_FR1 - ((PIT_FR2 - PIT_MIN) * 4) - ((PIT_FR1 - PIT_FR2) * 2); T0_frac = 0; } /* find T0_min and T0_max for subframe 1 or 3 */ T0_min = T0 - 8; if (T0_min < PIT_MIN) { T0_min = PIT_MIN; } T0_max = T0_min + 15; if (T0_max > PIT_MAX) { T0_max = PIT_MAX; T0_min = T0_max - 15; } *pT0_min = T0_min; *pT0_max = T0_max; } *pT0 = T0; *pT0_frac = T0_frac; return error; } static void ConcealPitchLag(CAcelpStaticMem *acelp_mem, const int PIT_MAX, int *pT0, int *pT0_frac) { USHORT *pold_T0 = &acelp_mem->old_T0; UCHAR *pold_T0_frac = &acelp_mem->old_T0_frac; if ((int)*pold_T0 >= PIT_MAX) { *pold_T0 = (UCHAR)(PIT_MAX - 5); } *pT0 = (int)*pold_T0; *pT0_frac = (int)*pold_T0_frac; } static UCHAR tab_coremode2nbits[8] = {20, 28, 36, 44, 52, 64, 12, 16}; static int MapCoreMode2NBits(int core_mode) { return (int)tab_coremode2nbits[core_mode]; } void CLpd_AcelpDecode(CAcelpStaticMem *acelp_mem, INT i_offset, const FIXP_LPC lsp_old[M_LP_FILTER_ORDER], const FIXP_LPC lsp_new[M_LP_FILTER_ORDER], FIXP_SGL stab_fac, CAcelpChannelData *pAcelpData, INT numLostSubframes, int lastLpcLost, int frameCnt, FIXP_DBL synth[], int pT[], FIXP_DBL *pit_gain, INT coreCoderFrameLength) { int i_subfr, subfr_nr, l_div, T; int T0 = -1, T0_frac = -1; /* mark invalid */ int pit_gain_index = 0; const int PIT_MAX = PIT_MAX_12k8 + (6 * i_offset); /* maximum pitch lag */ FIXP_COD *code; FIXP_DBL *exc2; FIXP_DBL *syn; FIXP_DBL *exc; FIXP_LPC A[M_LP_FILTER_ORDER]; INT A_exp; FIXP_DBL period_fac; FIXP_SGL gain_pit; FIXP_DBL gain_code, gain_code_smooth, Ener_code; int Ener_code_e; int n; int bfi = (numLostSubframes > 0) ? 1 : 0; C_ALLOC_SCRATCH_START( exc_buf, FIXP_DBL, PIT_MAX_MAX + L_INTERPOL + L_DIV + 1); /* 411 + 17 + 256 + 1 = 685 */ C_ALLOC_SCRATCH_START(syn_buf, FIXP_DBL, M_LP_FILTER_ORDER + L_DIV); /* 16 + 256 = 272 */ /* use same memory for code[L_SUBFR] and exc2[L_SUBFR] */ C_ALLOC_SCRATCH_START(tmp_buf, FIXP_DBL, L_SUBFR); /* 64 */ /* make sure they don't overlap if they are accessed alternatingly in * BuildAdaptiveExcitation() */ #if (COD_BITS == FRACT_BITS) code = (FIXP_COD *)(tmp_buf + L_SUBFR / 2); #elif (COD_BITS == DFRACT_BITS) code = (FIXP_COD *)tmp_buf; #endif exc2 = (FIXP_DBL *)tmp_buf; syn = syn_buf + M_LP_FILTER_ORDER; exc = exc_buf + PIT_MAX_MAX + L_INTERPOL; FDKmemcpy(syn_buf, acelp_mem->old_syn_mem, M_LP_FILTER_ORDER * sizeof(FIXP_DBL)); FDKmemcpy(exc_buf, acelp_mem->old_exc_mem, (PIT_MAX_MAX + L_INTERPOL) * sizeof(FIXP_DBL)); FDKmemclear(exc_buf + (PIT_MAX_MAX + L_INTERPOL), (L_DIV + 1) * sizeof(FIXP_DBL)); l_div = coreCoderFrameLength / NB_DIV; for (i_subfr = 0, subfr_nr = 0; i_subfr < l_div; i_subfr += L_SUBFR, subfr_nr++) { /*-------------------------------------------------* * - Decode pitch lag (T0 and T0_frac) * *-------------------------------------------------*/ if (bfi) { ConcealPitchLag(acelp_mem, PIT_MAX, &T0, &T0_frac); } else { T0 = (int)pAcelpData->T0[subfr_nr]; T0_frac = (int)pAcelpData->T0_frac[subfr_nr]; } /*-------------------------------------------------* * - Find the pitch gain, the interpolation filter * * and the adaptive codebook vector. * *-------------------------------------------------*/ Pred_lt4(&exc[i_subfr], T0, T0_frac); if ((!bfi && pAcelpData->ltp_filtering_flag[subfr_nr] == 0) || (bfi && numLostSubframes == 1 && stab_fac < FL2FXCONST_SGL(0.25f))) { /* find pitch excitation with lp filter: v'(n) => v(n) */ Pred_lt4_postfilter(&exc[i_subfr]); } /*-------------------------------------------------------* * - Decode innovative codebook. * * - Add the fixed-gain pitch contribution to code[]. * *-------------------------------------------------------*/ if (bfi) { for (n = 0; n < L_SUBFR; n++) { code[n] = FX_SGL2FX_COD((FIXP_SGL)E_UTIL_random(&acelp_mem->seed_ace)) >> 4; } } else { int nbits = MapCoreMode2NBits((int)pAcelpData->acelp_core_mode); D_ACELP_decode_4t64(pAcelpData->icb_index[subfr_nr], nbits, &code[0]); } T = T0; if (T0_frac > 2) { T += 1; } Preemph_code(code); Pit_shrp(code, T); /* Output pitch lag for bass post-filter */ if (T > PIT_MAX) { pT[subfr_nr] = PIT_MAX; } else { pT[subfr_nr] = T; } D_gain2_plus( pAcelpData->gains[subfr_nr], code, /* (i) : Innovative code vector, exponent = SF_CODE */ &gain_pit, /* (o) : Quantized pitch gain, exponent = SF_GAIN_P */ &gain_code, /* (o) : Quantized codebook gain */ pAcelpData ->mean_energy, /* (i) : mean_ener defined in open-loop (2 bits) */ bfi, &acelp_mem->past_gpit, &acelp_mem->past_gcode, &Ener_code, /* (o) : Innovative code vector energy */ &Ener_code_e); /* (o) : Innovative code vector energy exponent */ pit_gain[pit_gain_index++] = FX_SGL2FX_DBL(gain_pit); /* calc periodicity factor r_v */ period_fac = calc_period_factor(/* (o) : factor (-1=unvoiced to 1=voiced) */ &exc[i_subfr], /* (i) : pitch excitation, exponent = SF_EXC */ gain_pit, /* (i) : gain of pitch, exponent = SF_GAIN_P */ gain_code, /* (i) : gain of code */ Ener_code, /* (i) : Energy of code[] */ Ener_code_e); /* (i) : Exponent of energy of code[] */ if (lastLpcLost && frameCnt == 0) { if (gain_pit > FL2FXCONST_SGL(1.0f / (1 << SF_GAIN_P))) { gain_pit = FL2FXCONST_SGL(1.0f / (1 << SF_GAIN_P)); } } gain_code_smooth = noise_enhancer(/* (o) : smoothed gain g_sc exponent = SF_GAIN_C */ gain_code, /* (i) : Quantized codebook gain */ period_fac, /* (i) : periodicity factor (-1=unvoiced to 1=voiced) */ stab_fac, /* (i) : stability factor (0 <= ... < 1), exponent = 1 */ &acelp_mem->gc_threshold); /* Compute adaptive codebook update u'(n), pitch enhancement c'(n) and * post-processed excitation u(n). */ BuildAdaptiveExcitation(code, exc + i_subfr, gain_pit, gain_code, gain_code_smooth, period_fac, exc2); /* Interpolate filter coeffs for current subframe in lsp domain and convert * to LP domain */ int_lpc_acelp(lsp_old, /* input : LSPs from past frame */ lsp_new, /* input : LSPs from present frame */ subfr_nr, /* input : ACELP subframe index */ coreCoderFrameLength / L_DIV, A, /* output: LP coefficients of this subframe */ &A_exp); Syn_filt(A, /* (i) : a[m] prediction coefficients */ A_exp, L_SUBFR, /* (i) : length */ exc2, /* (i) : input signal */ &syn[i_subfr] /* (i/o) : filter states / output signal */ ); } /* end of subframe loop */ /* update pitch value for bfi procedure */ acelp_mem->old_T0_frac = T0_frac; acelp_mem->old_T0 = T0; /* save old excitation and old synthesis memory for next ACELP frame */ FDKmemcpy(acelp_mem->old_exc_mem, exc + l_div - (PIT_MAX_MAX + L_INTERPOL), sizeof(FIXP_DBL) * (PIT_MAX_MAX + L_INTERPOL)); FDKmemcpy(acelp_mem->old_syn_mem, syn_buf + l_div, sizeof(FIXP_DBL) * M_LP_FILTER_ORDER); Deemph(syn, synth, l_div, &acelp_mem->de_emph_mem); /* ref soft: mem = synth[-1] */ scaleValues(synth, l_div, -ACELP_OUTSCALE); acelp_mem->deemph_mem_wsyn = acelp_mem->de_emph_mem; C_ALLOC_SCRATCH_END(tmp_buf, FIXP_DBL, L_SUBFR); C_ALLOC_SCRATCH_END(syn_buf, FIXP_DBL, M_LP_FILTER_ORDER + L_DIV); C_ALLOC_SCRATCH_END(exc_buf, FIXP_DBL, PIT_MAX_MAX + L_INTERPOL + L_DIV + 1); return; } void CLpd_AcelpReset(CAcelpStaticMem *acelp) { acelp->gc_threshold = (FIXP_DBL)0; acelp->past_gpit = (FIXP_SGL)0; acelp->past_gcode = (FIXP_DBL)0; acelp->old_T0 = 64; acelp->old_T0_frac = 0; acelp->deemph_mem_wsyn = (FIXP_DBL)0; acelp->wsyn_rms = (FIXP_DBL)0; acelp->seed_ace = 0; } /* TCX time domain concealment */ /* Compare to figure 13a on page 54 in 3GPP TS 26.290 */ void CLpd_TcxTDConceal(CAcelpStaticMem *acelp_mem, SHORT *pitch, const FIXP_LPC lsp_old[M_LP_FILTER_ORDER], const FIXP_LPC lsp_new[M_LP_FILTER_ORDER], const FIXP_SGL stab_fac, INT nLostSf, FIXP_DBL synth[], INT coreCoderFrameLength, UCHAR last_tcx_noise_factor) { /* repeat past excitation with pitch from previous decoded TCX frame */ C_ALLOC_SCRATCH_START( exc_buf, FIXP_DBL, PIT_MAX_MAX + L_INTERPOL + L_DIV); /* 411 + 17 + 256 + 1 = */ C_ALLOC_SCRATCH_START(syn_buf, FIXP_DBL, M_LP_FILTER_ORDER + L_DIV); /* 256 + 16 = */ /* += */ FIXP_DBL ns_buf[L_DIV + 1]; FIXP_DBL *syn = syn_buf + M_LP_FILTER_ORDER; FIXP_DBL *exc = exc_buf + PIT_MAX_MAX + L_INTERPOL; FIXP_DBL *ns = ns_buf + 1; FIXP_DBL tmp, fact_exc; INT T = fMin(*pitch, (SHORT)PIT_MAX_MAX); int i, i_subfr, subfr_nr; int lDiv = coreCoderFrameLength / NB_DIV; FDKmemcpy(syn_buf, acelp_mem->old_syn_mem, M_LP_FILTER_ORDER * sizeof(FIXP_DBL)); FDKmemcpy(exc_buf, acelp_mem->old_exc_mem, (PIT_MAX_MAX + L_INTERPOL) * sizeof(FIXP_DBL)); /* if we lost all packets (i.e. 1 packet of TCX-20 ms, 2 packets of the TCX-40 ms or 4 packets of the TCX-80ms), we lost the whole coded frame extrapolation strategy: repeat lost excitation and use extrapolated LSFs */ /* AMR-WB+ like TCX TD concealment */ /* number of lost frame cmpt */ if (nLostSf < 2) { fact_exc = FL2FXCONST_DBL(0.8f); } else { fact_exc = FL2FXCONST_DBL(0.4f); } /* repeat past excitation */ for (i = 0; i < lDiv; i++) { exc[i] = fMult(fact_exc, exc[i - T]); } tmp = fMult(fact_exc, acelp_mem->wsyn_rms); acelp_mem->wsyn_rms = tmp; /* init deemph_mem_wsyn */ acelp_mem->deemph_mem_wsyn = exc[-1]; ns[-1] = acelp_mem->deemph_mem_wsyn; for (i_subfr = 0, subfr_nr = 0; i_subfr < lDiv; i_subfr += L_SUBFR, subfr_nr++) { FIXP_DBL tRes[L_SUBFR]; FIXP_LPC A[M_LP_FILTER_ORDER]; INT A_exp; /* interpolate LPC coefficients */ int_lpc_acelp(lsp_old, lsp_new, subfr_nr, lDiv / L_SUBFR, A, &A_exp); Syn_filt(A, /* (i) : a[m] prediction coefficients */ A_exp, L_SUBFR, /* (i) : length */ &exc[i_subfr], /* (i) : input signal */ &syn[i_subfr] /* (i/o) : filter states / output signal */ ); E_LPC_a_weight( A, A, M_LP_FILTER_ORDER); /* overwrite A as it is not needed any longer */ E_UTIL_residu(A, A_exp, &syn[i_subfr], tRes, L_SUBFR); Deemph(tRes, &ns[i_subfr], L_SUBFR, &acelp_mem->deemph_mem_wsyn); /* Amplitude limiter (saturate at wsyn_rms) */ for (i = i_subfr; i < i_subfr + L_SUBFR; i++) { if (ns[i] > tmp) { ns[i] = tmp; } else { if (ns[i] < -tmp) { ns[i] = -tmp; } } } E_UTIL_preemph(&ns[i_subfr], tRes, L_SUBFR); Syn_filt(A, /* (i) : a[m] prediction coefficients */ A_exp, L_SUBFR, /* (i) : length */ tRes, /* (i) : input signal */ &syn[i_subfr] /* (i/o) : filter states / output signal */ ); FDKmemmove(&synth[i_subfr], &syn[i_subfr], L_SUBFR * sizeof(FIXP_DBL)); } /* save old excitation and old synthesis memory for next ACELP frame */ FDKmemcpy(acelp_mem->old_exc_mem, exc + lDiv - (PIT_MAX_MAX + L_INTERPOL), sizeof(FIXP_DBL) * (PIT_MAX_MAX + L_INTERPOL)); FDKmemcpy(acelp_mem->old_syn_mem, syn_buf + lDiv, sizeof(FIXP_DBL) * M_LP_FILTER_ORDER); acelp_mem->de_emph_mem = acelp_mem->deemph_mem_wsyn; C_ALLOC_SCRATCH_END(syn_buf, FIXP_DBL, M_LP_FILTER_ORDER + L_DIV); C_ALLOC_SCRATCH_END(exc_buf, FIXP_DBL, PIT_MAX_MAX + L_INTERPOL + L_DIV); } void Acelp_PreProcessing(FIXP_DBL *synth_buf, FIXP_DBL *old_synth, INT *pitch, INT *old_T_pf, FIXP_DBL *pit_gain, FIXP_DBL *old_gain_pf, INT samplingRate, INT *i_offset, INT coreCoderFrameLength, INT synSfd, INT nbSubfrSuperfr) { int n; /* init beginning of synth_buf with old synthesis from previous frame */ FDKmemcpy(synth_buf, old_synth, sizeof(FIXP_DBL) * (PIT_MAX_MAX - BPF_DELAY)); /* calculate pitch lag offset for ACELP decoder */ *i_offset = (samplingRate * PIT_MIN_12k8 + (FSCALE_DENOM / 2)) / FSCALE_DENOM - PIT_MIN_12k8; /* for bass postfilter */ for (n = 0; n < synSfd; n++) { pitch[n] = old_T_pf[n]; pit_gain[n] = old_gain_pf[n]; } for (n = 0; n < nbSubfrSuperfr; n++) { pitch[n + synSfd] = L_SUBFR; pit_gain[n + synSfd] = (FIXP_DBL)0; } } void Acelp_PostProcessing(FIXP_DBL *synth_buf, FIXP_DBL *old_synth, INT *pitch, INT *old_T_pf, INT coreCoderFrameLength, INT synSfd, INT nbSubfrSuperfr) { int n; /* store last part of synth_buf (which is not handled by the IMDCT overlap) * for next frame */ FDKmemcpy(old_synth, synth_buf + coreCoderFrameLength, sizeof(FIXP_DBL) * (PIT_MAX_MAX - BPF_DELAY)); /* for bass postfilter */ for (n = 0; n < synSfd; n++) { old_T_pf[n] = pitch[nbSubfrSuperfr + n]; } } #define L_FAC_ZIR (LFAC) void CLpd_Acelp_Zir(const FIXP_LPC A[], const INT A_exp, CAcelpStaticMem *acelp_mem, const INT length, FIXP_DBL zir[], int doDeemph) { C_ALLOC_SCRATCH_START(tmp_buf, FIXP_DBL, L_FAC_ZIR + M_LP_FILTER_ORDER); FDK_ASSERT(length <= L_FAC_ZIR); FDKmemcpy(tmp_buf, acelp_mem->old_syn_mem, M_LP_FILTER_ORDER * sizeof(FIXP_DBL)); FDKmemset(tmp_buf + M_LP_FILTER_ORDER, 0, L_FAC_ZIR * sizeof(FIXP_DBL)); Syn_filt(A, A_exp, length, &tmp_buf[M_LP_FILTER_ORDER], &tmp_buf[M_LP_FILTER_ORDER]); if (!doDeemph) { /* if last lpd mode was TD concealment, then bypass deemph */ FDKmemcpy(zir, tmp_buf, length * sizeof(*zir)); } else { Deemph(&tmp_buf[M_LP_FILTER_ORDER], &zir[0], length, &acelp_mem->de_emph_mem); scaleValues(zir, length, -ACELP_OUTSCALE); } C_ALLOC_SCRATCH_END(tmp_buf, FIXP_DBL, L_FAC_ZIR + M_LP_FILTER_ORDER); } void CLpd_AcelpPrepareInternalMem(const FIXP_DBL *synth, UCHAR last_lpd_mode, UCHAR last_last_lpd_mode, const FIXP_LPC *A_new, const INT A_new_exp, const FIXP_LPC *A_old, const INT A_old_exp, CAcelpStaticMem *acelp_mem, INT coreCoderFrameLength, INT clearOldExc, UCHAR lpd_mode) { int l_div = coreCoderFrameLength / NB_DIV; /* length of one ACELP/TCX20 frame */ int l_div_partial; FIXP_DBL *syn, *old_exc_mem; C_ALLOC_SCRATCH_START(synth_buf, FIXP_DBL, PIT_MAX_MAX + L_INTERPOL + M_LP_FILTER_ORDER); syn = &synth_buf[M_LP_FILTER_ORDER]; l_div_partial = PIT_MAX_MAX + L_INTERPOL - l_div; old_exc_mem = acelp_mem->old_exc_mem; if (lpd_mode == 4) { /* Bypass Domain conversion. TCXTD Concealment does no deemphasis in the * end. */ FDKmemcpy( synth_buf, &synth[-(PIT_MAX_MAX + L_INTERPOL + M_LP_FILTER_ORDER)], (PIT_MAX_MAX + L_INTERPOL + M_LP_FILTER_ORDER) * sizeof(FIXP_DBL)); /* Set deemphasis memory state for TD concealment */ acelp_mem->deemph_mem_wsyn = scaleValueSaturate(synth[-1], ACELP_OUTSCALE); } else { /* convert past [PIT_MAX_MAX+L_INTERPOL+M_LP_FILTER_ORDER] synthesis to * preemph domain */ E_UTIL_preemph(&synth[-(PIT_MAX_MAX + L_INTERPOL + M_LP_FILTER_ORDER)], synth_buf, PIT_MAX_MAX + L_INTERPOL + M_LP_FILTER_ORDER); scaleValuesSaturate(synth_buf, PIT_MAX_MAX + L_INTERPOL + M_LP_FILTER_ORDER, ACELP_OUTSCALE); } /* Set deemphasis memory state */ acelp_mem->de_emph_mem = scaleValueSaturate(synth[-1], ACELP_OUTSCALE); /* update acelp synth filter memory */ FDKmemcpy(acelp_mem->old_syn_mem, &syn[PIT_MAX_MAX + L_INTERPOL - M_LP_FILTER_ORDER], M_LP_FILTER_ORDER * sizeof(FIXP_DBL)); if (clearOldExc) { FDKmemclear(old_exc_mem, (PIT_MAX_MAX + L_INTERPOL) * sizeof(FIXP_DBL)); C_ALLOC_SCRATCH_END(synth_buf, FIXP_DBL, PIT_MAX_MAX + L_INTERPOL + M_LP_FILTER_ORDER); return; } /* update past [PIT_MAX_MAX+L_INTERPOL] samples of exc memory */ if (last_lpd_mode == 1) { /* last frame was TCX20 */ if (last_last_lpd_mode == 0) { /* ACELP -> TCX20 -> ACELP transition */ /* Delay valid part of excitation buffer (from previous ACELP frame) by * l_div samples */ FDKmemmove(old_exc_mem, old_exc_mem + l_div, sizeof(FIXP_DBL) * l_div_partial); } else if (last_last_lpd_mode > 0) { /* TCX -> TCX20 -> ACELP transition */ E_UTIL_residu(A_old, A_old_exp, syn, old_exc_mem, l_div_partial); } E_UTIL_residu(A_new, A_new_exp, syn + l_div_partial, old_exc_mem + l_div_partial, l_div); } else { /* prev frame was FD, TCX40 or TCX80 */ int exc_A_new_length = (coreCoderFrameLength / 2 > PIT_MAX_MAX + L_INTERPOL) ? PIT_MAX_MAX + L_INTERPOL : coreCoderFrameLength / 2; int exc_A_old_length = PIT_MAX_MAX + L_INTERPOL - exc_A_new_length; E_UTIL_residu(A_old, A_old_exp, syn, old_exc_mem, exc_A_old_length); E_UTIL_residu(A_new, A_new_exp, &syn[exc_A_old_length], &old_exc_mem[exc_A_old_length], exc_A_new_length); } C_ALLOC_SCRATCH_END(synth_buf, FIXP_DBL, PIT_MAX_MAX + L_INTERPOL + M_LP_FILTER_ORDER); return; } FIXP_DBL *CLpd_ACELP_GetFreeExcMem(CAcelpStaticMem *acelp_mem, INT length) { FDK_ASSERT(length <= PIT_MAX_MAX + L_INTERPOL); return acelp_mem->old_exc_mem; } INT CLpd_AcelpRead(HANDLE_FDK_BITSTREAM hBs, CAcelpChannelData *acelp, INT acelp_core_mode, INT coreCoderFrameLength, INT i_offset) { int nb_subfr = coreCoderFrameLength / L_DIV; const UCHAR *num_acb_index_bits = (nb_subfr == 4) ? num_acb_idx_bits_table[0] : num_acb_idx_bits_table[1]; int nbits; int error = 0; const int PIT_MIN = PIT_MIN_12k8 + i_offset; const int PIT_FR2 = PIT_FR2_12k8 - i_offset; const int PIT_FR1 = PIT_FR1_12k8; const int PIT_MAX = PIT_MAX_12k8 + (6 * i_offset); int T0, T0_frac, T0_min = 0, T0_max; if (PIT_MAX > PIT_MAX_MAX) { error = AAC_DEC_DECODE_FRAME_ERROR; goto bail; } acelp->acelp_core_mode = acelp_core_mode; nbits = MapCoreMode2NBits(acelp_core_mode); /* decode mean energy with 2 bits : 18, 30, 42 or 54 dB */ acelp->mean_energy = FDKreadBits(hBs, 2); for (int sfr = 0; sfr < nb_subfr; sfr++) { /* read ACB index and store T0 and T0_frac for each ACELP subframe. */ error = DecodePitchLag(hBs, num_acb_index_bits[sfr], PIT_MIN, PIT_FR2, PIT_FR1, PIT_MAX, &T0, &T0_frac, &T0_min, &T0_max); if (error) { goto bail; } acelp->T0[sfr] = (USHORT)T0; acelp->T0_frac[sfr] = (UCHAR)T0_frac; acelp->ltp_filtering_flag[sfr] = FDKreadBits(hBs, 1); switch (nbits) { case 12: /* 12 bits AMR-WB codebook is used */ acelp->icb_index[sfr][0] = FDKreadBits(hBs, 1); acelp->icb_index[sfr][1] = FDKreadBits(hBs, 5); acelp->icb_index[sfr][2] = FDKreadBits(hBs, 1); acelp->icb_index[sfr][3] = FDKreadBits(hBs, 5); break; case 16: /* 16 bits AMR-WB codebook is used */ acelp->icb_index[sfr][0] = FDKreadBits(hBs, 1); acelp->icb_index[sfr][1] = FDKreadBits(hBs, 5); acelp->icb_index[sfr][2] = FDKreadBits(hBs, 5); acelp->icb_index[sfr][3] = FDKreadBits(hBs, 5); break; case 20: /* 20 bits AMR-WB codebook is used */ acelp->icb_index[sfr][0] = FDKreadBits(hBs, 5); acelp->icb_index[sfr][1] = FDKreadBits(hBs, 5); acelp->icb_index[sfr][2] = FDKreadBits(hBs, 5); acelp->icb_index[sfr][3] = FDKreadBits(hBs, 5); break; case 28: /* 28 bits AMR-WB codebook is used */ acelp->icb_index[sfr][0] = FDKreadBits(hBs, 9); acelp->icb_index[sfr][1] = FDKreadBits(hBs, 9); acelp->icb_index[sfr][2] = FDKreadBits(hBs, 5); acelp->icb_index[sfr][3] = FDKreadBits(hBs, 5); break; case 36: /* 36 bits AMR-WB codebook is used */ acelp->icb_index[sfr][0] = FDKreadBits(hBs, 9); acelp->icb_index[sfr][1] = FDKreadBits(hBs, 9); acelp->icb_index[sfr][2] = FDKreadBits(hBs, 9); acelp->icb_index[sfr][3] = FDKreadBits(hBs, 9); break; case 44: /* 44 bits AMR-WB codebook is used */ acelp->icb_index[sfr][0] = FDKreadBits(hBs, 13); acelp->icb_index[sfr][1] = FDKreadBits(hBs, 13); acelp->icb_index[sfr][2] = FDKreadBits(hBs, 9); acelp->icb_index[sfr][3] = FDKreadBits(hBs, 9); break; case 52: /* 52 bits AMR-WB codebook is used */ acelp->icb_index[sfr][0] = FDKreadBits(hBs, 13); acelp->icb_index[sfr][1] = FDKreadBits(hBs, 13); acelp->icb_index[sfr][2] = FDKreadBits(hBs, 13); acelp->icb_index[sfr][3] = FDKreadBits(hBs, 13); break; case 64: /* 64 bits AMR-WB codebook is used */ acelp->icb_index[sfr][0] = FDKreadBits(hBs, 2); acelp->icb_index[sfr][1] = FDKreadBits(hBs, 2); acelp->icb_index[sfr][2] = FDKreadBits(hBs, 2); acelp->icb_index[sfr][3] = FDKreadBits(hBs, 2); acelp->icb_index[sfr][4] = FDKreadBits(hBs, 14); acelp->icb_index[sfr][5] = FDKreadBits(hBs, 14); acelp->icb_index[sfr][6] = FDKreadBits(hBs, 14); acelp->icb_index[sfr][7] = FDKreadBits(hBs, 14); break; default: FDK_ASSERT(0); break; } acelp->gains[sfr] = FDKreadBits(hBs, 7); } bail: return error; } fdk-aac-2.0.2/libAACdec/src/usacdec_acelp.h000066400000000000000000000307121404225310700202440ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Matthias Hildenbrand Description: USAC ACELP frame decoder *******************************************************************************/ #ifndef USACDEC_ACELP_H #define USACDEC_ACELP_H #include "common_fix.h" #include "FDK_bitstream.h" #include "usacdec_const.h" #include "usacdec_rom.h" //#define ENHANCED_TCX_TD_CONCEAL_ENABLE /** Structure which holds the ACELP internal persistent memory */ typedef struct { FIXP_DBL old_exc_mem[PIT_MAX_MAX + L_INTERPOL]; FIXP_DBL old_syn_mem[M_LP_FILTER_ORDER]; /* synthesis filter states */ FIXP_SGL A[M_LP_FILTER_ORDER]; INT A_exp; FIXP_DBL gc_threshold; FIXP_DBL de_emph_mem; FIXP_SGL past_gpit; FIXP_DBL past_gcode; USHORT old_T0; UCHAR old_T0_frac; FIXP_DBL deemph_mem_wsyn; FIXP_DBL wsyn_rms; SHORT seed_ace; } CAcelpStaticMem; /** Structure which holds the parameter data needed to decode one ACELP frame. */ typedef struct { UCHAR acelp_core_mode; /**< mean excitation energy index for whole ACELP frame */ UCHAR mean_energy; /**< acelp core mode for whole ACELP frame */ USHORT T0[NB_SUBFR]; UCHAR T0_frac[NB_SUBFR]; UCHAR ltp_filtering_flag[NB_SUBFR]; /**< controlls whether LTP postfilter is active for each ACELP subframe */ SHORT icb_index[NB_SUBFR] [8]; /**< innovative codebook index for each ACELP subframe */ UCHAR gains[NB_SUBFR]; /**< gain index for each ACELP subframe */ } CAcelpChannelData; /** * \brief Read the acelp_coding() bitstream part. * \param[in] hBs bitstream handle to read data from. * \param[out] acelpData pointer to structure to store the parsed data of one * ACELP frame. * \param[in] acelp_core_mode the ACELP core mode index. * \param[in] coreCoderFrameLength length of core coder frame (1024|768) */ INT CLpd_AcelpRead(HANDLE_FDK_BITSTREAM hBs, CAcelpChannelData *acelpData, INT acelp_core_mode, INT i_offset, INT coreCoderFrameLength); /** * \brief Initialization of memory before one LPD frame is decoded * \param[out] synth_buf synthesis buffer to be initialized, exponent = SF_SYNTH * \param[in] old_synth past synthesis of previous LPD frame, exponent = * SF_SYNTH * \param[out] synth_buf_fb fullband synthesis buffer to be initialized, * exponent = SF_SYNTH * \param[in] old_synth_fb past fullband synthesis of previous LPD frame, * exponent = SF_SYNTH * \param[out] pitch vector where decoded pitch lag values are stored * \param[in] old_T_pf past pitch lag values of previous LPD frame * \param[in] samplingRate sampling rate for pitch lag offset calculation * \param[out] i_offset pitch lag offset for the decoding of the pitch lag * \param[in] coreCoderFrameLength length of core coder frame (1024|768) */ void Acelp_PreProcessing(FIXP_DBL *synth_buf, FIXP_DBL *old_synth, INT *pitch, INT *old_T_pf, FIXP_DBL *pit_gain, FIXP_DBL *old_gain_pf, INT samplingRate, INT *i_offset, INT coreCoderFrameLength, INT synSfd, INT nbSubfrSuperfr); /** * \brief Save tail of buffers for the initialization of the next LPD frame * \param[in] synth_buf synthesis of current LPD frame, exponent = SF_SYNTH * \param[out] old_synth memory where tail of fullband synth_buf is stored, * exponent = SF_SYNTH * \param[in] synth_buf_fb fullband synthesis of current LPD frame, exponent = * SF_SYNTH * \param[out] old_synth_fb memory where tail of fullband synth_buf is stored, * exponent = SF_SYNTH * \param[in] pitch decoded pitch lag values of current LPD frame * \param[out] old_T_pf memory where last SYN_SFD pitch lag values are stored */ void Acelp_PostProcessing(FIXP_DBL *synth_buf, FIXP_DBL *old_synth, INT *pitch, INT *old_T_pf, INT coreCoderFrameLength, INT synSfd, INT nbSubfrSuperfr); /** * \brief Decode one ACELP frame (three or four ACELP subframes with 64 samples * each) * \param[in,out] acelp_mem pointer to ACELP memory structure * \param[in] i_offset pitch lag offset * \param[in] lsp_old LPC filter in LSP domain corresponding to previous frame * \param[in] lsp_new LPC filter in LSP domain corresponding to current frame * \param[in] stab_fac stability factor constrained by 0<=stab_fac<=1.0, * exponent = SF_STAB * \param[in] acelpData pointer to struct with data which is needed for decoding * one ACELP frame * \param[out] synth ACELP output signal * \param[out] pT four decoded pitch lag values * \param[in] coreCoderFrameLength length of core coder frame (1024|768) */ void CLpd_AcelpDecode(CAcelpStaticMem *acelp_mem, INT i_offset, const FIXP_LPC lsp_old[M_LP_FILTER_ORDER], const FIXP_LPC lsp_new[M_LP_FILTER_ORDER], FIXP_SGL stab_fac, CAcelpChannelData *acelpData, INT numLostSubframes, int lastLpcLost, int frameCnt, FIXP_DBL synth[], int pT[], FIXP_DBL *pit_gain, INT coreCoderFrameLength); /** * \brief Reset ACELP internal memory. * \param[out] acelp_mem pointer to ACELP memory structure */ void CLpd_AcelpReset(CAcelpStaticMem *acelp_mem); /** * \brief Initialize ACELP internal memory in case of FAC before ACELP decoder * is called * \param[in] synth points to end+1 of past valid synthesis signal, exponent = * SF_SYNTH * \param[in] last_lpd_mode last lpd mode * \param[in] last_last_lpd_mode lpd mode before last_lpd_mode * \param[in] A_new LP synthesis filter coeffs corresponding to last frame, * exponent = SF_A_COEFFS * \param[in] A_old LP synthesis filter coeffs corresponding to the frame before * last frame, exponent = SF_A_COEFFS * \param[in,out] acelp_mem pointer to ACELP memory structure * \param[in] coreCoderFrameLength length of core coder frame (1024|768) */ void CLpd_AcelpPrepareInternalMem(const FIXP_DBL *synth, UCHAR last_lpd_mode, UCHAR last_last_lpd_mode, const FIXP_LPC *A_new, const INT A_new_exp, const FIXP_LPC *A_old, const INT A_old_exp, CAcelpStaticMem *acelp_mem, INT coreCoderFrameLength, INT clearOldExc, UCHAR lpd_mode); /** * \brief Calculate zero input response (zir) of the acelp synthesis filter * \param[in] A LP synthesis filter coefficients, exponent = SF_A_COEFFS * \param[in,out] acelp_mem pointer to ACELP memory structure * \param[in] length length of zir * \param[out] zir pointer to zir output buffer, exponent = SF_SYNTH */ void CLpd_Acelp_Zir(const FIXP_LPC A[], const INT A_exp, CAcelpStaticMem *acelp_mem, const INT length, FIXP_DBL zir[], int doDeemph); /** * \brief Borrow static excitation memory from ACELP decoder * \param[in] acelp_mem pointer to ACELP memory structure * \param[in] length number of requested FIXP_DBL values * \return pointer to requested memory * * The caller has to take care not to overwrite valid memory areas. * During TCX/FAC calculations and before CLpd_AcelpPrepareInternalMem() is * called, the following memory size is available: * - 256 samples in case of ACELP -> TCX20 -> ACELP transition * - PIT_MAX_MAX+L_INTERPOL samples in all other cases */ FIXP_DBL *CLpd_ACELP_GetFreeExcMem(CAcelpStaticMem *acelp_mem, INT length); void CLpd_TcxTDConceal(CAcelpStaticMem *acelp_mem, SHORT *pitch, const FIXP_LPC lsp_old[M_LP_FILTER_ORDER], const FIXP_LPC lsp_new[M_LP_FILTER_ORDER], const FIXP_SGL stab_fac, INT numLostSubframes, FIXP_DBL synth[], INT coreCoderFrameLength, UCHAR last_tcx_noise_factor); inline SHORT E_UTIL_random(SHORT *seed) { *seed = (SHORT)((((LONG)*seed * (LONG)31821) >> 1) + (LONG)13849); return (*seed); } #endif /* USACDEC_ACELP_H */ fdk-aac-2.0.2/libAACdec/src/usacdec_const.h000066400000000000000000000223321404225310700203050ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Manuel Jander Description: USAC related constants *******************************************************************************/ #ifndef USACDEC_CONST_H #define USACDEC_CONST_H /* scale factors */ #define SF_CODE 6 /* exponent of code[], fixed codebook vector */ #define SF_GAIN_C 16 /* exponent of gain code and smoothed gain code */ #define SF_EXC 16 /* exponent of exc[] and exc2[], excitation buffer */ #define SF_GAIN_P 1 /* exponent of gain_pit */ #define SF_PFAC 0 /* exponent of period/voicing factor */ #define SF_SYNTH SF_EXC /* exponent of synthesis buffer */ #define SF_A_COEFFS 3 /* exponent of LP domain synthesis filter coefficient */ #define SF_STAB 1 /* exponent of stability factor */ /* definitions which are independent of coreCoderFrameLength */ #define M_LP_FILTER_ORDER 16 /* LP filter order */ #define LP_FILTER_SCALE 4 /* LP filter scale */ #define PIT_MIN_12k8 34 /* Minimum pitch lag with resolution 1/4 */ #define PIT_MAX_12k8 231 /* Maximum pitch lag for fs=12.8kHz */ #define FSCALE_DENOM 12800 /* Frequency scale denominator */ #define FAC_FSCALE_MIN \ 6000 /* Minimum allowed frequency scale for acelp decoder */ #if !defined(LPD_MAX_CORE_SR) #define LPD_MAX_CORE_SR 24000 /* Default value from ref soft */ #endif #define FAC_FSCALE_MAX \ LPD_MAX_CORE_SR /* Maximum allowed frequency scale for acelp decoder */ /* Maximum pitch lag (= 411 for fs_max = 24000) */ #define PIT_MAX_TMP \ (PIT_MAX_12k8 + \ (6 * \ ((((FAC_FSCALE_MAX * PIT_MIN_12k8) + (FSCALE_DENOM / 2)) / FSCALE_DENOM) - \ PIT_MIN_12k8))) #if (PIT_MAX_TMP < \ 256) /* cannot be smaller because of tcx time domain concealment */ #define PIT_MAX_MAX 256 #else #define PIT_MAX_MAX PIT_MAX_TMP #endif #define NB_DIV 4 /* number of division (20ms) per 80ms frame */ #define L_SUBFR 64 /* subframe size (5ms) */ #define BPF_SFD 1 /* bass postfilter delay (subframe) */ #define BPF_DELAY (BPF_SFD * L_SUBFR) /* bass postfilter delay (samples) */ #define L_FILT 12 /* Delay of up-sampling filter (bass post-filter) */ #define L_EXTRA 96 /* for bass post-filter */ #define L_INTERPOL \ (16 + 1) /* Length of filter for interpolation (acelp decoder) */ /* definitions for coreCoderFrameLength = 1024 */ #define L_FRAME_PLUS_1024 1024 /* length of one 80ms superframe */ #define L_DIV_1024 \ (L_FRAME_PLUS_1024 / NB_DIV) /* length of one acelp or tcx20 frame */ #define NB_SUBFR_1024 \ (L_DIV_1024 / L_SUBFR) /* number of 5ms subframe per division */ #define NB_SUBFR_SUPERFR_1024 \ (L_FRAME_PLUS_1024 / L_SUBFR) /* number of 5ms subframe per 80ms frame */ #define AAC_SFD_1024 (NB_SUBFR_SUPERFR_1024 / 2) /* AAC delay (subframe) */ #define AAC_DELAY_1024 (AAC_SFD_1024 * L_SUBFR) /* AAC delay (samples) */ #define SYN_SFD_1024 (AAC_SFD_1024 - BPF_SFD) /* synthesis delay (subframe) */ #define SYN_DELAY_1024 \ (SYN_SFD_1024 * L_SUBFR) /* synthesis delay (samples) \ */ #define LFAC_1024 (L_DIV_1024 / 2) /* FAC frame length */ #define LFAC_SHORT_1024 \ (L_DIV_1024 / 4) /* for transitions EIGHT_SHORT FD->LPD and vv. */ #define FDNS_NPTS_1024 64 /* FD noise shaping resolution (64=100Hz/point) */ /* definitions for coreCoderFrameLength = 768 */ #define L_FRAME_PLUS_768 768 #define L_DIV_768 \ (L_FRAME_PLUS_768 / NB_DIV) /* length of one acelp or tcx20 frame */ #define NB_SUBFR_768 \ (L_DIV_768 / L_SUBFR) /* number of 5ms subframe per division */ #define NB_SUBFR_SUPERFR_768 \ (L_FRAME_PLUS_768 / L_SUBFR) /* number of 5ms subframe per 80ms frame */ #define AAC_SFD_768 (NB_SUBFR_SUPERFR_768 / 2) /* AAC delay (subframe) */ #define AAC_DELAY_768 (AAC_SFD_768 * L_SUBFR) /* AAC delay (samples) */ #define SYN_SFD_768 (AAC_SFD_768 - BPF_SFD) /* synthesis delay (subframe) */ #define SYN_DELAY_768 (SYN_SFD_768 * L_SUBFR) /* synthesis delay (samples) */ #define LFAC_768 (L_DIV_768 / 2) /* FAC frame length */ #define LFAC_SHORT_768 \ (L_DIV_768 / 4) /* for transitions EIGHT_SHORT FD->LPD and vv. */ /* maximum (used for memory allocation) */ #define L_FRAME_PLUS L_FRAME_PLUS_1024 #define L_DIV L_DIV_1024 #define NB_SUBFR NB_SUBFR_1024 #define NB_SUBFR_SUPERFR NB_SUBFR_SUPERFR_1024 #define AAC_SFD AAC_SFD_1024 #define AAC_DELAY AAC_DELAY_1024 #define SYN_SFD SYN_SFD_1024 #define SYN_DELAY SYN_DELAY_1024 #define LFAC LFAC_1024 #define LFAC_SHORT LFAC_SHORT_1024 #define FDNS_NPTS FDNS_NPTS_1024 #endif /* USACDEC_CONST_H */ fdk-aac-2.0.2/libAACdec/src/usacdec_fac.cpp000066400000000000000000000564601404225310700202540ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Manuel Jander Description: USAC FAC *******************************************************************************/ #include "usacdec_fac.h" #include "usacdec_const.h" #include "usacdec_lpc.h" #include "usacdec_acelp.h" #include "usacdec_rom.h" #include "dct.h" #include "FDK_tools_rom.h" #include "mdct.h" #define SPEC_FAC(ptr, i, gl) ((ptr) + ((i) * (gl))) FIXP_DBL *CLpd_FAC_GetMemory(CAacDecoderChannelInfo *pAacDecoderChannelInfo, UCHAR mod[NB_DIV], int *pState) { FIXP_DBL *ptr; int i; int k = 0; int max_windows = 8; FDK_ASSERT(*pState >= 0 && *pState < max_windows); /* Look for free space to store FAC data. 2 FAC data blocks fit into each TCX * spectral data block. */ for (i = *pState; i < max_windows; i++) { if (mod[i >> 1] == 0) { break; } } *pState = i + 1; if (i == max_windows) { ptr = pAacDecoderChannelInfo->data.usac.fac_data0; } else { FDK_ASSERT(mod[(i >> 1)] == 0); ptr = SPEC_FAC(pAacDecoderChannelInfo->pSpectralCoefficient, i, pAacDecoderChannelInfo->granuleLength << k); } return ptr; } int CLpd_FAC_Read(HANDLE_FDK_BITSTREAM hBs, FIXP_DBL *pFac, SCHAR *pFacScale, int length, int use_gain, int frame) { FIXP_DBL fac_gain; int fac_gain_e = 0; if (use_gain) { CLpd_DecodeGain(&fac_gain, &fac_gain_e, FDKreadBits(hBs, 7)); } if (CLpc_DecodeAVQ(hBs, pFac, 1, 1, length) != 0) { return -1; } { int scale; scale = getScalefactor(pFac, length); scaleValues(pFac, length, scale); pFacScale[frame] = DFRACT_BITS - 1 - scale; } if (use_gain) { int i; pFacScale[frame] += fac_gain_e; for (i = 0; i < length; i++) { pFac[i] = fMult(pFac[i], fac_gain); } } return 0; } /** * \brief Apply synthesis filter with zero input to x. The overall filter gain * is 1.0. * \param a LPC filter coefficients. * \param length length of the input/output data vector x. * \param x input/output vector, where the synthesis filter is applied in place. */ static void Syn_filt_zero(const FIXP_LPC a[], const INT a_exp, INT length, FIXP_DBL x[]) { int i, j; FIXP_DBL L_tmp; for (i = 0; i < length; i++) { L_tmp = (FIXP_DBL)0; for (j = 0; j < fMin(i, M_LP_FILTER_ORDER); j++) { L_tmp -= fMultDiv2(a[j], x[i - (j + 1)]) >> (LP_FILTER_SCALE - 1); } L_tmp = scaleValue(L_tmp, a_exp + LP_FILTER_SCALE); x[i] = fAddSaturate(x[i], L_tmp); } } /* Table is also correct for coreCoderFrameLength = 768. Factor 3/4 is canceled out: gainFac = 0.5 * sqrt(fac_length/lFrame) */ static const FIXP_DBL gainFac[4] = {0x40000000, 0x2d413ccd, 0x20000000, 0x16a09e66}; void CFac_ApplyGains(FIXP_DBL fac_data[LFAC], const INT fac_length, const FIXP_DBL tcx_gain, const FIXP_DBL alfd_gains[], const INT mod) { FIXP_DBL facFactor; int i; FDK_ASSERT((fac_length == 128) || (fac_length == 96)); /* 2) Apply gain factor to FAC data */ facFactor = fMult(gainFac[mod], tcx_gain); for (i = 0; i < fac_length; i++) { fac_data[i] = fMult(fac_data[i], facFactor); } /* 3) Apply spectrum deshaping using alfd_gains */ for (i = 0; i < fac_length / 4; i++) { int k; k = i >> (3 - mod); fac_data[i] = fMult(fac_data[i], alfd_gains[k]) << 1; /* alfd_gains is scaled by one bit. */ } } static void CFac_CalcFacSignal(FIXP_DBL *pOut, FIXP_DBL *pFac, const int fac_scale, const int fac_length, const FIXP_LPC A[M_LP_FILTER_ORDER], const INT A_exp, const int fAddZir, const int isFdFac) { FIXP_LPC wA[M_LP_FILTER_ORDER]; FIXP_DBL tf_gain = (FIXP_DBL)0; int wlength; int scale = fac_scale; /* obtain tranform gain. */ imdct_gain(&tf_gain, &scale, isFdFac ? 0 : fac_length); /* 4) Compute inverse DCT-IV of FAC data. Output scale of DCT IV is 16 bits. */ dct_IV(pFac, fac_length, &scale); /* dct_IV scale = log2(fac_length). "- 7" is a factor of 2/128 */ if (tf_gain != (FIXP_DBL)0) { /* non-radix 2 transform gain */ int i; for (i = 0; i < fac_length; i++) { pFac[i] = fMult(tf_gain, pFac[i]); } } scaleValuesSaturate(pOut, pFac, fac_length, scale); /* Avoid overflow issues and saturate. */ E_LPC_a_weight(wA, A, M_LP_FILTER_ORDER); /* We need the output of the IIR filter to be longer than "fac_length". For this reason we run it with zero input appended to the end of the input sequence, i.e. we generate its ZIR and extend the output signal.*/ FDKmemclear(pOut + fac_length, fac_length * sizeof(FIXP_DBL)); wlength = 2 * fac_length; /* 5) Apply weighted synthesis filter to FAC data, including optional Zir (5. * item 4). */ Syn_filt_zero(wA, A_exp, wlength, pOut); } INT CLpd_FAC_Mdct2Acelp(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *pFac, const int fac_scale, FIXP_LPC *A, INT A_exp, INT nrOutSamples, const INT fac_length, const INT isFdFac, UCHAR prevWindowShape) { FIXP_DBL *pOvl; FIXP_DBL *pOut0; const FIXP_WTP *pWindow; int i, fl, nrSamples = 0; FDK_ASSERT(fac_length <= 1024 / (4 * 2)); fl = fac_length * 2; pWindow = FDKgetWindowSlope(fl, prevWindowShape); /* Adapt window slope length in case of frame loss. */ if (hMdct->prev_fr != fl) { int nl = 0; imdct_adapt_parameters(hMdct, &fl, &nl, fac_length, pWindow, nrOutSamples); FDK_ASSERT(nl == 0); } if (nrSamples < nrOutSamples) { pOut0 = output; nrSamples += hMdct->ov_offset; /* Purge buffered output. */ FDKmemcpy(pOut0, hMdct->overlap.time, hMdct->ov_offset * sizeof(pOut0[0])); hMdct->ov_offset = 0; } pOvl = hMdct->overlap.freq + hMdct->ov_size - 1; if (nrSamples >= nrOutSamples) { pOut0 = hMdct->overlap.time + hMdct->ov_offset; hMdct->ov_offset += hMdct->prev_nr + fl / 2; } else { pOut0 = output + nrSamples; nrSamples += hMdct->prev_nr + fl / 2; } if (hMdct->prevPrevAliasSymmetry == 0) { for (i = 0; i < hMdct->prev_nr; i++) { FIXP_DBL x = -(*pOvl--); *pOut0 = IMDCT_SCALE_DBL(x); pOut0++; } } else { for (i = 0; i < hMdct->prev_nr; i++) { FIXP_DBL x = (*pOvl--); *pOut0 = IMDCT_SCALE_DBL(x); pOut0++; } } hMdct->prev_nr = 0; { if (pFac != NULL) { /* Note: The FAC gain might have been applied directly after bit stream * parse in this case. */ CFac_CalcFacSignal(pOut0, pFac, fac_scale, fac_length, A, A_exp, 0, isFdFac); } else { /* Clear buffer because of the overlap and ADD! */ FDKmemclear(pOut0, fac_length * sizeof(FIXP_DBL)); } } i = 0; if (hMdct->prevPrevAliasSymmetry == 0) { for (; i < fl / 2; i++) { FIXP_DBL x0; /* Overlap Add */ x0 = -fMult(*pOvl--, pWindow[i].v.re); *pOut0 = fAddSaturate(*pOut0, IMDCT_SCALE_DBL(x0)); pOut0++; } } else { for (; i < fl / 2; i++) { FIXP_DBL x0; /* Overlap Add */ x0 = fMult(*pOvl--, pWindow[i].v.re); *pOut0 = fAddSaturate(*pOut0, IMDCT_SCALE_DBL(x0)); pOut0++; } } if (hMdct->pFacZir != 0) { /* this should only happen for ACELP -> TCX20 -> ACELP transition */ FIXP_DBL *pOut = pOut0 - fl / 2; /* fl/2 == fac_length */ for (i = 0; i < fl / 2; i++) { pOut[i] = fAddSaturate(pOut[i], IMDCT_SCALE_DBL(hMdct->pFacZir[i])); } hMdct->pFacZir = NULL; } hMdct->prev_fr = 0; hMdct->prev_nr = 0; hMdct->prev_tl = 0; hMdct->prevPrevAliasSymmetry = hMdct->prevAliasSymmetry; return nrSamples; } INT CLpd_FAC_Acelp2Mdct(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *_pSpec, const SHORT spec_scale[], const int nSpec, FIXP_DBL *pFac, const int fac_scale, const INT fac_length, INT noOutSamples, const INT tl, const FIXP_WTP *wrs, const INT fr, FIXP_LPC A[16], INT A_exp, CAcelpStaticMem *acelp_mem, const FIXP_DBL gain, const int last_frame_lost, const int isFdFac, const UCHAR last_lpd_mode, const int k, int currAliasingSymmetry) { FIXP_DBL *pCurr, *pOvl, *pSpec; const FIXP_WTP *pWindow; const FIXP_WTB *FacWindowZir_conceal; UCHAR doFacZirConceal = 0; int doDeemph = 1; const FIXP_WTB *FacWindowZir, *FacWindowSynth; FIXP_DBL *pOut0 = output, *pOut1; int w, i, fl, nl, nr, f_len, nrSamples = 0, s = 0, scale, total_gain_e; FIXP_DBL *pF, *pFAC_and_FAC_ZIR = NULL; FIXP_DBL total_gain = gain; FDK_ASSERT(fac_length <= 1024 / (4 * 2)); switch (fac_length) { /* coreCoderFrameLength = 1024 */ case 128: pWindow = SineWindow256; FacWindowZir = FacWindowZir128; FacWindowSynth = FacWindowSynth128; break; case 64: pWindow = SineWindow128; FacWindowZir = FacWindowZir64; FacWindowSynth = FacWindowSynth64; break; case 32: pWindow = SineWindow64; FacWindowZir = FacWindowZir32; FacWindowSynth = FacWindowSynth32; break; /* coreCoderFrameLength = 768 */ case 96: pWindow = SineWindow192; FacWindowZir = FacWindowZir96; FacWindowSynth = FacWindowSynth96; break; case 48: pWindow = SineWindow96; FacWindowZir = FacWindowZir48; FacWindowSynth = FacWindowSynth48; break; default: FDK_ASSERT(0); return 0; } FacWindowZir_conceal = FacWindowSynth; /* Derive NR and NL */ fl = fac_length * 2; nl = (tl - fl) >> 1; nr = (tl - fr) >> 1; if (noOutSamples > nrSamples) { /* Purge buffered output. */ FDKmemcpy(pOut0, hMdct->overlap.time, hMdct->ov_offset * sizeof(pOut0[0])); nrSamples = hMdct->ov_offset; hMdct->ov_offset = 0; } if (nrSamples >= noOutSamples) { pOut1 = hMdct->overlap.time + hMdct->ov_offset; if (hMdct->ov_offset < fac_length) { pOut0 = output + nrSamples; } else { pOut0 = pOut1; } hMdct->ov_offset += fac_length + nl; } else { pOut1 = output + nrSamples; pOut0 = output + nrSamples; } { pFAC_and_FAC_ZIR = CLpd_ACELP_GetFreeExcMem(acelp_mem, 2 * fac_length); { const FIXP_DBL *pTmp1, *pTmp2; doFacZirConceal |= ((last_frame_lost != 0) && (k == 0)); doDeemph &= (last_lpd_mode != 4); if (doFacZirConceal) { /* ACELP contribution in concealment case: Use ZIR with a modified ZIR window to preserve some more energy. Dont use FAC, which contains wrong information for concealed frame Dont use last ACELP samples, but double ZIR, instead (afterwards) */ FDKmemclear(pFAC_and_FAC_ZIR, 2 * fac_length * sizeof(FIXP_DBL)); FacWindowSynth = (FIXP_WTB *)pFAC_and_FAC_ZIR; FacWindowZir = FacWindowZir_conceal; } else { CFac_CalcFacSignal(pFAC_and_FAC_ZIR, pFac, fac_scale + s, fac_length, A, A_exp, 1, isFdFac); } /* 6) Get windowed past ACELP samples and ACELP ZIR signal */ /* * Get ACELP ZIR (pFac[]) and ACELP past samples (pOut0[]) and add them * to the FAC synth signal contribution on pOut1[]. */ { { CLpd_Acelp_Zir(A, A_exp, acelp_mem, fac_length, pFac, doDeemph); pTmp1 = pOut0; pTmp2 = pFac; } for (i = 0, w = 0; i < fac_length; i++) { FIXP_DBL x; /* Div2 is compensated by table scaling */ x = fMultDiv2(pTmp2[i], FacWindowZir[w]); x += fMultDiv2(pTmp1[-i - 1], FacWindowSynth[w]); pOut1[i] = fAddSaturate(x, pFAC_and_FAC_ZIR[i]); w++; } } if (doFacZirConceal) { /* ZIR is the only ACELP contribution, so double it */ scaleValues(pOut1, fac_length, 1); } } } if (nrSamples < noOutSamples) { nrSamples += fac_length + nl; } /* Obtain transform gain */ total_gain = gain; total_gain_e = 0; imdct_gain(&total_gain, &total_gain_e, tl); /* IMDCT overlap add */ scale = total_gain_e; pSpec = _pSpec; /* Note:when comming from an LPD frame (TCX/ACELP) the previous alisaing * symmetry must always be 0 */ if (currAliasingSymmetry == 0) { dct_IV(pSpec, tl, &scale); } else { FIXP_DBL _tmp[1024 + ALIGNMENT_DEFAULT / sizeof(FIXP_DBL)]; FIXP_DBL *tmp = (FIXP_DBL *)ALIGN_PTR(_tmp); C_ALLOC_ALIGNED_REGISTER(tmp, sizeof(_tmp)); dst_III(pSpec, tmp, tl, &scale); C_ALLOC_ALIGNED_UNREGISTER(tmp); } /* Optional scaling of time domain - no yet windowed - of current spectrum */ if (total_gain != (FIXP_DBL)0) { for (i = 0; i < tl; i++) { pSpec[i] = fMult(pSpec[i], total_gain); } } int loc_scale = fixmin_I(spec_scale[0] + scale, (INT)DFRACT_BITS - 1); scaleValuesSaturate(pSpec, tl, loc_scale); pOut1 += fl / 2 - 1; pCurr = pSpec + tl - fl / 2; for (i = 0; i < fl / 2; i++) { FIXP_DBL x1; /* FAC signal is already on pOut1, because of that the += operator. */ x1 = fMult(*pCurr++, pWindow[i].v.re); FDK_ASSERT((pOut1 >= hMdct->overlap.time && pOut1 < hMdct->overlap.time + hMdct->ov_size) || (pOut1 >= output && pOut1 < output + 1024)); *pOut1 = fAddSaturate(*pOut1, IMDCT_SCALE_DBL(-x1)); pOut1--; } /* NL output samples TL/2+FL/2..TL. - current[FL/2..0] */ pOut1 += (fl / 2) + 1; pFAC_and_FAC_ZIR += fac_length; /* set pointer to beginning of FAC ZIR */ if (nl == 0) { /* save pointer to write FAC ZIR data later */ hMdct->pFacZir = pFAC_and_FAC_ZIR; } else { FDK_ASSERT(nl >= fac_length); /* FAC ZIR will be added now ... */ hMdct->pFacZir = NULL; } pF = pFAC_and_FAC_ZIR; f_len = fac_length; pCurr = pSpec + tl - fl / 2 - 1; for (i = 0; i < nl; i++) { FIXP_DBL x = -(*pCurr--); /* 5) (item 4) Synthesis filter Zir component, FAC ZIR (another one). */ if (i < f_len) { x = fAddSaturate(x, *pF++); } FDK_ASSERT((pOut1 >= hMdct->overlap.time && pOut1 < hMdct->overlap.time + hMdct->ov_size) || (pOut1 >= output && pOut1 < output + 1024)); *pOut1 = IMDCT_SCALE_DBL(x); pOut1++; } hMdct->prev_nr = nr; hMdct->prev_fr = fr; hMdct->prev_wrs = wrs; hMdct->prev_tl = tl; hMdct->prevPrevAliasSymmetry = hMdct->prevAliasSymmetry; hMdct->prevAliasSymmetry = currAliasingSymmetry; fl = fr; nl = nr; pOvl = pSpec + tl / 2 - 1; pOut0 = pOut1; for (w = 1; w < nSpec; w++) /* for ACELP -> FD short */ { const FIXP_WTP *pWindow_prev; /* Setup window pointers */ pWindow_prev = hMdct->prev_wrs; /* Current spectrum */ pSpec = _pSpec + w * tl; scale = total_gain_e; /* For the second, third, etc. short frames the alisaing symmetry is equal, * either (0,0) or (1,1) */ if (currAliasingSymmetry == 0) { /* DCT IV of current spectrum */ dct_IV(pSpec, tl, &scale); } else { dst_IV(pSpec, tl, &scale); } /* Optional scaling of time domain - no yet windowed - of current spectrum */ /* and de-scale current spectrum signal (time domain, no yet windowed) */ if (total_gain != (FIXP_DBL)0) { for (i = 0; i < tl; i++) { pSpec[i] = fMult(pSpec[i], total_gain); } } loc_scale = fixmin_I(spec_scale[w] + scale, (INT)DFRACT_BITS - 1); scaleValuesSaturate(pSpec, tl, loc_scale); if (noOutSamples <= nrSamples) { /* Divert output first half to overlap buffer if we already got enough * output samples. */ pOut0 = hMdct->overlap.time + hMdct->ov_offset; hMdct->ov_offset += hMdct->prev_nr + fl / 2; } else { /* Account output samples */ nrSamples += hMdct->prev_nr + fl / 2; } /* NR output samples 0 .. NR. -overlap[TL/2..TL/2-NR] */ for (i = 0; i < hMdct->prev_nr; i++) { FIXP_DBL x = -(*pOvl--); *pOut0 = IMDCT_SCALE_DBL(x); pOut0++; } if (noOutSamples <= nrSamples) { /* Divert output second half to overlap buffer if we already got enough * output samples. */ pOut1 = hMdct->overlap.time + hMdct->ov_offset + fl / 2 - 1; hMdct->ov_offset += fl / 2 + nl; } else { pOut1 = pOut0 + (fl - 1); nrSamples += fl / 2 + nl; } /* output samples before window crossing point NR .. TL/2. * -overlap[TL/2-NR..TL/2-NR-FL/2] + current[NR..TL/2] */ /* output samples after window crossing point TL/2 .. TL/2+FL/2. * -overlap[0..FL/2] - current[TL/2..FL/2] */ pCurr = pSpec + tl - fl / 2; if (currAliasingSymmetry == 0) { for (i = 0; i < fl / 2; i++) { FIXP_DBL x0, x1; cplxMultDiv2(&x1, &x0, *pCurr++, -*pOvl--, pWindow_prev[i]); *pOut0 = IMDCT_SCALE_DBL_LSH1(x0); *pOut1 = IMDCT_SCALE_DBL_LSH1(-x1); pOut0++; pOut1--; } } else { if (hMdct->prevPrevAliasSymmetry == 0) { /* Jump DST II -> DST IV for the second window */ for (i = 0; i < fl / 2; i++) { FIXP_DBL x0, x1; cplxMultDiv2(&x1, &x0, *pCurr++, -*pOvl--, pWindow_prev[i]); *pOut0 = IMDCT_SCALE_DBL_LSH1(x0); *pOut1 = IMDCT_SCALE_DBL_LSH1(x1); pOut0++; pOut1--; } } else { /* Jump DST IV -> DST IV from the second window on */ for (i = 0; i < fl / 2; i++) { FIXP_DBL x0, x1; cplxMultDiv2(&x1, &x0, *pCurr++, *pOvl--, pWindow_prev[i]); *pOut0 = IMDCT_SCALE_DBL_LSH1(x0); *pOut1 = IMDCT_SCALE_DBL_LSH1(x1); pOut0++; pOut1--; } } } if (hMdct->pFacZir != 0) { /* add FAC ZIR of previous ACELP -> mdct transition */ FIXP_DBL *pOut = pOut0 - fl / 2; FDK_ASSERT(fl / 2 <= 128); for (i = 0; i < fl / 2; i++) { pOut[i] = fAddSaturate(pOut[i], IMDCT_SCALE_DBL(hMdct->pFacZir[i])); } hMdct->pFacZir = NULL; } pOut0 += (fl / 2); /* NL output samples TL/2+FL/2..TL. - current[FL/2..0] */ pOut1 += (fl / 2) + 1; pCurr = pSpec + tl - fl / 2 - 1; for (i = 0; i < nl; i++) { FIXP_DBL x = -(*pCurr--); *pOut1 = IMDCT_SCALE_DBL(x); pOut1++; } /* Set overlap source pointer for next window pOvl = pSpec + tl/2 - 1; */ pOvl = pSpec + tl / 2 - 1; /* Previous window values. */ hMdct->prev_nr = nr; hMdct->prev_fr = fr; hMdct->prev_tl = tl; hMdct->prev_wrs = pWindow_prev; hMdct->prevPrevAliasSymmetry = hMdct->prevAliasSymmetry; hMdct->prevAliasSymmetry = currAliasingSymmetry; } /* Save overlap */ pOvl = hMdct->overlap.freq + hMdct->ov_size - tl / 2; FDK_ASSERT(pOvl >= hMdct->overlap.time + hMdct->ov_offset); FDK_ASSERT(tl / 2 <= hMdct->ov_size); for (i = 0; i < tl / 2; i++) { pOvl[i] = _pSpec[i + (w - 1) * tl]; } return nrSamples; } fdk-aac-2.0.2/libAACdec/src/usacdec_fac.h000066400000000000000000000210301404225310700177020ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Manuel Jander Description: USAC FAC *******************************************************************************/ #ifndef USACDEC_FAC_H #define USACDEC_FAC_H #include "channelinfo.h" #include "FDK_bitstream.h" /** * \brief Get the address of a memory area of the spectral data memory were the * FAC data can be stored into. * \param spec SPECTRAL_PTR pointing to the current spectral data. * \param mod the current LPD mod array. * \param pState pointer to a private state variable which must be 0 for the * first call and not changed externally. * \param isFullbandLPD is 1 if fullband LPD mode is on, otherwise it is 0. */ FIXP_DBL *CLpd_FAC_GetMemory(CAacDecoderChannelInfo *pAacDecoderChannelInfo, UCHAR mod[NB_SUBFR], int *pState); /** * \brief read a fac bitstream data block. * \param hBs a bit stream handle, where the fac bitstream data is located. * \param pFac pointer to were the FAC data will be stored into. * \param pFacScale pointer to were the FAC data scale value will be stored * into. * \param tcx_gain value to be used as FAC gain. If zero, read fac_gain from * bitstream. * \param tcx_gain_e exponen value of tcx_gain. * \param frame the subframe to be considered from the current superframe. * Always 0 for FD case. * \return 0 on success, -1 on error. */ int CLpd_FAC_Read(HANDLE_FDK_BITSTREAM hBs, FIXP_DBL *pFac, SCHAR *pFacScale, int length, int use_gain, int frame); /** * \brief Apply TCX and ALFD gains to FAC data. * \param fac_data pointer to FAC data. * \param fac_length FAC length (128 or 96). * \param tcx_gain TCX gain * \param alfd_gains pointer to alfd gains. * \param mod mod value (1,2,3) of TCX frame where the FAC signal needs to be * applied. */ void CFac_ApplyGains(FIXP_DBL fac_data[LFAC], const INT fac_length, const FIXP_DBL tcx_gain, const FIXP_DBL alfd_gains[], const INT mod); /** * \brief Do FAC transition from frequency domain to ACELP domain. */ INT CLpd_FAC_Mdct2Acelp(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *pFac_data, const int fac_data_e, FIXP_LPC *A, INT A_exp, INT nrOutSamples, const INT fac_length, const INT isFdFac, UCHAR prevWindowShape); /** * \brief Do FAC transition from ACELP domain to frequency domain. * \param hMdct MDCT context. * \param output pointer for time domain output. * \param pSpec pointer to MDCT spectrum input. * \param spec_scale MDCT spectrum exponents. * \param nSpec amount of contiguos MDCT spectra. * \param pFac pointer to FAC MDCT domain data. * \param fac_scale exponent of FAC data. * \param fac_length length of FAC data. * \param nrSamples room in samples in output buffer. * \param tl MDCT transform length of pSpec. * \param wrs right MDCT window slope. * \param fr right MDCT window slope length. * \param A LP domain filter coefficients. * \param deemph_mem deemphasis filter state. * \param gain gain to be applied to FAC data before overlap add. * \param old_syn_mem Synthesis filter state. * \param isFdFac indicates fac processing from or to FD. * \param pFacData fac data stored for fullband LPD. * \param elFlags element specific parser guidance flags. * \param isFacForFullband indicates that fac is processed for fullband LPD. */ INT CLpd_FAC_Acelp2Mdct(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *pSpec, const SHORT spec_scale[], const int nSpec, FIXP_DBL *pFac_data, const int fac_data_e, const INT fac_length, INT nrSamples, const INT tl, const FIXP_WTP *wrs, const INT fr, FIXP_LPC A[16], INT A_exp, CAcelpStaticMem *acelp_mem, const FIXP_DBL gain, const int last_frame_lost, const int isFdFac, const UCHAR last_lpd, const int k, int currAliasingSymmetry); #endif /* USACDEC_FAC_H */ fdk-aac-2.0.2/libAACdec/src/usacdec_lpc.cpp000066400000000000000000001036571404225310700203020ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. 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IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Matthias Hildenbrand, Manuel Jander Description: USAC LPC/AVQ decode *******************************************************************************/ #include "usacdec_lpc.h" #include "usacdec_rom.h" #include "FDK_trigFcts.h" #define NQ_MAX 36 /* * Helper functions. */ /** * \brief Read unary code. * \param hBs bitstream handle as data source. * \return decoded value. */ static int get_vlclbf(HANDLE_FDK_BITSTREAM hBs) { int result = 0; while (FDKreadBits(hBs, 1) && result <= NQ_MAX) { result++; } return result; } /** * \brief Read bit count limited unary code. * \param hBs bitstream handle as data source * \param n max amount of bits to be read. * \return decoded value. */ static int get_vlclbf_n(HANDLE_FDK_BITSTREAM hBs, int n) { int result = 0; while (FDKreadBits(hBs, 1)) { result++; n--; if (n <= 0) { break; } } return result; } /* * Algebraic Vector Quantizer */ /* ZF_SCALE must be greater than (number of FIXP_ZF)/2 because the loss of precision caused by fPow2Div2 in RE8_PPV() */ //#define ZF_SCALE ((NQ_MAX-3)>>1) #define ZF_SCALE ((DFRACT_BITS / 2)) #define FIXP_ZF FIXP_DBL #define INT2ZF(x, s) (FIXP_ZF)((x) << (ZF_SCALE - (s))) #define ZF2INT(x) (INT)((x) >> ZF_SCALE) /* 1.0 in ZF format format */ #define ONEZF ((FIXP_ZF)INT2ZF(1, 0)) /* static */ void nearest_neighbor_2D8(FIXP_ZF x[8], int y[8]) { FIXP_ZF s, em, e[8]; int i, j, sum; /* round x into 2Z^8 i.e. compute y=(y1,...,y8) such that yi = 2[xi/2] where [.] is the nearest integer operator in the mean time, compute sum = y1+...+y8 */ sum = 0; for (i = 0; i < 8; i++) { FIXP_ZF tmp; /* round to ..., -2, 0, 2, ... ([-1..1[ --> 0) */ if (x[i] < (FIXP_ZF)0) { tmp = ONEZF - x[i]; y[i] = -2 * ((ZF2INT(tmp)) >> 1); } else { tmp = ONEZF + x[i]; y[i] = 2 * ((ZF2INT(tmp)) >> 1); } sum += y[i]; } /* check if y1+...+y8 is a multiple of 4 if not, y is not round xj in the wrong way where j is defined by j = arg max_i | xi -yi| (this is called the Wagner rule) */ if (sum % 4) { /* find j = arg max_i | xi -yi| */ em = (FIXP_SGL)0; j = 0; for (i = 0; i < 8; i++) { /* compute ei = xi-yi */ e[i] = x[i] - INT2ZF(y[i], 0); } for (i = 0; i < 8; i++) { /* compute |ei| = | xi-yi | */ if (e[i] < (FIXP_ZF)0) { s = -e[i]; } else { s = e[i]; } /* check if |ei| is maximal, if so, set j=i */ if (em < s) { em = s; j = i; } } /* round xj in the "wrong way" */ if (e[j] < (FIXP_ZF)0) { y[j] -= 2; } else { y[j] += 2; } } } /*-------------------------------------------------------------- RE8_PPV(x,y) NEAREST NEIGHBOR SEARCH IN INFINITE LATTICE RE8 the algorithm is based on the definition of RE8 as RE8 = (2D8) U (2D8+[1,1,1,1,1,1,1,1]) it applies the coset decoding of Sloane and Conway (i) x: point in R^8 in 32-ZF_SCALE.ZF_SCALE format (o) y: point in RE8 (8-dimensional integer vector) -------------------------------------------------------------- */ /* static */ void RE8_PPV(FIXP_ZF x[], SHORT y[], int r) { int i, y0[8], y1[8]; FIXP_ZF x1[8], tmp; INT64 e; /* find the nearest neighbor y0 of x in 2D8 */ nearest_neighbor_2D8(x, y0); /* find the nearest neighbor y1 of x in 2D8+(1,...,1) (by coset decoding) */ for (i = 0; i < 8; i++) { x1[i] = x[i] - ONEZF; } nearest_neighbor_2D8(x1, y1); for (i = 0; i < 8; i++) { y1[i] += 1; } /* compute e0=||x-y0||^2 and e1=||x-y1||^2 */ e = 0; for (i = 0; i < 8; i++) { tmp = x[i] - INT2ZF(y0[i], 0); e += (INT64)fPow2Div2( tmp << r); /* shift left to ensure that no fract part bits get lost. */ tmp = x[i] - INT2ZF(y1[i], 0); e -= (INT64)fPow2Div2(tmp << r); } /* select best candidate y0 or y1 to minimize distortion */ if (e < 0) { for (i = 0; i < 8; i++) { y[i] = y0[i]; } } else { for (i = 0; i < 8; i++) { y[i] = y1[i]; } } } /* table look-up of unsigned value: find i where index >= table[i] Note: range must be >= 2, index must be >= table[0] */ static int table_lookup(const USHORT *table, unsigned int index, int range) { int i; for (i = 4; i < range; i += 4) { if (index < table[i]) { break; } } if (i > range) { i = range; } if (index < table[i - 2]) { i -= 2; } if (index < table[i - 1]) { i--; } i--; return (i); /* index >= table[i] */ } /*-------------------------------------------------------------------------- re8_decode_rank_of_permutation(rank, xs, x) DECODING OF THE RANK OF THE PERMUTATION OF xs (i) rank: index (rank) of a permutation (i) xs: signed leader in RE8 (8-dimensional integer vector) (o) x: point in RE8 (8-dimensional integer vector) -------------------------------------------------------------------------- */ static void re8_decode_rank_of_permutation(int rank, int *xs, SHORT x[8]) { INT a[8], w[8], B, fac, fac_B, target; int i, j; /* --- pre-processing based on the signed leader xs --- - compute the alphabet a=[a[0] ... a[q-1]] of x (q elements) such that a[0]!=...!=a[q-1] it is assumed that xs is sorted in the form of a signed leader which can be summarized in 2 requirements: a) |xs[0]| >= |xs[1]| >= |xs[2]| >= ... >= |xs[7]| b) if |xs[i]|=|xs[i-1]|, xs[i]>=xs[i+1] where |.| indicates the absolute value operator - compute q (the number of symbols in the alphabet) - compute w[0..q-1] where w[j] counts the number of occurences of the symbol a[j] in xs - compute B = prod_j=0..q-1 (w[j]!) where .! is the factorial */ /* xs[i], xs[i-1] and ptr_w/a*/ j = 0; w[j] = 1; a[j] = xs[0]; B = 1; for (i = 1; i < 8; i++) { if (xs[i] != xs[i - 1]) { j++; w[j] = 1; a[j] = xs[i]; } else { w[j]++; B *= w[j]; } } /* --- actual rank decoding --- the rank of x (where x is a permutation of xs) is based on Schalkwijk's formula it is given by rank=sum_{k=0..7} (A_k * fac_k/B_k) the decoding of this rank is sequential and reconstructs x[0..7] element by element from x[0] to x[7] [the tricky part is the inference of A_k for each k...] */ if (w[0] == 8) { for (i = 0; i < 8; i++) { x[i] = a[0]; /* avoid fac of 40320 */ } } else { target = rank * B; fac_B = 1; /* decode x element by element */ for (i = 0; i < 8; i++) { fac = fac_B * fdk_dec_tab_factorial[i]; /* fac = 1..5040 */ j = -1; do { target -= w[++j] * fac; } while (target >= 0); /* max of 30 tests / SV */ x[i] = a[j]; /* update rank, denominator B (B_k) and counter w[j] */ target += w[j] * fac; /* target = fac_B*B*rank */ fac_B *= w[j]; w[j]--; } } } /*-------------------------------------------------------------------------- re8_decode_base_index(n, I, y) DECODING OF AN INDEX IN Qn (n=0,2,3 or 4) (i) n: codebook number (*n is an integer defined in {0,2,3,4}) (i) I: index of c (pointer to unsigned 16-bit word) (o) y: point in RE8 (8-dimensional integer vector) note: the index I is defined as a 32-bit word, but only 16 bits are required (long can be replaced by unsigned integer) -------------------------------------------------------------------------- */ static void re8_decode_base_index(int *n, UINT index, SHORT y[8]) { int i, im, t, sign_code, ka, ks, rank, leader[8]; if (*n < 2) { for (i = 0; i < 8; i++) { y[i] = 0; } } else { // index = (unsigned int)*I; /* search for the identifier ka of the absolute leader (table-lookup) Q2 is a subset of Q3 - the two cases are considered in the same branch */ switch (*n) { case 2: case 3: i = table_lookup(fdk_dec_I3, index, NB_LDQ3); ka = fdk_dec_A3[i]; break; case 4: i = table_lookup(fdk_dec_I4, index, NB_LDQ4); ka = fdk_dec_A4[i]; break; default: FDK_ASSERT(0); return; } /* reconstruct the absolute leader */ for (i = 0; i < 8; i++) { leader[i] = fdk_dec_Da[ka][i]; } /* search for the identifier ks of the signed leader (table look-up) (this search is focused based on the identifier ka of the absolute leader)*/ t = fdk_dec_Ia[ka]; im = fdk_dec_Ns[ka]; ks = table_lookup(fdk_dec_Is + t, index, im); /* reconstruct the signed leader from its sign code */ sign_code = 2 * fdk_dec_Ds[t + ks]; for (i = 7; i >= 0; i--) { leader[i] *= (1 - (sign_code & 2)); sign_code >>= 1; } /* compute and decode the rank of the permutation */ rank = index - fdk_dec_Is[t + ks]; /* rank = index - cardinality offset */ re8_decode_rank_of_permutation(rank, leader, y); } return; } /* re8_y2k(y,m,k) VORONOI INDEXING (INDEX DECODING) k -> y (i) k: Voronoi index k[0..7] (i) m: Voronoi modulo (m = 2^r = 1<=2) (i) r: Voronoi order (m = 2^r = 1<=2) (o) y: 8-dimensional point y[0..7] in RE8 */ static void re8_k2y(int *k, int r, SHORT *y) { int i, tmp, sum; SHORT v[8]; FIXP_ZF zf[8]; FDK_ASSERT(r <= ZF_SCALE); /* compute y = k M and z=(y-a)/m, where M = [4 ] [2 2 ] [| \ ] [2 2 ] [1 1 _ 1 1] a=(2,0,...,0) m = 1<= 1; i--) { tmp = 2 * k[i]; sum += tmp; y[i] += tmp; zf[i] = INT2ZF(y[i], r); } y[0] += (4 * k[0] + sum); zf[0] = INT2ZF(y[0] - 2, r); /* find nearest neighbor v of z in infinite RE8 */ RE8_PPV(zf, v, r); /* compute y -= m v */ for (i = 0; i < 8; i++) { y[i] -= (SHORT)(v[i] << r); } } /*-------------------------------------------------------------------------- RE8_dec(n, I, k, y) MULTI-RATE INDEXING OF A POINT y in THE LATTICE RE8 (INDEX DECODING) (i) n: codebook number (*n is an integer defined in {0,2,3,4,..,n_max}). n_max = 36 (i) I: index of c (pointer to unsigned 16-bit word) (i) k: index of v (8-dimensional vector of binary indices) = Voronoi index (o) y: point in RE8 (8-dimensional integer vector) note: the index I is defined as a 32-bit word, but only 16 bits are required (long can be replaced by unsigned integer) return 0 on success, -1 on error. -------------------------------------------------------------------------- */ static int RE8_dec(int n, int I, int *k, FIXP_DBL *y) { SHORT v[8]; SHORT _y[8]; UINT r; int i; /* Check bound of codebook qn */ if (n > NQ_MAX) { return -1; } /* decode the sub-indices I and kv[] according to the codebook number n: if n=0,2,3,4, decode I (no Voronoi extension) if n>4, Voronoi extension is used, decode I and kv[] */ if (n <= 4) { re8_decode_base_index(&n, I, _y); for (i = 0; i < 8; i++) { y[i] = (LONG)_y[i]; } } else { /* compute the Voronoi modulo m = 2^r where r is extension order */ r = ((n - 3) >> 1); while (n > 4) { n -= 2; } /* decode base codebook index I into c (c is an element of Q3 or Q4) [here c is stored in y to save memory] */ re8_decode_base_index(&n, I, _y); /* decode Voronoi index k[] into v */ re8_k2y(k, r, v); /* reconstruct y as y = m c + v (with m=2^r, r integer >=1) */ for (i = 0; i < 8; i++) { y[i] = (LONG)((_y[i] << r) + v[i]); } } return 0; } /**************************/ /* start LPC decode stuff */ /**************************/ //#define M 16 #define FREQ_MAX 6400.0f #define FREQ_DIV 400.0f #define LSF_GAP 50.0f /** * \brief calculate inverse weighting factor and add non-weighted residual * LSF vector to first stage LSF approximation * \param lsfq first stage LSF approximation values. * \param xq weighted residual LSF vector * \param nk_mode code book number coding mode. */ static void lsf_weight_2st(FIXP_LPC *lsfq, FIXP_DBL *xq, int nk_mode) { FIXP_LPC d[M_LP_FILTER_ORDER + 1]; FIXP_SGL factor; LONG w; /* inverse weight factor */ int i; /* compute lsf distance */ d[0] = lsfq[0]; d[M_LP_FILTER_ORDER] = FL2FXCONST_LPC(FREQ_MAX / (1 << LSF_SCALE)) - lsfq[M_LP_FILTER_ORDER - 1]; for (i = 1; i < M_LP_FILTER_ORDER; i++) { d[i] = lsfq[i] - lsfq[i - 1]; } switch (nk_mode) { case 0: factor = FL2FXCONST_SGL(2.0f * 60.0f / FREQ_DIV); break; /* abs */ case 1: factor = FL2FXCONST_SGL(2.0f * 65.0f / FREQ_DIV); break; /* mid */ case 2: factor = FL2FXCONST_SGL(2.0f * 64.0f / FREQ_DIV); break; /* rel1 */ default: factor = FL2FXCONST_SGL(2.0f * 63.0f / FREQ_DIV); break; /* rel2 */ } /* add non-weighted residual LSF vector to LSF1st */ for (i = 0; i < M_LP_FILTER_ORDER; i++) { w = (LONG)fMultDiv2(factor, sqrtFixp(fMult(d[i], d[i + 1]))); lsfq[i] = fAddSaturate(lsfq[i], FX_DBL2FX_LPC((FIXP_DBL)((INT64)w * (LONG)xq[i]))); } return; } /** * \brief decode nqn amount of code book numbers. These values determine the * amount of following bits for nqn AVQ RE8 vectors. * \param nk_mode quantization mode. * \param nqn amount code book number to read. * \param qn pointer to output buffer to hold decoded code book numbers qn. */ static void decode_qn(HANDLE_FDK_BITSTREAM hBs, int nk_mode, int nqn, int qn[]) { int n; if (nk_mode == 1) { /* nk mode 1 */ /* Unary code for mid LPC1/LPC3 */ /* Q0=0, Q2=10, Q3=110, ... */ for (n = 0; n < nqn; n++) { qn[n] = get_vlclbf(hBs); if (qn[n] > 0) { qn[n]++; } } } else { /* nk_mode 0, 3 and 2 */ /* 2 bits to specify Q2,Q3,Q4,ext */ for (n = 0; n < nqn; n++) { qn[n] = 2 + FDKreadBits(hBs, 2); } if (nk_mode == 2) { /* Unary code for rel LPC1/LPC3 */ /* Q0 = 0, Q5=10, Q6=110, ... */ for (n = 0; n < nqn; n++) { if (qn[n] > 4) { qn[n] = get_vlclbf(hBs); if (qn[n] > 0) qn[n] += 4; } } } else { /* nk_mode == (0 and 3) */ /* Unary code for abs and rel LPC0/LPC2 */ /* Q5 = 0, Q6=10, Q0=110, Q7=1110, ... */ for (n = 0; n < nqn; n++) { if (qn[n] > 4) { qn[n] = get_vlclbf(hBs); switch (qn[n]) { case 0: qn[n] = 5; break; case 1: qn[n] = 6; break; case 2: qn[n] = 0; break; default: qn[n] += 4; break; } } } } } } /** * \brief reorder LSF coefficients to minimum distance. * \param lsf pointer to buffer containing LSF coefficients and where reordered * LSF coefficients will be stored into, scaled by LSF_SCALE. * \param min_dist min distance scaled by LSF_SCALE * \param n number of LSF/LSP coefficients. */ static void reorder_lsf(FIXP_LPC *lsf, FIXP_LPC min_dist, int n) { FIXP_LPC lsf_min; int i; lsf_min = min_dist; for (i = 0; i < n; i++) { if (lsf[i] < lsf_min) { lsf[i] = lsf_min; } lsf_min = fAddSaturate(lsf[i], min_dist); } /* reverse */ lsf_min = FL2FXCONST_LPC(FREQ_MAX / (1 << LSF_SCALE)) - min_dist; for (i = n - 1; i >= 0; i--) { if (lsf[i] > lsf_min) { lsf[i] = lsf_min; } lsf_min = lsf[i] - min_dist; } } /** * \brief First stage approximation * \param hBs bitstream handle as data source * \param lsfq pointer to output buffer to hold LPC coefficients scaled by * LSF_SCALE. */ static void vlpc_1st_dec( HANDLE_FDK_BITSTREAM hBs, /* input: codebook index */ FIXP_LPC *lsfq /* i/o: i:prediction o:quantized lsf */ ) { const FIXP_LPC *p_dico; int i, index; index = FDKreadBits(hBs, 8); p_dico = &fdk_dec_dico_lsf_abs_8b[index * M_LP_FILTER_ORDER]; for (i = 0; i < M_LP_FILTER_ORDER; i++) { lsfq[i] = p_dico[i]; } } /** * \brief Do first stage approximation weighting and multiply with AVQ * refinement. * \param hBs bitstream handle data ssource. * \param lsfq buffer holding 1st stage approx, 2nd stage approx is added to * this values. * \param nk_mode quantization mode. * \return 0 on success, -1 on error. */ static int vlpc_2st_dec( HANDLE_FDK_BITSTREAM hBs, FIXP_LPC *lsfq, /* i/o: i:1st stage o:1st+2nd stage */ int nk_mode /* input: 0=abs, >0=rel */ ) { int err; FIXP_DBL xq[M_LP_FILTER_ORDER]; /* weighted residual LSF vector */ /* Decode AVQ refinement */ { err = CLpc_DecodeAVQ(hBs, xq, nk_mode, 2, 8); } if (err != 0) { return -1; } /* add non-weighted residual LSF vector to LSF1st */ lsf_weight_2st(lsfq, xq, nk_mode); /* reorder */ reorder_lsf(lsfq, FL2FXCONST_LPC(LSF_GAP / (1 << LSF_SCALE)), M_LP_FILTER_ORDER); return 0; } /* * Externally visible functions */ int CLpc_DecodeAVQ(HANDLE_FDK_BITSTREAM hBs, FIXP_DBL *pOutput, int nk_mode, int no_qn, int length) { int i, l; for (i = 0; i < length; i += 8 * no_qn) { int qn[2], nk, n, I; int kv[8] = {0}; decode_qn(hBs, nk_mode, no_qn, qn); for (l = 0; l < no_qn; l++) { if (qn[l] == 0) { FDKmemclear(&pOutput[i + l * 8], 8 * sizeof(FIXP_DBL)); } /* Voronoi extension order ( nk ) */ nk = 0; n = qn[l]; if (qn[l] > 4) { nk = (qn[l] - 3) >> 1; n = qn[l] - nk * 2; } /* Base codebook index, in reverse bit group order (!) */ I = FDKreadBits(hBs, 4 * n); if (nk > 0) { int j; for (j = 0; j < 8; j++) { kv[j] = FDKreadBits(hBs, nk); } } if (RE8_dec(qn[l], I, kv, &pOutput[i + l * 8]) != 0) { return -1; } } } return 0; } int CLpc_Read(HANDLE_FDK_BITSTREAM hBs, FIXP_LPC lsp[][M_LP_FILTER_ORDER], FIXP_LPC lpc4_lsf[M_LP_FILTER_ORDER], FIXP_LPC lsf_adaptive_mean_cand[M_LP_FILTER_ORDER], FIXP_SGL pStability[], UCHAR *mod, int first_lpd_flag, int last_lpc_lost, int last_frame_ok) { int i, k, err; int mode_lpc_bin = 0; /* mode_lpc bitstream representation */ int lpc_present[5] = {0, 0, 0, 0, 0}; int lpc0_available = 1; int s = 0; int l = 3; const int nbDiv = NB_DIV; lpc_present[4 >> s] = 1; /* LPC4 */ /* Decode LPC filters in the following order: LPC 4,0,2,1,3 */ /*** Decode LPC4 ***/ vlpc_1st_dec(hBs, lsp[4 >> s]); err = vlpc_2st_dec(hBs, lsp[4 >> s], 0); /* nk_mode = 0 */ if (err != 0) { return err; } /*** Decode LPC0 and LPC2 ***/ k = 0; if (!first_lpd_flag) { lpc_present[0] = 1; lpc0_available = !last_lpc_lost; /* old LPC4 is new LPC0 */ for (i = 0; i < M_LP_FILTER_ORDER; i++) { lsp[0][i] = lpc4_lsf[i]; } /* skip LPC0 and continue with LPC2 */ k = 2; } for (; k < l; k += 2) { int nk_mode = 0; if ((k == 2) && (mod[0] == 3)) { break; /* skip LPC2 */ } lpc_present[k >> s] = 1; mode_lpc_bin = FDKreadBit(hBs); if (mode_lpc_bin == 0) { /* LPC0/LPC2: Abs */ vlpc_1st_dec(hBs, lsp[k >> s]); } else { /* LPC0/LPC2: RelR */ for (i = 0; i < M_LP_FILTER_ORDER; i++) { lsp[k >> s][i] = lsp[4 >> s][i]; } nk_mode = 3; } err = vlpc_2st_dec(hBs, lsp[k >> s], nk_mode); if (err != 0) { return err; } } /*** Decode LPC1 ***/ if (mod[0] < 2) { /* else: skip LPC1 */ lpc_present[1] = 1; mode_lpc_bin = get_vlclbf_n(hBs, 2); switch (mode_lpc_bin) { case 1: /* LPC1: abs */ vlpc_1st_dec(hBs, lsp[1]); err = vlpc_2st_dec(hBs, lsp[1], 0); if (err != 0) { return err; } break; case 2: /* LPC1: mid0 (no second stage AVQ quantizer in this case) */ if (lpc0_available) { /* LPC0/lsf[0] might be zero some times */ for (i = 0; i < M_LP_FILTER_ORDER; i++) { lsp[1][i] = (lsp[0][i] >> 1) + (lsp[2][i] >> 1); } } else { for (i = 0; i < M_LP_FILTER_ORDER; i++) { lsp[1][i] = lsp[2][i]; } } break; case 0: /* LPC1: RelR */ for (i = 0; i < M_LP_FILTER_ORDER; i++) { lsp[1][i] = lsp[2][i]; } err = vlpc_2st_dec(hBs, lsp[1], 2 << s); if (err != 0) { return err; } break; } } /*** Decode LPC3 ***/ if ((mod[2] < 2)) { /* else: skip LPC3 */ int nk_mode = 0; lpc_present[3] = 1; mode_lpc_bin = get_vlclbf_n(hBs, 3); switch (mode_lpc_bin) { case 1: /* LPC3: abs */ vlpc_1st_dec(hBs, lsp[3]); break; case 0: /* LPC3: mid */ for (i = 0; i < M_LP_FILTER_ORDER; i++) { lsp[3][i] = (lsp[2][i] >> 1) + (lsp[4][i] >> 1); } nk_mode = 1; break; case 2: /* LPC3: relL */ for (i = 0; i < M_LP_FILTER_ORDER; i++) { lsp[3][i] = lsp[2][i]; } nk_mode = 2; break; case 3: /* LPC3: relR */ for (i = 0; i < M_LP_FILTER_ORDER; i++) { lsp[3][i] = lsp[4][i]; } nk_mode = 2; break; } err = vlpc_2st_dec(hBs, lsp[3], nk_mode); if (err != 0) { return err; } } if (!lpc0_available && !last_frame_ok) { /* LPC(0) was lost. Use next available LPC(k) instead */ for (k = 1; k < (nbDiv + 1); k++) { if (lpc_present[k]) { for (i = 0; i < M_LP_FILTER_ORDER; i++) { #define LSF_INIT_TILT (0.25f) if (mod[0] > 0) { lsp[0][i] = FX_DBL2FX_LPC( fMult(lsp[k][i], FL2FXCONST_SGL(1.0f - LSF_INIT_TILT)) + fMult(fdk_dec_lsf_init[i], FL2FXCONST_SGL(LSF_INIT_TILT))); } else { lsp[0][i] = lsp[k][i]; } } break; } } } for (i = 0; i < M_LP_FILTER_ORDER; i++) { lpc4_lsf[i] = lsp[4 >> s][i]; } { FIXP_DBL divFac; int last, numLpc = 0; i = nbDiv; do { numLpc += lpc_present[i--]; } while (i >= 0 && numLpc < 3); last = i; switch (numLpc) { case 3: divFac = FL2FXCONST_DBL(1.0f / 3.0f); break; case 2: divFac = FL2FXCONST_DBL(1.0f / 2.0f); break; default: divFac = FL2FXCONST_DBL(1.0f); break; } /* get the adaptive mean for the next (bad) frame */ for (k = 0; k < M_LP_FILTER_ORDER; k++) { FIXP_DBL tmp = (FIXP_DBL)0; for (i = nbDiv; i > last; i--) { if (lpc_present[i]) { tmp = fMultAdd(tmp >> 1, lsp[i][k], divFac); } } lsf_adaptive_mean_cand[k] = FX_DBL2FX_LPC(tmp); } } /* calculate stability factor Theta. Needed for ACELP decoder and concealment */ { FIXP_LPC *lsf_prev, *lsf_curr; k = 0; FDK_ASSERT(lpc_present[0] == 1 && lpc_present[4 >> s] == 1); lsf_prev = lsp[0]; for (i = 1; i < (nbDiv + 1); i++) { if (lpc_present[i]) { FIXP_DBL tmp = (FIXP_DBL)0; int j; lsf_curr = lsp[i]; /* sum = tmp * 2^(LSF_SCALE*2 + 4) */ for (j = 0; j < M_LP_FILTER_ORDER; j++) { tmp += fPow2Div2((FIXP_SGL)(lsf_curr[j] - lsf_prev[j])) >> 3; } /* tmp = (float)(FL2FXCONST_DBL(1.25f) - fMult(tmp, * FL2FXCONST_DBL(1/400000.0f))); */ tmp = FL2FXCONST_DBL(1.25f / (1 << LSF_SCALE)) - fMult(tmp, FL2FXCONST_DBL((1 << (LSF_SCALE + 4)) / 400000.0f)); if (tmp >= FL2FXCONST_DBL(1.0f / (1 << LSF_SCALE))) { pStability[k] = FL2FXCONST_SGL(1.0f / 2.0f); } else if (tmp < FL2FXCONST_DBL(0.0f)) { pStability[k] = FL2FXCONST_SGL(0.0f); } else { pStability[k] = FX_DBL2FX_SGL(tmp << (LSF_SCALE - 1)); } lsf_prev = lsf_curr; k = i; } else { /* Mark stability value as undefined. */ pStability[i] = (FIXP_SGL)-1; } } } /* convert into LSP domain */ for (i = 0; i < (nbDiv + 1); i++) { if (lpc_present[i]) { for (k = 0; k < M_LP_FILTER_ORDER; k++) { lsp[i][k] = FX_DBL2FX_LPC( fixp_cos(fMult(lsp[i][k], FL2FXCONST_SGL((1 << LSPARG_SCALE) * M_PI / 6400.0)), LSF_SCALE - LSPARG_SCALE)); } } } return 0; } void CLpc_Conceal(FIXP_LPC lsp[][M_LP_FILTER_ORDER], FIXP_LPC lpc4_lsf[M_LP_FILTER_ORDER], FIXP_LPC lsf_adaptive_mean[M_LP_FILTER_ORDER], const int first_lpd_flag) { int i, j; #define BETA (FL2FXCONST_SGL(0.25f)) #define ONE_BETA (FL2FXCONST_SGL(0.75f)) #define BFI_FAC (FL2FXCONST_SGL(0.90f)) #define ONE_BFI_FAC (FL2FXCONST_SGL(0.10f)) /* Frame loss concealment (could be improved) */ if (first_lpd_flag) { /* Reset past LSF values */ for (i = 0; i < M_LP_FILTER_ORDER; i++) { lsp[0][i] = lpc4_lsf[i] = fdk_dec_lsf_init[i]; } } else { /* old LPC4 is new LPC0 */ for (i = 0; i < M_LP_FILTER_ORDER; i++) { lsp[0][i] = lpc4_lsf[i]; } } /* LPC1 */ for (i = 0; i < M_LP_FILTER_ORDER; i++) { FIXP_LPC lsf_mean = FX_DBL2FX_LPC(fMult(BETA, fdk_dec_lsf_init[i]) + fMult(ONE_BETA, lsf_adaptive_mean[i])); lsp[1][i] = FX_DBL2FX_LPC(fMult(BFI_FAC, lpc4_lsf[i]) + fMult(ONE_BFI_FAC, lsf_mean)); } /* LPC2 - LPC4 */ for (j = 2; j <= 4; j++) { for (i = 0; i < M_LP_FILTER_ORDER; i++) { /* lsf_mean[i] = FX_DBL2FX_LPC(fMult((FIXP_LPC)(BETA + j * FL2FXCONST_LPC(0.1f)), fdk_dec_lsf_init[i]) + fMult((FIXP_LPC)(ONE_BETA - j * FL2FXCONST_LPC(0.1f)), lsf_adaptive_mean[i])); */ FIXP_LPC lsf_mean = FX_DBL2FX_LPC( fMult((FIXP_SGL)(BETA + (FIXP_SGL)(j * (INT)FL2FXCONST_SGL(0.1f))), (FIXP_SGL)fdk_dec_lsf_init[i]) + fMult( (FIXP_SGL)(ONE_BETA - (FIXP_SGL)(j * (INT)FL2FXCONST_SGL(0.1f))), lsf_adaptive_mean[i])); lsp[j][i] = FX_DBL2FX_LPC(fMult(BFI_FAC, lsp[j - 1][i]) + fMult(ONE_BFI_FAC, lsf_mean)); } } /* Update past values for the future */ for (i = 0; i < M_LP_FILTER_ORDER; i++) { lpc4_lsf[i] = lsp[4][i]; } /* convert into LSP domain */ for (j = 0; j < 5; j++) { for (i = 0; i < M_LP_FILTER_ORDER; i++) { lsp[j][i] = FX_DBL2FX_LPC(fixp_cos( fMult(lsp[j][i], FL2FXCONST_SGL((1 << LSPARG_SCALE) * M_PI / 6400.0)), LSF_SCALE - LSPARG_SCALE)); } } } void E_LPC_a_weight(FIXP_LPC *wA, const FIXP_LPC *A, int m) { FIXP_DBL f; int i; f = FL2FXCONST_DBL(0.92f); for (i = 0; i < m; i++) { wA[i] = FX_DBL2FX_LPC(fMult(A[i], f)); f = fMult(f, FL2FXCONST_DBL(0.92f)); } } void CLpd_DecodeGain(FIXP_DBL *gain, INT *gain_e, int gain_code) { /* gain * 2^(gain_e) = 10^(gain_code/28) */ *gain = fLdPow( FL2FXCONST_DBL(3.3219280948873623478703194294894 / 4.0), /* log2(10)*/ 2, fMultDiv2((FIXP_DBL)gain_code << (DFRACT_BITS - 1 - 7), FL2FXCONST_DBL(2.0f / 28.0f)), 7, gain_e); } /** * \brief * Find the polynomial F1(z) or F2(z) from the LSPs. * This is performed by expanding the product polynomials: * * F1(z) = product ( 1 - 2 LSP_i z^-1 + z^-2 ) * i=0,2,4,6,8 * F2(z) = product ( 1 - 2 LSP_i z^-1 + z^-2 ) * i=1,3,5,7,9 * * where LSP_i are the LSPs in the cosine domain. * R.A.Salami October 1990 * \param lsp input, line spectral freq. (cosine domain) * \param f output, the coefficients of F1 or F2, scaled by 8 bits * \param n no of coefficients (m/2) * \param flag 1 : F1(z) ; 2 : F2(z) */ #define SF_F 8 static void get_lsppol(FIXP_LPC lsp[], FIXP_DBL f[], int n, int flag) { FIXP_DBL b; FIXP_LPC *plsp; int i, j; plsp = lsp + flag - 1; f[0] = FL2FXCONST_DBL(1.0f / (1 << SF_F)); b = -FX_LPC2FX_DBL(*plsp); f[1] = b >> (SF_F - 1); for (i = 2; i <= n; i++) { plsp += 2; b = -FX_LPC2FX_DBL(*plsp); f[i] = SATURATE_LEFT_SHIFT((fMultDiv2(b, f[i - 1]) + (f[i - 2] >> 1)), 2, DFRACT_BITS); for (j = i - 1; j > 1; j--) { f[j] = SATURATE_LEFT_SHIFT( ((f[j] >> 2) + fMultDiv2(b, f[j - 1]) + (f[j - 2] >> 2)), 2, DFRACT_BITS); } f[1] = f[1] + (b >> (SF_F - 1)); } return; } #define NC M_LP_FILTER_ORDER / 2 /** * \brief lsp input LSP vector * \brief a output LP filter coefficient vector scaled by SF_A_COEFFS. */ void E_LPC_f_lsp_a_conversion(FIXP_LPC *lsp, FIXP_LPC *a, INT *a_exp) { FIXP_DBL f1[NC + 1], f2[NC + 1]; int i, k; /*-----------------------------------------------------* * Find the polynomials F1(z) and F2(z) * *-----------------------------------------------------*/ get_lsppol(lsp, f1, NC, 1); get_lsppol(lsp, f2, NC, 2); /*-----------------------------------------------------* * Multiply F1(z) by (1+z^-1) and F2(z) by (1-z^-1) * *-----------------------------------------------------*/ scaleValues(f1, NC + 1, -2); scaleValues(f2, NC + 1, -2); for (i = NC; i > 0; i--) { f1[i] += f1[i - 1]; f2[i] -= f2[i - 1]; } FIXP_DBL aDBL[M_LP_FILTER_ORDER]; for (i = 1, k = M_LP_FILTER_ORDER - 1; i <= NC; i++, k--) { aDBL[i - 1] = f1[i] + f2[i]; aDBL[k] = f1[i] - f2[i]; } int headroom_a = getScalefactor(aDBL, M_LP_FILTER_ORDER); for (i = 0; i < M_LP_FILTER_ORDER; i++) { a[i] = FX_DBL2FX_LPC(aDBL[i] << headroom_a); } *a_exp = SF_F + (2 - 1) - headroom_a; } fdk-aac-2.0.2/libAACdec/src/usacdec_lpc.h000066400000000000000000000200111404225310700177250ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Matthias Hildenbrand, Manuel Jander Description: USAC LPC/AVQ decode *******************************************************************************/ #ifndef USACDEC_LPC_H #define USACDEC_LPC_H #include "channelinfo.h" #include "common_fix.h" #include "FDK_bitstream.h" #include "usacdec_rom.h" #define LSPARG_SCALE 10 /** * \brief AVQ (refinement) decode * \param hBs bitstream handle * \param lsfq buffer for AVQ decode output. * \param nk_mode quantization mode. * \param nqn amount of split/interleaved RE8 vectors. * \param total amount of individual data values to decode. * \return 0 on success, -1 on error. */ int CLpc_DecodeAVQ(HANDLE_FDK_BITSTREAM hBs, FIXP_DBL *lsfq, int nk_mode, int nqn, int length); /** * \brief Read and decode LPC coeficient sets. First stage approximation + AVQ * decode. * \param[in] hBs bitstream handle to read data from. * \param[out] lsp buffer into which the decoded LSP coefficients will be stored * into. * \param[in,out] lpc4_lsf buffer into which the decoded LCP4 LSF coefficients * will be stored into (persistent). * \param[out] lsf_adaptive_mean_cand lsf adaptive mean vector needed for * concealment. * \param[out] pStability array with stability values for the ACELP decoder (and * concealment). * \param[in] mod array which defines modes (ACELP, TCX20|40|80) are used in * the current superframe. * \param[in] first_lpd_flag indicates the presence of LPC0 * \param[in] last_lpc_lost indicate that LPC4 of previous frame was lost. * \param[in] last_frame_ok indicate that the last frame was ok. * \return 0 on success, -1 on error. */ int CLpc_Read(HANDLE_FDK_BITSTREAM hBs, FIXP_LPC lsp[][M_LP_FILTER_ORDER], FIXP_LPC lpc4_lsf[M_LP_FILTER_ORDER], FIXP_LPC lsf_adaptive_mean_cand[M_LP_FILTER_ORDER], FIXP_SGL pStability[], UCHAR *mod, int first_lpd_flag, int last_lpc_lost, int last_frame_ok); /** * \brief Generate LPC coefficient sets in case frame loss. * \param lsp buffer into which the decoded LSP coefficients will be stored * into. * \param lpc4_lsf buffer into which the decoded LCP4 LSF coefficients will be * stored into (persistent). * \param isf_adaptive_mean * \param first_lpd_flag indicates the previous LSF4 coefficients lpc4_lsf[] are * not valid. */ void CLpc_Conceal(FIXP_LPC lsp[][M_LP_FILTER_ORDER], FIXP_LPC lpc4_lsf[M_LP_FILTER_ORDER], FIXP_LPC isf_adaptive_mean[M_LP_FILTER_ORDER], const int first_lpd_flag); /** * \brief apply absolute weighting * \param A weighted LPC coefficient vector output. The first coeffcient is * implicitly 1.0 * \param A LPC coefficient vector. The first coeffcient is implicitly 1.0 * \param m length of vector A */ /* static */ void E_LPC_a_weight(FIXP_LPC *wA, const FIXP_LPC *A, const int m); /** * \brief decode TCX/FAC gain. In case of TCX the lg/sqrt(rms) part * must still be applied to obtain the gain value. * \param gain (o) pointer were the gain mantissa is stored into. * \param gain_e (o) pointer were the gain exponent is stored into. * \param gain_code (i) the 7 bit binary word from the bitstream * representing the gain. */ void CLpd_DecodeGain(FIXP_DBL *gain, INT *gain_e, int gain_code); /** * \brief convert LSP coefficients into LP domain. */ void E_LPC_f_lsp_a_conversion(FIXP_LPC *lsp, FIXP_LPC *a, INT *a_exp); #endif /* USACDEC_LPC_H */ fdk-aac-2.0.2/libAACdec/src/usacdec_lpd.cpp000066400000000000000000002133451404225310700202770ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Manuel Jander Description: USAC Linear Prediction Domain coding *******************************************************************************/ #include "usacdec_lpd.h" #include "usacdec_rom.h" #include "usacdec_fac.h" #include "usacdec_lpc.h" #include "FDK_tools_rom.h" #include "fft.h" #include "mdct.h" #include "usacdec_acelp.h" #include "overlapadd.h" #include "conceal.h" #include "block.h" #define SF_PITCH_TRACK 6 #define SF_GAIN 3 #define MIN_VAL FL2FXCONST_DBL(0.0f) #define MAX_VAL (FIXP_DBL) MAXVAL_DBL #include "ac_arith_coder.h" void filtLP(const FIXP_DBL *syn, PCM_DEC *syn_out, FIXP_DBL *noise, const FIXP_SGL *filt, const INT aacOutDataHeadroom, INT stop, int len) { INT i, j; FIXP_DBL tmp; FDK_ASSERT((aacOutDataHeadroom - 1) >= -(MDCT_OUTPUT_SCALE)); for (i = 0; i < stop; i++) { tmp = fMultDiv2(noise[i], filt[0]); // Filt in Q-1.16 for (j = 1; j <= len; j++) { tmp += fMult((noise[i - j] >> 1) + (noise[i + j] >> 1), filt[j]); } syn_out[i] = (PCM_DEC)( IMDCT_SCALE((syn[i] >> 1) - (tmp >> 1), aacOutDataHeadroom - 1)); } } void bass_pf_1sf_delay( FIXP_DBL *syn, /* (i) : 12.8kHz synthesis to postfilter */ const INT *T_sf, /* (i) : Pitch period for all subframes (T_sf[16]) */ FIXP_DBL *pit_gain, const int frame_length, /* (i) : frame length (should be 768|1024) */ const INT l_frame, const INT l_next, /* (i) : look ahead for symmetric filtering */ PCM_DEC *synth_out, /* (o) : filtered synthesis (with delay of 1 subfr) */ const INT aacOutDataHeadroom, /* (i) : headroom of the output time signal to prevent clipping */ FIXP_DBL mem_bpf[]) /* i/o : memory state [L_FILT+L_SUBFR] */ { INT i, sf, i_subfr, T, T2, lg; FIXP_DBL tmp, ener, corr, gain; FIXP_DBL *noise, *noise_in; FIXP_DBL noise_buf[L_FILT + (2 * L_SUBFR)]; // L_FILT = 12, L_SUBFR = 64 => 140 const FIXP_DBL *x, *y; { noise = noise_buf + L_FILT; // L_FILT = 12 delay of upsampling filter noise_in = noise_buf + L_FILT + L_SUBFR; /* Input scaling of the BPF memory */ scaleValues(mem_bpf, (L_FILT + L_SUBFR), 1); } int gain_exp = 17; sf = 0; for (i_subfr = 0; i_subfr < l_frame; i_subfr += L_SUBFR, sf++) { T = T_sf[sf]; gain = pit_gain[sf]; /* Gain is in Q17.14 */ /* If gain > 1 set to 1 */ if (gain > (FIXP_DBL)(1 << 14)) gain = (FIXP_DBL)(1 << 14); /* If gain < 0 set to 0 */ if (gain < (FIXP_DBL)0) gain = (FIXP_DBL)0; if (gain > (FIXP_DBL)0) { /* pitch tracker: test pitch/2 to avoid continuous pitch doubling */ /* Note: pitch is limited to PIT_MIN (34 = 376Hz) at the encoder */ T2 = T >> 1; x = &syn[i_subfr - L_EXTRA]; y = &syn[i_subfr - T2 - L_EXTRA]; ener = (FIXP_DBL)0; corr = (FIXP_DBL)0; tmp = (FIXP_DBL)0; int headroom_x = getScalefactor(x, L_SUBFR + L_EXTRA); int headroom_y = getScalefactor(y, L_SUBFR + L_EXTRA); int width_shift = 7; for (i = 0; i < (L_SUBFR + L_EXTRA); i++) { ener += fPow2Div2((x[i] << headroom_x)) >> width_shift; corr += fMultDiv2((x[i] << headroom_x), (y[i] << headroom_y)) >> width_shift; tmp += fPow2Div2((y[i] << headroom_y)) >> width_shift; } int exp_ener = ((17 - headroom_x) << 1) + width_shift + 1; int exp_corr = (17 - headroom_x) + (17 - headroom_y) + width_shift + 1; int exp_tmp = ((17 - headroom_y) << 1) + width_shift + 1; /* Add 0.01 to "ener". Adjust exponents */ FIXP_DBL point_zero_one = (FIXP_DBL)0x51eb851f; /* In Q-6.37 */ int diff; ener = fAddNorm(ener, exp_ener, point_zero_one, -6, &exp_ener); corr = fAddNorm(corr, exp_corr, point_zero_one, -6, &exp_corr); tmp = fAddNorm(tmp, exp_tmp, point_zero_one, -6, &exp_tmp); /* use T2 if normalized correlation > 0.95 */ INT s1, s2; s1 = CntLeadingZeros(ener) - 1; s2 = CntLeadingZeros(tmp) - 1; FIXP_DBL ener_by_tmp = fMultDiv2(ener << s1, tmp << s2); int ener_by_tmp_exp = (exp_ener - s1) + (exp_tmp - s2) + 1; if (ener_by_tmp_exp & 1) { ener_by_tmp <<= 1; ener_by_tmp_exp -= 1; } int temp_exp = 0; FIXP_DBL temp1 = invSqrtNorm2(ener_by_tmp, &temp_exp); int temp1_exp = temp_exp - (ener_by_tmp_exp >> 1); FIXP_DBL tmp_result = fMult(corr, temp1); int tmp_result_exp = exp_corr + temp1_exp; diff = tmp_result_exp - 0; FIXP_DBL point95 = FL2FXCONST_DBL(0.95f); if (diff >= 0) { diff = fMin(diff, 31); point95 = FL2FXCONST_DBL(0.95f) >> diff; } else { diff = fMax(diff, -31); tmp_result >>= (-diff); } if (tmp_result > point95) T = T2; /* prevent that noise calculation below reaches into not defined signal parts at the end of the synth_buf or in other words restrict the below used index (i+i_subfr+T) < l_frame + l_next */ lg = l_frame + l_next - T - i_subfr; if (lg > L_SUBFR) lg = L_SUBFR; else if (lg < 0) lg = 0; /* limit gain to avoid problem on burst */ if (lg > 0) { FIXP_DBL tmp1; /* max(lg) = 64 => scale with 6 bits minus 1 (fPow2Div2) */ s1 = getScalefactor(&syn[i_subfr], lg); s2 = getScalefactor(&syn[i_subfr + T], lg); INT s = fixMin(s1, s2); tmp = (FIXP_DBL)0; ener = (FIXP_DBL)0; for (i = 0; i < lg; i++) { tmp += fPow2Div2(syn[i + i_subfr] << s1) >> (SF_PITCH_TRACK); ener += fPow2Div2(syn[i + i_subfr + T] << s2) >> (SF_PITCH_TRACK); } tmp = tmp >> fMin(DFRACT_BITS - 1, (2 * (s1 - s))); ener = ener >> fMin(DFRACT_BITS - 1, (2 * (s2 - s))); /* error robustness: for the specific case syn[...] == -1.0f for all 64 samples ener or tmp might overflow and become negative. For all sane cases we have enough headroom. */ if (ener <= (FIXP_DBL)0) { ener = (FIXP_DBL)1; } if (tmp <= (FIXP_DBL)0) { tmp = (FIXP_DBL)1; } FDK_ASSERT(ener > (FIXP_DBL)0); /* tmp = sqrt(tmp/ener) */ int result_e = 0; tmp1 = fDivNorm(tmp, ener, &result_e); if (result_e & 1) { tmp1 >>= 1; result_e += 1; } tmp = sqrtFixp(tmp1); result_e >>= 1; gain_exp = 17; diff = result_e - gain_exp; FIXP_DBL gain1 = gain; if (diff >= 0) { diff = fMin(diff, 31); gain1 >>= diff; } else { result_e += (-diff); diff = fMax(diff, -31); tmp >>= (-diff); } if (tmp < gain1) { gain = tmp; gain_exp = result_e; } } /* calculate noise based on voiced pitch */ /* fMultDiv2() replaces weighting of gain with 0.5 */ diff = gain_exp - 17; if (diff >= 0) { gain <<= diff; } else { gain >>= (-diff); } s1 = CntLeadingZeros(gain) - 1; s1 -= 16; /* Leading bits for SGL */ FIXP_SGL gainSGL = FX_DBL2FX_SGL(gain << 16); gainSGL = gainSGL << s1; { for (i = 0; i < lg; i++) { /* scaled with SF_SYNTH + gain_sf + 1; composition of scalefactor 2: * one additional shift of syn values + fMult => fMultDiv2 */ noise_in[i] = scaleValue(fMultDiv2(gainSGL, (syn[i + i_subfr] >> 1) - (syn[i + i_subfr - T] >> 2) - (syn[i + i_subfr + T] >> 2)), 2 - s1); } for (i = lg; i < L_SUBFR; i++) { /* scaled with SF_SYNTH + gain_sf + 1; composition of scalefactor 2: * one additional shift of syn values + fMult => fMultDiv2 */ noise_in[i] = scaleValue(fMultDiv2(gainSGL, (syn[i + i_subfr] >> 1) - (syn[i + i_subfr - T] >> 1)), 2 - s1); } } } else { FDKmemset(noise_in, (FIXP_DBL)0, L_SUBFR * sizeof(FIXP_DBL)); } { FDKmemcpy(noise_buf, mem_bpf, (L_FILT + L_SUBFR) * sizeof(FIXP_DBL)); FDKmemcpy(mem_bpf, noise_buf + L_SUBFR, (L_FILT + L_SUBFR) * sizeof(FIXP_DBL)); } /* substract from voiced speech low-pass filtered noise */ /* filter coefficients are scaled with factor SF_FILT_LP (1) */ { filtLP(&syn[i_subfr - L_SUBFR], &synth_out[i_subfr], noise, fdk_dec_filt_lp, aacOutDataHeadroom, L_SUBFR, L_FILT); } } { } // To be determined (info from Ben) { /* Output scaling of the BPF memory */ scaleValues(mem_bpf, (L_FILT + L_SUBFR), -1); /* Copy the rest of the signal (after the fac) */ scaleValuesSaturate( (PCM_DEC *)&synth_out[l_frame], (FIXP_DBL *)&syn[l_frame - L_SUBFR], (frame_length - l_frame), MDCT_OUT_HEADROOM - aacOutDataHeadroom); } return; } /* * Frequency Domain Noise Shaping */ /** * \brief Adaptive Low Frequencies Deemphasis of spectral coefficients. * * Ensure quantization of low frequencies in case where the * signal dynamic is higher than the LPC noise shaping. * This is the inverse operation of adap_low_freq_emph(). * Output gain of all blocks. * * \param x pointer to the spectral coefficients, requires 1 bit headroom. * \param lg length of x. * \param bUseNewAlfe if set, apply ALFD for fullband lpd. * \param gainLpc1 pointer to gain based on old input LPC coefficients. * \param gainLpc2 pointer to gain based on new input LPC coefficients. * \param alfd_gains pointer to output gains. * \param s current scale shift factor of x. */ #define ALFDPOW2_SCALE 3 /*static*/ void CLpd_AdaptLowFreqDeemph(FIXP_DBL x[], int lg, FIXP_DBL alfd_gains[], INT s) { { int i, j, k, i_max; FIXP_DBL max, fac; /* Note: This stack array saves temporary accumulation results to be used in * a second run */ /* The size should be limited to (1024/4)/8=32 */ FIXP_DBL tmp_pow2[32]; s = s * 2 + ALFDPOW2_SCALE; s = fMin(31, s); k = 8; i_max = lg / 4; /* ALFD range = 1600Hz (lg = 6400Hz) */ /* find spectral peak */ max = FL2FX_DBL(0.01f) >> s; for (i = 0; i < i_max; i += k) { FIXP_DBL tmp; tmp = FIXP_DBL(0); FIXP_DBL *pX = &x[i]; j = 8; do { FIXP_DBL x0 = *pX++; FIXP_DBL x1 = *pX++; x0 = fPow2Div2(x0); x1 = fPow2Div2(x1); tmp = tmp + (x0 >> (ALFDPOW2_SCALE - 1)); tmp = tmp + (x1 >> (ALFDPOW2_SCALE - 1)); } while ((j = j - 2) != 0); tmp = fMax(tmp, (FL2FX_DBL(0.01f) >> s)); tmp_pow2[i >> 3] = tmp; if (tmp > max) { max = tmp; } } /* deemphasis of all blocks below the peak */ fac = FL2FX_DBL(0.1f) >> 1; for (i = 0; i < i_max; i += k) { FIXP_DBL tmp; INT shifti; tmp = tmp_pow2[i >> 3]; /* tmp = (float)sqrt(tmp/max); */ /* value of tmp is between 8/2*max^2 and max^2 / 2. */ /* required shift factor of division can grow up to 27 (grows exponentially for values toward zero) thus using normalized division to assure valid result. */ { INT sd; if (tmp != (FIXP_DBL)0) { tmp = fDivNorm(max, tmp, &sd); if (sd & 1) { sd++; tmp >>= 1; } } else { tmp = (FIXP_DBL)MAXVAL_DBL; sd = 0; } tmp = invSqrtNorm2(tmp, &shifti); tmp = scaleValue(tmp, shifti - 1 - (sd / 2)); } if (tmp > fac) { fac = tmp; } FIXP_DBL *pX = &x[i]; j = 8; do { FIXP_DBL x0 = pX[0]; FIXP_DBL x1 = pX[1]; x0 = fMultDiv2(x0, fac); x1 = fMultDiv2(x1, fac); x0 = x0 << 2; x1 = x1 << 2; *pX++ = x0; *pX++ = x1; } while ((j = j - 2) != 0); /* Store gains for FAC */ *alfd_gains++ = fac; } } } /** * \brief Interpolated Noise Shaping for mdct coefficients. * This algorithm shapes temporally the spectral noise between * the two spectral noise represention (FDNS_NPTS of resolution). * The noise is shaped monotonically between the two points * using a curved shape to favor the lower gain in mid-frame. * ODFT and amplitud calculation are applied to the 2 LPC coefficients first. * * \param r pointer to spectrum data. * \param rms RMS of output spectrum. * \param lg length of r. * \param A1 pointer to old input LPC coefficients of length M_LP_FILTER_ORDER * scaled by SF_A_COEFFS. * \param A2 pointer to new input LPC coefficients of length M_LP_FILTER_ORDER * scaled by SF_A_COEFFS. * \param bLpc2Mdct flags control lpc2mdct conversion and noise shaping. * \param gainLpc1 pointer to gain based on old input LPC coefficients. * \param gainLpc2 pointer to gain based on new input LPC coefficients. * \param gLpc_e pointer to exponent of gainLpc1 and gainLpc2. */ /* static */ #define NSHAPE_SCALE (4) #define LPC2MDCT_CALC (1) #define LPC2MDCT_GAIN_LOAD (2) #define LPC2MDCT_GAIN_SAVE (4) #define LPC2MDCT_APPLY_NSHAPE (8) void lpc2mdctAndNoiseShaping(FIXP_DBL *r, SHORT *pScale, const INT lg, const INT fdns_npts, const FIXP_LPC *A1, const INT A1_exp, const FIXP_LPC *A2, const INT A2_exp) { FIXP_DBL *tmp2 = NULL; FIXP_DBL rr_minus_one; int i, k, s, step; C_AALLOC_SCRATCH_START(tmp1, FIXP_DBL, FDNS_NPTS * 8) { tmp2 = tmp1 + fdns_npts * 4; /* lpc2mdct() */ /* ODFT. E_LPC_a_weight() for A1 and A2 vectors is included into the loop * below. */ FIXP_DBL f = FL2FXCONST_DBL(0.92f); const FIXP_STP *SinTab; int k_step; /* needed values: sin(phi), cos(phi); phi = i*PI/(2*fdns_npts), i = 0 ... * M_LP_FILTER_ORDER */ switch (fdns_npts) { case 64: SinTab = SineTable512; k_step = (512 / 64); FDK_ASSERT(512 >= 64); break; case 48: SinTab = SineTable384; k_step = 384 / 48; FDK_ASSERT(384 >= 48); break; default: FDK_ASSERT(0); return; } for (i = 0, k = k_step; i < M_LP_FILTER_ORDER; i++, k += k_step) { FIXP_STP cs = SinTab[k]; FIXP_DBL wA1, wA2; wA1 = fMult(A1[i], f); wA2 = fMult(A2[i], f); /* r[i] = A[i]*cos() */ tmp1[2 + i * 2] = fMult(wA1, cs.v.re); tmp2[2 + i * 2] = fMult(wA2, cs.v.re); /* i[i] = A[i]*sin() */ tmp1[3 + i * 2] = -fMult(wA1, cs.v.im); tmp2[3 + i * 2] = -fMult(wA2, cs.v.im); f = fMult(f, FL2FXCONST_DBL(0.92f)); } /* Guarantee at least 2 bits of headroom for the FFT */ /* "3" stands for 1.0 with 2 bits of headroom; (A1_exp + 2) guarantess 2 * bits of headroom if A1_exp > 1 */ int A1_exp_fix = fMax(3, A1_exp + 2); int A2_exp_fix = fMax(3, A2_exp + 2); /* Set 1.0 in the proper format */ tmp1[0] = (FIXP_DBL)(INT)((ULONG)0x80000000 >> A1_exp_fix); tmp2[0] = (FIXP_DBL)(INT)((ULONG)0x80000000 >> A2_exp_fix); tmp1[1] = tmp2[1] = (FIXP_DBL)0; /* Clear the resto of the array */ FDKmemclear( tmp1 + 2 * (M_LP_FILTER_ORDER + 1), 2 * (fdns_npts * 2 - (M_LP_FILTER_ORDER + 1)) * sizeof(FIXP_DBL)); FDKmemclear( tmp2 + 2 * (M_LP_FILTER_ORDER + 1), 2 * (fdns_npts * 2 - (M_LP_FILTER_ORDER + 1)) * sizeof(FIXP_DBL)); /* Guarantee 2 bits of headroom for FFT */ scaleValues(&tmp1[2], (2 * M_LP_FILTER_ORDER), (A1_exp - A1_exp_fix)); scaleValues(&tmp2[2], (2 * M_LP_FILTER_ORDER), (A2_exp - A2_exp_fix)); INT s2; s = A1_exp_fix; s2 = A2_exp_fix; fft(2 * fdns_npts, tmp1, &s); fft(2 * fdns_npts, tmp2, &s2); /* Adjust the exponents of both fft outputs if necessary*/ if (s > s2) { scaleValues(tmp2, 2 * fdns_npts, s2 - s); s2 = s; } else if (s < s2) { scaleValues(tmp1, 2 * fdns_npts, s - s2); s = s2; } FDK_ASSERT(s == s2); } /* Get amplitude and apply gains */ step = lg / fdns_npts; rr_minus_one = (FIXP_DBL)0; for (k = 0; k < fdns_npts; k++) { FIXP_DBL g1, g2, inv_g1_g2, a, b; INT inv_g1_g2_e; int g_e, shift; { FIXP_DBL real, imag; int si1, si2, sInput; real = tmp1[k * 2]; imag = tmp1[k * 2 + 1]; sInput = fMax(fMin(fNorm(real), fNorm(imag)) - 1, 0); real <<= sInput; imag <<= sInput; /* g1_e = si1 - 2*s/2 */ g1 = invSqrtNorm2(fPow2(real) + fPow2(imag), &si1); si1 += sInput; real = tmp2[k * 2]; imag = tmp2[k * 2 + 1]; sInput = fMax(fMin(fNorm(real), fNorm(imag)) - 1, 0); real <<= sInput; imag <<= sInput; /* g2_e = si2 - 2*s/2 */ g2 = invSqrtNorm2(fPow2(real) + fPow2(imag), &si2); si2 += sInput; /* Pick a common scale factor for g1 and g2 */ if (si1 > si2) { g2 >>= si1 - si2; g_e = si1 - s; } else { g1 >>= si2 - si1; g_e = si2 - s; } } /* end of lpc2mdct() */ FDK_ASSERT(g1 >= (FIXP_DBL)0); FDK_ASSERT(g2 >= (FIXP_DBL)0); /* mdct_IntNoiseShaping() */ { /* inv_g1_g2 * 2^inv_g1_g2_e = 1/(g1+g2) */ inv_g1_g2 = (g1 >> 1) + (g2 >> 1); if (inv_g1_g2 != (FIXP_DBL)0) { inv_g1_g2 = fDivNorm(FL2FXCONST_DBL(0.5f), inv_g1_g2, &inv_g1_g2_e); inv_g1_g2_e = inv_g1_g2_e - g_e; } else { inv_g1_g2 = (FIXP_DBL)MAXVAL_DBL; inv_g1_g2_e = 0; } if (g_e < 0) { /* a_e = g_e + inv_g1_g2_e + 1 */ a = scaleValue(fMult(fMult(g1, g2), inv_g1_g2), g_e); /* b_e = g_e + inv_g1_g2_e */ b = fMult(g2 - g1, inv_g1_g2); shift = g_e + inv_g1_g2_e + 1 - NSHAPE_SCALE; } else { /* a_e = (g_e+g_e) + inv_g1_g2_e + 1 */ a = fMult(fMult(g1, g2), inv_g1_g2); /* b_e = (g_e+g_e) + inv_g1_g2_e */ b = scaleValue(fMult(g2 - g1, inv_g1_g2), -g_e); shift = (g_e + g_e) + inv_g1_g2_e + 1 - NSHAPE_SCALE; } for (i = k * step; i < (k + 1) * step; i++) { FIXP_DBL tmp; /* rr[i] = 2*a*r[i] + b*rr[i-1] */ tmp = fMult(a, r[i]); tmp += scaleValue(fMultDiv2(b, rr_minus_one), NSHAPE_SCALE); tmp = scaleValueSaturate(tmp, shift); rr_minus_one = tmp; r[i] = tmp; } } } /* end of mdct_IntNoiseShaping() */ { *pScale += NSHAPE_SCALE; } C_AALLOC_SCRATCH_END(tmp1, FIXP_DBL, FDNS_NPTS * 8) } /** * \brief Calculates the energy. * \param r pointer to spectrum. * \param rs scale factor of spectrum r. * \param lg frame length in audio samples. * \param rms_e pointer to exponent of energy value. * \return mantissa of energy value. */ static FIXP_DBL calcEnergy(const FIXP_DBL *r, const SHORT rs, const INT lg, INT *rms_e) { int headroom = getScalefactor(r, lg); FIXP_DBL rms_m = 0; /* Calculate number of growth bits due to addition */ INT shift = (INT)(fNormz((FIXP_DBL)lg)); shift = 31 - shift; /* Generate 1e-2 in Q-6.37 */ const FIXP_DBL value0_01 = 0x51eb851e; const INT value0_01_exp = -6; /* Find the exponent of the resulting energy value */ *rms_e = ((rs - headroom) << 1) + shift + 1; INT delta = *rms_e - value0_01_exp; if (delta > 0) { /* Limit shift_to 31*/ delta = fMin(31, delta); rms_m = value0_01 >> delta; } else { rms_m = value0_01; *rms_e = value0_01_exp; shift = shift - delta; /* Limit shift_to 31*/ shift = fMin(31, shift); } for (int i = 0; i < lg; i++) { rms_m += fPow2Div2(r[i] << headroom) >> shift; } return rms_m; } /** * \brief TCX gain calculation. * \param pAacDecoderChannelInfo channel context data. * \param r output spectrum. * \param rms_e pointer to mantissa of energy value. * \param rms_e pointer to exponent of energy value. * \param frame the frame index of the LPD super frame. * \param lg the frame length in audio samples. * \param gain_m pointer to mantissa of TCX gain. * \param gain_e pointer to exponent of TCX gain. * \param elFlags element specific parser guidance flags. * \param lg_fb the fullband frame length in audio samples. * \param IGF_bgn the IGF start index. */ static void calcTCXGain(CAacDecoderChannelInfo *pAacDecoderChannelInfo, FIXP_DBL *r, FIXP_DBL rms_m, INT rms_e, const INT frame, const INT lg) { if ((rms_m != (FIXP_DBL)0)) { FIXP_DBL tcx_gain_m; INT tcx_gain_e; CLpd_DecodeGain(&tcx_gain_m, &tcx_gain_e, pAacDecoderChannelInfo->pDynData->specificTo.usac .tcx_global_gain[frame]); /* rms * 2^rms_e = lg/sqrt(sum(spec^2)) */ if (rms_e & 1) { rms_m >>= 1; rms_e++; } { FIXP_DBL fx_lg; INT fx_lg_e, s; INT inv_e; /* lg = fx_lg * 2^fx_lg_e */ s = fNorm((FIXP_DBL)lg); fx_lg = (FIXP_DBL)lg << s; fx_lg_e = DFRACT_BITS - 1 - s; /* 1/sqrt(rms) */ rms_m = invSqrtNorm2(rms_m, &inv_e); rms_m = fMult(rms_m, fx_lg); rms_e = inv_e - (rms_e >> 1) + fx_lg_e; } { int s = fNorm(tcx_gain_m); tcx_gain_m = tcx_gain_m << s; tcx_gain_e -= s; } tcx_gain_m = fMultDiv2(tcx_gain_m, rms_m); tcx_gain_e = tcx_gain_e + rms_e; /* global_gain * 2^(global_gain_e+rms_e) = (10^(global_gain/28)) * rms * * 2^rms_e */ { { tcx_gain_e += 1; } } pAacDecoderChannelInfo->data.usac.tcx_gain[frame] = tcx_gain_m; pAacDecoderChannelInfo->data.usac.tcx_gain_e[frame] = tcx_gain_e; pAacDecoderChannelInfo->specScale[frame] += tcx_gain_e; } } /** * \brief FDNS decoding. * \param pAacDecoderChannelInfo channel context data. * \param pAacDecoderStaticChannelInfo channel context static data. * \param r output spectrum. * \param lg the frame length in audio samples. * \param frame the frame index of the LPD super frame. * \param pScale pointer to current scale shift factor of r[]. * \param A1 old input LPC coefficients of length M_LP_FILTER_ORDER. * \param A2 new input LPC coefficients of length M_LP_FILTER_ORDER. * \param pAlfdGains pointer for ALFD gains output scaled by 1. * \param fdns_npts number of lines (FDNS_NPTS). * \param inf_mask pointer to noise mask. * \param IGF_win_mode IGF window mode (LONG, SHORT, TCX10, TCX20). * \param frameType (IGF_FRAME_DIVISION_AAC_OR_TCX_LONG or * IGF_FRAME_DIVISION_TCX_SHORT_1). * \param elFlags element specific parser guidance flags. * \param lg_fb the fullband frame length in audio samples. * \param IGF_bgn the IGF start index. * \param rms_m mantisse of energy. * \param rms_e exponent of energy. */ /* static */ void CLpd_FdnsDecode(CAacDecoderChannelInfo *pAacDecoderChannelInfo, CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, FIXP_DBL r[], const INT lg, const INT frame, SHORT *pScale, const FIXP_LPC A1[M_LP_FILTER_ORDER], const INT A1_exp, const FIXP_LPC A2[M_LP_FILTER_ORDER], const INT A2_exp, FIXP_DBL pAlfdGains[LFAC / 4], const INT fdns_npts) { /* Weight LPC coefficients using Rm values */ CLpd_AdaptLowFreqDeemph(r, lg, pAlfdGains, *pScale); FIXP_DBL rms_m = (FIXP_DBL)0; INT rms_e = 0; { /* Calculate Energy */ rms_m = calcEnergy(r, *pScale, lg, &rms_e); } calcTCXGain(pAacDecoderChannelInfo, r, rms_m, rms_e, frame, lg); /* Apply ODFT and Noise Shaping. LP coefficient (A1, A2) weighting is done * inside on the fly. */ lpc2mdctAndNoiseShaping(r, pScale, lg, fdns_npts, A1, A1_exp, A2, A2_exp); } /** * find pitch for TCX20 (time domain) concealment. */ static int find_mpitch(FIXP_DBL xri[], int lg) { FIXP_DBL max, pitch; INT pitch_e; int i, n; max = (FIXP_DBL)0; n = 2; /* find maximum below 400Hz */ for (i = 2; i < (lg >> 4); i += 2) { FIXP_DBL tmp = fPow2Div2(xri[i]) + fPow2Div2(xri[i + 1]); if (tmp > max) { max = tmp; n = i; } } // pitch = ((float)lg<<1)/(float)n; pitch = fDivNorm((FIXP_DBL)lg << 1, (FIXP_DBL)n, &pitch_e); pitch >>= fixMax(0, DFRACT_BITS - 1 - pitch_e - 16); /* find pitch multiple under 20ms */ if (pitch >= (FIXP_DBL)((256 << 16) - 1)) { /*231.0f*/ n = 256; } else { FIXP_DBL mpitch = pitch; while (mpitch < (FIXP_DBL)(255 << 16)) { mpitch += pitch; } n = (int)(mpitch - pitch) >> 16; } return (n); } /** * number of spectral coefficients / time domain samples using frame mode as * index. */ static const int lg_table_ccfl[2][4] = { {256, 256, 512, 1024}, /* coreCoderFrameLength = 1024 */ {192, 192, 384, 768} /* coreCoderFrameLength = 768 */ }; /** * \brief Decode and render one MDCT-TCX frame. * \param pAacDecoderChannelInfo channel context data. * \param lg the frame length in audio samples. * \param frame the frame index of the LPD super frame. */ static void CLpd_TcxDecode( CAacDecoderChannelInfo *pAacDecoderChannelInfo, CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, UINT flags, int mod, int last_mod, int frame, int frameOk) { FIXP_DBL *pAlfd_gains = pAacDecoderStaticChannelInfo->last_alfd_gains; ULONG *pSeed = &pAacDecoderStaticChannelInfo->nfRandomSeed; int lg = (pAacDecoderChannelInfo->granuleLength == 128) ? lg_table_ccfl[0][mod + 0] : lg_table_ccfl[1][mod + 0]; int next_frame = frame + (1 << (mod - 1)); int isFullBandLpd = 0; /* Obtain r[] vector by combining the quant[] and noise[] vectors */ { FIXP_DBL noise_level; FIXP_DBL *coeffs = SPEC_TCX(pAacDecoderChannelInfo->pSpectralCoefficient, frame, pAacDecoderChannelInfo->granuleLength, isFullBandLpd); int scale = pAacDecoderChannelInfo->specScale[frame]; int i, nfBgn, nfEnd; UCHAR tcx_noise_factor = pAacDecoderChannelInfo->pDynData->specificTo.usac .tcx_noise_factor[frame]; /* find pitch for bfi case */ pAacDecoderStaticChannelInfo->last_tcx_pitch = find_mpitch(coeffs, lg); if (frameOk) { /* store for concealment */ pAacDecoderStaticChannelInfo->last_tcx_noise_factor = tcx_noise_factor; } else { /* restore last frames value */ tcx_noise_factor = pAacDecoderStaticChannelInfo->last_tcx_noise_factor; } noise_level = (FIXP_DBL)((LONG)FL2FXCONST_DBL(0.0625f) * (8 - tcx_noise_factor)); noise_level = scaleValue(noise_level, -scale); const FIXP_DBL neg_noise_level = -noise_level; { nfBgn = lg / 6; nfEnd = lg; } for (i = nfBgn; i < nfEnd - 7; i += 8) { LONG tmp; /* Fill all 8 consecutive zero coeffs with noise */ tmp = coeffs[i + 0] | coeffs[i + 1] | coeffs[i + 2] | coeffs[i + 3] | coeffs[i + 4] | coeffs[i + 5] | coeffs[i + 6] | coeffs[i + 7]; if (tmp == 0) { for (int k = i; k < i + 8; k++) { UsacRandomSign(pSeed) ? (coeffs[k] = neg_noise_level) : (coeffs[k] = noise_level); } } } if ((nfEnd - i) > 0) { /* noise filling for last "band" with less than 8 bins */ LONG tmp = (LONG)coeffs[i]; int k; FDK_ASSERT((nfEnd - i) < 8); for (k = 1; k < (nfEnd - i); k++) { tmp |= (LONG)coeffs[i + k]; } if (tmp == 0) { for (k = i; k < nfEnd; k++) { UsacRandomSign(pSeed) ? (coeffs[k] = neg_noise_level) : (coeffs[k] = noise_level); } } } } { /* Convert LPC to LP domain */ if (last_mod == 0) { /* Note: The case where last_mod == 255 is handled by other means * in CLpdChannelStream_Read() */ E_LPC_f_lsp_a_conversion( pAacDecoderChannelInfo->data.usac.lsp_coeff[frame], pAacDecoderChannelInfo->data.usac.lp_coeff[frame], &pAacDecoderChannelInfo->data.usac.lp_coeff_exp[frame]); } E_LPC_f_lsp_a_conversion( pAacDecoderChannelInfo->data.usac.lsp_coeff[next_frame], pAacDecoderChannelInfo->data.usac.lp_coeff[next_frame], &pAacDecoderChannelInfo->data.usac.lp_coeff_exp[next_frame]); /* FDNS decoding */ CLpd_FdnsDecode( pAacDecoderChannelInfo, pAacDecoderStaticChannelInfo, SPEC_TCX(pAacDecoderChannelInfo->pSpectralCoefficient, frame, pAacDecoderChannelInfo->granuleLength, isFullBandLpd), lg, frame, pAacDecoderChannelInfo->specScale + frame, pAacDecoderChannelInfo->data.usac.lp_coeff[frame], pAacDecoderChannelInfo->data.usac.lp_coeff_exp[frame], pAacDecoderChannelInfo->data.usac.lp_coeff[next_frame], pAacDecoderChannelInfo->data.usac.lp_coeff_exp[next_frame], pAlfd_gains, pAacDecoderChannelInfo->granuleLength / 2 /* == FDNS_NPTS(ccfl) */ ); } } /** * \brief Read the tcx_coding bitstream part * \param hBs bitstream handle to read from. * \param pAacDecoderChannelInfo channel context info to store data into. * \param lg the frame length in audio samples. * \param first_tcx_flag flag indicating that this is the first TCX frame. * \param frame the frame index of the LPD super frame. */ static AAC_DECODER_ERROR CLpd_TCX_Read( HANDLE_FDK_BITSTREAM hBs, CAacDecoderChannelInfo *pAacDecoderChannelInfo, CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, int lg, int first_tcx_flag, int frame, UINT flags) { AAC_DECODER_ERROR errorAAC = AAC_DEC_OK; ARITH_CODING_ERROR error = ARITH_CODER_OK; FIXP_DBL *pSpec; int arith_reset_flag = 0; int isFullBandLpd = 0; pSpec = SPEC_TCX(pAacDecoderChannelInfo->pSpectralCoefficient, frame, pAacDecoderChannelInfo->granuleLength, isFullBandLpd); /* TCX noise level */ { pAacDecoderChannelInfo->pDynData->specificTo.usac.tcx_noise_factor[frame] = FDKreadBits(hBs, 3); } /* TCX global gain */ pAacDecoderChannelInfo->pDynData->specificTo.usac.tcx_global_gain[frame] = FDKreadBits(hBs, 7); /* Arithmetic coded residual/spectrum */ if (first_tcx_flag) { if (flags & AC_INDEP) { arith_reset_flag = 1; } else { arith_reset_flag = FDKreadBits(hBs, 1); } } /* CArco_DecodeArithData() output scale of "pSpec" is DFRACT_BITS-1 */ error = CArco_DecodeArithData(pAacDecoderStaticChannelInfo->hArCo, hBs, pSpec, lg, lg, arith_reset_flag); /* Rescale residual/spectrum */ { int scale = getScalefactor(pSpec, lg) - 2; /* Leave 2 bits headroom */ /* Exponent of CArco_DecodeArithData() output is DFRACT_BITS; integer * values. */ scaleValues(pSpec, lg, scale); scale = DFRACT_BITS - 1 - scale; pAacDecoderChannelInfo->specScale[frame] = scale; } if (error == ARITH_CODER_ERROR) errorAAC = AAC_DEC_UNKNOWN; return errorAAC; } /** * \brief translate lpd_mode into the mod[] array which describes the mode of * each each LPD frame * \param mod[] the array that will be filled with the mode indexes of the * inidividual frames. * \param lpd_mode the lpd_mode field read from the lpd_channel_stream */ static AAC_DECODER_ERROR CLpd_ReadAndMapLpdModeToModArray( UCHAR mod[4], HANDLE_FDK_BITSTREAM hBs, UINT elFlags) { int lpd_mode; { lpd_mode = FDKreadBits(hBs, 5); if (lpd_mode > 25 || lpd_mode < 0) { return AAC_DEC_PARSE_ERROR; } switch (lpd_mode) { case 25: /* 1 80MS frame */ mod[0] = mod[1] = mod[2] = mod[3] = 3; break; case 24: /* 2 40MS frames */ mod[0] = mod[1] = mod[2] = mod[3] = 2; break; default: switch (lpd_mode >> 2) { case 4: /* lpd_mode 19 - 16 => 1 40MS and 2 20MS frames */ mod[0] = mod[1] = 2; mod[2] = (lpd_mode & 1) ? 1 : 0; mod[3] = (lpd_mode & 2) ? 1 : 0; break; case 5: /* lpd_mode 23 - 20 => 2 20MS and 1 40MS frames */ mod[2] = mod[3] = 2; mod[0] = (lpd_mode & 1) ? 1 : 0; mod[1] = (lpd_mode & 2) ? 1 : 0; break; default: /* lpd_mode < 16 => 4 20MS frames */ mod[0] = (lpd_mode & 1) ? 1 : 0; mod[1] = (lpd_mode & 2) ? 1 : 0; mod[2] = (lpd_mode & 4) ? 1 : 0; mod[3] = (lpd_mode & 8) ? 1 : 0; break; } break; } } return AAC_DEC_OK; } static void CLpd_Reset( CAacDecoderChannelInfo *pAacDecoderChannelInfo, CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, int keep_past_signal) { int i; /* Reset TCX / ACELP common memory */ if (!keep_past_signal) { FDKmemclear(pAacDecoderStaticChannelInfo->old_synth, sizeof(pAacDecoderStaticChannelInfo->old_synth)); } /* Initialize the LSFs */ for (i = 0; i < M_LP_FILTER_ORDER; i++) { pAacDecoderStaticChannelInfo->lpc4_lsf[i] = fdk_dec_lsf_init[i]; } /* Reset memory needed by bass post-filter */ FDKmemclear(pAacDecoderStaticChannelInfo->mem_bpf, sizeof(pAacDecoderStaticChannelInfo->mem_bpf)); pAacDecoderStaticChannelInfo->old_bpf_control_info = 0; for (i = 0; i < SYN_SFD; i++) { pAacDecoderStaticChannelInfo->old_T_pf[i] = 64; pAacDecoderStaticChannelInfo->old_gain_pf[i] = (FIXP_DBL)0; } /* Reset ACELP memory */ CLpd_AcelpReset(&pAacDecoderStaticChannelInfo->acelp); pAacDecoderStaticChannelInfo->last_lpc_lost = 0; /* prev_lpc_lost */ pAacDecoderStaticChannelInfo->last_tcx_pitch = L_DIV; /* pitch_tcx */ pAacDecoderStaticChannelInfo->numLostLpdFrames = 0; /* nbLostCmpt */ } /* * Externally visible functions */ AAC_DECODER_ERROR CLpdChannelStream_Read( HANDLE_FDK_BITSTREAM hBs, CAacDecoderChannelInfo *pAacDecoderChannelInfo, CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, const SamplingRateInfo *pSamplingRateInfo, UINT flags) { AAC_DECODER_ERROR error = AAC_DEC_OK; int first_tcx_flag; int k, nbDiv, fFacDataPresent, first_lpd_flag, acelp_core_mode, facGetMemState = 0; UCHAR *mod = pAacDecoderChannelInfo->data.usac.mod; int lpd_mode_last, prev_frame_was_lpd; USAC_COREMODE core_mode_last; const int lg_table_offset = 0; const int *lg_table = (pAacDecoderChannelInfo->granuleLength == 128) ? &lg_table_ccfl[0][lg_table_offset] : &lg_table_ccfl[1][lg_table_offset]; int last_lpc_lost = pAacDecoderStaticChannelInfo->last_lpc_lost; int last_frame_ok = CConcealment_GetLastFrameOk( &pAacDecoderStaticChannelInfo->concealmentInfo, 1); INT i_offset; UINT samplingRate; samplingRate = pSamplingRateInfo->samplingRate; i_offset = (INT)(samplingRate * PIT_MIN_12k8 + (FSCALE_DENOM / 2)) / FSCALE_DENOM - (INT)PIT_MIN_12k8; if ((samplingRate < FAC_FSCALE_MIN) || (samplingRate > FAC_FSCALE_MAX)) { error = AAC_DEC_PARSE_ERROR; goto bail; } acelp_core_mode = FDKreadBits(hBs, 3); /* lpd_mode */ error = CLpd_ReadAndMapLpdModeToModArray(mod, hBs, 0); if (error != AAC_DEC_OK) { goto bail; } /* bpf_control_info */ pAacDecoderChannelInfo->data.usac.bpf_control_info = FDKreadBit(hBs); /* last_core_mode */ prev_frame_was_lpd = FDKreadBit(hBs); /* fac_data_present */ fFacDataPresent = FDKreadBit(hBs); /* Set valid values from * pAacDecoderStaticChannelInfo->{last_core_mode,last_lpd_mode} */ pAacDecoderChannelInfo->data.usac.core_mode_last = pAacDecoderStaticChannelInfo->last_core_mode; lpd_mode_last = pAacDecoderChannelInfo->data.usac.lpd_mode_last = pAacDecoderStaticChannelInfo->last_lpd_mode; if (prev_frame_was_lpd == 0) { /* Last frame was FD */ pAacDecoderChannelInfo->data.usac.core_mode_last = FD_LONG; pAacDecoderChannelInfo->data.usac.lpd_mode_last = 255; } else { /* Last frame was LPD */ pAacDecoderChannelInfo->data.usac.core_mode_last = LPD; if (((mod[0] == 0) && fFacDataPresent) || ((mod[0] != 0) && !fFacDataPresent)) { /* Currend mod is ACELP, fac data present -> TCX, current mod TCX, no fac * data -> TCX */ if (lpd_mode_last == 0) { /* Bit stream interruption detected. Assume last TCX mode as TCX20. */ pAacDecoderChannelInfo->data.usac.lpd_mode_last = 1; } /* Else assume that remembered TCX mode is correct. */ } else { pAacDecoderChannelInfo->data.usac.lpd_mode_last = 0; } } first_lpd_flag = (pAacDecoderChannelInfo->data.usac.core_mode_last != LPD); /* Depends on bitstream configuration */ first_tcx_flag = 1; if (pAacDecoderStaticChannelInfo->last_core_mode != LPD) { /* ATTENTION: Reset depends on what we rendered before! */ CLpd_Reset(pAacDecoderChannelInfo, pAacDecoderStaticChannelInfo, 0); if (!last_frame_ok) { /* If last rendered frame was not LPD and first lpd flag is not set, this * must be an error - set last_lpc_lost flag */ last_lpc_lost |= (first_lpd_flag) ? 0 : 1; } } core_mode_last = pAacDecoderChannelInfo->data.usac.core_mode_last; lpd_mode_last = pAacDecoderChannelInfo->data.usac.lpd_mode_last; nbDiv = NB_DIV; /* k is the frame index. If a frame is of size 40MS or 80MS, this frame index is incremented 2 or 4 instead of 1 respectively. */ k = 0; while (k < nbDiv) { /* Reset FAC data pointers in order to avoid applying old random FAC data. */ pAacDecoderChannelInfo->data.usac.fac_data[k] = NULL; if ((k == 0 && core_mode_last == LPD && fFacDataPresent) || (lpd_mode_last == 0 && mod[k] > 0) || ((lpd_mode_last != 255) && lpd_mode_last > 0 && mod[k] == 0)) { int err; /* Assign FAC memory */ pAacDecoderChannelInfo->data.usac.fac_data[k] = CLpd_FAC_GetMemory(pAacDecoderChannelInfo, mod, &facGetMemState); /* FAC for (ACELP -> TCX) or (TCX -> ACELP) */ err = CLpd_FAC_Read( hBs, pAacDecoderChannelInfo->data.usac.fac_data[k], pAacDecoderChannelInfo->data.usac.fac_data_e, pAacDecoderChannelInfo->granuleLength, /* == fac_length */ 0, k); if (err != 0) { error = AAC_DEC_PARSE_ERROR; goto bail; } } if (mod[k] == 0) /* acelp-mode */ { int err; err = CLpd_AcelpRead( hBs, &pAacDecoderChannelInfo->data.usac.acelp[k], acelp_core_mode, pAacDecoderChannelInfo->granuleLength * 8 /* coreCoderFrameLength */, i_offset); if (err != 0) { error = AAC_DEC_PARSE_ERROR; goto bail; } lpd_mode_last = 0; k++; } else /* mode != 0 => TCX */ { error = CLpd_TCX_Read(hBs, pAacDecoderChannelInfo, pAacDecoderStaticChannelInfo, lg_table[mod[k]], first_tcx_flag, k, flags); lpd_mode_last = mod[k]; first_tcx_flag = 0; k += 1 << (mod[k] - 1); } if (error != AAC_DEC_OK) { error = AAC_DEC_PARSE_ERROR; goto bail; } } { int err; /* Read LPC coefficients */ err = CLpc_Read( hBs, pAacDecoderChannelInfo->data.usac.lsp_coeff, pAacDecoderStaticChannelInfo->lpc4_lsf, pAacDecoderChannelInfo->data.usac.lsf_adaptive_mean_cand, pAacDecoderChannelInfo->data.usac.aStability, mod, first_lpd_flag, /* if last lpc4 is available from concealment do not extrapolate lpc0 from lpc2 */ (mod[0] & 0x3) ? 0 : (last_lpc_lost && pAacDecoderStaticChannelInfo->last_core_mode != LPD), last_frame_ok); if (err != 0) { error = AAC_DEC_PARSE_ERROR; goto bail; } } /* adjust old lsp[] following to a bad frame (to avoid overshoot) (ref: * dec_LPD.c) */ if (last_lpc_lost && !last_frame_ok) { int k_next; k = 0; while (k < nbDiv) { int i; k_next = k + (((mod[k] & 0x3) == 0) ? 1 : (1 << (mod[k] - 1))); FIXP_LPC *lsp_old = pAacDecoderChannelInfo->data.usac.lsp_coeff[k]; FIXP_LPC *lsp_new = pAacDecoderChannelInfo->data.usac.lsp_coeff[k_next]; for (i = 0; i < M_LP_FILTER_ORDER; i++) { if (lsp_new[i] < lsp_old[i]) { lsp_old[i] = lsp_new[i]; } } k = k_next; } } if (!CConcealment_GetLastFrameOk( &pAacDecoderStaticChannelInfo->concealmentInfo, 1)) { E_LPC_f_lsp_a_conversion( pAacDecoderChannelInfo->data.usac.lsp_coeff[0], pAacDecoderChannelInfo->data.usac.lp_coeff[0], &pAacDecoderChannelInfo->data.usac.lp_coeff_exp[0]); } else if (pAacDecoderStaticChannelInfo->last_lpd_mode != 0) { if (pAacDecoderStaticChannelInfo->last_lpd_mode == 255) { /* We need it for TCX decoding or ACELP excitation update */ E_LPC_f_lsp_a_conversion( pAacDecoderChannelInfo->data.usac.lsp_coeff[0], pAacDecoderChannelInfo->data.usac.lp_coeff[0], &pAacDecoderChannelInfo->data.usac.lp_coeff_exp[0]); } else { /* last_lpd_mode was TCX */ /* Copy old LPC4 LP domain coefficients to LPC0 LP domain buffer (to avoid * converting LSP coefficients again). */ FDKmemcpy(pAacDecoderChannelInfo->data.usac.lp_coeff[0], pAacDecoderStaticChannelInfo->lp_coeff_old[0], M_LP_FILTER_ORDER * sizeof(FIXP_LPC)); pAacDecoderChannelInfo->data.usac.lp_coeff_exp[0] = pAacDecoderStaticChannelInfo->lp_coeff_old_exp[0]; } } /* case last_lpd_mode was ACELP is handled by CLpd_TcxDecode() */ if (fFacDataPresent && (core_mode_last != LPD)) { int prev_frame_was_short; prev_frame_was_short = FDKreadBit(hBs); if (prev_frame_was_short) { core_mode_last = pAacDecoderChannelInfo->data.usac.core_mode_last = FD_SHORT; pAacDecoderChannelInfo->data.usac.lpd_mode_last = 255; if ((pAacDecoderStaticChannelInfo->last_core_mode != FD_SHORT) && CConcealment_GetLastFrameOk( &pAacDecoderStaticChannelInfo->concealmentInfo, 1)) { /* USAC Conformance document: short_fac_flag shall be encoded with a value of 1 if the window_sequence of the previous frame was 2 (EIGHT_SHORT_SEQUENCE). Otherwise short_fac_flag shall be encoded with a value of 0. */ error = AAC_DEC_PARSE_ERROR; goto bail; } } /* Assign memory */ pAacDecoderChannelInfo->data.usac.fac_data[0] = CLpd_FAC_GetMemory(pAacDecoderChannelInfo, mod, &facGetMemState); { int err; /* FAC for FD -> ACELP */ err = CLpd_FAC_Read( hBs, pAacDecoderChannelInfo->data.usac.fac_data[0], pAacDecoderChannelInfo->data.usac.fac_data_e, CLpd_FAC_getLength(core_mode_last != FD_SHORT, pAacDecoderChannelInfo->granuleLength), 1, 0); if (err != 0) { error = AAC_DEC_PARSE_ERROR; goto bail; } } } bail: if (error == AAC_DEC_OK) { /* check consitency of last core/lpd mode values */ if ((pAacDecoderChannelInfo->data.usac.core_mode_last != pAacDecoderStaticChannelInfo->last_core_mode) && (pAacDecoderStaticChannelInfo->last_lpc_lost == 0)) { /* Something got wrong! */ /* error = AAC_DEC_PARSE_ERROR; */ /* Throwing errors does not help */ } else if ((pAacDecoderChannelInfo->data.usac.core_mode_last == LPD) && (pAacDecoderChannelInfo->data.usac.lpd_mode_last != pAacDecoderStaticChannelInfo->last_lpd_mode) && (pAacDecoderStaticChannelInfo->last_lpc_lost == 0)) { /* Something got wrong! */ /* error = AAC_DEC_PARSE_ERROR; */ /* Throwing errors does not help */ } } return error; } void CLpdChannelStream_Decode( CAacDecoderChannelInfo *pAacDecoderChannelInfo, CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, UINT flags) { UCHAR *mod = pAacDecoderChannelInfo->data.usac.mod; int k; UCHAR last_lpd_mode; int nbDiv = NB_DIV; /* k is the frame index. If a frame is of size 40MS or 80MS, this frame index is incremented 2 or 4 instead of 1 respectively. */ k = 0; last_lpd_mode = pAacDecoderChannelInfo->data.usac .lpd_mode_last; /* could be different to what has been rendered */ while (k < nbDiv) { if (mod[k] == 0) { /* ACELP */ /* If FAC (fac_data[k] != NULL), and previous frame was TCX, apply (TCX) * gains to FAC data */ if (last_lpd_mode > 0 && last_lpd_mode != 255 && pAacDecoderChannelInfo->data.usac.fac_data[k]) { CFac_ApplyGains(pAacDecoderChannelInfo->data.usac.fac_data[k], pAacDecoderChannelInfo->granuleLength, pAacDecoderStaticChannelInfo->last_tcx_gain, pAacDecoderStaticChannelInfo->last_alfd_gains, (last_lpd_mode < 4) ? last_lpd_mode : 3); pAacDecoderChannelInfo->data.usac.fac_data_e[k] += pAacDecoderStaticChannelInfo->last_tcx_gain_e; } } else { /* TCX */ CLpd_TcxDecode(pAacDecoderChannelInfo, pAacDecoderStaticChannelInfo, flags, mod[k], last_lpd_mode, k, 1 /* frameOk == 1 */ ); /* Store TCX gain scale for next possible FAC transition. */ pAacDecoderStaticChannelInfo->last_tcx_gain = pAacDecoderChannelInfo->data.usac.tcx_gain[k]; pAacDecoderStaticChannelInfo->last_tcx_gain_e = pAacDecoderChannelInfo->data.usac.tcx_gain_e[k]; /* If FAC (fac_data[k] != NULL), apply gains */ if (last_lpd_mode == 0 && pAacDecoderChannelInfo->data.usac.fac_data[k]) { CFac_ApplyGains( pAacDecoderChannelInfo->data.usac.fac_data[k], pAacDecoderChannelInfo->granuleLength /* == fac_length */, pAacDecoderChannelInfo->data.usac.tcx_gain[k], pAacDecoderStaticChannelInfo->last_alfd_gains, mod[k]); pAacDecoderChannelInfo->data.usac.fac_data_e[k] += pAacDecoderChannelInfo->data.usac.tcx_gain_e[k]; } } /* remember previous mode */ last_lpd_mode = mod[k]; /* Increase k to next frame */ k += (mod[k] == 0) ? 1 : (1 << (mod[k] - 1)); } } AAC_DECODER_ERROR CLpd_RenderTimeSignal( CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, CAacDecoderChannelInfo *pAacDecoderChannelInfo, PCM_DEC *pTimeData, INT lFrame, SamplingRateInfo *pSamplingRateInfo, UINT frameOk, const INT aacOutDataHeadroom, UINT flags, UINT strmFlags) { UCHAR *mod = pAacDecoderChannelInfo->data.usac.mod; AAC_DECODER_ERROR error = AAC_DEC_OK; int k, i_offset; int last_k; int nrSamples = 0; int facFB = 1; int nbDiv = NB_DIV; int lDiv = lFrame / nbDiv; /* length of division (acelp or tcx20 frame)*/ int lFac = lDiv / 2; int nbSubfr = lFrame / (nbDiv * L_SUBFR); /* number of subframes per division */ int nbSubfrSuperfr = nbDiv * nbSubfr; int synSfd = (nbSubfrSuperfr / 2) - BPF_SFD; int SynDelay = synSfd * L_SUBFR; int aacDelay = lFrame / 2; /* In respect to the reference software, the synth pointer here is lagging by aacDelay ( == SYN_DELAY + BPF_DELAY ) samples. The corresponding old synthesis samples are handled by the IMDCT overlap. */ FIXP_DBL *synth_buf = pAacDecoderChannelInfo->pComStaticData->pWorkBufferCore1->synth_buf; FIXP_DBL *synth = synth_buf + PIT_MAX_MAX - BPF_DELAY; UCHAR last_lpd_mode, last_last_lpd_mode, last_lpc_lost, last_frame_lost; INT pitch[NB_SUBFR_SUPERFR + SYN_SFD]; FIXP_DBL pit_gain[NB_SUBFR_SUPERFR + SYN_SFD]; const int *lg_table; int lg_table_offset = 0; UINT samplingRate = pSamplingRateInfo->samplingRate; FDKmemclear(pitch, (NB_SUBFR_SUPERFR + SYN_SFD) * sizeof(INT)); if (flags & AACDEC_FLUSH) { CLpd_Reset(pAacDecoderChannelInfo, pAacDecoderStaticChannelInfo, flags & AACDEC_FLUSH); frameOk = 0; } switch (lFrame) { case 1024: lg_table = &lg_table_ccfl[0][lg_table_offset]; break; case 768: lg_table = &lg_table_ccfl[1][lg_table_offset]; break; default: FDK_ASSERT(0); return AAC_DEC_UNKNOWN; } last_frame_lost = !CConcealment_GetLastFrameOk( &pAacDecoderStaticChannelInfo->concealmentInfo, 0); /* Maintain LPD mode from previous frame */ if ((pAacDecoderStaticChannelInfo->last_core_mode == FD_LONG) || (pAacDecoderStaticChannelInfo->last_core_mode == FD_SHORT)) { pAacDecoderStaticChannelInfo->last_lpd_mode = 255; } if (!frameOk) { FIXP_DBL old_tcx_gain; FIXP_SGL old_stab; SCHAR old_tcx_gain_e; int nLostSf; last_lpd_mode = pAacDecoderStaticChannelInfo->last_lpd_mode; old_tcx_gain = pAacDecoderStaticChannelInfo->last_tcx_gain; old_tcx_gain_e = pAacDecoderStaticChannelInfo->last_tcx_gain_e; old_stab = pAacDecoderStaticChannelInfo->oldStability; nLostSf = pAacDecoderStaticChannelInfo->numLostLpdFrames; /* patch the last LPD mode */ pAacDecoderChannelInfo->data.usac.lpd_mode_last = last_lpd_mode; /* Do mode extrapolation and repeat the previous mode: if previous mode = ACELP -> ACELP if previous mode = TCX-20/40 -> TCX-20 if previous mode = TCX-80 -> TCX-80 notes: - ACELP is not allowed after TCX (no pitch information to reuse) - TCX-40 is not allowed in the mode repetition to keep the logic simple */ switch (last_lpd_mode) { case 0: mod[0] = mod[1] = mod[2] = mod[3] = 0; /* -> ACELP concealment */ break; case 3: mod[0] = mod[1] = mod[2] = mod[3] = 3; /* -> TCX FD concealment */ break; case 2: mod[0] = mod[1] = mod[2] = mod[3] = 2; /* -> TCX FD concealment */ break; case 1: default: mod[0] = mod[1] = mod[2] = mod[3] = 4; /* -> TCX TD concealment */ break; } /* LPC extrapolation */ CLpc_Conceal(pAacDecoderChannelInfo->data.usac.lsp_coeff, pAacDecoderStaticChannelInfo->lpc4_lsf, pAacDecoderStaticChannelInfo->lsf_adaptive_mean, /*(pAacDecoderStaticChannelInfo->numLostLpdFrames == 0) ||*/ (last_lpd_mode == 255)); if ((last_lpd_mode > 0) && (last_lpd_mode < 255)) { /* Copy old LPC4 LP domain coefficients to LPC0 LP domain buffer (to avoid * converting LSP coefficients again). */ FDKmemcpy(pAacDecoderChannelInfo->data.usac.lp_coeff[0], pAacDecoderStaticChannelInfo->lp_coeff_old[0], M_LP_FILTER_ORDER * sizeof(FIXP_LPC)); pAacDecoderChannelInfo->data.usac.lp_coeff_exp[0] = pAacDecoderStaticChannelInfo->lp_coeff_old_exp[0]; } /* case last_lpd_mode was ACELP is handled by CLpd_TcxDecode() */ /* case last_lpd_mode was Time domain TCX concealment is handled after this * "if (!frameOk)"-block */ /* k is the frame index. If a frame is of size 40MS or 80MS, this frame index is incremented 2 or 4 instead of 1 respectively. */ k = 0; while (k < nbDiv) { pAacDecoderChannelInfo->data.usac.tcx_gain[k] = old_tcx_gain; pAacDecoderChannelInfo->data.usac.tcx_gain_e[k] = old_tcx_gain_e; /* restore stability value from last frame */ pAacDecoderChannelInfo->data.usac.aStability[k] = old_stab; /* Increase k to next frame */ k += ((mod[k] & 0x3) == 0) ? 1 : (1 << ((mod[k] & 0x3) - 1)); nLostSf++; } } else { if ((pAacDecoderStaticChannelInfo->last_lpd_mode == 4) && (mod[0] > 0)) { /* Copy old LPC4 LP domain coefficients to LPC0 LP domain buffer (to avoid * converting LSP coefficients again). */ FDKmemcpy(pAacDecoderChannelInfo->data.usac.lp_coeff[0], pAacDecoderStaticChannelInfo->lp_coeff_old[0], M_LP_FILTER_ORDER * sizeof(FIXP_LPC)); pAacDecoderChannelInfo->data.usac.lp_coeff_exp[0] = pAacDecoderStaticChannelInfo->lp_coeff_old_exp[0]; } } Acelp_PreProcessing(synth_buf, pAacDecoderStaticChannelInfo->old_synth, pitch, pAacDecoderStaticChannelInfo->old_T_pf, pit_gain, pAacDecoderStaticChannelInfo->old_gain_pf, samplingRate, &i_offset, lFrame, synSfd, nbSubfrSuperfr); /* k is the frame index. If a frame is of size 40MS or 80MS, this frame index is incremented 2 or 4 instead of 1 respectively. */ k = 0; last_k = -1; /* mark invalid */ last_lpd_mode = pAacDecoderStaticChannelInfo->last_lpd_mode; last_last_lpd_mode = pAacDecoderStaticChannelInfo->last_last_lpd_mode; last_lpc_lost = pAacDecoderStaticChannelInfo->last_lpc_lost | last_frame_lost; /* This buffer must be avalable for the case of FD->ACELP transition. The beginning of the buffer is used after the BPF to overwrite the output signal. Only the FAC area must be affected by the BPF */ while (k < nbDiv) { if (frameOk == 0) { pAacDecoderStaticChannelInfo->numLostLpdFrames++; } else { last_frame_lost |= (pAacDecoderStaticChannelInfo->numLostLpdFrames > 0) ? 1 : 0; pAacDecoderStaticChannelInfo->numLostLpdFrames = 0; } if (mod[k] == 0 || mod[k] == 4) { /* ACELP or TCX time domain concealment */ FIXP_DBL *acelp_out; /* FAC management */ if ((last_lpd_mode != 0) && (last_lpd_mode != 4)) /* TCX TD concealment */ { FIXP_DBL *pFacData = NULL; if (frameOk && !last_frame_lost) { pFacData = pAacDecoderChannelInfo->data.usac.fac_data[k]; } nrSamples += CLpd_FAC_Mdct2Acelp( &pAacDecoderStaticChannelInfo->IMdct, synth + nrSamples, pFacData, pAacDecoderChannelInfo->data.usac.fac_data_e[k], pAacDecoderChannelInfo->data.usac.lp_coeff[k], pAacDecoderChannelInfo->data.usac.lp_coeff_exp[k], lFrame - nrSamples, CLpd_FAC_getLength( (pAacDecoderStaticChannelInfo->last_core_mode != FD_SHORT) || (k > 0), lFac), (pAacDecoderStaticChannelInfo->last_core_mode != LPD) && (k == 0), 0); FDKmemcpy( synth + nrSamples, pAacDecoderStaticChannelInfo->IMdct.overlap.time, pAacDecoderStaticChannelInfo->IMdct.ov_offset * sizeof(FIXP_DBL)); { FIXP_LPC *lp_prev = pAacDecoderChannelInfo->data.usac .lp_coeff[0]; /* init value does not real matter */ INT lp_prev_exp = pAacDecoderChannelInfo->data.usac.lp_coeff_exp[0]; if (last_lpd_mode != 255) { /* last mode was tcx */ last_k = k - (1 << (last_lpd_mode - 1)); if (last_k < 0) { lp_prev = pAacDecoderStaticChannelInfo->lp_coeff_old[1]; lp_prev_exp = pAacDecoderStaticChannelInfo->lp_coeff_old_exp[1]; } else { lp_prev = pAacDecoderChannelInfo->data.usac.lp_coeff[last_k]; lp_prev_exp = pAacDecoderChannelInfo->data.usac.lp_coeff_exp[last_k]; } } CLpd_AcelpPrepareInternalMem( synth + aacDelay + k * lDiv, last_lpd_mode, (last_last_lpd_mode == 4) ? 0 : last_last_lpd_mode, pAacDecoderChannelInfo->data.usac.lp_coeff[k], pAacDecoderChannelInfo->data.usac.lp_coeff_exp[k], lp_prev, lp_prev_exp, &pAacDecoderStaticChannelInfo->acelp, lFrame, (last_frame_lost && k < 2), mod[k]); } } else { if (k == 0 && pAacDecoderStaticChannelInfo->IMdct.ov_offset != lFrame / facFB / 2) { pAacDecoderStaticChannelInfo->IMdct.ov_offset = lFrame / facFB / 2; } nrSamples += imdct_drain(&pAacDecoderStaticChannelInfo->IMdct, synth + nrSamples, lFrame / facFB - nrSamples); } if (nrSamples >= lFrame / facFB) { /* Write ACELP time domain samples into IMDCT overlap buffer at * pAacDecoderStaticChannelInfo->IMdct.overlap.time + * pAacDecoderStaticChannelInfo->IMdct.ov_offset */ acelp_out = pAacDecoderStaticChannelInfo->IMdct.overlap.time + pAacDecoderStaticChannelInfo->IMdct.ov_offset; /* Account ACELP time domain output samples to overlap buffer */ pAacDecoderStaticChannelInfo->IMdct.ov_offset += lDiv; } else { /* Write ACELP time domain samples into output buffer at pTimeData + * nrSamples */ acelp_out = synth + nrSamples; /* Account ACELP time domain output samples to output buffer */ nrSamples += lDiv; } if (mod[k] == 4) { pAacDecoderStaticChannelInfo->acelp.wsyn_rms = scaleValue( pAacDecoderChannelInfo->data.usac.tcx_gain[k], fixMin(0, pAacDecoderChannelInfo->data.usac.tcx_gain_e[k] - SF_EXC)); CLpd_TcxTDConceal(&pAacDecoderStaticChannelInfo->acelp, &pAacDecoderStaticChannelInfo->last_tcx_pitch, pAacDecoderChannelInfo->data.usac.lsp_coeff[k], pAacDecoderChannelInfo->data.usac.lsp_coeff[k + 1], pAacDecoderChannelInfo->data.usac.aStability[k], pAacDecoderStaticChannelInfo->numLostLpdFrames, acelp_out, lFrame, pAacDecoderStaticChannelInfo->last_tcx_noise_factor); } else { FDK_ASSERT(pAacDecoderChannelInfo->data.usac.aStability[k] >= (FIXP_SGL)0); CLpd_AcelpDecode(&pAacDecoderStaticChannelInfo->acelp, i_offset, pAacDecoderChannelInfo->data.usac.lsp_coeff[k], pAacDecoderChannelInfo->data.usac.lsp_coeff[k + 1], pAacDecoderChannelInfo->data.usac.aStability[k], &pAacDecoderChannelInfo->data.usac.acelp[k], pAacDecoderStaticChannelInfo->numLostLpdFrames, last_lpc_lost, k, acelp_out, &pitch[(k * nbSubfr) + synSfd], &pit_gain[(k * nbSubfr) + synSfd], lFrame); } if (mod[k] != 4) { if (last_lpd_mode != 0 && pAacDecoderChannelInfo->data.usac .bpf_control_info) { /* FD/TCX -> ACELP transition */ /* bass post-filter past FAC area (past two (one for FD short) * subframes) */ int currentSf = synSfd + k * nbSubfr; if ((k > 0) || (pAacDecoderStaticChannelInfo->last_core_mode != FD_SHORT)) { /* TCX or FD long -> ACELP */ pitch[currentSf - 2] = pitch[currentSf - 1] = pitch[currentSf]; pit_gain[currentSf - 2] = pit_gain[currentSf - 1] = pit_gain[currentSf]; } else { /* FD short -> ACELP */ pitch[currentSf - 1] = pitch[currentSf]; pit_gain[currentSf - 1] = pit_gain[currentSf]; } } } } else { /* TCX */ int lg = lg_table[mod[k]]; int isFullBandLpd = 0; /* FAC management */ if ((last_lpd_mode == 0) || (last_lpd_mode == 4)) /* TCX TD concealment */ { C_AALLOC_SCRATCH_START(fac_buf, FIXP_DBL, 1024 / 8); /* pAacDecoderChannelInfo->data.usac.fac_data[k] == NULL means no FAC * data available. */ if (last_frame_lost == 1 || pAacDecoderChannelInfo->data.usac.fac_data[k] == NULL) { FDKmemclear(fac_buf, 1024 / 8 * sizeof(FIXP_DBL)); pAacDecoderChannelInfo->data.usac.fac_data[k] = fac_buf; pAacDecoderChannelInfo->data.usac.fac_data_e[k] = 0; } nrSamples += CLpd_FAC_Acelp2Mdct( &pAacDecoderStaticChannelInfo->IMdct, synth + nrSamples, SPEC_TCX(pAacDecoderChannelInfo->pSpectralCoefficient, k, pAacDecoderChannelInfo->granuleLength, isFullBandLpd), pAacDecoderChannelInfo->specScale + k, 1, pAacDecoderChannelInfo->data.usac.fac_data[k], pAacDecoderChannelInfo->data.usac.fac_data_e[k], pAacDecoderChannelInfo->granuleLength /* == fac_length */, lFrame - nrSamples, lg, FDKgetWindowSlope(lDiv, GetWindowShape(&pAacDecoderChannelInfo->icsInfo)), lDiv, pAacDecoderChannelInfo->data.usac.lp_coeff[k], pAacDecoderChannelInfo->data.usac.lp_coeff_exp[k], &pAacDecoderStaticChannelInfo->acelp, pAacDecoderChannelInfo->data.usac.tcx_gain[k], (last_frame_lost || !frameOk), 0 /* is not FD FAC */ , last_lpd_mode, k, pAacDecoderChannelInfo ->currAliasingSymmetry /* Note: The current aliasing symmetry for a TCX (i.e. LPD) frame must always be 0 */ ); pitch[(k * nbSubfr) + synSfd + 1] = pitch[(k * nbSubfr) + synSfd] = pitch[(k * nbSubfr) + synSfd - 1]; pit_gain[(k * nbSubfr) + synSfd + 1] = pit_gain[(k * nbSubfr) + synSfd] = pit_gain[(k * nbSubfr) + synSfd - 1]; C_AALLOC_SCRATCH_END(fac_buf, FIXP_DBL, 1024 / 8); } else { int tl = lg; int fl = lDiv; int fr = lDiv; nrSamples += imlt_block( &pAacDecoderStaticChannelInfo->IMdct, synth + nrSamples, SPEC_TCX(pAacDecoderChannelInfo->pSpectralCoefficient, k, pAacDecoderChannelInfo->granuleLength, isFullBandLpd), pAacDecoderChannelInfo->specScale + k, 1, lFrame - nrSamples, tl, FDKgetWindowSlope(fl, GetWindowShape(&pAacDecoderChannelInfo->icsInfo)), fl, FDKgetWindowSlope(fr, GetWindowShape(&pAacDecoderChannelInfo->icsInfo)), fr, pAacDecoderChannelInfo->data.usac.tcx_gain[k], pAacDecoderChannelInfo->currAliasingSymmetry ? MLT_FLAG_CURR_ALIAS_SYMMETRY : 0); } } /* remember previous mode */ last_last_lpd_mode = last_lpd_mode; last_lpd_mode = mod[k]; last_lpc_lost = (frameOk == 0) ? 1 : 0; /* Increase k to next frame */ last_k = k; k += ((mod[k] & 0x3) == 0) ? 1 : (1 << (mod[k] - 1)); } if (frameOk) { /* assume data was ok => store for concealment */ FDK_ASSERT(pAacDecoderChannelInfo->data.usac.aStability[last_k] >= (FIXP_SGL)0); pAacDecoderStaticChannelInfo->oldStability = pAacDecoderChannelInfo->data.usac.aStability[last_k]; FDKmemcpy(pAacDecoderStaticChannelInfo->lsf_adaptive_mean, pAacDecoderChannelInfo->data.usac.lsf_adaptive_mean_cand, M_LP_FILTER_ORDER * sizeof(FIXP_LPC)); } /* store past lp coeffs for next superframe (they are only valid and needed if * last_lpd_mode was tcx) */ if (last_lpd_mode > 0) { FDKmemcpy(pAacDecoderStaticChannelInfo->lp_coeff_old[0], pAacDecoderChannelInfo->data.usac.lp_coeff[nbDiv], M_LP_FILTER_ORDER * sizeof(FIXP_LPC)); pAacDecoderStaticChannelInfo->lp_coeff_old_exp[0] = pAacDecoderChannelInfo->data.usac.lp_coeff_exp[nbDiv]; FDKmemcpy(pAacDecoderStaticChannelInfo->lp_coeff_old[1], pAacDecoderChannelInfo->data.usac.lp_coeff[last_k], M_LP_FILTER_ORDER * sizeof(FIXP_LPC)); pAacDecoderStaticChannelInfo->lp_coeff_old_exp[1] = pAacDecoderChannelInfo->data.usac.lp_coeff_exp[last_k]; } FDK_ASSERT(nrSamples == lFrame); /* check whether usage of bass postfilter was de-activated in the bitstream; if yes, set pitch gain to 0 */ if (!(pAacDecoderChannelInfo->data.usac.bpf_control_info)) { if (mod[0] != 0 && (pAacDecoderStaticChannelInfo->old_bpf_control_info)) { for (int i = 2; i < nbSubfrSuperfr; i++) pit_gain[synSfd + i] = (FIXP_DBL)0; } else { for (int i = 0; i < nbSubfrSuperfr; i++) pit_gain[synSfd + i] = (FIXP_DBL)0; } } /* for bass postfilter */ for (int n = 0; n < synSfd; n++) { pAacDecoderStaticChannelInfo->old_T_pf[n] = pitch[nbSubfrSuperfr + n]; pAacDecoderStaticChannelInfo->old_gain_pf[n] = pit_gain[nbSubfrSuperfr + n]; } pAacDecoderStaticChannelInfo->old_bpf_control_info = pAacDecoderChannelInfo->data.usac.bpf_control_info; { INT lookahead = -BPF_DELAY; int copySamp = (mod[nbDiv - 1] == 0) ? (aacDelay) : (aacDelay - lFac); /* Copy enough time domain samples from MDCT to synthesis buffer as needed * by the bass postfilter */ lookahead += imdct_copy_ov_and_nr(&pAacDecoderStaticChannelInfo->IMdct, synth + nrSamples, copySamp); FDK_ASSERT(lookahead == copySamp - BPF_DELAY); FIXP_DBL *p2_synth = synth + BPF_DELAY; /* recalculate pitch gain to allow postfilering on FAC area */ for (int i = 0; i < nbSubfrSuperfr; i++) { int T = pitch[i]; FIXP_DBL gain = pit_gain[i]; if (gain > (FIXP_DBL)0) { gain = get_gain(&p2_synth[i * L_SUBFR], &p2_synth[(i * L_SUBFR) - T], L_SUBFR); pit_gain[i] = gain; } } { bass_pf_1sf_delay(p2_synth, pitch, pit_gain, lFrame, lFrame / facFB, mod[nbDiv - 1] ? (SynDelay - (lDiv / 2)) : SynDelay, pTimeData, aacOutDataHeadroom, pAacDecoderStaticChannelInfo->mem_bpf); } } Acelp_PostProcessing(synth_buf, pAacDecoderStaticChannelInfo->old_synth, pitch, pAacDecoderStaticChannelInfo->old_T_pf, lFrame, synSfd, nbSubfrSuperfr); /* Store last mode for next super frame */ { pAacDecoderStaticChannelInfo->last_core_mode = LPD; } pAacDecoderStaticChannelInfo->last_lpd_mode = last_lpd_mode; pAacDecoderStaticChannelInfo->last_last_lpd_mode = last_last_lpd_mode; pAacDecoderStaticChannelInfo->last_lpc_lost = last_lpc_lost; return error; } fdk-aac-2.0.2/libAACdec/src/usacdec_lpd.h000066400000000000000000000210171404225310700177350ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): Manuel Jander Description: USAC Linear Prediction Domain coding *******************************************************************************/ #ifndef USACDEC_LPD_H #define USACDEC_LPD_H #include "channelinfo.h" #define OPTIMIZE_AVG_PERFORMANCE /** * \brief read a lpd_channel_stream. * \param hBs a bit stream handle, where the lpd_channel_stream is located. * \param pAacDecoderChannelInfo the channel context structure for storing read * data. * \param flags bit stream syntax flags. * \return AAC_DECODER_ERROR error code. */ AAC_DECODER_ERROR CLpdChannelStream_Read( HANDLE_FDK_BITSTREAM hBs, CAacDecoderChannelInfo *pAacDecoderChannelInfo, CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, const SamplingRateInfo *pSamplingRateInfo, UINT flags); /** * \brief decode one lpd_channel_stream and render the audio output. * \param pAacDecoderChannelInfo struct holding the channel information to be * rendered. * \param pAacDecoderStaticChannelInfo struct holding the persistent channel * information to be rendered. * \param pSamplingRateInfo holds the sampling rate information * \param elFlags holds the internal decoder flags */ void CLpdChannelStream_Decode( CAacDecoderChannelInfo *pAacDecoderChannelInfo, CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, UINT flags); /** * \brief generate time domain output signal for LPD channel streams * \param pAacDecoderStaticChannelInfo * \param pAacDecoderChannelInfo * \param pTimeData pointer to output buffer * \param samplesPerFrame amount of output samples * \param pSamplingRateInfo holds the sampling rate information * \param aacOutDataHeadroom headroom of the output time signal to prevent * clipping */ AAC_DECODER_ERROR CLpd_RenderTimeSignal( CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, CAacDecoderChannelInfo *pAacDecoderChannelInfo, PCM_DEC *pTimeData, INT samplesPerFrame, SamplingRateInfo *pSamplingRateInfo, UINT frameOk, const INT aacOutDataHeadroom, UINT flags, UINT strmFlags); static inline INT CLpd_FAC_getLength(int fNotShortBlock, int fac_length_long) { if (fNotShortBlock) { return (fac_length_long); } else { return fac_length_long / 2; } } void filtLP(const FIXP_DBL *syn, PCM_DEC *syn_out, FIXP_DBL *noise, const FIXP_SGL *filt, const INT aacOutDataHeadroom, INT stop, int len); /** * \brief perform a low-frequency pitch enhancement on time domain signal * \param[in] syn pointer to time domain input signal * \param[in] synFB pointer to time domain input signal * \param[in] upsampling factor * \param[in] T_sf array with past decoded pitch period values for each subframe * \param[in] non_zero_gain_flags indicates whether pitch gains of past * subframes are zero or not, msb -> [1 BPF_DELAY subfr][7 SYN_DELAY subfr][16 * new subfr] <- lsb * \param[in] l_frame length of filtering, must be multiple of L_SUBFR * \param[in] l_next length of allowed look ahead on syn[i], i < l_frame+l_next * \param[out] synth_out pointer to time domain output signal * \param[in] headroom of the output time signal to prevent clipping * \param[in,out] mem_bpf pointer to filter memory (L_FILT+L_SUBFR) */ void bass_pf_1sf_delay(FIXP_DBL syn[], const INT T_sf[], FIXP_DBL *pit_gain, const int frame_length, const INT l_frame, const INT l_next, PCM_DEC *synth_out, const INT aacOutDataHeadroom, FIXP_DBL mem_bpf[]); /** * \brief random sign generator for FD and TCX noise filling * \param[in,out] seed pointer to random seed * \return if return value is zero use positive sign * \Note: This code is also implemented as a copy in block.cpp, grep for * "UsacRandomSign" */ FDK_INLINE int UsacRandomSign(ULONG *seed) { *seed = (ULONG)((UINT64)(*seed) * 69069 + 5); return (int)((*seed) & 0x10000); } void CFdp_Reset(CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo); #endif /* USACDEC_LPD_H */ fdk-aac-2.0.2/libAACdec/src/usacdec_rom.cpp000066400000000000000000003166231404225310700203200ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): M. Jander Description: *******************************************************************************/ #include "usacdec_rom.h" #define NB_SPHERE 32 #define NB_LEADER 37 #define NB_LDSIGN 226 #define NB_LDQ3 9 #define NB_LDQ4 28 /* For bass post filter */ #define FL2FXCONST_SGL_FILT(a) FL2FXCONST_SGL(a*(1 << SF_FILT_LP)) #define SF_FILT_LP 1 /* table of factorial */ const UINT fdk_dec_tab_factorial[8] = {5040, 720, 120, 24, 6, 2, 1, 1}; /* Da - Absolute leaders */ const UCHAR fdk_dec_Da[NB_LEADER][8] = { {1, 1, 1, 1, 1, 1, 1, 1}, {2, 2, 0, 0, 0, 0, 0, 0}, {2, 2, 2, 2, 0, 0, 0, 0}, {3, 1, 1, 1, 1, 1, 1, 1}, {4, 0, 0, 0, 0, 0, 0, 0}, {2, 2, 2, 2, 2, 2, 0, 0}, {3, 3, 1, 1, 1, 1, 1, 1}, {4, 2, 2, 0, 0, 0, 0, 0}, {2, 2, 2, 2, 2, 2, 2, 2}, {3, 3, 3, 1, 1, 1, 1, 1}, {4, 2, 2, 2, 2, 0, 0, 0}, {4, 4, 0, 0, 0, 0, 0, 0}, {5, 1, 1, 1, 1, 1, 1, 1}, {3, 3, 3, 3, 1, 1, 1, 1}, {4, 2, 2, 2, 2, 2, 2, 0}, {4, 4, 2, 2, 0, 0, 0, 0}, {5, 3, 1, 1, 1, 1, 1, 1}, {6, 2, 0, 0, 0, 0, 0, 0}, {4, 4, 4, 0, 0, 0, 0, 0}, {6, 2, 2, 2, 0, 0, 0, 0}, {6, 4, 2, 0, 0, 0, 0, 0}, {7, 1, 1, 1, 1, 1, 1, 1}, {8, 0, 0, 0, 0, 0, 0, 0}, {6, 6, 0, 0, 0, 0, 0, 0}, {8, 2, 2, 0, 0, 0, 0, 0}, {8, 4, 0, 0, 0, 0, 0, 0}, {9, 1, 1, 1, 1, 1, 1, 1}, {10, 2, 0, 0, 0, 0, 0, 0}, {8, 8, 0, 0, 0, 0, 0, 0}, {10, 6, 0, 0, 0, 0, 0, 0}, {12, 0, 0, 0, 0, 0, 0, 0}, {12, 4, 0, 0, 0, 0, 0, 0}, {10, 10, 0, 0, 0, 0, 0, 0}, {14, 2, 0, 0, 0, 0, 0, 0}, {12, 8, 0, 0, 0, 0, 0, 0}, {16, 0, 0, 0, 0, 0, 0, 0}, {20, 0, 0, 0, 0, 0, 0, 0}}; /* Ds - Sign codes of all signed leaders */ const UCHAR fdk_dec_Ds[NB_LDSIGN] = { 0, 3, 15, 63, 255, 0, 64, 192, 0, 16, 48, 112, 240, 1, 7, 31, 127, 128, 131, 143, 191, 0, 128, 0, 4, 12, 28, 60, 124, 252, 0, 3, 15, 63, 65, 71, 95, 192, 195, 207, 255, 0, 32, 96, 128, 160, 224, 0, 1, 3, 7, 15, 31, 63, 127, 255, 1, 7, 31, 32, 35, 47, 97, 103, 127, 224, 227, 239, 0, 8, 24, 56, 120, 128, 136, 152, 184, 248, 0, 64, 192, 0, 3, 15, 63, 129, 135, 159, 255, 0, 3, 15, 17, 23, 48, 51, 63, 113, 119, 240, 243, 255, 0, 2, 6, 14, 30, 62, 126, 128, 130, 134, 142, 158, 190, 254, 0, 16, 48, 64, 80, 112, 192, 208, 240, 1, 7, 31, 64, 67, 79, 127, 128, 131, 143, 191, 193, 199, 223, 0, 64, 128, 192, 0, 32, 96, 224, 0, 16, 48, 112, 128, 144, 176, 240, 0, 32, 64, 96, 128, 160, 192, 224, 1, 7, 31, 127, 128, 131, 143, 191, 0, 128, 0, 64, 192, 0, 32, 96, 128, 160, 224, 0, 64, 128, 192, 0, 3, 15, 63, 129, 135, 159, 255, 0, 64, 128, 192, 0, 64, 192, 0, 64, 128, 192, 0, 128, 0, 64, 128, 192, 0, 64, 192, 0, 64, 128, 192, 0, 64, 128, 192, 0, 128, 0, 128}; /* Ns - Number of signed leader associated to a given absolute leader */ const UCHAR fdk_dec_Ns[NB_LEADER] = { 5, 3, 5, 8, 2, 7, 11, 6, 9, 12, 10, 3, 8, 13, 14, 9, 14, 4, 4, 8, 8, 8, 2, 3, 6, 4, 8, 4, 3, 4, 2, 4, 3, 4, 4, 2, 2}; /* Ia - Position of the first signed leader associated to an absolute leader */ const UCHAR fdk_dec_Ia[NB_LEADER] = { 0, 5, 8, 13, 21, 23, 30, 41, 47, 56, 68, 78, 81, 89, 102, 116, 125, 139, 143, 147, 155, 163, 171, 173, 176, 182, 186, 194, 198, 201, 205, 207, 211, 214, 218, 222, 224}; /* Is - Cardinalite offset of signed leaders */ const USHORT fdk_dec_Is[NB_LDSIGN] = { 0, 1, 29, 99, 127, 128, 156, 212, 256, 326, 606, 1026, 1306, 1376, 1432, 1712, 1880, 1888, 1896, 2064, 2344, 240, 248, 0, 28, 196, 616, 1176, 1596, 1764, 1792, 1820, 2240, 2660, 2688, 3024, 4144, 4480, 4508, 4928, 5348, 2400, 2568, 2904, 3072, 3240, 3576, 5376, 5377, 5385, 5413, 5469, 5539, 5595, 5623, 5631, 5632, 5912, 6472, 6528, 6696, 8376, 9216, 10056, 11736, 11904, 11960, 12520, 12800, 13080, 14200, 15880, 17000, 17280, 17560, 18680, 20360, 21480, 3744, 3772, 3828, 21760, 21768, 21936, 22216, 22272, 22328, 22608, 22776, 22784, 22854, 23274, 23344, 24464, 25584, 26004, 28524, 28944, 30064, 31184, 31254, 31674, 31744, 31800, 32136, 32976, 34096, 34936, 35272, 35328, 35384, 35720, 36560, 37680, 38520, 38856, 38912, 39332, 40172, 40592, 41432, 43112, 43952, 44372, 45212, 45632, 45968, 47088, 47424, 47480, 48320, 49160, 49216, 49272, 50112, 50952, 51008, 51344, 52464, 3856, 3912, 3968, 4024, 52800, 52856, 53024, 53192, 53248, 53528, 54368, 55208, 55488, 55768, 56608, 57448, 57728, 58064, 58400, 58736, 59072, 59408, 59744, 60080, 60416, 60472, 60752, 60920, 60928, 60936, 61104, 61384, 4080, 4088, 61440, 61468, 61524, 61552, 61720, 62056, 62224, 62392, 62728, 62896, 62952, 63008, 63064, 63120, 63128, 63296, 63576, 63632, 63688, 63968, 64136, 64144, 64200, 64256, 64312, 64368, 64396, 64452, 64480, 64536, 64592, 64648, 64704, 64712, 64720, 64776, 64832, 64888, 64944, 64972, 65028, 65056, 65112, 65168, 65224, 65280, 65336, 65392, 65448, 65504, 65512, 65520, 65528}; /* A3 - Number of the absolute leaders in codebooks Q2 and Q3 */ const UCHAR fdk_dec_A3[NB_LDQ3] = {0, 1, 4, 2, 3, 7, 11, 17, 22}; /* A4 - Number of the absolute leaders in codebook Q4 */ const UCHAR fdk_dec_A4[NB_LDQ4] = {5, 6, 8, 9, 10, 12, 13, 14, 15, 16, 18, 19, 20, 21, 23, 24, 25, 26, 27, 28, 29, 30, 31, 32, 33, 34, 35, 36}; /* I3 - Cardinality offsets for absolute leaders in Q3 */ const USHORT fdk_dec_I3[NB_LDQ3] = {0, 128, 240, 256, 1376, 2400, 3744, 3856, 4080}; /* I4 - Cardinality offset for absolute leaders in Q4 */ const USHORT fdk_dec_I4[NB_LDQ4] = { 0, 1792, 5376, 5632, 12800, 21760, 22784, 31744, 38912, 45632, 52800, 53248, 57728, 60416, 61440, 61552, 62896, 63120, 64144, 64368, 64480, 64704, 64720, 64944, 65056, 65280, 65504, 65520}; /* Initial ISF memory for concealment case */ #define LSFI(x) ((x) << (FRACT_BITS - LSF_SCALE - 1)) const FIXP_LPC fdk_dec_lsf_init[16] = {1506, 3012, 4518, 6024, 7529, 9035, 10541, 12047, 13553, 15059, 16565, 18071, 19576, 21082, 22588, 24094}; /* dico_lsf_abs_8b is scaled by 1/(1<<13) */ #define DICO(x) FX_DBL2FXCONST_LPC(x >> (LSF_SCALE - 13)) const FIXP_LPC fdk_dec_dico_lsf_abs_8b[] = { DICO(0x05e57fe8), DICO(0x0ac00810), DICO(0x11ed8500), DICO(0x16d42ce0), DICO(0x1beb1e20), DICO(0x217eaf40), DICO(0x2768c740), DICO(0x2d26f600), DICO(0x32fe68c0), DICO(0x38b1d980), DICO(0x3e95bd80), DICO(0x446dab00), DICO(0x4abfd280), DICO(0x5094b380), DICO(0x56ccb800), DICO(0x5c9aba00), DICO(0x09660ca0), DICO(0x10ab4c00), DICO(0x15a16f20), DICO(0x19d3c780), DICO(0x1ee99060), DICO(0x241d1200), DICO(0x29c83700), DICO(0x2f098f00), DICO(0x34803fc0), DICO(0x3a37bc00), DICO(0x3ff55580), DICO(0x45da9280), DICO(0x4bec6700), DICO(0x5169e300), DICO(0x57797c80), DICO(0x5d09ae80), DICO(0x08a203b0), DICO(0x0d6ed1a0), DICO(0x152ccf20), DICO(0x19639dc0), DICO(0x1d7e3e60), DICO(0x21f4a7c0), DICO(0x27b2f8c0), DICO(0x2dbb4480), DICO(0x33ecde80), DICO(0x3982e100), DICO(0x3ea16100), DICO(0x43ab6080), DICO(0x49534a80), DICO(0x4ea7e100), DICO(0x550d6300), DICO(0x5bcdcc80), DICO(0x072dd048), DICO(0x0c654690), DICO(0x1436e940), DICO(0x19459680), DICO(0x1e0041c0), DICO(0x2240dc80), DICO(0x26de4040), DICO(0x2b509b00), DICO(0x309d8780), DICO(0x36151180), DICO(0x3c6c1200), DICO(0x42df6b80), DICO(0x4a144400), DICO(0x50541280), DICO(0x56c34b80), DICO(0x5cb6c600), DICO(0x051fef00), DICO(0x06b9fb48), DICO(0x0b4f9cc0), DICO(0x17e27800), DICO(0x1b8c7340), DICO(0x1f772ca0), DICO(0x2478dc80), DICO(0x28242240), DICO(0x2f27c640), DICO(0x33b03e80), DICO(0x381f20c0), DICO(0x3c662c00), DICO(0x49565080), DICO(0x529b0f00), DICO(0x583ed080), DICO(0x5d8cec00), DICO(0x071c4d18), DICO(0x097853b0), DICO(0x0f0f0690), DICO(0x157bf980), DICO(0x1801f580), DICO(0x1deb0c20), DICO(0x2523da40), DICO(0x28534600), DICO(0x2eb499c0), DICO(0x32eb5ac0), DICO(0x36749580), DICO(0x3a748200), DICO(0x4325f700), DICO(0x515d8300), DICO(0x58a18700), DICO(0x5d722100), DICO(0x06cbcd88), DICO(0x08bb6740), DICO(0x0dead310), DICO(0x152f0cc0), DICO(0x18427640), DICO(0x1d9f2f20), DICO(0x22ba3b40), DICO(0x271a6e80), DICO(0x2c677ec0), DICO(0x31061b00), DICO(0x349eef40), DICO(0x3c531b80), DICO(0x4aed0580), DICO(0x4f8bbf80), DICO(0x54b74980), DICO(0x5bc9b700), DICO(0x046410c8), DICO(0x06522ab0), DICO(0x0b6528c0), DICO(0x0f94bd90), DICO(0x1a8f8b80), DICO(0x1ea57820), DICO(0x233ee180), DICO(0x27b3acc0), DICO(0x2bd1d240), DICO(0x2fc4bcc0), DICO(0x3a98ea40), DICO(0x43d3f500), DICO(0x49b37580), DICO(0x4e2afd00), DICO(0x55953300), DICO(0x5d36f600), DICO(0x05d0f6c8), DICO(0x07e56d90), DICO(0x0be98080), DICO(0x0f956f30), DICO(0x1259b3c0), DICO(0x1f08b240), DICO(0x25008c00), DICO(0x2900b180), DICO(0x31ea6f00), DICO(0x352d1e00), DICO(0x3c970c80), DICO(0x45271200), DICO(0x4b632280), DICO(0x5098a480), DICO(0x5672fc80), DICO(0x5c163180), DICO(0x05bd81a0), DICO(0x07d4b8f0), DICO(0x0ce224b0), DICO(0x110abe20), DICO(0x13dfeac0), DICO(0x17dedae0), DICO(0x2535c0c0), DICO(0x2a19da80), DICO(0x2e5224c0), DICO(0x38ddeec0), DICO(0x3da99d80), DICO(0x42799100), DICO(0x48973b00), DICO(0x4ea62880), DICO(0x53f77e80), DICO(0x5bd9c100), DICO(0x0395cd50), DICO(0x058244b8), DICO(0x0af45520), DICO(0x1329cea0), DICO(0x1a3970c0), DICO(0x1d9f2e00), DICO(0x21704400), DICO(0x277a34c0), DICO(0x30215b40), DICO(0x33875040), DICO(0x3c159840), DICO(0x452fea00), DICO(0x4981d200), DICO(0x4e15a980), DICO(0x54e84780), DICO(0x5c79ea00), DICO(0x05413b98), DICO(0x08132a80), DICO(0x0dc7f050), DICO(0x13e25460), DICO(0x1784bf80), DICO(0x1d630200), DICO(0x238bc880), DICO(0x28cc0880), DICO(0x30da1a40), DICO(0x391e2200), DICO(0x415d8d00), DICO(0x48f13280), DICO(0x4e300300), DICO(0x52e56580), DICO(0x5849fe80), DICO(0x5cdef400), DICO(0x04a058c8), DICO(0x07569b88), DICO(0x0ef26610), DICO(0x13208140), DICO(0x168c0500), DICO(0x1afec080), DICO(0x22a0abc0), DICO(0x2a057880), DICO(0x2fd1c840), DICO(0x3703c680), DICO(0x3d326b80), DICO(0x43df2e80), DICO(0x4a6f9000), DICO(0x50900d80), DICO(0x56c73f00), DICO(0x5cc3da80), DICO(0x065c99e8), DICO(0x09060c50), DICO(0x0d1ef1c0), DICO(0x16bd9020), DICO(0x1a04dae0), DICO(0x1e3c0580), DICO(0x25783700), DICO(0x29710ac0), DICO(0x309cbb80), DICO(0x36c66280), DICO(0x3adb0580), DICO(0x41b37e00), DICO(0x496ca700), DICO(0x4dab7600), DICO(0x52be6280), DICO(0x58fec480), DICO(0x04640880), DICO(0x05a75ab8), DICO(0x0edba410), DICO(0x16e076a0), DICO(0x198acec0), DICO(0x1eb5fae0), DICO(0x228c9000), DICO(0x29986c00), DICO(0x2c780c80), DICO(0x38078dc0), DICO(0x3f42dc00), DICO(0x441ba900), DICO(0x492f8080), DICO(0x4ed85d00), DICO(0x54605800), DICO(0x5d106a80), DICO(0x045cb970), DICO(0x0627a828), DICO(0x0db35290), DICO(0x1778f780), DICO(0x1a243c60), DICO(0x23c2dd40), DICO(0x27c57840), DICO(0x2f53cd80), DICO(0x36f65600), DICO(0x3bc1b2c0), DICO(0x40c36500), DICO(0x46074180), DICO(0x4b551b80), DICO(0x50a99700), DICO(0x569b6c80), DICO(0x5ca25780), DICO(0x05ef2828), DICO(0x07d3adf8), DICO(0x0b5416d0), DICO(0x0f9adb70), DICO(0x126e7360), DICO(0x1baff460), DICO(0x2b5decc0), DICO(0x31036200), DICO(0x34ca7500), DICO(0x39681340), DICO(0x3da97100), DICO(0x4161ee00), DICO(0x46a62e80), DICO(0x4d1b9380), DICO(0x530e0300), DICO(0x59ff0480), DICO(0x04f5bc50), DICO(0x06e90d18), DICO(0x0c2af480), DICO(0x123f7400), DICO(0x1530a160), DICO(0x18aa3dc0), DICO(0x1cc0a240), DICO(0x2cdb02c0), DICO(0x32909a00), DICO(0x36bae640), DICO(0x3c917a80), DICO(0x40121900), DICO(0x48a90d80), DICO(0x51ccc180), DICO(0x5884ea00), DICO(0x5dbc4280), DICO(0x05791410), DICO(0x07b0dd80), DICO(0x0bec4190), DICO(0x13c30520), DICO(0x17ac1900), DICO(0x1b6f1d00), DICO(0x26e54f40), DICO(0x2d4a8040), DICO(0x311c6840), DICO(0x38ec4180), DICO(0x3f0c4340), DICO(0x427c5b00), DICO(0x4886e480), DICO(0x504a0b00), DICO(0x56d48700), DICO(0x5c80f600), DICO(0x04b58880), DICO(0x0743f0d8), DICO(0x0be95e20), DICO(0x0fd0d9b0), DICO(0x1c2e11a0), DICO(0x2241af80), DICO(0x296e83c0), DICO(0x2f16adc0), DICO(0x32cd6fc0), DICO(0x374ddec0), DICO(0x3da95f80), DICO(0x45d56c80), DICO(0x4c6afa80), DICO(0x5141f380), DICO(0x5616b380), DICO(0x5c58f580), DICO(0x03f4b368), DICO(0x05939890), DICO(0x09d95480), DICO(0x122cac60), DICO(0x17e27e00), DICO(0x1f9dc680), DICO(0x26e26680), DICO(0x2ae64040), DICO(0x2dd6cf40), DICO(0x3295c400), DICO(0x3e23b400), DICO(0x44fd0380), DICO(0x4ad7a700), DICO(0x51295e80), DICO(0x594a9400), DICO(0x5e41aa00), DICO(0x0424b9d8), DICO(0x05b30508), DICO(0x09380f20), DICO(0x0c9509c0), DICO(0x18730860), DICO(0x219a9d40), DICO(0x24f699c0), DICO(0x289b2680), DICO(0x2cb62240), DICO(0x36e88180), DICO(0x3e968800), DICO(0x48053c80), DICO(0x4d6dca80), DICO(0x51d9a580), DICO(0x563e5a80), DICO(0x5c0b2b80), DICO(0x03456ae8), DICO(0x04e49948), DICO(0x07dd0e88), DICO(0x0ed5cd30), DICO(0x1b06e980), DICO(0x1de2b9c0), DICO(0x21160540), DICO(0x270a8240), DICO(0x3352a280), DICO(0x3b8b6c00), DICO(0x40241400), DICO(0x43f60f80), DICO(0x4a897900), DICO(0x51692a00), DICO(0x57449d00), DICO(0x5d497480), DICO(0x04b94290), DICO(0x067e99d0), DICO(0x0ab06840), DICO(0x0e697070), DICO(0x1745c460), DICO(0x22ee8040), DICO(0x2647e8c0), DICO(0x2bc2c680), DICO(0x2fd57d00), DICO(0x37186680), DICO(0x3d074500), DICO(0x412b2800), DICO(0x4579af00), DICO(0x4caff980), DICO(0x557add00), DICO(0x5c6ae780), DICO(0x0423a090), DICO(0x05b9bca0), DICO(0x091b45d0), DICO(0x0c5b6d60), DICO(0x194dd1c0), DICO(0x1fc85020), DICO(0x2486b080), DICO(0x2920af80), DICO(0x2dd4f140), DICO(0x3598be40), DICO(0x3b9c1440), DICO(0x42d19280), DICO(0x4a314280), DICO(0x50b00a00), DICO(0x56c55400), DICO(0x5d5ba300), DICO(0x03e68b28), DICO(0x05a7b190), DICO(0x0917f000), DICO(0x0d247050), DICO(0x19e637a0), DICO(0x2221a540), DICO(0x2777e540), DICO(0x2c103380), DICO(0x30c2e040), DICO(0x389f1240), DICO(0x3f4a2c80), DICO(0x454a4c00), DICO(0x4b0ab680), DICO(0x50cf6000), DICO(0x571c0700), DICO(0x5d2ef600), DICO(0x04886f18), DICO(0x065103e8), DICO(0x0a607d40), DICO(0x0db91960), DICO(0x13546f20), DICO(0x22f5e200), DICO(0x27064240), DICO(0x2e371d40), DICO(0x33659240), DICO(0x38aa1c40), DICO(0x417bb280), DICO(0x47ca9480), DICO(0x4dd6fb80), DICO(0x528e3480), DICO(0x57c49d80), DICO(0x5cc98100), DICO(0x02db2370), DICO(0x04398848), DICO(0x07a8da38), DICO(0x10b90280), DICO(0x1a2a4a20), DICO(0x20b1f640), DICO(0x277096c0), DICO(0x2dc568c0), DICO(0x341b33c0), DICO(0x3a000640), DICO(0x40152880), DICO(0x45eeee00), DICO(0x4c08c480), DICO(0x51bf0600), DICO(0x5799a180), DICO(0x5d23db80), DICO(0x047b1498), DICO(0x06089848), DICO(0x0905af20), DICO(0x0bf13c20), DICO(0x11fcf620), DICO(0x1f79cd00), DICO(0x257f6b40), DICO(0x2cfc2600), DICO(0x31610040), DICO(0x35ea8280), DICO(0x3c774bc0), DICO(0x44417280), DICO(0x4b432500), DICO(0x510e9480), DICO(0x56f2e480), DICO(0x5d282780), DICO(0x02cfd0b0), DICO(0x042845d8), DICO(0x0a1fa610), DICO(0x15911fc0), DICO(0x1bc07f00), DICO(0x2281d640), DICO(0x287abcc0), DICO(0x2ec6b400), DICO(0x34a0d040), DICO(0x3aa4dcc0), DICO(0x4074d980), DICO(0x46726b80), DICO(0x4c3bf900), DICO(0x52055100), DICO(0x57b20500), DICO(0x5d34da80), DICO(0x04d4f768), DICO(0x06cad828), DICO(0x0b52a540), DICO(0x0ea224e0), DICO(0x13c3f460), DICO(0x23808900), DICO(0x27d1cec0), DICO(0x2d6051c0), DICO(0x33c5ff00), DICO(0x37ef2440), DICO(0x3d2a5300), DICO(0x43266000), DICO(0x4a53a100), DICO(0x50acce80), DICO(0x57612100), DICO(0x5cdee380), DICO(0x04039a88), DICO(0x0626dcb0), DICO(0x0c059620), DICO(0x12c3db20), DICO(0x1bb9eb40), DICO(0x240fda00), DICO(0x2baab840), DICO(0x3177c5c0), DICO(0x36cf2e40), DICO(0x3c025100), DICO(0x40bb8d00), DICO(0x45960800), DICO(0x4adaca00), DICO(0x505a7300), DICO(0x566a6400), DICO(0x5c8ce000), DICO(0x062891e8), DICO(0x09680810), DICO(0x0e9a11b0), DICO(0x1523e320), DICO(0x1c57db00), DICO(0x21f22c80), DICO(0x28aeeb00), DICO(0x2e4fd600), DICO(0x341cf000), DICO(0x3a5034c0), DICO(0x40600f80), DICO(0x461fde00), DICO(0x4c368480), DICO(0x51dbbc00), DICO(0x57709780), DICO(0x5cce9880), DICO(0x05d41f70), DICO(0x0a65bb30), DICO(0x132ddfa0), DICO(0x17d26820), DICO(0x1e6d8380), DICO(0x24e68dc0), DICO(0x2b68c4c0), DICO(0x30fa2880), DICO(0x361998c0), DICO(0x3aa1d640), DICO(0x3f942400), DICO(0x44d11680), DICO(0x4ab8e580), DICO(0x50643b80), DICO(0x5697fe00), DICO(0x5cb3a780), DICO(0x0707fa10), DICO(0x0cb8beb0), DICO(0x15011d20), DICO(0x1a4ad300), DICO(0x20997080), DICO(0x26dbe240), DICO(0x2d907880), DICO(0x3307a3c0), DICO(0x38819740), DICO(0x3d3e89c0), DICO(0x41ea2300), DICO(0x469ce200), DICO(0x4be61680), DICO(0x51261b80), DICO(0x5716ef80), DICO(0x5cba2900), DICO(0x084dc830), DICO(0x0f16f610), DICO(0x16ca2420), DICO(0x1bb58380), DICO(0x22f00f00), DICO(0x296ba4c0), DICO(0x306d2600), DICO(0x362ca080), DICO(0x3b86d280), DICO(0x3ffa96c0), DICO(0x446a5300), DICO(0x48d0fd00), DICO(0x4d8a0800), DICO(0x525bf200), DICO(0x57f5aa00), DICO(0x5d569480), DICO(0x08d664f0), DICO(0x110c8520), DICO(0x1865fa40), DICO(0x1efe3160), DICO(0x26f38740), DICO(0x2d4608c0), DICO(0x32862500), DICO(0x374f8840), DICO(0x3bfa9900), DICO(0x3ff5c8c0), DICO(0x4450c500), DICO(0x4918e680), DICO(0x4e1d0f00), DICO(0x53342600), DICO(0x58a38e00), DICO(0x5dbbff00), DICO(0x09143fd0), DICO(0x0f401c30), DICO(0x169c1ee0), DICO(0x1bcfb280), DICO(0x2190dd00), DICO(0x27bf56c0), DICO(0x2e8e0640), DICO(0x34b67080), DICO(0x3b534dc0), DICO(0x41134c00), DICO(0x467a3280), DICO(0x4bd63600), DICO(0x50de8700), DICO(0x55657580), DICO(0x5a0cef00), DICO(0x5e8aa200), DICO(0x06b5d860), DICO(0x0c8a5000), DICO(0x13343620), DICO(0x17a2abe0), DICO(0x1caf7340), DICO(0x22a3f740), DICO(0x29059980), DICO(0x2ecff880), DICO(0x34ce0f00), 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DICO(0x12e05320), DICO(0x155997e0), DICO(0x195df540), DICO(0x1c1b82c0), DICO(0x22c29400), DICO(0x307edec0), DICO(0x34e829c0), DICO(0x3b1b5040), DICO(0x434de400), DICO(0x496b9900), DICO(0x4ff88080), DICO(0x5604c480), DICO(0x5c84b080), DICO(0x03679fa4), DICO(0x050d4a20), DICO(0x09feec10), DICO(0x100a21e0), DICO(0x1483b260), DICO(0x1d76c780), DICO(0x24e75c80), DICO(0x2bd62880), DICO(0x32350a40), DICO(0x38be01c0), DICO(0x3f05a280), DICO(0x45295e80), DICO(0x4b554800), DICO(0x51618880), DICO(0x575a1500), DICO(0x5d109d80), DICO(0x034ffd08), DICO(0x04dccea8), DICO(0x07e289b8), DICO(0x0d6950d0), DICO(0x189acec0), DICO(0x1e3bf240), DICO(0x2535aa40), DICO(0x2b88d140), DICO(0x329bbb00), DICO(0x386c3500), DICO(0x3e950800), DICO(0x44c43f00), DICO(0x4b089780), DICO(0x51223280), DICO(0x574c9780), DICO(0x5d366400), DICO(0x036d8db8), DICO(0x04fd88b8), DICO(0x09700a70), DICO(0x116b73c0), DICO(0x17258c40), DICO(0x1fd03000), DICO(0x23fdf400), DICO(0x28e08dc0), DICO(0x32d688c0), DICO(0x37920b00), DICO(0x3daaa080), DICO(0x46a16c00), DICO(0x4f8c3100), DICO(0x54a13700), DICO(0x59d24b80), DICO(0x5cea4d80), DICO(0x05797c88), DICO(0x080dc8f0), DICO(0x0bd21520), DICO(0x1095b540), DICO(0x138fd400), DICO(0x1aed19c0), DICO(0x29fead00), DICO(0x2f70cec0), DICO(0x3327df00), DICO(0x3a812d00), DICO(0x400a8380), DICO(0x449daa00), DICO(0x4cd6b600), DICO(0x51f12400), DICO(0x56bdfd80), DICO(0x5be0a100), DICO(0x04de6d78), DICO(0x072aed40), DICO(0x0c6fc460), DICO(0x0f260220), DICO(0x179d00c0), DICO(0x1e8244e0), DICO(0x23867240), DICO(0x2baf7680), DICO(0x2fa6d240), DICO(0x37e83d40), DICO(0x3d1cbfc0), DICO(0x439d0a00), DICO(0x49fd3e00), DICO(0x50095e80), DICO(0x559f2080), DICO(0x5b889a00), DICO(0x042f5c10), DICO(0x061ab3e8), DICO(0x0dd8f160), DICO(0x126e0b40), DICO(0x16ea9040), DICO(0x20a31700), DICO(0x24608140), DICO(0x2ec65080), DICO(0x334e1a40), DICO(0x38252100), DICO(0x3fff0900), DICO(0x46519a80), DICO(0x4c5e0d80), DICO(0x518c5800), DICO(0x56c5b080), DICO(0x5c6ebb80), DICO(0x045ce230), DICO(0x067547b0), DICO(0x0a21c670), DICO(0x1408b820), DICO(0x1c0505c0), DICO(0x1e806c80), DICO(0x26d3c640), DICO(0x2ca573c0), DICO(0x3014d880), DICO(0x377108c0), DICO(0x3f35cf80), DICO(0x43566100), DICO(0x4a612900), DICO(0x5160c780), DICO(0x57d3e300), DICO(0x5d414b80), DICO(0x04c8eda8), DICO(0x06a32320), DICO(0x09ab2590), DICO(0x14230760), DICO(0x20fb23c0), DICO(0x24aa2880), DICO(0x285a2580), DICO(0x2c464ac0), DICO(0x30098280), DICO(0x36232780), DICO(0x3e428d40), DICO(0x42982280), DICO(0x47e60d80), DICO(0x4e1ecc00), DICO(0x55eb9200), DICO(0x5c81ed00), DICO(0x040d0b90), DICO(0x0586c5c0), DICO(0x0bea2190), DICO(0x1dc8dd60), DICO(0x20a07c40), DICO(0x2437e580), DICO(0x27ca5fc0), DICO(0x2d017980), DICO(0x34100040), DICO(0x38d3afc0), DICO(0x3da9b700), DICO(0x42082480), DICO(0x46586f00), DICO(0x4e3c3d80), DICO(0x55e1ee00), DICO(0x5c938d00), DICO(0x03d18c64), DICO(0x05941d60), DICO(0x116b2560), DICO(0x19cc8820), DICO(0x1ce930c0), DICO(0x22626080), DICO(0x26d2cf80), DICO(0x2ce2c980), DICO(0x305361c0), DICO(0x34b65900), DICO(0x39ee5b40), DICO(0x41508400), DICO(0x47ee1e80), DICO(0x50311180), DICO(0x56cb0c00), DICO(0x5d561680), DICO(0x0af22cf0), DICO(0x154e1c60), DICO(0x1b44ff60), DICO(0x2087d0c0), DICO(0x252f7380), DICO(0x28fe66c0), DICO(0x2d0e9800), DICO(0x30f7ee00), DICO(0x3606d640), DICO(0x3bac0a40), DICO(0x417c3700), DICO(0x470c2900), DICO(0x4cc20d00), DICO(0x51e55e80), DICO(0x576bc200), DICO(0x5caa7080), DICO(0x043314e0), DICO(0x05cb47b0), DICO(0x0985df20), DICO(0x185a22c0), DICO(0x1ea68300), DICO(0x21af9100), DICO(0x25cdcb40), DICO(0x297e0a80), DICO(0x3142ac80), DICO(0x358bb040), DICO(0x3a1345c0), DICO(0x402ca580), DICO(0x4d964780), DICO(0x5420cf80), DICO(0x592adf80), DICO(0x5dfe7a80), DICO(0x04c70e20), DICO(0x062fd6f0), DICO(0x113c60a0), DICO(0x159e9900), DICO(0x1933e5a0), DICO(0x1decffa0), DICO(0x22373400), DICO(0x27ed7180), DICO(0x2a8349c0), DICO(0x2f78f940), DICO(0x3d5c8540), DICO(0x429c3500), DICO(0x4936e300), DICO(0x4de08d00), DICO(0x52627700), DICO(0x5d822680), DICO(0x04982770), DICO(0x06ab1ad8), DICO(0x0cba1090), DICO(0x10839d60), DICO(0x1acd6a60), DICO(0x1f554560), DICO(0x2431c1c0), DICO(0x295db800), DICO(0x2d374c00), DICO(0x31f2fd80), DICO(0x3a200480), DICO(0x416bea80), DICO(0x4ba1ce00), DICO(0x52a88000), DICO(0x58d0db00), DICO(0x5d3e5800), DICO(0x059e0e70), DICO(0x08166ba0), DICO(0x0c9df590), DICO(0x11de5f40), DICO(0x14c08ea0), DICO(0x1c5ad6e0), DICO(0x24178ec0), DICO(0x28eb7640), DICO(0x31626280), DICO(0x35d43100), DICO(0x3cad10c0), DICO(0x422e1480), DICO(0x49baee00), DICO(0x53dd7180), DICO(0x5a17bb80), DICO(0x5e4bb180), DICO(0x03c64d00), DICO(0x05de0b28), DICO(0x0ccb82b0), DICO(0x144b1c00), DICO(0x19e39ec0), DICO(0x21e795c0), DICO(0x28149380), DICO(0x2f1ff7c0), DICO(0x35b7c900), DICO(0x3c5f4800), DICO(0x42799c00), DICO(0x48746180), DICO(0x4e8f4d00), DICO(0x53b98380), DICO(0x58d60000), DICO(0x5d71a000), DICO(0x050a2990), DICO(0x071bdb88), DICO(0x0f0d76b0), DICO(0x17a78de0), DICO(0x1aa20720), DICO(0x22eea3c0), DICO(0x276e9640), DICO(0x2ecc4d80), DICO(0x335f7900), DICO(0x37838640), DICO(0x3c81be80), DICO(0x41a4e400), DICO(0x476fa280), DICO(0x4f2ab780), DICO(0x56e87b80), DICO(0x5cbf2900), DICO(0x06724c98), DICO(0x099f5e20), DICO(0x119bcec0), DICO(0x16be11a0), DICO(0x19d53b00), DICO(0x1eb51360), DICO(0x2302c300), DICO(0x29c20d40), DICO(0x30dd5200), DICO(0x36576a80), DICO(0x3e3e7540), DICO(0x45aa9f80), DICO(0x4c833300), DICO(0x51d7ed00), DICO(0x57a4a380), DICO(0x5cce9100), DICO(0x05588b60), DICO(0x075a70a0), DICO(0x0ef96e30), DICO(0x12ac1100), DICO(0x1649dde0), DICO(0x1a50cdc0), DICO(0x22486140), DICO(0x27c7c800), DICO(0x2e08bd80), DICO(0x355d20c0), DICO(0x3925a9c0), DICO(0x44e2b480), DICO(0x4fa4e680), DICO(0x5470e180), DICO(0x595fee80), DICO(0x5d672f00), DICO(0x04a781c0), DICO(0x060a89f0), DICO(0x111f56a0), DICO(0x16b93be0), DICO(0x19ce9120), DICO(0x1e1fda60), DICO(0x2259f880), DICO(0x285cad80), DICO(0x2b140ec0), DICO(0x2f069940), DICO(0x33bbce40), DICO(0x3dd3e6c0), DICO(0x49154880), DICO(0x5008e380), DICO(0x568c4680), DICO(0x5da43780), DICO(0x053ade38), DICO(0x0708a200), DICO(0x0bad43d0), DICO(0x1860cf40), DICO(0x1c40dbe0), DICO(0x1f5a0120), DICO(0x25bcf7c0), DICO(0x29e0c840), DICO(0x2f223840), DICO(0x390c8940), DICO(0x3e1d7340), DICO(0x41c6a000), DICO(0x46cc2880), DICO(0x5006e280), DICO(0x5744e180), DICO(0x5d297780), DICO(0x0401a3f0), DICO(0x07be4a08), DICO(0x14f5f1a0), DICO(0x1c348080), DICO(0x226753c0), DICO(0x274b1b80), DICO(0x2c11a300), DICO(0x30798c00), DICO(0x35598c80), DICO(0x3aab18c0), DICO(0x4030f380), DICO(0x45c7b400), DICO(0x4be5be00), DICO(0x5180db80), DICO(0x57613b00), DICO(0x5cffff80), DICO(0x084b3090), DICO(0x0fe5b3b0), DICO(0x16dfd480), DICO(0x1d0aa700), DICO(0x24e08180), DICO(0x2b498640), DICO(0x30885b80), DICO(0x34ccc480), DICO(0x38fedec0), DICO(0x3cdf9c40), DICO(0x41737600), DICO(0x46ab9800), DICO(0x4c772e80), DICO(0x51e5b980), DICO(0x57df9900), DICO(0x5d5d7180), DICO(0x09549f00), DICO(0x12b84da0), DICO(0x1aeaf1c0), DICO(0x2142a9c0), DICO(0x275c9a00), DICO(0x2c260c80), DICO(0x3038c700), DICO(0x34081d00), DICO(0x38612f40), DICO(0x3d0bf8c0), DICO(0x42473e00), DICO(0x47ecad80), DICO(0x4db34380), DICO(0x52f0ab00), DICO(0x584cc980), DICO(0x5d62ae80), DICO(0x0aeca1e0), DICO(0x14354700), DICO(0x19955ba0), DICO(0x1de331e0), DICO(0x21cd60c0), DICO(0x25e72d80), DICO(0x2a402880), DICO(0x2efa64c0), DICO(0x3478d9c0), DICO(0x3a889e80), DICO(0x40744e00), DICO(0x4626fe80), DICO(0x4bd80900), DICO(0x51120f00), DICO(0x56d8d280), DICO(0x5c654400), DICO(0x07a40af8), DICO(0x0e65ec20), DICO(0x14c76780), DICO(0x19b35a60), DICO(0x1f76b7c0), DICO(0x24f512c0), DICO(0x2ae89f00), DICO(0x3036c3c0), DICO(0x36041800), DICO(0x3bc4df80), DICO(0x41adb080), DICO(0x477fbb80), DICO(0x4d5aa480), DICO(0x52cb0600), DICO(0x586f7280), DICO(0x5db40180), DICO(0x03997610), DICO(0x06175db0), DICO(0x0ea8c9c0), DICO(0x14397000), DICO(0x1ba34f60), DICO(0x22346680), DICO(0x28958080), DICO(0x2ea76840), DICO(0x33ee68c0), DICO(0x39e769c0), DICO(0x3f862500), DICO(0x4579b080), DICO(0x4b49cd00), DICO(0x5107da80), DICO(0x573cde00), DICO(0x5d090780), DICO(0x046fc2f8), DICO(0x0640dbc0), DICO(0x09da7ab0), DICO(0x174e4220), DICO(0x23dc8cc0), DICO(0x27016200), DICO(0x2a9a5240), DICO(0x2e6cc7c0), DICO(0x321ced40), DICO(0x38ca2d00), DICO(0x41482680), DICO(0x451e3700), DICO(0x4b90d800), DICO(0x50d0ba80), DICO(0x55602e80), DICO(0x5a465200), DICO(0x03ca7e28), DICO(0x057c9dd0), DICO(0x0c9ff0e0), DICO(0x1957fae0), DICO(0x1fef7860), DICO(0x27920c80), DICO(0x2cb233c0), DICO(0x32015c80), DICO(0x36af4f40), DICO(0x3ac18240), DICO(0x3f93d8c0), DICO(0x44eaef80), DICO(0x4aab5d80), DICO(0x50840e80), DICO(0x56c4cc80), DICO(0x5cb26600), DICO(0x038d53f8), DICO(0x050d1118), DICO(0x0bc14690), DICO(0x18918000), DICO(0x1e2a6ee0), DICO(0x24cc0c00), DICO(0x2a767d40), DICO(0x2e614940), DICO(0x32859c40), DICO(0x377fd940), DICO(0x3d3a3e40), DICO(0x43e81380), DICO(0x4aaac080), DICO(0x509e7800), DICO(0x57023a00), DICO(0x5d733c00), DICO(0x04cfa5e0), DICO(0x06deeb38), DICO(0x0a501c40), DICO(0x136d8aa0), DICO(0x17f16e40), DICO(0x1c119300), DICO(0x26154b00), DICO(0x2a0da100), DICO(0x2f5935c0), DICO(0x37108d40), DICO(0x3aef07c0), DICO(0x3fccf340), DICO(0x47e4a080), DICO(0x4d8de100), DICO(0x54eb6980), DICO(0x5cdb5380)}; /* ACELP: table for decoding adaptive codebook gain g_p (left column). Scaled by 2.0f. innovative codebook gain g_c (right column). Scaled by 16.0f. */ const FIXP_SGL fdk_t_qua_gain7b[128 * 2] = { 204, 441, 464, 1977, 869, 1077, 1072, 3062, 1281, 4759, 1647, 1539, 1845, 7020, 1853, 634, 1995, 2336, 2351, 15400, 2661, 1165, 2702, 3900, 2710, 10133, 3195, 1752, 3498, 2624, 3663, 849, 3984, 5697, 4214, 3399, 4415, 1304, 4695, 2056, 5376, 4558, 5386, 676, 5518, 23554, 5567, 7794, 5644, 3061, 5672, 1513, 5957, 2338, 6533, 1060, 6804, 5998, 6820, 1767, 6937, 3837, 7277, 414, 7305, 2665, 7466, 11304, 7942, 794, 8007, 1982, 8007, 1366, 8326, 3105, 8336, 4810, 8708, 7954, 8989, 2279, 9031, 1055, 9247, 3568, 9283, 1631, 9654, 6311, 9811, 2605, 10120, 683, 10143, 4179, 10245, 1946, 10335, 1218, 10468, 9960, 10651, 3000, 10951, 1530, 10969, 5290, 11203, 2305, 11325, 3562, 11771, 6754, 11839, 1849, 11941, 4495, 11954, 1298, 11975, 15223, 11977, 883, 11986, 2842, 12438, 2141, 12593, 3665, 12636, 8367, 12658, 1594, 12886, 2628, 12984, 4942, 13146, 1115, 13224, 524, 13341, 3163, 13399, 1923, 13549, 5961, 13606, 1401, 13655, 2399, 13782, 3909, 13868, 10923, 14226, 1723, 14232, 2939, 14278, 7528, 14439, 4598, 14451, 984, 14458, 2265, 14792, 1403, 14818, 3445, 14899, 5709, 15017, 15362, 15048, 1946, 15069, 2655, 15405, 9591, 15405, 4079, 15570, 7183, 15687, 2286, 15691, 1624, 15699, 3068, 15772, 5149, 15868, 1205, 15970, 696, 16249, 3584, 16338, 1917, 16424, 2560, 16483, 4438, 16529, 6410, 16620, 11966, 16839, 8780, 17030, 3050, 17033, 18325, 17092, 1568, 17123, 5197, 17351, 2113, 17374, 980, 17566, 26214, 17609, 3912, 17639, 32767, 18151, 7871, 18197, 2516, 18202, 5649, 18679, 3283, 18930, 1370, 19271, 13757, 19317, 4120, 19460, 1973, 19654, 10018, 19764, 6792, 19912, 5135, 20040, 2841, 21234, 19833}; /* ACELP: factor table for interpolation of LPC coeffs in LSP domain */ const FIXP_SGL lsp_interpol_factor[2][NB_SUBFR] = { {FL2FXCONST_SGL(0.125f), FL2FXCONST_SGL(0.375f), FL2FXCONST_SGL(0.625f), FL2FXCONST_SGL(0.875f)}, /* for coreCoderFrameLength = 1024 */ {FL2FXCONST_SGL(0.166667f), FL2FXCONST_SGL(0.5f), FL2FXCONST_SGL(0.833333f), 0x0} /* for coreCoderFrameLength = 768 */ }; /* For bass post filter */ #ifndef TABLE_filt_lp const FIXP_SGL fdk_dec_filt_lp[1 + L_FILT] = { FL2FXCONST_SGL_FILT(0.088250f), FL2FXCONST_SGL_FILT(0.086410f), FL2FXCONST_SGL_FILT(0.081074f), FL2FXCONST_SGL_FILT(0.072768f), FL2FXCONST_SGL_FILT(0.062294f), FL2FXCONST_SGL_FILT(0.050623f), FL2FXCONST_SGL_FILT(0.038774f), FL2FXCONST_SGL_FILT(0.027692f), FL2FXCONST_SGL_FILT(0.018130f), FL2FXCONST_SGL_FILT(0.010578f), FL2FXCONST_SGL_FILT(0.005221f), FL2FXCONST_SGL_FILT(0.001946f), FL2FXCONST_SGL_FILT(0.000385f)}; #endif /* FAC window tables for coreCoderFrameLength = 1024 */ const FIXP_WTB FacWindowSynth128[] = { WTC(0x7fff6216), WTC(0x7ffa72d1), WTC(0x7ff09478), WTC(0x7fe1c76b), WTC(0x7fce0c3e), WTC(0x7fb563b3), WTC(0x7f97cebd), WTC(0x7f754e80), WTC(0x7f4de451), WTC(0x7f2191b4), WTC(0x7ef05860), WTC(0x7eba3a39), WTC(0x7e7f3957), WTC(0x7e3f57ff), WTC(0x7dfa98a8), WTC(0x7db0fdf8), WTC(0x7d628ac6), WTC(0x7d0f4218), WTC(0x7cb72724), WTC(0x7c5a3d50), WTC(0x7bf88830), WTC(0x7b920b89), WTC(0x7b26cb4f), WTC(0x7ab6cba4), WTC(0x7a4210d8), WTC(0x79c89f6e), WTC(0x794a7c12), WTC(0x78c7aba2), WTC(0x78403329), WTC(0x77b417df), WTC(0x77235f2d), WTC(0x768e0ea6), WTC(0x75f42c0b), WTC(0x7555bd4c), WTC(0x74b2c884), WTC(0x740b53fb), WTC(0x735f6626), WTC(0x72af05a7), WTC(0x71fa3949), WTC(0x71410805), WTC(0x708378ff), WTC(0x6fc19385), WTC(0x6efb5f12), WTC(0x6e30e34a), WTC(0x6d6227fa), WTC(0x6c8f351c), WTC(0x6bb812d1), WTC(0x6adcc964), WTC(0x69fd614a), WTC(0x6919e320), WTC(0x683257ab), WTC(0x6746c7d8), WTC(0x66573cbb), WTC(0x6563bf92), WTC(0x646c59bf), WTC(0x637114cc), WTC(0x6271fa69), WTC(0x616f146c), WTC(0x60686ccf), WTC(0x5f5e0db3), WTC(0x5e50015d), WTC(0x5d3e5237), WTC(0x5c290acc), WTC(0x5b1035cf), WTC(0x59f3de12), WTC(0x58d40e8c), WTC(0x57b0d256), WTC(0x568a34a9), WTC(0x556040e2), WTC(0x5433027d), WTC(0x53028518), WTC(0x51ced46e), WTC(0x5097fc5e), WTC(0x4f5e08e3), WTC(0x4e210617), WTC(0x4ce10034), WTC(0x4b9e0390), WTC(0x4a581c9e), WTC(0x490f57ee), WTC(0x47c3c22f), WTC(0x46756828), WTC(0x452456bd), WTC(0x43d09aed), WTC(0x427a41d0), WTC(0x4121589b), WTC(0x3fc5ec98), WTC(0x3e680b2c), WTC(0x3d07c1d6), WTC(0x3ba51e29), WTC(0x3a402dd2), WTC(0x38d8fe93), WTC(0x376f9e46), WTC(0x36041ad9), WTC(0x34968250), WTC(0x3326e2c3), WTC(0x31b54a5e), WTC(0x3041c761), WTC(0x2ecc681e), WTC(0x2d553afc), WTC(0x2bdc4e6f), WTC(0x2a61b101), WTC(0x28e5714b), WTC(0x27679df4), WTC(0x25e845b6), WTC(0x24677758), WTC(0x22e541af), WTC(0x2161b3a0), WTC(0x1fdcdc1b), WTC(0x1e56ca1e), WTC(0x1ccf8cb3), WTC(0x1b4732ef), WTC(0x19bdcbf3), WTC(0x183366e9), WTC(0x16a81305), WTC(0x151bdf86), WTC(0x138edbb1), WTC(0x120116d5), WTC(0x1072a048), WTC(0x0ee38766), WTC(0x0d53db92), WTC(0x0bc3ac35), WTC(0x0a3308bd), WTC(0x08a2009a), WTC(0x0710a345), WTC(0x057f0035), WTC(0x03ed26e6), WTC(0x025b26d7), WTC(0x00c90f88), }; const FIXP_WTB FacWindowZir128[] = { WTC(0x7f36f078), WTC(0x7da4d929), WTC(0x7c12d91a), WTC(0x7a80ffcb), WTC(0x78ef5cbb), WTC(0x775dff66), WTC(0x75ccf743), WTC(0x743c53cb), WTC(0x72ac246e), WTC(0x711c789a), WTC(0x6f8d5fb8), WTC(0x6dfee92b), WTC(0x6c71244f), WTC(0x6ae4207a), WTC(0x6957ecfb), WTC(0x67cc9917), WTC(0x6642340d), WTC(0x64b8cd11), WTC(0x6330734d), WTC(0x61a935e2), WTC(0x602323e5), WTC(0x5e9e4c60), WTC(0x5d1abe51), WTC(0x5b9888a8), WTC(0x5a17ba4a), WTC(0x5898620c), WTC(0x571a8eb5), WTC(0x559e4eff), WTC(0x5423b191), WTC(0x52aac504), WTC(0x513397e2), WTC(0x4fbe389f), WTC(0x4e4ab5a2), WTC(0x4cd91d3d), WTC(0x4b697db0), WTC(0x49fbe527), WTC(0x489061ba), WTC(0x4727016d), WTC(0x45bfd22e), WTC(0x445ae1d7), WTC(0x42f83e2a), WTC(0x4197f4d4), WTC(0x403a1368), WTC(0x3edea765), WTC(0x3d85be30), WTC(0x3c2f6513), WTC(0x3adba943), WTC(0x398a97d8), WTC(0x383c3dd1), WTC(0x36f0a812), WTC(0x35a7e362), WTC(0x3461fc70), WTC(0x331effcc), WTC(0x31def9e9), WTC(0x30a1f71d), WTC(0x2f6803a2), WTC(0x2e312b92), WTC(0x2cfd7ae8), WTC(0x2bccfd83), WTC(0x2a9fbf1e), WTC(0x2975cb57), WTC(0x284f2daa), WTC(0x272bf174), WTC(0x260c21ee), WTC(0x24efca31), WTC(0x23d6f534), WTC(0x22c1adc9), WTC(0x21affea3), WTC(0x20a1f24d), WTC(0x1f979331), WTC(0x1e90eb94), WTC(0x1d8e0597), WTC(0x1c8eeb34), WTC(0x1b93a641), WTC(0x1a9c406e), WTC(0x19a8c345), WTC(0x18b93828), WTC(0x17cda855), WTC(0x16e61ce0), WTC(0x16029eb6), WTC(0x1523369c), WTC(0x1447ed2f), WTC(0x1370cae4), WTC(0x129dd806), WTC(0x11cf1cb6), WTC(0x1104a0ee), WTC(0x103e6c7b), WTC(0x0f7c8701), WTC(0x0ebef7fb), WTC(0x0e05c6b7), WTC(0x0d50fa59), WTC(0x0ca099da), WTC(0x0bf4ac05), WTC(0x0b4d377c), WTC(0x0aaa42b4), WTC(0x0a0bd3f5), WTC(0x0971f15a), WTC(0x08dca0d3), WTC(0x084be821), WTC(0x07bfccd7), WTC(0x0738545e), WTC(0x06b583ee), WTC(0x06376092), WTC(0x05bdef28), WTC(0x0549345c), WTC(0x04d934b1), WTC(0x046df477), WTC(0x040777d0), WTC(0x03a5c2b0), WTC(0x0348d8dc), WTC(0x02f0bde8), WTC(0x029d753a), WTC(0x024f0208), WTC(0x02056758), WTC(0x01c0a801), WTC(0x0180c6a9), WTC(0x0145c5c7), WTC(0x010fa7a0), WTC(0x00de6e4c), WTC(0x00b21baf), WTC(0x008ab180), WTC(0x00683143), WTC(0x004a9c4d), WTC(0x0031f3c2), WTC(0x001e3895), WTC(0x000f6b88), WTC(0x00058d2f), WTC(0x00009dea), }; const FIXP_WTB FacWindowSynth64[] = { WTC(0x7ffd885a), WTC(0x7fe9cbc0), WTC(0x7fc25596), WTC(0x7f872bf3), WTC(0x7f3857f6), WTC(0x7ed5e5c6), WTC(0x7e5fe493), WTC(0x7dd6668f), WTC(0x7d3980ec), WTC(0x7c894bde), WTC(0x7bc5e290), WTC(0x7aef6323), WTC(0x7a05eead), WTC(0x7909a92d), WTC(0x77fab989), WTC(0x76d94989), WTC(0x75a585cf), WTC(0x745f9dd1), WTC(0x7307c3d0), WTC(0x719e2cd2), WTC(0x7023109a), WTC(0x6e96a99d), WTC(0x6cf934fc), WTC(0x6b4af279), WTC(0x698c246c), WTC(0x67bd0fbd), WTC(0x65ddfbd3), WTC(0x63ef3290), WTC(0x61f1003f), WTC(0x5fe3b38d), WTC(0x5dc79d7c), WTC(0x5b9d1154), WTC(0x59646498), WTC(0x571deefa), WTC(0x54ca0a4b), WTC(0x5269126e), WTC(0x4ffb654d), WTC(0x4d8162c4), WTC(0x4afb6c98), WTC(0x4869e665), WTC(0x45cd358f), WTC(0x4325c135), WTC(0x4073f21d), WTC(0x3db832a6), WTC(0x3af2eeb7), WTC(0x382493b0), WTC(0x354d9057), WTC(0x326e54c7), WTC(0x2f875262), WTC(0x2c98fbba), WTC(0x29a3c485), WTC(0x26a82186), WTC(0x23a6887f), WTC(0x209f701c), WTC(0x1d934fe5), WTC(0x1a82a026), WTC(0x176dd9de), WTC(0x145576b1), WTC(0x1139f0cf), WTC(0x0e1bc2e4), WTC(0x0afb6805), WTC(0x07d95b9e), WTC(0x04b6195d), WTC(0x01921d20), }; const FIXP_WTB FacWindowZir64[] = { WTC(0x7e6de2e0), WTC(0x7b49e6a3), WTC(0x7826a462), WTC(0x750497fb), WTC(0x71e43d1c), WTC(0x6ec60f31), WTC(0x6baa894f), WTC(0x68922622), WTC(0x657d5fda), WTC(0x626cb01b), WTC(0x5f608fe4), WTC(0x5c597781), WTC(0x5957de7a), WTC(0x565c3b7b), WTC(0x53670446), WTC(0x5078ad9e), WTC(0x4d91ab39), WTC(0x4ab26fa9), WTC(0x47db6c50), WTC(0x450d1149), WTC(0x4247cd5a), WTC(0x3f8c0de3), WTC(0x3cda3ecb), WTC(0x3a32ca71), WTC(0x3796199b), WTC(0x35049368), WTC(0x327e9d3c), WTC(0x30049ab3), WTC(0x2d96ed92), WTC(0x2b35f5b5), WTC(0x28e21106), WTC(0x269b9b68), WTC(0x2462eeac), WTC(0x22386284), WTC(0x201c4c73), WTC(0x1e0effc1), WTC(0x1c10cd70), WTC(0x1a22042d), WTC(0x1842f043), WTC(0x1673db94), WTC(0x14b50d87), WTC(0x1306cb04), WTC(0x11695663), WTC(0x0fdcef66), WTC(0x0e61d32e), WTC(0x0cf83c30), WTC(0x0ba0622f), WTC(0x0a5a7a31), WTC(0x0926b677), WTC(0x08054677), WTC(0x06f656d3), WTC(0x05fa1153), WTC(0x05109cdd), WTC(0x043a1d70), WTC(0x0376b422), WTC(0x02c67f14), WTC(0x02299971), WTC(0x01a01b6d), WTC(0x012a1a3a), WTC(0x00c7a80a), WTC(0x0078d40d), WTC(0x003daa6a), WTC(0x00163440), WTC(0x000277a6), }; const FIXP_WTB FacWindowSynth32[] = { WTC(0x7ff62182), WTC(0x7fa736b4), WTC(0x7f0991c4), WTC(0x7e1d93ea), WTC(0x7ce3ceb2), WTC(0x7b5d039e), WTC(0x798a23b1), WTC(0x776c4edb), WTC(0x7504d345), WTC(0x72552c85), WTC(0x6f5f02b2), WTC(0x6c242960), WTC(0x68a69e81), WTC(0x64e88926), WTC(0x60ec3830), WTC(0x5cb420e0), WTC(0x5842dd54), WTC(0x539b2af0), WTC(0x4ebfe8a5), WTC(0x49b41533), WTC(0x447acd50), WTC(0x3f1749b8), WTC(0x398cdd32), WTC(0x33def287), WTC(0x2e110a62), WTC(0x2826b928), WTC(0x2223a4c5), WTC(0x1c0b826a), WTC(0x15e21445), WTC(0x0fab272b), WTC(0x096a9049), WTC(0x03242abf), }; const FIXP_WTB FacWindowZir32[] = { WTC(0x7cdbd541), WTC(0x76956fb7), WTC(0x7054d8d5), WTC(0x6a1debbb), WTC(0x63f47d96), WTC(0x5ddc5b3b), WTC(0x57d946d8), WTC(0x51eef59e), WTC(0x4c210d79), WTC(0x467322ce), WTC(0x40e8b648), WTC(0x3b8532b0), WTC(0x364beacd), WTC(0x3140175b), WTC(0x2c64d510), WTC(0x27bd22ac), WTC(0x234bdf20), WTC(0x1f13c7d0), WTC(0x1b1776da), WTC(0x1759617f), WTC(0x13dbd6a0), WTC(0x10a0fd4e), WTC(0x0daad37b), WTC(0x0afb2cbb), WTC(0x0893b125), WTC(0x0675dc4f), WTC(0x04a2fc62), WTC(0x031c314e), WTC(0x01e26c16), WTC(0x00f66e3c), WTC(0x0058c94c), WTC(0x0009de7e), }; /* FAC window tables for coreCoderFrameLength = 768 */ const FIXP_WTB FacWindowSynth96[] = { WTC(0x7ffee744), WTC(0x7ff62182), WTC(0x7fe49698), WTC(0x7fca47b9), WTC(0x7fa736b4), WTC(0x7f7b65ef), WTC(0x7f46d86c), WTC(0x7f0991c4), WTC(0x7ec3962a), WTC(0x7e74ea6a), WTC(0x7e1d93ea), WTC(0x7dbd98a4), WTC(0x7d54ff2e), WTC(0x7ce3ceb2), WTC(0x7c6a0ef2), WTC(0x7be7c847), WTC(0x7b5d039e), WTC(0x7ac9ca7a), WTC(0x7a2e26f2), WTC(0x798a23b1), WTC(0x78ddcbf5), WTC(0x78292b8d), WTC(0x776c4edb), WTC(0x76a742d1), WTC(0x75da14ef), WTC(0x7504d345), WTC(0x74278c72), WTC(0x73424fa0), WTC(0x72552c85), WTC(0x71603361), WTC(0x706374ff), WTC(0x6f5f02b2), WTC(0x6e52ee52), WTC(0x6d3f4a40), WTC(0x6c242960), WTC(0x6b019f1a), WTC(0x69d7bf57), WTC(0x68a69e81), WTC(0x676e5183), WTC(0x662eedc3), WTC(0x64e88926), WTC(0x639b3a0b), WTC(0x62471749), WTC(0x60ec3830), WTC(0x5f8ab487), WTC(0x5e22a487), WTC(0x5cb420e0), WTC(0x5b3f42ae), WTC(0x59c42381), WTC(0x5842dd54), WTC(0x56bb8a90), WTC(0x552e4605), WTC(0x539b2af0), WTC(0x520254ef), WTC(0x5063e008), WTC(0x4ebfe8a5), WTC(0x4d168b8b), WTC(0x4b67e5e4), WTC(0x49b41533), WTC(0x47fb3757), WTC(0x463d6a87), WTC(0x447acd50), WTC(0x42b37e96), WTC(0x40e79d8c), WTC(0x3f1749b8), WTC(0x3d42a2ec), WTC(0x3b69c947), WTC(0x398cdd32), WTC(0x37abff5d), WTC(0x35c750bc), WTC(0x33def287), WTC(0x31f30638), WTC(0x3003ad85), WTC(0x2e110a62), WTC(0x2c1b3efb), WTC(0x2a226db5), WTC(0x2826b928), WTC(0x26284422), WTC(0x2427319d), WTC(0x2223a4c5), WTC(0x201dc0ef), WTC(0x1e15a99a), WTC(0x1c0b826a), WTC(0x19ff6f2a), WTC(0x17f193c5), WTC(0x15e21445), WTC(0x13d114d0), WTC(0x11beb9aa), WTC(0x0fab272b), WTC(0x0d9681c2), WTC(0x0b80edf1), WTC(0x096a9049), WTC(0x07538d6b), WTC(0x053c0a01), WTC(0x03242abf), WTC(0x010c1460), }; const FIXP_WTB FacWindowZir96[] = { WTC(0x7ef3eba0), WTC(0x7cdbd541), WTC(0x7ac3f5ff), WTC(0x78ac7295), WTC(0x76956fb7), WTC(0x747f120f), WTC(0x72697e3e), WTC(0x7054d8d5), WTC(0x6e414656), WTC(0x6c2eeb30), WTC(0x6a1debbb), WTC(0x680e6c3b), WTC(0x660090d6), WTC(0x63f47d96), WTC(0x61ea5666), WTC(0x5fe23f11), WTC(0x5ddc5b3b), WTC(0x5bd8ce63), WTC(0x59d7bbde), WTC(0x57d946d8), WTC(0x55dd924b), WTC(0x53e4c105), WTC(0x51eef59e), WTC(0x4ffc527b), WTC(0x4e0cf9c8), WTC(0x4c210d79), WTC(0x4a38af44), WTC(0x485400a3), WTC(0x467322ce), WTC(0x449636b9), WTC(0x42bd5d14), WTC(0x40e8b648), WTC(0x3f186274), WTC(0x3d4c816a), WTC(0x3b8532b0), WTC(0x39c29579), WTC(0x3804c8a9), WTC(0x364beacd), WTC(0x34981a1c), WTC(0x32e97475), WTC(0x3140175b), WTC(0x2f9c1ff8), WTC(0x2dfdab11), WTC(0x2c64d510), WTC(0x2ad1b9fb), WTC(0x29447570), WTC(0x27bd22ac), WTC(0x263bdc7f), WTC(0x24c0bd52), WTC(0x234bdf20), WTC(0x21dd5b79), WTC(0x20754b79), WTC(0x1f13c7d0), WTC(0x1db8e8b7), WTC(0x1c64c5f5), WTC(0x1b1776da), WTC(0x19d1123d), WTC(0x1891ae7d), WTC(0x1759617f), WTC(0x162840a9), WTC(0x14fe60e6), WTC(0x13dbd6a0), WTC(0x12c0b5c0), WTC(0x11ad11ae), WTC(0x10a0fd4e), WTC(0x0f9c8b01), WTC(0x0e9fcc9f), WTC(0x0daad37b), WTC(0x0cbdb060), WTC(0x0bd8738e), WTC(0x0afb2cbb), WTC(0x0a25eb11), WTC(0x0958bd2f), WTC(0x0893b125), WTC(0x07d6d473), WTC(0x0722340b), WTC(0x0675dc4f), WTC(0x05d1d90e), WTC(0x05363586), WTC(0x04a2fc62), WTC(0x041837b9), WTC(0x0395f10e), WTC(0x031c314e), WTC(0x02ab00d2), WTC(0x0242675c), WTC(0x01e26c16), WTC(0x018b1596), WTC(0x013c69d6), WTC(0x00f66e3c), WTC(0x00b92794), WTC(0x00849a11), WTC(0x0058c94c), WTC(0x0035b847), WTC(0x001b6968), WTC(0x0009de7e), WTC(0x000118bc), }; const FIXP_WTB FacWindowSynth48[] = { WTC(0x7ffb9d15), WTC(0x7fd8878e), WTC(0x7f92661d), WTC(0x7f294bfd), WTC(0x7e9d55fc), WTC(0x7deeaa7a), WTC(0x7d1d7958), WTC(0x7c29fbee), WTC(0x7b1474fd), WTC(0x79dd3098), WTC(0x78848414), WTC(0x770acdec), WTC(0x757075ac), WTC(0x73b5ebd1), WTC(0x71dba9ab), WTC(0x6fe2313c), WTC(0x6dca0d14), WTC(0x6b93d02e), WTC(0x694015c3), WTC(0x66cf8120), WTC(0x6442bd7e), WTC(0x619a7dce), WTC(0x5ed77c8a), WTC(0x5bfa7b82), WTC(0x590443a7), WTC(0x55f5a4d2), WTC(0x52cf758f), WTC(0x4f9292dc), WTC(0x4c3fdff4), WTC(0x48d84609), WTC(0x455cb40c), WTC(0x41ce1e65), WTC(0x3e2d7eb1), WTC(0x3a7bd382), WTC(0x36ba2014), WTC(0x32e96c09), WTC(0x2f0ac320), WTC(0x2b1f34eb), WTC(0x2727d486), WTC(0x2325b847), WTC(0x1f19f97b), WTC(0x1b05b40f), WTC(0x16ea0646), WTC(0x12c8106f), WTC(0x0ea0f48c), WTC(0x0a75d60e), WTC(0x0647d97c), WTC(0x02182427), }; const FIXP_WTB FacWindowZir48[] = { WTC(0x7de7dbd9), WTC(0x79b82684), WTC(0x758a29f2), WTC(0x715f0b74), WTC(0x6d37ef91), WTC(0x6915f9ba), WTC(0x64fa4bf1), WTC(0x60e60685), WTC(0x5cda47b9), WTC(0x58d82b7a), WTC(0x54e0cb15), WTC(0x50f53ce0), WTC(0x4d1693f7), WTC(0x4945dfec), WTC(0x45842c7e), WTC(0x41d2814f), WTC(0x3e31e19b), WTC(0x3aa34bf4), WTC(0x3727b9f7), WTC(0x33c0200c), WTC(0x306d6d24), WTC(0x2d308a71), WTC(0x2a0a5b2e), WTC(0x26fbbc59), WTC(0x2405847e), WTC(0x21288376), WTC(0x1e658232), WTC(0x1bbd4282), WTC(0x19307ee0), WTC(0x16bfea3d), WTC(0x146c2fd2), WTC(0x1235f2ec), WTC(0x101dcec4), WTC(0x0e245655), WTC(0x0c4a142f), WTC(0x0a8f8a54), WTC(0x08f53214), WTC(0x077b7bec), WTC(0x0622cf68), WTC(0x04eb8b03), WTC(0x03d60412), WTC(0x02e286a8), WTC(0x02115586), WTC(0x0162aa04), WTC(0x00d6b403), WTC(0x006d99e3), WTC(0x00277872), WTC(0x000462eb), }; fdk-aac-2.0.2/libAACdec/src/usacdec_rom.h000066400000000000000000000143251404225310700177570ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC decoder library ****************************** Author(s): M. Jander Description: re8.h *******************************************************************************/ #ifndef USACDEC_ROM_H #define USACDEC_ROM_H #include "common_fix.h" #include "FDK_lpc.h" #include "usacdec_const.h" /* RE8 lattice quantiser constants */ #define NB_SPHERE 32 #define NB_LEADER 37 #define NB_LDSIGN 226 #define NB_LDQ3 9 #define NB_LDQ4 28 #define LSF_SCALE 13 /* RE8 lattice quantiser tables */ extern const UINT fdk_dec_tab_factorial[8]; extern const UCHAR fdk_dec_Ia[NB_LEADER]; extern const UCHAR fdk_dec_Ds[NB_LDSIGN]; extern const USHORT fdk_dec_Is[NB_LDSIGN]; extern const UCHAR fdk_dec_Ns[], fdk_dec_A3[], fdk_dec_A4[]; extern const UCHAR fdk_dec_Da[][8]; extern const USHORT fdk_dec_I3[], fdk_dec_I4[]; /* temp float tables for LPC decoding */ extern const FIXP_LPC fdk_dec_lsf_init[16]; extern const FIXP_LPC fdk_dec_dico_lsf_abs_8b[16 * 256]; /* ACELP tables */ #define SF_QUA_GAIN7B 4 extern const FIXP_SGL fdk_t_qua_gain7b[128 * 2]; extern const FIXP_SGL lsp_interpol_factor[2][NB_SUBFR]; /* For bass post filter */ #define L_FILT 12 /* Delay of up-sampling filter */ extern const FIXP_SGL fdk_dec_filt_lp[1 + L_FILT]; extern const FIXP_WTB FacWindowSynth128[128]; extern const FIXP_WTB FacWindowZir128[128]; extern const FIXP_WTB FacWindowSynth64[64]; extern const FIXP_WTB FacWindowZir64[64]; extern const FIXP_WTB FacWindowSynth32[32]; extern const FIXP_WTB FacWindowZir32[32]; extern const FIXP_WTB FacWindowSynth96[96]; extern const FIXP_WTB FacWindowZir96[96]; extern const FIXP_WTB FacWindowSynth48[48]; extern const FIXP_WTB FacWindowZir48[48]; #endif /* USACDEC_ROM_H */ fdk-aac-2.0.2/libAACenc/000077500000000000000000000000001404225310700145405ustar00rootroot00000000000000fdk-aac-2.0.2/libAACenc/include/000077500000000000000000000000001404225310700161635ustar00rootroot00000000000000fdk-aac-2.0.2/libAACenc/include/aacenc_lib.h000066400000000000000000002536401404225310700204060ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Lohwasser Description: *******************************************************************************/ /** * \file aacenc_lib.h * \brief FDK AAC Encoder library interface header file. * \mainpage Introduction \section Scope This document describes the high-level interface and usage of the ISO/MPEG-2/4 AAC Encoder library developed by the Fraunhofer Institute for Integrated Circuits (IIS). The library implements encoding on the basis of the MPEG-2 and MPEG-4 AAC Low-Complexity standard, and depending on the library's configuration, MPEG-4 High-Efficiency AAC v2 and/or AAC-ELD standard. All references to SBR (Spectral Band Replication) are only applicable to HE-AAC or AAC-ELD versions of the library. All references to PS (Parametric Stereo) are only applicable to HE-AAC v2 versions of the library. \section encBasics Encoder Basics This document can only give a rough overview about the ISO/MPEG-2 and ISO/MPEG-4 AAC audio coding standard. To understand all the terms in this document, you are encouraged to read the following documents. - ISO/IEC 13818-7 (MPEG-2 AAC), which defines the syntax of MPEG-2 AAC audio bitstreams. - ISO/IEC 14496-3 (MPEG-4 AAC, subparts 1 and 4), which defines the syntax of MPEG-4 AAC audio bitstreams. - Lutzky, Schuller, Gayer, Krämer, Wabnik, "A guideline to audio codec delay", 116th AES Convention, May 8, 2004 MPEG Advanced Audio Coding is based on a time-to-frequency mapping of the signal. The signal is partitioned into overlapping portions and transformed into frequency domain. The spectral components are then quantized and coded. \n An MPEG-2 or MPEG-4 AAC audio bitstream is composed of frames. Contrary to MPEG-1/2 Layer-3 (mp3), the length of individual frames is not restricted to a fixed number of bytes, but can take on any length between 1 and 768 bytes. \page LIBUSE Library Usage \section InterfaceDescription API Files All API header files are located in the folder /include of the release package. All header files are provided for usage in C/C++ programs. The AAC encoder library API functions are located in aacenc_lib.h. \section CallingSequence Calling Sequence For encoding of ISO/MPEG-2/4 AAC bitstreams the following sequence is mandatory. Input read and output write functions as well as the corresponding open and close functions are left out, since they may be implemented differently according to the user's specific requirements. The example implementation uses file-based input/output. -# Call aacEncOpen() to allocate encoder instance with required \ref encOpen "configuration". \code HANDLE_AACENCODER hAacEncoder = NULL; if ( (ErrorStatus = aacEncOpen(&hAacEncoder,0,0)) != AACENC_OK ) { \endcode -# Call aacEncoder_SetParam() for each parameter to be set. AOT, samplingrate, channelMode, bitrate and transport type are \ref encParams "mandatory". \code ErrorStatus = aacEncoder_SetParam(hAacEncoder, parameter, value); \endcode -# Call aacEncEncode() with NULL parameters to \ref encReconf "initialize" encoder instance with present parameter set. \code ErrorStatus = aacEncEncode(hAacEncoder, NULL, NULL, NULL, NULL); \endcode -# Call aacEncInfo() to retrieve a configuration data block to be transmitted out of band. This is required when using RFC3640 or RFC3016 like transport. \code AACENC_InfoStruct encInfo; aacEncInfo(hAacEncoder, &encInfo); \endcode -# Encode input audio data in loop. \code do { \endcode Feed \ref feedInBuf "input buffer" with new audio data and provide input/output \ref bufDes "arguments" to aacEncEncode(). \code ErrorStatus = aacEncEncode(hAacEncoder, &inBufDesc, &outBufDesc, &inargs, &outargs); \endcode Write \ref writeOutData "output data" to file or audio device. \code } while (ErrorStatus==AACENC_OK); \endcode -# Call aacEncClose() and destroy encoder instance. \code aacEncClose(&hAacEncoder); \endcode \section encOpen Encoder Instance Allocation The assignment of the aacEncOpen() function is very flexible and can be used in the following way. - If the amount of memory consumption is not an issue, the encoder instance can be allocated for the maximum number of possible audio channels (for example 6 or 8) with the full functional range supported by the library. This is the default open procedure for the AAC encoder if memory consumption does not need to be minimized. \code aacEncOpen(&hAacEncoder,0,0) \endcode - If the required MPEG-4 AOTs do not call for the full functional range of the library, encoder modules can be allocated selectively. \verbatim ------------------------------------------------------ AAC | SBR | PS | MD | FLAGS | value -----+-----+-----+----+-----------------------+------- X | - | - | - | (0x01) | 0x01 X | X | - | - | (0x01|0x02) | 0x03 X | X | X | - | (0x01|0x02|0x04) | 0x07 X | - | - | X | (0x01 |0x10) | 0x11 X | X | - | X | (0x01|0x02 |0x10) | 0x13 X | X | X | X | (0x01|0x02|0x04|0x10) | 0x17 ------------------------------------------------------ - AAC: Allocate AAC Core Encoder module. - SBR: Allocate Spectral Band Replication module. - PS: Allocate Parametric Stereo module. - MD: Allocate Meta Data module within AAC encoder. \endverbatim \code aacEncOpen(&hAacEncoder,value,0) \endcode - Specifying the maximum number of channels to be supported in the encoder instance can be done as follows. - For example allocate an encoder instance which supports 2 channels for all supported AOTs. The library itself may be capable of encoding up to 6 or 8 channels but in this example only 2 channel encoding is required and thus only buffers for 2 channels are allocated to save data memory. \code aacEncOpen(&hAacEncoder,0,2) \endcode - Additionally the maximum number of supported channels in the SBR module can be denoted separately.\n In this example the encoder instance provides a maximum of 6 channels out of which up to 2 channels support SBR. This encoder instance can produce for example 5.1 channel AAC-LC streams or stereo HE-AAC (v2) streams. HE-AAC 5.1 multi channel is not possible since only 2 out of 6 channels support SBR, which saves data memory. \code aacEncOpen(&hAacEncoder,0,6|(2<<8)) \endcode \n \section bufDes Input/Output Arguments \subsection allocIOBufs Provide Buffer Descriptors In the present encoder API, the input and output buffers are described with \ref AACENC_BufDesc "buffer descriptors". This mechanism allows a flexible handling of input and output buffers without impact to the actual encoding call. Optional buffers are necessary e.g. for ancillary data, meta data input or additional output buffers describing superframing data in DAB+ or DRM+.\n At least one input buffer for audio input data and one output buffer for bitstream data must be allocated. The input buffer size can be a user defined multiple of the number of input channels. PCM input data will be copied from the user defined PCM buffer to an internal input buffer and so input data can be less than one AAC audio frame. The output buffer size should be 6144 bits per channel excluding the LFE channel. If the output data does not fit into the provided buffer, an AACENC_ERROR will be returned by aacEncEncode(). \code static INT_PCM inputBuffer[8*2048]; static UCHAR ancillaryBuffer[50]; static AACENC_MetaData metaDataSetup; static UCHAR outputBuffer[8192]; \endcode All input and output buffer must be clustered in input and output buffer arrays. \code static void* inBuffer[] = { inputBuffer, ancillaryBuffer, &metaDataSetup }; static INT inBufferIds[] = { IN_AUDIO_DATA, IN_ANCILLRY_DATA, IN_METADATA_SETUP }; static INT inBufferSize[] = { sizeof(inputBuffer), sizeof(ancillaryBuffer), sizeof(metaDataSetup) }; static INT inBufferElSize[] = { sizeof(INT_PCM), sizeof(UCHAR), sizeof(AACENC_MetaData) }; static void* outBuffer[] = { outputBuffer }; static INT outBufferIds[] = { OUT_BITSTREAM_DATA }; static INT outBufferSize[] = { sizeof(outputBuffer) }; static INT outBufferElSize[] = { sizeof(UCHAR) }; \endcode Allocate buffer descriptors \code AACENC_BufDesc inBufDesc; AACENC_BufDesc outBufDesc; \endcode Initialize input buffer descriptor \code inBufDesc.numBufs = sizeof(inBuffer)/sizeof(void*); inBufDesc.bufs = (void**)&inBuffer; inBufDesc.bufferIdentifiers = inBufferIds; inBufDesc.bufSizes = inBufferSize; inBufDesc.bufElSizes = inBufferElSize; \endcode Initialize output buffer descriptor \code outBufDesc.numBufs = sizeof(outBuffer)/sizeof(void*); outBufDesc.bufs = (void**)&outBuffer; outBufDesc.bufferIdentifiers = outBufferIds; outBufDesc.bufSizes = outBufferSize; outBufDesc.bufElSizes = outBufferElSize; \endcode \subsection argLists Provide Input/Output Argument Lists The input and output arguments of an aacEncEncode() call are described in argument structures. \code AACENC_InArgs inargs; AACENC_OutArgs outargs; \endcode \section feedInBuf Feed Input Buffer The input buffer should be handled as a modulo buffer. New audio data in the form of pulse-code- modulated samples (PCM) must be read from external and be fed to the input buffer depending on its fill level. The required sample bitrate (represented by the data type INT_PCM which is 16, 24 or 32 bits wide) is fixed and depends on library configuration (usually 16 bit). \code inargs.numInSamples += WAV_InputRead ( wavIn, &inputBuffer[inargs.numInSamples], FDKmin(encInfo.inputChannels*encInfo.frameLength, sizeof(inputBuffer) / sizeof(INT_PCM)-inargs.numInSamples), SAMPLE_BITS ); \endcode After the encoder's internal buffer is fed with incoming audio samples, and aacEncEncode() processed the new input data, update/move remaining samples in input buffer, simulating a modulo buffer: \code if (outargs.numInSamples>0) { FDKmemmove( inputBuffer, &inputBuffer[outargs.numInSamples], sizeof(INT_PCM)*(inargs.numInSamples-outargs.numInSamples) ); inargs.numInSamples -= outargs.numInSamples; } \endcode \section writeOutData Output Bitstream Data If any AAC bitstream data is available, write it to output file or device as follows. \code if (outargs.numOutBytes>0) { FDKfwrite(outputBuffer, outargs.numOutBytes, 1, pOutFile); } \endcode \section cfgMetaData Meta Data Configuration If the present library is configured with Metadata support, it is possible to insert meta data side info into the generated audio bitstream while encoding. To work with meta data the encoder instance has to be \ref encOpen "allocated" with meta data support. The meta data mode must be configured with the ::AACENC_METADATA_MODE parameter and aacEncoder_SetParam() function. \code aacEncoder_SetParam(hAacEncoder, AACENC_METADATA_MODE, 0-3); \endcode This configuration indicates how to embed meta data into bitstrem. Either no insertion, MPEG or ETSI style. The meta data itself must be specified within the meta data setup structure AACENC_MetaData. Changing one of the AACENC_MetaData setup parameters can be achieved from outside the library within ::IN_METADATA_SETUP input buffer. There is no need to supply meta data setup structure every frame. If there is no new meta setup data available, the encoder uses the previous setup or the default configuration in initial state. In general the audio compressor and limiter within the encoder library can be configured with the ::AACENC_METADATA_DRC_PROFILE parameter AACENC_MetaData::drc_profile and and AACENC_MetaData::comp_profile. \n \section encReconf Encoder Reconfiguration The encoder library allows reconfiguration of the encoder instance with new settings continuously between encoding frames. Each parameter to be changed must be set with a single aacEncoder_SetParam() call. The internal status of each parameter can be retrieved with an aacEncoder_GetParam() call.\n There is no stand-alone reconfiguration function available. When parameters were modified from outside the library, an internal control mechanism triggers the necessary reconfiguration process which will be applied at the beginning of the following aacEncEncode() call. This state can be observed from external via the AACENC_INIT_STATUS and aacEncoder_GetParam() function. The reconfiguration process can also be applied immediately when all parameters of an aacEncEncode() call are NULL with a valid encoder handle.\n\n The internal reconfiguration process can be controlled from extern with the following access. \code aacEncoder_SetParam(hAacEncoder, AACENC_CONTROL_STATE, AACENC_CTRLFLAGS); \endcode \section encParams Encoder Parametrization All parameteres listed in ::AACENC_PARAM can be modified within an encoder instance. \subsection encMandatory Mandatory Encoder Parameters The following parameters must be specified when the encoder instance is initialized. \code aacEncoder_SetParam(hAacEncoder, AACENC_AOT, value); aacEncoder_SetParam(hAacEncoder, AACENC_BITRATE, value); aacEncoder_SetParam(hAacEncoder, AACENC_SAMPLERATE, value); aacEncoder_SetParam(hAacEncoder, AACENC_CHANNELMODE, value); \endcode Beyond that is an internal auto mode which preinitizializes the ::AACENC_BITRATE parameter if the parameter was not set from extern. The bitrate depends on the number of effective channels and sampling rate and is determined as follows. \code AAC-LC (AOT_AAC_LC): 1.5 bits per sample HE-AAC (AOT_SBR): 0.625 bits per sample (dualrate sbr) HE-AAC (AOT_SBR): 1.125 bits per sample (downsampled sbr) HE-AAC v2 (AOT_PS): 0.5 bits per sample \endcode \subsection channelMode Channel Mode Configuration The input audio data is described with the ::AACENC_CHANNELMODE parameter in the aacEncoder_SetParam() call. It is not possible to use the encoder instance with a 'number of input channels' argument. Instead, the channelMode must be set as follows. \code aacEncoder_SetParam(hAacEncoder, AACENC_CHANNELMODE, value); \endcode The parameter is specified in ::CHANNEL_MODE and can be mapped from the number of input channels in the following way. \code CHANNEL_MODE chMode = MODE_INVALID; switch (nChannels) { case 1: chMode = MODE_1; break; case 2: chMode = MODE_2; break; case 3: chMode = MODE_1_2; break; case 4: chMode = MODE_1_2_1; break; case 5: chMode = MODE_1_2_2; break; case 6: chMode = MODE_1_2_2_1; break; case 7: chMode = MODE_6_1; break; case 8: chMode = MODE_7_1_BACK; break; default: chMode = MODE_INVALID; } return chMode; \endcode \subsection peakbitrate Peak Bitrate Configuration In AAC, the default bitreservoir configuration depends on the chosen bitrate per frame and the number of effective channels. The size can be determined as below. \f[ bitreservoir = nEffChannels*6144 - (bitrate*framelength/samplerate) \f] Due to audio quality concerns it is not recommended to change the bitreservoir size to a lower value than the default setting! However, for minimizing the delay for streaming applications or for achieving a constant size of the bitstream packages in each frame, it may be necessaray to limit the maximum bits per frame size. This can be done with the ::AACENC_PEAK_BITRATE parameter. \code aacEncoder_SetParam(hAacEncoder, AACENC_PEAK_BITRATE, value); \endcode To achieve acceptable audio quality with a reduced bitreservoir size setting at least 1000 bits per audio channel is recommended. For a multichannel audio file with 5.1 channels the bitreservoir reduced to 5000 bits results in acceptable audio quality. \subsection vbrmode Variable Bitrate Mode The variable bitrate (VBR) mode coding adapts the bit consumption to the psychoacoustic requirements of the signal. The encoder ignores the user-defined bit rate and selects a suitable pre-defined configuration based on the provided AOT. The VBR mode 1 is tuned for HE-AACv2, for VBR mode 2, HE-AACv1 should be used. VBR modes 3-5 should be used with Low-Complexity AAC. When encoding AAC-ELD, the best mode is selected automatically. The bitrates given in the table are averages over time and different encoder settings. They strongly depend on the type of audio signal. The VBR configurations can be adjusted with the ::AACENC_BITRATEMODE encoder parameter. \verbatim ----------------------------------------------- VBR_MODE | Approx. Bitrate in kbps for stereo | AAC-LC | AAC-ELD ----------+---------------+-------------------- VBR_1 | 32 (HE-AACv2) | 48 VBR_2 | 72 (HE-AACv1) | 56 VBR_3 | 112 | 72 VBR_4 | 148 | 148 VBR_5 | 228 | 224 -------------------------------------------- \endverbatim Note that these figures are valid for stereo encoding only. VBR modes 2-5 will yield much lower bit rates when encoding single-channel input. For configurations which are making use of downmix modules the AAC core channels respectively downmix channels shall be considered. \subsection encQual Audio Quality Considerations The default encoder configuration is suggested to be used. Encoder tools such as TNS and PNS are activated by default and are internally controlled (see \ref BEHAVIOUR_TOOLS). There is an additional quality parameter called ::AACENC_AFTERBURNER. In the default configuration this quality switch is deactivated because it would cause a workload increase which might be significant. If workload is not an issue in the application we recommended to activate this feature. \code aacEncoder_SetParam(hAacEncoder, AACENC_AFTERBURNER, 0/1); \endcode \subsection encELD ELD Auto Configuration Mode For ELD configuration a so called auto configurator is available which configures SBR and the SBR ratio by itself. The configurator is used when the encoder parameter ::AACENC_SBR_MODE and ::AACENC_SBR_RATIO are not set explicitly. Based on sampling rate and chosen bitrate a reasonable SBR configuration will be used. \verbatim ------------------------------------------------------------------ Sampling Rate | Total Bitrate | No. of | SBR | SBR Ratio [kHz] | [bit/s] | Chan | | | | | | ---------------+-----------------+--------+-----+----------------- ]min, 16[ | min - max | 1 | off | --- ---------------+-----------------+--------------+----------------- [16] | min - 27999 | 1 | on | downsampled SBR | 28000 - max | 1 | off | --- ---------------+-----------------+--------------+----------------- ]16 - 24] | min - 39999 | 1 | on | downsampled SBR | 40000 - max | 1 | off | --- ---------------+-----------------+--------------+----------------- ]24 - 32] | min - 27999 | 1 | on | dualrate SBR | 28000 - 55999 | 1 | on | downsampled SBR | 56000 - max | 1 | off | --- ---------------+-----------------+--------------+----------------- ]32 - 44.1] | min - 63999 | 1 | on | dualrate SBR | 64000 - max | 1 | off | --- ---------------+-----------------+--------------+----------------- ]44.1 - 48] | min - 63999 | 1 | on | dualrate SBR | 64000 - max | 1 | off | --- | | | | ---------------+-----------------+--------+-----+----------------- ]min, 16[ | min - max | 2 | off | --- ---------------+-----------------+--------------+----------------- [16] | min - 31999 | 2 | on | downsampled SBR | 32000 - 63999 | 2 | on | downsampled SBR | 64000 - max | 2 | off | --- ---------------+-----------------+--------------+----------------- ]16 - 24] | min - 47999 | 2 | on | downsampled SBR | 48000 - 79999 | 2 | on | downsampled SBR | 80000 - max | 2 | off | --- ---------------+-----------------+--------------+----------------- ]24 - 32] | min - 31999 | 2 | on | dualrate SBR | 32000 - 67999 | 2 | on | dualrate SBR | 68000 - 95999 | 2 | on | downsampled SBR | 96000 - max | 2 | off | --- ---------------+-----------------+--------------+----------------- ]32 - 44.1] | min - 43999 | 2 | on | dualrate SBR | 44000 - 127999 | 2 | on | dualrate SBR | 128000 - max | 2 | off | --- ---------------+-----------------+--------------+----------------- ]44.1 - 48] | min - 43999 | 2 | on | dualrate SBR | 44000 - 127999 | 2 | on | dualrate SBR | 128000 - max | 2 | off | --- | | | ------------------------------------------------------------------ \endverbatim \subsection encDsELD Reduced Delay (Downscaled) Mode The downscaled mode of AAC-ELD reduces the algorithmic delay of AAC-ELD by virtually increasing the sampling rate. When using the downscaled mode, the bitrate should be increased for keeping the same audio quality level. For common signals, the bitrate should be increased by 25% for a downscale factor of 2. Currently, downscaling factors 2 and 4 are supported. To enable the downscaled mode in the encoder, the framelength parameter AACENC_GRANULE_LENGTH must be set accordingly to 256 or 240 for a downscale factor of 2 or 128 or 120 for a downscale factor of 4. The default values of 512 or 480 mean that no downscaling is applied. \code aacEncoder_SetParam(hAacEncoder, AACENC_GRANULE_LENGTH, 256); aacEncoder_SetParam(hAacEncoder, AACENC_GRANULE_LENGTH, 128); \endcode Downscaled bitstreams are fully backwards compatible. However, the legacy decoder needs to support high sample rate, e.g. 96kHz. The signaled sampling rate is multiplied by the downscale factor. Although not required, downscaling should be applied when decoding downscaled bitstreams. It reduces CPU workload and the output will have the same sampling rate as the input. In an ideal configuration both encoder and decoder should run with the same downscale factor. The following table shows approximate filter bank delays in ms for common sampling rates(sr) at framesize(fs), and downscale factor(dsf), based on this formula: \f[ 1000 * fs / (dsf * sr) \f] \verbatim -------------------------------------- | 512/2 | 512/4 | 480/2 | 480/4 ------+-------+-------+-------+------- 22050 | 17.41 | 8.71 | 16.33 | 8.16 32000 | 12.00 | 6.00 | 11.25 | 5.62 44100 | 8.71 | 4.35 | 8.16 | 4.08 48000 | 8.00 | 4.00 | 7.50 | 3.75 -------------------------------------- \endverbatim \section audiochCfg Audio Channel Configuration The MPEG standard refers often to the so-called Channel Configuration. This Channel Configuration is used for a fixed Channel Mapping. The configurations 1-7 and 11,12,14 are predefined in MPEG standard and used for implicit signalling within the encoded bitstream. For user defined Configurations the Channel Configuration is set to 0 and the Channel Mapping must be explecitly described with an appropriate Program Config Element. The present Encoder implementation does not allow the user to configure this Channel Configuration from extern. The Encoder implementation supports fixed Channel Modes which are mapped to Channel Configuration as follow. \verbatim ---------------------------------------------------------------------------------------- ChannelMode | ChCfg | Height | front_El | side_El | back_El | lfe_El -----------------------+-------+--------+---------------+----------+----------+--------- MODE_1 | 1 | NORM | SCE | | | MODE_2 | 2 | NORM | CPE | | | MODE_1_2 | 3 | NORM | SCE, CPE | | | MODE_1_2_1 | 4 | NORM | SCE, CPE | | SCE | MODE_1_2_2 | 5 | NORM | SCE, CPE | | CPE | MODE_1_2_2_1 | 6 | NORM | SCE, CPE | | CPE | LFE MODE_1_2_2_2_1 | 7 | NORM | SCE, CPE, CPE | | CPE | LFE MODE_6_1 | 11 | NORM | SCE, CPE | | CPE, SCE | LFE MODE_7_1_BACK | 12 | NORM | SCE, CPE | | CPE, CPE | LFE -----------------------+-------+--------+---------------+----------+----------+--------- MODE_7_1_TOP_FRONT | 14 | NORM | SCE, CPE | | CPE | LFE | | TOP | CPE | | | -----------------------+-------+--------+---------------+----------+----------+--------- MODE_7_1_REAR_SURROUND | 0 | NORM | SCE, CPE | | CPE, CPE | LFE MODE_7_1_FRONT_CENTER | 0 | NORM | SCE, CPE, CPE | | CPE | LFE ---------------------------------------------------------------------------------------- - NORM: Normal Height Layer. - TOP: Top Height Layer. - BTM: Bottom Height Layer. - SCE: Single Channel Element. - CPE: Channel Pair. - LFE: Low Frequency Element. \endverbatim The Table describes all fixed Channel Elements for each Channel Mode which are assigned to a speaker arrangement. The arrangement includes front, side, back and lfe Audio Channel Elements in the normal height layer, possibly followed by front, side, and back elements in the top and bottom layer (Channel Configuration 14). \n This mapping of Audio Channel Elements is defined in MPEG standard for Channel Config 1-7 and 11,12,14.\n In case of Channel Config 0 or writing matrix mixdown coefficients, the encoder enables the writing of Program Config Element itself as described in \ref encPCE. The configuration used in Program Config Element refers to the denoted Table.\n Beside the Channel Element assignment the Channel Modes are resposible for audio input data channel mapping. The Channel Mapping of the audio data depends on the selected ::AACENC_CHANNELORDER which can be MPEG or WAV like order.\n Following table describes the complete channel mapping for both Channel Order configurations. \verbatim --------------------------------------------------------------------------------------- ChannelMode | MPEG-Channelorder | WAV-Channelorder -----------------------+---+---+---+---+---+---+---+---+---+---+---+---+---+---+---+--- MODE_1 | 0 | | | | | | | | 0 | | | | | | | MODE_2 | 0 | 1 | | | | | | | 0 | 1 | | | | | | MODE_1_2 | 0 | 1 | 2 | | | | | | 2 | 0 | 1 | | | | | MODE_1_2_1 | 0 | 1 | 2 | 3 | | | | | 2 | 0 | 1 | 3 | | | | MODE_1_2_2 | 0 | 1 | 2 | 3 | 4 | | | | 2 | 0 | 1 | 3 | 4 | | | MODE_1_2_2_1 | 0 | 1 | 2 | 3 | 4 | 5 | | | 2 | 0 | 1 | 4 | 5 | 3 | | MODE_1_2_2_2_1 | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 2 | 6 | 7 | 0 | 1 | 4 | 5 | 3 MODE_6_1 | 0 | 1 | 2 | 3 | 4 | 5 | 6 | | 2 | 0 | 1 | 4 | 5 | 6 | 3 | MODE_7_1_BACK | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 2 | 0 | 1 | 6 | 7 | 4 | 5 | 3 MODE_7_1_TOP_FRONT | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 2 | 0 | 1 | 4 | 5 | 3 | 6 | 7 -----------------------+---+---+---+---+---+---+---+---+---+---+---+---+---+---+---+--- MODE_7_1_REAR_SURROUND | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 2 | 0 | 1 | 6 | 7 | 4 | 5 | 3 MODE_7_1_FRONT_CENTER | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 2 | 6 | 7 | 0 | 1 | 4 | 5 | 3 --------------------------------------------------------------------------------------- \endverbatim The denoted mapping is important for correct audio channel assignment when using MPEG or WAV ordering. The incoming audio channels are distributed MPEG like starting at the front channels and ending at the back channels. The distribution is used as described in Table concering Channel Config and fix channel elements. Please see the following example for clarification. \verbatim Example: MODE_1_2_2_1 - WAV-Channelorder 5.1 ------------------------------------------ Input Channel | Coder Channel --------------------+--------------------- 2 (front center) | 0 (SCE channel) 0 (left center) | 1 (1st of 1st CPE) 1 (right center) | 2 (2nd of 1st CPE) 4 (left surround) | 3 (1st of 2nd CPE) 5 (right surround) | 4 (2nd of 2nd CPE) 3 (LFE) | 5 (LFE) ------------------------------------------ \endverbatim \section suppBitrates Supported Bitrates The FDK AAC Encoder provides a wide range of supported bitrates. The minimum and maximum allowed bitrate depends on the Audio Object Type. For AAC-LC the minimum bitrate is the bitrate that is required to write the most basic and minimal valid bitstream. It consists of the bitstream format header information and other static/mandatory information within the AAC payload. The maximum AAC framesize allowed by the MPEG-4 standard determines the maximum allowed bitrate for AAC-LC. For HE-AAC and HE-AAC v2 a library internal look-up table is used. A good working point in terms of audio quality, sampling rate and bitrate, is at 1 to 1.5 bits/audio sample for AAC-LC, 0.625 bits/audio sample for dualrate HE-AAC, 1.125 bits/audio sample for downsampled HE-AAC and 0.5 bits/audio sample for HE-AAC v2. For example for one channel with a sampling frequency of 48 kHz, the range from 48 kbit/s to 72 kbit/s achieves reasonable audio quality for AAC-LC. For HE-AAC and HE-AAC v2 the lowest possible audio input sampling frequency is 16 kHz because then the AAC-LC core encoder operates in dual rate mode at its lowest possible sampling frequency, which is 8 kHz. HE-AAC v2 requires stereo input audio data. Please note that in HE-AAC or HE-AAC v2 mode the encoder supports much higher bitrates than are appropriate for HE-AAC or HE-AAC v2. For example, at a bitrate of more than 64 kbit/s for a stereo audio signal at 44.1 kHz it usually makes sense to use AAC-LC, which will produce better audio quality at that bitrate than HE-AAC or HE-AAC v2. \section reommendedConfig Recommended Sampling Rate and Bitrate Combinations The following table provides an overview of recommended encoder configuration parameters which we determined by virtue of numerous listening tests. \subsection reommendedConfigLC AAC-LC, HE-AAC, HE-AACv2 in Dualrate SBR mode. \verbatim ----------------------------------------------------------------------------------- Audio Object Type | Bit Rate Range | Supported | Preferred | No. of | [bit/s] | Sampling Rates | Sampl. | Chan. | | [kHz] | Rate | | | | [kHz] | -------------------+------------------+-----------------------+------------+------- AAC LC + SBR + PS | 8000 - 11999 | 22.05, 24.00 | 24.00 | 2 AAC LC + SBR + PS | 12000 - 17999 | 32.00 | 32.00 | 2 AAC LC + SBR + PS | 18000 - 39999 | 32.00, 44.10, 48.00 | 44.10 | 2 AAC LC + SBR + PS | 40000 - 64000 | 32.00, 44.10, 48.00 | 48.00 | 2 -------------------+------------------+-----------------------+------------+------- AAC LC + SBR | 8000 - 11999 | 22.05, 24.00 | 24.00 | 1 AAC LC + SBR | 12000 - 17999 | 32.00 | 32.00 | 1 AAC LC + SBR | 18000 - 39999 | 32.00, 44.10, 48.00 | 44.10 | 1 AAC LC + SBR | 40000 - 64000 | 32.00, 44.10, 48.00 | 48.00 | 1 -------------------+------------------+-----------------------+------------+------- AAC LC + SBR | 16000 - 27999 | 32.00, 44.10, 48.00 | 32.00 | 2 AAC LC + SBR | 28000 - 63999 | 32.00, 44.10, 48.00 | 44.10 | 2 AAC LC + SBR | 64000 - 128000 | 32.00, 44.10, 48.00 | 48.00 | 2 -------------------+------------------+-----------------------+------------+------- AAC LC + SBR | 64000 - 69999 | 32.00, 44.10, 48.00 | 32.00 | 5, 5.1 AAC LC + SBR | 70000 - 239999 | 32.00, 44.10, 48.00 | 44.10 | 5, 5.1 AAC LC + SBR | 240000 - 319999 | 32.00, 44.10, 48.00 | 48.00 | 5, 5.1 -------------------+------------------+-----------------------+------------+------- AAC LC | 8000 - 15999 | 11.025, 12.00, 16.00 | 12.00 | 1 AAC LC | 16000 - 23999 | 16.00 | 16.00 | 1 AAC LC | 24000 - 31999 | 16.00, 22.05, 24.00 | 24.00 | 1 AAC LC | 32000 - 55999 | 32.00 | 32.00 | 1 AAC LC | 56000 - 160000 | 32.00, 44.10, 48.00 | 44.10 | 1 AAC LC | 160001 - 288000 | 48.00 | 48.00 | 1 -------------------+------------------+-----------------------+------------+------- AAC LC | 16000 - 23999 | 11.025, 12.00, 16.00 | 12.00 | 2 AAC LC | 24000 - 31999 | 16.00 | 16.00 | 2 AAC LC | 32000 - 39999 | 16.00, 22.05, 24.00 | 22.05 | 2 AAC LC | 40000 - 95999 | 32.00 | 32.00 | 2 AAC LC | 96000 - 111999 | 32.00, 44.10, 48.00 | 32.00 | 2 AAC LC | 112000 - 320001 | 32.00, 44.10, 48.00 | 44.10 | 2 AAC LC | 320002 - 576000 | 48.00 | 48.00 | 2 -------------------+------------------+-----------------------+------------+------- AAC LC | 160000 - 239999 | 32.00 | 32.00 | 5, 5.1 AAC LC | 240000 - 279999 | 32.00, 44.10, 48.00 | 32.00 | 5, 5.1 AAC LC | 280000 - 800000 | 32.00, 44.10, 48.00 | 44.10 | 5, 5.1 ----------------------------------------------------------------------------------- \endverbatim \n \subsection reommendedConfigLD AAC-LD, AAC-ELD, AAC-ELD with SBR in Dualrate SBR mode. Unlike to HE-AAC configuration the SBR is not covered by ELD audio object type and needs to be enabled explicitly. Use ::AACENC_SBR_MODE to configure SBR and its samplingrate ratio with ::AACENC_SBR_RATIO parameter. \verbatim ----------------------------------------------------------------------------------- Audio Object Type | Bit Rate Range | Supported | Preferred | No. of | [bit/s] | Sampling Rates | Sampl. | Chan. | | [kHz] | Rate | | | | [kHz] | -------------------+------------------+-----------------------+------------+------- ELD + SBR | 18000 - 24999 | 32.00 - 44.10 | 32.00 | 1 ELD + SBR | 25000 - 31999 | 32.00 - 48.00 | 32.00 | 1 ELD + SBR | 32000 - 64000 | 32.00 - 48.00 | 48.00 | 1 -------------------+------------------+-----------------------+------------+------- ELD + SBR | 32000 - 51999 | 32.00 - 48.00 | 44.10 | 2 ELD + SBR | 52000 - 128000 | 32.00 - 48.00 | 48.00 | 2 -------------------+------------------+-----------------------+------------+------- ELD + SBR | 78000 - 160000 | 32.00 - 48.00 | 48.00 | 3 -------------------+------------------+-----------------------+------------+------- ELD + SBR | 104000 - 212000 | 32.00 - 48.00 | 48.00 | 4 -------------------+------------------+-----------------------+------------+------- ELD + SBR | 130000 - 246000 | 32.00 - 48.00 | 48.00 | 5, 5.1 -------------------+------------------+-----------------------+------------+------- LD, ELD | 16000 - 19999 | 16.00 - 24.00 | 16.00 | 1 LD, ELD | 20000 - 39999 | 16.00 - 32.00 | 24.00 | 1 LD, ELD | 40000 - 49999 | 22.05 - 32.00 | 32.00 | 1 LD, ELD | 50000 - 61999 | 24.00 - 44.10 | 32.00 | 1 LD, ELD | 62000 - 84999 | 32.00 - 48.00 | 44.10 | 1 LD, ELD | 85000 - 192000 | 44.10 - 48.00 | 48.00 | 1 -------------------+------------------+-----------------------+------------+------- LD, ELD | 64000 - 75999 | 24.00 - 32.00 | 32.00 | 2 LD, ELD | 76000 - 97999 | 24.00 - 44.10 | 32.00 | 2 LD, ELD | 98000 - 135999 | 32.00 - 48.00 | 44.10 | 2 LD, ELD | 136000 - 384000 | 44.10 - 48.00 | 48.00 | 2 -------------------+------------------+-----------------------+------------+------- LD, ELD | 96000 - 113999 | 24.00 - 32.00 | 32.00 | 3 LD, ELD | 114000 - 146999 | 24.00 - 44.10 | 32.00 | 3 LD, ELD | 147000 - 203999 | 32.00 - 48.00 | 44.10 | 3 LD, ELD | 204000 - 576000 | 44.10 - 48.00 | 48.00 | 3 -------------------+------------------+-----------------------+------------+------- LD, ELD | 128000 - 151999 | 24.00 - 32.00 | 32.00 | 4 LD, ELD | 152000 - 195999 | 24.00 - 44.10 | 32.00 | 4 LD, ELD | 196000 - 271999 | 32.00 - 48.00 | 44.10 | 4 LD, ELD | 272000 - 768000 | 44.10 - 48.00 | 48.00 | 4 -------------------+------------------+-----------------------+------------+------- LD, ELD | 160000 - 189999 | 24.00 - 32.00 | 32.00 | 5, 5.1 LD, ELD | 190000 - 244999 | 24.00 - 44.10 | 32.00 | 5, 5.1 LD, ELD | 245000 - 339999 | 32.00 - 48.00 | 44.10 | 5, 5.1 LD, ELD | 340000 - 960000 | 44.10 - 48.00 | 48.00 | 5, 5.1 ----------------------------------------------------------------------------------- \endverbatim \n \subsection reommendedConfigELD AAC-ELD with SBR in Downsampled SBR mode. \verbatim ----------------------------------------------------------------------------------- Audio Object Type | Bit Rate Range | Supported | Preferred | No. of | [bit/s] | Sampling Rates | Sampl. | Chan. | | [kHz] | Rate | | | | [kHz] | -------------------+------------------+-----------------------+------------+------- ELD + SBR | 18000 - 24999 | 16.00 - 22.05 | 22.05 | 1 (downsampled SBR) | 25000 - 31999 | 16.00 - 24.00 | 24.00 | 1 | 32000 - 47999 | 22.05 - 32.00 | 32.00 | 1 | 48000 - 64000 | 22.05 - 48.00 | 32.00 | 1 -------------------+------------------+-----------------------+------------+------- ELD + SBR | 32000 - 51999 | 16.00 - 24.00 | 24.00 | 2 (downsampled SBR) | 52000 - 59999 | 22.05 - 24.00 | 24.00 | 2 | 60000 - 95999 | 22.05 - 32.00 | 32.00 | 2 | 96000 - 128000 | 22.05 - 48.00 | 32.00 | 2 -------------------+------------------+-----------------------+------------+------- ELD + SBR | 78000 - 99999 | 22.05 - 24.00 | 24.00 | 3 (downsampled SBR) | 100000 - 143999 | 22.05 - 32.00 | 32.00 | 3 | 144000 - 159999 | 22.05 - 48.00 | 32.00 | 3 | 160000 - 192000 | 32.00 - 48.00 | 32.00 | 3 -------------------+------------------+-----------------------+------------+------- ELD + SBR | 104000 - 149999 | 22.05 - 24.00 | 24.00 | 4 (downsampled SBR) | 150000 - 191999 | 22.05 - 32.00 | 32.00 | 4 | 192000 - 211999 | 22.05 - 48.00 | 32.00 | 4 | 212000 - 256000 | 32.00 - 48.00 | 32.00 | 4 -------------------+------------------+-----------------------+------------+------- ELD + SBR | 130000 - 171999 | 22.05 - 24.00 | 24.00 | 5, 5.1 (downsampled SBR) | 172000 - 239999 | 22.05 - 32.00 | 32.00 | 5, 5.1 | 240000 - 320000 | 32.00 - 48.00 | 32.00 | 5, 5.1 ----------------------------------------------------------------------------------- \endverbatim \n \subsection reommendedConfigELDv2 AAC-ELD v2, AAC-ELD v2 with SBR. The ELD v2 212 configuration must be configured explicitly with ::AACENC_CHANNELMODE parameter according MODE_212 value. SBR can be configured separately through ::AACENC_SBR_MODE and ::AACENC_SBR_RATIO parameter. Following configurations shall apply to both framelengths 480 and 512. For ELD v2 configuration without SBR and framelength 480 the supported sampling rate is restricted to the range from 16 kHz up to 24 kHz. \verbatim ----------------------------------------------------------------------------------- Audio Object Type | Bit Rate Range | Supported | Preferred | No. of | [bit/s] | Sampling Rates | Sampl. | Chan. | | [kHz] | Rate | | | | [kHz] | -------------------+------------------+-----------------------+------------+------- ELD-212 | 16000 - 19999 | 16.00 - 24.00 | 16.00 | 2 (without SBR) | 20000 - 39999 | 16.00 - 32.00 | 24.00 | 2 | 40000 - 49999 | 22.05 - 32.00 | 32.00 | 2 | 50000 - 61999 | 24.00 - 44.10 | 32.00 | 2 | 62000 - 84999 | 32.00 - 48.00 | 44.10 | 2 | 85000 - 192000 | 44.10 - 48.00 | 48.00 | 2 -------------------+------------------+-----------------------+------------+------- ELD-212 + SBR | 18000 - 20999 | 32.00 | 32.00 | 2 (dualrate SBR) | 21000 - 25999 | 32.00 - 44.10 | 32.00 | 2 | 26000 - 31999 | 32.00 - 48.00 | 44.10 | 2 | 32000 - 64000 | 32.00 - 48.00 | 48.00 | 2 -------------------+------------------+-----------------------+------------+------- ELD-212 + SBR | 18000 - 19999 | 16.00 - 22.05 | 22.05 | 2 (downsampled SBR) | 20000 - 24999 | 16.00 - 24.00 | 22.05 | 2 | 25000 - 31999 | 16.00 - 24.00 | 24.00 | 2 | 32000 - 64000 | 24.00 - 24.00 | 24.00 | 2 -------------------+------------------+-----------------------+------------+------- \endverbatim \n \page ENCODERBEHAVIOUR Encoder Behaviour \section BEHAVIOUR_BANDWIDTH Bandwidth The FDK AAC encoder usually does not use the full frequency range of the input signal, but restricts the bandwidth according to certain library-internal settings. They can be changed in the table "bandWidthTable" in the file bandwidth.cpp (if available). The encoder API provides the ::AACENC_BANDWIDTH parameter to adjust the bandwidth explicitly. \code aacEncoder_SetParam(hAacEncoder, AACENC_BANDWIDTH, value); \endcode However it is not recommended to change these settings, because they are based on numerous listening tests and careful tweaks to ensure the best overall encoding quality. Also, the maximum bandwidth that can be set manually by the user is 20kHz or fs/2, whichever value is smaller. Theoretically a signal of for example 48 kHz can contain frequencies up to 24 kHz, but to use this full range in an audio encoder usually does not make sense. Usually the encoder has a very limited amount of bits to spend (typically 128 kbit/s for stereo 48 kHz content) and to allow full range bandwidth would waste a lot of these bits for frequencies the human ear is hardly able to perceive anyway, if at all. Hence it is wise to use the available bits for the really important frequency range and just skip the rest. At lower bitrates (e. g. <= 80 kbit/s for stereo 48 kHz content) the encoder will choose an even smaller bandwidth, because an encoded signal with smaller bandwidth and hence less artifacts sounds better than a signal with higher bandwidth but then more coding artefacts across all frequencies. These artefacts would occur if small bitrates and high bandwidths are chosen because the available bits are just not enough to encode all frequencies well. Unfortunately some people evaluate encoding quality based on possible bandwidth as well, but it is a double-edged sword considering the trade-off described above. Another aspect is workload consumption. The higher the allowed bandwidth, the more frequency lines have to be processed, which in turn increases the workload. \section FRAMESIZES_AND_BIT_RESERVOIR Frame Sizes & Bit Reservoir For AAC there is a difference between constant bit rate and constant frame length due to the so-called bit reservoir technique, which allows the encoder to use less bits in an AAC frame for those audio signal sections which are easy to encode, and then spend them at a later point in time for more complex audio sections. The extent to which this "bit exchange" is done is limited to allow for reliable and relatively low delay real time streaming. Therefore, for AAC-ELD, the bitreservoir is limited. It varies between 500 and 4000 bits/frame, depending on the bitrate/channel. - For a bitrate of 12kbps/channel and below, the AAC-ELD bitreservoir is 500 bits/frame. - For a bitrate of 70kbps/channel and above, the AAC-ELD bitreservoir is 4000 bits/frame. - Between 12kbps/channel and 70kbps/channel, the AAC-ELD bitrervoir is increased linearly. - For AAC-LC, the bitrate is only limited by the maximum AAC frame length. It is, regardless of the available bit reservoir, defined as 6144 bits per channel. Over a longer period in time the bitrate will be constant in the AAC constant bitrate mode, e.g. for ISDN transmission. This means that in AAC each bitstream frame will in general have a different length in bytes but over time it will reach the target bitrate. One could also make an MPEG compliant AAC encoder which always produces constant length packages for each AAC frame, but the audio quality would be considerably worse since the bit reservoir technique would have to be switched off completely. A higher bit rate would have to be used to get the same audio quality as with an enabled bit reservoir. For mp3 by the way, the same bit reservoir technique exists, but there each bit stream frame has a constant length for a given bit rate (ignoring the padding byte). In mp3 there is a so-called "back pointer" which tells the decoder which bits belong to the current mp3 frame - and in general some or many bits have been transmitted in an earlier mp3 frame. Basically this leads to the same "bit exchange between mp3 frames" as in AAC but with virtually constant length frames. This variable frame length at "constant bit rate" is not something special in this Fraunhofer IIS AAC encoder. AAC has been designed in that way. \subsection BEHAVIOUR_ESTIM_AVG_FRAMESIZES Estimating Average Frame Sizes A HE-AAC v1 or v2 audio frame contains 2048 PCM samples per channel. The number of HE-AAC frames \f$N\_FRAMES\f$ per second at 44.1 kHz is: \f[ N\_FRAMES = 44100 / 2048 = 21.5332 \f] At a bit rate of 8 kbps the average number of bits per frame \f$N\_BITS\_PER\_FRAME\f$ is: \f[ N\_BITS\_PER\_FRAME = 8000 / 21.5332 = 371.52 \f] which is about 46.44 bytes per encoded frame. At a bit rate of 32 kbps, which is quite high for single channel HE-AAC v1, it is: \f[ N\_BITS\_PER\_FRAME = 32000 / 21.5332 = 1486 \f] which is about 185.76 bytes per encoded frame. These bits/frame figures are average figures where each AAC frame generally has a different size in bytes. To calculate the same for AAC-LC just use 1024 instead of 2048 PCM samples per frame and channel. For AAC-LD/ELD it is either 480 or 512 PCM samples per frame and channel. \section BEHAVIOUR_TOOLS Encoder Tools The AAC encoder supports TNS, PNS, MS, Intensity and activates these tools depending on the audio signal and the encoder configuration (i.e. bitrate or AOT). It is not required to configure these tools manually. PNS improves encoding quality only for certain bitrates. Therefore it makes sense to activate PNS only for these bitrates and save the processing power required for PNS (about 10 % of the encoder) when using other bitrates. This is done automatically inside the encoder library. PNS is disabled inside the encoder library if an MPEG-2 AOT is choosen since PNS is an MPEG-4 AAC feature. If SBR is activated, the encoder automatically deactivates PNS internally. If TNS is disabled but PNS is allowed, the encoder deactivates PNS calculation internally. */ #ifndef AACENC_LIB_H #define AACENC_LIB_H #include "machine_type.h" #include "FDK_audio.h" #define AACENCODER_LIB_VL0 4 #define AACENCODER_LIB_VL1 0 #define AACENCODER_LIB_VL2 1 /** * AAC encoder error codes. */ typedef enum { AACENC_OK = 0x0000, /*!< No error happened. All fine. */ AACENC_INVALID_HANDLE = 0x0020, /*!< Handle passed to function call was invalid. */ AACENC_MEMORY_ERROR = 0x0021, /*!< Memory allocation failed. */ AACENC_UNSUPPORTED_PARAMETER = 0x0022, /*!< Parameter not available. */ AACENC_INVALID_CONFIG = 0x0023, /*!< Configuration not provided. */ AACENC_INIT_ERROR = 0x0040, /*!< General initialization error. */ AACENC_INIT_AAC_ERROR = 0x0041, /*!< AAC library initialization error. */ AACENC_INIT_SBR_ERROR = 0x0042, /*!< SBR library initialization error. */ AACENC_INIT_TP_ERROR = 0x0043, /*!< Transport library initialization error. */ AACENC_INIT_META_ERROR = 0x0044, /*!< Meta data library initialization error. */ AACENC_INIT_MPS_ERROR = 0x0045, /*!< MPS library initialization error. */ AACENC_ENCODE_ERROR = 0x0060, /*!< The encoding process was interrupted by an unexpected error. */ AACENC_ENCODE_EOF = 0x0080 /*!< End of file reached. */ } AACENC_ERROR; /** * AAC encoder buffer descriptors identifier. * This identifier are used within buffer descriptors * AACENC_BufDesc::bufferIdentifiers. */ typedef enum { /* Input buffer identifier. */ IN_AUDIO_DATA = 0, /*!< Audio input buffer, interleaved INT_PCM samples. */ IN_ANCILLRY_DATA = 1, /*!< Ancillary data to be embedded into bitstream. */ IN_METADATA_SETUP = 2, /*!< Setup structure for embedding meta data. */ /* Output buffer identifier. */ OUT_BITSTREAM_DATA = 3, /*!< Buffer holds bitstream output data. */ OUT_AU_SIZES = 4 /*!< Buffer contains sizes of each access unit. This information is necessary for superframing. */ } AACENC_BufferIdentifier; /** * AAC encoder handle. */ typedef struct AACENCODER *HANDLE_AACENCODER; /** * Provides some info about the encoder configuration. */ typedef struct { UINT maxOutBufBytes; /*!< Maximum number of encoder bitstream bytes within one frame. Size depends on maximum number of supported channels in encoder instance. */ UINT maxAncBytes; /*!< Maximum number of ancillary data bytes which can be inserted into bitstream within one frame. */ UINT inBufFillLevel; /*!< Internal input buffer fill level in samples per channel. This parameter will automatically be cleared if samplingrate or channel(Mode/Order) changes. */ UINT inputChannels; /*!< Number of input channels expected in encoding process. */ UINT frameLength; /*!< Amount of input audio samples consumed each frame per channel, depending on audio object type configuration. */ UINT nDelay; /*!< Codec delay in PCM samples/channel. Depends on framelength and AOT. Does not include framing delay for filling up encoder PCM input buffer. */ UINT nDelayCore; /*!< Codec delay in PCM samples/channel, w/o delay caused by the decoder SBR module. This delay is needed to correctly write edit lists for gapless playback. The decoder may not know how much delay is introdcued by SBR, since it may not know if SBR is active at all (implicit signaling), therefore the deocder must take into account any delay caused by the SBR module. */ UCHAR confBuf[64]; /*!< Configuration buffer in binary format as an AudioSpecificConfig or StreamMuxConfig according to the selected transport type. */ UINT confSize; /*!< Number of valid bytes in confBuf. */ } AACENC_InfoStruct; /** * Describes the input and output buffers for an aacEncEncode() call. */ typedef struct { INT numBufs; /*!< Number of buffers. */ void **bufs; /*!< Pointer to vector containing buffer addresses. */ INT *bufferIdentifiers; /*!< Identifier of each buffer element. See ::AACENC_BufferIdentifier. */ INT *bufSizes; /*!< Size of each buffer in 8-bit bytes. */ INT *bufElSizes; /*!< Size of each buffer element in bytes. */ } AACENC_BufDesc; /** * Defines the input arguments for an aacEncEncode() call. */ typedef struct { INT numInSamples; /*!< Number of valid input audio samples (multiple of input channels). */ INT numAncBytes; /*!< Number of ancillary data bytes to be encoded. */ } AACENC_InArgs; /** * Defines the output arguments for an aacEncEncode() call. */ typedef struct { INT numOutBytes; /*!< Number of valid bitstream bytes generated during aacEncEncode(). */ INT numInSamples; /*!< Number of input audio samples consumed by the encoder. */ INT numAncBytes; /*!< Number of ancillary data bytes consumed by the encoder. */ INT bitResState; /*!< State of the bit reservoir in bits. */ } AACENC_OutArgs; /** * Meta Data Compression Profiles. */ typedef enum { AACENC_METADATA_DRC_NONE = 0, /*!< None. */ AACENC_METADATA_DRC_FILMSTANDARD = 1, /*!< Film standard. */ AACENC_METADATA_DRC_FILMLIGHT = 2, /*!< Film light. */ AACENC_METADATA_DRC_MUSICSTANDARD = 3, /*!< Music standard. */ AACENC_METADATA_DRC_MUSICLIGHT = 4, /*!< Music light. */ AACENC_METADATA_DRC_SPEECH = 5, /*!< Speech. */ AACENC_METADATA_DRC_NOT_PRESENT = 256 /*!< Disable writing gain factor (used for comp_profile only). */ } AACENC_METADATA_DRC_PROFILE; /** * Meta Data setup structure. */ typedef struct { AACENC_METADATA_DRC_PROFILE drc_profile; /*!< MPEG DRC compression profile. See ::AACENC_METADATA_DRC_PROFILE. */ AACENC_METADATA_DRC_PROFILE comp_profile; /*!< ETSI heavy compression profile. See ::AACENC_METADATA_DRC_PROFILE. */ INT drc_TargetRefLevel; /*!< Used to define expected level to: Scaled with 16 bit. x*2^16. */ INT comp_TargetRefLevel; /*!< Adjust limiter to avoid overload. Scaled with 16 bit. x*2^16. */ INT prog_ref_level_present; /*!< Flag, if prog_ref_level is present */ INT prog_ref_level; /*!< Programme Reference Level = Dialogue Level: -31.75dB .. 0 dB ; stepsize: 0.25dB Scaled with 16 bit. x*2^16.*/ UCHAR PCE_mixdown_idx_present; /*!< Flag, if dmx-idx should be written in programme config element */ UCHAR ETSI_DmxLvl_present; /*!< Flag, if dmx-lvl should be written in ETSI-ancData */ SCHAR centerMixLevel; /*!< Center downmix level (0...7, according to table) */ SCHAR surroundMixLevel; /*!< Surround downmix level (0...7, according to table) */ UCHAR dolbySurroundMode; /*!< Indication for Dolby Surround Encoding Mode. - 0: Dolby Surround mode not indicated - 1: 2-ch audio part is not Dolby surround encoded - 2: 2-ch audio part is Dolby surround encoded */ UCHAR drcPresentationMode; /*!< Indicatin for DRC Presentation Mode. - 0: Presentation mode not inticated - 1: Presentation mode 1 - 2: Presentation mode 2 */ struct { /* extended ancillary data */ UCHAR extAncDataEnable; /*< Indicates if MPEG4_ext_ancillary_data() exists. - 0: No MPEG4_ext_ancillary_data(). - 1: Insert MPEG4_ext_ancillary_data(). */ UCHAR extDownmixLevelEnable; /*< Indicates if ext_downmixing_levels() exists. - 0: No ext_downmixing_levels(). - 1: Insert ext_downmixing_levels(). */ UCHAR extDownmixLevel_A; /*< Downmix level index A (0...7, according to table) */ UCHAR extDownmixLevel_B; /*< Downmix level index B (0...7, according to table) */ UCHAR dmxGainEnable; /*< Indicates if ext_downmixing_global_gains() exists. - 0: No ext_downmixing_global_gains(). - 1: Insert ext_downmixing_global_gains(). */ INT dmxGain5; /*< Gain factor for downmix to 5 channels. -15.75dB .. -15.75dB; stepsize: 0.25dB Scaled with 16 bit. x*2^16.*/ INT dmxGain2; /*< Gain factor for downmix to 2 channels. -15.75dB .. -15.75dB; stepsize: 0.25dB Scaled with 16 bit. x*2^16.*/ UCHAR lfeDmxEnable; /*< Indicates if ext_downmixing_lfe_level() exists. - 0: No ext_downmixing_lfe_level(). - 1: Insert ext_downmixing_lfe_level(). */ UCHAR lfeDmxLevel; /*< Downmix level index for LFE (0..15, according to table) */ } ExtMetaData; } AACENC_MetaData; /** * AAC encoder control flags. * * In interaction with the ::AACENC_CONTROL_STATE parameter it is possible to * get information about the internal initialization process. It is also * possible to overwrite the internal state from extern when necessary. */ typedef enum { AACENC_INIT_NONE = 0x0000, /*!< Do not trigger initialization. */ AACENC_INIT_CONFIG = 0x0001, /*!< Initialize all encoder modules configuration. */ AACENC_INIT_STATES = 0x0002, /*!< Reset all encoder modules history buffer. */ AACENC_INIT_TRANSPORT = 0x1000, /*!< Initialize transport lib with new parameters. */ AACENC_RESET_INBUFFER = 0x2000, /*!< Reset fill level of internal input buffer. */ AACENC_INIT_ALL = 0xFFFF /*!< Initialize all. */ } AACENC_CTRLFLAGS; /** * \brief AAC encoder setting parameters. * * Use aacEncoder_SetParam() function to configure, or use aacEncoder_GetParam() * function to read the internal status of the following parameters. */ typedef enum { AACENC_AOT = 0x0100, /*!< Audio object type. See ::AUDIO_OBJECT_TYPE in FDK_audio.h. - 2: MPEG-4 AAC Low Complexity. - 5: MPEG-4 AAC Low Complexity with Spectral Band Replication (HE-AAC). - 29: MPEG-4 AAC Low Complexity with Spectral Band Replication and Parametric Stereo (HE-AAC v2). This configuration can be used only with stereo input audio data. - 23: MPEG-4 AAC Low-Delay. - 39: MPEG-4 AAC Enhanced Low-Delay. Since there is no ::AUDIO_OBJECT_TYPE for ELD in combination with SBR defined, enable SBR explicitely by ::AACENC_SBR_MODE parameter. The ELD v2 212 configuration can be configured by ::AACENC_CHANNELMODE parameter. - 129: MPEG-2 AAC Low Complexity. - 132: MPEG-2 AAC Low Complexity with Spectral Band Replication (HE-AAC). Please note that the virtual MPEG-2 AOT's basically disables non-existing Perceptual Noise Substitution tool in AAC encoder and controls the MPEG_ID flag in adts header. The virtual MPEG-2 AOT doesn't prohibit specific transport formats. */ AACENC_BITRATE = 0x0101, /*!< Total encoder bitrate. This parameter is mandatory and interacts with ::AACENC_BITRATEMODE. - CBR: Bitrate in bits/second. - VBR: Variable bitrate. Bitrate argument will be ignored. See \ref suppBitrates for details. */ AACENC_BITRATEMODE = 0x0102, /*!< Bitrate mode. Configuration can be different kind of bitrate configurations: - 0: Constant bitrate, use bitrate according to ::AACENC_BITRATE. (default) Within none LD/ELD ::AUDIO_OBJECT_TYPE, the CBR mode makes use of full allowed bitreservoir. In contrast, at Low-Delay ::AUDIO_OBJECT_TYPE the bitreservoir is kept very small. - 1: Variable bitrate mode, \ref vbrmode "very low bitrate". - 2: Variable bitrate mode, \ref vbrmode "low bitrate". - 3: Variable bitrate mode, \ref vbrmode "medium bitrate". - 4: Variable bitrate mode, \ref vbrmode "high bitrate". - 5: Variable bitrate mode, \ref vbrmode "very high bitrate". */ AACENC_SAMPLERATE = 0x0103, /*!< Audio input data sampling rate. Encoder supports following sampling rates: 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 64000, 88200, 96000 */ AACENC_SBR_MODE = 0x0104, /*!< Configure SBR independently of the chosen Audio Object Type ::AUDIO_OBJECT_TYPE. This parameter is for ELD audio object type only. - -1: Use ELD SBR auto configurator (default). - 0: Disable Spectral Band Replication. - 1: Enable Spectral Band Replication. */ AACENC_GRANULE_LENGTH = 0x0105, /*!< Core encoder (AAC) audio frame length in samples: - 1024: Default configuration. - 512: Default length in LD/ELD configuration. - 480: Length in LD/ELD configuration. - 256: Length for ELD reduced delay mode (x2). - 240: Length for ELD reduced delay mode (x2). - 128: Length for ELD reduced delay mode (x4). - 120: Length for ELD reduced delay mode (x4). */ AACENC_CHANNELMODE = 0x0106, /*!< Set explicit channel mode. Channel mode must match with number of input channels. - 1-7, 11,12,14 and 33,34: MPEG channel modes supported, see ::CHANNEL_MODE in FDK_audio.h. */ AACENC_CHANNELORDER = 0x0107, /*!< Input audio data channel ordering scheme: - 0: MPEG channel ordering (e. g. 5.1: C, L, R, SL, SR, LFE). (default) - 1: WAVE file format channel ordering (e. g. 5.1: L, R, C, LFE, SL, SR). */ AACENC_SBR_RATIO = 0x0108, /*!< Controls activation of downsampled SBR. With downsampled SBR, the delay will be shorter. On the other hand, for achieving the same quality level, downsampled SBR needs more bits than dual-rate SBR. With downsampled SBR, the AAC encoder will work at the same sampling rate as the SBR encoder (single rate). Downsampled SBR is supported for AAC-ELD and HE-AACv1. - 1: Downsampled SBR (default for ELD). - 2: Dual-rate SBR (default for HE-AAC). */ AACENC_AFTERBURNER = 0x0200, /*!< This parameter controls the use of the afterburner feature. The afterburner is a type of analysis by synthesis algorithm which increases the audio quality but also the required processing power. It is recommended to always activate this if additional memory consumption and processing power consumption is not a problem. If increased MHz and memory consumption are an issue then the MHz and memory cost of this optional module need to be evaluated against the improvement in audio quality on a case by case basis. - 0: Disable afterburner (default). - 1: Enable afterburner. */ AACENC_BANDWIDTH = 0x0203, /*!< Core encoder audio bandwidth: - 0: Determine audio bandwidth internally (default, see chapter \ref BEHAVIOUR_BANDWIDTH). - 1 to fs/2: Audio bandwidth in Hertz. Limited to 20kHz max. Not usable if SBR is active. This setting is for experts only, better do not touch this value to avoid degraded audio quality. */ AACENC_PEAK_BITRATE = 0x0207, /*!< Peak bitrate configuration parameter to adjust maximum bits per audio frame. Bitrate is in bits/second. The peak bitrate will internally be limited to the chosen bitrate ::AACENC_BITRATE as lower limit and the number_of_effective_channels*6144 bit as upper limit. Setting the peak bitrate equal to ::AACENC_BITRATE does not necessarily mean that the audio frames will be of constant size. Since the peak bitate is in bits/second, the frame sizes can vary by one byte in one or the other direction over various frames. However, it is not recommended to reduce the peak pitrate to ::AACENC_BITRATE - it would disable the bitreservoir, which would affect the audio quality by a large amount. */ AACENC_TRANSMUX = 0x0300, /*!< Transport type to be used. See ::TRANSPORT_TYPE in FDK_audio.h. Following types can be configured in encoder library: - 0: raw access units - 1: ADIF bitstream format - 2: ADTS bitstream format - 6: Audio Mux Elements (LATM) with muxConfigPresent = 1 - 7: Audio Mux Elements (LATM) with muxConfigPresent = 0, out of band StreamMuxConfig - 10: Audio Sync Stream (LOAS) */ AACENC_HEADER_PERIOD = 0x0301, /*!< Frame count period for sending in-band configuration buffers within LATM/LOAS transport layer. Additionally this parameter configures the PCE repetition period in raw_data_block(). See \ref encPCE. - 0xFF: auto-mode default 10 for TT_MP4_ADTS, TT_MP4_LOAS and TT_MP4_LATM_MCP1, otherwise 0. - n: Frame count period. */ AACENC_SIGNALING_MODE = 0x0302, /*!< Signaling mode of the extension AOT: - 0: Implicit backward compatible signaling (default for non-MPEG-4 based AOT's and for the transport formats ADIF and ADTS) - A stream that uses implicit signaling can be decoded by every AAC decoder, even AAC-LC-only decoders - An AAC-LC-only decoder will only decode the low-frequency part of the stream, resulting in a band-limited output - This method works with all transport formats - This method does not work with downsampled SBR - 1: Explicit backward compatible signaling - A stream that uses explicit backward compatible signaling can be decoded by every AAC decoder, even AAC-LC-only decoders - An AAC-LC-only decoder will only decode the low-frequency part of the stream, resulting in a band-limited output - A decoder not capable of decoding PS will only decode the AAC-LC+SBR part. If the stream contained PS, the result will be a a decoded mono downmix - This method does not work with ADIF or ADTS. For LOAS/LATM, it only works with AudioMuxVersion==1 - This method does work with downsampled SBR - 2: Explicit hierarchical signaling (default for MPEG-4 based AOT's and for all transport formats excluding ADIF and ADTS) - A stream that uses explicit hierarchical signaling can be decoded only by HE-AAC decoders - An AAC-LC-only decoder will not decode a stream that uses explicit hierarchical signaling - A decoder not capable of decoding PS will not decode the stream at all if it contained PS - This method does not work with ADIF or ADTS. It works with LOAS/LATM and the MPEG-4 File format - This method does work with downsampled SBR For making sure that the listener always experiences the best audio quality, explicit hierarchical signaling should be used. This makes sure that only a full HE-AAC-capable decoder will decode those streams. The audio is played at full bandwidth. For best backwards compatibility, it is recommended to encode with implicit SBR signaling. A decoder capable of AAC-LC only will then only decode the AAC part, which means the decoded audio will sound band-limited. For MPEG-2 transport types (ADTS,ADIF), only implicit signaling is possible. For LOAS and LATM, explicit backwards compatible signaling only works together with AudioMuxVersion==1. The reason is that, for explicit backwards compatible signaling, additional information will be appended to the ASC. A decoder that is only capable of decoding AAC-LC will skip this part. Nevertheless, for jumping to the end of the ASC, it needs to know the ASC length. Transmitting the length of the ASC is a feature of AudioMuxVersion==1, it is not possible to transmit the length of the ASC with AudioMuxVersion==0, therefore an AAC-LC-only decoder will not be able to parse a LOAS/LATM stream that was being encoded with AudioMuxVersion==0. For downsampled SBR, explicit signaling is mandatory. The reason for this is that the extension sampling frequency (which is in case of SBR the sampling frequqncy of the SBR part) can only be signaled in explicit mode. For AAC-ELD, the SBR information is transmitted in the ELDSpecific Config, which is part of the AudioSpecificConfig. Therefore, the settings here will have no effect on AAC-ELD.*/ AACENC_TPSUBFRAMES = 0x0303, /*!< Number of sub frames in a transport frame for LOAS/LATM or ADTS (default 1). - ADTS: Maximum number of sub frames restricted to 4. - LOAS/LATM: Maximum number of sub frames restricted to 2.*/ AACENC_AUDIOMUXVER = 0x0304, /*!< AudioMuxVersion to be used for LATM. (AudioMuxVersionA, currently not implemented): - 0: Default, no transmission of tara Buffer fullness, no ASC length and including actual latm Buffer fullnes. - 1: Transmission of tara Buffer fullness, ASC length and actual latm Buffer fullness. - 2: Transmission of tara Buffer fullness, ASC length and maximum level of latm Buffer fullness. */ AACENC_PROTECTION = 0x0306, /*!< Configure protection in transport layer: - 0: No protection. (default) - 1: CRC active for ADTS transport format. */ AACENC_ANCILLARY_BITRATE = 0x0500, /*!< Constant ancillary data bitrate in bits/second. - 0: Either no ancillary data or insert exact number of bytes, denoted via input parameter, numAncBytes in AACENC_InArgs. - else: Insert ancillary data with specified bitrate. */ AACENC_METADATA_MODE = 0x0600, /*!< Configure Meta Data. See ::AACENC_MetaData for further details: - 0: Do not embed any metadata. - 1: Embed dynamic_range_info metadata. - 2: Embed dynamic_range_info and ancillary_data metadata. - 3: Embed ancillary_data metadata. */ AACENC_CONTROL_STATE = 0xFF00, /*!< There is an automatic process which internally reconfigures the encoder instance when a configuration parameter changed or an error occured. This paramerter allows overwriting or getting the control status of this process. See ::AACENC_CTRLFLAGS. */ AACENC_NONE = 0xFFFF /*!< ------ */ } AACENC_PARAM; #ifdef __cplusplus extern "C" { #endif /** * \brief Open an instance of the encoder. * * Allocate memory for an encoder instance with a functional range denoted by * the function parameters. Preinitialize encoder instance with default * configuration. * * \param phAacEncoder A pointer to an encoder handle. Initialized on return. * \param encModules Specify encoder modules to be supported in this encoder * instance: * - 0x0: Allocate memory for all available encoder * modules. * - else: Select memory allocation regarding encoder * modules. Following flags are possible and can be combined. * - 0x01: AAC module. * - 0x02: SBR module. * - 0x04: PS module. * - 0x08: MPS module. * - 0x10: Metadata module. * - example: (0x01|0x02|0x04|0x08|0x10) allocates * all modules and is equivalent to default configuration denotet by 0x0. * \param maxChannels Number of channels to be allocated. This parameter can * be used in different ways: * - 0: Allocate maximum number of AAC and SBR channels as * supported by the library. * - nChannels: Use same maximum number of channels for * allocating memory in AAC and SBR module. * - nChannels | (nSbrCh<<8): Number of SBR channels can be * different to AAC channels to save data memory. * * \return * - AACENC_OK, on succes. * - AACENC_INVALID_HANDLE, AACENC_MEMORY_ERROR, AACENC_INVALID_CONFIG, * on failure. */ AACENC_ERROR aacEncOpen(HANDLE_AACENCODER *phAacEncoder, const UINT encModules, const UINT maxChannels); /** * \brief Close the encoder instance. * * Deallocate encoder instance and free whole memory. * * \param phAacEncoder Pointer to the encoder handle to be deallocated. * * \return * - AACENC_OK, on success. * - AACENC_INVALID_HANDLE, on failure. */ AACENC_ERROR aacEncClose(HANDLE_AACENCODER *phAacEncoder); /** * \brief Encode audio data. * * This function is mainly for encoding audio data. In addition the function can * be used for an encoder (re)configuration process. * - PCM input data will be retrieved from external input buffer until the fill * level allows encoding a single frame. This functionality allows an external * buffer with reduced size in comparison to the AAC or HE-AAC audio frame * length. * - If the value of the input samples argument is zero, just internal * reinitialization will be applied if it is requested. * - At the end of a file the flushing process can be triggerd via setting the * value of the input samples argument to -1. The encoder delay lines are fully * flushed when the encoder returns no valid bitstream data * AACENC_OutArgs::numOutBytes. Furthermore the end of file is signaled by the * return value AACENC_ENCODE_EOF. * - If an error occured in the previous frame or any of the encoder parameters * changed, an internal reinitialization process will be applied before encoding * the incoming audio samples. * - The function can also be used for an independent reconfiguration process * without encoding. The first parameter has to be a valid encoder handle and * all other parameters can be set to NULL. * - If the size of the external bitbuffer in outBufDesc is not sufficient for * writing the whole bitstream, an internal error will be the return value and a * reconfiguration will be triggered. * * \param hAacEncoder A valid AAC encoder handle. * \param inBufDesc Input buffer descriptor, see AACENC_BufDesc: * - At least one input buffer with audio data is * expected. * - Optionally a second input buffer with * ancillary data can be fed. * \param outBufDesc Output buffer descriptor, see AACENC_BufDesc: * - Provide one output buffer for the encoded * bitstream. * \param inargs Input arguments, see AACENC_InArgs. * \param outargs Output arguments, AACENC_OutArgs. * * \return * - AACENC_OK, on success. * - AACENC_INVALID_HANDLE, AACENC_ENCODE_ERROR, on failure in encoding * process. * - AACENC_INVALID_CONFIG, AACENC_INIT_ERROR, AACENC_INIT_AAC_ERROR, * AACENC_INIT_SBR_ERROR, AACENC_INIT_TP_ERROR, AACENC_INIT_META_ERROR, * AACENC_INIT_MPS_ERROR, on failure in encoder initialization. * - AACENC_UNSUPPORTED_PARAMETER, on incorrect input or output buffer * descriptor initialization. * - AACENC_ENCODE_EOF, when flushing fully concluded. */ AACENC_ERROR aacEncEncode(const HANDLE_AACENCODER hAacEncoder, const AACENC_BufDesc *inBufDesc, const AACENC_BufDesc *outBufDesc, const AACENC_InArgs *inargs, AACENC_OutArgs *outargs); /** * \brief Acquire info about present encoder instance. * * This function retrieves information of the encoder configuration. In addition * to informative internal states, a configuration data block of the current * encoder settings will be returned. The format is either Audio Specific Config * in case of Raw Packets transport format or StreamMuxConfig in case of * LOAS/LATM transport format. The configuration data block is binary coded as * specified in ISO/IEC 14496-3 (MPEG-4 audio), to be used directly for MPEG-4 * File Format or RFC3016 or RFC3640 applications. * * \param hAacEncoder A valid AAC encoder handle. * \param pInfo Pointer to AACENC_InfoStruct. Filled on return. * * \return * - AACENC_OK, on succes. * - AACENC_INIT_ERROR, on failure. */ AACENC_ERROR aacEncInfo(const HANDLE_AACENCODER hAacEncoder, AACENC_InfoStruct *pInfo); /** * \brief Set one single AAC encoder parameter. * * This function allows configuration of all encoder parameters specified in * ::AACENC_PARAM. Each parameter must be set with a separate function call. An * internal validation of the configuration value range will be done and an * internal reconfiguration will be signaled. The actual configuration adoption * is part of the subsequent aacEncEncode() call. * * \param hAacEncoder A valid AAC encoder handle. * \param param Parameter to be set. See ::AACENC_PARAM. * \param value Parameter value. See parameter description in * ::AACENC_PARAM. * * \return * - AACENC_OK, on success. * - AACENC_INVALID_HANDLE, AACENC_UNSUPPORTED_PARAMETER, * AACENC_INVALID_CONFIG, on failure. */ AACENC_ERROR aacEncoder_SetParam(const HANDLE_AACENCODER hAacEncoder, const AACENC_PARAM param, const UINT value); /** * \brief Get one single AAC encoder parameter. * * This function is the complement to aacEncoder_SetParam(). After encoder * reinitialization with user defined settings, the internal status can be * obtained of each parameter, specified with ::AACENC_PARAM. * * \param hAacEncoder A valid AAC encoder handle. * \param param Parameter to be returned. See ::AACENC_PARAM. * * \return Internal configuration value of specifed parameter ::AACENC_PARAM. */ UINT aacEncoder_GetParam(const HANDLE_AACENCODER hAacEncoder, const AACENC_PARAM param); /** * \brief Get information about encoder library build. * * Fill a given LIB_INFO structure with library version information. * * \param info Pointer to an allocated LIB_INFO struct. * * \return * - AACENC_OK, on success. * - AACENC_INVALID_HANDLE, AACENC_INIT_ERROR, on failure. */ AACENC_ERROR aacEncGetLibInfo(LIB_INFO *info); #ifdef __cplusplus } #endif #endif /* AACENC_LIB_H */ fdk-aac-2.0.2/libAACenc/src/000077500000000000000000000000001404225310700153275ustar00rootroot00000000000000fdk-aac-2.0.2/libAACenc/src/aacEnc_ram.cpp000066400000000000000000000213151404225310700200460ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Lohwasser, M. Gayer Description: *******************************************************************************/ /*! \file \brief Memory layout \author Markus Lohwasser */ #include "aacEnc_ram.h" C_AALLOC_MEM(AACdynamic_RAM, FIXP_DBL, AAC_ENC_DYN_RAM_SIZE / sizeof(FIXP_DBL)) /* Static memory areas, must not be overwritten in other sections of the decoder ! */ /* The structure AacEncoder contains all Encoder structures. */ C_ALLOC_MEM(Ram_aacEnc_AacEncoder, struct AAC_ENC, 1) /* The structure PSY_INTERNAl contains all psych configuration and data pointer. * PsyStatic holds last and current Psych data. * PsyInputBuffer contains time input. Signal is needed at the beginning of Psych. Memory can be reused after signal is in time domain. * PsyData contains spectral, nrg and threshold information. Necessary data are copied into PsyOut, so memory is available after leaving psych. * TnsData, ChaosMeasure, PnsData are temporarily necessary, e.g. use memory from PsyInputBuffer. */ C_ALLOC_MEM2(Ram_aacEnc_PsyElement, PSY_ELEMENT, 1, ((8))) C_ALLOC_MEM(Ram_aacEnc_PsyInternal, PSY_INTERNAL, 1) C_ALLOC_MEM2(Ram_aacEnc_PsyStatic, PSY_STATIC, 1, (8)) C_ALLOC_MEM2(Ram_aacEnc_PsyInputBuffer, INT_PCM, MAX_INPUT_BUFFER_SIZE, (8)) PSY_DYNAMIC *GetRam_aacEnc_PsyDynamic(int n, UCHAR *dynamic_RAM) { FDK_ASSERT(dynamic_RAM != 0); /* The reinterpret_cast is used to suppress a compiler warning. We know that * (dynamic_RAM + P_BUF_1 + n*sizeof(PSY_DYNAMIC)) is sufficiently aligned, so * the cast is safe */ return reinterpret_cast(reinterpret_cast( dynamic_RAM + P_BUF_1 + n * sizeof(PSY_DYNAMIC))); } /* The structure PSY_OUT holds all psychoaccoustic data needed in quantization module */ C_ALLOC_MEM2(Ram_aacEnc_PsyOut, PSY_OUT, 1, (1)) C_ALLOC_MEM2(Ram_aacEnc_PsyOutElements, PSY_OUT_ELEMENT, 1, (1) * ((8))) C_ALLOC_MEM2(Ram_aacEnc_PsyOutChannel, PSY_OUT_CHANNEL, 1, (1) * (8)) /* The structure QC_STATE contains preinitialized settings and quantizer structures. * AdjustThreshold structure contains element-wise settings. * ElementBits contains elemnt-wise bit consumption settings. * When CRC is active, lookup table is necessary for fast crc calculation. * Bitcounter contains buffer to find optimal codebooks and minimal bit consumption. Values are temporarily, so dynamic memory can be used. */ C_ALLOC_MEM(Ram_aacEnc_QCstate, QC_STATE, 1) C_ALLOC_MEM(Ram_aacEnc_AdjustThreshold, ADJ_THR_STATE, 1) C_ALLOC_MEM2(Ram_aacEnc_AdjThrStateElement, ATS_ELEMENT, 1, ((8))) C_ALLOC_MEM2(Ram_aacEnc_ElementBits, ELEMENT_BITS, 1, ((8))) C_ALLOC_MEM(Ram_aacEnc_BitCntrState, struct BITCNTR_STATE, 1) INT *GetRam_aacEnc_BitLookUp(int n, UCHAR *dynamic_RAM) { FDK_ASSERT(dynamic_RAM != 0); /* The reinterpret_cast is used to suppress a compiler warning. We know that * (dynamic_RAM + P_BUF_1) is sufficiently aligned, so the cast is safe */ return reinterpret_cast( reinterpret_cast(dynamic_RAM + P_BUF_1)); } INT *GetRam_aacEnc_MergeGainLookUp(int n, UCHAR *dynamic_RAM) { FDK_ASSERT(dynamic_RAM != 0); /* The reinterpret_cast is used to suppress a compiler warning. We know that * (dynamic_RAM + P_BUF_1 + sizeof(INT)*(MAX_SFB_LONG*(CODE_BOOK_ESC_NDX+1))) * is sufficiently aligned, so the cast is safe */ return reinterpret_cast(reinterpret_cast( dynamic_RAM + P_BUF_1 + sizeof(INT) * (MAX_SFB_LONG * (CODE_BOOK_ESC_NDX + 1)))); } /* The structure QC_OUT contains settings and structures holding all necessary information needed in bitstreamwriter. */ C_ALLOC_MEM2(Ram_aacEnc_QCout, QC_OUT, 1, (1)) C_ALLOC_MEM2(Ram_aacEnc_QCelement, QC_OUT_ELEMENT, 1, (1) * ((8))) QC_OUT_CHANNEL *GetRam_aacEnc_QCchannel(int n, UCHAR *dynamic_RAM) { FDK_ASSERT(dynamic_RAM != 0); /* The reinterpret_cast is used to suppress a compiler warning. We know that * (dynamic_RAM + P_BUF_0 + n*sizeof(QC_OUT_CHANNEL)) is sufficiently aligned, * so the cast is safe */ return reinterpret_cast(reinterpret_cast( dynamic_RAM + P_BUF_0 + n * ALIGN_SIZE(sizeof(QC_OUT_CHANNEL)))); } fdk-aac-2.0.2/libAACenc/src/aacEnc_ram.h000066400000000000000000000227331404225310700175200ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Lohwasser, M. Gayer Description: *******************************************************************************/ /*! \file \brief Memory layout \author Markus Lohwasser */ #ifndef AACENC_RAM_H #define AACENC_RAM_H #include "common_fix.h" #include "aacenc.h" #include "psy_data.h" #include "interface.h" #include "psy_main.h" #include "bitenc.h" #include "bit_cnt.h" #include "psy_const.h" #define OUTPUTBUFFER_SIZE \ (8192) /*!< Output buffer size has to be at least 6144 bits per channel \ (768 bytes). FDK bitbuffer implementation expects buffer of \ size 2^n. */ /* Moved AAC_ENC struct definition from aac_enc.cpp into aacEnc_ram.h to get size and respective static memory in aacEnc_ram.cpp. aac_enc.h is the outward visible header file and putting the struct into would cause necessity of additional visible header files outside library. */ /* define hBitstream size: max AAC framelength is 6144 bits/channel */ /*#define BUFFER_BITSTR_SIZE ((6400*(8)/bbWordSize) +((bbWordSize - 1) / * bbWordSize))*/ struct AAC_ENC { AACENC_CONFIG *config; INT ancillaryBitsPerFrame; /* ancillary bits per frame calculated from ancillary rate */ CHANNEL_MAPPING channelMapping; QC_STATE *qcKernel; QC_OUT *qcOut[(1)]; PSY_OUT *psyOut[(1)]; PSY_INTERNAL *psyKernel; /* lifetime vars */ CHANNEL_MODE encoderMode; INT bandwidth90dB; AACENC_BITRATE_MODE bitrateMode; INT dontWriteAdif; /* use: write ADIF header only before 1st frame */ FIXP_DBL *dynamic_RAM; INT maxChannels; /* used while allocation */ INT maxElements; INT maxFrames; AUDIO_OBJECT_TYPE aot; /* AOT to be used while encoding. */ }; #define maxSize(a, b) (((a) > (b)) ? (a) : (b)) #define BIT_LOOK_UP_SIZE \ (sizeof(INT) * (MAX_SFB_LONG * (CODE_BOOK_ESC_NDX + 1))) #define MERGE_GAIN_LOOK_UP_SIZE (sizeof(INT) * MAX_SFB_LONG) /* Size of AhFlag buffer in function FDKaacEnc_adaptThresholdsToPe() */ #define ADJ_THR_AH_FLAG_SIZE (sizeof(UCHAR) * ((8)) * (2) * MAX_GROUPED_SFB) /* Size of ThrExp buffer in function FDKaacEnc_adaptThresholdsToPe() */ #define ADJ_THR_THR_EXP_SIZE (sizeof(FIXP_DBL) * ((8)) * (2) * MAX_GROUPED_SFB) /* Size of sfbNActiveLinesLdData buffer in function FDKaacEnc_correctThresh() */ #define ADJ_THR_ACT_LIN_LD_DATA_SIZE \ (sizeof(FIXP_DBL) * ((8)) * (2) * MAX_GROUPED_SFB) /* Total amount of dynamic buffer needed in adjust thresholds functionality */ #define ADJ_THR_SIZE \ (ADJ_THR_AH_FLAG_SIZE + ADJ_THR_THR_EXP_SIZE + ADJ_THR_ACT_LIN_LD_DATA_SIZE) /* Dynamic RAM - Allocation */ /* +++++++++++++++++++++++++++++++++++++++++++++++++++++ | P_BUF_0 | P_BUF_1 | +++++++++++++++++++++++++++++++++++++++++++++++++++++ | QC_OUT_CH | PSY_DYN | +++++++++++++++++++++++++++++++++++++++++++++++++++++ | | BitLookUp+MergeGainLookUp | +++++++++++++++++++++++++++++++++++++++++++++++++++++ | | AH_FLAG | THR_EXP | ACT_LIN_LD_DATA | +++++++++++++++++++++++++++++++++++++++++++++++++++++ | | Bitstream output buffer | +++++++++++++++++++++++++++++++++++++++++++++++++++++ */ #define BUF_SIZE_0 (ALIGN_SIZE(sizeof(QC_OUT_CHANNEL)) * (8)) #define BUF_SIZE_1 \ (ALIGN_SIZE(maxSize(maxSize(sizeof(PSY_DYNAMIC), \ (BIT_LOOK_UP_SIZE + MERGE_GAIN_LOOK_UP_SIZE)), \ ADJ_THR_SIZE))) #define P_BUF_0 (0) #define P_BUF_1 (P_BUF_0 + BUF_SIZE_0) #define AAC_ENC_DYN_RAM_SIZE (BUF_SIZE_0 + BUF_SIZE_1) H_ALLOC_MEM(AACdynamic_RAM, FIXP_DBL) /* ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ END - Dynamic RAM - Allocation */ /* See further Memory Allocation details in aacEnc_ram.cpp */ H_ALLOC_MEM(Ram_aacEnc_AacEncoder, AAC_ENC) H_ALLOC_MEM(Ram_aacEnc_PsyElement, PSY_ELEMENT) H_ALLOC_MEM(Ram_aacEnc_PsyInternal, PSY_INTERNAL) H_ALLOC_MEM(Ram_aacEnc_PsyStatic, PSY_STATIC) H_ALLOC_MEM(Ram_aacEnc_PsyInputBuffer, INT_PCM) PSY_DYNAMIC *GetRam_aacEnc_PsyDynamic(int n, UCHAR *dynamic_RAM); H_ALLOC_MEM(Ram_aacEnc_PsyOutChannel, PSY_OUT_CHANNEL) H_ALLOC_MEM(Ram_aacEnc_PsyOut, PSY_OUT) H_ALLOC_MEM(Ram_aacEnc_PsyOutElements, PSY_OUT_ELEMENT) H_ALLOC_MEM(Ram_aacEnc_QCstate, QC_STATE) H_ALLOC_MEM(Ram_aacEnc_AdjustThreshold, ADJ_THR_STATE) H_ALLOC_MEM(Ram_aacEnc_AdjThrStateElement, ATS_ELEMENT) H_ALLOC_MEM(Ram_aacEnc_ElementBits, ELEMENT_BITS) H_ALLOC_MEM(Ram_aacEnc_BitCntrState, BITCNTR_STATE) INT *GetRam_aacEnc_BitLookUp(int n, UCHAR *dynamic_RAM); INT *GetRam_aacEnc_MergeGainLookUp(int n, UCHAR *dynamic_RAM); QC_OUT_CHANNEL *GetRam_aacEnc_QCchannel(int n, UCHAR *dynamic_RAM); H_ALLOC_MEM(Ram_aacEnc_QCout, QC_OUT) H_ALLOC_MEM(Ram_aacEnc_QCelement, QC_OUT_ELEMENT) #endif /* #ifndef AACENC_RAM_H */ fdk-aac-2.0.2/libAACenc/src/aacEnc_rom.cpp000066400000000000000000005131071404225310700200710ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Lohwasser, M. Gayer Description: *******************************************************************************/ #include "aacEnc_rom.h" /* Huffman Tables */ const ULONG FDKaacEnc_huff_ltab1_2[3][3][3][3] = { {{{0x000b0009, 0x00090007, 0x000b0009}, {0x000a0008, 0x00070006, 0x000a0008}, {0x000b0009, 0x00090008, 0x000b0009}}, {{0x000a0008, 0x00070006, 0x000a0007}, {0x00070006, 0x00050005, 0x00070006}, {0x00090007, 0x00070006, 0x000a0008}}, {{0x000b0009, 0x00090007, 0x000b0008}, {0x00090008, 0x00070006, 0x00090008}, {0x000b0009, 0x00090007, 0x000b0009}}}, {{{0x00090008, 0x00070006, 0x00090007}, {0x00070006, 0x00050005, 0x00070006}, {0x00090007, 0x00070006, 0x00090008}}, {{0x00070006, 0x00050005, 0x00070006}, {0x00050005, 0x00010003, 0x00050005}, {0x00070006, 0x00050005, 0x00070006}}, {{0x00090008, 0x00070006, 0x00090007}, {0x00070006, 0x00050005, 0x00070006}, {0x00090008, 0x00070006, 0x00090008}}}, {{{0x000b0009, 0x00090007, 0x000b0009}, {0x00090008, 0x00070006, 0x00090008}, {0x000b0008, 0x00090007, 0x000b0009}}, {{0x000a0008, 0x00070006, 0x00090007}, {0x00070006, 0x00050004, 0x00070006}, {0x00090008, 0x00070006, 0x000a0007}}, {{0x000b0009, 0x00090007, 0x000b0009}, {0x000a0007, 0x00070006, 0x00090008}, {0x000b0009, 0x00090007, 0x000b0009}}}}; const ULONG FDKaacEnc_huff_ltab3_4[3][3][3][3] = { {{{0x00010004, 0x00040005, 0x00080008}, {0x00040005, 0x00050004, 0x00080008}, {0x00090009, 0x00090008, 0x000a000b}}, {{0x00040005, 0x00060005, 0x00090008}, {0x00060005, 0x00060004, 0x00090008}, {0x00090008, 0x00090007, 0x000a000a}}, {{0x00090009, 0x000a0008, 0x000d000b}, {0x00090008, 0x00090008, 0x000b000a}, {0x000b000b, 0x000a000a, 0x000c000b}}}, {{{0x00040004, 0x00060005, 0x000a0008}, {0x00060004, 0x00070004, 0x000a0008}, {0x000a0008, 0x000a0008, 0x000c000a}}, {{0x00050004, 0x00070004, 0x000b0008}, {0x00060004, 0x00070004, 0x000a0007}, {0x00090008, 0x00090007, 0x000b0009}}, {{0x00090008, 0x000a0008, 0x000d000a}, {0x00080007, 0x00090007, 0x000c0009}, {0x000a000a, 0x000b0009, 0x000c000a}}}, {{{0x00080008, 0x000a0008, 0x000f000b}, {0x00090008, 0x000b0007, 0x000f000a}, {0x000d000b, 0x000e000a, 0x0010000c}}, {{0x00080008, 0x000a0007, 0x000e000a}, {0x00090007, 0x000a0007, 0x000e0009}, {0x000c000a, 0x000c0009, 0x000f000b}}, {{0x000b000b, 0x000c000a, 0x0010000c}, {0x000a000a, 0x000b0009, 0x000f000b}, {0x000c000b, 0x000c000a, 0x000f000b}}}}; const ULONG FDKaacEnc_huff_ltab5_6[9][9] = { {0x000d000b, 0x000c000a, 0x000b0009, 0x000b0009, 0x000a0009, 0x000b0009, 0x000b0009, 0x000c000a, 0x000d000b}, {0x000c000a, 0x000b0009, 0x000a0008, 0x00090007, 0x00080007, 0x00090007, 0x000a0008, 0x000b0009, 0x000c000a}, {0x000c0009, 0x000a0008, 0x00090006, 0x00080006, 0x00070006, 0x00080006, 0x00090006, 0x000a0008, 0x000b0009}, {0x000b0009, 0x00090007, 0x00080006, 0x00050004, 0x00040004, 0x00050004, 0x00080006, 0x00090007, 0x000b0009}, {0x000a0009, 0x00080007, 0x00070006, 0x00040004, 0x00010004, 0x00040004, 0x00070006, 0x00080007, 0x000b0009}, {0x000b0009, 0x00090007, 0x00080006, 0x00050004, 0x00040004, 0x00050004, 0x00080006, 0x00090007, 0x000b0009}, {0x000b0009, 0x000a0008, 0x00090006, 0x00080006, 0x00070006, 0x00080006, 0x00090006, 0x000a0008, 0x000b0009}, {0x000c000a, 0x000b0009, 0x000a0008, 0x00090007, 0x00080007, 0x00090007, 0x000a0007, 0x000b0008, 0x000c000a}, {0x000d000b, 0x000c000a, 0x000c0009, 0x000b0009, 0x000a0009, 0x000a0009, 0x000b0009, 0x000c000a, 0x000d000b}}; const ULONG FDKaacEnc_huff_ltab7_8[8][8] = { {0x00010005, 0x00030004, 0x00060005, 0x00070006, 0x00080007, 0x00090008, 0x000a0009, 0x000b000a}, {0x00030004, 0x00040003, 0x00060004, 0x00070005, 0x00080006, 0x00080007, 0x00090007, 0x00090008}, {0x00060005, 0x00060004, 0x00070004, 0x00080005, 0x00080006, 0x00090007, 0x00090007, 0x000a0008}, {0x00070006, 0x00070005, 0x00080005, 0x00080006, 0x00090006, 0x00090007, 0x000a0008, 0x000a0008}, {0x00080007, 0x00080006, 0x00090006, 0x00090006, 0x000a0007, 0x000a0007, 0x000a0008, 0x000b0009}, {0x00090008, 0x00080007, 0x00090006, 0x00090007, 0x000a0007, 0x000a0008, 0x000b0008, 0x000b000a}, {0x000a0009, 0x00090007, 0x00090007, 0x000a0008, 0x000a0008, 0x000b0008, 0x000c0009, 0x000c0009}, {0x000b000a, 0x000a0008, 0x000a0008, 0x000a0008, 0x000b0009, 0x000b0009, 0x000c0009, 0x000c000a}}; const ULONG FDKaacEnc_huff_ltab9_10[13][13] = { {0x00010006, 0x00030005, 0x00060006, 0x00080006, 0x00090007, 0x000a0008, 0x000a0009, 0x000b000a, 0x000b000a, 0x000c000a, 0x000c000b, 0x000d000b, 0x000d000c}, {0x00030005, 0x00040004, 0x00060004, 0x00070005, 0x00080006, 0x00080007, 0x00090007, 0x000a0008, 0x000a0008, 0x000a0009, 0x000b000a, 0x000c000a, 0x000c000b}, {0x00060006, 0x00060004, 0x00070005, 0x00080005, 0x00080006, 0x00090006, 0x000a0007, 0x000a0008, 0x000a0008, 0x000b0009, 0x000c0009, 0x000c000a, 0x000c000a}, {0x00080006, 0x00070005, 0x00080005, 0x00090005, 0x00090006, 0x000a0007, 0x000a0007, 0x000b0008, 0x000b0008, 0x000b0009, 0x000c0009, 0x000c000a, 0x000d000a}, {0x00090007, 0x00080006, 0x00090006, 0x00090006, 0x000a0006, 0x000a0007, 0x000b0007, 0x000b0008, 0x000b0008, 0x000c0009, 0x000c0009, 0x000c000a, 0x000d000a}, {0x000a0008, 0x00090007, 0x00090006, 0x000a0007, 0x000b0007, 0x000b0007, 0x000b0008, 0x000c0008, 0x000b0008, 0x000c0009, 0x000c000a, 0x000d000a, 0x000d000b}, {0x000b0009, 0x00090007, 0x000a0007, 0x000b0007, 0x000b0007, 0x000b0008, 0x000c0008, 0x000c0009, 0x000c0009, 0x000c0009, 0x000d000a, 0x000d000a, 0x000d000b}, {0x000b0009, 0x000a0008, 0x000a0008, 0x000b0008, 0x000b0008, 0x000c0008, 0x000c0009, 0x000d0009, 0x000d0009, 0x000d000a, 0x000d000a, 0x000d000b, 0x000d000b}, {0x000b0009, 0x000a0008, 0x000a0008, 0x000b0008, 0x000b0008, 0x000b0008, 0x000c0009, 0x000c0009, 0x000d000a, 0x000d000a, 0x000e000a, 0x000d000b, 0x000e000b}, {0x000b000a, 0x000a0009, 0x000b0009, 0x000b0009, 0x000c0009, 0x000c0009, 0x000c0009, 0x000c000a, 0x000d000a, 0x000d000a, 0x000e000b, 0x000e000b, 0x000e000c}, {0x000c000a, 0x000b0009, 0x000b0009, 0x000c0009, 0x000c0009, 0x000c000a, 0x000d000a, 0x000d000a, 0x000d000a, 0x000e000b, 0x000e000b, 0x000e000b, 0x000f000c}, {0x000c000b, 0x000b000a, 0x000c0009, 0x000c000a, 0x000c000a, 0x000d000a, 0x000d000a, 0x000d000a, 0x000d000b, 0x000e000b, 0x000e000b, 0x000f000b, 0x000f000c}, {0x000d000b, 0x000c000a, 0x000c000a, 0x000c000a, 0x000d000a, 0x000d000a, 0x000d000a, 0x000d000b, 0x000e000b, 0x000e000c, 0x000e000c, 0x000e000c, 0x000f000c}}; const UCHAR FDKaacEnc_huff_ltab11[17][17] = { {0x04, 0x05, 0x06, 0x07, 0x08, 0x08, 0x09, 0x0a, 0x0a, 0x0a, 0x0b, 0x0b, 0x0c, 0x0b, 0x0c, 0x0c, 0x0a}, {0x05, 0x04, 0x05, 0x06, 0x07, 0x07, 0x08, 0x08, 0x09, 0x09, 0x09, 0x0a, 0x0a, 0x0a, 0x0a, 0x0b, 0x08}, {0x06, 0x05, 0x05, 0x06, 0x07, 0x07, 0x08, 0x08, 0x08, 0x09, 0x09, 0x09, 0x0a, 0x0a, 0x0a, 0x0a, 0x08}, {0x07, 0x06, 0x06, 0x06, 0x07, 0x07, 0x08, 0x08, 0x08, 0x09, 0x09, 0x09, 0x0a, 0x0a, 0x0a, 0x0a, 0x08}, {0x08, 0x07, 0x07, 0x07, 0x07, 0x08, 0x08, 0x08, 0x08, 0x09, 0x09, 0x09, 0x0a, 0x0a, 0x0a, 0x0a, 0x08}, {0x08, 0x07, 0x07, 0x07, 0x07, 0x08, 0x08, 0x08, 0x09, 0x09, 0x09, 0x09, 0x0a, 0x0a, 0x0a, 0x0a, 0x08}, {0x09, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x09, 0x09, 0x09, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x08}, {0x09, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x09, 0x09, 0x09, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x08}, {0x0a, 0x09, 0x08, 0x08, 0x09, 0x09, 0x09, 0x09, 0x09, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0b, 0x08}, {0x0a, 0x09, 0x09, 0x09, 0x09, 0x09, 0x09, 0x09, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0b, 0x0b, 0x08}, {0x0b, 0x09, 0x09, 0x09, 0x09, 0x09, 0x09, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0b, 0x0a, 0x0b, 0x0b, 0x08}, {0x0b, 0x0a, 0x09, 0x09, 0x0a, 0x09, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x08}, {0x0b, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x09}, {0x0b, 0x0a, 0x09, 0x09, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x09}, {0x0b, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x09}, {0x0c, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0c, 0x0c, 0x09}, {0x09, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x09, 0x05}}; const UCHAR FDKaacEnc_huff_ltabscf[121] = { 0x12, 0x12, 0x12, 0x12, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x12, 0x13, 0x12, 0x11, 0x11, 0x10, 0x11, 0x10, 0x10, 0x10, 0x10, 0x0f, 0x0f, 0x0e, 0x0e, 0x0e, 0x0e, 0x0e, 0x0e, 0x0d, 0x0d, 0x0c, 0x0c, 0x0c, 0x0b, 0x0c, 0x0b, 0x0a, 0x0a, 0x0a, 0x09, 0x09, 0x08, 0x08, 0x08, 0x07, 0x06, 0x06, 0x05, 0x04, 0x03, 0x01, 0x04, 0x04, 0x05, 0x06, 0x06, 0x07, 0x07, 0x08, 0x08, 0x09, 0x09, 0x0a, 0x0a, 0x0a, 0x0b, 0x0b, 0x0b, 0x0b, 0x0c, 0x0c, 0x0d, 0x0d, 0x0d, 0x0e, 0x0e, 0x10, 0x0f, 0x10, 0x0f, 0x12, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13}; const USHORT FDKaacEnc_huff_ctab1[3][3][3][3] = {{{{0x07f8, 0x01f1, 0x07fd}, {0x03f5, 0x0068, 0x03f0}, {0x07f7, 0x01ec, 0x07f5}}, {{0x03f1, 0x0072, 0x03f4}, {0x0074, 0x0011, 0x0076}, {0x01eb, 0x006c, 0x03f6}}, {{0x07fc, 0x01e1, 0x07f1}, {0x01f0, 0x0061, 0x01f6}, {0x07f2, 0x01ea, 0x07fb}}}, {{{0x01f2, 0x0069, 0x01ed}, {0x0077, 0x0017, 0x006f}, {0x01e6, 0x0064, 0x01e5}}, {{0x0067, 0x0015, 0x0062}, {0x0012, 0x0000, 0x0014}, {0x0065, 0x0016, 0x006d}}, {{0x01e9, 0x0063, 0x01e4}, {0x006b, 0x0013, 0x0071}, {0x01e3, 0x0070, 0x01f3}}}, {{{0x07fe, 0x01e7, 0x07f3}, {0x01ef, 0x0060, 0x01ee}, {0x07f0, 0x01e2, 0x07fa}}, {{0x03f3, 0x006a, 0x01e8}, {0x0075, 0x0010, 0x0073}, {0x01f4, 0x006e, 0x03f7}}, {{0x07f6, 0x01e0, 0x07f9}, {0x03f2, 0x0066, 0x01f5}, {0x07ff, 0x01f7, 0x07f4}}}}; const USHORT FDKaacEnc_huff_ctab2[3][3][3][3] = {{{{0x01f3, 0x006f, 0x01fd}, {0x00eb, 0x0023, 0x00ea}, {0x01f7, 0x00e8, 0x01fa}}, {{0x00f2, 0x002d, 0x0070}, {0x0020, 0x0006, 0x002b}, {0x006e, 0x0028, 0x00e9}}, {{0x01f9, 0x0066, 0x00f8}, {0x00e7, 0x001b, 0x00f1}, {0x01f4, 0x006b, 0x01f5}}}, {{{0x00ec, 0x002a, 0x006c}, {0x002c, 0x000a, 0x0027}, {0x0067, 0x001a, 0x00f5}}, {{0x0024, 0x0008, 0x001f}, {0x0009, 0x0000, 0x0007}, {0x001d, 0x000b, 0x0030}}, {{0x00ef, 0x001c, 0x0064}, {0x001e, 0x000c, 0x0029}, {0x00f3, 0x002f, 0x00f0}}}, {{{0x01fc, 0x0071, 0x01f2}, {0x00f4, 0x0021, 0x00e6}, {0x00f7, 0x0068, 0x01f8}}, {{0x00ee, 0x0022, 0x0065}, {0x0031, 0x0002, 0x0026}, {0x00ed, 0x0025, 0x006a}}, {{0x01fb, 0x0072, 0x01fe}, {0x0069, 0x002e, 0x00f6}, {0x01ff, 0x006d, 0x01f6}}}}; const USHORT FDKaacEnc_huff_ctab3[3][3][3][3] = {{{{0x0000, 0x0009, 0x00ef}, {0x000b, 0x0019, 0x00f0}, {0x01eb, 0x01e6, 0x03f2}}, {{0x000a, 0x0035, 0x01ef}, {0x0034, 0x0037, 0x01e9}, {0x01ed, 0x01e7, 0x03f3}}, {{0x01ee, 0x03ed, 0x1ffa}, {0x01ec, 0x01f2, 0x07f9}, {0x07f8, 0x03f8, 0x0ff8}}}, {{{0x0008, 0x0038, 0x03f6}, {0x0036, 0x0075, 0x03f1}, {0x03eb, 0x03ec, 0x0ff4}}, {{0x0018, 0x0076, 0x07f4}, {0x0039, 0x0074, 0x03ef}, {0x01f3, 0x01f4, 0x07f6}}, {{0x01e8, 0x03ea, 0x1ffc}, {0x00f2, 0x01f1, 0x0ffb}, {0x03f5, 0x07f3, 0x0ffc}}}, {{{0x00ee, 0x03f7, 0x7ffe}, {0x01f0, 0x07f5, 0x7ffd}, {0x1ffb, 0x3ffa, 0xffff}}, {{0x00f1, 0x03f0, 0x3ffc}, {0x01ea, 0x03ee, 0x3ffb}, {0x0ff6, 0x0ffa, 0x7ffc}}, {{0x07f2, 0x0ff5, 0xfffe}, {0x03f4, 0x07f7, 0x7ffb}, {0x0ff7, 0x0ff9, 0x7ffa}}}}; const USHORT FDKaacEnc_huff_ctab4[3][3][3][3] = {{{{0x0007, 0x0016, 0x00f6}, {0x0018, 0x0008, 0x00ef}, {0x01ef, 0x00f3, 0x07f8}}, {{0x0019, 0x0017, 0x00ed}, {0x0015, 0x0001, 0x00e2}, {0x00f0, 0x0070, 0x03f0}}, {{0x01ee, 0x00f1, 0x07fa}, {0x00ee, 0x00e4, 0x03f2}, {0x07f6, 0x03ef, 0x07fd}}}, {{{0x0005, 0x0014, 0x00f2}, {0x0009, 0x0004, 0x00e5}, {0x00f4, 0x00e8, 0x03f4}}, {{0x0006, 0x0002, 0x00e7}, {0x0003, 0x0000, 0x006b}, {0x00e3, 0x0069, 0x01f3}}, {{0x00eb, 0x00e6, 0x03f6}, {0x006e, 0x006a, 0x01f4}, {0x03ec, 0x01f0, 0x03f9}}}, {{{0x00f5, 0x00ec, 0x07fb}, {0x00ea, 0x006f, 0x03f7}, {0x07f9, 0x03f3, 0x0fff}}, {{0x00e9, 0x006d, 0x03f8}, {0x006c, 0x0068, 0x01f5}, {0x03ee, 0x01f2, 0x07f4}}, {{0x07f7, 0x03f1, 0x0ffe}, {0x03ed, 0x01f1, 0x07f5}, {0x07fe, 0x03f5, 0x07fc}}}}; const USHORT FDKaacEnc_huff_ctab5[9][9] = { {0x1fff, 0x0ff7, 0x07f4, 0x07e8, 0x03f1, 0x07ee, 0x07f9, 0x0ff8, 0x1ffd}, {0x0ffd, 0x07f1, 0x03e8, 0x01e8, 0x00f0, 0x01ec, 0x03ee, 0x07f2, 0x0ffa}, {0x0ff4, 0x03ef, 0x01f2, 0x00e8, 0x0070, 0x00ec, 0x01f0, 0x03ea, 0x07f3}, {0x07eb, 0x01eb, 0x00ea, 0x001a, 0x0008, 0x0019, 0x00ee, 0x01ef, 0x07ed}, {0x03f0, 0x00f2, 0x0073, 0x000b, 0x0000, 0x000a, 0x0071, 0x00f3, 0x07e9}, {0x07ef, 0x01ee, 0x00ef, 0x0018, 0x0009, 0x001b, 0x00eb, 0x01e9, 0x07ec}, {0x07f6, 0x03eb, 0x01f3, 0x00ed, 0x0072, 0x00e9, 0x01f1, 0x03ed, 0x07f7}, {0x0ff6, 0x07f0, 0x03e9, 0x01ed, 0x00f1, 0x01ea, 0x03ec, 0x07f8, 0x0ff9}, {0x1ffc, 0x0ffc, 0x0ff5, 0x07ea, 0x03f3, 0x03f2, 0x07f5, 0x0ffb, 0x1ffe}}; const USHORT FDKaacEnc_huff_ctab6[9][9] = { {0x07fe, 0x03fd, 0x01f1, 0x01eb, 0x01f4, 0x01ea, 0x01f0, 0x03fc, 0x07fd}, {0x03f6, 0x01e5, 0x00ea, 0x006c, 0x0071, 0x0068, 0x00f0, 0x01e6, 0x03f7}, {0x01f3, 0x00ef, 0x0032, 0x0027, 0x0028, 0x0026, 0x0031, 0x00eb, 0x01f7}, {0x01e8, 0x006f, 0x002e, 0x0008, 0x0004, 0x0006, 0x0029, 0x006b, 0x01ee}, {0x01ef, 0x0072, 0x002d, 0x0002, 0x0000, 0x0003, 0x002f, 0x0073, 0x01fa}, {0x01e7, 0x006e, 0x002b, 0x0007, 0x0001, 0x0005, 0x002c, 0x006d, 0x01ec}, {0x01f9, 0x00ee, 0x0030, 0x0024, 0x002a, 0x0025, 0x0033, 0x00ec, 0x01f2}, {0x03f8, 0x01e4, 0x00ed, 0x006a, 0x0070, 0x0069, 0x0074, 0x00f1, 0x03fa}, {0x07ff, 0x03f9, 0x01f6, 0x01ed, 0x01f8, 0x01e9, 0x01f5, 0x03fb, 0x07fc}}; const USHORT FDKaacEnc_huff_ctab7[8][8] = { {0x0000, 0x0005, 0x0037, 0x0074, 0x00f2, 0x01eb, 0x03ed, 0x07f7}, {0x0004, 0x000c, 0x0035, 0x0071, 0x00ec, 0x00ee, 0x01ee, 0x01f5}, {0x0036, 0x0034, 0x0072, 0x00ea, 0x00f1, 0x01e9, 0x01f3, 0x03f5}, {0x0073, 0x0070, 0x00eb, 0x00f0, 0x01f1, 0x01f0, 0x03ec, 0x03fa}, {0x00f3, 0x00ed, 0x01e8, 0x01ef, 0x03ef, 0x03f1, 0x03f9, 0x07fb}, {0x01ed, 0x00ef, 0x01ea, 0x01f2, 0x03f3, 0x03f8, 0x07f9, 0x07fc}, {0x03ee, 0x01ec, 0x01f4, 0x03f4, 0x03f7, 0x07f8, 0x0ffd, 0x0ffe}, {0x07f6, 0x03f0, 0x03f2, 0x03f6, 0x07fa, 0x07fd, 0x0ffc, 0x0fff}}; const USHORT FDKaacEnc_huff_ctab8[8][8] = { {0x000e, 0x0005, 0x0010, 0x0030, 0x006f, 0x00f1, 0x01fa, 0x03fe}, {0x0003, 0x0000, 0x0004, 0x0012, 0x002c, 0x006a, 0x0075, 0x00f8}, {0x000f, 0x0002, 0x0006, 0x0014, 0x002e, 0x0069, 0x0072, 0x00f5}, {0x002f, 0x0011, 0x0013, 0x002a, 0x0032, 0x006c, 0x00ec, 0x00fa}, {0x0071, 0x002b, 0x002d, 0x0031, 0x006d, 0x0070, 0x00f2, 0x01f9}, {0x00ef, 0x0068, 0x0033, 0x006b, 0x006e, 0x00ee, 0x00f9, 0x03fc}, {0x01f8, 0x0074, 0x0073, 0x00ed, 0x00f0, 0x00f6, 0x01f6, 0x01fd}, {0x03fd, 0x00f3, 0x00f4, 0x00f7, 0x01f7, 0x01fb, 0x01fc, 0x03ff}}; const USHORT FDKaacEnc_huff_ctab9[13][13] = { {0x0000, 0x0005, 0x0037, 0x00e7, 0x01de, 0x03ce, 0x03d9, 0x07c8, 0x07cd, 0x0fc8, 0x0fdd, 0x1fe4, 0x1fec}, {0x0004, 0x000c, 0x0035, 0x0072, 0x00ea, 0x00ed, 0x01e2, 0x03d1, 0x03d3, 0x03e0, 0x07d8, 0x0fcf, 0x0fd5}, {0x0036, 0x0034, 0x0071, 0x00e8, 0x00ec, 0x01e1, 0x03cf, 0x03dd, 0x03db, 0x07d0, 0x0fc7, 0x0fd4, 0x0fe4}, {0x00e6, 0x0070, 0x00e9, 0x01dd, 0x01e3, 0x03d2, 0x03dc, 0x07cc, 0x07ca, 0x07de, 0x0fd8, 0x0fea, 0x1fdb}, {0x01df, 0x00eb, 0x01dc, 0x01e6, 0x03d5, 0x03de, 0x07cb, 0x07dd, 0x07dc, 0x0fcd, 0x0fe2, 0x0fe7, 0x1fe1}, {0x03d0, 0x01e0, 0x01e4, 0x03d6, 0x07c5, 0x07d1, 0x07db, 0x0fd2, 0x07e0, 0x0fd9, 0x0feb, 0x1fe3, 0x1fe9}, {0x07c4, 0x01e5, 0x03d7, 0x07c6, 0x07cf, 0x07da, 0x0fcb, 0x0fda, 0x0fe3, 0x0fe9, 0x1fe6, 0x1ff3, 0x1ff7}, {0x07d3, 0x03d8, 0x03e1, 0x07d4, 0x07d9, 0x0fd3, 0x0fde, 0x1fdd, 0x1fd9, 0x1fe2, 0x1fea, 0x1ff1, 0x1ff6}, {0x07d2, 0x03d4, 0x03da, 0x07c7, 0x07d7, 0x07e2, 0x0fce, 0x0fdb, 0x1fd8, 0x1fee, 0x3ff0, 0x1ff4, 0x3ff2}, {0x07e1, 0x03df, 0x07c9, 0x07d6, 0x0fca, 0x0fd0, 0x0fe5, 0x0fe6, 0x1feb, 0x1fef, 0x3ff3, 0x3ff4, 0x3ff5}, {0x0fe0, 0x07ce, 0x07d5, 0x0fc6, 0x0fd1, 0x0fe1, 0x1fe0, 0x1fe8, 0x1ff0, 0x3ff1, 0x3ff8, 0x3ff6, 0x7ffc}, {0x0fe8, 0x07df, 0x0fc9, 0x0fd7, 0x0fdc, 0x1fdc, 0x1fdf, 0x1fed, 0x1ff5, 0x3ff9, 0x3ffb, 0x7ffd, 0x7ffe}, {0x1fe7, 0x0fcc, 0x0fd6, 0x0fdf, 0x1fde, 0x1fda, 0x1fe5, 0x1ff2, 0x3ffa, 0x3ff7, 0x3ffc, 0x3ffd, 0x7fff}}; const USHORT FDKaacEnc_huff_ctab10[13][13] = { {0x0022, 0x0008, 0x001d, 0x0026, 0x005f, 0x00d3, 0x01cf, 0x03d0, 0x03d7, 0x03ed, 0x07f0, 0x07f6, 0x0ffd}, {0x0007, 0x0000, 0x0001, 0x0009, 0x0020, 0x0054, 0x0060, 0x00d5, 0x00dc, 0x01d4, 0x03cd, 0x03de, 0x07e7}, {0x001c, 0x0002, 0x0006, 0x000c, 0x001e, 0x0028, 0x005b, 0x00cd, 0x00d9, 0x01ce, 0x01dc, 0x03d9, 0x03f1}, {0x0025, 0x000b, 0x000a, 0x000d, 0x0024, 0x0057, 0x0061, 0x00cc, 0x00dd, 0x01cc, 0x01de, 0x03d3, 0x03e7}, {0x005d, 0x0021, 0x001f, 0x0023, 0x0027, 0x0059, 0x0064, 0x00d8, 0x00df, 0x01d2, 0x01e2, 0x03dd, 0x03ee}, {0x00d1, 0x0055, 0x0029, 0x0056, 0x0058, 0x0062, 0x00ce, 0x00e0, 0x00e2, 0x01da, 0x03d4, 0x03e3, 0x07eb}, {0x01c9, 0x005e, 0x005a, 0x005c, 0x0063, 0x00ca, 0x00da, 0x01c7, 0x01ca, 0x01e0, 0x03db, 0x03e8, 0x07ec}, {0x01e3, 0x00d2, 0x00cb, 0x00d0, 0x00d7, 0x00db, 0x01c6, 0x01d5, 0x01d8, 0x03ca, 0x03da, 0x07ea, 0x07f1}, {0x01e1, 0x00d4, 0x00cf, 0x00d6, 0x00de, 0x00e1, 0x01d0, 0x01d6, 0x03d1, 0x03d5, 0x03f2, 0x07ee, 0x07fb}, {0x03e9, 0x01cd, 0x01c8, 0x01cb, 0x01d1, 0x01d7, 0x01df, 0x03cf, 0x03e0, 0x03ef, 0x07e6, 0x07f8, 0x0ffa}, {0x03eb, 0x01dd, 0x01d3, 0x01d9, 0x01db, 0x03d2, 0x03cc, 0x03dc, 0x03ea, 0x07ed, 0x07f3, 0x07f9, 0x0ff9}, {0x07f2, 0x03ce, 0x01e4, 0x03cb, 0x03d8, 0x03d6, 0x03e2, 0x03e5, 0x07e8, 0x07f4, 0x07f5, 0x07f7, 0x0ffb}, {0x07fa, 0x03ec, 0x03df, 0x03e1, 0x03e4, 0x03e6, 0x03f0, 0x07e9, 0x07ef, 0x0ff8, 0x0ffe, 0x0ffc, 0x0fff}}; const USHORT FDKaacEnc_huff_ctab11[21][17] = { {0x0000, 0x0006, 0x0019, 0x003d, 0x009c, 0x00c6, 0x01a7, 0x0390, 0x03c2, 0x03df, 0x07e6, 0x07f3, 0x0ffb, 0x07ec, 0x0ffa, 0x0ffe, 0x038e}, {0x0005, 0x0001, 0x0008, 0x0014, 0x0037, 0x0042, 0x0092, 0x00af, 0x0191, 0x01a5, 0x01b5, 0x039e, 0x03c0, 0x03a2, 0x03cd, 0x07d6, 0x00ae}, {0x0017, 0x0007, 0x0009, 0x0018, 0x0039, 0x0040, 0x008e, 0x00a3, 0x00b8, 0x0199, 0x01ac, 0x01c1, 0x03b1, 0x0396, 0x03be, 0x03ca, 0x009d}, {0x003c, 0x0015, 0x0016, 0x001a, 0x003b, 0x0044, 0x0091, 0x00a5, 0x00be, 0x0196, 0x01ae, 0x01b9, 0x03a1, 0x0391, 0x03a5, 0x03d5, 0x0094}, {0x009a, 0x0036, 0x0038, 0x003a, 0x0041, 0x008c, 0x009b, 0x00b0, 0x00c3, 0x019e, 0x01ab, 0x01bc, 0x039f, 0x038f, 0x03a9, 0x03cf, 0x0093}, {0x00bf, 0x003e, 0x003f, 0x0043, 0x0045, 0x009e, 0x00a7, 0x00b9, 0x0194, 0x01a2, 0x01ba, 0x01c3, 0x03a6, 0x03a7, 0x03bb, 0x03d4, 0x009f}, {0x01a0, 0x008f, 0x008d, 0x0090, 0x0098, 0x00a6, 0x00b6, 0x00c4, 0x019f, 0x01af, 0x01bf, 0x0399, 0x03bf, 0x03b4, 0x03c9, 0x03e7, 0x00a8}, {0x01b6, 0x00ab, 0x00a4, 0x00aa, 0x00b2, 0x00c2, 0x00c5, 0x0198, 0x01a4, 0x01b8, 0x038c, 0x03a4, 0x03c4, 0x03c6, 0x03dd, 0x03e8, 0x00ad}, {0x03af, 0x0192, 0x00bd, 0x00bc, 0x018e, 0x0197, 0x019a, 0x01a3, 0x01b1, 0x038d, 0x0398, 0x03b7, 0x03d3, 0x03d1, 0x03db, 0x07dd, 0x00b4}, {0x03de, 0x01a9, 0x019b, 0x019c, 0x01a1, 0x01aa, 0x01ad, 0x01b3, 0x038b, 0x03b2, 0x03b8, 0x03ce, 0x03e1, 0x03e0, 0x07d2, 0x07e5, 0x00b7}, {0x07e3, 0x01bb, 0x01a8, 0x01a6, 0x01b0, 0x01b2, 0x01b7, 0x039b, 0x039a, 0x03ba, 0x03b5, 0x03d6, 0x07d7, 0x03e4, 0x07d8, 0x07ea, 0x00ba}, {0x07e8, 0x03a0, 0x01bd, 0x01b4, 0x038a, 0x01c4, 0x0392, 0x03aa, 0x03b0, 0x03bc, 0x03d7, 0x07d4, 0x07dc, 0x07db, 0x07d5, 0x07f0, 0x00c1}, {0x07fb, 0x03c8, 0x03a3, 0x0395, 0x039d, 0x03ac, 0x03ae, 0x03c5, 0x03d8, 0x03e2, 0x03e6, 0x07e4, 0x07e7, 0x07e0, 0x07e9, 0x07f7, 0x0190}, {0x07f2, 0x0393, 0x01be, 0x01c0, 0x0394, 0x0397, 0x03ad, 0x03c3, 0x03c1, 0x03d2, 0x07da, 0x07d9, 0x07df, 0x07eb, 0x07f4, 0x07fa, 0x0195}, {0x07f8, 0x03bd, 0x039c, 0x03ab, 0x03a8, 0x03b3, 0x03b9, 0x03d0, 0x03e3, 0x03e5, 0x07e2, 0x07de, 0x07ed, 0x07f1, 0x07f9, 0x07fc, 0x0193}, {0x0ffd, 0x03dc, 0x03b6, 0x03c7, 0x03cc, 0x03cb, 0x03d9, 0x03da, 0x07d3, 0x07e1, 0x07ee, 0x07ef, 0x07f5, 0x07f6, 0x0ffc, 0x0fff, 0x019d}, {0x01c2, 0x00b5, 0x00a1, 0x0096, 0x0097, 0x0095, 0x0099, 0x00a0, 0x00a2, 0x00ac, 0x00a9, 0x00b1, 0x00b3, 0x00bb, 0x00c0, 0x018f, 0x0004}, {0x0018, 0x002e, 0x0000, 0x005a, 0x00a5, 0x00f8, 0x00b7, 0x0094, 0x00f9, 0x004d, 0x0021, 0x002b, 0x004f, 0x007b, 0x00bc, 0x0046, 0x0015}, {0x0042, 0x0037, 0x0078, 0x000d, 0x0068, 0x005f, 0x000d, 0x005e, 0x005a, 0x00be, 0x0063, 0x007e, 0x001f, 0x0092, 0x001a, 0x00ab, 0x0032}, {0x00e6, 0x0037, 0x0000, 0x0058, 0x000b, 0x005a, 0x00e1, 0x005d, 0x0029, 0x0017, 0x007e, 0x0069, 0x00aa, 0x0054, 0x0029, 0x0032, 0x0041}, {0x0046, 0x00ea, 0x0034, 0x00ea, 0x0011, 0x001b, 0x00a9, 0x0094, 0x00e2, 0x0031, 0x00d0, 0x00e5, 0x0007, 0x0070, 0x0069, 0x003e, 0x0021}}; const ULONG FDKaacEnc_huff_ctabscf[121] = { 0x0003ffe8, 0x0003ffe6, 0x0003ffe7, 0x0003ffe5, 0x0007fff5, 0x0007fff1, 0x0007ffed, 0x0007fff6, 0x0007ffee, 0x0007ffef, 0x0007fff0, 0x0007fffc, 0x0007fffd, 0x0007ffff, 0x0007fffe, 0x0007fff7, 0x0007fff8, 0x0007fffb, 0x0007fff9, 0x0003ffe4, 0x0007fffa, 0x0003ffe3, 0x0001ffef, 0x0001fff0, 0x0000fff5, 0x0001ffee, 0x0000fff2, 0x0000fff3, 0x0000fff4, 0x0000fff1, 0x00007ff6, 0x00007ff7, 0x00003ff9, 0x00003ff5, 0x00003ff7, 0x00003ff3, 0x00003ff6, 0x00003ff2, 0x00001ff7, 0x00001ff5, 0x00000ff9, 0x00000ff7, 0x00000ff6, 0x000007f9, 0x00000ff4, 0x000007f8, 0x000003f9, 0x000003f7, 0x000003f5, 0x000001f8, 0x000001f7, 0x000000fa, 0x000000f8, 0x000000f6, 0x00000079, 0x0000003a, 0x00000038, 0x0000001a, 0x0000000b, 0x00000004, 0x00000000, 0x0000000a, 0x0000000c, 0x0000001b, 0x00000039, 0x0000003b, 0x00000078, 0x0000007a, 0x000000f7, 0x000000f9, 0x000001f6, 0x000001f9, 0x000003f4, 0x000003f6, 0x000003f8, 0x000007f5, 0x000007f4, 0x000007f6, 0x000007f7, 0x00000ff5, 0x00000ff8, 0x00001ff4, 0x00001ff6, 0x00001ff8, 0x00003ff8, 0x00003ff4, 0x0000fff0, 0x00007ff4, 0x0000fff6, 0x00007ff5, 0x0003ffe2, 0x0007ffd9, 0x0007ffda, 0x0007ffdb, 0x0007ffdc, 0x0007ffdd, 0x0007ffde, 0x0007ffd8, 0x0007ffd2, 0x0007ffd3, 0x0007ffd4, 0x0007ffd5, 0x0007ffd6, 0x0007fff2, 0x0007ffdf, 0x0007ffe7, 0x0007ffe8, 0x0007ffe9, 0x0007ffea, 0x0007ffeb, 0x0007ffe6, 0x0007ffe0, 0x0007ffe1, 0x0007ffe2, 0x0007ffe3, 0x0007ffe4, 0x0007ffe5, 0x0007ffd7, 0x0007ffec, 0x0007fff4, 0x0007fff3}; /* table of (0.50000...1.00000) ^0.75 */ const FIXP_QTD FDKaacEnc_mTab_3_4[MANT_SIZE] = { QTC(0x4c1bf829), QTC(0x4c3880de), QTC(0x4c550603), QTC(0x4c71879c), QTC(0x4c8e05aa), QTC(0x4caa8030), QTC(0x4cc6f72f), QTC(0x4ce36aab), QTC(0x4cffdaa4), QTC(0x4d1c471d), QTC(0x4d38b019), QTC(0x4d55159a), QTC(0x4d7177a1), QTC(0x4d8dd631), QTC(0x4daa314b), QTC(0x4dc688f3), QTC(0x4de2dd2a), QTC(0x4dff2df2), QTC(0x4e1b7b4d), QTC(0x4e37c53d), QTC(0x4e540bc5), QTC(0x4e704ee6), QTC(0x4e8c8ea3), QTC(0x4ea8cafd), QTC(0x4ec503f7), QTC(0x4ee13992), QTC(0x4efd6bd0), QTC(0x4f199ab4), QTC(0x4f35c640), QTC(0x4f51ee75), QTC(0x4f6e1356), QTC(0x4f8a34e4), QTC(0x4fa65321), QTC(0x4fc26e10), QTC(0x4fde85b2), QTC(0x4ffa9a0a), QTC(0x5016ab18), QTC(0x5032b8e0), QTC(0x504ec362), QTC(0x506acaa1), QTC(0x5086cea0), QTC(0x50a2cf5e), QTC(0x50becce0), QTC(0x50dac725), QTC(0x50f6be31), QTC(0x5112b205), QTC(0x512ea2a3), QTC(0x514a900d), QTC(0x51667a45), QTC(0x5182614c), QTC(0x519e4524), QTC(0x51ba25cf), QTC(0x51d60350), QTC(0x51f1dda7), QTC(0x520db4d6), QTC(0x522988e0), QTC(0x524559c6), QTC(0x52612789), QTC(0x527cf22d), QTC(0x5298b9b1), QTC(0x52b47e19), QTC(0x52d03f65), QTC(0x52ebfd98), QTC(0x5307b8b4), QTC(0x532370b9), QTC(0x533f25aa), QTC(0x535ad789), QTC(0x53768656), QTC(0x53923215), QTC(0x53addac6), QTC(0x53c9806b), QTC(0x53e52306), QTC(0x5400c298), QTC(0x541c5f24), QTC(0x5437f8ab), QTC(0x54538f2e), QTC(0x546f22af), QTC(0x548ab330), QTC(0x54a640b3), QTC(0x54c1cb38), QTC(0x54dd52c2), QTC(0x54f8d753), QTC(0x551458eb), QTC(0x552fd78d), QTC(0x554b5339), QTC(0x5566cbf3), QTC(0x558241bb), QTC(0x559db492), QTC(0x55b9247b), QTC(0x55d49177), QTC(0x55effb87), QTC(0x560b62ad), QTC(0x5626c6eb), QTC(0x56422842), QTC(0x565d86b4), QTC(0x5678e242), QTC(0x56943aee), QTC(0x56af90b9), QTC(0x56cae3a4), QTC(0x56e633b2), QTC(0x570180e4), QTC(0x571ccb3b), QTC(0x573812b8), QTC(0x5753575e), QTC(0x576e992e), QTC(0x5789d829), QTC(0x57a51450), QTC(0x57c04da6), QTC(0x57db842b), QTC(0x57f6b7e1), QTC(0x5811e8c9), QTC(0x582d16e6), QTC(0x58484238), QTC(0x58636ac0), QTC(0x587e9081), QTC(0x5899b37c), QTC(0x58b4d3b1), QTC(0x58cff123), QTC(0x58eb0bd3), QTC(0x590623c2), QTC(0x592138f2), QTC(0x593c4b63), QTC(0x59575b19), QTC(0x59726812), QTC(0x598d7253), QTC(0x59a879da), QTC(0x59c37eab), QTC(0x59de80c6), QTC(0x59f9802d), QTC(0x5a147ce0), QTC(0x5a2f76e2), QTC(0x5a4a6e34), QTC(0x5a6562d6), QTC(0x5a8054cb), QTC(0x5a9b4414), QTC(0x5ab630b2), QTC(0x5ad11aa6), QTC(0x5aec01f1), QTC(0x5b06e696), QTC(0x5b21c895), QTC(0x5b3ca7ef), QTC(0x5b5784a6), QTC(0x5b725ebc), QTC(0x5b8d3631), QTC(0x5ba80b06), QTC(0x5bc2dd3e), QTC(0x5bddacd9), QTC(0x5bf879d8), QTC(0x5c13443d), QTC(0x5c2e0c09), QTC(0x5c48d13e), QTC(0x5c6393dc), QTC(0x5c7e53e5), QTC(0x5c99115a), QTC(0x5cb3cc3c), QTC(0x5cce848d), QTC(0x5ce93a4e), QTC(0x5d03ed80), QTC(0x5d1e9e24), QTC(0x5d394c3b), QTC(0x5d53f7c7), QTC(0x5d6ea0c9), QTC(0x5d894742), QTC(0x5da3eb33), QTC(0x5dbe8c9e), QTC(0x5dd92b84), QTC(0x5df3c7e5), QTC(0x5e0e61c3), QTC(0x5e28f920), QTC(0x5e438dfc), QTC(0x5e5e2059), QTC(0x5e78b037), QTC(0x5e933d99), QTC(0x5eadc87e), QTC(0x5ec850e9), QTC(0x5ee2d6da), QTC(0x5efd5a53), QTC(0x5f17db54), QTC(0x5f3259e0), QTC(0x5f4cd5f6), QTC(0x5f674f99), QTC(0x5f81c6c8), QTC(0x5f9c3b87), QTC(0x5fb6add4), QTC(0x5fd11db3), QTC(0x5feb8b23), QTC(0x6005f626), QTC(0x60205ebd), QTC(0x603ac4e9), QTC(0x605528ac), QTC(0x606f8a05), QTC(0x6089e8f7), QTC(0x60a44583), QTC(0x60be9fa9), QTC(0x60d8f76b), QTC(0x60f34cca), QTC(0x610d9fc7), QTC(0x6127f062), QTC(0x61423e9e), QTC(0x615c8a7a), QTC(0x6176d3f9), QTC(0x61911b1b), QTC(0x61ab5fe1), QTC(0x61c5a24d), QTC(0x61dfe25f), QTC(0x61fa2018), QTC(0x62145b7a), QTC(0x622e9485), QTC(0x6248cb3b), QTC(0x6262ff9d), QTC(0x627d31ab), QTC(0x62976167), QTC(0x62b18ed1), QTC(0x62cbb9eb), QTC(0x62e5e2b6), QTC(0x63000933), QTC(0x631a2d62), QTC(0x63344f45), QTC(0x634e6edd), QTC(0x63688c2b), QTC(0x6382a730), QTC(0x639cbfec), QTC(0x63b6d661), QTC(0x63d0ea90), QTC(0x63eafc7a), QTC(0x64050c1f), QTC(0x641f1982), QTC(0x643924a2), QTC(0x64532d80), QTC(0x646d341f), QTC(0x6487387e), QTC(0x64a13a9e), QTC(0x64bb3a81), QTC(0x64d53828), QTC(0x64ef3393), QTC(0x65092cc4), QTC(0x652323bb), QTC(0x653d1879), QTC(0x65570b00), QTC(0x6570fb50), QTC(0x658ae96b), QTC(0x65a4d550), QTC(0x65bebf01), QTC(0x65d8a680), QTC(0x65f28bcc), QTC(0x660c6ee8), QTC(0x66264fd3), QTC(0x66402e8f), QTC(0x665a0b1c), QTC(0x6673e57d), QTC(0x668dbdb0), QTC(0x66a793b8), QTC(0x66c16795), QTC(0x66db3949), QTC(0x66f508d4), QTC(0x670ed636), QTC(0x6728a172), QTC(0x67426a87), QTC(0x675c3177), QTC(0x6775f643), QTC(0x678fb8eb), QTC(0x67a97971), QTC(0x67c337d5), QTC(0x67dcf418), QTC(0x67f6ae3b), QTC(0x6810663f), QTC(0x682a1c25), QTC(0x6843cfed), QTC(0x685d8199), QTC(0x68773129), QTC(0x6890de9f), QTC(0x68aa89fa), QTC(0x68c4333d), QTC(0x68ddda67), QTC(0x68f77f7a), QTC(0x69112277), QTC(0x692ac35e), QTC(0x69446230), QTC(0x695dfeee), QTC(0x6977999a), QTC(0x69913232), QTC(0x69aac8ba), QTC(0x69c45d31), QTC(0x69ddef98), QTC(0x69f77ff0), QTC(0x6a110e3a), QTC(0x6a2a9a77), QTC(0x6a4424a8), QTC(0x6a5daccc), QTC(0x6a7732e6), QTC(0x6a90b6f6), QTC(0x6aaa38fd), QTC(0x6ac3b8fb), QTC(0x6add36f2), QTC(0x6af6b2e2), QTC(0x6b102ccd), QTC(0x6b29a4b2), QTC(0x6b431a92), QTC(0x6b5c8e6f), QTC(0x6b76004a), QTC(0x6b8f7022), QTC(0x6ba8ddf9), QTC(0x6bc249d0), QTC(0x6bdbb3a7), QTC(0x6bf51b80), QTC(0x6c0e815a), QTC(0x6c27e537), QTC(0x6c414718), QTC(0x6c5aa6fd), QTC(0x6c7404e7), QTC(0x6c8d60d7), QTC(0x6ca6bace), QTC(0x6cc012cc), QTC(0x6cd968d2), QTC(0x6cf2bce1), QTC(0x6d0c0ef9), QTC(0x6d255f1d), QTC(0x6d3ead4b), QTC(0x6d57f985), QTC(0x6d7143cc), QTC(0x6d8a8c21), QTC(0x6da3d283), QTC(0x6dbd16f5), QTC(0x6dd65976), QTC(0x6def9a08), QTC(0x6e08d8ab), QTC(0x6e221560), QTC(0x6e3b5027), QTC(0x6e548902), QTC(0x6e6dbff1), QTC(0x6e86f4f5), QTC(0x6ea0280e), QTC(0x6eb9593e), QTC(0x6ed28885), QTC(0x6eebb5e3), QTC(0x6f04e15a), QTC(0x6f1e0aea), QTC(0x6f373294), QTC(0x6f505859), QTC(0x6f697c39), QTC(0x6f829e35), QTC(0x6f9bbe4e), QTC(0x6fb4dc85), QTC(0x6fcdf8d9), QTC(0x6fe7134d), QTC(0x70002be0), QTC(0x70194293), QTC(0x70325767), QTC(0x704b6a5d), QTC(0x70647b76), QTC(0x707d8ab1), QTC(0x70969811), QTC(0x70afa394), QTC(0x70c8ad3d), QTC(0x70e1b50c), QTC(0x70fabb01), QTC(0x7113bf1d), QTC(0x712cc161), QTC(0x7145c1ce), QTC(0x715ec064), QTC(0x7177bd24), QTC(0x7190b80f), QTC(0x71a9b124), QTC(0x71c2a866), QTC(0x71db9dd4), QTC(0x71f49170), QTC(0x720d8339), QTC(0x72267331), QTC(0x723f6159), QTC(0x72584db0), QTC(0x72713838), QTC(0x728a20f1), QTC(0x72a307db), QTC(0x72bbecf9), QTC(0x72d4d049), QTC(0x72edb1ce), QTC(0x73069187), QTC(0x731f6f75), QTC(0x73384b98), QTC(0x735125f3), QTC(0x7369fe84), QTC(0x7382d54d), QTC(0x739baa4e), QTC(0x73b47d89), QTC(0x73cd4efd), QTC(0x73e61eab), QTC(0x73feec94), QTC(0x7417b8b8), QTC(0x74308319), QTC(0x74494bb6), QTC(0x74621291), QTC(0x747ad7aa), QTC(0x74939b02), QTC(0x74ac5c98), QTC(0x74c51c6f), QTC(0x74ddda86), QTC(0x74f696de), QTC(0x750f5178), QTC(0x75280a54), QTC(0x7540c174), QTC(0x755976d7), QTC(0x75722a7e), QTC(0x758adc69), QTC(0x75a38c9b), QTC(0x75bc3b12), QTC(0x75d4e7cf), QTC(0x75ed92d4), QTC(0x76063c21), QTC(0x761ee3b6), QTC(0x76378994), QTC(0x76502dbc), QTC(0x7668d02e), QTC(0x768170eb), QTC(0x769a0ff3), QTC(0x76b2ad47), QTC(0x76cb48e7), QTC(0x76e3e2d5), QTC(0x76fc7b10), QTC(0x7715119a), QTC(0x772da673), QTC(0x7746399b), QTC(0x775ecb13), QTC(0x77775adc), QTC(0x778fe8f6), QTC(0x77a87561), QTC(0x77c1001f), QTC(0x77d98930), QTC(0x77f21095), QTC(0x780a964d), QTC(0x78231a5b), QTC(0x783b9cbd), QTC(0x78541d75), QTC(0x786c9c84), QTC(0x788519e9), QTC(0x789d95a6), QTC(0x78b60fbb), QTC(0x78ce8828), QTC(0x78e6feef), QTC(0x78ff740f), QTC(0x7917e78a), QTC(0x7930595f), QTC(0x7948c990), QTC(0x7961381d), QTC(0x7979a506), QTC(0x7992104c), QTC(0x79aa79f0), QTC(0x79c2e1f1), QTC(0x79db4852), QTC(0x79f3ad11), QTC(0x7a0c1031), QTC(0x7a2471b0), QTC(0x7a3cd191), QTC(0x7a552fd3), QTC(0x7a6d8c76), QTC(0x7a85e77d), QTC(0x7a9e40e6), QTC(0x7ab698b2), QTC(0x7aceeee3), QTC(0x7ae74378), QTC(0x7aff9673), QTC(0x7b17e7d2), QTC(0x7b303799), QTC(0x7b4885c5), QTC(0x7b60d259), QTC(0x7b791d55), QTC(0x7b9166b9), QTC(0x7ba9ae86), QTC(0x7bc1f4bc), QTC(0x7bda395c), QTC(0x7bf27c66), QTC(0x7c0abddb), QTC(0x7c22fdbb), QTC(0x7c3b3c07), QTC(0x7c5378c0), QTC(0x7c6bb3e5), QTC(0x7c83ed78), QTC(0x7c9c2579), QTC(0x7cb45be9), QTC(0x7ccc90c7), QTC(0x7ce4c414), QTC(0x7cfcf5d2), QTC(0x7d152600), QTC(0x7d2d549f), QTC(0x7d4581b0), QTC(0x7d5dad32), QTC(0x7d75d727), QTC(0x7d8dff8f), QTC(0x7da6266a), QTC(0x7dbe4bba), QTC(0x7dd66f7d), QTC(0x7dee91b6), QTC(0x7e06b264), QTC(0x7e1ed188), QTC(0x7e36ef22), QTC(0x7e4f0b34), QTC(0x7e6725bd), QTC(0x7e7f3ebd), QTC(0x7e975636), QTC(0x7eaf6c28), QTC(0x7ec78093), QTC(0x7edf9378), QTC(0x7ef7a4d7), QTC(0x7f0fb4b1), QTC(0x7f27c307), QTC(0x7f3fcfd8), QTC(0x7f57db25), QTC(0x7f6fe4ef), QTC(0x7f87ed36), QTC(0x7f9ff3fb), QTC(0x7fb7f93e), QTC(0x7fcffcff), QTC(0x7fe7ff40)}; /* table of pow(2.0,0.25*q)/2.0, q[0..4) */ const FIXP_QTD FDKaacEnc_quantTableQ[4] = {QTC(0x40000000), QTC(0x4c1bf7ff), QTC(0x5a82797f), QTC(0x6ba27e7f)}; /* table of pow(2.0,0.75*e)/8.0, e[0..4) */ const FIXP_QTD FDKaacEnc_quantTableE[4] = {QTC(0x10000000), QTC(0x1ae89f99), QTC(0x2d413ccd), QTC(0x4c1bf828)}; /* table to count used number of bits */ const SHORT FDKaacEnc_sideInfoTabLong[] = { 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x000e, 0x000e, 0x000e, 0x000e, 0x000e, 0x000e, 0x000e, 0x000e, 0x000e, 0x000e, 0x000e, 0x000e, 0x000e, 0x000e, 0x000e, 0x000e, 0x000e, 0x000e, 0x000e, 0x000e, 0x000e}; const SHORT FDKaacEnc_sideInfoTabShort[] = { 0x0007, 0x0007, 0x0007, 0x0007, 0x0007, 0x0007, 0x0007, 0x000a, 0x000a, 0x000a, 0x000a, 0x000a, 0x000a, 0x000a, 0x000d, 0x000d}; /* Psy Configuration constants */ const SFB_PARAM_LONG p_FDKaacEnc_8000_long_1024 = { 40, {12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28, 32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80}}; const SFB_PARAM_SHORT p_FDKaacEnc_8000_short_128 = { 15, {4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20}}; const SFB_PARAM_LONG p_FDKaacEnc_11025_long_1024 = { 43, {8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28, 32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64}}; const SFB_PARAM_SHORT p_FDKaacEnc_11025_short_128 = { 15, {4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20}}; const SFB_PARAM_LONG p_FDKaacEnc_12000_long_1024 = { 43, {8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28, 32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64}}; const SFB_PARAM_SHORT p_FDKaacEnc_12000_short_128 = { 15, {4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20}}; const SFB_PARAM_LONG p_FDKaacEnc_16000_long_1024 = { 43, {8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28, 32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64}}; const SFB_PARAM_SHORT p_FDKaacEnc_16000_short_128 = { 15, {4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20}}; const SFB_PARAM_LONG p_FDKaacEnc_22050_long_1024 = { 47, {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28, 32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64}}; const SFB_PARAM_SHORT p_FDKaacEnc_22050_short_128 = { 15, {4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20}}; const SFB_PARAM_LONG p_FDKaacEnc_24000_long_1024 = { 47, {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28, 32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64}}; const SFB_PARAM_SHORT p_FDKaacEnc_24000_short_128 = { 15, {4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20}}; const SFB_PARAM_LONG p_FDKaacEnc_32000_long_1024 = { 51, {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32}}; const SFB_PARAM_SHORT p_FDKaacEnc_32000_short_128 = { 14, {4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16}}; const SFB_PARAM_LONG p_FDKaacEnc_44100_long_1024 = { 49, {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 96}}; const SFB_PARAM_SHORT p_FDKaacEnc_44100_short_128 = { 14, {4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16}}; const SFB_PARAM_LONG p_FDKaacEnc_48000_long_1024 = { 49, {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 96}}; const SFB_PARAM_SHORT p_FDKaacEnc_48000_short_128 = { 14, {4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16}}; const SFB_PARAM_LONG p_FDKaacEnc_64000_long_1024 = { 47, {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40}}; const SFB_PARAM_SHORT p_FDKaacEnc_64000_short_128 = { 12, {4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36}}; const SFB_PARAM_LONG p_FDKaacEnc_88200_long_1024 = { 41, {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64}}; const SFB_PARAM_SHORT p_FDKaacEnc_88200_short_128 = { 12, {4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36}}; const SFB_PARAM_LONG p_FDKaacEnc_96000_long_1024 = { 41, {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64}}; const SFB_PARAM_SHORT p_FDKaacEnc_96000_short_128 = { 12, {4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36}}; /* TNS filter coefficients */ /* 3 bit resolution */ const FIXP_LPC FDKaacEnc_tnsEncCoeff3[8] = { FX_DBL2FXCONST_LPC(0x81f1d201), FX_DBL2FXCONST_LPC(0x91261481), FX_DBL2FXCONST_LPC(0xadb92301), FX_DBL2FXCONST_LPC(0xd438af00), FX_DBL2FXCONST_LPC(0x00000000), FX_DBL2FXCONST_LPC(0x37898080), FX_DBL2FXCONST_LPC(0x64130dff), FX_DBL2FXCONST_LPC(0x7cca6fff)}; const FIXP_LPC FDKaacEnc_tnsCoeff3Borders[8] = { FX_DBL2FXCONST_LPC(0x80000001) /*-4*/, FX_DBL2FXCONST_LPC(0x87b826df) /*-3*/, FX_DBL2FXCONST_LPC(0x9df24154) /*-2*/, FX_DBL2FXCONST_LPC(0xbfffffe5) /*-1*/, FX_DBL2FXCONST_LPC(0xe9c5e578) /* 0*/, FX_DBL2FXCONST_LPC(0x1c7b90f0) /* 1*/, FX_DBL2FXCONST_LPC(0x4fce83a9) /* 2*/, FX_DBL2FXCONST_LPC(0x7352f2c3) /* 3*/ }; /* 4 bit resolution */ const FIXP_LPC FDKaacEnc_tnsEncCoeff4[16] = { FX_DBL2FXCONST_LPC(0x808bc881), FX_DBL2FXCONST_LPC(0x84e2e581), FX_DBL2FXCONST_LPC(0x8d6b4a01), FX_DBL2FXCONST_LPC(0x99da9201), FX_DBL2FXCONST_LPC(0xa9c45701), FX_DBL2FXCONST_LPC(0xbc9dde81), FX_DBL2FXCONST_LPC(0xd1c2d500), FX_DBL2FXCONST_LPC(0xe87ae540), FX_DBL2FXCONST_LPC(0x00000000), FX_DBL2FXCONST_LPC(0x1a9cd9c0), FX_DBL2FXCONST_LPC(0x340ff240), FX_DBL2FXCONST_LPC(0x4b3c8bff), FX_DBL2FXCONST_LPC(0x5f1f5e7f), FX_DBL2FXCONST_LPC(0x6ed9eb7f), FX_DBL2FXCONST_LPC(0x79bc387f), FX_DBL2FXCONST_LPC(0x7f4c7e7f)}; const FIXP_LPC FDKaacEnc_tnsCoeff4Borders[16] = { FX_DBL2FXCONST_LPC(0x80000001) /*-8*/, FX_DBL2FXCONST_LPC(0x822deff0) /*-7*/, FX_DBL2FXCONST_LPC(0x88a4bfe6) /*-6*/, FX_DBL2FXCONST_LPC(0x932c159d) /*-5*/, FX_DBL2FXCONST_LPC(0xa16827c2) /*-4*/, FX_DBL2FXCONST_LPC(0xb2dcde27) /*-3*/, FX_DBL2FXCONST_LPC(0xc6f20b91) /*-2*/, FX_DBL2FXCONST_LPC(0xdcf89c64) /*-1*/, FX_DBL2FXCONST_LPC(0xf4308ce1) /* 0*/, FX_DBL2FXCONST_LPC(0x0d613054) /* 1*/, FX_DBL2FXCONST_LPC(0x278dde80) /* 2*/, FX_DBL2FXCONST_LPC(0x4000001b) /* 3*/, FX_DBL2FXCONST_LPC(0x55a6127b) /* 4*/, FX_DBL2FXCONST_LPC(0x678dde8f) /* 5*/, FX_DBL2FXCONST_LPC(0x74ef0ed7) /* 6*/, FX_DBL2FXCONST_LPC(0x7d33f0da) /* 7*/ }; const FIXP_DBL FDKaacEnc_mTab_4_3Elc[512] = { FL2FXCONST_DBL(0.3968502629920499), FL2FXCONST_DBL(0.3978840634868335), FL2FXCONST_DBL(0.3989185359354711), FL2FXCONST_DBL(0.3999536794661432), FL2FXCONST_DBL(0.4009894932098531), FL2FXCONST_DBL(0.4020259763004115), FL2FXCONST_DBL(0.4030631278744227), FL2FXCONST_DBL(0.4041009470712695), FL2FXCONST_DBL(0.4051394330330996), FL2FXCONST_DBL(0.4061785849048110), FL2FXCONST_DBL(0.4072184018340380), FL2FXCONST_DBL(0.4082588829711372), FL2FXCONST_DBL(0.4093000274691739), FL2FXCONST_DBL(0.4103418344839078), FL2FXCONST_DBL(0.4113843031737798), FL2FXCONST_DBL(0.4124274326998980), FL2FXCONST_DBL(0.4134712222260245), FL2FXCONST_DBL(0.4145156709185620), FL2FXCONST_DBL(0.4155607779465400), FL2FXCONST_DBL(0.4166065424816022), FL2FXCONST_DBL(0.4176529636979932), FL2FXCONST_DBL(0.4187000407725452), FL2FXCONST_DBL(0.4197477728846652), FL2FXCONST_DBL(0.4207961592163222), FL2FXCONST_DBL(0.4218451989520345), FL2FXCONST_DBL(0.4228948912788567), FL2FXCONST_DBL(0.4239452353863673), FL2FXCONST_DBL(0.4249962304666564), FL2FXCONST_DBL(0.4260478757143130), FL2FXCONST_DBL(0.4271001703264124), FL2FXCONST_DBL(0.4281531135025046), FL2FXCONST_DBL(0.4292067044446017), FL2FXCONST_DBL(0.4302609423571658), FL2FXCONST_DBL(0.4313158264470970), FL2FXCONST_DBL(0.4323713559237216), FL2FXCONST_DBL(0.4334275299987803), FL2FXCONST_DBL(0.4344843478864161), FL2FXCONST_DBL(0.4355418088031630), FL2FXCONST_DBL(0.4365999119679339), FL2FXCONST_DBL(0.4376586566020096), FL2FXCONST_DBL(0.4387180419290272), FL2FXCONST_DBL(0.4397780671749683), FL2FXCONST_DBL(0.4408387315681480), FL2FXCONST_DBL(0.4419000343392039), FL2FXCONST_DBL(0.4429619747210847), FL2FXCONST_DBL(0.4440245519490388), FL2FXCONST_DBL(0.4450877652606038), FL2FXCONST_DBL(0.4461516138955953), FL2FXCONST_DBL(0.4472160970960963), FL2FXCONST_DBL(0.4482812141064458), FL2FXCONST_DBL(0.4493469641732286), FL2FXCONST_DBL(0.4504133465452648), FL2FXCONST_DBL(0.4514803604735984), FL2FXCONST_DBL(0.4525480052114875), FL2FXCONST_DBL(0.4536162800143939), FL2FXCONST_DBL(0.4546851841399719), FL2FXCONST_DBL(0.4557547168480591), FL2FXCONST_DBL(0.4568248774006652), FL2FXCONST_DBL(0.4578956650619623), FL2FXCONST_DBL(0.4589670790982746), FL2FXCONST_DBL(0.4600391187780688), FL2FXCONST_DBL(0.4611117833719430), FL2FXCONST_DBL(0.4621850721526184), FL2FXCONST_DBL(0.4632589843949278), FL2FXCONST_DBL(0.4643335193758069), FL2FXCONST_DBL(0.4654086763742842), FL2FXCONST_DBL(0.4664844546714713), FL2FXCONST_DBL(0.4675608535505532), FL2FXCONST_DBL(0.4686378722967790), FL2FXCONST_DBL(0.4697155101974522), FL2FXCONST_DBL(0.4707937665419216), FL2FXCONST_DBL(0.4718726406215713), FL2FXCONST_DBL(0.4729521317298118), FL2FXCONST_DBL(0.4740322391620711), FL2FXCONST_DBL(0.4751129622157845), FL2FXCONST_DBL(0.4761943001903867), FL2FXCONST_DBL(0.4772762523873015), FL2FXCONST_DBL(0.4783588181099338), FL2FXCONST_DBL(0.4794419966636599), FL2FXCONST_DBL(0.4805257873558190), FL2FXCONST_DBL(0.4816101894957042), FL2FXCONST_DBL(0.4826952023945537), FL2FXCONST_DBL(0.4837808253655421), FL2FXCONST_DBL(0.4848670577237714), FL2FXCONST_DBL(0.4859538987862632), FL2FXCONST_DBL(0.4870413478719488), FL2FXCONST_DBL(0.4881294043016621), FL2FXCONST_DBL(0.4892180673981298), FL2FXCONST_DBL(0.4903073364859640), FL2FXCONST_DBL(0.4913972108916533), FL2FXCONST_DBL(0.4924876899435545), FL2FXCONST_DBL(0.4935787729718844), FL2FXCONST_DBL(0.4946704593087116), FL2FXCONST_DBL(0.4957627482879484), FL2FXCONST_DBL(0.4968556392453423), FL2FXCONST_DBL(0.4979491315184684), FL2FXCONST_DBL(0.4990432244467211), FL2FXCONST_DBL(0.5001379173713062), FL2FXCONST_DBL(0.5012332096352328), FL2FXCONST_DBL(0.5023291005833056), FL2FXCONST_DBL(0.5034255895621171), FL2FXCONST_DBL(0.5045226759200399), FL2FXCONST_DBL(0.5056203590072181), FL2FXCONST_DBL(0.5067186381755611), FL2FXCONST_DBL(0.5078175127787346), FL2FXCONST_DBL(0.5089169821721536), FL2FXCONST_DBL(0.5100170457129749), FL2FXCONST_DBL(0.5111177027600893), FL2FXCONST_DBL(0.5122189526741143), FL2FXCONST_DBL(0.5133207948173868), FL2FXCONST_DBL(0.5144232285539552), FL2FXCONST_DBL(0.5155262532495726), FL2FXCONST_DBL(0.5166298682716894), FL2FXCONST_DBL(0.5177340729894460), FL2FXCONST_DBL(0.5188388667736652), FL2FXCONST_DBL(0.5199442489968457), FL2FXCONST_DBL(0.5210502190331544), FL2FXCONST_DBL(0.5221567762584198), FL2FXCONST_DBL(0.5232639200501247), FL2FXCONST_DBL(0.5243716497873989), FL2FXCONST_DBL(0.5254799648510130), FL2FXCONST_DBL(0.5265888646233705), FL2FXCONST_DBL(0.5276983484885021), FL2FXCONST_DBL(0.5288084158320574), FL2FXCONST_DBL(0.5299190660412995), FL2FXCONST_DBL(0.5310302985050975), FL2FXCONST_DBL(0.5321421126139198), FL2FXCONST_DBL(0.5332545077598274), FL2FXCONST_DBL(0.5343674833364678), FL2FXCONST_DBL(0.5354810387390675), FL2FXCONST_DBL(0.5365951733644262), FL2FXCONST_DBL(0.5377098866109097), FL2FXCONST_DBL(0.5388251778784438), FL2FXCONST_DBL(0.5399410465685075), FL2FXCONST_DBL(0.5410574920841272), FL2FXCONST_DBL(0.5421745138298695), FL2FXCONST_DBL(0.5432921112118353), FL2FXCONST_DBL(0.5444102836376534), FL2FXCONST_DBL(0.5455290305164744), FL2FXCONST_DBL(0.5466483512589642), FL2FXCONST_DBL(0.5477682452772976), FL2FXCONST_DBL(0.5488887119851529), FL2FXCONST_DBL(0.5500097507977050), FL2FXCONST_DBL(0.5511313611316194), FL2FXCONST_DBL(0.5522535424050467), FL2FXCONST_DBL(0.5533762940376158), FL2FXCONST_DBL(0.5544996154504284), FL2FXCONST_DBL(0.5556235060660528), FL2FXCONST_DBL(0.5567479653085183), FL2FXCONST_DBL(0.5578729926033087), FL2FXCONST_DBL(0.5589985873773569), FL2FXCONST_DBL(0.5601247490590389), FL2FXCONST_DBL(0.5612514770781683), FL2FXCONST_DBL(0.5623787708659898), FL2FXCONST_DBL(0.5635066298551742), FL2FXCONST_DBL(0.5646350534798125), FL2FXCONST_DBL(0.5657640411754097), FL2FXCONST_DBL(0.5668935923788799), FL2FXCONST_DBL(0.5680237065285404), FL2FXCONST_DBL(0.5691543830641059), FL2FXCONST_DBL(0.5702856214266832), FL2FXCONST_DBL(0.5714174210587655), FL2FXCONST_DBL(0.5725497814042271), FL2FXCONST_DBL(0.5736827019083177), FL2FXCONST_DBL(0.5748161820176573), FL2FXCONST_DBL(0.5759502211802304), FL2FXCONST_DBL(0.5770848188453810), FL2FXCONST_DBL(0.5782199744638067), FL2FXCONST_DBL(0.5793556874875542), FL2FXCONST_DBL(0.5804919573700131), FL2FXCONST_DBL(0.5816287835659116), FL2FXCONST_DBL(0.5827661655313104), FL2FXCONST_DBL(0.5839041027235979), FL2FXCONST_DBL(0.5850425946014850), FL2FXCONST_DBL(0.5861816406250000), FL2FXCONST_DBL(0.5873212402554834), FL2FXCONST_DBL(0.5884613929555826), FL2FXCONST_DBL(0.5896020981892474), FL2FXCONST_DBL(0.5907433554217242), FL2FXCONST_DBL(0.5918851641195517), FL2FXCONST_DBL(0.5930275237505556), FL2FXCONST_DBL(0.5941704337838434), FL2FXCONST_DBL(0.5953138936897999), FL2FXCONST_DBL(0.5964579029400819), FL2FXCONST_DBL(0.5976024610076139), FL2FXCONST_DBL(0.5987475673665825), FL2FXCONST_DBL(0.5998932214924321), FL2FXCONST_DBL(0.6010394228618597), FL2FXCONST_DBL(0.6021861709528106), FL2FXCONST_DBL(0.6033334652444733), FL2FXCONST_DBL(0.6044813052172748), FL2FXCONST_DBL(0.6056296903528761), FL2FXCONST_DBL(0.6067786201341671), FL2FXCONST_DBL(0.6079280940452625), FL2FXCONST_DBL(0.6090781115714966), FL2FXCONST_DBL(0.6102286721994192), FL2FXCONST_DBL(0.6113797754167908), FL2FXCONST_DBL(0.6125314207125777), FL2FXCONST_DBL(0.6136836075769482), FL2FXCONST_DBL(0.6148363355012674), FL2FXCONST_DBL(0.6159896039780929), FL2FXCONST_DBL(0.6171434125011708), FL2FXCONST_DBL(0.6182977605654305), FL2FXCONST_DBL(0.6194526476669808), FL2FXCONST_DBL(0.6206080733031054), FL2FXCONST_DBL(0.6217640369722584), FL2FXCONST_DBL(0.6229205381740598), FL2FXCONST_DBL(0.6240775764092919), FL2FXCONST_DBL(0.6252351511798939), FL2FXCONST_DBL(0.6263932619889586), FL2FXCONST_DBL(0.6275519083407275), FL2FXCONST_DBL(0.6287110897405869), FL2FXCONST_DBL(0.6298708056950635), FL2FXCONST_DBL(0.6310310557118203), FL2FXCONST_DBL(0.6321918392996523), FL2FXCONST_DBL(0.6333531559684823), FL2FXCONST_DBL(0.6345150052293571), FL2FXCONST_DBL(0.6356773865944432), FL2FXCONST_DBL(0.6368402995770224), FL2FXCONST_DBL(0.6380037436914881), FL2FXCONST_DBL(0.6391677184533411), FL2FXCONST_DBL(0.6403322233791856), FL2FXCONST_DBL(0.6414972579867254), FL2FXCONST_DBL(0.6426628217947594), FL2FXCONST_DBL(0.6438289143231779), FL2FXCONST_DBL(0.6449955350929588), FL2FXCONST_DBL(0.6461626836261636), FL2FXCONST_DBL(0.6473303594459330), FL2FXCONST_DBL(0.6484985620764839), FL2FXCONST_DBL(0.6496672910431047), FL2FXCONST_DBL(0.6508365458721518), FL2FXCONST_DBL(0.6520063260910459), FL2FXCONST_DBL(0.6531766312282679), FL2FXCONST_DBL(0.6543474608133552), FL2FXCONST_DBL(0.6555188143768979), FL2FXCONST_DBL(0.6566906914505349), FL2FXCONST_DBL(0.6578630915669509), FL2FXCONST_DBL(0.6590360142598715), FL2FXCONST_DBL(0.6602094590640603), FL2FXCONST_DBL(0.6613834255153149), FL2FXCONST_DBL(0.6625579131504635), FL2FXCONST_DBL(0.6637329215073610), FL2FXCONST_DBL(0.6649084501248851), FL2FXCONST_DBL(0.6660844985429335), FL2FXCONST_DBL(0.6672610663024197), FL2FXCONST_DBL(0.6684381529452691), FL2FXCONST_DBL(0.6696157580144163), FL2FXCONST_DBL(0.6707938810538011), FL2FXCONST_DBL(0.6719725216083646), FL2FXCONST_DBL(0.6731516792240465), FL2FXCONST_DBL(0.6743313534477807), FL2FXCONST_DBL(0.6755115438274927), FL2FXCONST_DBL(0.6766922499120955), FL2FXCONST_DBL(0.6778734712514865), FL2FXCONST_DBL(0.6790552073965435), FL2FXCONST_DBL(0.6802374578991223), FL2FXCONST_DBL(0.6814202223120524), FL2FXCONST_DBL(0.6826035001891340), FL2FXCONST_DBL(0.6837872910851345), FL2FXCONST_DBL(0.6849715945557853), FL2FXCONST_DBL(0.6861564101577784), FL2FXCONST_DBL(0.6873417374487629), FL2FXCONST_DBL(0.6885275759873420), FL2FXCONST_DBL(0.6897139253330697), FL2FXCONST_DBL(0.6909007850464473), FL2FXCONST_DBL(0.6920881546889198), FL2FXCONST_DBL(0.6932760338228737), FL2FXCONST_DBL(0.6944644220116332), FL2FXCONST_DBL(0.6956533188194565), FL2FXCONST_DBL(0.6968427238115332), FL2FXCONST_DBL(0.6980326365539813), FL2FXCONST_DBL(0.6992230566138435), FL2FXCONST_DBL(0.7004139835590845), FL2FXCONST_DBL(0.7016054169585869), FL2FXCONST_DBL(0.7027973563821499), FL2FXCONST_DBL(0.7039898014004843), FL2FXCONST_DBL(0.7051827515852106), FL2FXCONST_DBL(0.7063762065088554), FL2FXCONST_DBL(0.7075701657448483), FL2FXCONST_DBL(0.7087646288675196), FL2FXCONST_DBL(0.7099595954520960), FL2FXCONST_DBL(0.7111550650746988), FL2FXCONST_DBL(0.7123510373123402), FL2FXCONST_DBL(0.7135475117429202), FL2FXCONST_DBL(0.7147444879452244), FL2FXCONST_DBL(0.7159419654989200), FL2FXCONST_DBL(0.7171399439845538), FL2FXCONST_DBL(0.7183384229835486), FL2FXCONST_DBL(0.7195374020782005), FL2FXCONST_DBL(0.7207368808516762), FL2FXCONST_DBL(0.7219368588880097), FL2FXCONST_DBL(0.7231373357720997), FL2FXCONST_DBL(0.7243383110897066), FL2FXCONST_DBL(0.7255397844274496), FL2FXCONST_DBL(0.7267417553728043), FL2FXCONST_DBL(0.7279442235140992), FL2FXCONST_DBL(0.7291471884405130), FL2FXCONST_DBL(0.7303506497420724), FL2FXCONST_DBL(0.7315546070096487), FL2FXCONST_DBL(0.7327590598349553), FL2FXCONST_DBL(0.7339640078105445), FL2FXCONST_DBL(0.7351694505298055), FL2FXCONST_DBL(0.7363753875869610), FL2FXCONST_DBL(0.7375818185770647), FL2FXCONST_DBL(0.7387887430959987), FL2FXCONST_DBL(0.7399961607404706), FL2FXCONST_DBL(0.7412040711080108), FL2FXCONST_DBL(0.7424124737969701), FL2FXCONST_DBL(0.7436213684065166), FL2FXCONST_DBL(0.7448307545366334), FL2FXCONST_DBL(0.7460406317881158), FL2FXCONST_DBL(0.7472509997625686), FL2FXCONST_DBL(0.7484618580624036), FL2FXCONST_DBL(0.7496732062908372), FL2FXCONST_DBL(0.7508850440518872), FL2FXCONST_DBL(0.7520973709503704), FL2FXCONST_DBL(0.7533101865919009), FL2FXCONST_DBL(0.7545234905828862), FL2FXCONST_DBL(0.7557372825305252), FL2FXCONST_DBL(0.7569515620428062), FL2FXCONST_DBL(0.7581663287285035), FL2FXCONST_DBL(0.7593815821971756), FL2FXCONST_DBL(0.7605973220591619), FL2FXCONST_DBL(0.7618135479255810), FL2FXCONST_DBL(0.7630302594083277), FL2FXCONST_DBL(0.7642474561200708), FL2FXCONST_DBL(0.7654651376742505), FL2FXCONST_DBL(0.7666833036850760), FL2FXCONST_DBL(0.7679019537675227), FL2FXCONST_DBL(0.7691210875373307), FL2FXCONST_DBL(0.7703407046110011), FL2FXCONST_DBL(0.7715608046057948), FL2FXCONST_DBL(0.7727813871397293), FL2FXCONST_DBL(0.7740024518315765), FL2FXCONST_DBL(0.7752239983008605), FL2FXCONST_DBL(0.7764460261678551), FL2FXCONST_DBL(0.7776685350535814), FL2FXCONST_DBL(0.7788915245798054), FL2FXCONST_DBL(0.7801149943690360), FL2FXCONST_DBL(0.7813389440445223), FL2FXCONST_DBL(0.7825633732302513), FL2FXCONST_DBL(0.7837882815509458), FL2FXCONST_DBL(0.7850136686320621), FL2FXCONST_DBL(0.7862395340997874), FL2FXCONST_DBL(0.7874658775810378), FL2FXCONST_DBL(0.7886926987034559), FL2FXCONST_DBL(0.7899199970954088), FL2FXCONST_DBL(0.7911477723859853), FL2FXCONST_DBL(0.7923760242049944), FL2FXCONST_DBL(0.7936047521829623), FL2FXCONST_DBL(0.7948339559511308), FL2FXCONST_DBL(0.7960636351414546), FL2FXCONST_DBL(0.7972937893865995), FL2FXCONST_DBL(0.7985244183199399), FL2FXCONST_DBL(0.7997555215755570), FL2FXCONST_DBL(0.8009870987882359), FL2FXCONST_DBL(0.8022191495934644), FL2FXCONST_DBL(0.8034516736274301), FL2FXCONST_DBL(0.8046846705270185), FL2FXCONST_DBL(0.8059181399298110), FL2FXCONST_DBL(0.8071520814740822), FL2FXCONST_DBL(0.8083864947987989), FL2FXCONST_DBL(0.8096213795436166), FL2FXCONST_DBL(0.8108567353488784), FL2FXCONST_DBL(0.8120925618556127), FL2FXCONST_DBL(0.8133288587055308), FL2FXCONST_DBL(0.8145656255410253), FL2FXCONST_DBL(0.8158028620051674), FL2FXCONST_DBL(0.8170405677417053), FL2FXCONST_DBL(0.8182787423950622), FL2FXCONST_DBL(0.8195173856103341), FL2FXCONST_DBL(0.8207564970332875), FL2FXCONST_DBL(0.8219960763103580), FL2FXCONST_DBL(0.8232361230886477), FL2FXCONST_DBL(0.8244766370159234), FL2FXCONST_DBL(0.8257176177406150), FL2FXCONST_DBL(0.8269590649118125), FL2FXCONST_DBL(0.8282009781792650), FL2FXCONST_DBL(0.8294433571933784), FL2FXCONST_DBL(0.8306862016052132), FL2FXCONST_DBL(0.8319295110664831), FL2FXCONST_DBL(0.8331732852295520), FL2FXCONST_DBL(0.8344175237474336), FL2FXCONST_DBL(0.8356622262737878), FL2FXCONST_DBL(0.8369073924629202), FL2FXCONST_DBL(0.8381530219697793), FL2FXCONST_DBL(0.8393991144499545), FL2FXCONST_DBL(0.8406456695596752), FL2FXCONST_DBL(0.8418926869558079), FL2FXCONST_DBL(0.8431401662958544), FL2FXCONST_DBL(0.8443881072379507), FL2FXCONST_DBL(0.8456365094408642), FL2FXCONST_DBL(0.8468853725639923), FL2FXCONST_DBL(0.8481346962673606), FL2FXCONST_DBL(0.8493844802116208), FL2FXCONST_DBL(0.8506347240580492), FL2FXCONST_DBL(0.8518854274685442), FL2FXCONST_DBL(0.8531365901056253), FL2FXCONST_DBL(0.8543882116324307), FL2FXCONST_DBL(0.8556402917127157), FL2FXCONST_DBL(0.8568928300108512), FL2FXCONST_DBL(0.8581458261918209), FL2FXCONST_DBL(0.8593992799212207), FL2FXCONST_DBL(0.8606531908652563), FL2FXCONST_DBL(0.8619075586907414), FL2FXCONST_DBL(0.8631623830650962), FL2FXCONST_DBL(0.8644176636563452), FL2FXCONST_DBL(0.8656734001331161), FL2FXCONST_DBL(0.8669295921646375), FL2FXCONST_DBL(0.8681862394207371), FL2FXCONST_DBL(0.8694433415718407), FL2FXCONST_DBL(0.8707008982889695), FL2FXCONST_DBL(0.8719589092437391), FL2FXCONST_DBL(0.8732173741083574), FL2FXCONST_DBL(0.8744762925556232), FL2FXCONST_DBL(0.8757356642589241), FL2FXCONST_DBL(0.8769954888922352), FL2FXCONST_DBL(0.8782557661301171), FL2FXCONST_DBL(0.8795164956477146), FL2FXCONST_DBL(0.8807776771207545), FL2FXCONST_DBL(0.8820393102255443), FL2FXCONST_DBL(0.8833013946389704), FL2FXCONST_DBL(0.8845639300384969), FL2FXCONST_DBL(0.8858269161021629), FL2FXCONST_DBL(0.8870903525085819), FL2FXCONST_DBL(0.8883542389369399), FL2FXCONST_DBL(0.8896185750669933), FL2FXCONST_DBL(0.8908833605790678), FL2FXCONST_DBL(0.8921485951540565), FL2FXCONST_DBL(0.8934142784734187), FL2FXCONST_DBL(0.8946804102191776), FL2FXCONST_DBL(0.8959469900739191), FL2FXCONST_DBL(0.8972140177207906), FL2FXCONST_DBL(0.8984814928434985), FL2FXCONST_DBL(0.8997494151263077), FL2FXCONST_DBL(0.9010177842540390), FL2FXCONST_DBL(0.9022865999120682), FL2FXCONST_DBL(0.9035558617863242), FL2FXCONST_DBL(0.9048255695632878), FL2FXCONST_DBL(0.9060957229299895), FL2FXCONST_DBL(0.9073663215740092), FL2FXCONST_DBL(0.9086373651834729), FL2FXCONST_DBL(0.9099088534470528), FL2FXCONST_DBL(0.9111807860539647), FL2FXCONST_DBL(0.9124531626939672), FL2FXCONST_DBL(0.9137259830573594), FL2FXCONST_DBL(0.9149992468349805), FL2FXCONST_DBL(0.9162729537182071), FL2FXCONST_DBL(0.9175471033989524), FL2FXCONST_DBL(0.9188216955696648), FL2FXCONST_DBL(0.9200967299233258), FL2FXCONST_DBL(0.9213722061534494), FL2FXCONST_DBL(0.9226481239540795), FL2FXCONST_DBL(0.9239244830197896), FL2FXCONST_DBL(0.9252012830456805), FL2FXCONST_DBL(0.9264785237273793), FL2FXCONST_DBL(0.9277562047610376), FL2FXCONST_DBL(0.9290343258433305), FL2FXCONST_DBL(0.9303128866714547), FL2FXCONST_DBL(0.9315918869431275), FL2FXCONST_DBL(0.9328713263565848), FL2FXCONST_DBL(0.9341512046105802), FL2FXCONST_DBL(0.9354315214043836), FL2FXCONST_DBL(0.9367122764377792), FL2FXCONST_DBL(0.9379934694110648), FL2FXCONST_DBL(0.9392751000250497), FL2FXCONST_DBL(0.9405571679810542), FL2FXCONST_DBL(0.9418396729809072), FL2FXCONST_DBL(0.9431226147269456), FL2FXCONST_DBL(0.9444059929220124), FL2FXCONST_DBL(0.9456898072694558), FL2FXCONST_DBL(0.9469740574731275), FL2FXCONST_DBL(0.9482587432373810), FL2FXCONST_DBL(0.9495438642670713), FL2FXCONST_DBL(0.9508294202675522), FL2FXCONST_DBL(0.9521154109446763), FL2FXCONST_DBL(0.9534018360047926), FL2FXCONST_DBL(0.9546886951547455), FL2FXCONST_DBL(0.9559759881018738), FL2FXCONST_DBL(0.9572637145540087), FL2FXCONST_DBL(0.9585518742194732), FL2FXCONST_DBL(0.9598404668070802), FL2FXCONST_DBL(0.9611294920261317), FL2FXCONST_DBL(0.9624189495864168), FL2FXCONST_DBL(0.9637088391982110), FL2FXCONST_DBL(0.9649991605722750), FL2FXCONST_DBL(0.9662899134198524), FL2FXCONST_DBL(0.9675810974526697), FL2FXCONST_DBL(0.9688727123829343), FL2FXCONST_DBL(0.9701647579233330), FL2FXCONST_DBL(0.9714572337870316), FL2FXCONST_DBL(0.9727501396876727), FL2FXCONST_DBL(0.9740434753393749), FL2FXCONST_DBL(0.9753372404567313), FL2FXCONST_DBL(0.9766314347548087), FL2FXCONST_DBL(0.9779260579491460), FL2FXCONST_DBL(0.9792211097557527), FL2FXCONST_DBL(0.9805165898911081), FL2FXCONST_DBL(0.9818124980721600), FL2FXCONST_DBL(0.9831088340163232), FL2FXCONST_DBL(0.9844055974414786), FL2FXCONST_DBL(0.9857027880659716), FL2FXCONST_DBL(0.9870004056086111), FL2FXCONST_DBL(0.9882984497886684), FL2FXCONST_DBL(0.9895969203258759), FL2FXCONST_DBL(0.9908958169404255), FL2FXCONST_DBL(0.9921951393529680), FL2FXCONST_DBL(0.9934948872846116), FL2FXCONST_DBL(0.9947950604569206), FL2FXCONST_DBL(0.9960956585919144), FL2FXCONST_DBL(0.9973966814120665), FL2FXCONST_DBL(0.9986981286403025)}; const FIXP_DBL FDKaacEnc_specExpMantTableCombElc[4][14] = { {FL2FXCONST_DBL(0.5000000000000000), FL2FXCONST_DBL(0.6299605249474366), FL2FXCONST_DBL(0.7937005259840998), FL2FXCONST_DBL(0.5000000000000000), FL2FXCONST_DBL(0.6299605249474366), FL2FXCONST_DBL(0.7937005259840998), FL2FXCONST_DBL(0.5000000000000000), FL2FXCONST_DBL(0.6299605249474366), FL2FXCONST_DBL(0.7937005259840998), FL2FXCONST_DBL(0.5000000000000000), FL2FXCONST_DBL(0.6299605249474366), FL2FXCONST_DBL(0.7937005259840998), FL2FXCONST_DBL(0.5000000000000000), FL2FXCONST_DBL(0.6299605249474366)}, {FL2FXCONST_DBL(0.5946035575013605), FL2FXCONST_DBL(0.7491535384383408), FL2FXCONST_DBL(0.9438743126816935), FL2FXCONST_DBL(0.5946035575013605), FL2FXCONST_DBL(0.7491535384383408), FL2FXCONST_DBL(0.9438743126816935), FL2FXCONST_DBL(0.5946035575013605), FL2FXCONST_DBL(0.7491535384383408), FL2FXCONST_DBL(0.9438743126816935), FL2FXCONST_DBL(0.5946035575013605), FL2FXCONST_DBL(0.7491535384383408), FL2FXCONST_DBL(0.9438743126816935), FL2FXCONST_DBL(0.5946035575013605), FL2FXCONST_DBL(0.7491535384383408)}, {FL2FXCONST_DBL(0.7071067811865476), FL2FXCONST_DBL(0.8908987181403393), FL2FXCONST_DBL(0.5612310241546865), FL2FXCONST_DBL(0.7071067811865476), FL2FXCONST_DBL(0.8908987181403393), FL2FXCONST_DBL(0.5612310241546865), FL2FXCONST_DBL(0.7071067811865476), FL2FXCONST_DBL(0.8908987181403393), FL2FXCONST_DBL(0.5612310241546865), FL2FXCONST_DBL(0.7071067811865476), FL2FXCONST_DBL(0.8908987181403393), FL2FXCONST_DBL(0.5612310241546865), FL2FXCONST_DBL(0.7071067811865476), FL2FXCONST_DBL(0.8908987181403393)}, {FL2FXCONST_DBL(0.8408964152537145), FL2FXCONST_DBL(0.5297315471796477), FL2FXCONST_DBL(0.6674199270850172), FL2FXCONST_DBL(0.8408964152537145), FL2FXCONST_DBL(0.5297315471796477), FL2FXCONST_DBL(0.6674199270850172), FL2FXCONST_DBL(0.8408964152537145), FL2FXCONST_DBL(0.5297315471796477), FL2FXCONST_DBL(0.6674199270850172), FL2FXCONST_DBL(0.8408964152537145), FL2FXCONST_DBL(0.5297315471796477), FL2FXCONST_DBL(0.6674199270850172), FL2FXCONST_DBL(0.8408964152537145), FL2FXCONST_DBL(0.5297315471796477)}}; const UCHAR FDKaacEnc_specExpTableComb[4][14] = { {1, 2, 3, 5, 6, 7, 9, 10, 11, 13, 14, 15, 17, 18}, {1, 2, 3, 5, 6, 7, 9, 10, 11, 13, 14, 15, 17, 18}, {1, 2, 4, 5, 6, 8, 9, 10, 12, 13, 14, 16, 17, 18}, {1, 3, 4, 5, 7, 8, 9, 11, 12, 13, 15, 16, 17, 19}}; #define WTS0 1 #define WTS1 0 #define WTS2 -2 const FIXP_WTB ELDAnalysis512[1536] = { /* part 0 */ WTC0(0xfac5a770), WTC0(0xfaafbab8), WTC0(0xfa996a40), WTC0(0xfa82bbd0), WTC0(0xfa6bb538), WTC0(0xfa545c38), WTC0(0xfa3cb698), WTC0(0xfa24ca28), WTC0(0xfa0c9ca8), WTC0(0xf9f433e8), WTC0(0xf9db9580), WTC0(0xf9c2c298), WTC0(0xf9a9b800), WTC0(0xf9907250), WTC0(0xf976ee38), WTC0(0xf95d2b88), WTC0(0xf9432d10), WTC0(0xf928f5c0), WTC0(0xf90e8868), WTC0(0xf8f3e400), WTC0(0xf8d903a0), WTC0(0xf8bde238), WTC0(0xf8a27af0), WTC0(0xf886cde8), WTC0(0xf86ae020), WTC0(0xf84eb6c0), WTC0(0xf83256f8), WTC0(0xf815c4b8), WTC0(0xf7f902c0), WTC0(0xf7dc13b0), WTC0(0xf7befa60), WTC0(0xf7a1ba40), WTC0(0xf78457c0), WTC0(0xf766d780), WTC0(0xf7493d90), WTC0(0xf72b8990), WTC0(0xf70db5f0), WTC0(0xf6efbd30), WTC0(0xf6d19a20), WTC0(0xf6b352e0), WTC0(0xf694f8c0), WTC0(0xf6769da0), WTC0(0xf6585310), WTC0(0xf63a28d0), WTC0(0xf61c2c60), WTC0(0xf5fe6b10), WTC0(0xf5e0f250), WTC0(0xf5c3ceb0), WTC0(0xf5a70be0), WTC0(0xf58ab5a0), WTC0(0xf56ed7b0), WTC0(0xf5537e40), WTC0(0xf538b610), WTC0(0xf51e8bf0), WTC0(0xf5050c90), WTC0(0xf4ec4330), WTC0(0xf4d439b0), WTC0(0xf4bcf9b0), WTC0(0xf4a68ce0), WTC0(0xf490fa80), WTC0(0xf47c4760), WTC0(0xf4687830), WTC0(0xf4558f00), WTC0(0xf4434fc0), WTC0(0xf4314070), WTC0(0xf41ee450), WTC0(0xf40bc130), WTC0(0xf3f799c0), WTC0(0xf3e26d30), WTC0(0xf3cc3d70), WTC0(0xf3b50c80), WTC0(0xf39cdd60), WTC0(0xf383b440), WTC0(0xf3699550), WTC0(0xf34e84c0), WTC0(0xf33286b0), WTC0(0xf3159f10), WTC0(0xf2f7d1b0), WTC0(0xf2d92290), WTC0(0xf2b994d0), WTC0(0xf2992ad0), WTC0(0xf277e6d0), WTC0(0xf255cb60), WTC0(0xf232dd00), WTC0(0xf20f2240), WTC0(0xf1eaa1d0), WTC0(0xf1c56240), WTC0(0xf19f63d0), WTC0(0xf178a0f0), WTC0(0xf15113a0), WTC0(0xf128b5c0), WTC0(0xf0ff7fd0), WTC0(0xf0d56860), WTC0(0xf0aa6610), WTC0(0xf07e6fd0), WTC0(0xf0518190), WTC0(0xf0239cd0), WTC0(0xeff4c320), WTC0(0xefc4f720), WTC0(0xef945080), WTC0(0xef62fce0), WTC0(0xef312a40), WTC0(0xeeff05c0), WTC0(0xeecca2c0), WTC0(0xee99faa0), WTC0(0xee6705a0), WTC0(0xee33bb60), WTC0(0xee000060), WTC0(0xedcba660), WTC0(0xed967e80), WTC0(0xed605b80), WTC0(0xed293b40), WTC0(0xecf146a0), WTC0(0xecb8a8a0), WTC0(0xec7f8bc0), WTC0(0xec461260), WTC0(0xec0c5720), WTC0(0xebd27440), WTC0(0xeb988220), WTC0(0xeb5e7040), WTC0(0xeb2404c0), WTC0(0xeae90440), WTC0(0xeaad33c0), WTC0(0xea7066c0), WTC0(0xea327f60), WTC0(0xe9f36000), WTC0(0xe9b2ed60), WTC0(0xe9713920), WTC0(0xe92e81e0), WTC0(0xe8eb08c0), WTC0(0xe8a70e60), WTC0(0xe862d8e0), WTC0(0xe81eb340), WTC0(0xe7dae8a0), WTC0(0xe797c1a0), WTC0(0xe7554ca0), WTC0(0xe7135dc0), WTC0(0xe6d1c6a0), WTC0(0xe6905720), WTC0(0xe64eb9c0), WTC0(0xe60c7300), WTC0(0xe5c90600), WTC0(0xe583f920), WTC0(0xe53d1ce0), WTC0(0xe4f48c80), WTC0(0xe4aa6640), WTC0(0xe45ecaa0), WTC0(0xe4120be0), WTC0(0xe3c4ae60), WTC0(0xe3773860), WTC0(0xe32a2ea0), WTC0(0xe2ddeea0), WTC0(0xe292af00), WTC0(0xe248a4a0), WTC0(0xe2000140), WTC0(0xe1b8b640), WTC0(0xe1727440), WTC0(0xe12ce900), WTC0(0xe0e7c280), WTC0(0xe0a2b420), WTC0(0xe05d76c0), WTC0(0xe017c360), WTC0(0xdfd15440), WTC0(0xdf8a0540), WTC0(0xdf41d300), WTC0(0xdef8bb40), WTC0(0xdeaebd40), WTC0(0xde63e7c0), WTC0(0xde185940), WTC0(0xddcc3180), WTC0(0xdd7f9000), WTC0(0xdd329e80), WTC0(0xdce58e80), WTC0(0xdc989300), WTC0(0xdc4bde40), WTC0(0xdbff96c0), WTC0(0xdbb3d780), WTC0(0xdb68bb80), WTC0(0xdb1e5c80), WTC0(0xdad4c380), WTC0(0xda8be840), WTC0(0xda43c1c0), WTC0(0xd9fc4740), WTC0(0xd9b56640), WTC0(0xd96f0440), WTC0(0xd9290600), WTC0(0xd8e35080), WTC0(0xd89dcd40), WTC0(0xd8586b40), WTC0(0xd8131940), WTC0(0xd7cdc640), WTC0(0xd7886180), WTC0(0xd742dc80), WTC0(0xd6fd2780), WTC0(0xd6b73400), WTC0(0xd670fd80), WTC0(0xd62a8a40), WTC0(0xd5e3e080), WTC0(0xd59d0840), WTC0(0xd5562b80), WTC0(0xd50f9540), WTC0(0xd4c992c0), WTC0(0xd4846f80), WTC0(0xd4405a80), WTC0(0xd3fd6580), WTC0(0xd3bba140), WTC0(0xd37b1c80), WTC0(0xd33bb780), WTC0(0xd2fd2400), WTC0(0xd2bf1240), WTC0(0xd2813300), WTC0(0xd2435ac0), WTC0(0xd2057fc0), WTC0(0xd1c79a00), WTC0(0xd189a240), WTC0(0xd14b9dc0), WTC0(0xd10d9e00), WTC0(0xd0cfb580), WTC0(0xd091f6c0), WTC0(0xd0548100), WTC0(0xd0177f40), WTC0(0xcfdb1cc0), WTC0(0xcf9f84c0), WTC0(0xcf64d780), WTC0(0xcf2b2b00), WTC0(0xcef29440), WTC0(0xcebb2640), WTC0(0xce84c000), WTC0(0xce4f0bc0), WTC0(0xce19b200), WTC0(0xcde45d40), WTC0(0xcdaeedc0), WTC0(0xcd7979c0), WTC0(0xcd4419c0), WTC0(0xcd0ee6c0), WTC0(0xccda0540), WTC0(0xcca5a500), WTC0(0xcc71f640), WTC0(0xcc3f2800), WTC0(0xcc0d4300), WTC0(0xcbdc2a00), WTC0(0xcbabbe80), WTC0(0xcb7be200), WTC0(0xcb4c8200), WTC0(0xcb1d9800), WTC0(0xcaef1d40), WTC0(0xcac10bc0), WTC0(0xca936440), WTC0(0xca662d00), WTC0(0xca396d40), WTC0(0xca0d2b80), WTC0(0xc9e16f80), WTC0(0xc9b63f80), WTC0(0xc98ba2c0), WTC0(0xc961a000), WTC0(0xc9383ec0), WTC0(0xc90a0440), WTC0(0xc8e0d280), WTC0(0xc8b73b80), WTC0(0xc88d4900), WTC0(0xc86304c0), WTC0(0xc83878c0), WTC0(0xc80dae80), WTC0(0xc7e2afc0), WTC0(0xc7b78640), WTC0(0xc78c3c40), WTC0(0xc760da80), WTC0(0xc7356640), WTC0(0xc709de40), WTC0(0xc6de41c0), WTC0(0xc6b28fc0), WTC0(0xc686bd40), WTC0(0xc65ab600), WTC0(0xc62e6580), WTC0(0xc601b880), WTC0(0xc5d4bac0), WTC0(0xc5a79640), WTC0(0xc57a76c0), WTC0(0xc54d8780), WTC0(0xc520e840), WTC0(0xc4f4acc0), WTC0(0xc4c8e880), WTC0(0xc49dad80), WTC0(0xc472e640), WTC0(0xc44856c0), WTC0(0xc41dc140), WTC0(0xc3f2e940), WTC0(0xc3c7bc00), WTC0(0xc39c4f00), WTC0(0xc370b9c0), WTC0(0xc34513c0), WTC0(0xc3197940), WTC0(0xc2ee0a00), WTC0(0xc2c2e640), WTC0(0xc2982d80), WTC0(0xc26df5c0), WTC0(0xc2444b00), WTC0(0xc21b3940), WTC0(0xc1f2cbc0), WTC0(0xc1cb05c0), WTC0(0xc1a3e340), WTC0(0xc17d5f00), WTC0(0xc15773c0), WTC0(0xc1320940), WTC0(0xc10cf480), WTC0(0xc0e80a00), WTC0(0xc0c31f00), WTC0(0xc09e2640), WTC0(0xc0792ec0), WTC0(0xc0544940), WTC0(0xc02f86c0), WTC0(0xc00b04c0), WTC0(0xbfe6ed01), WTC0(0xbfc36a01), WTC0(0xbfa0a581), WTC0(0xbf7eb581), WTC0(0xbf5d9a81), WTC0(0xbf3d5501), WTC0(0xbf1de601), WTC0(0xbeff4801), WTC0(0xbee17201), WTC0(0xbec45881), WTC0(0xbea7f301), WTC0(0xbe8c3781), WTC0(0xbe712001), WTC0(0xbe56a381), WTC0(0xbe3cbc01), WTC0(0xbe236001), WTC0(0xbe0a8581), WTC0(0xbdf22181), WTC0(0xbdda2a01), WTC0(0xbdc29a81), WTC0(0xbdab7181), WTC0(0xbd94b001), WTC0(0xbd7e5581), WTC0(0xbd686681), WTC0(0xbd52eb01), WTC0(0xbd3deb81), WTC0(0xbd297181), WTC0(0xbd158801), WTC0(0xbd023f01), WTC0(0xbcefa601), WTC0(0xbcddcc81), WTC0(0xbcccbd01), WTC0(0xbcbc7e01), WTC0(0xbcad1501), WTC0(0xbc9e8801), WTC0(0xbc90d481), WTC0(0xbc83f201), WTC0(0xbc77d601), WTC0(0xbc6c7781), WTC0(0xbc61c401), WTC0(0xbc57a301), WTC0(0xbc4dfb81), WTC0(0xbc44b481), WTC0(0xbc3bbc01), WTC0(0xbc330781), WTC0(0xbc2a8c81), WTC0(0xbc224181), WTC0(0xbc1a2401), WTC0(0xbc123b81), WTC0(0xbc0a8f01), WTC0(0xbc032601), WTC0(0xbbfc0f81), WTC0(0xbbf56181), WTC0(0xbbef3301), WTC0(0xbbe99981), WTC0(0xbbe49d01), WTC0(0xbbe03801), WTC0(0xbbdc6481), WTC0(0xbbd91b81), WTC0(0xbbd64d01), WTC0(0xbbd3e101), WTC0(0xbbd1bd81), WTC0(0xbbcfca81), WTC0(0xbbce0601), WTC0(0xbbcc8201), WTC0(0xbbcb5301), WTC0(0xbbca8d01), WTC0(0xbbca5081), WTC0(0xbbcaca01), WTC0(0xbbcc2681), WTC0(0xbbce9181), WTC0(0xbbd21281), WTC0(0xbbd68c81), WTC0(0xbbdbe201), WTC0(0xbbe1f401), WTC0(0xbbe89901), WTC0(0xbbef9b81), WTC0(0xbbf6c601), WTC0(0xbbfde481), WTC0(0xbc04e381), WTC0(0xbc0bcf81), WTC0(0xbc12b801), WTC0(0xbc19ab01), WTC0(0xbc20ae01), WTC0(0xbc27bd81), WTC0(0xbc2ed681), WTC0(0xbc35f501), WTC0(0xbc3d1801), WTC0(0xbc444081), WTC0(0xbc4b6e81), WTC0(0xbc52a381), WTC0(0xbc59df81), WTC0(0xbc612301), WTC0(0xbc686e01), WTC0(0xbc6fc101), WTC0(0xbc771c01), WTC0(0xbc7e7e01), WTC0(0xbc85e801), WTC0(0xbc8d5901), WTC0(0xbc94d201), WTC0(0xbc9c5281), WTC0(0xbca3db01), WTC0(0xbcab6c01), WTC0(0xbcb30601), WTC0(0xbcbaa801), WTC0(0xbcc25181), WTC0(0xbcca0301), WTC0(0xbcd1bb81), WTC0(0xbcd97c81), WTC0(0xbce14601), WTC0(0xbce91801), WTC0(0xbcf0f381), WTC0(0xbcf8d781), WTC0(0xbd00c381), WTC0(0xbd08b781), WTC0(0xbd10b381), WTC0(0xbd18b781), WTC0(0xbd20c401), WTC0(0xbd28d981), WTC0(0xbd30f881), WTC0(0xbd391f81), WTC0(0xbd414f01), WTC0(0xbd498601), WTC0(0xbd51c481), WTC0(0xbd5a0b01), WTC0(0xbd625981), WTC0(0xbd6ab101), WTC0(0xbd731081), WTC0(0xbd7b7781), WTC0(0xbd83e681), WTC0(0xbd8c5c01), WTC0(0xbd94d801), WTC0(0xbd9d5b81), WTC0(0xbda5e601), WTC0(0xbdae7881), WTC0(0xbdb71201), WTC0(0xbdbfb281), WTC0(0xbdc85981), WTC0(0xbdd10681), WTC0(0xbdd9b981), WTC0(0xbde27201), WTC0(0xbdeb3101), WTC0(0xbdf3f701), WTC0(0xbdfcc301), WTC0(0xbe059481), WTC0(0xbe0e6c01), WTC0(0xbe174781), WTC0(0xbe202801), WTC0(0xbe290d01), WTC0(0xbe31f701), WTC0(0xbe3ae601), WTC0(0xbe43da81), WTC0(0xbe4cd381), WTC0(0xbe55d001), WTC0(0xbe5ed081), WTC0(0xbe67d381), WTC0(0xbe70da01), WTC0(0xbe79e481), WTC0(0xbe82f301), WTC0(0xbe8c0501), WTC0(0xbe951a81), WTC0(0xbe9e3281), WTC0(0xbea74c81), WTC0(0xbeb06881), WTC0(0xbeb98681), WTC0(0xbec2a781), WTC0(0xbecbca81), WTC0(0xbed4f081), WTC0(0xbede1901), WTC0(0xbee74281), WTC0(0xbef06d01), WTC0(0xbef99901), WTC0(0xbf02c581), WTC0(0xbf0bf381), WTC0(0xbf152381), WTC0(0xbf1e5501), WTC0(0xbf278801), WTC0(0xbf30bb01), WTC0(0xbf39ee81), WTC0(0xbf432281), WTC0(0xbf4c5681), WTC0(0xbf558b01), WTC0(0xbf5ec101), WTC0(0xbf67f801), WTC0(0xbf712f01), WTC0(0xbf7a6681), WTC0(0xbf839d81), WTC0(0xbf8cd481), WTC0(0xbf960b01), WTC0(0xbf9f4181), WTC0(0xbfa87901), WTC0(0xbfb1b101), WTC0(0xbfbae981), WTC0(0xbfc42201), WTC0(0xbfcd5a01), WTC0(0xbfd69101), WTC0(0xbfdfc781), WTC0(0xbfe8fc01), WTC0(0xbff22f81), WTC0(0xbffb6081), /* part 1 */ WTC1(0x80093e01), WTC1(0x801b9b01), WTC1(0x802df701), WTC1(0x80405101), WTC1(0x8052a881), WTC1(0x8064fc81), WTC1(0x80774c81), WTC1(0x80899881), WTC1(0x809bdf01), WTC1(0x80ae1f81), WTC1(0x80c05a01), WTC1(0x80d28d81), WTC1(0x80e4bb81), WTC1(0x80f6e481), WTC1(0x81090981), WTC1(0x811b2981), WTC1(0x812d4481), WTC1(0x813f5981), WTC1(0x81516701), WTC1(0x81636d81), WTC1(0x81756d81), WTC1(0x81876781), WTC1(0x81995c01), WTC1(0x81ab4b01), WTC1(0x81bd3401), WTC1(0x81cf1581), WTC1(0x81e0ee81), WTC1(0x81f2bf81), WTC1(0x82048881), WTC1(0x82164a81), WTC1(0x82280581), WTC1(0x8239b981), WTC1(0x824b6601), WTC1(0x825d0901), WTC1(0x826ea201), WTC1(0x82803101), WTC1(0x8291b601), WTC1(0x82a33281), WTC1(0x82b4a601), WTC1(0x82c61101), WTC1(0x82d77201), WTC1(0x82e8c801), WTC1(0x82fa1181), WTC1(0x830b4f81), WTC1(0x831c8101), WTC1(0x832da781), WTC1(0x833ec381), WTC1(0x834fd481), WTC1(0x8360d901), WTC1(0x8371d081), WTC1(0x8382ba01), WTC1(0x83939501), WTC1(0x83a46181), WTC1(0x83b52101), WTC1(0x83c5d381), WTC1(0x83d67881), WTC1(0x83e70f01), WTC1(0x83f79681), WTC1(0x84080d81), WTC1(0x84187401), WTC1(0x8428ca01), WTC1(0x84391081), WTC1(0x84494881), WTC1(0x84597081), WTC1(0x84698881), WTC1(0x84798f81), WTC1(0x84898481), WTC1(0x84996701), WTC1(0x84a93801), WTC1(0x84b8f801), WTC1(0x84c8a701), WTC1(0x84d84601), WTC1(0x84e7d381), WTC1(0x84f74e01), WTC1(0x8506b581), WTC1(0x85160981), WTC1(0x85254a81), WTC1(0x85347901), WTC1(0x85439601), WTC1(0x8552a181), WTC1(0x85619a01), WTC1(0x85707f81), WTC1(0x857f5101), WTC1(0x858e0e01), WTC1(0x859cb781), WTC1(0x85ab4f01), WTC1(0x85b9d481), WTC1(0x85c84801), WTC1(0x85d6a981), WTC1(0x85e4f801), WTC1(0x85f33281), WTC1(0x86015981), WTC1(0x860f6e01), WTC1(0x861d7081), WTC1(0x862b6201), WTC1(0x86394301), WTC1(0x86471281), WTC1(0x8654d001), WTC1(0x86627b01), WTC1(0x86701381), WTC1(0x867d9a81), WTC1(0x868b1001), WTC1(0x86987581), WTC1(0x86a5ca81), WTC1(0x86b30f01), WTC1(0x86c04381), WTC1(0x86cd6681), WTC1(0x86da7901), WTC1(0x86e77b81), WTC1(0x86f46d81), WTC1(0x87014f81), WTC1(0x870e2301), WTC1(0x871ae981), WTC1(0x8727a381), WTC1(0x87345381), WTC1(0x8740f681), WTC1(0x874d8681), WTC1(0x8759fd01), WTC1(0x87665481), WTC1(0x87729701), WTC1(0x877ede01), WTC1(0x878b4301), WTC1(0x8797dd81), WTC1(0x87a48b01), WTC1(0x87b0ef01), WTC1(0x87bcab81), WTC1(0x87c76201), WTC1(0x87d0ca81), WTC1(0x87fdd781), WTC1(0x881dd301), WTC1(0x88423301), WTC1(0x886a8a81), WTC1(0x88962981), WTC1(0x88c45e81), WTC1(0x88f47901), WTC1(0x8925f101), WTC1(0x89586901), WTC1(0x898b8301), WTC1(0x89bee581), WTC1(0x89f26101), WTC1(0x8a25f301), WTC1(0x8a599a81), WTC1(0x8a8d5801), WTC1(0x8ac13381), WTC1(0x8af53e81), WTC1(0x8b298b81), WTC1(0x8b5e2c81), WTC1(0x8b933001), WTC1(0x8bc8a401), WTC1(0x8bfe9401), WTC1(0x8c350d01), WTC1(0x8c6c1b01), WTC1(0x8ca3cb01), WTC1(0x8cdc2901), WTC1(0x8d154081), WTC1(0x8d4f1b01), WTC1(0x8d89be81), WTC1(0x8dc53001), WTC1(0x8e017581), WTC1(0x8e3e9481), WTC1(0x8e7c9301), WTC1(0x8ebb7581), WTC1(0x8efb4181), WTC1(0x8f3bfb01), WTC1(0x8f7da401), WTC1(0x8fc03f01), WTC1(0x9003ce81), WTC1(0x90485401), WTC1(0x908dd101), WTC1(0x90d44781), WTC1(0x911bb981), WTC1(0x91642781), WTC1(0x91ad9281), WTC1(0x91f7f981), WTC1(0x92435d01), WTC1(0x928fbe01), WTC1(0x92dd1b01), WTC1(0x932b7501), WTC1(0x937acb01), WTC1(0x93cb1c81), WTC1(0x941c6901), WTC1(0x946eaf81), WTC1(0x94c1ee01), WTC1(0x95162381), WTC1(0x956b4f81), WTC1(0x95c17081), WTC1(0x96188501), WTC1(0x96708b81), WTC1(0x96c98381), WTC1(0x97236b01), WTC1(0x977e4181), WTC1(0x97da0481), WTC1(0x9836b201), WTC1(0x98944901), WTC1(0x98f2c601), WTC1(0x99522801), WTC1(0x99b26c81), WTC1(0x9a139101), WTC1(0x9a759301), WTC1(0x9ad87081), WTC1(0x9b3c2801), WTC1(0x9ba0b701), WTC1(0x9c061b81), WTC1(0x9c6c5481), WTC1(0x9cd35f81), WTC1(0x9d3b3b81), WTC1(0x9da3e601), WTC1(0x9e0d5e01), WTC1(0x9e779f81), WTC1(0x9ee2a901), WTC1(0x9f4e7801), WTC1(0x9fbb0981), WTC1(0xa0285d81), WTC1(0xa0967201), WTC1(0xa1054701), WTC1(0xa174da81), WTC1(0xa1e52a81), WTC1(0xa2563501), WTC1(0xa2c7f801), WTC1(0xa33a7201), WTC1(0xa3ada281), WTC1(0xa4218801), WTC1(0xa4962181), WTC1(0xa50b6e81), WTC1(0xa5816e81), WTC1(0xa5f81f81), WTC1(0xa66f8201), WTC1(0xa6e79401), WTC1(0xa7605601), WTC1(0xa7d9c681), WTC1(0xa853e501), WTC1(0xa8ceb201), WTC1(0xa94a2c01), WTC1(0xa9c65401), WTC1(0xaa432981), WTC1(0xaac0ad01), WTC1(0xab3edf01), WTC1(0xabbdc001), WTC1(0xac3d5001), WTC1(0xacbd9081), WTC1(0xad3e8101), WTC1(0xadc02281), WTC1(0xae427481), WTC1(0xaec57801), WTC1(0xaf492f01), WTC1(0xafcd9a81), WTC1(0xb052bc01), WTC1(0xb0d89401), WTC1(0xb15f2381), WTC1(0xb1e66a01), WTC1(0xb26e6881), WTC1(0xb2f71f01), WTC1(0xb3808d81), WTC1(0xb40ab501), WTC1(0xb4959501), WTC1(0xb5212e81), WTC1(0x4a6cf67f), WTC1(0x49dffeff), WTC1(0x495265ff), WTC1(0x48c4277f), WTC1(0x4835407f), WTC1(0x47a5aeff), WTC1(0x471570ff), WTC1(0x468484ff), WTC1(0x45f2eaff), WTC1(0x4560a2ff), WTC1(0x44cdad7f), WTC1(0x443a0c7f), WTC1(0x43a5c07f), WTC1(0x4310caff), WTC1(0x427b2bff), WTC1(0x41e4e3ff), WTC1(0x414df2ff), WTC1(0x40b6557f), WTC1(0x401e06ff), WTC1(0x3f8503c0), WTC1(0x3eeb4e00), WTC1(0x3e50ebc0), WTC1(0x3db5e680), WTC1(0x3d1a4680), WTC1(0x3c7e10c0), WTC1(0x3be14cc0), WTC1(0x3b4402c0), WTC1(0x3aa63800), WTC1(0x3a07e840), WTC1(0x39690880), WTC1(0x38c98700), WTC1(0x38295b40), WTC1(0x37888a80), WTC1(0x36e71d40), WTC1(0x36451d80), WTC1(0x35a29400), WTC1(0x34ff8800), WTC1(0x345c04c0), WTC1(0x33b81940), WTC1(0x3313d200), WTC1(0x326f3800), WTC1(0x31ca5600), WTC1(0x31253840), WTC1(0x307fe8c0), WTC1(0x2fda6e40), WTC1(0x2f34ce40), WTC1(0x2e8f0e40), WTC1(0x2de92ec0), WTC1(0x2d432780), WTC1(0x2c9cea40), WTC1(0x2bf66300), WTC1(0x2b4f88c0), WTC1(0x2aa864c0), WTC1(0x2a010240), WTC1(0x29596e40), WTC1(0x28b1ba80), WTC1(0x2809ff40), WTC1(0x27625b80), WTC1(0x26baf580), WTC1(0x2613e7c0), WTC1(0x256d3dc0), WTC1(0x24c70300), WTC1(0x24214380), WTC1(0x237c0800), WTC1(0x22d75400), WTC1(0x22332a80), WTC1(0x218f8cc0), WTC1(0x20ec7e40), WTC1(0x204a04c0), WTC1(0x1fa82540), WTC1(0x1f06e300), WTC1(0x1e664000), WTC1(0x1dc63bc0), WTC1(0x1d26d3c0), WTC1(0x1c8803a0), WTC1(0x1be9cc40), WTC1(0x1b4c34c0), WTC1(0x1aaf4480), WTC1(0x1a130260), WTC1(0x197774a0), WTC1(0x18dca260), WTC1(0x184294e0), WTC1(0x17a95840), WTC1(0x1710fd80), WTC1(0x16799ce0), WTC1(0x15e35340), WTC1(0x154e41a0), WTC1(0x14ba8360), WTC1(0x14282be0), WTC1(0x13975100), WTC1(0x13080aa0), WTC1(0x127a6240), WTC1(0x11ee50a0), WTC1(0x1163cc80), WTC1(0x10dacb20), WTC1(0x105333a0), WTC1(0x0fccdb30), WTC1(0x0f478f40), WTC1(0x0ec31700), WTC1(0x0e3f4e80), WTC1(0x0dbc27f0), WTC1(0x0d399000), WTC1(0x0cb76d00), WTC1(0x0c359d50), WTC1(0x0bb3fd50), WTC1(0x0b326bd0), WTC1(0x0ab0ca80), WTC1(0x0a2f0dc0), WTC1(0x09ad40c0), WTC1(0x092b7a90), WTC1(0x08a9db80), WTC1(0x08285c80), WTC1(0x07a6c7b8), WTC1(0x0724e4e0), WTC1(0x06a27b80), WTC1(0x061f52f8), WTC1(0x059b2ad0), WTC1(0x0515b568), WTC1(0x048ea058), WTC1(0x04066408), WTC1(0x037e52d8), WTC1(0x02f7d3c8), WTC1(0x0274614c), WTC1(0x01f63008), WTC1(0x0180403a), WTC1(0x0115c442), WTC1(0x00ba09e2), WTC1(0x006f077c), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), /* part 2 */ WTC2(0xfff36be1), WTC2(0xffdafbc1), WTC2(0xffc28035), WTC2(0xffa9fe8a), WTC2(0xff917c08), WTC2(0xff78fdfc), WTC2(0xff6089af), WTC2(0xff48246c), WTC2(0xff2fd37f), WTC2(0xff179c31), WTC2(0xfeff83b6), WTC2(0xfee78d18), WTC2(0xfecfb93e), WTC2(0xfeb808f2), WTC2(0xfea07d06), WTC2(0xfe8916b4), WTC2(0xfe71d7a0), WTC2(0xfe5ac174), WTC2(0xfe43d5d6), WTC2(0xfe2d167e), WTC2(0xfe16852e), WTC2(0xfe0023a6), WTC2(0xfde9f3f8), WTC2(0xfdd3ff7c), WTC2(0xfdbe56c0), WTC2(0xfda90aa8), WTC2(0xfd942b78), WTC2(0xfd7fbb20), WTC2(0xfd6bad50), WTC2(0xfd57f510), WTC2(0xfd44857c), WTC2(0xfd3153fc), WTC2(0xfd1e5840), WTC2(0xfd0b8a0c), WTC2(0xfcf8e180), WTC2(0xfce65eec), WTC2(0xfcd40ad0), WTC2(0xfcc1ee0c), WTC2(0xfcb011e8), WTC2(0xfc9e896c), WTC2(0xfc8d716c), WTC2(0xfc7ce720), WTC2(0xfc6d072c), WTC2(0xfc5de09c), WTC2(0xfc4f74e8), WTC2(0xfc41c4e8), WTC2(0xfc34d0dc), WTC2(0xfc288a68), WTC2(0xfc1cd49c), WTC2(0xfc1191e0), WTC2(0xfc06a4d0), WTC2(0xfbfbf3e8), WTC2(0xfbf16990), WTC2(0xfbe6f068), WTC2(0xfbdc7428), WTC2(0xfbd1fc68), WTC2(0xfbc7ac50), WTC2(0xfbbda868), WTC2(0xfbb41500), WTC2(0xfbab1438), WTC2(0xfba2c5f8), WTC2(0xfb9b4a00), WTC2(0xfb94bfa8), WTC2(0xfb8f3b48), WTC2(0xfb8ac638), WTC2(0xfb876970), WTC2(0xfb852d20), WTC2(0xfb840ae0), WTC2(0xfb83ed60), WTC2(0xfb84bec0), WTC2(0xfb866918), WTC2(0xfb88d4a8), WTC2(0xfb8be810), WTC2(0xfb8f89d0), WTC2(0xfb93a080), WTC2(0xfb981418), WTC2(0xfb9ccdf0), WTC2(0xfba1b770), WTC2(0xfba6bae0), WTC2(0xfbabd5c0), WTC2(0xfbb118d8), WTC2(0xfbb695c0), WTC2(0xfbbc5e90), WTC2(0xfbc29030), WTC2(0xfbc95268), WTC2(0xfbd0cd78), WTC2(0xfbd929c8), WTC2(0xfbe294d0), WTC2(0xfbed4108), WTC2(0xfbf96118), WTC2(0xfc0726c8), WTC2(0xfc16b064), WTC2(0xfc280890), WTC2(0xfc3b3920), WTC2(0xfc504a98), WTC2(0xfc67271c), WTC2(0xfc7f9a74), WTC2(0xfc996f18), WTC2(0xfcb46eb8), WTC2(0xfcd050b0), WTC2(0xfcecba24), WTC2(0xfd094f64), WTC2(0xfd25b720), WTC2(0xfd41ce40), WTC2(0xfd5da7f8), WTC2(0xfd7959d8), WTC2(0xfd94fb74), WTC2(0xfdb0d3fc), WTC2(0xfdcd5a34), WTC2(0xfdeb06e4), WTC2(0xfe0a5184), WTC2(0xfe2b92c4), WTC2(0xfe4f0486), WTC2(0xfe74df54), WTC2(0xfe9d5886), WTC2(0xfec85b92), WTC2(0xfef58a16), WTC2(0xff248275), WTC2(0xff54e401), WTC2(0xff866330), WTC2(0xffb8c99b), WTC2(0xffebe1c9), WTC2(0x001f786a), WTC2(0x00538bf9), WTC2(0x00884cbc), WTC2(0x00bded23), WTC2(0x00f49f54), WTC2(0x012c8ee4), WTC2(0x0165e0d2), WTC2(0x01a0b9d6), WTC2(0x01dd3d80), WTC2(0x021b74d4), WTC2(0x025b4e48), WTC2(0x029cb730), WTC2(0x02df9d0c), WTC2(0x0323f1a4), WTC2(0x0369ab00), WTC2(0x03b0bf5c), WTC2(0x03f925a0), WTC2(0x0442e3d8), WTC2(0x048e0f40), WTC2(0x04dabdb0), WTC2(0x05290430), WTC2(0x0578e428), WTC2(0x05ca4b60), WTC2(0x061d26c0), WTC2(0x067163d8), WTC2(0x06c6ff10), WTC2(0x071e03b0), WTC2(0x07767da0), WTC2(0x07d07918), WTC2(0x082c08e0), WTC2(0x08894660), WTC2(0x08e84b70), WTC2(0x094930b0), WTC2(0x09abf8d0), WTC2(0x0a109020), WTC2(0x0a76e210), WTC2(0x0adeda50), WTC2(0x0b486b80), WTC2(0x0bb38f00), WTC2(0x0c203e80), WTC2(0x0c8e73e0), WTC2(0x0cfe2c30), WTC2(0x0d6f6820), WTC2(0x0de22850), WTC2(0x0e566d90), WTC2(0x0ecc3dd0), WTC2(0x0f43a3a0), WTC2(0x0fbca9f0), WTC2(0x10375b80), WTC2(0x10b3be20), WTC2(0x1131d280), WTC2(0x11b19960), WTC2(0x123313a0), WTC2(0x12b64380), WTC2(0x133b2d00), WTC2(0x13c1d440), WTC2(0x144a3d60), WTC2(0x14d46900), WTC2(0x15605480), WTC2(0x15edfd20), WTC2(0x167d6040), WTC2(0x170e7e80), WTC2(0x17a15b80), WTC2(0x1835fb00), WTC2(0x18cc60a0), WTC2(0x19648dc0), WTC2(0x19fe80e0), WTC2(0x1a9a38a0), WTC2(0x1b37b3e0), WTC2(0x1bd6f400), WTC2(0x1c77fd20), WTC2(0x1d1ad400), WTC2(0x1dbf7c80), WTC2(0x1e65f820), WTC2(0x1f0e4540), WTC2(0x1fb861e0), WTC2(0x20644cc0), WTC2(0x21120640), WTC2(0x21c19240), WTC2(0x2272f480), WTC2(0x23263000), WTC2(0x23db4580), WTC2(0x24923340), WTC2(0x254af700), WTC2(0x26058e80), WTC2(0x26c1fa00), WTC2(0x27803d00), WTC2(0x28405a40), WTC2(0x29025500), WTC2(0x29c62d40), WTC2(0x2a8be0c0), WTC2(0x2b536cc0), WTC2(0x2c1ccf80), WTC2(0x2ce80840), WTC2(0x2db519c0), WTC2(0x2e840600), WTC2(0x2f54cf80), WTC2(0x302775c0), WTC2(0x30fbf640), WTC2(0x31d24e00), WTC2(0x32aa7a00), WTC2(0x338479c0), WTC2(0x34604e40), WTC2(0x353df900), WTC2(0x361d7ac0), WTC2(0x36fed200), WTC2(0x37e1fb40), WTC2(0x38c6f240), WTC2(0x39adb2c0), WTC2(0x3a963a00), WTC2(0x3b808740), WTC2(0x3c6c9880), WTC2(0x3d5a6cc0), WTC2(0x3e4a0040), WTC2(0x3f3b4bc0), WTC2(0x402e48ff), WTC2(0x4122f17f), WTC2(0x42193f7f), WTC2(0x43112eff), WTC2(0x440abbff), WTC2(0x4505e2ff), WTC2(0x46029e7f), WTC2(0x4700e9ff), WTC2(0x4800bfff), WTC2(0x49021bff), WTC2(0x4a050eff), WTC2(0x4b09bc7f), WTC2(0x4c104aff), WTC2(0x4d18df7f), WTC2(0x4e23a07f), WTC2(0x4f30b2ff), WTC2(0x50403c7f), WTC2(0x515262ff), WTC2(0x52674b7f), WTC2(0x001678b2), WTC2(0x00061a3b), WTC2(0xfffb4622), WTC2(0xfff5ea94), WTC2(0xfff5f5b9), WTC2(0xfffb55bd), WTC2(0x0005f8cb), WTC2(0x0015cd0c), WTC2(0x002ac0ac), WTC2(0x0044c1d5), WTC2(0x0063beb2), WTC2(0x0087a56d), WTC2(0x00b06431), WTC2(0x00dde929), WTC2(0x01102280), WTC2(0x0146fe5e), WTC2(0x01826af2), WTC2(0x01c25662), WTC2(0x0206aedc), WTC2(0x024f6288), WTC2(0x029c5f94), WTC2(0x02ed9424), WTC2(0x0342ee6c), WTC2(0x039c5c90), WTC2(0x03f9ccbc), WTC2(0x045b2d18), WTC2(0x04c06bd8), WTC2(0x05297718), WTC2(0x05963d10), WTC2(0x0606abe8), WTC2(0x067ab1c0), WTC2(0x06f23cd0), WTC2(0x076d3b40), WTC2(0x07eb9b38), WTC2(0x086d4ae0), WTC2(0x08f23860), WTC2(0x097a51f0), WTC2(0x0a0585b0), WTC2(0x0a93c1d0), WTC2(0x0b24f470), WTC2(0x0bb90bc0), WTC2(0x0c4ff5f0), WTC2(0x0ce9a130), WTC2(0x0d85fb90), WTC2(0x0e24f360), WTC2(0x0ec676b0), WTC2(0x0f6a73b0), WTC2(0x1010d880), WTC2(0x10b99360), WTC2(0x11649280), WTC2(0x1211c400), WTC2(0x12c115e0), WTC2(0x137276a0), WTC2(0x1425d420), WTC2(0x14db1ca0), WTC2(0x15923e60), WTC2(0x164b2780), WTC2(0x1705c620), WTC2(0x17c20860), WTC2(0x187fdca0), WTC2(0x193f30e0), WTC2(0x19fff340), WTC2(0x1ac21200), WTC2(0x1b857b40), WTC2(0x1c4a1d40), WTC2(0x1d0fe600), WTC2(0x1dd6c3e0), WTC2(0x1e9ea4e0), WTC2(0x1f677740), WTC2(0x20312940), WTC2(0x20fba8c0), WTC2(0x21c6e440), WTC2(0x2292c9c0), WTC2(0x235f4780), WTC2(0x242c4b80), WTC2(0x24f9c400), WTC2(0x25c79f40), WTC2(0x2695cb40), WTC2(0x27643680), WTC2(0x2832cec0), WTC2(0x29018240), WTC2(0x29d03f80), WTC2(0x2a9ef480), WTC2(0x2b6d8f00), WTC2(0x2c3bfdc0), WTC2(0x2d0a2ec0), WTC2(0x2dd81000), WTC2(0x2ea58fc0), WTC2(0x2f729c40), WTC2(0x303f2380), WTC2(0x310b1400), WTC2(0x31d65b80), WTC2(0x32a0e840), WTC2(0x336aa8c0), WTC2(0x34338ac0), WTC2(0x34fb7cc0), WTC2(0x35c26cc0), WTC2(0x36884900), WTC2(0x374cff80), WTC2(0x38107e80), WTC2(0x38d2b440), WTC2(0x39938ec0), WTC2(0x3a52fc40), WTC2(0x3b10eb00), WTC2(0x3bcd4900), WTC2(0x3c880480), WTC2(0x3d410bc0), WTC2(0x3df84d00), WTC2(0x3eadb600), WTC2(0x3f613540), WTC2(0x4012b8ff), WTC2(0x40c22eff), WTC2(0x416f85ff), WTC2(0x421aab7f), WTC2(0x42c38e7f), WTC2(0x436a1c7f), WTC2(0x440e437f), WTC2(0x44aff27f), WTC2(0x454f167f), WTC2(0x45eb9eff), WTC2(0x468578ff), WTC2(0x471c937f), WTC2(0x47b0dc7f), WTC2(0x484241ff), WTC2(0x48d0b1ff), WTC2(0x495c1a7f), WTC2(0x49e46a7f), WTC2(0x4a698f7f), WTC2(0x4aeb77ff), WTC2(0x4b6a11ff), WTC2(0x4be54b7f), WTC2(0x4c5d12ff), WTC2(0x4cd155ff), WTC2(0x4d4203ff), WTC2(0x4daf09ff), WTC2(0x4e18567f), WTC2(0x4e7dd77f), WTC2(0x4edf7b7f), WTC2(0x4f3d307f), WTC2(0x4f96e47f), WTC2(0x4fec85ff), WTC2(0x503e02ff), WTC2(0x508b497f), WTC2(0x50d447ff), WTC2(0x5118ec7f), WTC2(0x515924ff), WTC2(0x5194dfff), WTC2(0x51cc0b7f), WTC2(0x51fe95ff), WTC2(0x522c6cff), WTC2(0x52557eff), WTC2(0x5279b9ff), WTC2(0x52990c7f), WTC2(0x52b364ff), WTC2(0x52c8b07f), WTC2(0x52d8ddff), WTC2(0x52e3db7f), WTC2(0x52e996ff), WTC2(0x52e9ff7f), WTC2(0x52e501ff), WTC2(0x52da8cff), WTC2(0x52ca8f7f), WTC2(0x52b4f67f), WTC2(0x5299b07f), WTC2(0x5278ac7f), WTC2(0x5251d77f), WTC2(0x52251fff), WTC2(0x51f274ff), WTC2(0x51b9c37f), WTC2(0x517af9ff), WTC2(0x5136077f), WTC2(0x50ead8ff), WTC2(0x50995cff), WTC2(0x504181ff), WTC2(0x4fe335ff), WTC2(0x4f7e677f), WTC2(0x4f1303ff), WTC2(0x4ea0f9ff), WTC2(0x4e2837ff), WTC2(0x4da8ab7f), WTC2(0x4d2242ff), WTC2(0x4c94ecff), WTC2(0x4c0096ff), WTC2(0x4b652f7f), WTC2(0x4ac2a4ff), WTC2(0x4a18e4ff), WTC2(0x4967ddff), WTC2(0x48af7e7f), WTC2(0x47efb3ff), WTC2(0x47286cff), WTC2(0x4659ad7f), WTC2(0x45856f7f), WTC2(0x44afa3ff), WTC2(0x43dc507f), WTC2(0x430f657f), WTC2(0x424ad47f), WTC2(0x418e927f), WTC2(0x40da7bff), WTC2(0x402e6f7f), WTC2(0x3f8a3100), WTC2(0x3eed6f40), WTC2(0x3e57d700), WTC2(0x3dc914c0), WTC2(0x3d40cc40), WTC2(0x3cbe98c0), WTC2(0x3c421540), WTC2(0x3bcadbc0), WTC2(0x3b588880), WTC2(0x3aeab780), WTC2(0x3a810540), WTC2(0x3a1b0e00), WTC2(0x39b86d00), WTC2(0x3958bcc0), WTC2(0x38fb9700), WTC2(0x38a095c0), WTC2(0x38473d80), WTC2(0x37eeff40), WTC2(0x37974b40), WTC2(0x373f9500), WTC2(0x36e7ae00), WTC2(0x368fc4c0), WTC2(0x36380b80), WTC2(0x35e0b300), WTC2(0x3589c140), WTC2(0x35331180), WTC2(0x34dc7c80), WTC2(0x3485dc80), WTC2(0x342f1600), WTC2(0x33d81780), WTC2(0x3380d0c0), WTC2(0x33293100), WTC2(0x32d11800), WTC2(0x32785780), WTC2(0x321ec0c0), WTC2(0x31c42680), WTC2(0x316885c0), WTC2(0x310c0580), WTC2(0x30aecec0), WTC2(0x30510940), WTC2(0x2ff2b8c0), WTC2(0x2f93bf40), WTC2(0x2f33fc00), WTC2(0x2ed350c0), WTC2(0x2e71ba80), WTC2(0x2e0f5340), WTC2(0x2dac35c0), WTC2(0x2d487c80), WTC2(0x2ce431c0), WTC2(0x2c7f4fc0), WTC2(0x2c19d080), WTC2(0x2bb3ad80), WTC2(0x2b4ce080), WTC2(0x2ae56340), WTC2(0x2a7d2f80), WTC2(0x2a143f00), WTC2(0x29aa8b40)}; const FIXP_WTB ELDAnalysis480[1440] = { WTC0(0xfacfbef0), WTC0(0xfab88c18), WTC0(0xfaa0e520), WTC0(0xfa88d110), WTC0(0xfa7056e8), WTC0(0xfa577db0), WTC0(0xfa3e4c70), WTC0(0xfa24ca28), WTC0(0xfa0afde0), WTC0(0xf9f0eea0), WTC0(0xf9d6a2c8), WTC0(0xf9bc1ab8), WTC0(0xf9a15230), WTC0(0xf9864510), WTC0(0xf96af058), WTC0(0xf94f55c0), WTC0(0xf93378e0), WTC0(0xf9175d80), WTC0(0xf8fb0468), WTC0(0xf8de68b8), WTC0(0xf8c18438), WTC0(0xf8a450d8), WTC0(0xf886cde8), WTC0(0xf8690148), WTC0(0xf84af148), WTC0(0xf82ca410), WTC0(0xf80e1e18), WTC0(0xf7ef62a0), WTC0(0xf7d074e0), WTC0(0xf7b15870), WTC0(0xf7921240), WTC0(0xf772a7a0), WTC0(0xf7531e50), WTC0(0xf7337820), WTC0(0xf713afd0), WTC0(0xf6f3bea0), WTC0(0xf6d39dc0), WTC0(0xf6b352e0), WTC0(0xf692f280), WTC0(0xf6729250), WTC0(0xf65247a0), WTC0(0xf63224c0), WTC0(0xf6123a00), WTC0(0xf5f297c0), WTC0(0xf5d34dd0), WTC0(0xf5b46b10), WTC0(0xf595fd90), WTC0(0xf5781390), WTC0(0xf55abba0), WTC0(0xf53e0510), WTC0(0xf521ff70), WTC0(0xf506ba30), WTC0(0xf4ec4330), WTC0(0xf4d2a680), WTC0(0xf4b9efe0), WTC0(0xf4a22ac0), WTC0(0xf48b5f70), WTC0(0xf4759310), WTC0(0xf460cde0), WTC0(0xf44cfcc0), WTC0(0xf439aff0), WTC0(0xf4264e00), WTC0(0xf4123d90), WTC0(0xf3fd1370), WTC0(0xf3e6be00), WTC0(0xf3cf41a0), WTC0(0xf3b6a030), WTC0(0xf39cdd60), WTC0(0xf381fe00), WTC0(0xf3660760), WTC0(0xf348fe70), WTC0(0xf32ae820), WTC0(0xf30bc940), WTC0(0xf2eba690), WTC0(0xf2ca8480), WTC0(0xf2a86670), WTC0(0xf2854f40), WTC0(0xf2614190), WTC0(0xf23c41e0), WTC0(0xf21657a0), WTC0(0xf1ef8ae0), WTC0(0xf1c7e3e0), WTC0(0xf19f63d0), WTC0(0xf1760450), WTC0(0xf14bbdf0), WTC0(0xf1208960), WTC0(0xf0f45cd0), WTC0(0xf0c72ce0), WTC0(0xf098ee00), WTC0(0xf06996f0), WTC0(0xf0392620), WTC0(0xf0079e10), WTC0(0xefd4ffc0), WTC0(0xefa15ca0), WTC0(0xef6ce600), WTC0(0xef37d460), WTC0(0xef025f80), WTC0(0xeecca2c0), WTC0(0xee969760), WTC0(0xee603440), WTC0(0xee296d20), WTC0(0xedf21c00), WTC0(0xedba07e0), WTC0(0xed80f640), WTC0(0xed46bf40), WTC0(0xed0b7b00), WTC0(0xeccf5fc0), WTC0(0xec92a120), WTC0(0xec556d60), WTC0(0xec17e700), WTC0(0xebda2d40), WTC0(0xeb9c5fa0), WTC0(0xeb5e7040), WTC0(0xeb201b20), WTC0(0xeae117c0), WTC0(0xeaa12000), WTC0(0xea600180), WTC0(0xea1d9940), WTC0(0xe9d9c160), WTC0(0xe99468a0), WTC0(0xe94dc040), WTC0(0xe9061940), WTC0(0xe8bdc140), WTC0(0xe8750ae0), WTC0(0xe82c4fa0), WTC0(0xe7e3ea40), WTC0(0xe79c35e0), WTC0(0xe7554ca0), WTC0(0xe70efc00), WTC0(0xe6c90c20), WTC0(0xe6833f00), WTC0(0xe63d2300), WTC0(0xe5f620a0), WTC0(0xe5ad9dc0), WTC0(0xe5632080), WTC0(0xe5169da0), WTC0(0xe4c83e60), WTC0(0xe4782400), WTC0(0xe4269840), WTC0(0xe3d42dc0), WTC0(0xe38188c0), WTC0(0xe32f4be0), WTC0(0xe2ddeea0), WTC0(0xe28db520), WTC0(0xe23ee000), WTC0(0xe1f1a580), WTC0(0xe1a5e3a0), WTC0(0xe15b35a0), WTC0(0xe1113860), WTC0(0xe0c78a00), WTC0(0xe07dd0e0), WTC0(0xe033b7c0), WTC0(0xdfe8e680), WTC0(0xdf9d1fc0), WTC0(0xdf5055c0), WTC0(0xdf0287c0), WTC0(0xdeb3b340), WTC0(0xde63e7c0), WTC0(0xde134a00), WTC0(0xddc20000), WTC0(0xdd703180), WTC0(0xdd1e1280), WTC0(0xdccbe080), WTC0(0xdc79d980), WTC0(0xdc283600), WTC0(0xdbd71e00), WTC0(0xdb86b140), WTC0(0xdb3710c0), WTC0(0xdae850c0), WTC0(0xda9a6bc0), WTC0(0xda4d5640), WTC0(0xda010640), WTC0(0xd9b56640), WTC0(0xd96a5700), WTC0(0xd91fb700), WTC0(0xd8d56600), WTC0(0xd88b4a40), WTC0(0xd8414f00), WTC0(0xd7f75f80), WTC0(0xd7ad6740), WTC0(0xd76352c0), WTC0(0xd7191040), WTC0(0xd6ce8c80), WTC0(0xd683bd00), WTC0(0xd638a5c0), WTC0(0xd5ed4f80), WTC0(0xd5a1c240), WTC0(0xd5562b80), WTC0(0xd50ae500), WTC0(0xd4c04c80), WTC0(0xd476bb40), WTC0(0xd42e62c0), WTC0(0xd3e75680), WTC0(0xd3a1ad00), WTC0(0xd35d6780), WTC0(0xd31a4300), WTC0(0xd2d7dc00), WTC0(0xd295d080), WTC0(0xd253d8c0), WTC0(0xd211df40), WTC0(0xd1cfdbc0), WTC0(0xd18dc480), WTC0(0xd14b9dc0), WTC0(0xd1097c80), WTC0(0xd0c77700), WTC0(0xd085a500), WTC0(0xd0442f40), WTC0(0xd0034a80), WTC0(0xcfc32c00), WTC0(0xcf840400), WTC0(0xcf45f400), WTC0(0xcf0913c0), WTC0(0xcecd8000), WTC0(0xce932c80), WTC0(0xce59bf40), WTC0(0xce20cd40), WTC0(0xcde7ec40), WTC0(0xcdaeedc0), WTC0(0xcd75ea00), WTC0(0xcd3cfec0), WTC0(0xcd044b40), WTC0(0xcccbff00), WTC0(0xcc945480), WTC0(0xcc5d8780), WTC0(0xcc27c3c0), WTC0(0xcbf2fc40), WTC0(0xcbbf0a00), WTC0(0xcb8bc7c0), WTC0(0xcb591880), WTC0(0xcb26f0c0), WTC0(0xcaf54980), WTC0(0xcac41ac0), WTC0(0xca936440), WTC0(0xca632d80), WTC0(0xca337f00), WTC0(0xca046180), WTC0(0xc9d5dd40), WTC0(0xc9a7fa80), WTC0(0xc97ac200), WTC0(0xc94e3c00), WTC0(0xc91d1840), WTC0(0xc8f15980), WTC0(0xc8c52340), WTC0(0xc8988100), WTC0(0xc86b7f00), WTC0(0xc83e28c0), WTC0(0xc8108a80), WTC0(0xc7e2afc0), WTC0(0xc7b4a480), WTC0(0xc7867480), WTC0(0xc7582b40), WTC0(0xc729cc80), WTC0(0xc6fb5700), WTC0(0xc6ccca40), WTC0(0xc69e2180), WTC0(0xc66f49c0), WTC0(0xc64029c0), WTC0(0xc610a740), WTC0(0xc5e0bfc0), WTC0(0xc5b09e80), WTC0(0xc5807900), WTC0(0xc5508440), WTC0(0xc520e840), WTC0(0xc4f1bdc0), WTC0(0xc4c31d00), WTC0(0xc4951780), WTC0(0xc4678a00), WTC0(0xc43a28c0), WTC0(0xc40ca800), WTC0(0xc3deccc0), WTC0(0xc3b09940), WTC0(0xc3822c00), WTC0(0xc353a0c0), WTC0(0xc3251740), WTC0(0xc2f6b500), WTC0(0xc2c8a140), WTC0(0xc29b02c0), WTC0(0xc26df5c0), WTC0(0xc2418940), WTC0(0xc215cbc0), WTC0(0xc1eaca00), WTC0(0xc1c08680), WTC0(0xc196fb00), WTC0(0xc16e22c0), WTC0(0xc145f040), WTC0(0xc11e3a80), WTC0(0xc0f6cc00), WTC0(0xc0cf6ec0), WTC0(0xc0a802c0), WTC0(0xc0809280), WTC0(0xc0593340), WTC0(0xc031f880), WTC0(0xc00b04c0), WTC0(0xbfe48981), WTC0(0xbfbebb81), WTC0(0xbf99cb01), WTC0(0xbf75cc81), WTC0(0xbf52c101), WTC0(0xbf30a901), WTC0(0xbf0f8301), WTC0(0xbeef4601), WTC0(0xbecfe601), WTC0(0xbeb15701), WTC0(0xbe938c81), WTC0(0xbe767e81), WTC0(0xbe5a2301), WTC0(0xbe3e7201), WTC0(0xbe236001), WTC0(0xbe08e181), WTC0(0xbdeee981), WTC0(0xbdd56b81), WTC0(0xbdbc6381), WTC0(0xbda3d081), WTC0(0xbd8bb281), WTC0(0xbd740b81), WTC0(0xbd5ce281), WTC0(0xbd464281), WTC0(0xbd303581), WTC0(0xbd1ac801), WTC0(0xbd060c81), WTC0(0xbcf21601), WTC0(0xbcdef701), WTC0(0xbcccbd01), WTC0(0xbcbb7001), WTC0(0xbcab1781), WTC0(0xbc9bb901), WTC0(0xbc8d5101), WTC0(0xbc7fd301), WTC0(0xbc733401), WTC0(0xbc676501), WTC0(0xbc5c4c81), WTC0(0xbc51cb01), WTC0(0xbc47c281), WTC0(0xbc3e1981), WTC0(0xbc34c081), WTC0(0xbc2bab01), WTC0(0xbc22cd81), WTC0(0xbc1a2401), WTC0(0xbc11b681), WTC0(0xbc098d81), WTC0(0xbc01b381), WTC0(0xbbfa3c01), WTC0(0xbbf34281), WTC0(0xbbece281), WTC0(0xbbe73201), WTC0(0xbbe23281), WTC0(0xbbdddb01), WTC0(0xbbda2501), WTC0(0xbbd70201), WTC0(0xbbd45601), WTC0(0xbbd20301), WTC0(0xbbcfea81), WTC0(0xbbce0601), WTC0(0xbbcc6b01), WTC0(0xbbcb3201), WTC0(0xbbca7481), WTC0(0xbbca5d01), WTC0(0xbbcb2281), WTC0(0xbbccfc81), WTC0(0xbbd01301), WTC0(0xbbd45881), WTC0(0xbbd9a781), WTC0(0xbbdfdb81), WTC0(0xbbe6c801), WTC0(0xbbee2f81), WTC0(0xbbf5d181), WTC0(0xbbfd6c01), WTC0(0xbc04e381), WTC0(0xbc0c4581), WTC0(0xbc13a481), WTC0(0xbc1b1081), WTC0(0xbc228f01), WTC0(0xbc2a1a81), WTC0(0xbc31af01), WTC0(0xbc394901), WTC0(0xbc40e881), WTC0(0xbc488e81), WTC0(0xbc503b81), WTC0(0xbc57f101), WTC0(0xbc5fae81), WTC0(0xbc677501), WTC0(0xbc6f4401), WTC0(0xbc771c01), WTC0(0xbc7efc81), WTC0(0xbc86e581), WTC0(0xbc8ed701), WTC0(0xbc96d101), WTC0(0xbc9ed481), WTC0(0xbca6e101), WTC0(0xbcaef701), WTC0(0xbcb71701), WTC0(0xbcbf4001), WTC0(0xbcc77181), WTC0(0xbccfac01), WTC0(0xbcd7ef01), WTC0(0xbce03b81), WTC0(0xbce89281), WTC0(0xbcf0f381), WTC0(0xbcf95e81), WTC0(0xbd01d281), WTC0(0xbd0a4f81), WTC0(0xbd12d581), WTC0(0xbd1b6501), WTC0(0xbd23ff01), WTC0(0xbd2ca281), WTC0(0xbd355081), WTC0(0xbd3e0801), WTC0(0xbd46c801), WTC0(0xbd4f9101), WTC0(0xbd586281), WTC0(0xbd613d81), WTC0(0xbd6a2201), WTC0(0xbd731081), WTC0(0xbd7c0781), WTC0(0xbd850701), WTC0(0xbd8e0e01), WTC0(0xbd971c81), WTC0(0xbda03381), WTC0(0xbda95301), WTC0(0xbdb27b01), WTC0(0xbdbbab01), WTC0(0xbdc4e301), WTC0(0xbdce2181), WTC0(0xbdd76701), WTC0(0xbde0b301), WTC0(0xbdea0681), WTC0(0xbdf36101), WTC0(0xbdfcc301), WTC0(0xbe062b81), WTC0(0xbe0f9a01), WTC0(0xbe190d81), WTC0(0xbe228681), WTC0(0xbe2c0501), WTC0(0xbe358901), WTC0(0xbe3f1381), WTC0(0xbe48a301), WTC0(0xbe523781), WTC0(0xbe5bd001), WTC0(0xbe656c01), WTC0(0xbe6f0c01), WTC0(0xbe78b001), WTC0(0xbe825801), WTC0(0xbe8c0501), WTC0(0xbe95b581), WTC0(0xbe9f6901), WTC0(0xbea91f01), WTC0(0xbeb2d681), WTC0(0xbebc9181), WTC0(0xbec64e81), WTC0(0xbed00f81), WTC0(0xbed9d281), WTC0(0xbee39801), WTC0(0xbeed5f01), WTC0(0xbef72681), WTC0(0xbf00ef81), WTC0(0xbf0aba01), WTC0(0xbf148681), WTC0(0xbf1e5501), WTC0(0xbf282501), WTC0(0xbf31f501), WTC0(0xbf3bc601), WTC0(0xbf459681), WTC0(0xbf4f6801), WTC0(0xbf593a01), WTC0(0xbf630d81), WTC0(0xbf6ce201), WTC0(0xbf76b701), WTC0(0xbf808b81), WTC0(0xbf8a5f81), WTC0(0xbf943301), WTC0(0xbf9e0701), WTC0(0xbfa7dc01), WTC0(0xbfb1b101), WTC0(0xbfbb8701), WTC0(0xbfc55c81), WTC0(0xbfcf3181), WTC0(0xbfd90601), WTC0(0xbfe2d901), WTC0(0xbfecaa81), WTC0(0xbff67a01), /* part 1 */ WTC1(0x80130981), WTC1(0x80269f81), WTC1(0x803a3381), WTC1(0x804dc481), WTC1(0x80615281), WTC1(0x8074dc01), WTC1(0x80886081), WTC1(0x809bdf01), WTC1(0x80af5701), WTC1(0x80c2c781), WTC1(0x80d63101), WTC1(0x80e99401), WTC1(0x80fcf181), WTC1(0x81104a01), WTC1(0x81239d81), WTC1(0x8136ea01), WTC1(0x814a2f81), WTC1(0x815d6c01), WTC1(0x8170a181), WTC1(0x8183cf81), WTC1(0x8196f781), WTC1(0x81aa1981), WTC1(0x81bd3401), WTC1(0x81d04681), WTC1(0x81e34f81), WTC1(0x81f64f01), WTC1(0x82094581), WTC1(0x821c3401), WTC1(0x822f1b01), WTC1(0x8241fa01), WTC1(0x8254cf01), WTC1(0x82679901), WTC1(0x827a5801), WTC1(0x828d0b01), WTC1(0x829fb401), WTC1(0x82b25301), WTC1(0x82c4e801), WTC1(0x82d77201), WTC1(0x82e9ef01), WTC1(0x82fc5f01), WTC1(0x830ec081), WTC1(0x83211501), WTC1(0x83335c81), WTC1(0x83459881), WTC1(0x8357c701), WTC1(0x8369e781), WTC1(0x837bf801), WTC1(0x838df801), WTC1(0x839fe801), WTC1(0x83b1c881), WTC1(0x83c39a81), WTC1(0x83d55d01), WTC1(0x83e70f01), WTC1(0x83f8b001), WTC1(0x840a3e81), WTC1(0x841bb981), WTC1(0x842d2281), WTC1(0x843e7a81), WTC1(0x844fc081), WTC1(0x8460f581), WTC1(0x84721701), WTC1(0x84832481), WTC1(0x84941d81), WTC1(0x84a50201), WTC1(0x84b5d301), WTC1(0x84c69101), WTC1(0x84d73c01), WTC1(0x84e7d381), WTC1(0x84f85581), WTC1(0x8508c181), WTC1(0x85191801), WTC1(0x85295881), WTC1(0x85398481), WTC1(0x85499d01), WTC1(0x8559a081), WTC1(0x85698e81), WTC1(0x85796601), WTC1(0x85892681), WTC1(0x8598d081), WTC1(0x85a86581), WTC1(0x85b7e601), WTC1(0x85c75201), WTC1(0x85d6a981), WTC1(0x85e5eb81), WTC1(0x85f51681), WTC1(0x86042c01), WTC1(0x86132c01), WTC1(0x86221801), WTC1(0x8630f181), WTC1(0x863fb701), WTC1(0x864e6901), WTC1(0x865d0581), WTC1(0x866b8d81), WTC1(0x867a0081), WTC1(0x86886001), WTC1(0x8696ad01), WTC1(0x86a4e781), WTC1(0x86b30f01), WTC1(0x86c12401), WTC1(0x86cf2601), WTC1(0x86dd1481), WTC1(0x86eaf081), WTC1(0x86f8ba81), WTC1(0x87067281), WTC1(0x87141b01), WTC1(0x8721b481), WTC1(0x872f4201), WTC1(0x873cc201), WTC1(0x874a2f01), WTC1(0x87578181), WTC1(0x8764b101), WTC1(0x8771c601), WTC1(0x877ede01), WTC1(0x878c1881), WTC1(0x87998f01), WTC1(0x87a70e81), WTC1(0x87b42481), WTC1(0x87c05e81), WTC1(0x87cb5101), WTC1(0x87d4ac81), WTC1(0x87e73d81), WTC1(0x88124281), WTC1(0x88353501), WTC1(0x885f8481), WTC1(0x888d3181), WTC1(0x88be1681), WTC1(0x88f13801), WTC1(0x8925f101), WTC1(0x895bcd01), WTC1(0x89925a81), WTC1(0x89c92f81), WTC1(0x8a001f01), WTC1(0x8a372881), WTC1(0x8a6e4a01), WTC1(0x8aa58681), WTC1(0x8adcee01), WTC1(0x8b149701), WTC1(0x8b4c9701), WTC1(0x8b850281), WTC1(0x8bbde981), WTC1(0x8bf75b01), WTC1(0x8c316681), WTC1(0x8c6c1b01), WTC1(0x8ca78781), WTC1(0x8ce3ba81), WTC1(0x8d20c301), WTC1(0x8d5eaa01), WTC1(0x8d9d7781), WTC1(0x8ddd3201), WTC1(0x8e1de001), WTC1(0x8e5f8881), WTC1(0x8ea23201), WTC1(0x8ee5e301), WTC1(0x8f2aa101), WTC1(0x8f706f01), WTC1(0x8fb74f81), WTC1(0x8fff4601), WTC1(0x90485401), WTC1(0x90927b81), WTC1(0x90ddc001), WTC1(0x912a2201), WTC1(0x9177a301), WTC1(0x91c64301), WTC1(0x92160301), WTC1(0x9266e281), WTC1(0x92b8e101), WTC1(0x930bff81), WTC1(0x93603d01), WTC1(0x93b59901), WTC1(0x940c1281), WTC1(0x9463a881), WTC1(0x94bc5981), WTC1(0x95162381), WTC1(0x95710601), WTC1(0x95ccff01), WTC1(0x962a0c81), WTC1(0x96882e01), WTC1(0x96e76101), WTC1(0x9747a481), WTC1(0x97a8f681), WTC1(0x980b5501), WTC1(0x986ebd81), WTC1(0x98d32d81), WTC1(0x9938a281), WTC1(0x999f1981), WTC1(0x9a069001), WTC1(0x9a6f0381), WTC1(0x9ad87081), WTC1(0x9b42d581), WTC1(0x9bae2f81), WTC1(0x9c1a7c81), WTC1(0x9c87ba81), WTC1(0x9cf5e701), WTC1(0x9d650081), WTC1(0x9dd50481), WTC1(0x9e45f081), WTC1(0x9eb7c101), WTC1(0x9f2a7281), WTC1(0x9f9e0301), WTC1(0xa0127081), WTC1(0xa087b981), WTC1(0xa0fddd81), WTC1(0xa174da81), WTC1(0xa1ecae01), WTC1(0xa2655581), WTC1(0xa2dece81), WTC1(0xa3591801), WTC1(0xa3d43001), WTC1(0xa4501601), WTC1(0xa4ccc901), WTC1(0xa54a4701), WTC1(0xa5c89001), WTC1(0xa647a301), WTC1(0xa6c77e01), WTC1(0xa7482101), WTC1(0xa7c98b01), WTC1(0xa84bbb81), WTC1(0xa8ceb201), WTC1(0xa9526d81), WTC1(0xa9d6ef01), WTC1(0xaa5c3601), WTC1(0xaae24301), WTC1(0xab691681), WTC1(0xabf0b181), WTC1(0xac791401), WTC1(0xad023f01), WTC1(0xad8c3301), WTC1(0xae16f001), WTC1(0xaea27681), WTC1(0xaf2ec901), WTC1(0xafbbe801), WTC1(0xb049d601), WTC1(0xb0d89401), WTC1(0xb1682281), WTC1(0xb1f88181), WTC1(0xb289b181), WTC1(0xb31bb301), WTC1(0xb3ae8601), WTC1(0xb4422b81), WTC1(0xb4d6a381), WTC1(0x4a5a327f), WTC1(0x49c4adff), WTC1(0x492e637f), WTC1(0x48974f7f), WTC1(0x47ff6d7f), WTC1(0x4766baff), WTC1(0x46cd35ff), WTC1(0x4632dd7f), WTC1(0x4597b0ff), WTC1(0x44fbb1ff), WTC1(0x445eeaff), WTC1(0x43c165ff), WTC1(0x4323227f), WTC1(0x4284277f), WTC1(0x41e48aff), WTC1(0x4144557f), WTC1(0x40a3867f), WTC1(0x4001f5ff), WTC1(0x3f5f5d80), WTC1(0x3ebbad00), WTC1(0x3e16ee40), WTC1(0x3d713d00), WTC1(0x3ccab700), WTC1(0x3c236500), WTC1(0x3b7b5800), WTC1(0x3ad2ecc0), WTC1(0x3a2a6540), WTC1(0x3981b7c0), WTC1(0x38d8ba00), WTC1(0x382f01c0), WTC1(0x37846240), WTC1(0x36d8eb00), WTC1(0x362c9ec0), WTC1(0x357f7a00), WTC1(0x34d18340), WTC1(0x3422c900), WTC1(0x33736c40), WTC1(0x32c39040), WTC1(0x32134280), WTC1(0x31629280), WTC1(0x30b1a000), WTC1(0x30008380), WTC1(0x2f4f4240), WTC1(0x2e9df180), WTC1(0x2decc780), WTC1(0x2d3bd640), WTC1(0x2c8b0cc0), WTC1(0x2bda3080), WTC1(0x2b28ec80), WTC1(0x2a773500), WTC1(0x29c51b40), WTC1(0x291293c0), WTC1(0x285f9280), WTC1(0x27ac35c0), WTC1(0x26f8ab40), WTC1(0x26454c00), WTC1(0x25925600), WTC1(0x24dfd580), WTC1(0x242ddd40), WTC1(0x237c87c0), WTC1(0x22cbe240), WTC1(0x221bef40), WTC1(0x216cb040), WTC1(0x20be2800), WTC1(0x20105c80), WTC1(0x1f6352a0), WTC1(0x1eb71240), WTC1(0x1e0ba140), WTC1(0x1d60fe40), WTC1(0x1cb723e0), WTC1(0x1c0e0300), WTC1(0x1b6596c0), WTC1(0x1abde8a0), WTC1(0x1a16fbe0), WTC1(0x1970c680), WTC1(0x18cb4840), WTC1(0x18268e20), WTC1(0x1782a0c0), WTC1(0x16df8960), WTC1(0x163d6300), WTC1(0x159c52c0), WTC1(0x14fc87e0), WTC1(0x145e2c80), WTC1(0x13c15b60), WTC1(0x13263240), WTC1(0x128cd9a0), WTC1(0x11f562a0), WTC1(0x115fc1c0), WTC1(0x10cbf160), WTC1(0x1039f200), WTC1(0x0fa9a080), WTC1(0x0f1abd90), WTC1(0x0e8d01d0), WTC1(0x0e003330), WTC1(0x0d743590), WTC1(0x0ce8ef40), WTC1(0x0c5e1900), WTC1(0x0bd35d70), WTC1(0x0b488eb0), WTC1(0x0abd8410), WTC1(0x0a320a00), WTC1(0x09a60e70), WTC1(0x0919ab00), WTC1(0x088d0de0), WTC1(0x080065e0), WTC1(0x07739710), WTC1(0x06e65808), WTC1(0x06588348), WTC1(0x05ca0ae0), WTC1(0x053aaaf8), WTC1(0x04a9faf0), WTC1(0x0417f698), WTC1(0x03859ff4), WTC1(0x02f49be4), WTC1(0x0266b668), WTC1(0x01de554e), WTC1(0x015f50ca), WTC1(0x00eb7e5d), WTC1(0x00904f24), WTC1(0x00212889), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), /* part 2 */ WTC2(0xfffece02), WTC2(0xffe4c3df), WTC2(0xffcaaa55), WTC2(0xffb087d1), WTC2(0xff9662bf), WTC2(0xff7c418b), WTC2(0xff622aa0), WTC2(0xff48246c), WTC2(0xff2e355a), WTC2(0xff1463db), WTC2(0xfefab608), WTC2(0xfee12f0a), WTC2(0xfec7cfd2), WTC2(0xfeae995a), WTC2(0xfe958cc4), WTC2(0xfe7cabce), WTC2(0xfe63f882), WTC2(0xfe4b74e0), WTC2(0xfe3322f6), WTC2(0xfe1b04dc), WTC2(0xfe031ccc), WTC2(0xfdeb6cf0), WTC2(0xfdd3ff7c), WTC2(0xfdbce834), WTC2(0xfda63bb8), WTC2(0xfd900c68), WTC2(0xfd7a590c), WTC2(0xfd6511b4), WTC2(0xfd5026c0), WTC2(0xfd3b8954), WTC2(0xfd272df0), WTC2(0xfd130adc), WTC2(0xfcff15ac), WTC2(0xfceb4a68), WTC2(0xfcd7b110), WTC2(0xfcc454d0), WTC2(0xfcb14064), WTC2(0xfc9e896c), WTC2(0xfc8c5264), WTC2(0xfc7abef0), WTC2(0xfc69f078), WTC2(0xfc59f5e8), WTC2(0xfc4acfec), WTC2(0xfc3c8060), WTC2(0xfc2f0264), WTC2(0xfc223b7c), WTC2(0xfc160714), WTC2(0xfc0a4150), WTC2(0xfbfec920), WTC2(0xfbf38320), WTC2(0xfbe855d0), WTC2(0xfbdd2740), WTC2(0xfbd1fc68), WTC2(0xfbc6fea0), WTC2(0xfbbc5a48), WTC2(0xfbb23b48), WTC2(0xfba8ca78), WTC2(0xfba02e50), WTC2(0xfb988de0), WTC2(0xfb920b40), WTC2(0xfb8cb870), WTC2(0xfb889f68), WTC2(0xfb85cbe8), WTC2(0xfb843dd0), WTC2(0xfb83df78), WTC2(0xfb8495d0), WTC2(0xfb864660), WTC2(0xfb88d4a8), WTC2(0xfb8c21e8), WTC2(0xfb900f28), WTC2(0xfb947dc0), WTC2(0xfb9950c0), WTC2(0xfb9e6d08), WTC2(0xfba3b658), WTC2(0xfba91908), WTC2(0xfbae9e08), WTC2(0xfbb45bd0), WTC2(0xfbba66f8), WTC2(0xfbc0dcf0), WTC2(0xfbc7ead8), WTC2(0xfbcfc200), WTC2(0xfbd89330), WTC2(0xfbe294d0), WTC2(0xfbee03d0), WTC2(0xfbfb1de8), WTC2(0xfc0a1da4), WTC2(0xfc1b22e0), WTC2(0xfc2e38f0), WTC2(0xfc436d48), WTC2(0xfc5abf7c), WTC2(0xfc74024c), WTC2(0xfc8ef2e8), WTC2(0xfcab51ac), WTC2(0xfcc8d024), WTC2(0xfce704f0), WTC2(0xfd0580cc), WTC2(0xfd23d4d0), WTC2(0xfd41ce40), WTC2(0xfd5f81b0), WTC2(0xfd7d08f0), WTC2(0xfd9a8560), WTC2(0xfdb85938), WTC2(0xfdd71798), WTC2(0xfdf753b8), WTC2(0xfe1993ee), WTC2(0xfe3e30f8), WTC2(0xfe656cba), WTC2(0xfe8f8fdc), WTC2(0xfebca8a4), WTC2(0xfeec590e), WTC2(0xff1e285c), WTC2(0xff51a0b7), WTC2(0xff866330), WTC2(0xffbc2cbb), WTC2(0xfff2bbff), WTC2(0x0029d79d), WTC2(0x00618a22), WTC2(0x009a1185), WTC2(0x00d3aa8c), WTC2(0x010e8ff6), WTC2(0x014af29e), WTC2(0x0188fe56), WTC2(0x01c8e108), WTC2(0x020ab3c4), WTC2(0x024e68a8), WTC2(0x0293e824), WTC2(0x02db1bc8), WTC2(0x0323f1a4), WTC2(0x036e5d6c), WTC2(0x03ba5320), WTC2(0x0407c938), WTC2(0x0456cad0), WTC2(0x04a77288), WTC2(0x04f9db88), WTC2(0x054e1888), WTC2(0x05a41ef0), WTC2(0x05fbd6e0), WTC2(0x065528c0), WTC2(0x06b00838), WTC2(0x070c7ee0), WTC2(0x076a9bb0), WTC2(0x07ca6d10), WTC2(0x082c08e0), WTC2(0x088f8da0), WTC2(0x08f51ac0), WTC2(0x095ccc20), WTC2(0x09c69f70), WTC2(0x0a327b40), WTC2(0x0aa046d0), WTC2(0x0b0febb0), WTC2(0x0b815dd0), WTC2(0x0bf49600), WTC2(0x0c698c50), WTC2(0x0ce03ba0), WTC2(0x0d58a380), WTC2(0x0dd2c510), WTC2(0x0e4ea110), WTC2(0x0ecc3dd0), WTC2(0x0f4ba800), WTC2(0x0fcced10), WTC2(0x10501960), WTC2(0x10d532a0), WTC2(0x115c39c0), WTC2(0x11e52fa0), WTC2(0x12701560), WTC2(0x12fcef20), WTC2(0x138bc200), WTC2(0x141c9300), WTC2(0x14af64a0), WTC2(0x154434e0), WTC2(0x15db0020), WTC2(0x1673c360), WTC2(0x170e7e80), WTC2(0x17ab35e0), WTC2(0x1849ee40), WTC2(0x18eaaba0), WTC2(0x198d6f00), WTC2(0x1a3236a0), WTC2(0x1ad90080), WTC2(0x1b81cc60), WTC2(0x1c2c9da0), WTC2(0x1cd97980), WTC2(0x1d8865c0), WTC2(0x1e396540), WTC2(0x1eec7700), WTC2(0x1fa198c0), WTC2(0x2058c840), WTC2(0x21120640), WTC2(0x21cd5700), WTC2(0x228abec0), WTC2(0x234a4180), WTC2(0x240bdf80), WTC2(0x24cf95c0), WTC2(0x259561c0), WTC2(0x265d4200), WTC2(0x27273840), WTC2(0x27f348c0), WTC2(0x28c17700), WTC2(0x2991c500), WTC2(0x2a643080), WTC2(0x2b38b680), WTC2(0x2c0f53c0), WTC2(0x2ce80840), WTC2(0x2dc2d680), WTC2(0x2e9fc100), WTC2(0x2f7ecac0), WTC2(0x305ff280), WTC2(0x314334c0), WTC2(0x32288e00), WTC2(0x330ffb80), WTC2(0x33f97d80), WTC2(0x34e515c0), WTC2(0x35d2c5c0), WTC2(0x36c28d00), WTC2(0x37b467c0), WTC2(0x38a85080), WTC2(0x399e4240), WTC2(0x3a963a00), WTC2(0x3b903600), WTC2(0x3c8c3480), WTC2(0x3d8a3380), WTC2(0x3e8a2dc0), WTC2(0x3f8c1b40), WTC2(0x408ff2ff), WTC2(0x4195ae7f), WTC2(0x429d477f), WTC2(0x43a6b87f), WTC2(0x44b1fdff), WTC2(0x45bf11ff), WTC2(0x46cdee7f), WTC2(0x47de8cff), WTC2(0x48f0e77f), WTC2(0x4a050eff), WTC2(0x4b1b2dff), WTC2(0x4c3372ff), WTC2(0x4d4e0bff), WTC2(0x4e6b257f), WTC2(0x4f8aedff), WTC2(0x50ad92ff), WTC2(0x51d341ff), WTC2(0x002006a9), WTC2(0x000bfb36), WTC2(0xfffe45ac), WTC2(0xfff6d064), WTC2(0xfff585bc), WTC2(0xfffa500d), WTC2(0x000519b4), WTC2(0x0015cd0c), WTC2(0x002c5470), WTC2(0x00489a3b), WTC2(0x006a88c8), WTC2(0x00920a74), WTC2(0x00bf0999), WTC2(0x00f17092), WTC2(0x012929bc), WTC2(0x01661f70), WTC2(0x01a83c0c), WTC2(0x01ef69e8), WTC2(0x023b9364), WTC2(0x028ca2d4), WTC2(0x02e2829c), WTC2(0x033d1d10), WTC2(0x039c5c90), WTC2(0x04002b78), WTC2(0x04687418), WTC2(0x04d520e0), WTC2(0x05461c18), WTC2(0x05bb5020), WTC2(0x0634a758), WTC2(0x06b20c20), WTC2(0x073368c8), WTC2(0x07b8a7b0), WTC2(0x0841b340), WTC2(0x08ce75b0), WTC2(0x095ed980), WTC2(0x09f2c900), WTC2(0x0a8a2e80), WTC2(0x0b24f470), WTC2(0x0bc30510), WTC2(0x0c644ad0), WTC2(0x0d08b010), WTC2(0x0db01f10), WTC2(0x0e5a8250), WTC2(0x0f07c400), WTC2(0x0fb7cea0), WTC2(0x106a8c80), WTC2(0x111fe800), WTC2(0x11d7cb60), WTC2(0x12922120), WTC2(0x134ed3a0), WTC2(0x140dcd00), WTC2(0x14cef7e0), WTC2(0x15923e60), WTC2(0x16578b00), WTC2(0x171ec820), WTC2(0x17e7e020), WTC2(0x18b2bd20), WTC2(0x197f49c0), WTC2(0x1a4d7040), WTC2(0x1b1d1b00), WTC2(0x1bee3460), WTC2(0x1cc0a6a0), WTC2(0x1d945c40), WTC2(0x1e693f80), WTC2(0x1f3f3ac0), WTC2(0x20163880), WTC2(0x20ee22c0), WTC2(0x21c6e440), WTC2(0x22a06740), WTC2(0x237a9600), WTC2(0x24555ac0), WTC2(0x2530a040), WTC2(0x260c5080), WTC2(0x26e85600), WTC2(0x27c49b00), WTC2(0x28a10a00), WTC2(0x297d8d80), WTC2(0x2a5a0f80), WTC2(0x2b367a80), WTC2(0x2c12b8c0), WTC2(0x2ceeb500), WTC2(0x2dca5940), WTC2(0x2ea58fc0), WTC2(0x2f804340), WTC2(0x305a5dc0), WTC2(0x3133ca00), WTC2(0x320c7200), WTC2(0x32e44000), WTC2(0x33bb1ec0), WTC2(0x3490f880), WTC2(0x3565b7c0), WTC2(0x36394640), WTC2(0x370b8f00), WTC2(0x37dc7c00), WTC2(0x38abf7c0), WTC2(0x3979ecc0), WTC2(0x3a464500), WTC2(0x3b10eb00), WTC2(0x3bd9c940), WTC2(0x3ca0c9c0), WTC2(0x3d65d740), WTC2(0x3e28dc00), WTC2(0x3ee9c240), WTC2(0x3fa87480), WTC2(0x4064dcff), WTC2(0x411ee67f), WTC2(0x41d67a7f), WTC2(0x428b847f), WTC2(0x433ded7f), WTC2(0x43eda0ff), WTC2(0x449a887f), WTC2(0x45448f7f), WTC2(0x45eb9eff), WTC2(0x468fa1ff), WTC2(0x473082ff), WTC2(0x47ce2c7f), WTC2(0x4868887f), WTC2(0x48ff80ff), WTC2(0x499300ff), WTC2(0x4a22f2ff), WTC2(0x4aaf407f), WTC2(0x4b37d47f), WTC2(0x4bbc997f), WTC2(0x4c3d78ff), WTC2(0x4cba5e7f), WTC2(0x4d33337f), WTC2(0x4da7e27f), WTC2(0x4e18567f), WTC2(0x4e8478ff), WTC2(0x4eec347f), WTC2(0x4f4f737f), WTC2(0x4fae20ff), WTC2(0x500825ff), WTC2(0x505d6dff), WTC2(0x50ade37f), WTC2(0x50f96f7f), WTC2(0x513ffdff), WTC2(0x518177ff), WTC2(0x51bdc87f), WTC2(0x51f4d9ff), WTC2(0x5226967f), WTC2(0x5252e87f), WTC2(0x5279b9ff), WTC2(0x529af5ff), WTC2(0x52b6867f), WTC2(0x52cc55ff), WTC2(0x52dc4eff), WTC2(0x52e65aff), WTC2(0x52ea657f), WTC2(0x52e857ff), WTC2(0x52e01d7f), WTC2(0x52d19fff), WTC2(0x52bcc9ff), WTC2(0x52a1857f), WTC2(0x527fbd7f), WTC2(0x52575b7f), WTC2(0x52284a7f), WTC2(0x51f274ff), WTC2(0x51b5c47f), WTC2(0x5172247f), WTC2(0x51277dff), WTC2(0x50d5bc7f), WTC2(0x507cc9ff), WTC2(0x501c90ff), WTC2(0x4fb4fb7f), WTC2(0x4f45f3ff), WTC2(0x4ecf64ff), WTC2(0x4e5138ff), WTC2(0x4dcb597f), WTC2(0x4d3db1ff), WTC2(0x4ca82bff), WTC2(0x4c0ab27f), WTC2(0x4b652f7f), WTC2(0x4ab78d7f), WTC2(0x4a01b67f), WTC2(0x4943957f), WTC2(0x487d12ff), WTC2(0x47ae1f7f), WTC2(0x46d68f7f), WTC2(0x45f7187f), WTC2(0x4513597f), WTC2(0x4430467f), WTC2(0x4352d2ff), WTC2(0x427e6bff), WTC2(0x41b390ff), WTC2(0x40f2077f), WTC2(0x4039a87f), WTC2(0x3f8a3100), WTC2(0x3ee33e00), WTC2(0x3e446ac0), WTC2(0x3dad5180), WTC2(0x3d1d7fc0), WTC2(0x3c947b00), WTC2(0x3c11c7c0), WTC2(0x3b94ebc0), WTC2(0x3b1d6dc0), WTC2(0x3aaad480), WTC2(0x3a3ca740), WTC2(0x39d26c40), WTC2(0x396ba8c0), WTC2(0x3907e080), WTC2(0x38a69800), WTC2(0x38473d80), WTC2(0x37e923c0), WTC2(0x378b9b80), WTC2(0x372e0380), WTC2(0x36d03a80), WTC2(0x36727f00), WTC2(0x36150e40), WTC2(0x35b81540), WTC2(0x355b8000), WTC2(0x34ff1dc0), WTC2(0x34a2bfc0), WTC2(0x34463e80), WTC2(0x33e982c0), WTC2(0x338c7880), WTC2(0x332f0bc0), WTC2(0x32d11800), WTC2(0x327265c0), WTC2(0x3212bbc0), WTC2(0x31b1e740), WTC2(0x314fef00), WTC2(0x30ed0540), WTC2(0x30895c80), WTC2(0x30251880), WTC2(0x2fc02880), WTC2(0x2f5a6480), WTC2(0x2ef3a480), WTC2(0x2e8bd640), WTC2(0x2e231100), WTC2(0x2db97680), WTC2(0x2d4f2700), WTC2(0x2ce431c0), WTC2(0x2c789080), WTC2(0x2c0c3bc0), WTC2(0x2b9f2bc0), WTC2(0x2b315940), WTC2(0x2ac2bc00), WTC2(0x2a534cc0), WTC2(0x29e303c0)}; const FIXP_WTB ELDAnalysis256[768] = { WTC(0xfababde8), WTC(0xfa8e1e6a), WTC(0xfa6012a9), WTC(0xfa30c8dd), WTC(0xfa006f4b), WTC(0xf9cf32c4), WTC(0xf99d1cc8), WTC(0xf96a148d), WTC(0xf936184d), WTC(0xf9013d5b), WTC(0xf8cb7b67), WTC(0xf894ace0), WTC(0xf85cd28e), WTC(0xf82413f8), WTC(0xf7ea90af), WTC(0xf7b05ee6), WTC(0xf7759b0b), WTC(0xf73a671f), WTC(0xf6febea3), WTC(0xf6c27a0e), WTC(0xf685ca33), WTC(0xf6493907), WTC(0xf60d437b), WTC(0xf5d2551f), WTC(0xf598d273), WTC(0xf561199e), WTC(0xf52b8c6f), WTC(0xf4f8907d), WTC(0xf4c87fdf), WTC(0xf49ba806), WTC(0xf4724286), WTC(0xf44c6127), WTC(0xf4282435), WTC(0xf401ceae), WTC(0xf3d775a1), WTC(0xf3a91477), WTC(0xf376c33f), WTC(0xf340a328), WTC(0xf306d4d6), WTC(0xf2c9775c), WTC(0xf288a3ed), WTC(0xf2446e2a), WTC(0xf1fcfa45), WTC(0xf1b27b2d), WTC(0xf164f3f4), WTC(0xf114365c), WTC(0xf0c00532), WTC(0xf06817a9), WTC(0xf00c4ea4), WTC(0xefacbc7f), WTC(0xef4a205f), WTC(0xeee5dc33), WTC(0xee808a0d), WTC(0xee19eeb2), WTC(0xedb12f6e), WTC(0xed44e8eb), WTC(0xecd50a13), WTC(0xec62d8dd), WTC(0xebef68b2), WTC(0xeb7b805c), WTC(0xeb069af4), WTC(0xea8eef1c), WTC(0xea131c86), WTC(0xe99234c6), WTC(0xe90cd9c2), WTC(0xe884f65b), WTC(0xe7fcbd6d), WTC(0xe7767300), WTC(0xe6f289d0), WTC(0xe66f958a), WTC(0xe5eae99f), WTC(0xe560c403), WTC(0xe4cfaaa1), WTC(0xe43887dc), WTC(0xe39dedc4), WTC(0xe303f190), WTC(0xe26d7f5d), WTC(0xe1dc34ff), WTC(0xe14f9ced), WTC(0xe0c53cd0), WTC(0xe03ab085), WTC(0xdfadc948), WTC(0xdf1d640c), WTC(0xde896bb6), WTC(0xddf256ad), WTC(0xdd591e3d), WTC(0xdcbf0aec), WTC(0xdc25ab0a), WTC(0xdb8e334c), WTC(0xdaf97794), WTC(0xda67bed9), WTC(0xd9d8c524), WTC(0xd94bfa62), WTC(0xd8c089b5), WTC(0xd835c151), WTC(0xd7ab1704), WTC(0xd7200906), WTC(0xd69420dc), WTC(0xd6073c0d), WTC(0xd5799615), WTC(0xd4ec7c87), WTC(0xd46241c9), WTC(0xd3dc5bde), WTC(0xd35b4a79), WTC(0xd2de1032), WTC(0xd26246f5), WTC(0xd1e68ed2), WTC(0xd16aa0a4), WTC(0xd0eea5d2), WTC(0xd073302b), WTC(0xcff93749), WTC(0xcf820f45), WTC(0xcf0ebb30), WTC(0xce9fd702), WTC(0xce34596c), WTC(0xcdc9a803), WTC(0xcd5ec5d6), WTC(0xccf468ec), WTC(0xcc8bb41e), WTC(0xcc2619cc), WTC(0xcbc3e090), WTC(0xcb6422f5), WTC(0xcb064d2f), WTC(0xcaaa2a6d), WTC(0xca4fbdc9), WTC(0xc9f73c43), WTC(0xc9a0dc9b), WTC(0xc94cdd02), WTC(0xc8f578a4), WTC(0xc8a24d15), WTC(0xc84dc71f), WTC(0xc7f83516), WTC(0xc7a1e4b9), WTC(0xc74b22b1), WTC(0xc6f41284), WTC(0xc69cabc1), WTC(0xc644986d), WTC(0xc5eb4167), WTC(0xc5910312), WTC(0xc5372c7f), WTC(0xc4deba2e), WTC(0xc4883eca), WTC(0xc43310f0), WTC(0xc3dd5c5a), WTC(0xc3868802), WTC(0xc32f431d), WTC(0xc2d86c9e), WTC(0xc28300a6), WTC(0xc22fae33), WTC(0xc1ded3f7), WTC(0xc1908d7d), WTC(0xc144b0ed), WTC(0xc0fa7cee), WTC(0xc0b0a3b5), WTC(0xc066b8d3), WTC(0xc01d3b32), WTC(0xbfd5161c), WTC(0xbf8f92af), WTC(0xbf4d5cea), WTC(0xbf0e7d5e), WTC(0xbed2ce3a), WTC(0xbe9a0062), WTC(0xbe63cec2), WTC(0xbe2ffd2f), WTC(0xbdfe4565), WTC(0xbdce5568), WTC(0xbda003df), WTC(0xbd735018), WTC(0xbd485b2c), WTC(0xbd1f69bd), WTC(0xbcf8db7c), WTC(0xbcd52b0a), WTC(0xbcb4ae4a), WTC(0xbc979382), WTC(0xbc7dcbab), WTC(0xbc6709dc), WTC(0xbc52c1b1), WTC(0xbc402f2b), WTC(0xbc2ec37b), WTC(0xbc1e2cb3), WTC(0xbc0e5d5f), WTC(0xbbff8f23), WTC(0xbbf238d2), WTC(0xbbe707d4), WTC(0xbbde3c63), WTC(0xbbd7a658), WTC(0xbbd2c7f0), WTC(0xbbcee18b), WTC(0xbbcbdebb), WTC(0xbbca5ab1), WTC(0xbbcb5622), WTC(0xbbd032e4), WTC(0xbbd91d4d), WTC(0xbbe53757), WTC(0xbbf32f54), WTC(0xbc016781), WTC(0xbc0f433a), WTC(0xbc1d2aa4), WTC(0xbc2b4912), WTC(0xbc3985df), WTC(0xbc47d6b9), WTC(0xbc564099), WTC(0xbc64c78a), WTC(0xbc736d96), WTC(0xbc823210), WTC(0xbc911484), WTC(0xbca015b8), WTC(0xbcaf37eb), WTC(0xbcbe7bc3), WTC(0xbccdde4d), WTC(0xbcdd6037), WTC(0xbced049a), WTC(0xbcfccc81), WTC(0xbd0cb482), WTC(0xbd1cbcaa), WTC(0xbd2ce7ea), WTC(0xbd3d363b), WTC(0xbd4da445), WTC(0xbd5e312d), WTC(0xbd6edfd1), WTC(0xbd7fae14), WTC(0xbd90991b), WTC(0xbda19fcf), WTC(0xbdb2c464), WTC(0xbdc4053b), WTC(0xbdd55f4b), WTC(0xbde6d0a0), WTC(0xbdf85c51), WTC(0xbe09ffa3), WTC(0xbe1bb724), WTC(0xbe2d8160), WTC(0xbe3f5f98), WTC(0xbe515144), WTC(0xbe6351a9), WTC(0xbe755ebd), WTC(0xbe877b8e), WTC(0xbe99a63d), WTC(0xbeabda45), WTC(0xbebe16b0), WTC(0xbed05d1c), WTC(0xbee2ada9), WTC(0xbef502e2), WTC(0xbf075c40), WTC(0xbf19bc0b), WTC(0xbf2c217f), WTC(0xbf3e887a), WTC(0xbf50f09d), WTC(0xbf635c77), WTC(0xbf75cac0), WTC(0xbf883905), WTC(0xbf9aa62b), WTC(0xbfad14f1), WTC(0xbfbf85c7), WTC(0xbfd1f592), WTC(0xbfe461fc), WTC(0xbff6c86a), WTC(0x80126c8d), WTC(0x80372448), WTC(0x805bd2fd), WTC(0x80807315), WTC(0x80a4fffa), WTC(0x80c9748d), WTC(0x80edd08b), WTC(0x81121a23), WTC(0x81364fde), WTC(0x815a6b16), WTC(0x817e6b36), WTC(0x81a25433), WTC(0x81c625c8), WTC(0x81e9d801), WTC(0x820d6a5c), WTC(0x8230e060), WTC(0x825438c0), WTC(0x82776ac7), WTC(0x829a7555), WTC(0x82bd5ca3), WTC(0x82e01e80), WTC(0x8302b200), WTC(0x83251590), WTC(0x83474d79), WTC(0x8369566f), WTC(0x838b2957), WTC(0x83acc2d9), WTC(0x83ce27c1), WTC(0x83ef54b9), WTC(0x841042d1), WTC(0x8430ef15), WTC(0x84515e84), WTC(0x84718e32), WTC(0x84917804), WTC(0x84b11a25), WTC(0x84d0788d), WTC(0x84ef9322), WTC(0x850e61ec), WTC(0x852ce400), WTC(0x854b1e0a), WTC(0x85690f2c), WTC(0x8586b207), WTC(0x85a4057b), WTC(0x85c1107d), WTC(0x85ddd335), WTC(0x85fa485e), WTC(0x86167172), WTC(0x8632549d), WTC(0x864df388), WTC(0x8669497e), WTC(0x86845757), WTC(0x869f2218), WTC(0x86b9ab5a), WTC(0x86d3f1bf), WTC(0x86edf68f), WTC(0x8707baf1), WTC(0x872147e0), WTC(0x873aa6fc), WTC(0x8753c571), WTC(0x876c76e6), WTC(0x87850ab7), WTC(0x879e373b), WTC(0x87b6ea37), WTC(0x87cc4188), WTC(0x880d4300), WTC(0x8855e9ff), WTC(0x88acfca0), WTC(0x890d0f94), WTC(0x8971e7d5), WTC(0x89d8a0c1), WTC(0x8a3fc425), WTC(0x8aa74105), WTC(0x8b0f5b93), WTC(0x8b78a107), WTC(0x8be38bb3), WTC(0x8c508092), WTC(0x8cbfe384), WTC(0x8d3214f1), WTC(0x8da75d21), WTC(0x8e1fe96c), WTC(0x8e9be76a), WTC(0x8f1b806c), WTC(0x8f9ed314), WTC(0x9025f26a), WTC(0x90b0ecea), WTC(0x913fd0eb), WTC(0x91d2a684), WTC(0x92696dea), WTC(0x93042868), WTC(0x93a2d456), WTC(0x94456d20), WTC(0x94ebe9e5), WTC(0x95964178), WTC(0x96446a05), WTC(0x96f65958), WTC(0x97ac059a), WTC(0x98656089), WTC(0x99225a80), WTC(0x99e2e2e8), WTC(0x9aa6e666), WTC(0x9b6e54b8), WTC(0x9c391d99), WTC(0x9d07338a), WTC(0x9dd8888d), WTC(0x9ead0b5c), WTC(0x9f84a871), WTC(0xa05f4fb3), WTC(0xa13cf913), WTC(0xa21d9891), WTC(0xa3011e27), WTC(0xa3e77eb4), WTC(0xa4d0b190), WTC(0xa5bcb0d7), WTC(0xa6ab750c), WTC(0xa79cf884), WTC(0xa89135cb), WTC(0xa9882a44), WTC(0xaa81d578), WTC(0xab7e39a6), WTC(0xac7d5a36), WTC(0xad7f3ba5), WTC(0xae83dfed), WTC(0xaf8b4e16), WTC(0xb095911c), WTC(0xb1a2afd1), WTC(0xb2b2ac9f), WTC(0xb3c58807), WTC(0xb4db4d5e), WTC(0x4a268ead), WTC(0x490b5ba7), WTC(0x47ed8d30), WTC(0x46cd10c5), WTC(0x45a9dcc1), WTC(0x4483f267), WTC(0x435b5aeb), WTC(0x42301d12), WTC(0x41023a15), WTC(0x3fd19bf1), WTC(0x3e9e31e1), WTC(0x3d682986), WTC(0x3c2fc001), WTC(0x3af52d8f), WTC(0x39b88b7d), WTC(0x38798642), WTC(0x3737e6d3), WTC(0x35f3e98a), WTC(0x34add45c), WTC(0x33660083), WTC(0x321ccf3a), WTC(0x30d2963e), WTC(0x2f87a28f), WTC(0x2e3c22cd), WTC(0x2cf010e5), WTC(0x2ba2ffe5), WTC(0x2a54ba93), WTC(0x290596f5), WTC(0x27b62806), WTC(0x266762b8), WTC(0x251a11b1), WTC(0x23ce94f9), WTC(0x22852ddb), WTC(0x213df340), WTC(0x1ff90185), WTC(0x1eb67d94), WTC(0x1d767485), WTC(0x1c38d477), WTC(0x1afda747), WTC(0x19c5248b), WTC(0x188f8259), WTC(0x175d0d40), WTC(0x162e5320), WTC(0x150436cd), WTC(0x13df8d3f), WTC(0x12c102f1), WTC(0x11a8dd65), WTC(0x1096d490), WTC(0x0f8a1755), WTC(0x0e811dcd), WTC(0x0d7acb9a), WTC(0x0c767d00), WTC(0x0b7334d9), WTC(0x0a6fef31), WTC(0x096c5a87), WTC(0x08691adb), WTC(0x0765e395), WTC(0x06610309), WTC(0x0558a0d2), WTC(0x044a946c), WTC(0x033acb52), WTC(0x0234706f), WTC(0x014939dc), WTC(0x00928577), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0xffe73593), WTC(0xffb63fcf), WTC(0xff853c1f), WTC(0xff5454d7), WTC(0xff23b44b), WTC(0xfef38417), WTC(0xfec3dc9a), WTC(0xfe94c511), WTC(0xfe664753), WTC(0xfe387086), WTC(0xfe0b4e63), WTC(0xfddef15c), WTC(0xfdb3a3f6), WTC(0xfd89e611), WTC(0xfd61c750), WTC(0xfd3ae585), WTC(0xfd14ec09), WTC(0xfcef9b06), WTC(0xfccaf509), WTC(0xfca74180), WTC(0xfc8518a3), WTC(0xfc655c7a), WTC(0xfc488545), WTC(0xfc2e9998), WTC(0xfc1726bb), WTC(0xfc01463f), WTC(0xfbec2c64), WTC(0xfbd735ce), WTC(0xfbc29e8e), WTC(0xfbaf8042), WTC(0xfb9eeba0), WTC(0xfb91dc05), WTC(0xfb88f420), WTC(0xfb8479eb), WTC(0xfb84398b), WTC(0xfb87884b), WTC(0xfb8da8bf), WTC(0xfb95d020), WTC(0xfb9f3e49), WTC(0xfba9448a), WTC(0xfbb3cf10), WTC(0xfbbf67e7), WTC(0xfbccf65d), WTC(0xfbddba58), WTC(0xfbf31f46), WTC(0xfc0eb236), WTC(0xfc3164f0), WTC(0xfc5b8269), WTC(0xfc8c5bcd), WTC(0xfcc248ee), WTC(0xfcfb056c), WTC(0xfd33cc26), WTC(0xfd6b84ee), WTC(0xfda2d9e7), WTC(0xfddc03fb), WTC(0xfe1aaf57), WTC(0xfe61a0af), WTC(0xfeb28df7), WTC(0xff0cd343), WTC(0xff6d8388), WTC(0xffd24331), WTC(0x00396fe3), WTC(0x00a2fb3e), WTC(0x01107050), WTC(0x01831900), WTC(0x01fc2377), WTC(0x027bd1fc), WTC(0x03019a2d), WTC(0x038d0a88), WTC(0x041dd88f), WTC(0x04b43495), WTC(0x0550c1ef), WTC(0x05f38bd6), WTC(0x069c0523), WTC(0x074a114e), WTC(0x07fe0ceb), WTC(0x08b88e33), WTC(0x097a5965), WTC(0x0a438318), WTC(0x0b137046), WTC(0x0be9b5ab), WTC(0x0cc61fa9), WTC(0x0da897b2), WTC(0x0e9123b3), WTC(0x0f7ff200), WTC(0x10755696), WTC(0x11717f94), WTC(0x127474a0), WTC(0x137e489d), WTC(0x148f1b02), WTC(0x15a6f15e), WTC(0x16c5b7c9), WTC(0x17eb72b1), WTC(0x19183e51), WTC(0x1a4c2444), WTC(0x1b871b1c), WTC(0x1cc92e92), WTC(0x1e127ffc), WTC(0x1f6319b9), WTC(0x20baef78), WTC(0x221a0861), WTC(0x23807f94), WTC(0x24ee5a89), WTC(0x2663898d), WTC(0x27e0101e), WTC(0x2964058d), WTC(0x2aef6bcf), WTC(0x2c8230fc), WTC(0x2e1c545b), WTC(0x2fbde72b), WTC(0x3166e76f), WTC(0x33173f5d), WTC(0x34cee8c3), WTC(0x368debe1), WTC(0x38543d4f), WTC(0x3a21bd94), WTC(0x3bf6576f), WTC(0x3dd1ff07), WTC(0x3fb4948e), WTC(0x419de414), WTC(0x438dc202), WTC(0x45840e7d), WTC(0x4780a435), WTC(0x4983609f), WTC(0x4b8cc548), WTC(0x4d9df796), WTC(0x4fb81f46), WTC(0x51dc8690), WTC(0x000d970d), WTC(0xfff7ea67), WTC(0xfff7fc3d), WTC(0x000d3de2), WTC(0x003720ad), WTC(0x007515f1), WTC(0x00c68f04), WTC(0x012afd3b), WTC(0x01a1d1ec), WTC(0x022a7e69), WTC(0x02c47408), WTC(0x036f2420), WTC(0x042a0001), WTC(0x04f47905), WTC(0x05ce007e), WTC(0x06b607be), WTC(0x07ac0028), WTC(0x08af5b01), WTC(0x09bf89a7), WTC(0x0adbfd6d), WTC(0x0c042798), WTC(0x0d377997), WTC(0x0e7564b5), WTC(0x0fbd5a3a), WTC(0x110ecb85), WTC(0x126929fb), WTC(0x13cbe6e6), WTC(0x15367376), WTC(0x16a8413f), WTC(0x1820c15f), WTC(0x199f6568), WTC(0x1b239e6b), WTC(0x1cacdde2), WTC(0x1e3a951a), WTC(0x1fcc356f), WTC(0x2161301f), WTC(0x22f8f6b7), WTC(0x2492fa4a), WTC(0x262eac3f), WTC(0x27cb7e20), WTC(0x2968e0c4), WTC(0x2b064625), WTC(0x2ca31f1a), WTC(0x2e3edd2a), WTC(0x2fd8f19f), WTC(0x3170ce00), WTC(0x3305e32c), WTC(0x3497a2df), WTC(0x36257e78), WTC(0x37aee70b), WTC(0x39334e05), WTC(0x3ab22498), WTC(0x3c2adc2c), WTC(0x3d9ce645), WTC(0x3f07b3ef), WTC(0x406ab6ca), WTC(0x41c56001), WTC(0x4317214a), WTC(0x445f6b34), WTC(0x459daf5d), WTC(0x46d15f56), WTC(0x47f9ed71), WTC(0x4916d11f), WTC(0x4a275770), WTC(0x4b2b2fff), WTC(0x4c219eae), WTC(0x4d0a20cb), WTC(0x4de4288e), WTC(0x4eaf263d), WTC(0x4f6a8bb8), WTC(0x5015ca33), WTC(0x50b052dd), WTC(0x51399757), WTC(0x51b108c6), WTC(0x5216190a), WTC(0x5268387c), WTC(0x52a6d933), WTC(0x52d16c19), WTC(0x52e7628b), WTC(0x52e82ea3), WTC(0x52d3407d), WTC(0x52a80a28), WTC(0x5265fd43), WTC(0x520c8a1d), WTC(0x519b22c8), WTC(0x511138e0), WTC(0x506e3c82), WTC(0x4fb1a037), WTC(0x4edad4e3), WTC(0x4de94c2d), WTC(0x4cdc76d8), WTC(0x4bb3c683), WTC(0x4a6eacd2), WTC(0x490c9abe), WTC(0x478d04f1), WTC(0x45f00420), WTC(0x4445673f), WTC(0x42ac0d2e), WTC(0x41338364), WTC(0x3fdb5b58), WTC(0x3ea1c30f), WTC(0x3d842780), WTC(0x3c7fa763), WTC(0x3b911b96), WTC(0x3ab560bf), WTC(0x39e95908), WTC(0x3929debb), WTC(0x3873bd4d), WTC(0x37c31db2), WTC(0x3713a59c), WTC(0x3663deb2), WTC(0x35b52f23), WTC(0x3507c61e), WTC(0x345a7f42), WTC(0x33ac7e0c), WTC(0x32fd366f), WTC(0x324baa28), WTC(0x319674e9), WTC(0x30dd7e1a), WTC(0x3021f3e8), WTC(0x2f63f903), WTC(0x2ea2a1aa), WTC(0x2dddd97b), WTC(0x2d166985), WTC(0x2c4ca42f), WTC(0x2b805cca), WTC(0x2ab162aa), WTC(0x29df7b17), }; const FIXP_WTB ELDAnalysis240[720] = { WTC(0xfab9477b), WTC(0xfa899344), WTC(0xfa5845dd), WTC(0xfa259762), WTC(0xf9f1c005), WTC(0xf9bcefe6), WTC(0xf9871e8b), WTC(0xf9503397), WTC(0xf9183f47), WTC(0xf8df4eac), WTC(0xf8a53ba7), WTC(0xf869f0be), WTC(0xf82d9759), WTC(0xf7f0593e), WTC(0xf7b2520a), WTC(0xf773a37c), WTC(0xf73475ce), WTC(0xf6f4bedd), WTC(0xf6b455a8), WTC(0xf6739525), WTC(0xf6332510), WTC(0xf5f3938b), WTC(0xf5b56073), WTC(0xf57900bd), WTC(0xf53ee82d), WTC(0xf5079149), WTC(0xf4d36ffc), WTC(0xf4a2e526), WTC(0xf4763d91), WTC(0xf44d9872), WTC(0xf426eaed), WTC(0xf3fdc161), WTC(0xf3d001ff), WTC(0xf39dafcc), WTC(0xf366eb43), WTC(0xf32bdcdc), WTC(0xf2ecab80), WTC(0xf2a97b34), WTC(0xf26265ae), WTC(0xf2178a6f), WTC(0xf1c92458), WTC(0xf17752b9), WTC(0xf121e6ac), WTC(0xf0c89a63), WTC(0xf06b15ef), WTC(0xf0092e86), WTC(0xefa2fd42), WTC(0xef397ebc), WTC(0xeece51c6), WTC(0xee61e8b6), WTC(0xedf3d92e), WTC(0xed82c330), WTC(0xed0d58bb), WTC(0xec94891b), WTC(0xec19d435), WTC(0xeb9e4e4e), WTC(0xeb221000), WTC(0xeaa32422), WTC(0xea1fb440), WTC(0xe99695d2), WTC(0xe90859ab), WTC(0xe8775114), WTC(0xe7e62b37), WTC(0xe7578147), WTC(0xe6cb3ac1), WTC(0xe63f5696), WTC(0xe5afe916), WTC(0xe519090f), WTC(0xe47aab0d), WTC(0xe3d6c2d0), WTC(0xe331dae7), WTC(0xe29031e1), WTC(0xe1f40926), WTC(0xe15d87d2), WTC(0xe0c9d727), WTC(0xe0360ad5), WTC(0xdf9f81af), WTC(0xdf04f9f9), WTC(0xde66697f), WTC(0xddc48ca1), WTC(0xdd20a42a), WTC(0xdc7c6853), WTC(0xdbd9a476), WTC(0xdb398a8c), WTC(0xda9cd7c2), WTC(0xda0365cf), WTC(0xd96cad85), WTC(0xd8d7b7a3), WTC(0xd8439e8c), WTC(0xd7afb73d), WTC(0xd71b6347), WTC(0xd686149a), WTC(0xd5efab2c), WTC(0xd558877e), WTC(0xd4c29dbc), WTC(0xd430a0aa), WTC(0xd3a3d490), WTC(0xd31c588f), WTC(0xd297e075), WTC(0xd213ef33), WTC(0xd18fd566), WTC(0xd10b8d3f), WTC(0xd087b250), WTC(0xd0054ef2), WTC(0xcf85f94a), WTC(0xcf0af5f7), WTC(0xce94faf5), WTC(0xce229409), WTC(0xcdb0b5f8), WTC(0xcd3ec554), WTC(0xcccdbf58), WTC(0xcc5f39d5), WTC(0xcbf49ef5), WTC(0xcb8d5f73), WTC(0xcb28801c), WTC(0xcac5a265), WTC(0xca64ad2e), WTC(0xca05d7fd), WTC(0xc9a96602), WTC(0xc94f9f79), WTC(0xc8f2b954), WTC(0xc899e795), WTC(0xc83f94aa), WTC(0xc7e41f63), WTC(0xc787e69f), WTC(0xc72b3fd0), WTC(0xc6ce3f0f), WTC(0xc670c175), WTC(0xc61224cf), WTC(0xc5b21fec), WTC(0xc55202a4), WTC(0xc4f3353a), WTC(0xc4968597), WTC(0xc43b93f2), WTC(0xc3e03d26), WTC(0xc383a011), WTC(0xc3268aed), WTC(0xc2ca1039), WTC(0xc26f5bcc), WTC(0xc21726c9), WTC(0xc1c1d5b2), WTC(0xc16f66ba), WTC(0xc11f76d9), WTC(0xc0d0a9f6), WTC(0xc081cddb), WTC(0xc0333180), WTC(0xbfe5bb54), WTC(0xbf9aee90), WTC(0xbf53d587), WTC(0xbf108855), WTC(0xbed0de05), WTC(0xbe9477d7), WTC(0xbe5b030f), WTC(0xbe243642), WTC(0xbdefb72f), WTC(0xbdbd29df), WTC(0xbd8c71ab), WTC(0xbd5d99cb), WTC(0xbd30e375), WTC(0xbd06afcc), WTC(0xbcdf8c7f), WTC(0xbcbbf704), WTC(0xbc9c307e), WTC(0xbc803b86), WTC(0xbc67c0c7), WTC(0xbc521d3d), WTC(0xbc3e6561), WTC(0xbc2bf2cb), WTC(0xbc1a6872), WTC(0xbc09ce15), WTC(0xbbfa764f), WTC(0xbbed1356), WTC(0xbbe257fa), WTC(0xbbda4099), WTC(0xbbd46a31), WTC(0xbbcffa76), WTC(0xbbcc766d), WTC(0xbbca782f), WTC(0xbbcb16c7), WTC(0xbbcff77c), WTC(0xbbd978e6), WTC(0xbbe68e5f), WTC(0xbbf593ed), WTC(0xbc04a834), WTC(0xbc136941), WTC(0xbc2252c3), WTC(0xbc31723d), WTC(0xbc40ab92), WTC(0xbc4ffe2d), WTC(0xbc5f7072), WTC(0xbc6f0520), WTC(0xbc7ebd23), WTC(0xbc8e9746), WTC(0xbc9e942f), WTC(0xbcaeb633), WTC(0xbcbefe8b), WTC(0xbccf69bb), WTC(0xbcdff92e), WTC(0xbcf0b04f), WTC(0xbd018ebd), WTC(0xbd129192), WTC(0xbd23b9b8), WTC(0xbd350afb), WTC(0xbd46820e), WTC(0xbd581bfc), WTC(0xbd69db11), WTC(0xbd7bbf57), WTC(0xbd8dc584), WTC(0xbd9feaad), WTC(0xbdb231a4), WTC(0xbdc498ea), WTC(0xbdd71cd1), WTC(0xbde9bb57), WTC(0xbdfc77d9), WTC(0xbe0f4e93), WTC(0xbe223ae5), WTC(0xbe353cf5), WTC(0xbe485689), WTC(0xbe5b8329), WTC(0xbe6ebe88), WTC(0xbe820afd), WTC(0xbe956811), WTC(0xbea8d109), WTC(0xbebc4352), WTC(0xbecfc0fb), WTC(0xbee34a07), WTC(0xbef6d884), WTC(0xbf0a6bb1), WTC(0xbf1e0685), WTC(0xbf31a685), WTC(0xbf45483c), WTC(0xbf58eb6b), WTC(0xbf6c9376), WTC(0xbf803c90), WTC(0xbf93e4b9), WTC(0xbfa78d05), WTC(0xbfbb3830), WTC(0xbfcee339), WTC(0xbfe28aa9), WTC(0xbff62b89), WTC(0x8013a5f4), WTC(0x803acfd6), WTC(0x8061eec7), WTC(0x8088fc73), WTC(0x80aff270), WTC(0x80d6cbe5), WTC(0x80fd8c2a), WTC(0x812437a8), WTC(0x814ac94f), WTC(0x81713adc), WTC(0x81979098), WTC(0x81bdccb7), WTC(0x81e3e738), WTC(0x8209dd04), WTC(0x822fb23a), WTC(0x825565bb), WTC(0x827aed94), WTC(0x82a04909), WTC(0x82c57c85), WTC(0x82ea831c), WTC(0x830f539d), WTC(0x8333eeba), WTC(0x8358585a), WTC(0x837c882e), WTC(0x83a07742), WTC(0x83c428a5), WTC(0x83e79c4c), WTC(0x840aca65), WTC(0x842dad81), WTC(0x84504ac0), WTC(0x84729fb1), WTC(0x8494a4f1), WTC(0x84b65932), WTC(0x84d7c0f8), WTC(0x84f8d936), WTC(0x85199a59), WTC(0x853a05a1), WTC(0x855a2023), WTC(0x8579e46e), WTC(0x85994d55), WTC(0x85b86190), WTC(0x85d723e6), WTC(0x85f58fa9), WTC(0x8613a3ce), WTC(0x863167b5), WTC(0x864eddfe), WTC(0x866c0138), WTC(0x8688d2e4), WTC(0x86a55901), WTC(0x86c19497), WTC(0x86dd8390), WTC(0x86f9288f), WTC(0x871487e0), WTC(0x872fadd0), WTC(0x874a9a1e), WTC(0x876519d0), WTC(0x877f471e), WTC(0x8799fb36), WTC(0x87b48b97), WTC(0x87cba021), WTC(0x880f67ae), WTC(0x885e0f91), WTC(0x88bc84cd), WTC(0x89244640), WTC(0x8990a45d), WTC(0x89fe6766), WTC(0x8a6c9065), WTC(0x8adb31e6), WTC(0x8b4ad5b3), WTC(0x8bbc2068), WTC(0x8c2f93ff), WTC(0x8ca5a922), WTC(0x8d1ed72d), WTC(0x8d9b7ddb), WTC(0x8e1bd6cc), WTC(0x8ea01924), WTC(0x8f287716), WTC(0x8fb5143e), WTC(0x9046074e), WTC(0x90db612b), WTC(0x91753263), WTC(0x92138094), WTC(0x92b64cf3), WTC(0x935d96c9), WTC(0x94095a56), WTC(0x94b98fd4), WTC(0x956e2a87), WTC(0x96271ff6), WTC(0x96e46309), WTC(0x97a5e80d), WTC(0x986b9e55), WTC(0x993572af), WTC(0x9a0350ce), WTC(0x9ad52154), WTC(0x9baad10f), WTC(0x9c844cdd), WTC(0x9d618437), WTC(0x9e4265b2), WTC(0x9f26d9ad), WTC(0xa00ec9b0), WTC(0xa0fa2916), WTC(0xa1e8ec20), WTC(0xa2daffa4), WTC(0xa3d05468), WTC(0xa4c8e007), WTC(0xa5c49ae4), WTC(0xa6c37c24), WTC(0xa7c57d03), WTC(0xa8ca9750), WTC(0xa9d2c7f2), WTC(0xaade0f6f), WTC(0xabec7177), WTC(0xacfdf2b1), WTC(0xae129740), WTC(0xaf2a6321), WTC(0xb04563a6), WTC(0xb163a2e6), WTC(0xb28524c4), WTC(0xb3a9eaf7), WTC(0xb4d1ff1b), WTC(0x4a1d2880), WTC(0x48eee56e), WTC(0x47bda882), WTC(0x46895c79), WTC(0x4551f8a1), WTC(0x4417817b), WTC(0x42da023d), WTC(0x419980ca), WTC(0x4055f463), WTC(0x3f0f3b51), WTC(0x3dc56e18), WTC(0x3c78d943), WTC(0x3b29bf3d), WTC(0x39d84ea0), WTC(0x3884337d), WTC(0x372d2371), WTC(0x35d364ea), WTC(0x34774cef), WTC(0x33194d3a), WTC(0x31b9d586), WTC(0x30594fcf), WTC(0x2ef80b63), WTC(0x2d9630d5), WTC(0x2c337c00), WTC(0x2acf6a9e), WTC(0x296a3205), WTC(0x28046825), WTC(0x269f1752), WTC(0x253b5314), WTC(0x23d9993f), WTC(0x227a3c77), WTC(0x211d59a0), WTC(0x1fc314fd), WTC(0x1e6b9834), WTC(0x1d16eb58), WTC(0x1bc4f82e), WTC(0x1a75e481), WTC(0x1929f389), WTC(0x17e16ee3), WTC(0x169cd758), WTC(0x155d1ae5), WTC(0x14235182), WTC(0x12f051de), WTC(0x11c4993b), WTC(0x109fdf4c), WTC(0x0f81351c), WTC(0x0e66c5e6), WTC(0x0d4f4b16), WTC(0x0c39f013), WTC(0x0b25765d), WTC(0x0a10c51e), WTC(0x08fbee35), WTC(0x07e7986f), WTC(0x06d25fe7), WTC(0x05ba1b52), WTC(0x049c33b7), WTC(0x0379ceb9), WTC(0x025ee7c7), WTC(0x015edc1c), WTC(0x00978deb), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0xffe59474), WTC(0xffb158f8), WTC(0xff7d1275), WTC(0xff48f44c), WTC(0xff1531e3), WTC(0xfee1faa9), WTC(0xfeaf626e), WTC(0xfe7d7227), WTC(0xfe4c383f), WTC(0xfe1bc4ff), WTC(0xfdec297f), WTC(0xfdbd9f6c), WTC(0xfd90bbf0), WTC(0xfd65ba73), WTC(0xfd3c2d32), WTC(0xfd13ab35), WTC(0xfcebe811), WTC(0xfcc4eeae), WTC(0xfc9f1d64), WTC(0xfc7b48b0), WTC(0xfc5a7282), WTC(0xfc3cef9b), WTC(0xfc229f4c), WTC(0xfc0a9e29), WTC(0xfbf3dccb), WTC(0xfbdd80c3), WTC(0xfbc7556c), WTC(0xfbb289cf), WTC(0xfba06f99), WTC(0xfb923aca), WTC(0xfb88bb57), WTC(0xfb84457f), WTC(0xfb848c2e), WTC(0xfb88bd28), WTC(0xfb8feda1), WTC(0xfb9928eb), WTC(0xfba38b9e), WTC(0xfbae711c), WTC(0xfbba3538), WTC(0xfbc7af9d), WTC(0xfbd8485e), WTC(0xfbeda238), WTC(0xfc099de4), WTC(0xfc2d981f), WTC(0xfc59fd0d), WTC(0xfc8e1583), WTC(0xfcc7e0d7), WTC(0xfd048d03), WTC(0xfd40dfaf), WTC(0xfd7c1d35), WTC(0xfdb767cc), WTC(0xfdf64a9f), WTC(0xfe3d0242), WTC(0xfe8e3316), WTC(0xfeead2b9), WTC(0xff4fff8c), WTC(0xffba7b11), WTC(0x00281cae), WTC(0x009847d1), WTC(0x010cb653), WTC(0x01870639), WTC(0x02089db5), WTC(0x0291b571), WTC(0x0321a4c1), WTC(0x03b7eda0), WTC(0x04544c7f), WTC(0x04f74134), WTC(0x05a16810), WTC(0x06525829), WTC(0x070994d4), WTC(0x07c767ba), WTC(0x088c69b8), WTC(0x09598634), WTC(0x0a2f14ca), WTC(0x0b0c677b), WTC(0x0bf0f576), WTC(0x0cdc7f73), WTC(0x0dceed54), WTC(0x0ec84a00), WTC(0x0fc8db86), WTC(0x10d10278), WTC(0x11e0e05e), WTC(0x12f880cc), WTC(0x1418049b), WTC(0x153f86b3), WTC(0x166ef5a3), WTC(0x17a64878), WTC(0x18e59dde), WTC(0x1a2d088b), WTC(0x1b7c7e41), WTC(0x1cd40ab7), WTC(0x1e33d542), WTC(0x1f9be7a5), WTC(0x210c344f), WTC(0x2284cb85), WTC(0x2405ca48), WTC(0x258f2b7f), WTC(0x2720e063), WTC(0x28bafd49), WTC(0x2a5d950c), WTC(0x2c0896e4), WTC(0x2dbbf7d4), WTC(0x2f77ca28), WTC(0x313c1273), WTC(0x3308b7e4), WTC(0x34ddb0ec), WTC(0x36bb06b7), WTC(0x38a0a935), WTC(0x3a8e7270), WTC(0x3c844ca9), WTC(0x3e82267e), WTC(0x4087ccfa), WTC(0x42950352), WTC(0x44a99ce7), WTC(0x46c57093), WTC(0x48e84dbe), WTC(0x4b127506), WTC(0x4d452d29), WTC(0x4f81e066), WTC(0x51ca11c4), WTC(0x000c82e8), WTC(0xfff6f40c), WTC(0xfffa1260), WTC(0x001530bf), WTC(0x0047a202), WTC(0x0090b903), WTC(0x00efc89f), WTC(0x016423af), WTC(0x01ed1d0e), WTC(0x028a0796), WTC(0x033a3620), WTC(0x03fcfb89), WTC(0x04d1aaaa), WTC(0x05b7965c), WTC(0x06ae1179), WTC(0x07b46ee8), WTC(0x08ca0173), WTC(0x09ee1c00), WTC(0x0b201162), WTC(0x0c5f346e), WTC(0x0daad808), WTC(0x0f024f17), WTC(0x1064ec4b), WTC(0x11d202c4), WTC(0x1348e514), WTC(0x14c8e62f), WTC(0x1651590a), WTC(0x17e19051), WTC(0x1978df27), WTC(0x1b169812), WTC(0x1cba0e15), WTC(0x1e629407), WTC(0x200f7cd4), WTC(0x21c01b29), WTC(0x2373c228), WTC(0x2529c453), WTC(0x26e174b9), WTC(0x289a262f), WTC(0x2a532bba), WTC(0x2c0bd7b2), WTC(0x2dc37d92), WTC(0x2f796fce), WTC(0x312d017a), WTC(0x32dd8513), WTC(0x348a4dde), WTC(0x3632aeb3), WTC(0x37d5fa29), WTC(0x39738334), WTC(0x3b0a9c99), WTC(0x3c9a9926), WTC(0x3e22cc21), WTC(0x3fa287dc), WTC(0x41191f89), WTC(0x4285e5fc), WTC(0x43e82e02), WTC(0x453f4a40), WTC(0x468a8dd9), WTC(0x47c94c23), WTC(0x48fadc7c), WTC(0x4a1e75f9), WTC(0x4b339ecf), WTC(0x4c3981b1), WTC(0x4d2f7cd3), WTC(0x4e14e381), WTC(0x4ee90804), WTC(0x4fab3d6a), WTC(0x505ad6bd), WTC(0x50f726a3), WTC(0x517f7fea), WTC(0x51f335fd), WTC(0x52519b0f), WTC(0x529a01f2), WTC(0x52cbbe31), WTC(0x52e621d9), WTC(0x52e880aa), WTC(0x52d22c7a), WTC(0x52a278a5), WTC(0x5258b880), WTC(0x51f43e1d), WTC(0x51745c38), WTC(0x50d8669e), WTC(0x501faf0e), WTC(0x4f49897e), WTC(0x4e554804), WTC(0x4d423d9e), WTC(0x4c0fbd8b), WTC(0x4abd1a4d), WTC(0x4949a698), WTC(0x47b4b7f9), WTC(0x45fe2b6d), WTC(0x44375019), WTC(0x4284e96e), WTC(0x40f7efa2), WTC(0x3f8f8b33), WTC(0x3e494311), WTC(0x3d21e35b), WTC(0x3c15c621), WTC(0x3b2115f3), WTC(0x3a4008aa), WTC(0x396ed2a6), WTC(0x38a99a1d), WTC(0x37ec1177), WTC(0x3730f154), WTC(0x36756c15), WTC(0x35bafb0d), WTC(0x35020093), WTC(0x34492381), WTC(0x338f6226), WTC(0x32d40a34), WTC(0x3215bd73), WTC(0x315302ce), WTC(0x308c7c41), WTC(0x2fc3532f), WTC(0x2ef6de8f), WTC(0x2e265a7f), WTC(0x2d527bfd), WTC(0x2c7bf035), WTC(0x2ba2975b), WTC(0x2ac63552), WTC(0x29e686ca), }; const FIXP_WTB ELDAnalysis128[384] = { WTC(0xfaa49e98), WTC(0xfa48929f), WTC(0xf9e7eb39), WTC(0xf983b829), WTC(0xf91bc5cb), WTC(0xf8b0376f), WTC(0xf8408d62), WTC(0xf7cd8c1e), WTC(0xf7580da3), WTC(0xf6e0b0dc), WTC(0xf667753c), WTC(0xf5efa4cf), WTC(0xf57cb6de), WTC(0xf511b62b), WTC(0xf4b1a860), WTC(0xf45ee8f8), WTC(0xf415710d), WTC(0xf3c0c4f3), WTC(0xf35c2af9), WTC(0xf2e89620), WTC(0xf266f3cb), WTC(0xf1d819bf), WTC(0xf13cff2f), WTC(0xf09489d2), WTC(0xefdcfa80), WTC(0xef182059), WTC(0xee4d6c60), WTC(0xed7b8da7), WTC(0xec9c27b1), WTC(0xebb57d0d), WTC(0xeacb3918), WTC(0xe9d35591), WTC(0xe8c9176e), WTC(0xe7b93e42), WTC(0xe6b10e47), WTC(0xe5a6b875), WTC(0xe484c345), WTC(0xe3509f1b), WTC(0xe224254c), WTC(0xe10a4e18), WTC(0xdff4a668), WTC(0xded3d881), WTC(0xdda5ed98), WTC(0xdc722d13), WTC(0xdb437360), WTC(0xda1fefed), WTC(0xd90623b2), WTC(0xd7f070f5), WTC(0xd6da361b), WTC(0xd5c0786f), WTC(0xd4a6e188), WTC(0xd39b37b4), WTC(0xd2a01ff2), WTC(0xd1a8a05c), WTC(0xd0b0cf47), WTC(0xcfbd3527), WTC(0xced6b8d7), WTC(0xcdff0a66), WTC(0xcd2978a4), WTC(0xcc587183), WTC(0xcb93bfdb), WTC(0xcad80773), WTC(0xca233c2b), WTC(0xc9768b5e), WTC(0xc8cc130c), WTC(0xc8231acd), WTC(0xc7768de4), WTC(0xc6c86bdf), WTC(0xc6181aa1), WTC(0xc563f6ce), WTC(0xc4b33a2a), WTC(0xc4085fcf), WTC(0xc35ae72e), WTC(0xc2ad7adf), WTC(0xc206ed94), WTC(0xc16a5744), WTC(0xc0d59625), WTC(0xc041e21b), WTC(0xbfb1ee05), WTC(0xbf2d82ea), WTC(0xbeb60fe9), WTC(0xbe499da8), WTC(0xbde61891), WTC(0xbd8975b6), WTC(0xbd339d36), WTC(0xbce6a08b), WTC(0xbca5b2f9), WTC(0xbc721002), WTC(0xbc494d41), WTC(0xbc266160), WTC(0xbc06d14f), WTC(0xbbec52c7), WTC(0xbbdaaf79), WTC(0xbbd0be99), WTC(0xbbcae139), WTC(0xbbcd359c), WTC(0xbbded5d3), WTC(0xbbfa58cf), WTC(0xbc162f9d), WTC(0xbc326534), WTC(0xbc4f081c), WTC(0xbc6c1678), WTC(0xbc899f93), WTC(0xbca7a263), WTC(0xbcc62954), WTC(0xbce52ddc), WTC(0xbd04bc7f), WTC(0xbd24cd8f), WTC(0xbd456998), WTC(0xbd668428), WTC(0xbd88207c), WTC(0xbdaa2e4b), WTC(0xbdccaf3e), WTC(0xbdef932c), WTC(0xbe12d936), WTC(0xbe366dd6), WTC(0xbe5a4fd9), WTC(0xbe7e6b49), WTC(0xbea2bf3f), WTC(0xbec738b5), WTC(0xbeebd791), WTC(0xbf108b49), WTC(0xbf3554aa), WTC(0xbf5a25cf), WTC(0xbf7f020b), WTC(0xbfa3dd25), WTC(0xbfc8be1e), WTC(0xbfed95f6), WTC(0x8024c933), WTC(0x806e24fb), WTC(0x80b73d96), WTC(0x80fff78c), WTC(0x8148612f), WTC(0x8190626e), WTC(0x81d8030a), WTC(0x821f28e1), WTC(0x8265d6be), WTC(0x82abed56), WTC(0x82f16e40), WTC(0x833636d2), WTC(0x837a470a), WTC(0x83bd7be6), WTC(0x83ffd415), WTC(0x84412e66), WTC(0x84818c4f), WTC(0x84c0d18c), WTC(0x84ff0429), WTC(0x853c09a5), WTC(0x8577eacf), WTC(0x85b29402), WTC(0x85ec17bf), WTC(0x86246b50), WTC(0x865ba7d7), WTC(0x8691c4c0), WTC(0x86c6d72a), WTC(0x86fae06c), WTC(0x872dfcd1), WTC(0x87602c6a), WTC(0x87918a27), WTC(0x87c22ef8), WTC(0x882f7c20), WTC(0x88dc38ab), WTC(0x89a52f47), WTC(0x8a737635), WTC(0x8b43d08d), WTC(0x8c19be9f), WTC(0x8cf89ce7), WTC(0x8de337e9), WTC(0x8edb3e02), WTC(0x8fe1e815), WTC(0x90f7e107), WTC(0x921d8b95), WTC(0x9353008c), WTC(0x94982fd7), WTC(0x95ecdc6d), WTC(0x9750b87f), WTC(0x98c36ad6), WTC(0x9a447674), WTC(0x9bd34e9c), WTC(0x9d6f76fa), WTC(0x9f18780d), WTC(0xa0cdc487), WTC(0xa28eff8e), WTC(0xa45bbe4b), WTC(0xa633baaf), WTC(0xa816bff0), WTC(0xaa04a90d), WTC(0xabfd7205), WTC(0xae01356b), WTC(0xb0101477), WTC(0xb22a5244), WTC(0xb4500b02), WTC(0x499946f3), WTC(0x475da5af), WTC(0x45173dce), WTC(0x42c610ad), WTC(0x406a44b0), WTC(0x3e037ce8), WTC(0x3b92b806), WTC(0x39195cd0), WTC(0x36962f17), WTC(0x340a1b28), WTC(0x3177cde6), WTC(0x2ee1f20d), WTC(0x2c49b0d6), WTC(0x29ad3d74), WTC(0x270e9ec1), WTC(0x2474132f), WTC(0x21e14a01), WTC(0x1f57704e), WTC(0x1cd758ce), WTC(0x1a610d48), WTC(0x17f5db88), WTC(0x1598a188), WTC(0x134f7a40), WTC(0x111f1ca0), WTC(0x0f053b83), WTC(0x0cf871db), WTC(0x0af19e60), WTC(0x08eaa56d), WTC(0x06e3c473), WTC(0x04d25cc9), WTC(0x02b59b4d), WTC(0x00e5bc4c), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0xffcebf14), WTC(0xff6cc249), WTC(0xff0b8bda), WTC(0xfeac3e5f), WTC(0xfe4f4638), WTC(0xfdf5052f), WTC(0xfd9e8cf4), WTC(0xfd4e34ea), WTC(0xfd023142), WTC(0xfcb8f6f7), WTC(0xfc74e090), WTC(0xfc3b33e8), WTC(0xfc0c1254), WTC(0xfbe1b2eb), WTC(0xfbb8ce73), WTC(0xfb97e511), WTC(0xfb86273d), WTC(0xfb857a3e), WTC(0xfb918813), WTC(0xfba43692), WTC(0xfbb96f24), WTC(0xfbd4dcc3), WTC(0xfc000cd5), WTC(0xfc458653), WTC(0xfca6cd98), WTC(0xfd178c8e), WTC(0xfd872979), WTC(0xfdfa721e), WTC(0xfe88c77c), WTC(0xff3c8b5c), WTC(0x00059ebc), WTC(0x00d91db0), WTC(0x01bec555), WTC(0x02bdfb80), WTC(0x03d4c846), WTC(0x0501ad53), WTC(0x064719b2), WTC(0x07a34a2b), WTC(0x0918814b), WTC(0x0aaaaa6b), WTC(0x0c5728b9), WTC(0x0e1c1a0f), WTC(0x0ff9ccd1), WTC(0x11f21ffb), WTC(0x1405d073), WTC(0x1635776b), WTC(0x1880f4e5), WTC(0x1ae8bdcb), WTC(0x1d6cede3), WTC(0x200e1d93), WTC(0x22cc56fd), WTC(0x25a80858), WTC(0x28a11bdd), WTC(0x2bb7e363), WTC(0x2eec2f0f), WTC(0x323e298d), WTC(0x35ad7f0b), WTC(0x393a1989), WTC(0x3ce34a65), WTC(0x40a8683d), WTC(0x44881c94), WTC(0x48813cb3), WTC(0x4c94523d), WTC(0x50c8eff0), WTC(0x00000000), WTC(0x00000000), WTC(0x0053a1ea), WTC(0x00f67066), WTC(0x01e3f61b), WTC(0x0317bdaf), WTC(0x048d51ca), WTC(0x06403d11), WTC(0x082c0a34), WTC(0x0a4c43d2), WTC(0x0c9c748e), WTC(0x0f182718), WTC(0x11bae616), WTC(0x14803c1b), WTC(0x1763b3e6), WTC(0x1a60d832), WTC(0x1d73336b), WTC(0x20965068), WTC(0x23c5b9d6), WTC(0x26fcfa1a), WTC(0x2a379c30), WTC(0x2d712a75), WTC(0x30a52fba), WTC(0x33cf36a9), WTC(0x36eaca15), WTC(0x39f3743b), WTC(0x3ce4bfc2), WTC(0x3fba37b8), WTC(0x426f6671), WTC(0x44ffd6e8), WTC(0x47671326), WTC(0x49a0b2d0), WTC(0x4ba81be1), WTC(0x4d78fd1e), WTC(0x4f0ed503), WTC(0x50652e28), WTC(0x51779351), WTC(0x52418fbe), WTC(0x52bead75), WTC(0x52ea76cc), WTC(0x52c07793), WTC(0x523c3918), WTC(0x5159470e), WTC(0x50132b36), WTC(0x4e657128), WTC(0x4c4ba31d), WTC(0x49c14b60), WTC(0x46c1e9e4), WTC(0x4374e179), WTC(0x408377b1), WTC(0x3e0fa0aa), WTC(0x3c05d584), WTC(0x3a4d9888), WTC(0x38cdd8fa), WTC(0x376b75c4), WTC(0x360c51a7), WTC(0x34b12fdc), WTC(0x33550d9f), WTC(0x31f1955c), WTC(0x307ffcb2), WTC(0x2f03c44d), WTC(0x2d7a6b86), WTC(0x2be6d3d4), WTC(0x2a48d219), }; const FIXP_WTB ELDAnalysis120[360] = { WTC(0xfaa1a40a), WTC(0xfa3f173d), WTC(0xf9d7760c), WTC(0xf96bcc12), WTC(0xf8fbe82c), WTC(0xf887bb26), WTC(0xf80f131a), WTC(0xf7930d03), WTC(0xf714aeaf), WTC(0xf693f5a1), WTC(0xf613385b), WTC(0xf596ef0e), WTC(0xf522dcc4), WTC(0xf4bab1f6), WTC(0xf461700c), WTC(0xf412e1fe), WTC(0xf3b769b1), WTC(0xf349eaca), WTC(0xf2cb9164), WTC(0xf23d6d7b), WTC(0xf1a0ab28), WTC(0xf0f5c20f), WTC(0xf03aadd9), WTC(0xef6e8c66), WTC(0xee984910), WTC(0xedbbcbf8), WTC(0xecd1435b), WTC(0xebdc1b77), WTC(0xeae31338), WTC(0xe9dbea9a), WTC(0xe8c005ea), WTC(0xe79e7068), WTC(0xe6856dce), WTC(0xe5657c79), WTC(0xe42928e9), WTC(0xe2e0756e), WTC(0xe1a83cf5), WTC(0xe0801fab), WTC(0xdf52bfeb), WTC(0xde15d1b1), WTC(0xdcce71ba), WTC(0xdb8931bf), WTC(0xda4fbbe8), WTC(0xd9220a98), WTC(0xd7f9af0c), WTC(0xd6d0e1df), WTC(0xd5a41f15), WTC(0xd4790330), WTC(0xd35f84b8), WTC(0xd255e881), WTC(0xd14daf00), WTC(0xd04638e2), WTC(0xcf47dff7), WTC(0xce5b8850), WTC(0xcd77b1d6), WTC(0xcc960ec0), WTC(0xcbc0a6a4), WTC(0xcaf6d4f7), WTC(0xca34fa7d), WTC(0xc97c26c0), WTC(0xc8c6868a), WTC(0xc811f873), WTC(0xc7599db8), WTC(0xc69f9780), WTC(0xc5e24043), WTC(0xc5226384), WTC(0xc468f547), WTC(0xc3b20be9), WTC(0xc2f826f0), WTC(0xc242ea02), WTC(0xc19844ae), WTC(0xc0f80714), WTC(0xc05a6da1), WTC(0xbfbfe6d3), WTC(0xbf31b5b4), WTC(0xbeb24843), WTC(0xbe3f4cb4), WTC(0xbdd63599), WTC(0xbd74c6b2), WTC(0xbd1b7128), WTC(0xbccd4b41), WTC(0xbc8dc08e), WTC(0xbc5ca2bd), WTC(0xbc3509f1), WTC(0xbc11f904), WTC(0xbbf37848), WTC(0xbbddfb6b), WTC(0xbbd214ea), WTC(0xbbcb3a11), WTC(0xbbcce581), WTC(0xbbdfa6ba), WTC(0xbbfd2fa5), WTC(0xbc1ad516), WTC(0xbc390c16), WTC(0xbc57b323), WTC(0xbc76dd1f), WTC(0xbc969172), WTC(0xbcb6d611), WTC(0xbcd7ac8e), WTC(0xbcf91af9), WTC(0xbd1b20bd), WTC(0xbd3dc1f5), WTC(0xbd60f678), WTC(0xbd84bec7), WTC(0xbda909c0), WTC(0xbdcdd76e), WTC(0xbdf3161c), WTC(0xbe18c1e5), WTC(0xbe3ec6c5), WTC(0xbe651f19), WTC(0xbe8bb78b), WTC(0xbeb288e1), WTC(0xbed9848b), WTC(0xbf00a16f), WTC(0xbf27d681), WTC(0xbf4f1958), WTC(0xbf76680e), WTC(0xbf9db85c), WTC(0xbfc50e09), WTC(0xbfec5bdf), WTC(0x80273bdb), WTC(0x807577de), WTC(0x80c3630d), WTC(0x8110e4cb), WTC(0x815e05da), WTC(0x81aab20f), WTC(0x81f6e68e), WTC(0x824290d0), WTC(0x828da04c), WTC(0x82d8061f), WTC(0x8321a740), WTC(0x836a77f9), WTC(0x83b2574f), WTC(0x83f93cc1), WTC(0x843f04a6), WTC(0x8483acd1), WTC(0x84c71639), WTC(0x850944da), WTC(0x854a1d3a), WTC(0x8589a437), WTC(0x85c7ccf9), WTC(0x8604a42f), WTC(0x86402cfb), WTC(0x867a740d), WTC(0x86b37ff2), WTC(0x86eb5f6f), WTC(0x87222109), WTC(0x8757eb5f), WTC(0x878c8341), WTC(0x87c0be51), WTC(0x88345fca), WTC(0x88ef97fb), WTC(0x89c778c5), WTC(0x8aa3cc20), WTC(0x8b833d56), WTC(0x8c6a42a9), WTC(0x8d5cb787), WTC(0x8e5d7787), WTC(0x8f6e3c5b), WTC(0x90902640), WTC(0x91c3ca28), WTC(0x9309622d), WTC(0x9460e7d0), WTC(0x95ca1ad2), WTC(0x97449dff), WTC(0x98d00612), WTC(0x9a6bbc19), WTC(0x9c17164d), WTC(0x9dd180fc), WTC(0x9f9a6304), WTC(0xa1711f56), WTC(0xa355425e), WTC(0xa5465846), WTC(0xa744190f), WTC(0xa94e4ca9), WTC(0xab64dcf0), WTC(0xad87e068), WTC(0xafb77bfa), WTC(0xb1f3fb2b), WTC(0xb43d885c), WTC(0x49866451), WTC(0x4723e56b), WTC(0x44b51e8d), WTC(0x423a2180), WTC(0x3fb2fe0f), WTC(0x3d1f78f2), WTC(0x3a8153b7), WTC(0x37d906ff), WTC(0x35259e7e), WTC(0x3269ba18), WTC(0x2fa8c1ea), WTC(0x2ce4fbab), WTC(0x2a1cebd5), WTC(0x27519ac1), WTC(0x248a2de9), WTC(0x21cb7a79), WTC(0x1f16fc13), WTC(0x1c6d9a50), WTC(0x19cf83c1), WTC(0x173e9943), WTC(0x14bf6a71), WTC(0x12598f74), WTC(0x100fe27f), WTC(0x0ddab46a), WTC(0x0bafab9d), WTC(0x09864fc7), WTC(0x075d3ac8), WTC(0x052c0fc2), WTC(0x02ea842b), WTC(0x00f21d19), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0xffcb7b4d), WTC(0xff62fb20), WTC(0xfefb82e2), WTC(0xfe965482), WTC(0xfe33e4bf), WTC(0xfdd4b9b5), WTC(0xfd7b04ec), WTC(0xfd27cfe4), WTC(0xfcd84cfd), WTC(0xfc8ce1d4), WTC(0xfc4b45e1), WTC(0xfc1666da), WTC(0xfbe8af26), WTC(0xfbbcad9e), WTC(0xfb98c6ad), WTC(0xfb85dd87), WTC(0xfb86355b), WTC(0xfb94589b), WTC(0xfba8ed8a), WTC(0xfbc0a735), WTC(0xfbe23f8d), WTC(0xfc1a92c4), WTC(0xfc73315f), WTC(0xfce611a1), WTC(0xfd5e94d9), WTC(0xfdd61cab), WTC(0xfe6427fd), WTC(0xff1c92da), WTC(0xfff10542), WTC(0x00d1d58a), WTC(0x01c6daab), WTC(0x02d8dbc3), WTC(0x040557a6), WTC(0x054b6f91), WTC(0x06ad2b2b), WTC(0x0828f4c5), WTC(0x09c348d1), WTC(0x0b7dcb56), WTC(0x0d54d98e), WTC(0x0f47a599), WTC(0x1157fa1b), WTC(0x138743ae), WTC(0x15d6423f), WTC(0x1844f0ae), WTC(0x1ad3c273), WTC(0x1d82e65c), WTC(0x205306e1), WTC(0x23443d5c), WTC(0x2656fae8), WTC(0x298b3a5a), WTC(0x2ce13a7d), WTC(0x3058e0e3), WTC(0x33f22950), WTC(0x37acd0b2), WTC(0x3b885e6b), WTC(0x3f840412), WTC(0x439e65ee), WTC(0x47d6014e), WTC(0x4c2aa857), WTC(0x50a46876), WTC(0xfffe9b02), WTC(0x0004ac61), WTC(0x0069639d), WTC(0x0127578b), WTC(0x02391efe), WTC(0x039950cc), WTC(0x054283bf), WTC(0x072f4eba), WTC(0x095a488c), WTC(0x0bbe0802), WTC(0x0e5523f9), WTC(0x111a332e), WTC(0x1407cca2), WTC(0x171886f5), WTC(0x1a46f927), WTC(0x1d8db9e1), WTC(0x20e7600d), WTC(0x244e82a3), WTC(0x27bdb846), WTC(0x2b2f97a8), WTC(0x2e9eb7ea), WTC(0x3205afd1), WTC(0x355f161c), WTC(0x38a581d7), WTC(0x3bd3894a), WTC(0x3ee3c398), WTC(0x41d0c7ae), WTC(0x44952cb8), WTC(0x472b8856), WTC(0x498e7eee), WTC(0x4bb88245), WTC(0x4da44d9f), WTC(0x4f4c6b7d), WTC(0x50ab7298), WTC(0x51bbfa11), WTC(0x527898fb), WTC(0x52dbe5f2), WTC(0x52e07778), WTC(0x5280e51f), WTC(0x51b7c4f1), WTC(0x507fadc7), WTC(0x4ed336dd), WTC(0x4cacf749), WTC(0x4a078534), WTC(0x46dd6dcc), WTC(0x43599e22), WTC(0x403f48da), WTC(0x3db1ed89), WTC(0x3b98bd24), WTC(0x39d5b003), WTC(0x384a3292), WTC(0x36d328ff), WTC(0x355e63a6), WTC(0x33ec69d8), WTC(0x32755ebd), WTC(0x30f01fce), WTC(0x2f5d9646), WTC(0x2dbcc615), WTC(0x2c0fa145), WTC(0x2a56ce53), }; fdk-aac-2.0.2/libAACenc/src/aacEnc_rom.h000066400000000000000000000210461404225310700175320ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Lohwasser, M. Gayer Description: *******************************************************************************/ /*! \file \brief Memory layout \author Markus Lohwasser */ #ifndef AACENC_ROM_H #define AACENC_ROM_H #include "common_fix.h" #include "psy_const.h" #include "psy_configuration.h" #include "FDK_tools_rom.h" #include "FDK_lpc.h" /* Huffman Tables */ extern const ULONG FDKaacEnc_huff_ltab1_2[3][3][3][3]; extern const ULONG FDKaacEnc_huff_ltab3_4[3][3][3][3]; extern const ULONG FDKaacEnc_huff_ltab5_6[9][9]; extern const ULONG FDKaacEnc_huff_ltab7_8[8][8]; extern const ULONG FDKaacEnc_huff_ltab9_10[13][13]; extern const UCHAR FDKaacEnc_huff_ltab11[17][17]; extern const UCHAR FDKaacEnc_huff_ltabscf[121]; extern const USHORT FDKaacEnc_huff_ctab1[3][3][3][3]; extern const USHORT FDKaacEnc_huff_ctab2[3][3][3][3]; extern const USHORT FDKaacEnc_huff_ctab3[3][3][3][3]; extern const USHORT FDKaacEnc_huff_ctab4[3][3][3][3]; extern const USHORT FDKaacEnc_huff_ctab5[9][9]; extern const USHORT FDKaacEnc_huff_ctab6[9][9]; extern const USHORT FDKaacEnc_huff_ctab7[8][8]; extern const USHORT FDKaacEnc_huff_ctab8[8][8]; extern const USHORT FDKaacEnc_huff_ctab9[13][13]; extern const USHORT FDKaacEnc_huff_ctab10[13][13]; extern const USHORT FDKaacEnc_huff_ctab11[21][17]; extern const ULONG FDKaacEnc_huff_ctabscf[121]; /* quantizer */ #define MANT_DIGITS 9 #define MANT_SIZE (1 << MANT_DIGITS) #if defined(ARCH_PREFER_MULT_32x16) #define FIXP_QTD FIXP_SGL #define QTC FX_DBL2FXCONST_SGL #else #define FIXP_QTD FIXP_DBL #define QTC #endif extern const FIXP_QTD FDKaacEnc_mTab_3_4[MANT_SIZE]; extern const FIXP_QTD FDKaacEnc_quantTableQ[4]; extern const FIXP_QTD FDKaacEnc_quantTableE[4]; extern const FIXP_DBL FDKaacEnc_mTab_4_3Elc[512]; extern const FIXP_DBL FDKaacEnc_specExpMantTableCombElc[4][14]; extern const UCHAR FDKaacEnc_specExpTableComb[4][14]; /* table to count used number of bits */ extern const SHORT FDKaacEnc_sideInfoTabLong[]; extern const SHORT FDKaacEnc_sideInfoTabShort[]; /* Psy Configuration constants */ extern const SFB_PARAM_LONG p_FDKaacEnc_8000_long_1024; extern const SFB_PARAM_SHORT p_FDKaacEnc_8000_short_128; extern const SFB_PARAM_LONG p_FDKaacEnc_11025_long_1024; extern const SFB_PARAM_SHORT p_FDKaacEnc_11025_short_128; extern const SFB_PARAM_LONG p_FDKaacEnc_12000_long_1024; extern const SFB_PARAM_SHORT p_FDKaacEnc_12000_short_128; extern const SFB_PARAM_LONG p_FDKaacEnc_16000_long_1024; extern const SFB_PARAM_SHORT p_FDKaacEnc_16000_short_128; extern const SFB_PARAM_LONG p_FDKaacEnc_22050_long_1024; extern const SFB_PARAM_SHORT p_FDKaacEnc_22050_short_128; extern const SFB_PARAM_LONG p_FDKaacEnc_24000_long_1024; extern const SFB_PARAM_SHORT p_FDKaacEnc_24000_short_128; extern const SFB_PARAM_LONG p_FDKaacEnc_32000_long_1024; extern const SFB_PARAM_SHORT p_FDKaacEnc_32000_short_128; extern const SFB_PARAM_LONG p_FDKaacEnc_44100_long_1024; extern const SFB_PARAM_SHORT p_FDKaacEnc_44100_short_128; extern const SFB_PARAM_LONG p_FDKaacEnc_48000_long_1024; extern const SFB_PARAM_SHORT p_FDKaacEnc_48000_short_128; extern const SFB_PARAM_LONG p_FDKaacEnc_64000_long_1024; extern const SFB_PARAM_SHORT p_FDKaacEnc_64000_short_128; extern const SFB_PARAM_LONG p_FDKaacEnc_88200_long_1024; extern const SFB_PARAM_SHORT p_FDKaacEnc_88200_short_128; extern const SFB_PARAM_LONG p_FDKaacEnc_96000_long_1024; extern const SFB_PARAM_SHORT p_FDKaacEnc_96000_short_128; /* TNS filter coefficients */ extern const FIXP_LPC FDKaacEnc_tnsEncCoeff3[8]; extern const FIXP_LPC FDKaacEnc_tnsCoeff3Borders[8]; extern const FIXP_LPC FDKaacEnc_tnsEncCoeff4[16]; extern const FIXP_LPC FDKaacEnc_tnsCoeff4Borders[16]; #define WTC0 WTC #define WTC1 WTC #define WTC2 WTC extern const FIXP_WTB ELDAnalysis512[1536]; extern const FIXP_WTB ELDAnalysis480[1440]; extern const FIXP_WTB ELDAnalysis256[768]; extern const FIXP_WTB ELDAnalysis240[720]; extern const FIXP_WTB ELDAnalysis128[384]; extern const FIXP_WTB ELDAnalysis120[360]; #endif /* #ifndef AACENC_ROM_H */ fdk-aac-2.0.2/libAACenc/src/aacenc.cpp000066400000000000000000001154161404225310700172550ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Schug / A. Groeschel Description: fast aac coder functions *******************************************************************************/ #include "aacenc.h" #include "bitenc.h" #include "interface.h" #include "psy_configuration.h" #include "psy_main.h" #include "qc_main.h" #include "bandwidth.h" #include "channel_map.h" #include "tns_func.h" #include "aacEnc_ram.h" #include "genericStds.h" #define BITRES_MIN \ 300 /* default threshold for using reduced/disabled bitres mode */ #define BITRES_MAX_LD 4000 #define BITRES_MIN_LD 500 #define BITRATE_MAX_LD 70000 /* Max assumed bitrate for bitres calculation */ #define BITRATE_MIN_LD 12000 /* Min assumed bitrate for bitres calculation */ INT FDKaacEnc_CalcBitsPerFrame(const INT bitRate, const INT frameLength, const INT samplingRate) { int shift = 0; while ((frameLength & ~((1 << (shift + 1)) - 1)) == frameLength && (samplingRate & ~((1 << (shift + 1)) - 1)) == samplingRate) { shift++; } return (bitRate * (frameLength >> shift)) / (samplingRate >> shift); } INT FDKaacEnc_CalcBitrate(const INT bitsPerFrame, const INT frameLength, const INT samplingRate) { int shift = 0; while ((frameLength & ~((1 << (shift + 1)) - 1)) == frameLength && (samplingRate & ~((1 << (shift + 1)) - 1)) == samplingRate) { shift++; } return (bitsPerFrame * (samplingRate >> shift)) / (frameLength >> shift); } static AAC_ENCODER_ERROR FDKaacEnc_InitCheckAncillary( INT bitRate, INT framelength, INT ancillaryRate, INT *ancillaryBitsPerFrame, INT sampleRate); INT FDKaacEnc_LimitBitrate(HANDLE_TRANSPORTENC hTpEnc, AUDIO_OBJECT_TYPE aot, INT coreSamplingRate, INT frameLength, INT nChannels, INT nChannelsEff, INT bitRate, INT averageBits, INT *pAverageBitsPerFrame, AACENC_BITRATE_MODE bitrateMode, INT nSubFrames) { INT transportBits, prevBitRate, averageBitsPerFrame, minBitrate = 0, iter = 0; INT minBitsPerFrame = 40 * nChannels; if (isLowDelay(aot)) { minBitrate = 8000 * nChannelsEff; } do { prevBitRate = bitRate; averageBitsPerFrame = FDKaacEnc_CalcBitsPerFrame(bitRate, frameLength, coreSamplingRate) / nSubFrames; if (pAverageBitsPerFrame != NULL) { *pAverageBitsPerFrame = averageBitsPerFrame; } if (hTpEnc != NULL) { transportBits = transportEnc_GetStaticBits(hTpEnc, averageBitsPerFrame); } else { /* Assume some worst case */ transportBits = 208; } bitRate = fMax(bitRate, fMax(minBitrate, FDKaacEnc_CalcBitrate((minBitsPerFrame + transportBits), frameLength, coreSamplingRate))); FDK_ASSERT(bitRate >= 0); bitRate = fMin(bitRate, FDKaacEnc_CalcBitrate( (nChannelsEff * MIN_BUFSIZE_PER_EFF_CHAN), frameLength, coreSamplingRate)); FDK_ASSERT(bitRate >= 0); } while (prevBitRate != bitRate && iter++ < 3); return bitRate; } typedef struct { AACENC_BITRATE_MODE bitrateMode; int chanBitrate[2]; /* mono/stereo settings */ } CONFIG_TAB_ENTRY_VBR; static const CONFIG_TAB_ENTRY_VBR configTabVBR[] = { {AACENC_BR_MODE_CBR, {0, 0}}, {AACENC_BR_MODE_VBR_1, {32000, 20000}}, {AACENC_BR_MODE_VBR_2, {40000, 32000}}, {AACENC_BR_MODE_VBR_3, {56000, 48000}}, {AACENC_BR_MODE_VBR_4, {72000, 64000}}, {AACENC_BR_MODE_VBR_5, {112000, 96000}}}; /*----------------------------------------------------------------------------- functionname: FDKaacEnc_GetVBRBitrate description: Get VBR bitrate from vbr quality input params: int vbrQuality (VBR0, VBR1, VBR2) channelMode returns: vbr bitrate ------------------------------------------------------------------------------*/ INT FDKaacEnc_GetVBRBitrate(AACENC_BITRATE_MODE bitrateMode, CHANNEL_MODE channelMode) { INT bitrate = 0; INT monoStereoMode = 0; /* default mono */ if (FDKaacEnc_GetMonoStereoMode(channelMode) == EL_MODE_STEREO) { monoStereoMode = 1; } switch (bitrateMode) { case AACENC_BR_MODE_VBR_1: case AACENC_BR_MODE_VBR_2: case AACENC_BR_MODE_VBR_3: case AACENC_BR_MODE_VBR_4: case AACENC_BR_MODE_VBR_5: bitrate = configTabVBR[bitrateMode].chanBitrate[monoStereoMode]; break; case AACENC_BR_MODE_INVALID: case AACENC_BR_MODE_CBR: case AACENC_BR_MODE_SFR: case AACENC_BR_MODE_FF: default: bitrate = 0; break; } /* convert channel bitrate to overall bitrate*/ bitrate *= FDKaacEnc_GetChannelModeConfiguration(channelMode)->nChannelsEff; return bitrate; } /*----------------------------------------------------------------------------- functionname: FDKaacEnc_AdjustVBRBitrateMode description: Adjust bitrate mode to given bitrate parameter input params: int vbrQuality (VBR0, VBR1, VBR2) bitrate channelMode returns: vbr bitrate mode ------------------------------------------------------------------------------*/ AACENC_BITRATE_MODE FDKaacEnc_AdjustVBRBitrateMode( AACENC_BITRATE_MODE bitrateMode, INT bitrate, CHANNEL_MODE channelMode) { AACENC_BITRATE_MODE newBitrateMode = bitrateMode; if (bitrate != -1) { const INT monoStereoMode = (FDKaacEnc_GetMonoStereoMode(channelMode) == EL_MODE_STEREO) ? 1 : 0; const INT nChannelsEff = FDKaacEnc_GetChannelModeConfiguration(channelMode)->nChannelsEff; newBitrateMode = AACENC_BR_MODE_INVALID; for (int idx = (int)(sizeof(configTabVBR) / sizeof(*configTabVBR)) - 1; idx >= 0; idx--) { if (bitrate >= configTabVBR[idx].chanBitrate[monoStereoMode] * nChannelsEff) { if (configTabVBR[idx].chanBitrate[monoStereoMode] * nChannelsEff < FDKaacEnc_GetVBRBitrate(bitrateMode, channelMode)) { newBitrateMode = configTabVBR[idx].bitrateMode; } else { newBitrateMode = bitrateMode; } break; } } } return AACENC_BR_MODE_IS_VBR(newBitrateMode) ? newBitrateMode : AACENC_BR_MODE_INVALID; } /** * \brief Convert encoder bitreservoir value for transport library. * * \param hAacEnc Encoder handle * * \return Corrected bitreservoir level used in transport library. */ static INT FDKaacEnc_EncBitresToTpBitres(const HANDLE_AAC_ENC hAacEnc) { INT transportBitreservoir = 0; switch (hAacEnc->bitrateMode) { case AACENC_BR_MODE_CBR: transportBitreservoir = hAacEnc->qcKernel->bitResTot; /* encoder bitreservoir level */ break; case AACENC_BR_MODE_VBR_1: case AACENC_BR_MODE_VBR_2: case AACENC_BR_MODE_VBR_3: case AACENC_BR_MODE_VBR_4: case AACENC_BR_MODE_VBR_5: transportBitreservoir = FDK_INT_MAX; /* signal variable bitrate */ break; case AACENC_BR_MODE_SFR: transportBitreservoir = 0; /* super framing and fixed framing */ break; /* without bitreservoir signaling */ default: case AACENC_BR_MODE_INVALID: transportBitreservoir = 0; /* invalid configuration*/ } if (hAacEnc->config->audioMuxVersion == 2) { transportBitreservoir = MIN_BUFSIZE_PER_EFF_CHAN * hAacEnc->channelMapping.nChannelsEff; } return transportBitreservoir; } INT FDKaacEnc_GetBitReservoirState(const HANDLE_AAC_ENC hAacEncoder) { return FDKaacEnc_EncBitresToTpBitres(hAacEncoder); } /*----------------------------------------------------------------------------- functionname: FDKaacEnc_AacInitDefaultConfig description: gives reasonable default configuration returns: --- ------------------------------------------------------------------------------*/ void FDKaacEnc_AacInitDefaultConfig(AACENC_CONFIG *config) { /* make the preinitialization of the structs flexible */ FDKmemclear(config, sizeof(AACENC_CONFIG)); /* default ancillary */ config->anc_Rate = 0; /* no ancillary data */ config->ancDataBitRate = 0; /* no additional consumed bitrate */ /* default configurations */ config->bitRate = -1; /* bitrate must be set*/ config->averageBits = -1; /* instead of bitrate/s we can configure bits/superframe */ config->bitrateMode = AACENC_BR_MODE_CBR; /* set bitrate mode to constant bitrate */ config->bandWidth = 0; /* get bandwidth from table */ config->useTns = TNS_ENABLE_MASK; /* tns enabled completly */ config->usePns = 1; /* depending on channelBitrate this might be set to 0 later */ config->useIS = 1; /* Intensity Stereo Configuration */ config->useMS = 1; /* MS Stereo tool */ config->framelength = -1; /* Framesize not configured */ config->syntaxFlags = 0; /* default syntax with no specialities */ config->epConfig = -1; /* no ER syntax -> no additional error protection */ config->nSubFrames = 1; /* default, no sub frames */ config->channelOrder = CH_ORDER_MPEG; /* Use MPEG channel ordering. */ config->channelMode = MODE_UNKNOWN; config->minBitsPerFrame = -1; /* minum number of bits in each AU */ config->maxBitsPerFrame = -1; /* minum number of bits in each AU */ config->audioMuxVersion = -1; /* audio mux version not configured */ config->downscaleFactor = 1; /* downscale factor for ELD reduced delay mode, 1 is normal ELD */ } /*--------------------------------------------------------------------------- functionname: FDKaacEnc_Open description: allocate and initialize a new encoder instance returns: error code ---------------------------------------------------------------------------*/ AAC_ENCODER_ERROR FDKaacEnc_Open(HANDLE_AAC_ENC *phAacEnc, const INT nElements, const INT nChannels, const INT nSubFrames) { AAC_ENCODER_ERROR ErrorStatus; AAC_ENC *hAacEnc = NULL; UCHAR *dynamicRAM = NULL; if (phAacEnc == NULL) { return AAC_ENC_INVALID_HANDLE; } /* allocate encoder structure */ hAacEnc = GetRam_aacEnc_AacEncoder(); if (hAacEnc == NULL) { ErrorStatus = AAC_ENC_NO_MEMORY; goto bail; } FDKmemclear(hAacEnc, sizeof(AAC_ENC)); if (NULL == (hAacEnc->dynamic_RAM = GetAACdynamic_RAM())) { ErrorStatus = AAC_ENC_NO_MEMORY; goto bail; } dynamicRAM = (UCHAR *)hAacEnc->dynamic_RAM; /* allocate the Psy aud Psy Out structure */ ErrorStatus = FDKaacEnc_PsyNew(&hAacEnc->psyKernel, nElements, nChannels, dynamicRAM); if (ErrorStatus != AAC_ENC_OK) goto bail; ErrorStatus = FDKaacEnc_PsyOutNew(hAacEnc->psyOut, nElements, nChannels, nSubFrames, dynamicRAM); if (ErrorStatus != AAC_ENC_OK) goto bail; /* allocate the Q&C Out structure */ ErrorStatus = FDKaacEnc_QCOutNew(hAacEnc->qcOut, nElements, nChannels, nSubFrames, dynamicRAM); if (ErrorStatus != AAC_ENC_OK) goto bail; /* allocate the Q&C kernel */ ErrorStatus = FDKaacEnc_QCNew(&hAacEnc->qcKernel, nElements, dynamicRAM); if (ErrorStatus != AAC_ENC_OK) goto bail; hAacEnc->maxChannels = nChannels; hAacEnc->maxElements = nElements; hAacEnc->maxFrames = nSubFrames; bail: *phAacEnc = hAacEnc; return ErrorStatus; } AAC_ENCODER_ERROR FDKaacEnc_Initialize( HANDLE_AAC_ENC hAacEnc, AACENC_CONFIG *config, /* pre-initialized config struct */ HANDLE_TRANSPORTENC hTpEnc, ULONG initFlags) { AAC_ENCODER_ERROR ErrorStatus; INT psyBitrate, tnsMask; // INT profile = 1; CHANNEL_MAPPING *cm = NULL; INT mbfac_e, qbw; FIXP_DBL mbfac, bw_ratio; QC_INIT qcInit; INT averageBitsPerFrame = 0; const CHANNEL_MODE prevChannelMode = hAacEnc->encoderMode; if (config == NULL) return AAC_ENC_INVALID_HANDLE; /******************* sanity checks *******************/ /* check config structure */ if (config->nChannels < 1 || config->nChannels > (8)) { return AAC_ENC_UNSUPPORTED_CHANNELCONFIG; } /* check sample rate */ switch (config->sampleRate) { case 8000: case 11025: case 12000: case 16000: case 22050: case 24000: case 32000: case 44100: case 48000: case 64000: case 88200: case 96000: break; default: return AAC_ENC_UNSUPPORTED_SAMPLINGRATE; } /* bitrate has to be set */ if (config->bitRate == -1) { return AAC_ENC_UNSUPPORTED_BITRATE; } /* check bit rate */ if (FDKaacEnc_LimitBitrate( hTpEnc, config->audioObjectType, config->sampleRate, config->framelength, config->nChannels, FDKaacEnc_GetChannelModeConfiguration(config->channelMode) ->nChannelsEff, config->bitRate, config->averageBits, &averageBitsPerFrame, config->bitrateMode, config->nSubFrames) != config->bitRate && !(AACENC_BR_MODE_IS_VBR(config->bitrateMode))) { return AAC_ENC_UNSUPPORTED_BITRATE; } if (config->syntaxFlags & AC_ER_VCB11) { return AAC_ENC_UNSUPPORTED_ER_FORMAT; } if (config->syntaxFlags & AC_ER_HCR) { return AAC_ENC_UNSUPPORTED_ER_FORMAT; } /* check frame length */ switch (config->framelength) { case 1024: if (isLowDelay(config->audioObjectType)) { return AAC_ENC_INVALID_FRAME_LENGTH; } break; case 128: case 256: case 512: case 120: case 240: case 480: if (!isLowDelay(config->audioObjectType)) { return AAC_ENC_INVALID_FRAME_LENGTH; } break; default: return AAC_ENC_INVALID_FRAME_LENGTH; } if (config->anc_Rate != 0) { ErrorStatus = FDKaacEnc_InitCheckAncillary( config->bitRate, config->framelength, config->anc_Rate, &hAacEnc->ancillaryBitsPerFrame, config->sampleRate); if (ErrorStatus != AAC_ENC_OK) goto bail; /* update estimated consumed bitrate */ config->ancDataBitRate += FDKaacEnc_CalcBitrate(hAacEnc->ancillaryBitsPerFrame, config->framelength, config->sampleRate); } /* maximal allowed DSE bytes in frame */ config->maxAncBytesPerAU = fMin((256), fMax(0, FDKaacEnc_CalcBitsPerFrame( (config->bitRate - (config->nChannels * 8000)), config->framelength, config->sampleRate) >> 3)); /* bind config to hAacEnc->config */ hAacEnc->config = config; /* set hAacEnc->bitrateMode */ hAacEnc->bitrateMode = config->bitrateMode; hAacEnc->encoderMode = config->channelMode; ErrorStatus = FDKaacEnc_InitChannelMapping( hAacEnc->encoderMode, config->channelOrder, &hAacEnc->channelMapping); if (ErrorStatus != AAC_ENC_OK) goto bail; cm = &hAacEnc->channelMapping; ErrorStatus = FDKaacEnc_DetermineBandWidth( config->bandWidth, config->bitRate - config->ancDataBitRate, hAacEnc->bitrateMode, config->sampleRate, config->framelength, cm, hAacEnc->encoderMode, &hAacEnc->config->bandWidth); if (ErrorStatus != AAC_ENC_OK) goto bail; hAacEnc->bandwidth90dB = (INT)hAacEnc->config->bandWidth; tnsMask = config->useTns ? TNS_ENABLE_MASK : 0x0; psyBitrate = config->bitRate - config->ancDataBitRate; if ((hAacEnc->encoderMode != prevChannelMode) || (initFlags != 0)) { /* Reinitialize psych states in case of channel configuration change ore if * full reset requested. */ ErrorStatus = FDKaacEnc_psyInit(hAacEnc->psyKernel, hAacEnc->psyOut, hAacEnc->maxFrames, hAacEnc->maxChannels, config->audioObjectType, cm); if (ErrorStatus != AAC_ENC_OK) goto bail; } ErrorStatus = FDKaacEnc_psyMainInit( hAacEnc->psyKernel, config->audioObjectType, cm, config->sampleRate, config->framelength, psyBitrate, tnsMask, hAacEnc->bandwidth90dB, config->usePns, config->useIS, config->useMS, config->syntaxFlags, initFlags); if (ErrorStatus != AAC_ENC_OK) goto bail; ErrorStatus = FDKaacEnc_QCOutInit(hAacEnc->qcOut, hAacEnc->maxFrames, cm); if (ErrorStatus != AAC_ENC_OK) goto bail; qcInit.channelMapping = &hAacEnc->channelMapping; qcInit.sceCpe = 0; if (AACENC_BR_MODE_IS_VBR(config->bitrateMode)) { qcInit.averageBits = (averageBitsPerFrame + 7) & ~7; qcInit.bitRes = MIN_BUFSIZE_PER_EFF_CHAN * cm->nChannelsEff; qcInit.maxBits = MIN_BUFSIZE_PER_EFF_CHAN * cm->nChannelsEff; qcInit.maxBits = (config->maxBitsPerFrame != -1) ? fixMin(qcInit.maxBits, config->maxBitsPerFrame) : qcInit.maxBits; qcInit.maxBits = fixMax(qcInit.maxBits, (averageBitsPerFrame + 7) & ~7); qcInit.minBits = (config->minBitsPerFrame != -1) ? config->minBitsPerFrame : 0; qcInit.minBits = fixMin(qcInit.minBits, averageBitsPerFrame & ~7); } else { INT bitreservoir = -1; /* default bitreservoir size*/ if (isLowDelay(config->audioObjectType)) { INT brPerChannel = config->bitRate / config->nChannels; brPerChannel = fMin(BITRATE_MAX_LD, fMax(BITRATE_MIN_LD, brPerChannel)); /* bitreservoir = * (maxBitRes-minBitRes)/(maxBitRate-minBitrate)*(bitRate-minBitrate)+minBitRes; */ FIXP_DBL slope = fDivNorm( (brPerChannel - BITRATE_MIN_LD), BITRATE_MAX_LD - BITRATE_MIN_LD); /* calc slope for interpolation */ bitreservoir = fMultI(slope, (INT)(BITRES_MAX_LD - BITRES_MIN_LD)) + BITRES_MIN_LD; /* interpolate */ bitreservoir = bitreservoir & ~7; /* align to bytes */ } int maxBitres; qcInit.averageBits = (averageBitsPerFrame + 7) & ~7; maxBitres = (MIN_BUFSIZE_PER_EFF_CHAN * cm->nChannelsEff) - qcInit.averageBits; qcInit.bitRes = (bitreservoir != -1) ? fMin(bitreservoir, maxBitres) : maxBitres; qcInit.maxBits = fixMin(MIN_BUFSIZE_PER_EFF_CHAN * cm->nChannelsEff, ((averageBitsPerFrame + 7) & ~7) + qcInit.bitRes); qcInit.maxBits = (config->maxBitsPerFrame != -1) ? fixMin(qcInit.maxBits, config->maxBitsPerFrame) : qcInit.maxBits; qcInit.maxBits = fixMin(MIN_BUFSIZE_PER_EFF_CHAN * cm->nChannelsEff, fixMax(qcInit.maxBits, (averageBitsPerFrame + 7 + 8) & ~7)); qcInit.minBits = fixMax( 0, ((averageBitsPerFrame - 1) & ~7) - qcInit.bitRes - transportEnc_GetStaticBits( hTpEnc, ((averageBitsPerFrame + 7) & ~7) + qcInit.bitRes)); qcInit.minBits = (config->minBitsPerFrame != -1) ? fixMax(qcInit.minBits, config->minBitsPerFrame) : qcInit.minBits; qcInit.minBits = fixMin( qcInit.minBits, (averageBitsPerFrame - transportEnc_GetStaticBits(hTpEnc, qcInit.maxBits)) & ~7); } qcInit.sampleRate = config->sampleRate; qcInit.isLowDelay = isLowDelay(config->audioObjectType) ? 1 : 0; qcInit.nSubFrames = config->nSubFrames; qcInit.padding.paddingRest = config->sampleRate; if (qcInit.maxBits - qcInit.averageBits >= ((qcInit.isLowDelay) ? BITRES_MIN_LD : BITRES_MIN) * config->nChannels) { qcInit.bitResMode = AACENC_BR_MODE_FULL; /* full bitreservoir */ } else if (qcInit.maxBits > qcInit.averageBits) { qcInit.bitResMode = AACENC_BR_MODE_REDUCED; /* reduced bitreservoir */ } else { qcInit.bitResMode = AACENC_BR_MODE_DISABLED; /* disabled bitreservoir */ } /* Configure bitrate distribution strategy. */ switch (config->channelMode) { case MODE_1_2: case MODE_1_2_1: case MODE_1_2_2: case MODE_1_2_2_1: case MODE_6_1: case MODE_1_2_2_2_1: case MODE_7_1_BACK: case MODE_7_1_TOP_FRONT: case MODE_7_1_REAR_SURROUND: case MODE_7_1_FRONT_CENTER: qcInit.bitDistributionMode = 0; /* over all elements bitrate estimation */ break; case MODE_1: case MODE_2: default: /* all non mpeg defined channel modes */ qcInit.bitDistributionMode = 1; /* element-wise bit bitrate estimation */ } /* config->channelMode */ /* Calc meanPe: qcInit.meanPe = 10.0f * FRAME_LEN_LONG * * hAacEnc->bandwidth90dB/(config->sampleRate/2.0f); */ bw_ratio = fDivNorm((FIXP_DBL)(10 * config->framelength * hAacEnc->bandwidth90dB), (FIXP_DBL)(config->sampleRate), &qbw); qcInit.meanPe = fMax((INT)scaleValue(bw_ratio, qbw + 1 - (DFRACT_BITS - 1)), 1); /* Calc maxBitFac, scale it to 24 bit accuracy */ mbfac = fDivNorm(qcInit.maxBits, qcInit.averageBits / qcInit.nSubFrames, &mbfac_e); qcInit.maxBitFac = scaleValue(mbfac, -(DFRACT_BITS - 1 - 24 - mbfac_e)); switch (config->bitrateMode) { case AACENC_BR_MODE_CBR: qcInit.bitrateMode = QCDATA_BR_MODE_CBR; break; case AACENC_BR_MODE_VBR_1: qcInit.bitrateMode = QCDATA_BR_MODE_VBR_1; break; case AACENC_BR_MODE_VBR_2: qcInit.bitrateMode = QCDATA_BR_MODE_VBR_2; break; case AACENC_BR_MODE_VBR_3: qcInit.bitrateMode = QCDATA_BR_MODE_VBR_3; break; case AACENC_BR_MODE_VBR_4: qcInit.bitrateMode = QCDATA_BR_MODE_VBR_4; break; case AACENC_BR_MODE_VBR_5: qcInit.bitrateMode = QCDATA_BR_MODE_VBR_5; break; case AACENC_BR_MODE_SFR: qcInit.bitrateMode = QCDATA_BR_MODE_SFR; break; case AACENC_BR_MODE_FF: qcInit.bitrateMode = QCDATA_BR_MODE_FF; break; default: ErrorStatus = AAC_ENC_UNSUPPORTED_BITRATE_MODE; goto bail; } qcInit.invQuant = (config->useRequant) ? 2 : 0; /* maxIterations should be set to the maximum number of requantization * iterations that are allowed before the crash recovery functionality is * activated. This setting should be adjusted to the processing power * available, i.e. to the processing power headroom in one frame that is still * left after normal encoding without requantization. Please note that if * activated this functionality is used most likely only in cases where the * encoder is operating beyond recommended settings, i.e. the audio quality is * suboptimal anyway. Activating the crash recovery does not further reduce * audio quality significantly in these cases. */ if (isLowDelay(config->audioObjectType)) { qcInit.maxIterations = 2; } else { qcInit.maxIterations = 5; } qcInit.bitrate = config->bitRate - config->ancDataBitRate; qcInit.staticBits = transportEnc_GetStaticBits( hTpEnc, qcInit.averageBits / qcInit.nSubFrames); ErrorStatus = FDKaacEnc_QCInit(hAacEnc->qcKernel, &qcInit, initFlags); if (ErrorStatus != AAC_ENC_OK) goto bail; /* Map virtual aot's to intern aot used in bitstream writer. */ switch (hAacEnc->config->audioObjectType) { case AOT_MP2_AAC_LC: hAacEnc->aot = AOT_AAC_LC; break; case AOT_MP2_SBR: hAacEnc->aot = AOT_SBR; break; default: hAacEnc->aot = hAacEnc->config->audioObjectType; } /* common things */ return AAC_ENC_OK; bail: return ErrorStatus; } /*--------------------------------------------------------------------------- functionname: FDKaacEnc_EncodeFrame description: encodes one frame returns: error code ---------------------------------------------------------------------------*/ AAC_ENCODER_ERROR FDKaacEnc_EncodeFrame( HANDLE_AAC_ENC hAacEnc, /* encoder handle */ HANDLE_TRANSPORTENC hTpEnc, INT_PCM *RESTRICT inputBuffer, const UINT inputBufferBufSize, INT *nOutBytes, AACENC_EXT_PAYLOAD extPayload[MAX_TOTAL_EXT_PAYLOADS]) { AAC_ENCODER_ERROR ErrorStatus; int el, n, c = 0; UCHAR extPayloadUsed[MAX_TOTAL_EXT_PAYLOADS]; CHANNEL_MAPPING *cm = &hAacEnc->channelMapping; PSY_OUT *psyOut = hAacEnc->psyOut[c]; QC_OUT *qcOut = hAacEnc->qcOut[c]; FDKmemclear(extPayloadUsed, MAX_TOTAL_EXT_PAYLOADS * sizeof(UCHAR)); qcOut->elementExtBits = 0; /* sum up all extended bit of each element */ qcOut->staticBits = 0; /* sum up side info bits of each element */ qcOut->totalNoRedPe = 0; /* sum up PE */ /* advance psychoacoustics */ for (el = 0; el < cm->nElements; el++) { ELEMENT_INFO elInfo = cm->elInfo[el]; if ((elInfo.elType == ID_SCE) || (elInfo.elType == ID_CPE) || (elInfo.elType == ID_LFE)) { int ch; /* update pointer!*/ for (ch = 0; ch < elInfo.nChannelsInEl; ch++) { PSY_OUT_CHANNEL *psyOutChan = psyOut->psyOutElement[el]->psyOutChannel[ch]; QC_OUT_CHANNEL *qcOutChan = qcOut->qcElement[el]->qcOutChannel[ch]; psyOutChan->mdctSpectrum = qcOutChan->mdctSpectrum; psyOutChan->sfbSpreadEnergy = qcOutChan->sfbSpreadEnergy; psyOutChan->sfbEnergy = qcOutChan->sfbEnergy; psyOutChan->sfbEnergyLdData = qcOutChan->sfbEnergyLdData; psyOutChan->sfbMinSnrLdData = qcOutChan->sfbMinSnrLdData; psyOutChan->sfbThresholdLdData = qcOutChan->sfbThresholdLdData; } ErrorStatus = FDKaacEnc_psyMain( elInfo.nChannelsInEl, hAacEnc->psyKernel->psyElement[el], hAacEnc->psyKernel->psyDynamic, hAacEnc->psyKernel->psyConf, psyOut->psyOutElement[el], inputBuffer, inputBufferBufSize, cm->elInfo[el].ChannelIndex, cm->nChannels); if (ErrorStatus != AAC_ENC_OK) return ErrorStatus; /* FormFactor, Pe and staticBitDemand calculation */ ErrorStatus = FDKaacEnc_QCMainPrepare( &elInfo, hAacEnc->qcKernel->hAdjThr->adjThrStateElem[el], psyOut->psyOutElement[el], qcOut->qcElement[el], hAacEnc->aot, hAacEnc->config->syntaxFlags, hAacEnc->config->epConfig); if (ErrorStatus != AAC_ENC_OK) return ErrorStatus; /*-------------------------------------------- */ qcOut->qcElement[el]->extBitsUsed = 0; qcOut->qcElement[el]->nExtensions = 0; /* reset extension payload */ FDKmemclear(&qcOut->qcElement[el]->extension, (1) * sizeof(QC_OUT_EXTENSION)); for (n = 0; n < MAX_TOTAL_EXT_PAYLOADS; n++) { if (!extPayloadUsed[n] && (extPayload[n].associatedChElement == el) && (extPayload[n].dataSize > 0) && (extPayload[n].pData != NULL)) { int idx = qcOut->qcElement[el]->nExtensions++; qcOut->qcElement[el]->extension[idx].type = extPayload[n].dataType; /* Perform a sanity check on the type? */ qcOut->qcElement[el]->extension[idx].nPayloadBits = extPayload[n].dataSize; qcOut->qcElement[el]->extension[idx].pPayload = extPayload[n].pData; /* Now ask the bitstream encoder how many bits we need to encode the * data with the current bitstream syntax: */ qcOut->qcElement[el]->extBitsUsed += FDKaacEnc_writeExtensionData( NULL, &qcOut->qcElement[el]->extension[idx], 0, 0, hAacEnc->config->syntaxFlags, hAacEnc->aot, hAacEnc->config->epConfig); extPayloadUsed[n] = 1; } } /* sum up extension and static bits for all channel elements */ qcOut->elementExtBits += qcOut->qcElement[el]->extBitsUsed; qcOut->staticBits += qcOut->qcElement[el]->staticBitsUsed; /* sum up pe */ qcOut->totalNoRedPe += qcOut->qcElement[el]->peData.pe; } } qcOut->nExtensions = 0; qcOut->globalExtBits = 0; /* reset extension payload */ FDKmemclear(&qcOut->extension, (2 + 2) * sizeof(QC_OUT_EXTENSION)); /* Add extension payload not assigned to an channel element (Ancillary data is the only supported type up to now) */ for (n = 0; n < MAX_TOTAL_EXT_PAYLOADS; n++) { if (!extPayloadUsed[n] && (extPayload[n].associatedChElement == -1) && (extPayload[n].pData != NULL)) { UINT payloadBits = 0; if (extPayload[n].dataType == EXT_DATA_ELEMENT) { if (hAacEnc->ancillaryBitsPerFrame) { /* granted frame dse bitrate */ payloadBits = hAacEnc->ancillaryBitsPerFrame; } else { /* write anc data if bitrate constraint fulfilled */ if ((extPayload[n].dataSize >> 3) <= hAacEnc->config->maxAncBytesPerAU) { payloadBits = extPayload[n].dataSize; } } payloadBits = fixMin(extPayload[n].dataSize, payloadBits); } else { payloadBits = extPayload[n].dataSize; } if (payloadBits > 0) { int idx = qcOut->nExtensions++; qcOut->extension[idx].type = extPayload[n].dataType; /* Perform a sanity check on the type? */ qcOut->extension[idx].nPayloadBits = payloadBits; qcOut->extension[idx].pPayload = extPayload[n].pData; /* Now ask the bitstream encoder how many bits we need to encode the * data with the current bitstream syntax: */ qcOut->globalExtBits += FDKaacEnc_writeExtensionData( NULL, &qcOut->extension[idx], 0, 0, hAacEnc->config->syntaxFlags, hAacEnc->aot, hAacEnc->config->epConfig); if (extPayload[n].dataType == EXT_DATA_ELEMENT) { /* substract the processed bits */ extPayload[n].dataSize -= payloadBits; } extPayloadUsed[n] = 1; } } } if (!(hAacEnc->config->syntaxFlags & (AC_SCALABLE | AC_ER))) { qcOut->globalExtBits += EL_ID_BITS; /* add bits for ID_END */ } /* build bitstream all nSubFrames */ { INT totalBits = 0; /* Total AU bits */ ; INT avgTotalBits = 0; /*-------------------------------------------- */ /* Get average total bits */ /*-------------------------------------------- */ { /* frame wise bitrate adaption */ FDKaacEnc_AdjustBitrate( hAacEnc->qcKernel, cm, &avgTotalBits, hAacEnc->config->bitRate, hAacEnc->config->sampleRate, hAacEnc->config->framelength); /* adjust super frame bitrate */ avgTotalBits *= hAacEnc->config->nSubFrames; } /* Make first estimate of transport header overhead. Take maximum possible frame size into account to prevent bitreservoir underrun. */ hAacEnc->qcKernel->globHdrBits = transportEnc_GetStaticBits( hTpEnc, avgTotalBits + hAacEnc->qcKernel->bitResTot); /*-------------------------------------------- */ /*-------------------------------------------- */ /*-------------------------------------------- */ ErrorStatus = FDKaacEnc_QCMain( hAacEnc->qcKernel, hAacEnc->psyOut, hAacEnc->qcOut, avgTotalBits, cm, hAacEnc->aot, hAacEnc->config->syntaxFlags, hAacEnc->config->epConfig); if (ErrorStatus != AAC_ENC_OK) return ErrorStatus; /*-------------------------------------------- */ /*-------------------------------------------- */ ErrorStatus = FDKaacEnc_updateFillBits( cm, hAacEnc->qcKernel, hAacEnc->qcKernel->elementBits, hAacEnc->qcOut); if (ErrorStatus != AAC_ENC_OK) return ErrorStatus; /*-------------------------------------------- */ ErrorStatus = FDKaacEnc_FinalizeBitConsumption( cm, hAacEnc->qcKernel, qcOut, qcOut->qcElement, hTpEnc, hAacEnc->aot, hAacEnc->config->syntaxFlags, hAacEnc->config->epConfig); if (ErrorStatus != AAC_ENC_OK) return ErrorStatus; /*-------------------------------------------- */ totalBits += qcOut->totalBits; /*-------------------------------------------- */ FDKaacEnc_updateBitres(cm, hAacEnc->qcKernel, hAacEnc->qcOut); /*-------------------------------------------- */ /* for ( all sub frames ) ... */ /* write bitstream header */ if (TRANSPORTENC_OK != transportEnc_WriteAccessUnit(hTpEnc, totalBits, FDKaacEnc_EncBitresToTpBitres(hAacEnc), cm->nChannelsEff)) { return AAC_ENC_UNKNOWN; } /* write bitstream */ ErrorStatus = FDKaacEnc_WriteBitstream( hTpEnc, cm, qcOut, psyOut, hAacEnc->qcKernel, hAacEnc->aot, hAacEnc->config->syntaxFlags, hAacEnc->config->epConfig); if (ErrorStatus != AAC_ENC_OK) return ErrorStatus; /* transportEnc_EndAccessUnit() is being called inside * FDKaacEnc_WriteBitstream() */ if (TRANSPORTENC_OK != transportEnc_GetFrame(hTpEnc, nOutBytes)) { return AAC_ENC_UNKNOWN; } } /* -end- if (curFrame==hAacEnc->qcKernel->nSubFrames) */ /*-------------------------------------------- */ return AAC_ENC_OK; } /*--------------------------------------------------------------------------- functionname:FDKaacEnc_Close description: delete encoder instance returns: ---------------------------------------------------------------------------*/ void FDKaacEnc_Close(HANDLE_AAC_ENC *phAacEnc) /* encoder handle */ { if (*phAacEnc == NULL) { return; } AAC_ENC *hAacEnc = (AAC_ENC *)*phAacEnc; if (hAacEnc->dynamic_RAM != NULL) FreeAACdynamic_RAM(&hAacEnc->dynamic_RAM); FDKaacEnc_PsyClose(&hAacEnc->psyKernel, hAacEnc->psyOut); FDKaacEnc_QCClose(&hAacEnc->qcKernel, hAacEnc->qcOut); FreeRam_aacEnc_AacEncoder(phAacEnc); } /* The following functions are in this source file only for convenience and */ /* need not be visible outside of a possible encoder library. */ /* basic defines for ancillary data */ #define MAX_ANCRATE 19200 /* ancillary rate >= 19200 isn't valid */ /*--------------------------------------------------------------------------- functionname: FDKaacEnc_InitCheckAncillary description: initialize and check ancillary data struct return: if success or NULL if error ---------------------------------------------------------------------------*/ static AAC_ENCODER_ERROR FDKaacEnc_InitCheckAncillary( INT bitRate, INT framelength, INT ancillaryRate, INT *ancillaryBitsPerFrame, INT sampleRate) { /* don't use negative ancillary rates */ if (ancillaryRate < -1) return AAC_ENC_UNSUPPORTED_ANC_BITRATE; /* check if ancillary rate is ok */ if ((ancillaryRate != (-1)) && (ancillaryRate != 0)) { /* ancRate <= 15% of bitrate && ancRate < 19200 */ if ((ancillaryRate >= MAX_ANCRATE) || ((ancillaryRate * 20) > (bitRate * 3))) { return AAC_ENC_UNSUPPORTED_ANC_BITRATE; } } else if (ancillaryRate == -1) { /* if no special ancRate is requested but a ancillary file is stated, then generate a ancillary rate matching to the bitrate */ if (bitRate >= (MAX_ANCRATE * 10)) { /* ancillary rate is 19199 */ ancillaryRate = (MAX_ANCRATE - 1); } else { /* 10% of bitrate */ ancillaryRate = bitRate / 10; } } /* make ancillaryBitsPerFrame byte align */ *ancillaryBitsPerFrame = FDKaacEnc_CalcBitsPerFrame(ancillaryRate, framelength, sampleRate) & ~0x7; return AAC_ENC_OK; } fdk-aac-2.0.2/libAACenc/src/aacenc.h000066400000000000000000000413441404225310700167200ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Schug / A. Groeschel Description: fast aac coder interface library functions *******************************************************************************/ #ifndef AACENC_H #define AACENC_H #include "common_fix.h" #include "FDK_audio.h" #include "tpenc_lib.h" #include "sbr_encoder.h" #define MIN_BUFSIZE_PER_EFF_CHAN 6144 #ifdef __cplusplus extern "C" { #endif /* * AAC-LC error codes. */ typedef enum { AAC_ENC_OK = 0x0000, /*!< All fine. */ AAC_ENC_UNKNOWN = 0x0002, /*!< Error condition is of unknown reason, or from another module. */ /* initialization errors */ aac_enc_init_error_start = 0x2000, AAC_ENC_INVALID_HANDLE = 0x2020, /*!< The handle passed to the function call was invalid (probably NULL). */ AAC_ENC_INVALID_FRAME_LENGTH = 0x2080, /*!< Invalid frame length (must be 1024 or 960). */ AAC_ENC_INVALID_N_CHANNELS = 0x20e0, /*!< Invalid amount of audio input channels. */ AAC_ENC_INVALID_SFB_TABLE = 0x2140, /*!< Internal encoder error. */ AAC_ENC_UNSUPPORTED_AOT = 0x3000, /*!< The Audio Object Type (AOT) is not supported. */ AAC_ENC_UNSUPPORTED_FILTERBANK = 0x3010, /*!< Filterbank type is not supported. */ AAC_ENC_UNSUPPORTED_BITRATE = 0x3020, /*!< The chosen bitrate is not supported. */ AAC_ENC_UNSUPPORTED_BITRATE_MODE = 0x3028, /*!< Unsupported bit rate mode (CBR or VBR). */ AAC_ENC_UNSUPPORTED_ANC_BITRATE = 0x3040, /*!< Unsupported ancillay bitrate. */ AAC_ENC_UNSUPPORTED_ANC_MODE = 0x3060, AAC_ENC_UNSUPPORTED_TRANSPORT_TYPE = 0x3080, /*!< The bitstream format is not supported. */ AAC_ENC_UNSUPPORTED_ER_FORMAT = 0x30a0, /*!< The error resilience tool format is not supported. */ AAC_ENC_UNSUPPORTED_EPCONFIG = 0x30c0, /*!< The error protection format is not supported. */ AAC_ENC_UNSUPPORTED_CHANNELCONFIG = 0x30e0, /*!< The channel configuration (either number or arrangement) is not supported. */ AAC_ENC_UNSUPPORTED_SAMPLINGRATE = 0x3100, /*!< Sample rate of audio input is not supported. */ AAC_ENC_NO_MEMORY = 0x3120, /*!< Could not allocate memory. */ AAC_ENC_PE_INIT_TABLE_NOT_FOUND = 0x3140, /*!< Internal encoder error. */ aac_enc_init_error_end, /* encode errors */ aac_enc_error_start = 0x4000, AAC_ENC_QUANT_ERROR = 0x4020, /*!< Too many bits used in quantization. */ AAC_ENC_WRITTEN_BITS_ERROR = 0x4040, /*!< Unexpected number of written bits, differs to calculated number of bits. */ AAC_ENC_PNS_TABLE_ERROR = 0x4060, /*!< PNS level out of range. */ AAC_ENC_GLOBAL_GAIN_TOO_HIGH = 0x4080, /*!< Internal quantizer error. */ AAC_ENC_BITRES_TOO_LOW = 0x40a0, /*!< Too few bits in bit reservoir. */ AAC_ENC_BITRES_TOO_HIGH = 0x40a1, /*!< Too many bits in bit reservoir. */ AAC_ENC_INVALID_CHANNEL_BITRATE = 0x4100, AAC_ENC_INVALID_ELEMENTINFO_TYPE = 0x4120, /*!< Internal encoder error. */ AAC_ENC_WRITE_SCAL_ERROR = 0x41e0, /*!< Error writing scalefacData. */ AAC_ENC_WRITE_SEC_ERROR = 0x4200, /*!< Error writing sectionData. */ AAC_ENC_WRITE_SPEC_ERROR = 0x4220, /*!< Error writing spectralData. */ aac_enc_error_end } AAC_ENCODER_ERROR; /*-------------------------- defines --------------------------------------*/ #define ANC_DATA_BUFFERSIZE 1024 /* ancBuffer size */ #define MAX_TOTAL_EXT_PAYLOADS ((((8)) * (1)) + (2 + 2)) typedef enum { AACENC_BR_MODE_INVALID = -1, /*!< Invalid bitrate mode. */ AACENC_BR_MODE_CBR = 0, /*!< Constant bitrate mode. */ AACENC_BR_MODE_VBR_1 = 1, /*!< Variable bitrate mode, very low bitrate. */ AACENC_BR_MODE_VBR_2 = 2, /*!< Variable bitrate mode, low bitrate. */ AACENC_BR_MODE_VBR_3 = 3, /*!< Variable bitrate mode, medium bitrate. */ AACENC_BR_MODE_VBR_4 = 4, /*!< Variable bitrate mode, high bitrate. */ AACENC_BR_MODE_VBR_5 = 5, /*!< Variable bitrate mode, very high bitrate. */ AACENC_BR_MODE_FF = 6, /*!< Fixed frame mode. */ AACENC_BR_MODE_SFR = 7 /*!< Superframe mode. */ } AACENC_BITRATE_MODE; #define AACENC_BR_MODE_IS_VBR(brMode) ((brMode >= 1) && (brMode <= 5)) typedef enum { CH_ORDER_MPEG = 0, /*!< MPEG channel ordering (e. g. 5.1: C, L, R, SL, SR, LFE) */ CH_ORDER_WAV, /*!< WAV fileformat channel ordering (e. g. 5.1: L, R, C, LFE, SL, SR) */ CH_ORDER_WG4 /*!< WG4 fileformat channel ordering (e. g. 5.1: L, R, SL, SR, C, LFE) */ } CHANNEL_ORDER; /*-------------------- structure definitions ------------------------------*/ struct AACENC_CONFIG { INT sampleRate; /* encoder sample rate */ INT bitRate; /* encoder bit rate in bits/sec */ INT ancDataBitRate; /* additional bits consumed by anc data or sbr have to be consiedered while configuration */ INT nSubFrames; /* number of frames in super frame (not ADTS/LATM subframes !) */ AUDIO_OBJECT_TYPE audioObjectType; /* Audio Object Type */ INT averageBits; /* encoder bit rate in bits/superframe */ AACENC_BITRATE_MODE bitrateMode; /* encoder bitrate mode (CBR/VBR) */ INT nChannels; /* number of channels to process */ CHANNEL_ORDER channelOrder; /* input Channel ordering scheme. */ INT bandWidth; /* targeted audio bandwidth in Hz */ CHANNEL_MODE channelMode; /* encoder channel mode configuration */ INT framelength; /* used frame size */ UINT syntaxFlags; /* bitstreams syntax configuration */ SCHAR epConfig; /* error protection configuration */ INT anc_Rate; /* ancillary rate, 0 (disabled), -1 (default) else desired rate */ UINT maxAncBytesPerAU; INT minBitsPerFrame; /* minimum number of bits in AU */ INT maxBitsPerFrame; /* maximum number of bits in AU */ INT audioMuxVersion; /* audio mux version in loas/latm transport format */ UINT sbrRatio; /* sbr sampling rate ratio: dual- or single-rate */ UCHAR useTns; /* flag: use temporal noise shaping */ UCHAR usePns; /* flag: use perceptual noise substitution */ UCHAR useIS; /* flag: use intensity coding */ UCHAR useMS; /* flag: use ms stereo tool */ UCHAR useRequant; /* flag: use afterburner */ UINT downscaleFactor; }; typedef struct { UCHAR *pData; /* pointer to extension payload data */ UINT dataSize; /* extension payload data size in bits */ EXT_PAYLOAD_TYPE dataType; /* extension payload data type */ INT associatedChElement; /* number of the channel element the data is assigned to */ } AACENC_EXT_PAYLOAD; typedef struct AAC_ENC *HANDLE_AAC_ENC; /** * \brief Calculate framesize in bits for given bit rate, frame length and * sampling rate. * * \param bitRate Ttarget bitrate in bits per second. * \param frameLength Number of audio samples in one frame. * \param samplingRate Sampling rate in Hz. * * \return Framesize in bits per frame. */ INT FDKaacEnc_CalcBitsPerFrame(const INT bitRate, const INT frameLength, const INT samplingRate); /** * \brief Calculate bitrate in bits per second for given framesize, frame length * and sampling rate. * * \param bitsPerFrame Framesize in bits per frame * \param frameLength Number of audio samples in one frame. * \param samplingRate Sampling rate in Hz. * * \return Bitrate in bits per second. */ INT FDKaacEnc_CalcBitrate(const INT bitsPerFrame, const INT frameLength, const INT samplingRate); /** * \brief Limit given bit rate to a valid value * \param hTpEnc transport encoder handle * \param aot audio object type * \param coreSamplingRate the sample rate to be used for the AAC encoder * \param frameLength the frameLength to be used for the AAC encoder * \param nChannels number of total channels * \param nChannelsEff number of effective channels * \param bitRate the initial bit rate value for which the closest valid bit * rate value is searched for * \param averageBits average bits per frame for fixed framing. Set to -1 if not * available. * \param optional pointer where the current bits per frame are stored into. * \param bitrateMode the current bit rate mode * \param nSubFrames number of sub frames for super framing (not transport * frames). * \return a valid bit rate value as close as possible or identical to bitRate */ INT FDKaacEnc_LimitBitrate(HANDLE_TRANSPORTENC hTpEnc, AUDIO_OBJECT_TYPE aot, INT coreSamplingRate, INT frameLength, INT nChannels, INT nChannelsEff, INT bitRate, INT averageBits, INT *pAverageBitsPerFrame, AACENC_BITRATE_MODE bitrateMode, INT nSubFrames); /** * \brief Get current state of the bit reservoir * \param hAacEncoder encoder handle * \return bit reservoir state in bits */ INT FDKaacEnc_GetBitReservoirState(const HANDLE_AAC_ENC hAacEncoder); /*----------------------------------------------------------------------------- functionname: FDKaacEnc_GetVBRBitrate description: Get VBR bitrate from vbr quality input params: int vbrQuality (VBR0, VBR1, VBR2) channelMode returns: vbr bitrate ------------------------------------------------------------------------------*/ INT FDKaacEnc_GetVBRBitrate(AACENC_BITRATE_MODE bitrateMode, CHANNEL_MODE channelMode); /*----------------------------------------------------------------------------- functionname: FDKaacEnc_AdjustVBRBitrateMode description: Adjust bitrate mode to given bitrate parameter input params: int vbrQuality (VBR0, VBR1, VBR2) bitrate channelMode returns: vbr bitrate mode ------------------------------------------------------------------------------*/ AACENC_BITRATE_MODE FDKaacEnc_AdjustVBRBitrateMode( AACENC_BITRATE_MODE bitrateMode, INT bitrate, CHANNEL_MODE channelMode); /*----------------------------------------------------------------------------- functionname: FDKaacEnc_AacInitDefaultConfig description: gives reasonable default configuration returns: --- ------------------------------------------------------------------------------*/ void FDKaacEnc_AacInitDefaultConfig(AACENC_CONFIG *config); /*--------------------------------------------------------------------------- functionname:FDKaacEnc_Open description: allocate and initialize a new encoder instance returns: 0 if success ---------------------------------------------------------------------------*/ AAC_ENCODER_ERROR FDKaacEnc_Open( HANDLE_AAC_ENC *phAacEnc, /* pointer to an encoder handle, initialized on return */ const INT nElements, /* number of maximal elements in instance to support */ const INT nChannels, /* number of maximal channels in instance to support */ const INT nSubFrames); /* support superframing in instance */ AAC_ENCODER_ERROR FDKaacEnc_Initialize( HANDLE_AAC_ENC hAacEncoder, /* pointer to an encoder handle, initialized on return */ AACENC_CONFIG *config, /* pre-initialized config struct */ HANDLE_TRANSPORTENC hTpEnc, ULONG initFlags); /*--------------------------------------------------------------------------- functionname: FDKaacEnc_EncodeFrame description: encode one frame returns: 0 if success ---------------------------------------------------------------------------*/ AAC_ENCODER_ERROR FDKaacEnc_EncodeFrame( HANDLE_AAC_ENC hAacEnc, /* encoder handle */ HANDLE_TRANSPORTENC hTpEnc, INT_PCM *inputBuffer, const UINT inputBufferBufSize, INT *numOutBytes, AACENC_EXT_PAYLOAD extPayload[MAX_TOTAL_EXT_PAYLOADS]); /*--------------------------------------------------------------------------- functionname:FDKaacEnc_Close description: delete encoder instance returns: ---------------------------------------------------------------------------*/ void FDKaacEnc_Close(HANDLE_AAC_ENC *phAacEnc); /* encoder handle */ #ifdef __cplusplus } #endif #endif /* AACENC_H */ fdk-aac-2.0.2/libAACenc/src/aacenc_lib.cpp000066400000000000000000002543661404225310700201130ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Lohwasser Description: FDK HE-AAC Encoder interface library functions *******************************************************************************/ #include "aacenc_lib.h" #include "FDK_audio.h" #include "aacenc.h" #include "aacEnc_ram.h" #include "FDK_core.h" /* FDK_tools versioning info */ /* Encoder library info */ #define AACENCODER_LIB_VL0 4 #define AACENCODER_LIB_VL1 0 #define AACENCODER_LIB_VL2 1 #define AACENCODER_LIB_TITLE "AAC Encoder" #ifdef SUPPRESS_BUILD_DATE_INFO #define AACENCODER_LIB_BUILD_DATE "" #define AACENCODER_LIB_BUILD_TIME "" #else #define AACENCODER_LIB_BUILD_DATE __DATE__ #define AACENCODER_LIB_BUILD_TIME __TIME__ #endif #include "pcm_utils.h" #include "sbr_encoder.h" #include "../src/sbrenc_ram.h" #include "channel_map.h" #include "psy_const.h" #include "bitenc.h" #include "tpenc_lib.h" #include "metadata_main.h" #include "mps_main.h" #include "sacenc_lib.h" #define SBL(fl) \ (fl / \ 8) /*!< Short block length (hardcoded to 8 short blocks per long block) */ #define BSLA(fl) \ (4 * SBL(fl) + SBL(fl) / 2) /*!< AAC block switching look-ahead */ #define DELAY_AAC(fl) (fl + BSLA(fl)) /*!< MDCT + blockswitching */ #define DELAY_AACLD(fl) (fl) /*!< MDCT delay (no framing delay included) */ #define DELAY_AACELD(fl) \ ((fl) / 2) /*!< ELD FB delay (no framing delay included) */ #define MAX_DS_DELAY (100) /*!< Maximum downsampler delay in SBR. */ #define INPUTBUFFER_SIZE \ (2 * (1024) + MAX_DS_DELAY + 1537) /*!< Audio input samples + downsampler \ delay + sbr/aac delay compensation */ #define DEFAULT_HEADER_PERIOD_REPETITION_RATE \ 10 /*!< Default header repetition rate used in transport library and for SBR \ header. */ //////////////////////////////////////////////////////////////////////////////////// /** * Flags to characterize encoder modules to be supported in present instance. */ enum { ENC_MODE_FLAG_AAC = 0x0001, ENC_MODE_FLAG_SBR = 0x0002, ENC_MODE_FLAG_PS = 0x0004, ENC_MODE_FLAG_SAC = 0x0008, ENC_MODE_FLAG_META = 0x0010 }; //////////////////////////////////////////////////////////////////////////////////// typedef struct { AUDIO_OBJECT_TYPE userAOT; /*!< Audio Object Type. */ UINT userSamplerate; /*!< Sampling frequency. */ UINT nChannels; /*!< will be set via channelMode. */ CHANNEL_MODE userChannelMode; UINT userBitrate; UINT userBitrateMode; UINT userBandwidth; UINT userAfterburner; UINT userFramelength; UINT userAncDataRate; UINT userPeakBitrate; UCHAR userTns; /*!< Use TNS coding. */ UCHAR userPns; /*!< Use PNS coding. */ UCHAR userIntensity; /*!< Use Intensity coding. */ TRANSPORT_TYPE userTpType; /*!< Transport type */ UCHAR userTpSignaling; /*!< Extension AOT signaling mode. */ UCHAR userTpNsubFrames; /*!< Number of sub frames in a transport frame for LOAS/LATM or ADTS (default 1). */ UCHAR userTpAmxv; /*!< AudioMuxVersion to be used for LATM (default 0). */ UCHAR userTpProtection; UCHAR userTpHeaderPeriod; /*!< Parameter used to configure LATM/LOAS SMC rate. Moreover this parameters is used to configure repetition rate of PCE in raw_data_block. */ UCHAR userErTools; /*!< Use VCB11, HCR and/or RVLC ER tool. */ UINT userPceAdditions; /*!< Configure additional bits in PCE. */ UCHAR userMetaDataMode; /*!< Meta data library configuration. */ UCHAR userSbrEnabled; /*!< Enable SBR for ELD. */ UINT userSbrRatio; /*!< SBR sampling rate ratio. Dual- or single-rate. */ UINT userDownscaleFactor; } USER_PARAM; /** * SBR extenxion payload struct provides buffers to be filled in SBR encoder * library. */ typedef struct { UCHAR data[(1)][(8)][MAX_PAYLOAD_SIZE]; /*!< extension payload data buffer */ UINT dataSize[(1)][(8)]; /*!< extension payload data size in bits */ } SBRENC_EXT_PAYLOAD; //////////////////////////////////////////////////////////////////////////////////// /**************************************************************************** Structure Definitions ****************************************************************************/ typedef struct AACENC_CONFIG *HANDLE_AACENC_CONFIG; struct AACENCODER { USER_PARAM extParam; CODER_CONFIG coderConfig; /* AAC */ AACENC_CONFIG aacConfig; HANDLE_AAC_ENC hAacEnc; /* SBR */ HANDLE_SBR_ENCODER hEnvEnc; /* SBR encoder */ SBRENC_EXT_PAYLOAD *pSbrPayload; /* SBR extension payload */ /* Meta Data */ HANDLE_FDK_METADATA_ENCODER hMetadataEnc; INT metaDataAllowed; /* Signal whether chosen configuration allows metadata. Necessary for delay compensation. Metadata mode is a separate parameter. */ HANDLE_MPS_ENCODER hMpsEnc; /* Transport */ HANDLE_TRANSPORTENC hTpEnc; INT_PCM *inputBuffer; /* Internal input buffer. Input source for AAC encoder */ UCHAR *outBuffer; /* Internal bitstream buffer */ INT inputBufferSize; /* Size of internal input buffer */ INT inputBufferSizePerChannel; /* Size of internal input buffer per channel */ INT outBufferInBytes; /* Size of internal bitstream buffer*/ INT inputBufferOffset; /* Where to write new input samples. */ INT nSamplesToRead; /* number of input samples neeeded for encoding one frame */ INT nSamplesRead; /* number of input samples already in input buffer */ INT nZerosAppended; /* appended zeros at end of file*/ INT nDelay; /* codec delay */ INT nDelayCore; /* codec delay, w/o the SBR decoder delay */ AACENC_EXT_PAYLOAD extPayload[MAX_TOTAL_EXT_PAYLOADS]; ULONG InitFlags; /* internal status to treggier re-initialization */ /* Memory allocation info. */ INT nMaxAacElements; INT nMaxAacChannels; INT nMaxSbrElements; INT nMaxSbrChannels; UINT encoder_modis; /* Capability flags */ UINT CAPF_tpEnc; }; typedef struct { /* input */ ULONG nChannels; /*!< Number of audio channels. */ ULONG samplingRate; /*!< Encoder output sampling rate. */ ULONG bitrateRange; /*!< Lower bitrate range for config entry. */ /* output*/ UCHAR sbrMode; /*!< 0: ELD sbr off, 1: ELD with downsampled sbr, 2: ELD with dualrate sbr. */ CHANNEL_MODE chMode; /*!< Channel mode. */ } ELD_SBR_CONFIGURATOR; /** * \brief This table defines ELD/SBR default configurations. */ static const ELD_SBR_CONFIGURATOR eldSbrAutoConfigTab[] = { {1, 48000, 0, 2, MODE_1}, {1, 48000, 64000, 0, MODE_1}, {1, 44100, 0, 2, MODE_1}, {1, 44100, 64000, 0, MODE_1}, {1, 32000, 0, 2, MODE_1}, {1, 32000, 28000, 1, MODE_1}, {1, 32000, 56000, 0, MODE_1}, {1, 24000, 0, 1, MODE_1}, {1, 24000, 40000, 0, MODE_1}, {1, 16000, 0, 1, MODE_1}, {1, 16000, 28000, 0, MODE_1}, {1, 15999, 0, 0, MODE_1}, {2, 48000, 0, 2, MODE_2}, {2, 48000, 44000, 2, MODE_2}, {2, 48000, 128000, 0, MODE_2}, {2, 44100, 0, 2, MODE_2}, {2, 44100, 44000, 2, MODE_2}, {2, 44100, 128000, 0, MODE_2}, {2, 32000, 0, 2, MODE_2}, {2, 32000, 32000, 2, MODE_2}, {2, 32000, 68000, 1, MODE_2}, {2, 32000, 96000, 0, MODE_2}, {2, 24000, 0, 1, MODE_2}, {2, 24000, 48000, 1, MODE_2}, {2, 24000, 80000, 0, MODE_2}, {2, 16000, 0, 1, MODE_2}, {2, 16000, 32000, 1, MODE_2}, {2, 16000, 64000, 0, MODE_2}, {2, 15999, 0, 0, MODE_2} }; /* * \brief Configure SBR for ELD configuration. * * This function finds default SBR configuration for ELD based on number of * channels, sampling rate and bitrate. * * \param nChannels Number of audio channels. * \param samplingRate Audio signal sampling rate. * \param bitrate Encoder bitrate. * * \return - pointer to eld sbr configuration. * - NULL, on failure. */ static const ELD_SBR_CONFIGURATOR *eldSbrConfigurator(const ULONG nChannels, const ULONG samplingRate, const ULONG bitrate) { int i; const ELD_SBR_CONFIGURATOR *pSetup = NULL; for (i = 0; i < (int)(sizeof(eldSbrAutoConfigTab) / sizeof(ELD_SBR_CONFIGURATOR)); i++) { if ((nChannels == eldSbrAutoConfigTab[i].nChannels) && (samplingRate <= eldSbrAutoConfigTab[i].samplingRate) && (bitrate >= eldSbrAutoConfigTab[i].bitrateRange)) { pSetup = &eldSbrAutoConfigTab[i]; } } return pSetup; } static inline INT isSbrActive(const HANDLE_AACENC_CONFIG hAacConfig) { INT sbrUsed = 0; /* Note: Even if implicit signalling was selected, The AOT itself here is not * AOT_AAC_LC */ if ((hAacConfig->audioObjectType == AOT_SBR) || (hAacConfig->audioObjectType == AOT_PS) || (hAacConfig->audioObjectType == AOT_MP2_SBR)) { sbrUsed = 1; } if (hAacConfig->audioObjectType == AOT_ER_AAC_ELD && (hAacConfig->syntaxFlags & AC_SBR_PRESENT)) { sbrUsed = 1; } return (sbrUsed); } static inline INT isPsActive(const AUDIO_OBJECT_TYPE audioObjectType) { INT psUsed = 0; if (audioObjectType == AOT_PS) { psUsed = 1; } return (psUsed); } static CHANNEL_MODE GetCoreChannelMode( const CHANNEL_MODE channelMode, const AUDIO_OBJECT_TYPE audioObjectType) { CHANNEL_MODE mappedChannelMode = channelMode; if ((isPsActive(audioObjectType) && (channelMode == MODE_2)) || (channelMode == MODE_212)) { mappedChannelMode = MODE_1; } return mappedChannelMode; } static SBR_PS_SIGNALING getSbrSignalingMode( const AUDIO_OBJECT_TYPE audioObjectType, const TRANSPORT_TYPE transportType, const UCHAR transportSignaling, const UINT sbrRatio) { SBR_PS_SIGNALING sbrSignaling; if (transportType == TT_UNKNOWN || sbrRatio == 0) { sbrSignaling = SIG_UNKNOWN; /* Needed parameters have not been set */ return sbrSignaling; } else { sbrSignaling = SIG_EXPLICIT_HIERARCHICAL; /* default: explicit hierarchical signaling */ } if ((audioObjectType == AOT_AAC_LC) || (audioObjectType == AOT_SBR) || (audioObjectType == AOT_PS) || (audioObjectType == AOT_MP2_AAC_LC) || (audioObjectType == AOT_MP2_SBR)) { switch (transportType) { case TT_MP4_ADIF: case TT_MP4_ADTS: sbrSignaling = SIG_IMPLICIT; /* For MPEG-2 transport types, only implicit signaling is possible */ break; case TT_MP4_RAW: case TT_MP4_LATM_MCP1: case TT_MP4_LATM_MCP0: case TT_MP4_LOAS: default: if (transportSignaling == 0xFF) { /* Defaults */ sbrSignaling = SIG_EXPLICIT_HIERARCHICAL; } else { /* User set parameters */ /* Attention: Backward compatible explicit signaling does only work * with AMV1 for LATM/LOAS */ sbrSignaling = (SBR_PS_SIGNALING)transportSignaling; } break; } } return sbrSignaling; } static inline INT getAssociatedChElement(SBR_ELEMENT_INFO *elInfoSbr, CHANNEL_MAPPING *channelMapping) { ELEMENT_INFO *elInfo = channelMapping->elInfo; INT nElements = channelMapping->nElements; INT associatedChElement = -1; int i; for (i = 0; i < nElements; i++) { if (elInfoSbr->elType == elInfo[i].elType && elInfoSbr->instanceTag == elInfo[i].instanceTag) { associatedChElement = i; break; } } return associatedChElement; } /**************************************************************************** Allocate Encoder ****************************************************************************/ H_ALLOC_MEM(_AacEncoder, AACENCODER) C_ALLOC_MEM(_AacEncoder, struct AACENCODER, 1) /* * Map Encoder specific config structures to CODER_CONFIG. */ static void FDKaacEnc_MapConfig(CODER_CONFIG *const cc, const USER_PARAM *const extCfg, const SBR_PS_SIGNALING sbrSignaling, const HANDLE_AACENC_CONFIG hAacConfig) { AUDIO_OBJECT_TYPE transport_AOT = AOT_NULL_OBJECT; FDKmemclear(cc, sizeof(CODER_CONFIG)); cc->flags = 0; cc->samplesPerFrame = hAacConfig->framelength; cc->samplingRate = hAacConfig->sampleRate; cc->extSamplingRate = extCfg->userSamplerate; /* Map virtual aot to transport aot. */ switch (hAacConfig->audioObjectType) { case AOT_MP2_AAC_LC: transport_AOT = AOT_AAC_LC; break; case AOT_MP2_SBR: transport_AOT = AOT_SBR; cc->flags |= CC_SBR; break; default: transport_AOT = hAacConfig->audioObjectType; } if (hAacConfig->audioObjectType == AOT_ER_AAC_ELD) { cc->flags |= (hAacConfig->syntaxFlags & AC_SBR_PRESENT) ? CC_SBR : 0; cc->flags |= (hAacConfig->syntaxFlags & AC_LD_MPS) ? CC_SAC : 0; } /* transport type is usually AAC-LC. */ if ((transport_AOT == AOT_SBR) || (transport_AOT == AOT_PS)) { cc->aot = AOT_AAC_LC; } else { cc->aot = transport_AOT; } /* Configure extension aot. */ if (sbrSignaling == SIG_IMPLICIT) { cc->extAOT = AOT_NULL_OBJECT; /* implicit */ } else { if ((sbrSignaling == SIG_EXPLICIT_BW_COMPATIBLE) && ((transport_AOT == AOT_SBR) || (transport_AOT == AOT_PS))) { cc->extAOT = AOT_SBR; /* explicit backward compatible */ } else { cc->extAOT = transport_AOT; /* explicit hierarchical */ } } if ((transport_AOT == AOT_SBR) || (transport_AOT == AOT_PS)) { cc->sbrPresent = 1; if (transport_AOT == AOT_PS) { cc->psPresent = 1; } } cc->sbrSignaling = sbrSignaling; if (hAacConfig->downscaleFactor > 1) { cc->downscaleSamplingRate = cc->samplingRate; cc->samplingRate *= hAacConfig->downscaleFactor; cc->extSamplingRate *= hAacConfig->downscaleFactor; } cc->bitRate = hAacConfig->bitRate; cc->noChannels = hAacConfig->nChannels; cc->flags |= CC_IS_BASELAYER; cc->channelMode = hAacConfig->channelMode; cc->nSubFrames = (hAacConfig->nSubFrames > 1 && extCfg->userTpNsubFrames == 1) ? hAacConfig->nSubFrames : extCfg->userTpNsubFrames; cc->flags |= (extCfg->userTpProtection) ? CC_PROTECTION : 0; if (extCfg->userTpHeaderPeriod != 0xFF) { cc->headerPeriod = extCfg->userTpHeaderPeriod; } else { /* auto-mode */ switch (extCfg->userTpType) { case TT_MP4_ADTS: case TT_MP4_LOAS: case TT_MP4_LATM_MCP1: cc->headerPeriod = DEFAULT_HEADER_PERIOD_REPETITION_RATE; break; default: cc->headerPeriod = 0; } } /* Mpeg-4 signaling for transport library. */ switch (hAacConfig->audioObjectType) { case AOT_MP2_AAC_LC: case AOT_MP2_SBR: cc->flags &= ~CC_MPEG_ID; /* Required for ADTS. */ cc->extAOT = AOT_NULL_OBJECT; break; default: cc->flags |= CC_MPEG_ID; } /* ER-tools signaling. */ cc->flags |= (hAacConfig->syntaxFlags & AC_ER_VCB11) ? CC_VCB11 : 0; cc->flags |= (hAacConfig->syntaxFlags & AC_ER_HCR) ? CC_HCR : 0; cc->flags |= (hAacConfig->syntaxFlags & AC_ER_RVLC) ? CC_RVLC : 0; /* Matrix mixdown coefficient configuration. */ if ((extCfg->userPceAdditions & 0x1) && (hAacConfig->epConfig == -1) && ((cc->channelMode == MODE_1_2_2) || (cc->channelMode == MODE_1_2_2_1))) { cc->matrixMixdownA = ((extCfg->userPceAdditions >> 1) & 0x3) + 1; cc->flags |= (extCfg->userPceAdditions >> 3) & 0x1 ? CC_PSEUDO_SURROUND : 0; } else { cc->matrixMixdownA = 0; } cc->channelConfigZero = 0; } /* * Validate prefilled pointers within buffer descriptor. * * \param pBufDesc Pointer to buffer descriptor * \return - AACENC_OK, all fine. * - AACENC_INVALID_HANDLE, on missing pointer initializiation. * - AACENC_UNSUPPORTED_PARAMETER, on incorrect buffer descriptor initialization. */ static AACENC_ERROR validateBufDesc(const AACENC_BufDesc *pBufDesc) { AACENC_ERROR err = AACENC_OK; if (pBufDesc != NULL) { int i; if ((pBufDesc->bufferIdentifiers == NULL) || (pBufDesc->bufSizes == NULL) || (pBufDesc->bufElSizes == NULL) || (pBufDesc->bufs == NULL)) { err = AACENC_UNSUPPORTED_PARAMETER; goto bail; } for (i = 0; i < pBufDesc->numBufs; i++) { if (pBufDesc->bufs[i] == NULL) { err = AACENC_UNSUPPORTED_PARAMETER; goto bail; } } } else { err = AACENC_INVALID_HANDLE; } bail: return err; } /* * Examine buffer descriptor regarding choosen identifier. * * \param pBufDesc Pointer to buffer descriptor * \param identifier Buffer identifier to look for. * \return - Buffer descriptor index. * -1, if there is no entry available. */ static INT getBufDescIdx(const AACENC_BufDesc *pBufDesc, const AACENC_BufferIdentifier identifier) { INT i, idx = -1; if (pBufDesc != NULL) { for (i = 0; i < pBufDesc->numBufs; i++) { if ((AACENC_BufferIdentifier)pBufDesc->bufferIdentifiers[i] == identifier) { idx = i; break; } } } return idx; } /**************************************************************************** Function Declarations ****************************************************************************/ AAC_ENCODER_ERROR aacEncDefaultConfig(HANDLE_AACENC_CONFIG hAacConfig, USER_PARAM *config) { /* make reasonable default settings */ FDKaacEnc_AacInitDefaultConfig(hAacConfig); /* clear configuration structure and copy default settings */ FDKmemclear(config, sizeof(USER_PARAM)); /* copy encoder configuration settings */ config->nChannels = hAacConfig->nChannels; config->userAOT = hAacConfig->audioObjectType = AOT_AAC_LC; config->userSamplerate = hAacConfig->sampleRate; config->userChannelMode = hAacConfig->channelMode; config->userBitrate = hAacConfig->bitRate; config->userBitrateMode = hAacConfig->bitrateMode; config->userPeakBitrate = (UINT)-1; config->userBandwidth = hAacConfig->bandWidth; config->userTns = hAacConfig->useTns; config->userPns = hAacConfig->usePns; config->userIntensity = hAacConfig->useIS; config->userAfterburner = hAacConfig->useRequant; config->userFramelength = (UINT)-1; config->userDownscaleFactor = 1; /* initialize transport parameters */ config->userTpType = TT_UNKNOWN; config->userTpAmxv = 0; config->userTpSignaling = 0xFF; /* choose signaling automatically */ config->userTpNsubFrames = 1; config->userTpProtection = 0; /* not crc protected*/ config->userTpHeaderPeriod = 0xFF; /* header period in auto mode */ config->userPceAdditions = 0; /* no matrix mixdown coefficient */ config->userMetaDataMode = 0; /* do not embed any meta data info */ config->userAncDataRate = 0; /* SBR rate is set to 0 here, which means it should be set automatically in FDKaacEnc_AdjustEncSettings() if the user did not set a rate expilicitely. */ config->userSbrRatio = 0; /* SBR enable set to -1 means to inquire ELD audio configurator for reasonable * configuration. */ config->userSbrEnabled = (UCHAR)-1; return AAC_ENC_OK; } static void aacEncDistributeSbrBits(CHANNEL_MAPPING *channelMapping, SBR_ELEMENT_INFO *sbrElInfo, INT bitRate) { INT codebits = bitRate; int el; /* Copy Element info */ for (el = 0; el < channelMapping->nElements; el++) { sbrElInfo[el].ChannelIndex[0] = channelMapping->elInfo[el].ChannelIndex[0]; sbrElInfo[el].ChannelIndex[1] = channelMapping->elInfo[el].ChannelIndex[1]; sbrElInfo[el].elType = channelMapping->elInfo[el].elType; sbrElInfo[el].bitRate = fMultIfloor(channelMapping->elInfo[el].relativeBits, bitRate); sbrElInfo[el].instanceTag = channelMapping->elInfo[el].instanceTag; sbrElInfo[el].nChannelsInEl = channelMapping->elInfo[el].nChannelsInEl; sbrElInfo[el].fParametricStereo = 0; sbrElInfo[el].fDualMono = 0; codebits -= sbrElInfo[el].bitRate; } sbrElInfo[0].bitRate += codebits; } static INT aacEncoder_LimitBitrate(const HANDLE_TRANSPORTENC hTpEnc, const INT samplingRate, const INT frameLength, const INT nChannels, const CHANNEL_MODE channelMode, INT bitRate, const INT nSubFrames, const INT sbrActive, const INT sbrDownSampleRate, const UINT syntaxFlags, const AUDIO_OBJECT_TYPE aot) { INT coreSamplingRate; CHANNEL_MAPPING cm; FDKaacEnc_InitChannelMapping(channelMode, CH_ORDER_MPEG, &cm); if (sbrActive) { coreSamplingRate = samplingRate >> (sbrEncoder_IsSingleRatePossible(aot) ? (sbrDownSampleRate - 1) : 1); } else { coreSamplingRate = samplingRate; } /* Limit bit rate in respect to the core coder */ bitRate = FDKaacEnc_LimitBitrate(hTpEnc, aot, coreSamplingRate, frameLength, nChannels, cm.nChannelsEff, bitRate, -1, NULL, AACENC_BR_MODE_INVALID, nSubFrames); /* Limit bit rate in respect to available SBR modes if active */ if (sbrActive) { int numIterations = 0; INT initialBitrate, adjustedBitrate; adjustedBitrate = bitRate; /* Find total bitrate which provides valid configuration for each SBR * element. */ do { int e; SBR_ELEMENT_INFO sbrElInfo[((8))]; FDK_ASSERT(cm.nElements <= ((8))); initialBitrate = adjustedBitrate; /* Get bit rate for each SBR element */ aacEncDistributeSbrBits(&cm, sbrElInfo, initialBitrate); for (e = 0; e < cm.nElements; e++) { INT sbrElementBitRateIn, sbrBitRateOut; if (cm.elInfo[e].elType != ID_SCE && cm.elInfo[e].elType != ID_CPE) { continue; } sbrElementBitRateIn = sbrElInfo[e].bitRate; sbrBitRateOut = sbrEncoder_LimitBitRate(sbrElementBitRateIn, cm.elInfo[e].nChannelsInEl, coreSamplingRate, aot); if (sbrBitRateOut == 0) { return 0; } /* If bitrates don't match, distribution and limiting needs to be determined again. Abort element loop and restart with adapted bitrate. */ if (sbrElementBitRateIn != sbrBitRateOut) { if (sbrElementBitRateIn < sbrBitRateOut) { adjustedBitrate = fMax(initialBitrate, (INT)fDivNorm((FIXP_DBL)(sbrBitRateOut + 8), cm.elInfo[e].relativeBits)); break; } if (sbrElementBitRateIn > sbrBitRateOut) { adjustedBitrate = fMin(initialBitrate, (INT)fDivNorm((FIXP_DBL)(sbrBitRateOut - 8), cm.elInfo[e].relativeBits)); break; } } /* sbrElementBitRateIn != sbrBitRateOut */ } /* elements */ numIterations++; /* restrict iteration to worst case of num elements */ } while ((initialBitrate != adjustedBitrate) && (numIterations <= cm.nElements)); /* Unequal bitrates mean that no reasonable bitrate configuration found. */ bitRate = (initialBitrate == adjustedBitrate) ? adjustedBitrate : 0; } /* Limit bit rate in respect to available MPS modes if active */ if ((aot == AOT_ER_AAC_ELD) && (syntaxFlags & AC_LD_MPS) && (channelMode == MODE_1)) { bitRate = FDK_MpegsEnc_GetClosestBitRate( aot, MODE_212, samplingRate, (sbrActive) ? sbrDownSampleRate : 0, bitRate); } return bitRate; } /* * \brief Get CBR bitrate * * \hAacConfig Internal encoder config * \return Bitrate */ static INT FDKaacEnc_GetCBRBitrate(const HANDLE_AACENC_CONFIG hAacConfig, const INT userSbrRatio) { INT bitrate = FDKaacEnc_GetChannelModeConfiguration(hAacConfig->channelMode) ->nChannelsEff * hAacConfig->sampleRate; if (isPsActive(hAacConfig->audioObjectType)) { bitrate = 1 * bitrate; /* 0.5 bit per sample */ } else if (isSbrActive(hAacConfig)) { if ((userSbrRatio == 2) || ((userSbrRatio == 0) && (hAacConfig->audioObjectType != AOT_ER_AAC_ELD))) { bitrate = (bitrate + (bitrate >> 2)) >> 1; /* 0.625 bits per sample */ } if ((userSbrRatio == 1) || ((userSbrRatio == 0) && (hAacConfig->audioObjectType == AOT_ER_AAC_ELD))) { bitrate = (bitrate + (bitrate >> 3)); /* 1.125 bits per sample */ } } else { bitrate = bitrate + (bitrate >> 1); /* 1.5 bits per sample */ } return bitrate; } /* * \brief Consistency check of given USER_PARAM struct and * copy back configuration from public struct into internal * encoder configuration struct. * * \hAacEncoder Internal encoder config which is to be updated * \param config User provided config (public struct) * \return returns always AAC_ENC_OK */ static AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder, USER_PARAM *config) { AACENC_ERROR err = AACENC_OK; /* Get struct pointers. */ HANDLE_AACENC_CONFIG hAacConfig = &hAacEncoder->aacConfig; /* Encoder settings update. */ hAacConfig->sampleRate = config->userSamplerate; if (config->userDownscaleFactor > 1) { hAacConfig->useTns = 0; hAacConfig->usePns = 0; hAacConfig->useIS = 0; } else { hAacConfig->useTns = config->userTns; hAacConfig->usePns = config->userPns; hAacConfig->useIS = config->userIntensity; } hAacConfig->audioObjectType = config->userAOT; hAacConfig->channelMode = GetCoreChannelMode(config->userChannelMode, hAacConfig->audioObjectType); hAacConfig->nChannels = FDKaacEnc_GetChannelModeConfiguration(hAacConfig->channelMode)->nChannels; hAacConfig->bitrateMode = (AACENC_BITRATE_MODE)config->userBitrateMode; hAacConfig->bandWidth = config->userBandwidth; hAacConfig->useRequant = config->userAfterburner; hAacConfig->anc_Rate = config->userAncDataRate; hAacConfig->syntaxFlags = 0; hAacConfig->epConfig = -1; if (hAacConfig->audioObjectType != AOT_ER_AAC_ELD && config->userDownscaleFactor > 1) { return AACENC_INVALID_CONFIG; /* downscaling only allowed for AOT_ER_AAC_ELD */ } if (config->userDownscaleFactor > 1 && config->userSbrEnabled == 1) { return AACENC_INVALID_CONFIG; /* downscaling only allowed for AOT_ER_AAC_ELD w/o SBR */ } if (config->userDownscaleFactor > 1 && config->userChannelMode == 128) { return AACENC_INVALID_CONFIG; /* disallow downscaling for AAC-ELDv2 */ } if (config->userTpType == TT_MP4_LATM_MCP1 || config->userTpType == TT_MP4_LATM_MCP0 || config->userTpType == TT_MP4_LOAS) { hAacConfig->audioMuxVersion = config->userTpAmxv; } else { hAacConfig->audioMuxVersion = -1; } /* Adapt internal AOT when necessary. */ switch (config->userAOT) { case AOT_MP2_AAC_LC: case AOT_MP2_SBR: hAacConfig->usePns = 0; FDK_FALLTHROUGH; case AOT_AAC_LC: case AOT_SBR: case AOT_PS: config->userTpType = (config->userTpType != TT_UNKNOWN) ? config->userTpType : TT_MP4_ADTS; hAacConfig->framelength = (config->userFramelength != (UINT)-1) ? config->userFramelength : 1024; if (hAacConfig->framelength != 1024 && hAacConfig->framelength != 960) { return AACENC_INVALID_CONFIG; } break; case AOT_ER_AAC_LD: hAacConfig->epConfig = 0; hAacConfig->syntaxFlags |= AC_ER | AC_LD; hAacConfig->syntaxFlags |= ((config->userErTools & 0x1) ? AC_ER_VCB11 : 0); hAacConfig->syntaxFlags |= ((config->userErTools & 0x2) ? AC_ER_HCR : 0); hAacConfig->syntaxFlags |= ((config->userErTools & 0x4) ? AC_ER_RVLC : 0); config->userTpType = (config->userTpType != TT_UNKNOWN) ? config->userTpType : TT_MP4_LOAS; hAacConfig->framelength = (config->userFramelength != (UINT)-1) ? config->userFramelength : 512; if (hAacConfig->framelength != 512 && hAacConfig->framelength != 480) { return AACENC_INVALID_CONFIG; } break; case AOT_ER_AAC_ELD: hAacConfig->epConfig = 0; hAacConfig->syntaxFlags |= AC_ER | AC_ELD; hAacConfig->syntaxFlags |= ((config->userErTools & 0x1) ? AC_ER_VCB11 : 0); hAacConfig->syntaxFlags |= ((config->userErTools & 0x2) ? AC_ER_HCR : 0); hAacConfig->syntaxFlags |= ((config->userErTools & 0x4) ? AC_ER_RVLC : 0); hAacConfig->syntaxFlags |= ((config->userSbrEnabled == 1) ? AC_SBR_PRESENT : 0); hAacConfig->syntaxFlags |= ((config->userChannelMode == MODE_212) ? AC_LD_MPS : 0); config->userTpType = (config->userTpType != TT_UNKNOWN) ? config->userTpType : TT_MP4_LOAS; hAacConfig->framelength = (config->userFramelength != (UINT)-1) ? config->userFramelength : 512; hAacConfig->downscaleFactor = config->userDownscaleFactor; switch (config->userDownscaleFactor) { case 1: break; case 2: case 4: hAacConfig->syntaxFlags |= AC_ELD_DOWNSCALE; break; default: return AACENC_INVALID_CONFIG; } if (hAacConfig->framelength != 512 && hAacConfig->framelength != 480 && hAacConfig->framelength != 256 && hAacConfig->framelength != 240 && hAacConfig->framelength != 128 && hAacConfig->framelength != 120) { return AACENC_INVALID_CONFIG; } break; default: break; } /* Initialize SBR parameters */ if ((config->userSbrRatio == 0) && (isSbrActive(hAacConfig))) { /* Automatic SBR ratio configuration * - downsampled SBR for ELD * - otherwise always dualrate SBR */ if (hAacConfig->audioObjectType == AOT_ER_AAC_ELD) { hAacConfig->sbrRatio = ((hAacConfig->syntaxFlags & AC_LD_MPS) && (hAacConfig->sampleRate >= 27713)) ? 2 : 1; } else { hAacConfig->sbrRatio = 2; } } else { /* SBR ratio has been set by the user, so use it. */ hAacConfig->sbrRatio = isSbrActive(hAacConfig) ? config->userSbrRatio : 0; } /* Set default bitrate */ hAacConfig->bitRate = config->userBitrate; switch (hAacConfig->bitrateMode) { case AACENC_BR_MODE_CBR: /* Set default bitrate if no external bitrate declared. */ if (config->userBitrate == (UINT)-1) { hAacConfig->bitRate = FDKaacEnc_GetCBRBitrate(hAacConfig, config->userSbrRatio); } hAacConfig->averageBits = -1; break; case AACENC_BR_MODE_VBR_1: case AACENC_BR_MODE_VBR_2: case AACENC_BR_MODE_VBR_3: case AACENC_BR_MODE_VBR_4: case AACENC_BR_MODE_VBR_5: /* Adjust bitrate mode in case given peak bitrate is lower than expected * VBR bitrate. */ if ((INT)config->userPeakBitrate != -1) { hAacConfig->bitrateMode = FDKaacEnc_AdjustVBRBitrateMode( hAacConfig->bitrateMode, config->userPeakBitrate, hAacConfig->channelMode); } /* Get bitrate in VBR configuration */ /* In VBR mode; SBR-modul depends on bitrate, core encoder on bitrateMode. */ hAacConfig->bitRate = FDKaacEnc_GetVBRBitrate(hAacConfig->bitrateMode, hAacConfig->channelMode); break; default: return AACENC_INVALID_CONFIG; } /* set bitreservoir size */ switch (hAacConfig->bitrateMode) { case AACENC_BR_MODE_VBR_1: case AACENC_BR_MODE_VBR_2: case AACENC_BR_MODE_VBR_3: case AACENC_BR_MODE_VBR_4: case AACENC_BR_MODE_VBR_5: case AACENC_BR_MODE_CBR: if ((INT)config->userPeakBitrate != -1) { hAacConfig->maxBitsPerFrame = (FDKaacEnc_CalcBitsPerFrame( fMax(hAacConfig->bitRate, (INT)config->userPeakBitrate), hAacConfig->framelength, hAacConfig->sampleRate) + 7) & ~7; } else { hAacConfig->maxBitsPerFrame = -1; } if (hAacConfig->audioMuxVersion == 2) { hAacConfig->minBitsPerFrame = fMin(32 * 8, FDKaacEnc_CalcBitsPerFrame(hAacConfig->bitRate, hAacConfig->framelength, hAacConfig->sampleRate)) & ~7; } break; default: return AACENC_INVALID_CONFIG; } /* Max bits per frame limitation depending on transport format. */ if ((config->userTpNsubFrames > 1)) { int maxFrameLength = 8 * hAacEncoder->outBufferInBytes; switch (config->userTpType) { case TT_MP4_LOAS: maxFrameLength = fMin(maxFrameLength, 8 * (1 << 13)) / config->userTpNsubFrames; break; case TT_MP4_ADTS: maxFrameLength = fMin(maxFrameLength, 8 * ((1 << 13) - 1)) / config->userTpNsubFrames; break; default: maxFrameLength = -1; } if (maxFrameLength != -1) { if (hAacConfig->maxBitsPerFrame > maxFrameLength) { return AACENC_INVALID_CONFIG; } else if (hAacConfig->maxBitsPerFrame == -1) { hAacConfig->maxBitsPerFrame = maxFrameLength; } } } if ((hAacConfig->audioObjectType == AOT_ER_AAC_ELD) && !(hAacConfig->syntaxFlags & AC_ELD_DOWNSCALE) && (config->userSbrEnabled == (UCHAR)-1) && (config->userSbrRatio == 0) && ((hAacConfig->syntaxFlags & AC_LD_MPS) == 0)) { const ELD_SBR_CONFIGURATOR *pConfig = NULL; if (NULL != (pConfig = eldSbrConfigurator( FDKaacEnc_GetChannelModeConfiguration(hAacConfig->channelMode) ->nChannels, hAacConfig->sampleRate, hAacConfig->bitRate))) { hAacConfig->syntaxFlags |= (pConfig->sbrMode == 0) ? 0 : AC_SBR_PRESENT; hAacConfig->syntaxFlags |= (pConfig->chMode == MODE_212) ? AC_LD_MPS : 0; hAacConfig->channelMode = GetCoreChannelMode(pConfig->chMode, hAacConfig->audioObjectType); hAacConfig->nChannels = FDKaacEnc_GetChannelModeConfiguration(hAacConfig->channelMode) ->nChannels; hAacConfig->sbrRatio = (pConfig->sbrMode == 0) ? 0 : (pConfig->sbrMode == 1) ? 1 : 2; } } { UCHAR tpSignaling = getSbrSignalingMode(hAacConfig->audioObjectType, config->userTpType, config->userTpSignaling, hAacConfig->sbrRatio); if ((hAacConfig->audioObjectType == AOT_AAC_LC || hAacConfig->audioObjectType == AOT_SBR || hAacConfig->audioObjectType == AOT_PS) && (config->userTpType == TT_MP4_LATM_MCP1 || config->userTpType == TT_MP4_LATM_MCP0 || config->userTpType == TT_MP4_LOAS) && (tpSignaling == 1) && (config->userTpAmxv == 0)) { /* For backward compatible explicit signaling, AMV1 has to be active */ return AACENC_INVALID_CONFIG; } if ((hAacConfig->audioObjectType == AOT_AAC_LC || hAacConfig->audioObjectType == AOT_SBR || hAacConfig->audioObjectType == AOT_PS) && (tpSignaling == 0) && (hAacConfig->sbrRatio == 1)) { /* Downsampled SBR has to be signaled explicitely (for transmission of SBR * sampling fequency) */ return AACENC_INVALID_CONFIG; } } switch (hAacConfig->bitrateMode) { case AACENC_BR_MODE_CBR: case AACENC_BR_MODE_VBR_1: case AACENC_BR_MODE_VBR_2: case AACENC_BR_MODE_VBR_3: case AACENC_BR_MODE_VBR_4: case AACENC_BR_MODE_VBR_5: /* We need the frame length to call aacEncoder_LimitBitrate() */ if (0 >= (hAacConfig->bitRate = aacEncoder_LimitBitrate( NULL, hAacConfig->sampleRate, hAacConfig->framelength, hAacConfig->nChannels, hAacConfig->channelMode, hAacConfig->bitRate, hAacConfig->nSubFrames, isSbrActive(hAacConfig), hAacConfig->sbrRatio, hAacConfig->syntaxFlags, hAacConfig->audioObjectType))) { return AACENC_INVALID_CONFIG; } break; default: break; } /* Configure PNS */ if (AACENC_BR_MODE_IS_VBR(hAacConfig->bitrateMode) /* VBR without PNS. */ || (hAacConfig->useTns == 0)) /* TNS required. */ { hAacConfig->usePns = 0; } if (hAacConfig->epConfig >= 0) { hAacConfig->syntaxFlags |= AC_ER; if (((INT)hAacConfig->channelMode < 1) || ((INT)hAacConfig->channelMode > 14)) { return AACENC_INVALID_CONFIG; /* Channel config 0 not supported. */ } } if ((hAacConfig->syntaxFlags & AC_LD_MPS) == 0) { if (FDKaacEnc_DetermineEncoderMode(&hAacConfig->channelMode, hAacConfig->nChannels) != AAC_ENC_OK) { return AACENC_INVALID_CONFIG; /* nChannels doesn't match chMode, this is just a check-up */ } } if ((hAacConfig->nChannels > hAacEncoder->nMaxAacChannels) || ((FDKaacEnc_GetChannelModeConfiguration(hAacConfig->channelMode) ->nChannelsEff > hAacEncoder->nMaxSbrChannels) && isSbrActive(hAacConfig))) { return AACENC_INVALID_CONFIG; /* not enough channels allocated */ } /* Meta data restriction. */ switch (hAacConfig->audioObjectType) { /* Allow metadata support */ case AOT_AAC_LC: case AOT_SBR: case AOT_PS: case AOT_MP2_AAC_LC: case AOT_MP2_SBR: hAacEncoder->metaDataAllowed = 1; if (!((((INT)hAacConfig->channelMode >= 1) && ((INT)hAacConfig->channelMode <= 14)) || (MODE_7_1_REAR_SURROUND == hAacConfig->channelMode) || (MODE_7_1_FRONT_CENTER == hAacConfig->channelMode))) { config->userMetaDataMode = 0; } break; /* Prohibit metadata support */ default: hAacEncoder->metaDataAllowed = 0; } return err; } static INT aacenc_SbrCallback(void *self, HANDLE_FDK_BITSTREAM hBs, const INT sampleRateIn, const INT sampleRateOut, const INT samplesPerFrame, const AUDIO_OBJECT_TYPE coreCodec, const MP4_ELEMENT_ID elementID, const INT elementIndex, const UCHAR harmonicSbr, const UCHAR stereoConfigIndex, const UCHAR configMode, UCHAR *configChanged, const INT downscaleFactor) { HANDLE_AACENCODER hAacEncoder = (HANDLE_AACENCODER)self; sbrEncoder_GetHeader(hAacEncoder->hEnvEnc, hBs, elementIndex, 0); return 0; } INT aacenc_SscCallback(void *self, HANDLE_FDK_BITSTREAM hBs, const AUDIO_OBJECT_TYPE coreCodec, const INT samplingRate, const INT frameSize, const INT stereoConfigIndex, const INT coreSbrFrameLengthIndex, const INT configBytes, const UCHAR configMode, UCHAR *configChanged) { HANDLE_AACENCODER hAacEncoder = (HANDLE_AACENCODER)self; return (FDK_MpegsEnc_WriteSpatialSpecificConfig(hAacEncoder->hMpsEnc, hBs)); } static AACENC_ERROR aacEncInit(HANDLE_AACENCODER hAacEncoder, ULONG InitFlags, USER_PARAM *config) { AACENC_ERROR err = AACENC_OK; INT aacBufferOffset = 0; HANDLE_SBR_ENCODER *hSbrEncoder = &hAacEncoder->hEnvEnc; HANDLE_AACENC_CONFIG hAacConfig = &hAacEncoder->aacConfig; hAacEncoder->nZerosAppended = 0; /* count appended zeros */ INT frameLength = hAacConfig->framelength; if ((InitFlags & AACENC_INIT_CONFIG)) { CHANNEL_MODE prevChMode = hAacConfig->channelMode; /* Verify settings and update: config -> heAacEncoder */ if ((err = FDKaacEnc_AdjustEncSettings(hAacEncoder, config)) != AACENC_OK) { return err; } frameLength = hAacConfig->framelength; /* adapt temporal framelength */ /* Seamless channel reconfiguration in sbr not fully implemented */ if ((prevChMode != hAacConfig->channelMode) && isSbrActive(hAacConfig)) { InitFlags |= AACENC_INIT_STATES; } } /* Clear input buffer */ if (InitFlags == AACENC_INIT_ALL) { FDKmemclear(hAacEncoder->inputBuffer, sizeof(INT_PCM) * hAacEncoder->inputBufferSize); } if ((InitFlags & AACENC_INIT_CONFIG)) { aacBufferOffset = 0; switch (hAacConfig->audioObjectType) { case AOT_ER_AAC_LD: hAacEncoder->nDelay = DELAY_AACLD(hAacConfig->framelength); break; case AOT_ER_AAC_ELD: hAacEncoder->nDelay = DELAY_AACELD(hAacConfig->framelength); break; default: hAacEncoder->nDelay = DELAY_AAC(hAacConfig->framelength); /* AAC encoder delay */ } hAacConfig->ancDataBitRate = 0; } if ((NULL != hAacEncoder->hEnvEnc) && isSbrActive(hAacConfig) && ((InitFlags & AACENC_INIT_CONFIG) || (InitFlags & AACENC_INIT_STATES))) { INT sbrError; UINT initFlag = 0; SBR_ELEMENT_INFO sbrElInfo[(8)]; CHANNEL_MAPPING channelMapping; CHANNEL_MODE channelMode = isPsActive(hAacConfig->audioObjectType) ? config->userChannelMode : hAacConfig->channelMode; INT numChannels = isPsActive(hAacConfig->audioObjectType) ? config->nChannels : hAacConfig->nChannels; if (FDKaacEnc_InitChannelMapping(channelMode, hAacConfig->channelOrder, &channelMapping) != AAC_ENC_OK) { return AACENC_INIT_ERROR; } /* Check return value and if the SBR encoder can handle enough elements */ if (channelMapping.nElements > (8)) { return AACENC_INIT_ERROR; } aacEncDistributeSbrBits(&channelMapping, sbrElInfo, hAacConfig->bitRate); initFlag += (InitFlags & AACENC_INIT_STATES) ? 1 : 0; /* Let the SBR encoder take a look at the configuration and change if * required. */ sbrError = sbrEncoder_Init( *hSbrEncoder, sbrElInfo, channelMapping.nElements, hAacEncoder->inputBuffer, hAacEncoder->inputBufferSizePerChannel, &hAacConfig->bandWidth, &aacBufferOffset, &numChannels, hAacConfig->syntaxFlags, &hAacConfig->sampleRate, &hAacConfig->sbrRatio, &frameLength, hAacConfig->audioObjectType, &hAacEncoder->nDelay, (hAacConfig->audioObjectType == AOT_ER_AAC_ELD) ? 1 : TRANS_FAC, (config->userTpHeaderPeriod != 0xFF) ? config->userTpHeaderPeriod : DEFAULT_HEADER_PERIOD_REPETITION_RATE, initFlag); /* Suppress AOT reconfiguration and check error status. */ if ((sbrError) || (numChannels != hAacConfig->nChannels)) { return AACENC_INIT_SBR_ERROR; } if (numChannels == 1) { hAacConfig->channelMode = MODE_1; } /* Never use PNS if SBR is active */ if (hAacConfig->usePns) { hAacConfig->usePns = 0; } /* estimated bitrate consumed by SBR or PS */ hAacConfig->ancDataBitRate = sbrEncoder_GetEstimateBitrate(*hSbrEncoder); } /* sbr initialization */ if ((hAacEncoder->hMpsEnc != NULL) && (hAacConfig->syntaxFlags & AC_LD_MPS)) { int coreCoderDelay = DELAY_AACELD(hAacConfig->framelength); if (isSbrActive(hAacConfig)) { coreCoderDelay = hAacConfig->sbrRatio * coreCoderDelay + sbrEncoder_GetInputDataDelay(*hSbrEncoder); } if (MPS_ENCODER_OK != FDK_MpegsEnc_Init(hAacEncoder->hMpsEnc, hAacConfig->audioObjectType, config->userSamplerate, hAacConfig->bitRate, isSbrActive(hAacConfig) ? hAacConfig->sbrRatio : 0, frameLength, /* for dual rate sbr this value is already multiplied by 2 */ hAacEncoder->inputBufferSizePerChannel, coreCoderDelay)) { return AACENC_INIT_MPS_ERROR; } } hAacEncoder->nDelay = fMax(FDK_MpegsEnc_GetDelay(hAacEncoder->hMpsEnc), hAacEncoder->nDelay); /* * Initialize Transport - Module. */ if ((InitFlags & AACENC_INIT_TRANSPORT)) { UINT flags = 0; FDKaacEnc_MapConfig( &hAacEncoder->coderConfig, config, getSbrSignalingMode(hAacConfig->audioObjectType, config->userTpType, config->userTpSignaling, hAacConfig->sbrRatio), hAacConfig); /* create flags for transport encoder */ if (config->userTpAmxv != 0) { flags |= TP_FLAG_LATM_AMV; } /* Clear output buffer */ FDKmemclear(hAacEncoder->outBuffer, hAacEncoder->outBufferInBytes * sizeof(UCHAR)); /* Initialize Bitstream encoder */ if (transportEnc_Init(hAacEncoder->hTpEnc, hAacEncoder->outBuffer, hAacEncoder->outBufferInBytes, config->userTpType, &hAacEncoder->coderConfig, flags) != 0) { return AACENC_INIT_TP_ERROR; } } /* transport initialization */ /* * Initialize AAC - Core. */ if ((InitFlags & AACENC_INIT_CONFIG) || (InitFlags & AACENC_INIT_STATES)) { if (FDKaacEnc_Initialize( hAacEncoder->hAacEnc, hAacConfig, hAacEncoder->hTpEnc, (InitFlags & AACENC_INIT_STATES) ? 1 : 0) != AAC_ENC_OK) { return AACENC_INIT_AAC_ERROR; } } /* aac initialization */ /* * Initialize Meta Data - Encoder. */ if (hAacEncoder->hMetadataEnc && (hAacEncoder->metaDataAllowed != 0) && ((InitFlags & AACENC_INIT_CONFIG) || (InitFlags & AACENC_INIT_STATES))) { INT inputDataDelay = DELAY_AAC(hAacConfig->framelength); if (isSbrActive(hAacConfig) && hSbrEncoder != NULL) { inputDataDelay = hAacConfig->sbrRatio * inputDataDelay + sbrEncoder_GetInputDataDelay(*hSbrEncoder); } if (FDK_MetadataEnc_Init(hAacEncoder->hMetadataEnc, ((InitFlags & AACENC_INIT_STATES) ? 1 : 0), config->userMetaDataMode, inputDataDelay, frameLength, config->userSamplerate, config->nChannels, config->userChannelMode, hAacConfig->channelOrder) != 0) { return AACENC_INIT_META_ERROR; } hAacEncoder->nDelay += FDK_MetadataEnc_GetDelay(hAacEncoder->hMetadataEnc); } /* Get custom delay, i.e. the codec delay w/o the decoder's SBR- or MPS delay */ if ((hAacEncoder->hMpsEnc != NULL) && (hAacConfig->syntaxFlags & AC_LD_MPS)) { hAacEncoder->nDelayCore = hAacEncoder->nDelay - fMax(0, FDK_MpegsEnc_GetDecDelay(hAacEncoder->hMpsEnc)); } else if (isSbrActive(hAacConfig) && hSbrEncoder != NULL) { hAacEncoder->nDelayCore = hAacEncoder->nDelay - fMax(0, sbrEncoder_GetSbrDecDelay(hAacEncoder->hEnvEnc)); } else { hAacEncoder->nDelayCore = hAacEncoder->nDelay; } /* * Update pointer to working buffer. */ if ((InitFlags & AACENC_INIT_CONFIG)) { hAacEncoder->inputBufferOffset = aacBufferOffset; hAacEncoder->nSamplesToRead = frameLength * config->nChannels; } /* parameter changed */ return AACENC_OK; } AACENC_ERROR aacEncOpen(HANDLE_AACENCODER *phAacEncoder, const UINT encModules, const UINT maxChannels) { AACENC_ERROR err = AACENC_OK; HANDLE_AACENCODER hAacEncoder = NULL; if (phAacEncoder == NULL) { err = AACENC_INVALID_HANDLE; goto bail; } /* allocate memory */ hAacEncoder = Get_AacEncoder(); if (hAacEncoder == NULL) { err = AACENC_MEMORY_ERROR; goto bail; } FDKmemclear(hAacEncoder, sizeof(AACENCODER)); /* Specify encoder modules to be allocated. */ if (encModules == 0) { C_ALLOC_SCRATCH_START(_pLibInfo, LIB_INFO, FDK_MODULE_LAST) LIB_INFO(*pLibInfo) [FDK_MODULE_LAST] = (LIB_INFO(*)[FDK_MODULE_LAST])_pLibInfo; FDKinitLibInfo(*pLibInfo); aacEncGetLibInfo(*pLibInfo); hAacEncoder->encoder_modis = ENC_MODE_FLAG_AAC; if (FDKlibInfo_getCapabilities(*pLibInfo, FDK_SBRENC) & CAPF_SBR_HQ) { hAacEncoder->encoder_modis |= ENC_MODE_FLAG_SBR; } if (FDKlibInfo_getCapabilities(*pLibInfo, FDK_SBRENC) & CAPF_SBR_PS_MPEG) { hAacEncoder->encoder_modis |= ENC_MODE_FLAG_PS; } if (FDKlibInfo_getCapabilities(*pLibInfo, FDK_AACENC) & CAPF_AAC_DRC) { hAacEncoder->encoder_modis |= ENC_MODE_FLAG_META; } hAacEncoder->encoder_modis |= ENC_MODE_FLAG_SAC; C_ALLOC_SCRATCH_END(_pLibInfo, LIB_INFO, FDK_MODULE_LAST) } else { hAacEncoder->encoder_modis = encModules; } /* Determine max channel configuration. */ if (maxChannels == 0) { hAacEncoder->nMaxAacChannels = (8); hAacEncoder->nMaxSbrChannels = (8); } else { hAacEncoder->nMaxAacChannels = (maxChannels & 0x00FF); if ((hAacEncoder->encoder_modis & ENC_MODE_FLAG_SBR)) { hAacEncoder->nMaxSbrChannels = (maxChannels & 0xFF00) ? (maxChannels >> 8) : hAacEncoder->nMaxAacChannels; } if ((hAacEncoder->nMaxAacChannels > (8)) || (hAacEncoder->nMaxSbrChannels > (8))) { err = AACENC_INVALID_CONFIG; goto bail; } } /* maxChannels==0 */ /* Max number of elements could be tuned any more. */ hAacEncoder->nMaxAacElements = fixMin(((8)), hAacEncoder->nMaxAacChannels); hAacEncoder->nMaxSbrElements = fixMin((8), hAacEncoder->nMaxSbrChannels); /* In case of memory overlay, allocate memory out of libraries */ if (hAacEncoder->encoder_modis & (ENC_MODE_FLAG_SBR | ENC_MODE_FLAG_PS)) hAacEncoder->inputBufferSizePerChannel = INPUTBUFFER_SIZE; else hAacEncoder->inputBufferSizePerChannel = (1024); hAacEncoder->inputBufferSize = hAacEncoder->nMaxAacChannels * hAacEncoder->inputBufferSizePerChannel; if (NULL == (hAacEncoder->inputBuffer = (INT_PCM *)FDKcalloc( hAacEncoder->inputBufferSize, sizeof(INT_PCM)))) { err = AACENC_MEMORY_ERROR; goto bail; } /* Open SBR Encoder */ if (hAacEncoder->encoder_modis & ENC_MODE_FLAG_SBR) { if (sbrEncoder_Open( &hAacEncoder->hEnvEnc, hAacEncoder->nMaxSbrElements, hAacEncoder->nMaxSbrChannels, (hAacEncoder->encoder_modis & ENC_MODE_FLAG_PS) ? 1 : 0)) { err = AACENC_MEMORY_ERROR; goto bail; } if (NULL == (hAacEncoder->pSbrPayload = (SBRENC_EXT_PAYLOAD *)FDKcalloc( 1, sizeof(SBRENC_EXT_PAYLOAD)))) { err = AACENC_MEMORY_ERROR; goto bail; } } /* (encoder_modis&ENC_MODE_FLAG_SBR) */ /* Open Aac Encoder */ if (FDKaacEnc_Open(&hAacEncoder->hAacEnc, hAacEncoder->nMaxAacElements, hAacEncoder->nMaxAacChannels, (1)) != AAC_ENC_OK) { err = AACENC_MEMORY_ERROR; goto bail; } /* Bitstream output buffer */ hAacEncoder->outBufferInBytes = 1 << (DFRACT_BITS - CntLeadingZeros(fixMax( 1, ((1) * hAacEncoder->nMaxAacChannels * 6144) >> 3))); /* buffer has to be 2^n */ if (NULL == (hAacEncoder->outBuffer = (UCHAR *)FDKcalloc( hAacEncoder->outBufferInBytes, sizeof(UCHAR)))) { err = AACENC_MEMORY_ERROR; goto bail; } /* Open Meta Data Encoder */ if (hAacEncoder->encoder_modis & ENC_MODE_FLAG_META) { if (FDK_MetadataEnc_Open(&hAacEncoder->hMetadataEnc, (UINT)hAacEncoder->nMaxAacChannels)) { err = AACENC_MEMORY_ERROR; goto bail; } } /* (encoder_modis&ENC_MODE_FLAG_META) */ /* Open MPEG Surround Encoder */ if (hAacEncoder->encoder_modis & ENC_MODE_FLAG_SAC) { if (MPS_ENCODER_OK != FDK_MpegsEnc_Open(&hAacEncoder->hMpsEnc)) { err = AACENC_MEMORY_ERROR; goto bail; } } /* (hAacEncoder->encoder_modis&ENC_MODE_FLAG_SAC) */ /* Open Transport Encoder */ if (transportEnc_Open(&hAacEncoder->hTpEnc) != 0) { err = AACENC_MEMORY_ERROR; goto bail; } else { C_ALLOC_SCRATCH_START(_pLibInfo, LIB_INFO, FDK_MODULE_LAST) LIB_INFO(*pLibInfo) [FDK_MODULE_LAST] = (LIB_INFO(*)[FDK_MODULE_LAST])_pLibInfo; FDKinitLibInfo(*pLibInfo); transportEnc_GetLibInfo(*pLibInfo); /* Get capabilty flag for transport encoder. */ hAacEncoder->CAPF_tpEnc = FDKlibInfo_getCapabilities(*pLibInfo, FDK_TPENC); C_ALLOC_SCRATCH_END(_pLibInfo, LIB_INFO, FDK_MODULE_LAST) } if (transportEnc_RegisterSbrCallback(hAacEncoder->hTpEnc, aacenc_SbrCallback, hAacEncoder) != 0) { err = AACENC_INIT_TP_ERROR; goto bail; } if (transportEnc_RegisterSscCallback(hAacEncoder->hTpEnc, aacenc_SscCallback, hAacEncoder) != 0) { err = AACENC_INIT_TP_ERROR; goto bail; } /* Initialize encoder instance with default parameters. */ aacEncDefaultConfig(&hAacEncoder->aacConfig, &hAacEncoder->extParam); /* Initialize headerPeriod in coderConfig for aacEncoder_GetParam(). */ hAacEncoder->coderConfig.headerPeriod = hAacEncoder->extParam.userTpHeaderPeriod; /* All encoder modules have to be initialized */ hAacEncoder->InitFlags = AACENC_INIT_ALL; /* Return encoder instance */ *phAacEncoder = hAacEncoder; return err; bail: aacEncClose(&hAacEncoder); return err; } AACENC_ERROR aacEncClose(HANDLE_AACENCODER *phAacEncoder) { AACENC_ERROR err = AACENC_OK; if (phAacEncoder == NULL) { err = AACENC_INVALID_HANDLE; goto bail; } if (*phAacEncoder != NULL) { HANDLE_AACENCODER hAacEncoder = *phAacEncoder; if (hAacEncoder->inputBuffer != NULL) { FDKfree(hAacEncoder->inputBuffer); hAacEncoder->inputBuffer = NULL; } if (hAacEncoder->outBuffer != NULL) { FDKfree(hAacEncoder->outBuffer); hAacEncoder->outBuffer = NULL; } if (hAacEncoder->hEnvEnc) { sbrEncoder_Close(&hAacEncoder->hEnvEnc); } if (hAacEncoder->pSbrPayload != NULL) { FDKfree(hAacEncoder->pSbrPayload); hAacEncoder->pSbrPayload = NULL; } if (hAacEncoder->hAacEnc) { FDKaacEnc_Close(&hAacEncoder->hAacEnc); } transportEnc_Close(&hAacEncoder->hTpEnc); if (hAacEncoder->hMetadataEnc) { FDK_MetadataEnc_Close(&hAacEncoder->hMetadataEnc); } if (hAacEncoder->hMpsEnc) { FDK_MpegsEnc_Close(&hAacEncoder->hMpsEnc); } Free_AacEncoder(phAacEncoder); } bail: return err; } AACENC_ERROR aacEncEncode(const HANDLE_AACENCODER hAacEncoder, const AACENC_BufDesc *inBufDesc, const AACENC_BufDesc *outBufDesc, const AACENC_InArgs *inargs, AACENC_OutArgs *outargs) { AACENC_ERROR err = AACENC_OK; INT i, nBsBytes = 0; INT outBytes[(1)]; int nExtensions = 0; int ancDataExtIdx = -1; /* deal with valid encoder handle */ if (hAacEncoder == NULL) { err = AACENC_INVALID_HANDLE; goto bail; } /* * Adjust user settings and trigger reinitialization. */ if (hAacEncoder->InitFlags != 0) { err = aacEncInit(hAacEncoder, hAacEncoder->InitFlags, &hAacEncoder->extParam); if (err != AACENC_OK) { /* keep init flags alive! */ goto bail; } hAacEncoder->InitFlags = AACENC_INIT_NONE; } if (outargs != NULL) { FDKmemclear(outargs, sizeof(AACENC_OutArgs)); } if (outBufDesc != NULL) { for (i = 0; i < outBufDesc->numBufs; i++) { if (outBufDesc->bufs[i] != NULL) { FDKmemclear(outBufDesc->bufs[i], outBufDesc->bufSizes[i]); } } } /* * If only encoder handle given, independent (re)initialization can be * triggered. */ if ((inBufDesc == NULL) && (outBufDesc == NULL) && (inargs == NULL) && (outargs == NULL)) { goto bail; } /* check if buffer descriptors are filled out properly. */ if ((inargs == NULL) || (outargs == NULL) || ((AACENC_OK != validateBufDesc(inBufDesc)) && (inargs->numInSamples > 0)) || (AACENC_OK != validateBufDesc(outBufDesc))) { err = AACENC_UNSUPPORTED_PARAMETER; goto bail; } /* reset buffer wich signals number of valid bytes in output bitstream buffer */ FDKmemclear(outBytes, hAacEncoder->aacConfig.nSubFrames * sizeof(INT)); /* * Manage incoming audio samples. */ if ((inBufDesc != NULL) && (inargs->numInSamples > 0) && (getBufDescIdx(inBufDesc, IN_AUDIO_DATA) != -1)) { /* Fetch data until nSamplesToRead reached */ INT idx = getBufDescIdx(inBufDesc, IN_AUDIO_DATA); INT newSamples = fixMax(0, fixMin(inargs->numInSamples, hAacEncoder->nSamplesToRead - hAacEncoder->nSamplesRead)); INT_PCM *pIn = hAacEncoder->inputBuffer + (hAacEncoder->inputBufferOffset + hAacEncoder->nSamplesRead) / hAacEncoder->aacConfig.nChannels; newSamples -= (newSamples % hAacEncoder->extParam .nChannels); /* process multiple samples of input channels */ /* Copy new input samples to internal buffer */ if (inBufDesc->bufElSizes[idx] == (INT)sizeof(INT_PCM)) { FDK_deinterleave((INT_PCM *)inBufDesc->bufs[idx], pIn, hAacEncoder->extParam.nChannels, newSamples / hAacEncoder->extParam.nChannels, hAacEncoder->inputBufferSizePerChannel); } else if (inBufDesc->bufElSizes[idx] > (INT)sizeof(INT_PCM)) { FDK_deinterleave((LONG *)inBufDesc->bufs[idx], pIn, hAacEncoder->extParam.nChannels, newSamples / hAacEncoder->extParam.nChannels, hAacEncoder->inputBufferSizePerChannel); } else { FDK_deinterleave((SHORT *)inBufDesc->bufs[idx], pIn, hAacEncoder->extParam.nChannels, newSamples / hAacEncoder->extParam.nChannels, hAacEncoder->inputBufferSizePerChannel); } hAacEncoder->nSamplesRead += newSamples; /* Number of fetched input buffer samples. */ outargs->numInSamples = newSamples; } /* input buffer completely filled ? */ if (hAacEncoder->nSamplesRead < hAacEncoder->nSamplesToRead) { /* - eof reached and flushing enabled, or - return to main and wait for further incoming audio samples */ if (inargs->numInSamples == -1) { if ((hAacEncoder->nZerosAppended < hAacEncoder->nDelay)) { int nZeros = (hAacEncoder->nSamplesToRead - hAacEncoder->nSamplesRead) / hAacEncoder->extParam.nChannels; FDK_ASSERT(nZeros >= 0); /* clear out until end-of-buffer */ if (nZeros) { for (i = 0; i < (int)hAacEncoder->extParam.nChannels; i++) { FDKmemclear(hAacEncoder->inputBuffer + i * hAacEncoder->inputBufferSizePerChannel + (hAacEncoder->inputBufferOffset + hAacEncoder->nSamplesRead) / hAacEncoder->extParam.nChannels, sizeof(INT_PCM) * nZeros); } hAacEncoder->nZerosAppended += nZeros; hAacEncoder->nSamplesRead = hAacEncoder->nSamplesToRead; } } else { /* flushing completed */ err = AACENC_ENCODE_EOF; /* eof reached */ goto bail; } } else { /* inargs->numInSamples!= -1 */ goto bail; /* not enough samples in input buffer and no flushing enabled */ } } /* init payload */ FDKmemclear(hAacEncoder->extPayload, sizeof(AACENC_EXT_PAYLOAD) * MAX_TOTAL_EXT_PAYLOADS); for (i = 0; i < MAX_TOTAL_EXT_PAYLOADS; i++) { hAacEncoder->extPayload[i].associatedChElement = -1; } if (hAacEncoder->pSbrPayload != NULL) { FDKmemclear(hAacEncoder->pSbrPayload, sizeof(*hAacEncoder->pSbrPayload)); } /* * Calculate Meta Data info. */ if ((hAacEncoder->hMetadataEnc != NULL) && (hAacEncoder->metaDataAllowed != 0)) { const AACENC_MetaData *pMetaData = NULL; AACENC_EXT_PAYLOAD *pMetaDataExtPayload = NULL; UINT nMetaDataExtensions = 0; INT matrix_mixdown_idx = 0; /* New meta data info available ? */ if (getBufDescIdx(inBufDesc, IN_METADATA_SETUP) != -1) { pMetaData = (AACENC_MetaData *) inBufDesc->bufs[getBufDescIdx(inBufDesc, IN_METADATA_SETUP)]; } FDK_MetadataEnc_Process( hAacEncoder->hMetadataEnc, hAacEncoder->inputBuffer + hAacEncoder->inputBufferOffset / hAacEncoder->coderConfig.noChannels, hAacEncoder->inputBufferSizePerChannel, hAacEncoder->nSamplesRead, pMetaData, &pMetaDataExtPayload, &nMetaDataExtensions, &matrix_mixdown_idx); for (i = 0; i < (INT)nMetaDataExtensions; i++) { /* Get meta data extension payload. */ hAacEncoder->extPayload[nExtensions++] = pMetaDataExtPayload[i]; } if ((matrix_mixdown_idx != -1) && ((hAacEncoder->extParam.userChannelMode == MODE_1_2_2) || (hAacEncoder->extParam.userChannelMode == MODE_1_2_2_1))) { /* Set matrix mixdown coefficient. */ UINT pceValue = (UINT)((0 << 3) | ((matrix_mixdown_idx & 0x3) << 1) | 1); if (hAacEncoder->extParam.userPceAdditions != pceValue) { hAacEncoder->extParam.userPceAdditions = pceValue; hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT; } } } /* * Encode MPS data. */ if ((hAacEncoder->hMpsEnc != NULL) && (hAacEncoder->aacConfig.syntaxFlags & AC_LD_MPS)) { AACENC_EXT_PAYLOAD mpsExtensionPayload; FDKmemclear(&mpsExtensionPayload, sizeof(AACENC_EXT_PAYLOAD)); if (MPS_ENCODER_OK != FDK_MpegsEnc_Process( hAacEncoder->hMpsEnc, hAacEncoder->inputBuffer + hAacEncoder->inputBufferOffset / hAacEncoder->coderConfig.noChannels, hAacEncoder->nSamplesRead, &mpsExtensionPayload)) { err = AACENC_ENCODE_ERROR; goto bail; } if ((mpsExtensionPayload.pData != NULL) && ((mpsExtensionPayload.dataSize != 0))) { hAacEncoder->extPayload[nExtensions++] = mpsExtensionPayload; } } if ((NULL != hAacEncoder->hEnvEnc) && (NULL != hAacEncoder->pSbrPayload) && isSbrActive(&hAacEncoder->aacConfig)) { INT nPayload = 0; /* * Encode SBR data. */ if (sbrEncoder_EncodeFrame(hAacEncoder->hEnvEnc, hAacEncoder->inputBuffer, hAacEncoder->inputBufferSizePerChannel, hAacEncoder->pSbrPayload->dataSize[nPayload], hAacEncoder->pSbrPayload->data[nPayload])) { err = AACENC_ENCODE_ERROR; goto bail; } else { /* Add SBR extension payload */ for (i = 0; i < (8); i++) { if (hAacEncoder->pSbrPayload->dataSize[nPayload][i] > 0) { hAacEncoder->extPayload[nExtensions].pData = hAacEncoder->pSbrPayload->data[nPayload][i]; { hAacEncoder->extPayload[nExtensions].dataSize = hAacEncoder->pSbrPayload->dataSize[nPayload][i]; hAacEncoder->extPayload[nExtensions].associatedChElement = getAssociatedChElement( &hAacEncoder->hEnvEnc->sbrElement[i]->elInfo, &hAacEncoder->hAacEnc->channelMapping); if (hAacEncoder->extPayload[nExtensions].associatedChElement == -1) { err = AACENC_ENCODE_ERROR; goto bail; } } hAacEncoder->extPayload[nExtensions].dataType = EXT_SBR_DATA; /* Once SBR Encoder supports SBR CRC set EXT_SBR_DATA_CRC */ nExtensions++; /* or EXT_SBR_DATA according to configuration. */ FDK_ASSERT(nExtensions <= MAX_TOTAL_EXT_PAYLOADS); } } nPayload++; } } /* sbrEnabled */ if ((inargs->numAncBytes > 0) && (getBufDescIdx(inBufDesc, IN_ANCILLRY_DATA) != -1)) { INT idx = getBufDescIdx(inBufDesc, IN_ANCILLRY_DATA); hAacEncoder->extPayload[nExtensions].dataSize = inargs->numAncBytes * 8; hAacEncoder->extPayload[nExtensions].pData = (UCHAR *)inBufDesc->bufs[idx]; hAacEncoder->extPayload[nExtensions].dataType = EXT_DATA_ELEMENT; hAacEncoder->extPayload[nExtensions].associatedChElement = -1; ancDataExtIdx = nExtensions; /* store index */ nExtensions++; } /* * Encode AAC - Core. */ if (FDKaacEnc_EncodeFrame(hAacEncoder->hAacEnc, hAacEncoder->hTpEnc, hAacEncoder->inputBuffer, hAacEncoder->inputBufferSizePerChannel, outBytes, hAacEncoder->extPayload) != AAC_ENC_OK) { err = AACENC_ENCODE_ERROR; goto bail; } if (ancDataExtIdx >= 0) { outargs->numAncBytes = inargs->numAncBytes - (hAacEncoder->extPayload[ancDataExtIdx].dataSize >> 3); } /* samples exhausted */ hAacEncoder->nSamplesRead -= hAacEncoder->nSamplesToRead; /* * Delay balancing buffer handling */ if (isSbrActive(&hAacEncoder->aacConfig)) { sbrEncoder_UpdateBuffers(hAacEncoder->hEnvEnc, hAacEncoder->inputBuffer, hAacEncoder->inputBufferSizePerChannel); } /* * Make bitstream public */ if ((outBufDesc != NULL) && (outBufDesc->numBufs >= 1)) { INT bsIdx = getBufDescIdx(outBufDesc, OUT_BITSTREAM_DATA); INT auIdx = getBufDescIdx(outBufDesc, OUT_AU_SIZES); for (i = 0, nBsBytes = 0; i < hAacEncoder->aacConfig.nSubFrames; i++) { nBsBytes += outBytes[i]; if (auIdx != -1) { ((INT *)outBufDesc->bufs[auIdx])[i] = outBytes[i]; } } if ((bsIdx != -1) && (outBufDesc->bufSizes[bsIdx] >= nBsBytes)) { FDKmemcpy(outBufDesc->bufs[bsIdx], hAacEncoder->outBuffer, sizeof(UCHAR) * nBsBytes); outargs->numOutBytes = nBsBytes; outargs->bitResState = FDKaacEnc_GetBitReservoirState(hAacEncoder->hAacEnc); } else { /* output buffer too small, can't write valid bitstream */ err = AACENC_ENCODE_ERROR; goto bail; } } bail: if (err == AACENC_ENCODE_ERROR) { /* All encoder modules have to be initialized */ hAacEncoder->InitFlags = AACENC_INIT_ALL; } return err; } static AAC_ENCODER_ERROR aacEncGetConf(HANDLE_AACENCODER hAacEncoder, UINT *size, UCHAR *confBuffer) { FDK_BITSTREAM tmpConf; UINT confType; UCHAR buf[64]; int err; /* Init bit buffer */ FDKinitBitStream(&tmpConf, buf, 64, 0, BS_WRITER); /* write conf in tmp buffer */ err = transportEnc_GetConf(hAacEncoder->hTpEnc, &hAacEncoder->coderConfig, &tmpConf, &confType); /* copy data to outbuffer: length in bytes */ FDKbyteAlign(&tmpConf, 0); /* Check buffer size */ if (FDKgetValidBits(&tmpConf) > ((*size) << 3)) return AAC_ENC_UNKNOWN; FDKfetchBuffer(&tmpConf, confBuffer, size); if (err != 0) return AAC_ENC_UNKNOWN; else return AAC_ENC_OK; } AACENC_ERROR aacEncGetLibInfo(LIB_INFO *info) { int i = 0; if (info == NULL) { return AACENC_INVALID_HANDLE; } FDK_toolsGetLibInfo(info); transportEnc_GetLibInfo(info); sbrEncoder_GetLibInfo(info); FDK_MpegsEnc_GetLibInfo(info); /* search for next free tab */ for (i = 0; i < FDK_MODULE_LAST; i++) { if (info[i].module_id == FDK_NONE) break; } if (i == FDK_MODULE_LAST) { return AACENC_INIT_ERROR; } info[i].module_id = FDK_AACENC; info[i].build_date = AACENCODER_LIB_BUILD_DATE; info[i].build_time = AACENCODER_LIB_BUILD_TIME; info[i].title = AACENCODER_LIB_TITLE; info[i].version = LIB_VERSION(AACENCODER_LIB_VL0, AACENCODER_LIB_VL1, AACENCODER_LIB_VL2); ; LIB_VERSION_STRING(&info[i]); /* Capability flags */ info[i].flags = 0 | CAPF_AAC_1024 | CAPF_AAC_LC | CAPF_AAC_512 | CAPF_AAC_480 | CAPF_AAC_DRC | CAPF_AAC_ELD_DOWNSCALE; /* End of flags */ return AACENC_OK; } AACENC_ERROR aacEncoder_SetParam(const HANDLE_AACENCODER hAacEncoder, const AACENC_PARAM param, const UINT value) { AACENC_ERROR err = AACENC_OK; USER_PARAM *settings = &hAacEncoder->extParam; /* check encoder handle */ if (hAacEncoder == NULL) { err = AACENC_INVALID_HANDLE; goto bail; } /* apply param value */ switch (param) { case AACENC_AOT: if (settings->userAOT != (AUDIO_OBJECT_TYPE)value) { /* check if AOT matches the allocated modules */ switch (value) { case AOT_PS: if (!(hAacEncoder->encoder_modis & (ENC_MODE_FLAG_PS))) { err = AACENC_INVALID_CONFIG; goto bail; } FDK_FALLTHROUGH; case AOT_SBR: case AOT_MP2_SBR: if (!(hAacEncoder->encoder_modis & (ENC_MODE_FLAG_SBR))) { err = AACENC_INVALID_CONFIG; goto bail; } FDK_FALLTHROUGH; case AOT_AAC_LC: case AOT_MP2_AAC_LC: case AOT_ER_AAC_LD: case AOT_ER_AAC_ELD: if (!(hAacEncoder->encoder_modis & (ENC_MODE_FLAG_AAC))) { err = AACENC_INVALID_CONFIG; goto bail; } break; default: err = AACENC_INVALID_CONFIG; goto bail; } /* switch value */ settings->userAOT = (AUDIO_OBJECT_TYPE)value; hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_STATES | AACENC_INIT_TRANSPORT; } break; case AACENC_BITRATE: if (settings->userBitrate != value) { settings->userBitrate = value; hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_TRANSPORT; } break; case AACENC_BITRATEMODE: if (settings->userBitrateMode != value) { switch (value) { case 0: case 1: case 2: case 3: case 4: case 5: settings->userBitrateMode = value; hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_TRANSPORT; break; default: err = AACENC_INVALID_CONFIG; break; } /* switch value */ } break; case AACENC_SAMPLERATE: if (settings->userSamplerate != value) { if (!((value == 8000) || (value == 11025) || (value == 12000) || (value == 16000) || (value == 22050) || (value == 24000) || (value == 32000) || (value == 44100) || (value == 48000) || (value == 64000) || (value == 88200) || (value == 96000))) { err = AACENC_INVALID_CONFIG; break; } settings->userSamplerate = value; hAacEncoder->nSamplesRead = 0; /* reset internal inputbuffer */ hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_STATES | AACENC_INIT_TRANSPORT; } break; case AACENC_CHANNELMODE: if (settings->userChannelMode != (CHANNEL_MODE)value) { if (((CHANNEL_MODE)value == MODE_212) && (NULL != hAacEncoder->hMpsEnc)) { settings->userChannelMode = (CHANNEL_MODE)value; settings->nChannels = 2; } else { const CHANNEL_MODE_CONFIG_TAB *pConfig = FDKaacEnc_GetChannelModeConfiguration((CHANNEL_MODE)value); if (pConfig == NULL) { err = AACENC_INVALID_CONFIG; break; } if ((pConfig->nElements > hAacEncoder->nMaxAacElements) || (pConfig->nChannelsEff > hAacEncoder->nMaxAacChannels)) { err = AACENC_INVALID_CONFIG; break; } settings->userChannelMode = (CHANNEL_MODE)value; settings->nChannels = pConfig->nChannels; } hAacEncoder->nSamplesRead = 0; /* reset internal inputbuffer */ hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_TRANSPORT; if (!((value >= 1) && (value <= 6))) { hAacEncoder->InitFlags |= AACENC_INIT_STATES; } } break; case AACENC_BANDWIDTH: if (settings->userBandwidth != value) { settings->userBandwidth = value; hAacEncoder->InitFlags |= AACENC_INIT_CONFIG; } break; case AACENC_CHANNELORDER: if (hAacEncoder->aacConfig.channelOrder != (CHANNEL_ORDER)value) { if (!((value == 0) || (value == 1) || (value == 2))) { err = AACENC_INVALID_CONFIG; break; } hAacEncoder->aacConfig.channelOrder = (CHANNEL_ORDER)value; hAacEncoder->nSamplesRead = 0; /* reset internal inputbuffer */ hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_STATES | AACENC_INIT_TRANSPORT; } break; case AACENC_AFTERBURNER: if (settings->userAfterburner != value) { if (!((value == 0) || (value == 1))) { err = AACENC_INVALID_CONFIG; break; } settings->userAfterburner = value; hAacEncoder->InitFlags |= AACENC_INIT_CONFIG; } break; case AACENC_GRANULE_LENGTH: if (settings->userFramelength != value) { switch (value) { case 1024: case 512: case 480: case 256: case 240: case 128: case 120: if ((value << 1) == 480 || (value << 1) == 512) { settings->userDownscaleFactor = 2; } else if ((value << 2) == 480 || (value << 2) == 512) { settings->userDownscaleFactor = 4; } settings->userFramelength = value; hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_TRANSPORT; break; default: err = AACENC_INVALID_CONFIG; break; } } break; case AACENC_SBR_RATIO: if (settings->userSbrRatio != value) { if (!((value == 0) || (value == 1) || (value == 2))) { err = AACENC_INVALID_CONFIG; break; } settings->userSbrRatio = value; hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_STATES | AACENC_INIT_TRANSPORT; } break; case AACENC_SBR_MODE: if ((settings->userSbrEnabled != value) && (NULL != hAacEncoder->hEnvEnc)) { settings->userSbrEnabled = value; hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_STATES | AACENC_INIT_TRANSPORT; } break; case AACENC_TRANSMUX: if (settings->userTpType != (TRANSPORT_TYPE)value) { TRANSPORT_TYPE type = (TRANSPORT_TYPE)value; UINT flags = hAacEncoder->CAPF_tpEnc; if (!(((type == TT_MP4_ADIF) && (flags & CAPF_ADIF)) || ((type == TT_MP4_ADTS) && (flags & CAPF_ADTS)) || ((type == TT_MP4_LATM_MCP0) && ((flags & CAPF_LATM) && (flags & CAPF_RAWPACKETS))) || ((type == TT_MP4_LATM_MCP1) && ((flags & CAPF_LATM) && (flags & CAPF_RAWPACKETS))) || ((type == TT_MP4_LOAS) && (flags & CAPF_LOAS)) || ((type == TT_MP4_RAW) && (flags & CAPF_RAWPACKETS)))) { err = AACENC_INVALID_CONFIG; break; } settings->userTpType = (TRANSPORT_TYPE)value; hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT; } break; case AACENC_SIGNALING_MODE: if (settings->userTpSignaling != value) { if (!((value == 0) || (value == 1) || (value == 2))) { err = AACENC_INVALID_CONFIG; break; } settings->userTpSignaling = value; hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT; } break; case AACENC_PROTECTION: if (settings->userTpProtection != value) { if (!((value == 0) || (value == 1))) { err = AACENC_INVALID_CONFIG; break; } settings->userTpProtection = value; hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT; } break; case AACENC_HEADER_PERIOD: if (settings->userTpHeaderPeriod != value) { if (!(((INT)value >= 0) && (value <= 255))) { err = AACENC_INVALID_CONFIG; break; } settings->userTpHeaderPeriod = value; hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT; } break; case AACENC_AUDIOMUXVER: if (settings->userTpAmxv != value) { if (!((value == 0) || (value == 1) || (value == 2))) { err = AACENC_INVALID_CONFIG; break; } settings->userTpAmxv = value; hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT; } break; case AACENC_TPSUBFRAMES: if (settings->userTpNsubFrames != value) { if (!((value >= 1) && (value <= 4))) { err = AACENC_INVALID_CONFIG; break; } settings->userTpNsubFrames = value; hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT; } break; case AACENC_ANCILLARY_BITRATE: if (settings->userAncDataRate != value) { settings->userAncDataRate = value; } break; case AACENC_CONTROL_STATE: if (hAacEncoder->InitFlags != value) { if (value & AACENC_RESET_INBUFFER) { hAacEncoder->nSamplesRead = 0; } hAacEncoder->InitFlags = value; } break; case AACENC_METADATA_MODE: if ((UINT)settings->userMetaDataMode != value) { if (!(((INT)value >= 0) && ((INT)value <= 3))) { err = AACENC_INVALID_CONFIG; break; } settings->userMetaDataMode = value; hAacEncoder->InitFlags |= AACENC_INIT_CONFIG; } break; case AACENC_PEAK_BITRATE: if (settings->userPeakBitrate != value) { settings->userPeakBitrate = value; hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_TRANSPORT; } break; default: err = AACENC_UNSUPPORTED_PARAMETER; break; } /* switch(param) */ bail: return err; } UINT aacEncoder_GetParam(const HANDLE_AACENCODER hAacEncoder, const AACENC_PARAM param) { UINT value = 0; USER_PARAM *settings = &hAacEncoder->extParam; /* check encoder handle */ if (hAacEncoder == NULL) { goto bail; } /* apply param value */ switch (param) { case AACENC_AOT: value = (UINT)hAacEncoder->aacConfig.audioObjectType; break; case AACENC_BITRATE: switch (hAacEncoder->aacConfig.bitrateMode) { case AACENC_BR_MODE_CBR: value = (UINT)hAacEncoder->aacConfig.bitRate; break; default: value = (UINT)-1; } break; case AACENC_BITRATEMODE: value = (UINT)((hAacEncoder->aacConfig.bitrateMode != AACENC_BR_MODE_FF) ? hAacEncoder->aacConfig.bitrateMode : AACENC_BR_MODE_CBR); break; case AACENC_SAMPLERATE: value = (UINT)hAacEncoder->coderConfig.extSamplingRate; break; case AACENC_CHANNELMODE: if ((MODE_1 == hAacEncoder->aacConfig.channelMode) && (hAacEncoder->aacConfig.syntaxFlags & AC_LD_MPS)) { value = MODE_212; } else { value = (UINT)hAacEncoder->aacConfig.channelMode; } break; case AACENC_BANDWIDTH: value = (UINT)hAacEncoder->aacConfig.bandWidth; break; case AACENC_CHANNELORDER: value = (UINT)hAacEncoder->aacConfig.channelOrder; break; case AACENC_AFTERBURNER: value = (UINT)hAacEncoder->aacConfig.useRequant; break; case AACENC_GRANULE_LENGTH: value = (UINT)hAacEncoder->aacConfig.framelength; break; case AACENC_SBR_RATIO: value = isSbrActive(&hAacEncoder->aacConfig) ? hAacEncoder->aacConfig.sbrRatio : 0; break; case AACENC_SBR_MODE: value = (UINT)(hAacEncoder->aacConfig.syntaxFlags & AC_SBR_PRESENT) ? 1 : 0; break; case AACENC_TRANSMUX: value = (UINT)settings->userTpType; break; case AACENC_SIGNALING_MODE: value = (UINT)getSbrSignalingMode( hAacEncoder->aacConfig.audioObjectType, settings->userTpType, settings->userTpSignaling, hAacEncoder->aacConfig.sbrRatio); break; case AACENC_PROTECTION: value = (UINT)settings->userTpProtection; break; case AACENC_HEADER_PERIOD: value = (UINT)hAacEncoder->coderConfig.headerPeriod; break; case AACENC_AUDIOMUXVER: value = (UINT)hAacEncoder->aacConfig.audioMuxVersion; break; case AACENC_TPSUBFRAMES: value = (UINT)settings->userTpNsubFrames; break; case AACENC_ANCILLARY_BITRATE: value = (UINT)hAacEncoder->aacConfig.anc_Rate; break; case AACENC_CONTROL_STATE: value = (UINT)hAacEncoder->InitFlags; break; case AACENC_METADATA_MODE: value = (hAacEncoder->metaDataAllowed == 0) ? 0 : (UINT)settings->userMetaDataMode; break; case AACENC_PEAK_BITRATE: value = (UINT)-1; /* peak bitrate parameter is meaningless */ if (((INT)hAacEncoder->extParam.userPeakBitrate != -1)) { value = (UINT)(fMax((INT)hAacEncoder->extParam.userPeakBitrate, hAacEncoder->aacConfig .bitRate)); /* peak bitrate parameter is in use */ } break; default: // err = MPS_INVALID_PARAMETER; break; } /* switch(param) */ bail: return value; } AACENC_ERROR aacEncInfo(const HANDLE_AACENCODER hAacEncoder, AACENC_InfoStruct *pInfo) { AACENC_ERROR err = AACENC_OK; FDKmemclear(pInfo, sizeof(AACENC_InfoStruct)); pInfo->confSize = 64; /* pre-initialize */ pInfo->maxOutBufBytes = ((hAacEncoder->nMaxAacChannels * 6144) + 7) >> 3; pInfo->maxAncBytes = hAacEncoder->aacConfig.maxAncBytesPerAU; pInfo->inBufFillLevel = hAacEncoder->nSamplesRead / hAacEncoder->extParam.nChannels; pInfo->inputChannels = hAacEncoder->extParam.nChannels; pInfo->frameLength = hAacEncoder->nSamplesToRead / hAacEncoder->extParam.nChannels; pInfo->nDelay = hAacEncoder->nDelay; pInfo->nDelayCore = hAacEncoder->nDelayCore; /* Get encoder configuration */ if (aacEncGetConf(hAacEncoder, &pInfo->confSize, &pInfo->confBuf[0]) != AAC_ENC_OK) { err = AACENC_INIT_ERROR; goto bail; } bail: return err; } fdk-aac-2.0.2/libAACenc/src/aacenc_pns.cpp000066400000000000000000000465761404225310700201470ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Lohwasser Description: pns.c *******************************************************************************/ #include "aacenc_pns.h" #include "psy_data.h" #include "pnsparam.h" #include "noisedet.h" #include "bit_cnt.h" #include "interface.h" /* minCorrelationEnergy = (1.0e-10f)^2 ~ 2^-67 = 2^-47 * 2^-20 */ static const FIXP_DBL minCorrelationEnergy = FL2FXCONST_DBL(0.0); /* FL2FXCONST_DBL((float)FDKpow(2.0,-47)); */ /* noiseCorrelationThresh = 0.6^2 */ static const FIXP_DBL noiseCorrelationThresh = FL2FXCONST_DBL(0.36); static void FDKaacEnc_FDKaacEnc_noiseDetection( PNS_CONFIG *pnsConf, PNS_DATA *pnsData, const INT sfbActive, const INT *sfbOffset, INT tnsOrder, INT tnsPredictionGain, INT tnsActive, FIXP_DBL *mdctSpectrum, INT *sfbMaxScaleSpec, FIXP_SGL *sfbtonality); static void FDKaacEnc_CalcNoiseNrgs(const INT sfbActive, INT *pnsFlag, FIXP_DBL *sfbEnergyLdData, INT *noiseNrg); /***************************************************************************** functionname: initPnsConfiguration description: fill pnsConf with pns parameters returns: error status input: PNS Config struct (modified) bitrate, samplerate, usePns, number of sfb's, pointer to sfb offset output: error code *****************************************************************************/ AAC_ENCODER_ERROR FDKaacEnc_InitPnsConfiguration( PNS_CONFIG *pnsConf, INT bitRate, INT sampleRate, INT usePns, INT sfbCnt, const INT *sfbOffset, const INT numChan, const INT isLC) { AAC_ENCODER_ERROR ErrorStatus; /* init noise detection */ ErrorStatus = FDKaacEnc_GetPnsParam(&pnsConf->np, bitRate, sampleRate, sfbCnt, sfbOffset, &usePns, numChan, isLC); if (ErrorStatus != AAC_ENC_OK) return ErrorStatus; pnsConf->minCorrelationEnergy = minCorrelationEnergy; pnsConf->noiseCorrelationThresh = noiseCorrelationThresh; pnsConf->usePns = usePns; return AAC_ENC_OK; } /***************************************************************************** functionname: FDKaacEnc_PnsDetect description: do decision, if PNS shall used or not returns: input: pns config structure pns data structure (modified), lastWindowSequence (long or short blocks) sfbActive pointer to Sfb Energy, Threshold, Offset pointer to mdct Spectrum length of each group pointer to tonality calculated in chaosmeasure tns order and prediction gain calculated noiseNrg at active PNS output: pnsFlag in pns data structure *****************************************************************************/ void FDKaacEnc_PnsDetect(PNS_CONFIG *pnsConf, PNS_DATA *pnsData, const INT lastWindowSequence, const INT sfbActive, const INT maxSfbPerGroup, FIXP_DBL *sfbThresholdLdData, const INT *sfbOffset, FIXP_DBL *mdctSpectrum, INT *sfbMaxScaleSpec, FIXP_SGL *sfbtonality, INT tnsOrder, INT tnsPredictionGain, INT tnsActive, FIXP_DBL *sfbEnergyLdData, INT *noiseNrg) { int sfb; int startNoiseSfb; /* Reset pns info. */ FDKmemclear(pnsData->pnsFlag, sizeof(pnsData->pnsFlag)); for (sfb = 0; sfb < MAX_GROUPED_SFB; sfb++) { noiseNrg[sfb] = NO_NOISE_PNS; } /* Disable PNS and skip detection in certain cases. */ if (pnsConf->usePns == 0) { return; } else { /* AAC - LC core encoder */ if ((pnsConf->np.detectionAlgorithmFlags & IS_LOW_COMPLEXITY) && (lastWindowSequence == SHORT_WINDOW)) { return; } /* AAC - (E)LD core encoder */ if (!(pnsConf->np.detectionAlgorithmFlags & IS_LOW_COMPLEXITY) && (pnsConf->np.detectionAlgorithmFlags & JUST_LONG_WINDOW) && (lastWindowSequence != LONG_WINDOW)) { return; } } /* call noise detection */ FDKaacEnc_FDKaacEnc_noiseDetection( pnsConf, pnsData, sfbActive, sfbOffset, tnsOrder, tnsPredictionGain, tnsActive, mdctSpectrum, sfbMaxScaleSpec, sfbtonality); /* set startNoiseSfb (long) */ startNoiseSfb = pnsConf->np.startSfb; /* Set noise substitution status */ for (sfb = 0; sfb < sfbActive; sfb++) { /* No PNS below startNoiseSfb */ if (sfb < startNoiseSfb) { pnsData->pnsFlag[sfb] = 0; continue; } /* do noise substitution if fuzzy measure is high enough sfb freq > minimum sfb freq signal in coder band is not masked */ if ((pnsData->noiseFuzzyMeasure[sfb] > FL2FXCONST_SGL(0.5)) && ((sfbThresholdLdData[sfb] + FL2FXCONST_DBL(0.5849625f / 64.0f)) /* thr * 1.5 = thrLd +ld(1.5)/64 */ < sfbEnergyLdData[sfb])) { /* mark in psyout flag array that we will code this band with PNS */ pnsData->pnsFlag[sfb] = 1; /* PNS_ON */ } else { pnsData->pnsFlag[sfb] = 0; /* PNS_OFF */ } /* no PNS if LTP is active */ } /* avoid PNS holes */ if ((pnsData->noiseFuzzyMeasure[0] > FL2FXCONST_SGL(0.5f)) && (pnsData->pnsFlag[1])) { pnsData->pnsFlag[0] = 1; } for (sfb = 1; sfb < maxSfbPerGroup - 1; sfb++) { if ((pnsData->noiseFuzzyMeasure[sfb] > pnsConf->np.gapFillThr) && (pnsData->pnsFlag[sfb - 1]) && (pnsData->pnsFlag[sfb + 1])) { pnsData->pnsFlag[sfb] = 1; } } if (maxSfbPerGroup > 0) { /* avoid PNS hole */ if ((pnsData->noiseFuzzyMeasure[maxSfbPerGroup - 1] > pnsConf->np.gapFillThr) && (pnsData->pnsFlag[maxSfbPerGroup - 2])) { pnsData->pnsFlag[maxSfbPerGroup - 1] = 1; } /* avoid single PNS band */ if (pnsData->pnsFlag[maxSfbPerGroup - 2] == 0) { pnsData->pnsFlag[maxSfbPerGroup - 1] = 0; } } /* avoid single PNS bands */ if (pnsData->pnsFlag[1] == 0) { pnsData->pnsFlag[0] = 0; } for (sfb = 1; sfb < maxSfbPerGroup - 1; sfb++) { if ((pnsData->pnsFlag[sfb - 1] == 0) && (pnsData->pnsFlag[sfb + 1] == 0)) { pnsData->pnsFlag[sfb] = 0; } } /* calculate noiseNrg's */ FDKaacEnc_CalcNoiseNrgs(sfbActive, pnsData->pnsFlag, sfbEnergyLdData, noiseNrg); } /***************************************************************************** functionname:FDKaacEnc_FDKaacEnc_noiseDetection description: wrapper for noisedet.c returns: input: pns config structure pns data structure (modified), sfbActive tns order and prediction gain pointer to mdct Spectrumand Sfb Energy pointer to Sfb tonality output: noiseFuzzyMeasure in structure pnsData flags tonal / nontonal *****************************************************************************/ static void FDKaacEnc_FDKaacEnc_noiseDetection( PNS_CONFIG *pnsConf, PNS_DATA *pnsData, const INT sfbActive, const INT *sfbOffset, int tnsOrder, INT tnsPredictionGain, INT tnsActive, FIXP_DBL *mdctSpectrum, INT *sfbMaxScaleSpec, FIXP_SGL *sfbtonality) { INT condition = TRUE; if (!(pnsConf->np.detectionAlgorithmFlags & IS_LOW_COMPLEXITY)) { condition = (tnsOrder > 3); } /* no PNS if heavy TNS activity clear pnsData->noiseFuzzyMeasure */ if ((pnsConf->np.detectionAlgorithmFlags & USE_TNS_GAIN_THR) && (tnsPredictionGain >= pnsConf->np.tnsGainThreshold) && condition && !((pnsConf->np.detectionAlgorithmFlags & USE_TNS_PNS) && (tnsPredictionGain >= pnsConf->np.tnsPNSGainThreshold) && (tnsActive))) { /* clear all noiseFuzzyMeasure */ FDKmemclear(pnsData->noiseFuzzyMeasure, sfbActive * sizeof(FIXP_SGL)); } else { /* call noise detection, output in pnsData->noiseFuzzyMeasure, use real mdct spectral data */ FDKaacEnc_noiseDetect(mdctSpectrum, sfbMaxScaleSpec, sfbActive, sfbOffset, pnsData->noiseFuzzyMeasure, &pnsConf->np, sfbtonality); } } /***************************************************************************** functionname:FDKaacEnc_CalcNoiseNrgs description: Calculate the NoiseNrg's returns: input: sfbActive if pnsFlag calculate NoiseNrg pointer to sfbEnergy and groupLen pointer to noiseNrg (modified) output: noiseNrg's in pnsFlaged sfb's *****************************************************************************/ static void FDKaacEnc_CalcNoiseNrgs(const INT sfbActive, INT *RESTRICT pnsFlag, FIXP_DBL *RESTRICT sfbEnergyLdData, INT *RESTRICT noiseNrg) { int sfb; INT tmp = (-LOG_NORM_PCM) << 2; for (sfb = 0; sfb < sfbActive; sfb++) { if (pnsFlag[sfb]) { INT nrg = (-sfbEnergyLdData[sfb] + FL2FXCONST_DBL(0.5f / 64.0f)) >> (DFRACT_BITS - 1 - 7); noiseNrg[sfb] = tmp - nrg; } } } /***************************************************************************** functionname:FDKaacEnc_CodePnsChannel description: Execute pns decission returns: input: sfbActive pns config structure use PNS if pnsFlag pointer to Sfb Energy, noiseNrg, Threshold output: set sfbThreshold high to code pe with 0, noiseNrg marks flag for pns coding *****************************************************************************/ void FDKaacEnc_CodePnsChannel(const INT sfbActive, PNS_CONFIG *pnsConf, INT *RESTRICT pnsFlag, FIXP_DBL *RESTRICT sfbEnergyLdData, INT *RESTRICT noiseNrg, FIXP_DBL *RESTRICT sfbThresholdLdData) { INT sfb; INT lastiNoiseEnergy = 0; INT firstPNSband = 1; /* TRUE for first PNS-coded band */ /* no PNS */ if (!pnsConf->usePns) { for (sfb = 0; sfb < sfbActive; sfb++) { /* no PNS coding */ noiseNrg[sfb] = NO_NOISE_PNS; } return; } /* code PNS */ for (sfb = 0; sfb < sfbActive; sfb++) { if (pnsFlag[sfb]) { /* high sfbThreshold causes pe = 0 */ if (noiseNrg[sfb] != NO_NOISE_PNS) sfbThresholdLdData[sfb] = sfbEnergyLdData[sfb] + FL2FXCONST_DBL(1.0f / LD_DATA_SCALING); /* set noiseNrg in valid region */ if (!firstPNSband) { INT deltaiNoiseEnergy = noiseNrg[sfb] - lastiNoiseEnergy; if (deltaiNoiseEnergy > CODE_BOOK_PNS_LAV) noiseNrg[sfb] -= deltaiNoiseEnergy - CODE_BOOK_PNS_LAV; else if (deltaiNoiseEnergy < -CODE_BOOK_PNS_LAV) noiseNrg[sfb] -= deltaiNoiseEnergy + CODE_BOOK_PNS_LAV; } else { firstPNSband = 0; } lastiNoiseEnergy = noiseNrg[sfb]; } else { /* no PNS coding */ noiseNrg[sfb] = NO_NOISE_PNS; } } } /***************************************************************************** functionname:FDKaacEnc_PreProcessPnsChannelPair description: Calculate the correlation of noise in a channel pair returns: input: sfbActive pointer to sfb energies left, right and mid channel pns config structure pns data structure left and right (modified) output: noiseEnergyCorrelation in pns data structure *****************************************************************************/ void FDKaacEnc_PreProcessPnsChannelPair( const INT sfbActive, FIXP_DBL *RESTRICT sfbEnergyLeft, FIXP_DBL *RESTRICT sfbEnergyRight, FIXP_DBL *RESTRICT sfbEnergyLeftLD, FIXP_DBL *RESTRICT sfbEnergyRightLD, FIXP_DBL *RESTRICT sfbEnergyMid, PNS_CONFIG *RESTRICT pnsConf, PNS_DATA *pnsDataLeft, PNS_DATA *pnsDataRight) { INT sfb; FIXP_DBL ccf; if (!pnsConf->usePns) return; FIXP_DBL *RESTRICT pNoiseEnergyCorrelationL = pnsDataLeft->noiseEnergyCorrelation; FIXP_DBL *RESTRICT pNoiseEnergyCorrelationR = pnsDataRight->noiseEnergyCorrelation; for (sfb = 0; sfb < sfbActive; sfb++) { FIXP_DBL quot = (sfbEnergyLeftLD[sfb] >> 1) + (sfbEnergyRightLD[sfb] >> 1); if (quot < FL2FXCONST_DBL(-32.0f / (float)LD_DATA_SCALING)) ccf = FL2FXCONST_DBL(0.0f); else { FIXP_DBL accu = sfbEnergyMid[sfb] - (((sfbEnergyLeft[sfb] >> 1) + (sfbEnergyRight[sfb] >> 1)) >> 1); INT sign = (accu < FL2FXCONST_DBL(0.0f)) ? 1 : 0; accu = fixp_abs(accu); ccf = CalcLdData(accu) + FL2FXCONST_DBL((float)1.0f / (float)LD_DATA_SCALING) - quot; /* ld(accu*2) = ld(accu) + 1 */ ccf = (ccf >= FL2FXCONST_DBL(0.0)) ? ((FIXP_DBL)MAXVAL_DBL) : (sign) ? -CalcInvLdData(ccf) : CalcInvLdData(ccf); } pNoiseEnergyCorrelationL[sfb] = ccf; pNoiseEnergyCorrelationR[sfb] = ccf; } } /***************************************************************************** functionname:FDKaacEnc_PostProcessPnsChannelPair description: if PNS used at left and right channel, use msMask to flag correlation returns: input: sfbActive pns config structure pns data structure left and right (modified) pointer to msMask, flags correlation by pns coding (modified) Digest of MS coding output: pnsFlag in pns data structure, msFlag in msMask (flags correlation) *****************************************************************************/ void FDKaacEnc_PostProcessPnsChannelPair(const INT sfbActive, PNS_CONFIG *pnsConf, PNS_DATA *pnsDataLeft, PNS_DATA *pnsDataRight, INT *RESTRICT msMask, INT *msDigest) { INT sfb; if (!pnsConf->usePns) return; for (sfb = 0; sfb < sfbActive; sfb++) { /* MS post processing */ if (msMask[sfb]) { if ((pnsDataLeft->pnsFlag[sfb]) && (pnsDataRight->pnsFlag[sfb])) { /* AAC only: Standard */ /* do this to avoid ms flags in layers that should not have it */ if (pnsDataLeft->noiseEnergyCorrelation[sfb] <= pnsConf->noiseCorrelationThresh) { msMask[sfb] = 0; *msDigest = MS_SOME; } } else { /* No PNS coding */ pnsDataLeft->pnsFlag[sfb] = 0; pnsDataRight->pnsFlag[sfb] = 0; } } /* Use MS flag to signal noise correlation if pns is active in both channels */ if ((pnsDataLeft->pnsFlag[sfb]) && (pnsDataRight->pnsFlag[sfb])) { if (pnsDataLeft->noiseEnergyCorrelation[sfb] > pnsConf->noiseCorrelationThresh) { msMask[sfb] = 1; *msDigest = MS_SOME; } } } } fdk-aac-2.0.2/libAACenc/src/aacenc_pns.h000066400000000000000000000122251404225310700175740ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Lohwasser Description: pns.h *******************************************************************************/ #ifndef AACENC_PNS_H #define AACENC_PNS_H #include "common_fix.h" #include "pnsparam.h" #define NO_NOISE_PNS FDK_INT_MIN typedef struct { NOISEPARAMS np; FIXP_DBL minCorrelationEnergy; FIXP_DBL noiseCorrelationThresh; INT usePns; } PNS_CONFIG; typedef struct { FIXP_SGL noiseFuzzyMeasure[MAX_GROUPED_SFB]; FIXP_DBL noiseEnergyCorrelation[MAX_GROUPED_SFB]; INT pnsFlag[MAX_GROUPED_SFB]; } PNS_DATA; #endif /* AACENC_PNS_H */ fdk-aac-2.0.2/libAACenc/src/aacenc_tns.cpp000066400000000000000000001250101404225310700201300ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): Alex Groeschel, Tobias Chalupka Description: Temporal noise shaping *******************************************************************************/ #include "aacenc_tns.h" #include "psy_const.h" #include "psy_configuration.h" #include "tns_func.h" #include "aacEnc_rom.h" #include "aacenc_tns.h" #include "FDK_lpc.h" #define FILTER_DIRECTION 0 /* 0 = up, 1 = down */ static const FIXP_DBL acfWindowLong[12 + 3 + 1] = { 0x7fffffff, 0x7fb80000, 0x7ee00000, 0x7d780000, 0x7b800000, 0x78f80000, 0x75e00000, 0x72380000, 0x6e000000, 0x69380000, 0x63e00000, 0x5df80000, 0x57800000, 0x50780000, 0x48e00000, 0x40b80000}; static const FIXP_DBL acfWindowShort[4 + 3 + 1] = { 0x7fffffff, 0x7e000000, 0x78000000, 0x6e000000, 0x60000000, 0x4e000000, 0x38000000, 0x1e000000}; typedef struct { INT bitRateFrom[2]; /* noneSbr=0, useSbr=1 */ INT bitRateTo[2]; /* noneSbr=0, useSbr=1 */ TNS_PARAMETER_TABULATED paramTab[2]; /* mono=0, stereo=1 */ } TNS_INFO_TAB; #define TNS_TIMERES_SCALE (1) #define FL2_TIMERES_FIX(a) (FL2FXCONST_DBL(a / (float)(1 << TNS_TIMERES_SCALE))) static const TNS_INFO_TAB tnsInfoTab[] = { {{16000, 13500}, {32000, 28000}, {{{1, 1}, {1437, 1500}, {1400, 600}, {12, 12}, {FILTER_DIRECTION, FILTER_DIRECTION}, {3, 1}, {FL2_TIMERES_FIX(0.4f), FL2_TIMERES_FIX(1.2f)}, 1}, {{1, 1}, {1437, 1500}, {1400, 600}, {12, 12}, {FILTER_DIRECTION, FILTER_DIRECTION}, {3, 1}, {FL2_TIMERES_FIX(0.4f), FL2_TIMERES_FIX(1.2f)}, 1}}}, {{32001, 28001}, {60000, 52000}, {{{1, 1}, {1437, 1500}, {1400, 600}, {12, 10}, {FILTER_DIRECTION, FILTER_DIRECTION}, {3, 1}, {FL2_TIMERES_FIX(0.4f), FL2_TIMERES_FIX(1.0f)}, 1}, {{1, 1}, {1437, 1500}, {1400, 600}, {12, 10}, {FILTER_DIRECTION, FILTER_DIRECTION}, {3, 1}, {FL2_TIMERES_FIX(0.4f), FL2_TIMERES_FIX(1.0f)}, 1}}}, {{60001, 52001}, {384000, 384000}, {{{1, 1}, {1437, 1500}, {1400, 600}, {12, 8}, {FILTER_DIRECTION, FILTER_DIRECTION}, {3, 1}, {FL2_TIMERES_FIX(0.4f), FL2_TIMERES_FIX(1.0f)}, 1}, {{1, 1}, {1437, 1500}, {1400, 600}, {12, 8}, {FILTER_DIRECTION, FILTER_DIRECTION}, {3, 1}, {FL2_TIMERES_FIX(0.4f), FL2_TIMERES_FIX(1.0f)}, 1}}}}; typedef struct { INT samplingRate; SCHAR maxBands[2]; /* long=0; short=1 */ } TNS_MAX_TAB_ENTRY; static const TNS_MAX_TAB_ENTRY tnsMaxBandsTab1024[] = { {96000, {31, 9}}, {88200, {31, 9}}, {64000, {34, 10}}, {48000, {40, 14}}, {44100, {42, 14}}, {32000, {51, 14}}, {24000, {46, 14}}, {22050, {46, 14}}, {16000, {42, 14}}, {12000, {42, 14}}, {11025, {42, 14}}, {8000, {39, 14}}}; static const TNS_MAX_TAB_ENTRY tnsMaxBandsTab120[] = { {48000, {12, -1}}, /* 48000 */ {44100, {12, -1}}, /* 44100 */ {32000, {15, -1}}, /* 32000 */ {24000, {15, -1}}, /* 24000 */ {22050, {15, -1}} /* 22050 */ }; static const TNS_MAX_TAB_ENTRY tnsMaxBandsTab128[] = { {48000, {12, -1}}, /* 48000 */ {44100, {12, -1}}, /* 44100 */ {32000, {15, -1}}, /* 32000 */ {24000, {15, -1}}, /* 24000 */ {22050, {15, -1}} /* 22050 */ }; static const TNS_MAX_TAB_ENTRY tnsMaxBandsTab240[] = { {96000, {22, -1}}, /* 96000 */ {48000, {22, -1}}, /* 48000 */ {44100, {22, -1}}, /* 44100 */ {32000, {21, -1}}, /* 32000 */ {24000, {21, -1}}, /* 24000 */ {22050, {21, -1}} /* 22050 */ }; static const TNS_MAX_TAB_ENTRY tnsMaxBandsTab256[] = { {96000, {25, -1}}, /* 96000 */ {48000, {25, -1}}, /* 48000 */ {44100, {25, -1}}, /* 44100 */ {32000, {24, -1}}, /* 32000 */ {24000, {24, -1}}, /* 24000 */ {22050, {24, -1}} /* 22050 */ }; static const TNS_MAX_TAB_ENTRY tnsMaxBandsTab480[] = {{48000, {31, -1}}, {44100, {32, -1}}, {32000, {37, -1}}, {24000, {30, -1}}, {22050, {30, -1}}}; static const TNS_MAX_TAB_ENTRY tnsMaxBandsTab512[] = {{48000, {31, -1}}, {44100, {32, -1}}, {32000, {37, -1}}, {24000, {31, -1}}, {22050, {31, -1}}}; static void FDKaacEnc_Parcor2Index(const FIXP_LPC *parcor, INT *RESTRICT index, const INT order, const INT bitsPerCoeff); static void FDKaacEnc_Index2Parcor(const INT *index, FIXP_LPC *RESTRICT parcor, const INT order, const INT bitsPerCoeff); static void FDKaacEnc_CalcGaussWindow(FIXP_DBL *win, const int winSize, const INT samplingRate, const INT transformResolution, const FIXP_DBL timeResolution, const INT timeResolution_e); static const TNS_PARAMETER_TABULATED *FDKaacEnc_GetTnsParam(const INT bitRate, const INT channels, const INT sbrLd) { int i; const TNS_PARAMETER_TABULATED *tnsConfigTab = NULL; for (i = 0; i < (int)(sizeof(tnsInfoTab) / sizeof(TNS_INFO_TAB)); i++) { if ((bitRate >= tnsInfoTab[i].bitRateFrom[sbrLd ? 1 : 0]) && bitRate <= tnsInfoTab[i].bitRateTo[sbrLd ? 1 : 0]) { tnsConfigTab = &tnsInfoTab[i].paramTab[(channels == 1) ? 0 : 1]; } } return tnsConfigTab; } static INT getTnsMaxBands(const INT sampleRate, const INT granuleLength, const INT isShortBlock) { int i; INT numBands = -1; const TNS_MAX_TAB_ENTRY *pMaxBandsTab = NULL; int maxBandsTabSize = 0; switch (granuleLength) { case 960: case 1024: pMaxBandsTab = tnsMaxBandsTab1024; maxBandsTabSize = sizeof(tnsMaxBandsTab1024) / sizeof(TNS_MAX_TAB_ENTRY); break; case 120: pMaxBandsTab = tnsMaxBandsTab120; maxBandsTabSize = sizeof(tnsMaxBandsTab120) / sizeof(TNS_MAX_TAB_ENTRY); break; case 128: pMaxBandsTab = tnsMaxBandsTab128; maxBandsTabSize = sizeof(tnsMaxBandsTab128) / sizeof(TNS_MAX_TAB_ENTRY); break; case 240: pMaxBandsTab = tnsMaxBandsTab240; maxBandsTabSize = sizeof(tnsMaxBandsTab240) / sizeof(TNS_MAX_TAB_ENTRY); break; case 256: pMaxBandsTab = tnsMaxBandsTab256; maxBandsTabSize = sizeof(tnsMaxBandsTab256) / sizeof(TNS_MAX_TAB_ENTRY); break; case 480: pMaxBandsTab = tnsMaxBandsTab480; maxBandsTabSize = sizeof(tnsMaxBandsTab480) / sizeof(TNS_MAX_TAB_ENTRY); break; case 512: pMaxBandsTab = tnsMaxBandsTab512; maxBandsTabSize = sizeof(tnsMaxBandsTab512) / sizeof(TNS_MAX_TAB_ENTRY); break; default: numBands = -1; } if (pMaxBandsTab != NULL) { for (i = 0; i < maxBandsTabSize; i++) { numBands = pMaxBandsTab[i].maxBands[(!isShortBlock) ? 0 : 1]; if (sampleRate >= pMaxBandsTab[i].samplingRate) { break; } } } return numBands; } /***************************************************************************/ /*! \brief FDKaacEnc_FreqToBandWidthRounding Returns index of nearest band border \param frequency \param sampling frequency \param total number of bands \param pointer to table of band borders \return band border ****************************************************************************/ INT FDKaacEnc_FreqToBandWidthRounding(const INT freq, const INT fs, const INT numOfBands, const INT *bandStartOffset) { INT lineNumber, band; /* assert(freq >= 0); */ lineNumber = (freq * bandStartOffset[numOfBands] * 4 / fs + 1) / 2; /* freq > fs/2 */ if (lineNumber >= bandStartOffset[numOfBands]) return numOfBands; /* find band the line number lies in */ for (band = 0; band < numOfBands; band++) { if (bandStartOffset[band + 1] > lineNumber) break; } /* round to nearest band border */ if (lineNumber - bandStartOffset[band] > bandStartOffset[band + 1] - lineNumber) { band++; } return (band); } /***************************************************************************** functionname: FDKaacEnc_InitTnsConfiguration description: fill TNS_CONFIG structure with sensible content returns: input: bitrate, samplerate, number of channels, blocktype (long or short), TNS Config struct (modified), psy config struct, tns active flag output: *****************************************************************************/ AAC_ENCODER_ERROR FDKaacEnc_InitTnsConfiguration( INT bitRate, INT sampleRate, INT channels, INT blockType, INT granuleLength, INT isLowDelay, INT ldSbrPresent, TNS_CONFIG *tC, PSY_CONFIGURATION *pC, INT active, INT useTnsPeak) { int i; // float acfTimeRes = (blockType == SHORT_WINDOW) ? 0.125f : 0.046875f; if (channels <= 0) return (AAC_ENCODER_ERROR)1; tC->isLowDelay = isLowDelay; /* initialize TNS filter flag, order, and coefficient resolution (in bits per * coeff) */ tC->tnsActive = (active) ? TRUE : FALSE; tC->maxOrder = (blockType == SHORT_WINDOW) ? 5 : 12; /* maximum: 7, 20 */ if (bitRate < 16000) tC->maxOrder -= 2; tC->coefRes = (blockType == SHORT_WINDOW) ? 3 : 4; /* LPC stop line: highest MDCT line to be coded, but do not go beyond * TNS_MAX_BANDS! */ tC->lpcStopBand = getTnsMaxBands(sampleRate, granuleLength, (blockType == SHORT_WINDOW) ? 1 : 0); if (tC->lpcStopBand < 0) { return (AAC_ENCODER_ERROR)1; } tC->lpcStopBand = fMin(tC->lpcStopBand, pC->sfbActive); tC->lpcStopLine = pC->sfbOffset[tC->lpcStopBand]; switch (granuleLength) { case 960: case 1024: /* TNS start line: skip lower MDCT lines to prevent artifacts due to * filter mismatch */ if (blockType == SHORT_WINDOW) { tC->lpcStartBand[LOFILT] = 0; } else { tC->lpcStartBand[LOFILT] = (sampleRate < 9391) ? 2 : ((sampleRate < 18783) ? 4 : 8); } tC->lpcStartLine[LOFILT] = pC->sfbOffset[tC->lpcStartBand[LOFILT]]; i = tC->lpcStopBand; while (pC->sfbOffset[i] > (tC->lpcStartLine[LOFILT] + (tC->lpcStopLine - tC->lpcStartLine[LOFILT]) / 4)) i--; tC->lpcStartBand[HIFILT] = i; tC->lpcStartLine[HIFILT] = pC->sfbOffset[i]; tC->confTab.threshOn[HIFILT] = 1437; tC->confTab.threshOn[LOFILT] = 1500; tC->confTab.tnsLimitOrder[HIFILT] = tC->maxOrder; tC->confTab.tnsLimitOrder[LOFILT] = fMax(0, tC->maxOrder - 7); tC->confTab.tnsFilterDirection[HIFILT] = FILTER_DIRECTION; tC->confTab.tnsFilterDirection[LOFILT] = FILTER_DIRECTION; tC->confTab.acfSplit[HIFILT] = -1; /* signal Merged4to2QuartersAutoCorrelation in FDKaacEnc_MergedAutoCorrelation*/ tC->confTab.acfSplit[LOFILT] = -1; /* signal Merged4to2QuartersAutoCorrelation in FDKaacEnc_MergedAutoCorrelation */ tC->confTab.filterEnabled[HIFILT] = 1; tC->confTab.filterEnabled[LOFILT] = 1; tC->confTab.seperateFiltersAllowed = 1; /* compute autocorrelation window based on maximum filter order for given * block type */ /* for (i = 0; i <= tC->maxOrder + 3; i++) { float acfWinTemp = acfTimeRes * i; acfWindow[i] = FL2FXCONST_DBL(1.0f - acfWinTemp * acfWinTemp); } */ if (blockType == SHORT_WINDOW) { FDKmemcpy(tC->acfWindow[HIFILT], acfWindowShort, fMin((LONG)sizeof(acfWindowShort), (LONG)sizeof(tC->acfWindow[HIFILT]))); FDKmemcpy(tC->acfWindow[LOFILT], acfWindowShort, fMin((LONG)sizeof(acfWindowShort), (LONG)sizeof(tC->acfWindow[HIFILT]))); } else { FDKmemcpy(tC->acfWindow[HIFILT], acfWindowLong, fMin((LONG)sizeof(acfWindowLong), (LONG)sizeof(tC->acfWindow[HIFILT]))); FDKmemcpy(tC->acfWindow[LOFILT], acfWindowLong, fMin((LONG)sizeof(acfWindowLong), (LONG)sizeof(tC->acfWindow[HIFILT]))); } break; case 480: case 512: { const TNS_PARAMETER_TABULATED *pCfg = FDKaacEnc_GetTnsParam(bitRate, channels, ldSbrPresent); if (pCfg != NULL) { FDKmemcpy(&(tC->confTab), pCfg, sizeof(tC->confTab)); tC->lpcStartBand[HIFILT] = FDKaacEnc_FreqToBandWidthRounding( pCfg->filterStartFreq[HIFILT], sampleRate, pC->sfbCnt, pC->sfbOffset); tC->lpcStartLine[HIFILT] = pC->sfbOffset[tC->lpcStartBand[HIFILT]]; tC->lpcStartBand[LOFILT] = FDKaacEnc_FreqToBandWidthRounding( pCfg->filterStartFreq[LOFILT], sampleRate, pC->sfbCnt, pC->sfbOffset); tC->lpcStartLine[LOFILT] = pC->sfbOffset[tC->lpcStartBand[LOFILT]]; FDKaacEnc_CalcGaussWindow( tC->acfWindow[HIFILT], tC->maxOrder + 1, sampleRate, granuleLength, pCfg->tnsTimeResolution[HIFILT], TNS_TIMERES_SCALE); FDKaacEnc_CalcGaussWindow( tC->acfWindow[LOFILT], tC->maxOrder + 1, sampleRate, granuleLength, pCfg->tnsTimeResolution[LOFILT], TNS_TIMERES_SCALE); } else { tC->tnsActive = FALSE; /* no configuration available, disable tns tool */ } } break; default: tC->tnsActive = FALSE; /* no configuration available, disable tns tool */ } return AAC_ENC_OK; } /***************************************************************************/ /*! \brief FDKaacEnc_ScaleUpSpectrum Scales up spectrum lines in a given frequency section \param scaled spectrum \param original spectrum \param frequency line to start scaling \param frequency line to enc scaling \return scale factor ****************************************************************************/ static inline INT FDKaacEnc_ScaleUpSpectrum(FIXP_DBL *dest, const FIXP_DBL *src, const INT startLine, const INT stopLine) { INT i, scale; FIXP_DBL maxVal = FL2FXCONST_DBL(0.f); /* Get highest value in given spectrum */ for (i = startLine; i < stopLine; i++) { maxVal = fixMax(maxVal, fixp_abs(src[i])); } scale = CountLeadingBits(maxVal); /* Scale spectrum according to highest value */ for (i = startLine; i < stopLine; i++) { dest[i] = src[i] << scale; } return scale; } /***************************************************************************/ /*! \brief FDKaacEnc_CalcAutoCorrValue Calculate autocorellation value for one lag \param pointer to spectrum \param start line \param stop line \param lag to be calculated \param scaling of the lag ****************************************************************************/ static inline FIXP_DBL FDKaacEnc_CalcAutoCorrValue(const FIXP_DBL *spectrum, const INT startLine, const INT stopLine, const INT lag, const INT scale) { int i; FIXP_DBL result = FL2FXCONST_DBL(0.f); /* This versions allows to save memory accesses, when computing pow2 */ /* It is of interest for ARM, XTENSA without parallel memory access */ if (lag == 0) { for (i = startLine; i < stopLine; i++) { result += (fPow2(spectrum[i]) >> scale); } } else { for (i = startLine; i < (stopLine - lag); i++) { result += (fMult(spectrum[i], spectrum[i + lag]) >> scale); } } return result; } /***************************************************************************/ /*! \brief FDKaacEnc_AutoCorrNormFac Autocorrelation function for 1st and 2nd half of the spectrum \param pointer to spectrum \param pointer to autocorrelation window \param filter start line ****************************************************************************/ static inline FIXP_DBL FDKaacEnc_AutoCorrNormFac(const FIXP_DBL value, const INT scale, INT *sc) { #define HLM_MIN_NRG 0.0000000037252902984619140625f /* 2^-28 */ #define MAX_INV_NRGFAC (1.f / HLM_MIN_NRG) FIXP_DBL retValue; FIXP_DBL A, B; if (scale >= 0) { A = value; B = FL2FXCONST_DBL(HLM_MIN_NRG) >> fixMin(DFRACT_BITS - 1, scale); } else { A = value >> fixMin(DFRACT_BITS - 1, (-scale)); B = FL2FXCONST_DBL(HLM_MIN_NRG); } if (A > B) { int shift = 0; FIXP_DBL tmp = invSqrtNorm2(value, &shift); retValue = fMult(tmp, tmp); *sc += (2 * shift); } else { /* MAX_INV_NRGFAC*FDKpow(2,-28) = 1/2^-28 * 2^-28 = 1.0 */ retValue = /*FL2FXCONST_DBL(MAX_INV_NRGFAC*FDKpow(2,-28))*/ (FIXP_DBL)MAXVAL_DBL; *sc += scale + 28; } return retValue; } static void FDKaacEnc_MergedAutoCorrelation( const FIXP_DBL *spectrum, const INT isLowDelay, const FIXP_DBL acfWindow[MAX_NUM_OF_FILTERS][TNS_MAX_ORDER + 3 + 1], const INT lpcStartLine[MAX_NUM_OF_FILTERS], const INT lpcStopLine, const INT maxOrder, const INT acfSplit[MAX_NUM_OF_FILTERS], FIXP_DBL *_rxx1, FIXP_DBL *_rxx2) { int i, idx0, idx1, idx2, idx3, idx4, lag; FIXP_DBL rxx1_0, rxx2_0, rxx3_0, rxx4_0; /* buffer for temporal spectrum */ C_ALLOC_SCRATCH_START(pSpectrum, FIXP_DBL, (1024)) /* MDCT line indices separating the 1st, 2nd, 3rd, and 4th analysis quarters */ if ((acfSplit[LOFILT] == -1) || (acfSplit[HIFILT] == -1)) { /* autocorrelation function for 1st, 2nd, 3rd, and 4th quarter of the * spectrum */ idx0 = lpcStartLine[LOFILT]; i = lpcStopLine - lpcStartLine[LOFILT]; idx1 = idx0 + i / 4; idx2 = idx0 + i / 2; idx3 = idx0 + i * 3 / 4; idx4 = lpcStopLine; } else { FDK_ASSERT(acfSplit[LOFILT] == 1); FDK_ASSERT(acfSplit[HIFILT] == 3); i = (lpcStopLine - lpcStartLine[HIFILT]) / 3; idx0 = lpcStartLine[LOFILT]; idx1 = lpcStartLine[HIFILT]; idx2 = idx1 + i; idx3 = idx2 + i; idx4 = lpcStopLine; } /* copy spectrum to temporal buffer and scale up as much as possible */ INT sc1 = FDKaacEnc_ScaleUpSpectrum(pSpectrum, spectrum, idx0, idx1); INT sc2 = FDKaacEnc_ScaleUpSpectrum(pSpectrum, spectrum, idx1, idx2); INT sc3 = FDKaacEnc_ScaleUpSpectrum(pSpectrum, spectrum, idx2, idx3); INT sc4 = FDKaacEnc_ScaleUpSpectrum(pSpectrum, spectrum, idx3, idx4); /* get scaling values for summation */ INT nsc1, nsc2, nsc3, nsc4; for (nsc1 = 1; (1 << nsc1) < (idx1 - idx0); nsc1++) ; for (nsc2 = 1; (1 << nsc2) < (idx2 - idx1); nsc2++) ; for (nsc3 = 1; (1 << nsc3) < (idx3 - idx2); nsc3++) ; for (nsc4 = 1; (1 << nsc4) < (idx4 - idx3); nsc4++) ; /* compute autocorrelation value at lag zero, i. e. energy, for each quarter */ rxx1_0 = FDKaacEnc_CalcAutoCorrValue(pSpectrum, idx0, idx1, 0, nsc1); rxx2_0 = FDKaacEnc_CalcAutoCorrValue(pSpectrum, idx1, idx2, 0, nsc2); rxx3_0 = FDKaacEnc_CalcAutoCorrValue(pSpectrum, idx2, idx3, 0, nsc3); rxx4_0 = FDKaacEnc_CalcAutoCorrValue(pSpectrum, idx3, idx4, 0, nsc4); /* compute energy normalization factors, i. e. 1/energy (saves some divisions) */ if (rxx1_0 != FL2FXCONST_DBL(0.f)) { INT sc_fac1 = -1; FIXP_DBL fac1 = FDKaacEnc_AutoCorrNormFac(rxx1_0, ((-2 * sc1) + nsc1), &sc_fac1); _rxx1[0] = scaleValue(fMult(rxx1_0, fac1), sc_fac1); if (isLowDelay) { for (lag = 1; lag <= maxOrder; lag++) { /* compute energy-normalized and windowed autocorrelation values at this * lag */ FIXP_DBL x1 = FDKaacEnc_CalcAutoCorrValue(pSpectrum, idx0, idx1, lag, nsc1); _rxx1[lag] = fMult(scaleValue(fMult(x1, fac1), sc_fac1), acfWindow[LOFILT][lag]); } } else { for (lag = 1; lag <= maxOrder; lag++) { if ((3 * lag) <= maxOrder + 3) { FIXP_DBL x1 = FDKaacEnc_CalcAutoCorrValue(pSpectrum, idx0, idx1, lag, nsc1); _rxx1[lag] = fMult(scaleValue(fMult(x1, fac1), sc_fac1), acfWindow[LOFILT][3 * lag]); } } } } /* auto corr over upper 3/4 of spectrum */ if (!((rxx2_0 == FL2FXCONST_DBL(0.f)) && (rxx3_0 == FL2FXCONST_DBL(0.f)) && (rxx4_0 == FL2FXCONST_DBL(0.f)))) { FIXP_DBL fac2, fac3, fac4; fac2 = fac3 = fac4 = FL2FXCONST_DBL(0.f); INT sc_fac2, sc_fac3, sc_fac4; sc_fac2 = sc_fac3 = sc_fac4 = 0; if (rxx2_0 != FL2FXCONST_DBL(0.f)) { fac2 = FDKaacEnc_AutoCorrNormFac(rxx2_0, ((-2 * sc2) + nsc2), &sc_fac2); sc_fac2 -= 2; } if (rxx3_0 != FL2FXCONST_DBL(0.f)) { fac3 = FDKaacEnc_AutoCorrNormFac(rxx3_0, ((-2 * sc3) + nsc3), &sc_fac3); sc_fac3 -= 2; } if (rxx4_0 != FL2FXCONST_DBL(0.f)) { fac4 = FDKaacEnc_AutoCorrNormFac(rxx4_0, ((-2 * sc4) + nsc4), &sc_fac4); sc_fac4 -= 2; } _rxx2[0] = scaleValue(fMult(rxx2_0, fac2), sc_fac2) + scaleValue(fMult(rxx3_0, fac3), sc_fac3) + scaleValue(fMult(rxx4_0, fac4), sc_fac4); for (lag = 1; lag <= maxOrder; lag++) { /* merge quarters 2, 3, 4 into one autocorrelation; quarter 1 stays * separate */ FIXP_DBL x2 = scaleValue(fMult(FDKaacEnc_CalcAutoCorrValue( pSpectrum, idx1, idx2, lag, nsc2), fac2), sc_fac2) + scaleValue(fMult(FDKaacEnc_CalcAutoCorrValue( pSpectrum, idx2, idx3, lag, nsc3), fac3), sc_fac3) + scaleValue(fMult(FDKaacEnc_CalcAutoCorrValue( pSpectrum, idx3, idx4, lag, nsc4), fac4), sc_fac4); _rxx2[lag] = fMult(x2, acfWindow[HIFILT][lag]); } } C_ALLOC_SCRATCH_END(pSpectrum, FIXP_DBL, (1024)) } /***************************************************************************** functionname: FDKaacEnc_TnsDetect description: do decision, if TNS shall be used or not returns: input: tns data structure (modified), tns config structure, scalefactor size and table, spectrum, subblock num, blocktype, sfb-wise energy. *****************************************************************************/ INT FDKaacEnc_TnsDetect(TNS_DATA *tnsData, const TNS_CONFIG *tC, TNS_INFO *tnsInfo, INT sfbCnt, const FIXP_DBL *spectrum, INT subBlockNumber, INT blockType) { /* autocorrelation function for 1st, 2nd, 3rd, and 4th quarter of the * spectrum. */ FIXP_DBL rxx1[TNS_MAX_ORDER + 1]; /* higher part */ FIXP_DBL rxx2[TNS_MAX_ORDER + 1]; /* lower part */ FIXP_LPC parcor_tmp[TNS_MAX_ORDER]; int i; FDKmemclear(rxx1, sizeof(rxx1)); FDKmemclear(rxx2, sizeof(rxx2)); TNS_SUBBLOCK_INFO *tsbi = (blockType == SHORT_WINDOW) ? &tnsData->dataRaw.Short.subBlockInfo[subBlockNumber] : &tnsData->dataRaw.Long.subBlockInfo; tnsData->filtersMerged = FALSE; tsbi->tnsActive[HIFILT] = FALSE; tsbi->predictionGain[HIFILT] = 1000; tsbi->tnsActive[LOFILT] = FALSE; tsbi->predictionGain[LOFILT] = 1000; tnsInfo->numOfFilters[subBlockNumber] = 0; tnsInfo->coefRes[subBlockNumber] = tC->coefRes; for (i = 0; i < tC->maxOrder; i++) { tnsInfo->coef[subBlockNumber][HIFILT][i] = tnsInfo->coef[subBlockNumber][LOFILT][i] = 0; } tnsInfo->length[subBlockNumber][HIFILT] = tnsInfo->length[subBlockNumber][LOFILT] = 0; tnsInfo->order[subBlockNumber][HIFILT] = tnsInfo->order[subBlockNumber][LOFILT] = 0; if ((tC->tnsActive) && (tC->maxOrder > 0)) { int sumSqrCoef; FDKaacEnc_MergedAutoCorrelation( spectrum, tC->isLowDelay, tC->acfWindow, tC->lpcStartLine, tC->lpcStopLine, tC->maxOrder, tC->confTab.acfSplit, rxx1, rxx2); /* compute higher TNS filter coefficients in lattice form (ParCor) with * LeRoux-Gueguen/Schur algorithm */ { FIXP_DBL predictionGain_m; INT predictionGain_e; CLpc_AutoToParcor(rxx2, 0, parcor_tmp, tC->confTab.tnsLimitOrder[HIFILT], &predictionGain_m, &predictionGain_e); tsbi->predictionGain[HIFILT] = (INT)fMultNorm(predictionGain_m, predictionGain_e, 1000, 31, 31); } /* non-linear quantization of TNS lattice coefficients with given resolution */ FDKaacEnc_Parcor2Index(parcor_tmp, tnsInfo->coef[subBlockNumber][HIFILT], tC->confTab.tnsLimitOrder[HIFILT], tC->coefRes); /* reduce filter order by truncating trailing zeros, compute sum(abs(coefs)) */ for (i = tC->confTab.tnsLimitOrder[HIFILT] - 1; i >= 0; i--) { if (tnsInfo->coef[subBlockNumber][HIFILT][i] != 0) { break; } } tnsInfo->order[subBlockNumber][HIFILT] = i + 1; sumSqrCoef = 0; for (; i >= 0; i--) { sumSqrCoef += tnsInfo->coef[subBlockNumber][HIFILT][i] * tnsInfo->coef[subBlockNumber][HIFILT][i]; } tnsInfo->direction[subBlockNumber][HIFILT] = tC->confTab.tnsFilterDirection[HIFILT]; tnsInfo->length[subBlockNumber][HIFILT] = sfbCnt - tC->lpcStartBand[HIFILT]; /* disable TNS if predictionGain is less than 3dB or sumSqrCoef is too small */ if ((tsbi->predictionGain[HIFILT] > tC->confTab.threshOn[HIFILT]) || (sumSqrCoef > (tC->confTab.tnsLimitOrder[HIFILT] / 2 + 2))) { tsbi->tnsActive[HIFILT] = TRUE; tnsInfo->numOfFilters[subBlockNumber]++; /* compute second filter for lower quarter; only allowed for long windows! */ if ((blockType != SHORT_WINDOW) && (tC->confTab.filterEnabled[LOFILT]) && (tC->confTab.seperateFiltersAllowed)) { /* compute second filter for lower frequencies */ /* compute TNS filter in lattice (ParCor) form with LeRoux-Gueguen * algorithm */ INT predGain; { FIXP_DBL predictionGain_m; INT predictionGain_e; CLpc_AutoToParcor(rxx1, 0, parcor_tmp, tC->confTab.tnsLimitOrder[LOFILT], &predictionGain_m, &predictionGain_e); predGain = (INT)fMultNorm(predictionGain_m, predictionGain_e, 1000, 31, 31); } /* non-linear quantization of TNS lattice coefficients with given * resolution */ FDKaacEnc_Parcor2Index(parcor_tmp, tnsInfo->coef[subBlockNumber][LOFILT], tC->confTab.tnsLimitOrder[LOFILT], tC->coefRes); /* reduce filter order by truncating trailing zeros, compute * sum(abs(coefs)) */ for (i = tC->confTab.tnsLimitOrder[LOFILT] - 1; i >= 0; i--) { if (tnsInfo->coef[subBlockNumber][LOFILT][i] != 0) { break; } } tnsInfo->order[subBlockNumber][LOFILT] = i + 1; sumSqrCoef = 0; for (; i >= 0; i--) { sumSqrCoef += tnsInfo->coef[subBlockNumber][LOFILT][i] * tnsInfo->coef[subBlockNumber][LOFILT][i]; } tnsInfo->direction[subBlockNumber][LOFILT] = tC->confTab.tnsFilterDirection[LOFILT]; tnsInfo->length[subBlockNumber][LOFILT] = tC->lpcStartBand[HIFILT] - tC->lpcStartBand[LOFILT]; /* filter lower quarter if gain is high enough, but not if it's too high */ if (((predGain > tC->confTab.threshOn[LOFILT]) && (predGain < (16000 * tC->confTab.tnsLimitOrder[LOFILT]))) || ((sumSqrCoef > 9) && (sumSqrCoef < 22 * tC->confTab.tnsLimitOrder[LOFILT]))) { /* compare lower to upper filter; if they are very similar, merge them */ tsbi->tnsActive[LOFILT] = TRUE; sumSqrCoef = 0; for (i = 0; i < tC->confTab.tnsLimitOrder[LOFILT]; i++) { sumSqrCoef += fAbs(tnsInfo->coef[subBlockNumber][HIFILT][i] - tnsInfo->coef[subBlockNumber][LOFILT][i]); } if ((sumSqrCoef < 2) && (tnsInfo->direction[subBlockNumber][LOFILT] == tnsInfo->direction[subBlockNumber][HIFILT])) { tnsData->filtersMerged = TRUE; tnsInfo->length[subBlockNumber][HIFILT] = sfbCnt - tC->lpcStartBand[LOFILT]; for (; i < tnsInfo->order[subBlockNumber][HIFILT]; i++) { if (fAbs(tnsInfo->coef[subBlockNumber][HIFILT][i]) > 1) { break; } } for (i--; i >= 0; i--) { if (tnsInfo->coef[subBlockNumber][HIFILT][i] != 0) { break; } } if (i < tnsInfo->order[subBlockNumber][HIFILT]) { tnsInfo->order[subBlockNumber][HIFILT] = i + 1; } } else { tnsInfo->numOfFilters[subBlockNumber]++; } } /* filter lower part */ tsbi->predictionGain[LOFILT] = predGain; } /* second filter allowed */ } /* if predictionGain > 1437 ... */ } /* maxOrder > 0 && tnsActive */ return 0; } /***************************************************************************/ /*! \brief FDKaacLdEnc_TnsSync synchronize TNS parameters when TNS gain difference small (relative) \param pointer to TNS data structure (destination) \param pointer to TNS data structure (source) \param pointer to TNS config structure \param number of sub-block \param block type \return void ****************************************************************************/ void FDKaacEnc_TnsSync(TNS_DATA *tnsDataDest, const TNS_DATA *tnsDataSrc, TNS_INFO *tnsInfoDest, TNS_INFO *tnsInfoSrc, const INT blockTypeDest, const INT blockTypeSrc, const TNS_CONFIG *tC) { int i, w, absDiff, nWindows; TNS_SUBBLOCK_INFO *sbInfoDest; const TNS_SUBBLOCK_INFO *sbInfoSrc; /* if one channel contains short blocks and the other not, do not synchronize */ if ((blockTypeSrc == SHORT_WINDOW && blockTypeDest != SHORT_WINDOW) || (blockTypeDest == SHORT_WINDOW && blockTypeSrc != SHORT_WINDOW)) { return; } if (blockTypeDest != SHORT_WINDOW) { sbInfoDest = &tnsDataDest->dataRaw.Long.subBlockInfo; sbInfoSrc = &tnsDataSrc->dataRaw.Long.subBlockInfo; nWindows = 1; } else { sbInfoDest = &tnsDataDest->dataRaw.Short.subBlockInfo[0]; sbInfoSrc = &tnsDataSrc->dataRaw.Short.subBlockInfo[0]; nWindows = 8; } for (w = 0; w < nWindows; w++) { const TNS_SUBBLOCK_INFO *pSbInfoSrcW = sbInfoSrc + w; TNS_SUBBLOCK_INFO *pSbInfoDestW = sbInfoDest + w; INT doSync = 1, absDiffSum = 0; /* if TNS is active in at least one channel, check if ParCor coefficients of * higher filter are similar */ if (pSbInfoDestW->tnsActive[HIFILT] || pSbInfoSrcW->tnsActive[HIFILT]) { for (i = 0; i < tC->maxOrder; i++) { absDiff = fAbs(tnsInfoDest->coef[w][HIFILT][i] - tnsInfoSrc->coef[w][HIFILT][i]); absDiffSum += absDiff; /* if coefficients diverge too much between channels, do not synchronize */ if ((absDiff > 1) || (absDiffSum > 2)) { doSync = 0; break; } } if (doSync) { /* if no significant difference was detected, synchronize coefficient * sets */ if (pSbInfoSrcW->tnsActive[HIFILT]) { /* no dest filter, or more dest than source filters: use one dest * filter */ if ((!pSbInfoDestW->tnsActive[HIFILT]) || ((pSbInfoDestW->tnsActive[HIFILT]) && (tnsInfoDest->numOfFilters[w] > tnsInfoSrc->numOfFilters[w]))) { pSbInfoDestW->tnsActive[HIFILT] = tnsInfoDest->numOfFilters[w] = 1; } tnsDataDest->filtersMerged = tnsDataSrc->filtersMerged; tnsInfoDest->order[w][HIFILT] = tnsInfoSrc->order[w][HIFILT]; tnsInfoDest->length[w][HIFILT] = tnsInfoSrc->length[w][HIFILT]; tnsInfoDest->direction[w][HIFILT] = tnsInfoSrc->direction[w][HIFILT]; tnsInfoDest->coefCompress[w][HIFILT] = tnsInfoSrc->coefCompress[w][HIFILT]; for (i = 0; i < tC->maxOrder; i++) { tnsInfoDest->coef[w][HIFILT][i] = tnsInfoSrc->coef[w][HIFILT][i]; } } else pSbInfoDestW->tnsActive[HIFILT] = tnsInfoDest->numOfFilters[w] = 0; } } } } /***************************************************************************/ /*! \brief FDKaacEnc_TnsEncode perform TNS encoding \param pointer to TNS info structure \param pointer to TNS data structure \param number of sfbs \param pointer to TNS config structure \param low-pass line \param pointer to spectrum \param number of sub-block \param block type \return ERROR STATUS ****************************************************************************/ INT FDKaacEnc_TnsEncode(TNS_INFO *tnsInfo, TNS_DATA *tnsData, const INT numOfSfb, const TNS_CONFIG *tC, const INT lowPassLine, FIXP_DBL *spectrum, const INT subBlockNumber, const INT blockType) { INT i, startLine, stopLine; if (((blockType == SHORT_WINDOW) && (!tnsData->dataRaw.Short.subBlockInfo[subBlockNumber] .tnsActive[HIFILT])) || ((blockType != SHORT_WINDOW) && (!tnsData->dataRaw.Long.subBlockInfo.tnsActive[HIFILT]))) { return 1; } startLine = (tnsData->filtersMerged) ? tC->lpcStartLine[LOFILT] : tC->lpcStartLine[HIFILT]; stopLine = tC->lpcStopLine; for (i = 0; i < tnsInfo->numOfFilters[subBlockNumber]; i++) { INT lpcGainFactor; FIXP_LPC LpcCoeff[TNS_MAX_ORDER]; FIXP_DBL workBuffer[TNS_MAX_ORDER]; FIXP_LPC parcor_tmp[TNS_MAX_ORDER]; FDKaacEnc_Index2Parcor(tnsInfo->coef[subBlockNumber][i], parcor_tmp, tnsInfo->order[subBlockNumber][i], tC->coefRes); lpcGainFactor = CLpc_ParcorToLpc( parcor_tmp, LpcCoeff, tnsInfo->order[subBlockNumber][i], workBuffer); FDKmemclear(workBuffer, TNS_MAX_ORDER * sizeof(FIXP_DBL)); CLpc_Analysis(&spectrum[startLine], stopLine - startLine, LpcCoeff, lpcGainFactor, tnsInfo->order[subBlockNumber][i], workBuffer, NULL); /* update for second filter */ startLine = tC->lpcStartLine[LOFILT]; stopLine = tC->lpcStartLine[HIFILT]; } return (0); } static void FDKaacEnc_CalcGaussWindow(FIXP_DBL *win, const int winSize, const INT samplingRate, const INT transformResolution, const FIXP_DBL timeResolution, const INT timeResolution_e) { #define PI_E (2) #define PI_M FL2FXCONST_DBL(3.1416f / (float)(1 << PI_E)) #define EULER_E (2) #define EULER_M FL2FXCONST_DBL(2.7183 / (float)(1 << EULER_E)) #define COEFF_LOOP_SCALE (4) INT i, e1, e2, gaussExp_e; FIXP_DBL gaussExp_m; /* calc. window exponent from time resolution: * * gaussExp = PI * samplingRate * 0.001f * timeResolution / * transformResolution; gaussExp = -0.5f * gaussExp * gaussExp; */ gaussExp_m = fMultNorm( timeResolution, fMult(PI_M, fDivNorm((FIXP_DBL)(samplingRate), (FIXP_DBL)(LONG)(transformResolution * 1000.f), &e1)), &e2); gaussExp_m = -fPow2Div2(gaussExp_m); gaussExp_e = 2 * (e1 + e2 + timeResolution_e + PI_E); FDK_ASSERT(winSize < (1 << COEFF_LOOP_SCALE)); /* calc. window coefficients * win[i] = (float)exp( gaussExp * (i+0.5) * (i+0.5) ); */ for (i = 0; i < winSize; i++) { win[i] = fPow( EULER_M, EULER_E, fMult(gaussExp_m, fPow2((i * FL2FXCONST_DBL(1.f / (float)(1 << COEFF_LOOP_SCALE)) + FL2FXCONST_DBL(.5f / (float)(1 << COEFF_LOOP_SCALE))))), gaussExp_e + 2 * COEFF_LOOP_SCALE, &e1); win[i] = scaleValueSaturate(win[i], e1); } } static INT FDKaacEnc_Search3(FIXP_LPC parcor) { INT i, index = 0; for (i = 0; i < 8; i++) { if (parcor > FDKaacEnc_tnsCoeff3Borders[i]) index = i; } return (index - 4); } static INT FDKaacEnc_Search4(FIXP_LPC parcor) { INT i, index = 0; for (i = 0; i < 16; i++) { if (parcor > FDKaacEnc_tnsCoeff4Borders[i]) index = i; } return (index - 8); } /***************************************************************************** functionname: FDKaacEnc_Parcor2Index *****************************************************************************/ static void FDKaacEnc_Parcor2Index(const FIXP_LPC *parcor, INT *RESTRICT index, const INT order, const INT bitsPerCoeff) { INT i; for (i = 0; i < order; i++) { if (bitsPerCoeff == 3) index[i] = FDKaacEnc_Search3(parcor[i]); else index[i] = FDKaacEnc_Search4(parcor[i]); } } /***************************************************************************** functionname: FDKaacEnc_Index2Parcor description: inverse quantization for reflection coefficients returns: - input: quantized values, ptr. to reflection coefficients, no. of coefficients, resolution output: reflection coefficients *****************************************************************************/ static void FDKaacEnc_Index2Parcor(const INT *index, FIXP_LPC *RESTRICT parcor, const INT order, const INT bitsPerCoeff) { INT i; for (i = 0; i < order; i++) parcor[i] = bitsPerCoeff == 4 ? FDKaacEnc_tnsEncCoeff4[index[i] + 8] : FDKaacEnc_tnsEncCoeff3[index[i] + 4]; } fdk-aac-2.0.2/libAACenc/src/aacenc_tns.h000066400000000000000000000204261404225310700176020ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): Alex Groeschel Description: Temporal noise shaping *******************************************************************************/ #ifndef AACENC_TNS_H #define AACENC_TNS_H #include "common_fix.h" #include "psy_const.h" #ifndef PI #define PI 3.1415926535897931f #endif /** * TNS_ENABLE_MASK * This bitfield defines which TNS features are enabled * The TNS mask is composed of 4 bits. * tnsMask |= 0x1; activate TNS short blocks * tnsMask |= 0x2; activate TNS for long blocks * tnsMask |= 0x4; activate TNS PEAK tool for short blocks * tnsMask |= 0x8; activate TNS PEAK tool for long blocks */ #define TNS_ENABLE_MASK 0xf /* TNS max filter order for Low Complexity MPEG4 profile */ #define TNS_MAX_ORDER 12 #define MAX_NUM_OF_FILTERS 2 #define HIFILT 0 /* index of higher filter */ #define LOFILT 1 /* index of lower filter */ typedef struct { /* stuff that is tabulated dependent on bitrate etc. */ INT filterEnabled[MAX_NUM_OF_FILTERS]; INT threshOn[MAX_NUM_OF_FILTERS]; /* min. prediction gain for using tns TABUL*/ INT filterStartFreq[MAX_NUM_OF_FILTERS]; /* lowest freq for lpc TABUL*/ INT tnsLimitOrder[MAX_NUM_OF_FILTERS]; /* Limit for TNS order TABUL*/ INT tnsFilterDirection[MAX_NUM_OF_FILTERS]; /* Filtering direction, 0=up, 1=down TABUL */ INT acfSplit[MAX_NUM_OF_FILTERS]; FIXP_DBL tnsTimeResolution[MAX_NUM_OF_FILTERS]; /* TNS max. time resolution TABUL. Should be fract but MSVC won't compile then */ INT seperateFiltersAllowed; } TNS_PARAMETER_TABULATED; typedef struct { /*assigned at InitTime*/ TNS_PARAMETER_TABULATED confTab; INT isLowDelay; INT tnsActive; INT maxOrder; /* max. order of tns filter */ INT coefRes; FIXP_DBL acfWindow[MAX_NUM_OF_FILTERS][TNS_MAX_ORDER + 3 + 1]; /* now some things that only probably can be done at Init time; could be they have to be split up for each individual (short) window or even filter. */ INT lpcStartBand[MAX_NUM_OF_FILTERS]; INT lpcStartLine[MAX_NUM_OF_FILTERS]; INT lpcStopBand; INT lpcStopLine; } TNS_CONFIG; typedef struct { INT tnsActive[MAX_NUM_OF_FILTERS]; INT predictionGain[MAX_NUM_OF_FILTERS]; } TNS_SUBBLOCK_INFO; typedef struct { /*changed at runTime*/ TNS_SUBBLOCK_INFO subBlockInfo[TRANS_FAC]; FIXP_DBL ratioMultTable[TRANS_FAC][MAX_SFB_SHORT]; } TNS_DATA_SHORT; typedef struct { /*changed at runTime*/ TNS_SUBBLOCK_INFO subBlockInfo; FIXP_DBL ratioMultTable[MAX_SFB_LONG]; } TNS_DATA_LONG; /* can be implemented as union */ typedef shouldBeUnion { TNS_DATA_LONG Long; TNS_DATA_SHORT Short; } TNS_DATA_RAW; typedef struct { INT numOfSubblocks; TNS_DATA_RAW dataRaw; INT tnsMaxScaleSpec; INT filtersMerged; } TNS_DATA; typedef struct { INT numOfFilters[TRANS_FAC]; INT coefRes[TRANS_FAC]; INT length[TRANS_FAC][MAX_NUM_OF_FILTERS]; INT order[TRANS_FAC][MAX_NUM_OF_FILTERS]; INT direction[TRANS_FAC][MAX_NUM_OF_FILTERS]; INT coefCompress[TRANS_FAC][MAX_NUM_OF_FILTERS]; /* for Long: length TNS_MAX_ORDER (12 for LC) is required -> 12 */ /* for Short: length TRANS_FAC*TNS_MAX_ORDER (only 5 for short LC) is required * -> 8*5=40 */ /* Currently TRANS_FAC*TNS_MAX_ORDER = 8*12 = 96 (for LC) is used (per * channel)! Memory could be saved here! */ INT coef[TRANS_FAC][MAX_NUM_OF_FILTERS][TNS_MAX_ORDER]; } TNS_INFO; INT FDKaacEnc_FreqToBandWidthRounding(const INT freq, const INT fs, const INT numOfBands, const INT *bandStartOffset); #endif /* AACENC_TNS_H */ fdk-aac-2.0.2/libAACenc/src/adj_thr.cpp000066400000000000000000003526401404225310700174600ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Werner Description: Threshold compensation *******************************************************************************/ #include "adj_thr.h" #include "sf_estim.h" #include "aacEnc_ram.h" #define NUM_NRG_LEVS (8) #define INV_INT_TAB_SIZE (8) static const FIXP_DBL invInt[INV_INT_TAB_SIZE] = { 0x7fffffff, 0x7fffffff, 0x40000000, 0x2aaaaaaa, 0x20000000, 0x19999999, 0x15555555, 0x12492492}; #define INV_SQRT4_TAB_SIZE (8) static const FIXP_DBL invSqrt4[INV_SQRT4_TAB_SIZE] = { 0x7fffffff, 0x7fffffff, 0x6ba27e65, 0x61424bb5, 0x5a827999, 0x55994845, 0x51c8e33c, 0x4eb160d1}; /*static const INT invRedExp = 4;*/ static const FIXP_DBL SnrLdMin1 = (FIXP_DBL)0xfcad0ddf; /*FL2FXCONST_DBL(FDKlog(0.316)/FDKlog(2.0)/LD_DATA_SCALING);*/ static const FIXP_DBL SnrLdMin2 = (FIXP_DBL)0x0351e1a2; /*FL2FXCONST_DBL(FDKlog(3.16) /FDKlog(2.0)/LD_DATA_SCALING);*/ static const FIXP_DBL SnrLdFac = (FIXP_DBL)0xff5b2c3e; /*FL2FXCONST_DBL(FDKlog(0.8) /FDKlog(2.0)/LD_DATA_SCALING);*/ static const FIXP_DBL SnrLdMin3 = (FIXP_DBL)0xfe000000; /*FL2FXCONST_DBL(FDKlog(0.5) /FDKlog(2.0)/LD_DATA_SCALING);*/ static const FIXP_DBL SnrLdMin4 = (FIXP_DBL)0x02000000; /*FL2FXCONST_DBL(FDKlog(2.0) /FDKlog(2.0)/LD_DATA_SCALING);*/ static const FIXP_DBL SnrLdMin5 = (FIXP_DBL)0xfc000000; /*FL2FXCONST_DBL(FDKlog(0.25) /FDKlog(2.0)/LD_DATA_SCALING);*/ /* The bits2Pe factors are choosen for the case that some times the crash recovery strategy will be activated once. */ #define AFTERBURNER_STATI 2 #define MAX_ALLOWED_EL_CHANNELS 2 typedef struct { INT bitrate; FIXP_DBL bits2PeFactor[AFTERBURNER_STATI][MAX_ALLOWED_EL_CHANNELS]; } BIT_PE_SFAC; typedef struct { INT sampleRate; const BIT_PE_SFAC *pPeTab; INT nEntries; } BITS2PE_CFG_TAB; #define FL2B2PE(value) FL2FXCONST_DBL((value) / (1 << 2)) static const BIT_PE_SFAC S_Bits2PeTab16000[] = { /* bitrate| afterburner off | afterburner on | | nCh=1 | nCh=2 | nCh=1 | nCh=2 */ {10000, {{FL2B2PE(1.60f), FL2B2PE(0.00f)}, {FL2B2PE(1.40f), FL2B2PE(0.00f)}}}, {24000, {{FL2B2PE(1.80f), FL2B2PE(1.40f)}, {FL2B2PE(1.60f), FL2B2PE(1.20f)}}}, {32000, {{FL2B2PE(1.80f), FL2B2PE(1.60f)}, {FL2B2PE(1.60f), FL2B2PE(1.40f)}}}, {48000, {{FL2B2PE(1.60f), FL2B2PE(1.80f)}, {FL2B2PE(1.60f), FL2B2PE(1.60f)}}}, {64000, {{FL2B2PE(1.20f), FL2B2PE(1.60f)}, {FL2B2PE(1.20f), FL2B2PE(1.60f)}}}, {96000, {{FL2B2PE(1.40f), FL2B2PE(1.80f)}, {FL2B2PE(1.40f), FL2B2PE(1.60f)}}}, {128000, {{FL2B2PE(1.40f), FL2B2PE(1.80f)}, {FL2B2PE(1.40f), FL2B2PE(1.80f)}}}, {148000, {{FL2B2PE(1.40f), FL2B2PE(1.80f)}, {FL2B2PE(1.40f), FL2B2PE(1.40f)}}}}; static const BIT_PE_SFAC S_Bits2PeTab22050[] = { /* bitrate| afterburner off | afterburner on | | nCh=1 | nCh=2 | nCh=1 | nCh=2 */ {16000, {{FL2B2PE(1.60f), FL2B2PE(1.40f)}, {FL2B2PE(1.20f), FL2B2PE(0.80f)}}}, {24000, {{FL2B2PE(1.60f), FL2B2PE(1.40f)}, {FL2B2PE(1.40f), FL2B2PE(1.00f)}}}, {32000, {{FL2B2PE(1.40f), FL2B2PE(1.40f)}, {FL2B2PE(1.40f), FL2B2PE(1.20f)}}}, {48000, {{FL2B2PE(1.20f), FL2B2PE(1.60f)}, {FL2B2PE(1.20f), FL2B2PE(1.40f)}}}, {64000, {{FL2B2PE(1.60f), FL2B2PE(1.60f)}, {FL2B2PE(1.60f), FL2B2PE(1.40f)}}}, {96000, {{FL2B2PE(1.80f), FL2B2PE(1.60f)}, {FL2B2PE(1.80f), FL2B2PE(1.60f)}}}, {128000, {{FL2B2PE(1.80f), FL2B2PE(1.80f)}, {FL2B2PE(1.60f), FL2B2PE(1.60f)}}}, {148000, {{FL2B2PE(1.40f), FL2B2PE(1.80f)}, {FL2B2PE(1.40f), FL2B2PE(1.60f)}}}}; static const BIT_PE_SFAC S_Bits2PeTab24000[] = { /* bitrate| afterburner off | afterburner on | | nCh=1 | nCh=2 | nCh=1 | nCh=2 */ {16000, {{FL2B2PE(1.40f), FL2B2PE(1.40f)}, {FL2B2PE(1.20f), FL2B2PE(0.80f)}}}, {24000, {{FL2B2PE(1.60f), FL2B2PE(1.20f)}, {FL2B2PE(1.40f), FL2B2PE(1.00f)}}}, {32000, {{FL2B2PE(1.40f), FL2B2PE(1.20f)}, {FL2B2PE(1.40f), FL2B2PE(0.80f)}}}, {48000, {{FL2B2PE(1.40f), FL2B2PE(1.60f)}, {FL2B2PE(1.40f), FL2B2PE(1.40f)}}}, {64000, {{FL2B2PE(1.60f), FL2B2PE(1.60f)}, {FL2B2PE(1.60f), FL2B2PE(1.40f)}}}, {96000, {{FL2B2PE(1.80f), FL2B2PE(1.60f)}, {FL2B2PE(1.80f), FL2B2PE(1.60f)}}}, {128000, {{FL2B2PE(1.40f), FL2B2PE(1.60f)}, {FL2B2PE(1.80f), FL2B2PE(1.80f)}}}, {148000, {{FL2B2PE(1.40f), FL2B2PE(1.60f)}, {FL2B2PE(1.40f), FL2B2PE(1.80f)}}}}; static const BIT_PE_SFAC S_Bits2PeTab32000[] = { /* bitrate| afterburner off | afterburner on | | nCh=1 | nCh=2 | nCh=1 | nCh=2 */ {16000, {{FL2B2PE(1.20f), FL2B2PE(1.40f)}, {FL2B2PE(0.80f), FL2B2PE(0.80f)}}}, {24000, {{FL2B2PE(1.40f), FL2B2PE(1.20f)}, {FL2B2PE(1.00f), FL2B2PE(0.60f)}}}, {32000, {{FL2B2PE(1.20f), FL2B2PE(1.20f)}, {FL2B2PE(1.00f), FL2B2PE(0.80f)}}}, {48000, {{FL2B2PE(1.40f), FL2B2PE(1.40f)}, {FL2B2PE(1.20f), FL2B2PE(1.20f)}}}, {64000, {{FL2B2PE(1.60f), FL2B2PE(1.40f)}, {FL2B2PE(1.60f), FL2B2PE(1.20f)}}}, {96000, {{FL2B2PE(1.60f), FL2B2PE(1.40f)}, {FL2B2PE(1.60f), FL2B2PE(1.40f)}}}, {128000, {{FL2B2PE(1.80f), FL2B2PE(1.60f)}, {FL2B2PE(1.80f), FL2B2PE(1.60f)}}}, {148000, {{FL2B2PE(1.80f), FL2B2PE(1.60f)}, {FL2B2PE(1.80f), FL2B2PE(1.60f)}}}, {160000, {{FL2B2PE(1.80f), FL2B2PE(1.60f)}, {FL2B2PE(1.80f), FL2B2PE(1.60f)}}}, {200000, {{FL2B2PE(1.40f), FL2B2PE(1.60f)}, {FL2B2PE(1.40f), FL2B2PE(1.60f)}}}, {320000, {{FL2B2PE(3.20f), FL2B2PE(1.80f)}, {FL2B2PE(3.20f), FL2B2PE(1.80f)}}}}; static const BIT_PE_SFAC S_Bits2PeTab44100[] = { /* bitrate| afterburner off | afterburner on | | nCh=1 | nCh=2 | nCh=1 | nCh=2 */ {16000, {{FL2B2PE(1.20f), FL2B2PE(1.60f)}, {FL2B2PE(0.80f), FL2B2PE(1.00f)}}}, {24000, {{FL2B2PE(1.00f), FL2B2PE(1.20f)}, {FL2B2PE(1.00f), FL2B2PE(0.80f)}}}, {32000, {{FL2B2PE(1.20f), FL2B2PE(1.20f)}, {FL2B2PE(0.80f), FL2B2PE(0.60f)}}}, {48000, {{FL2B2PE(1.20f), FL2B2PE(1.20f)}, {FL2B2PE(1.20f), FL2B2PE(0.80f)}}}, {64000, {{FL2B2PE(1.40f), FL2B2PE(1.20f)}, {FL2B2PE(1.20f), FL2B2PE(1.00f)}}}, {96000, {{FL2B2PE(1.60f), FL2B2PE(1.20f)}, {FL2B2PE(1.60f), FL2B2PE(1.20f)}}}, {128000, {{FL2B2PE(1.60f), FL2B2PE(1.60f)}, {FL2B2PE(1.60f), FL2B2PE(1.40f)}}}, {148000, {{FL2B2PE(1.60f), FL2B2PE(1.60f)}, {FL2B2PE(1.60f), FL2B2PE(1.60f)}}}, {160000, {{FL2B2PE(1.60f), FL2B2PE(1.60f)}, {FL2B2PE(1.60f), FL2B2PE(1.60f)}}}, {200000, {{FL2B2PE(1.80f), FL2B2PE(1.60f)}, {FL2B2PE(1.60f), FL2B2PE(1.60f)}}}, {320000, {{FL2B2PE(3.20f), FL2B2PE(1.60f)}, {FL2B2PE(3.20f), FL2B2PE(1.60f)}}}}; static const BIT_PE_SFAC S_Bits2PeTab48000[] = { /* bitrate| afterburner off | afterburner on | | nCh=1 | nCh=2 | nCh=1 | nCh=2 */ {16000, {{FL2B2PE(1.40f), FL2B2PE(0.00f)}, {FL2B2PE(0.80f), FL2B2PE(0.00f)}}}, {24000, {{FL2B2PE(1.40f), FL2B2PE(1.20f)}, {FL2B2PE(1.00f), FL2B2PE(0.80f)}}}, {32000, {{FL2B2PE(1.00f), FL2B2PE(1.20f)}, {FL2B2PE(0.60f), FL2B2PE(0.80f)}}}, {48000, {{FL2B2PE(1.20f), FL2B2PE(1.00f)}, {FL2B2PE(0.80f), FL2B2PE(0.80f)}}}, {64000, {{FL2B2PE(1.20f), FL2B2PE(1.20f)}, {FL2B2PE(1.20f), FL2B2PE(1.00f)}}}, {96000, {{FL2B2PE(1.60f), FL2B2PE(1.40f)}, {FL2B2PE(1.60f), FL2B2PE(1.20f)}}}, {128000, {{FL2B2PE(1.60f), FL2B2PE(1.60f)}, {FL2B2PE(1.60f), FL2B2PE(1.40f)}}}, {148000, {{FL2B2PE(1.60f), FL2B2PE(1.60f)}, {FL2B2PE(1.60f), FL2B2PE(1.40f)}}}, {160000, {{FL2B2PE(1.60f), FL2B2PE(1.60f)}, {FL2B2PE(1.60f), FL2B2PE(1.40f)}}}, {200000, {{FL2B2PE(1.20f), FL2B2PE(1.60f)}, {FL2B2PE(1.60f), FL2B2PE(1.40f)}}}, {320000, {{FL2B2PE(3.20f), FL2B2PE(1.60f)}, {FL2B2PE(3.20f), FL2B2PE(1.60f)}}}}; static const BITS2PE_CFG_TAB bits2PeConfigTab[] = { {16000, S_Bits2PeTab16000, sizeof(S_Bits2PeTab16000) / sizeof(BIT_PE_SFAC)}, {22050, S_Bits2PeTab22050, sizeof(S_Bits2PeTab22050) / sizeof(BIT_PE_SFAC)}, {24000, S_Bits2PeTab24000, sizeof(S_Bits2PeTab24000) / sizeof(BIT_PE_SFAC)}, {32000, S_Bits2PeTab32000, sizeof(S_Bits2PeTab32000) / sizeof(BIT_PE_SFAC)}, {44100, S_Bits2PeTab44100, sizeof(S_Bits2PeTab44100) / sizeof(BIT_PE_SFAC)}, {48000, S_Bits2PeTab48000, sizeof(S_Bits2PeTab48000) / sizeof(BIT_PE_SFAC)}}; /* values for avoid hole flag */ enum _avoid_hole_state { NO_AH = 0, AH_INACTIVE = 1, AH_ACTIVE = 2 }; /* Q format definitions */ #define Q_BITFAC \ (24) /* Q scaling used in FDKaacEnc_bitresCalcBitFac() calculation */ #define Q_AVGBITS (17) /* scale bit values */ /***************************************************************************** functionname: FDKaacEnc_InitBits2PeFactor description: retrieve bits2PeFactor from table *****************************************************************************/ static void FDKaacEnc_InitBits2PeFactor( FIXP_DBL *bits2PeFactor_m, INT *bits2PeFactor_e, const INT bitRate, const INT nChannels, const INT sampleRate, const INT advancedBitsToPe, const INT dZoneQuantEnable, const INT invQuant) { /**** 1) Set default bits2pe factor ****/ FIXP_DBL bit2PE_m = FL2FXCONST_DBL(1.18f / (1 << (1))); INT bit2PE_e = 1; /**** 2) For AAC-(E)LD, make use of advanced bits to pe factor table ****/ if (advancedBitsToPe && nChannels <= (2)) { int i; const BIT_PE_SFAC *peTab = NULL; INT size = 0; /*** 2.1) Get correct table entry ***/ for (i = 0; i < (INT)(sizeof(bits2PeConfigTab) / sizeof(BITS2PE_CFG_TAB)); i++) { if (sampleRate >= bits2PeConfigTab[i].sampleRate) { peTab = bits2PeConfigTab[i].pPeTab; size = bits2PeConfigTab[i].nEntries; } } if ((peTab != NULL) && (size != 0)) { INT startB = -1; /* bitrate entry in table that is the next-lower to actual bitrate */ INT stopB = -1; /* bitrate entry in table that is the next-higher to actual bitrate */ FIXP_DBL startPF = FL2FXCONST_DBL(0.0f); /* bits2PE factor entry in table that is the next-lower to actual bits2PE factor */ FIXP_DBL stopPF = FL2FXCONST_DBL(0.0f); /* bits2PE factor entry in table that is the next-higher to actual bits2PE factor */ FIXP_DBL slope = FL2FXCONST_DBL( 0.0f); /* the slope from the start bits2Pe entry to the next one */ const int qualityIdx = (invQuant == 0) ? 0 : 1; if (bitRate >= peTab[size - 1].bitrate) { /* Chosen bitrate is higher than the highest bitrate in table. The slope for extrapolating the bits2PE factor must be zero. Values are set accordingly. */ startB = peTab[size - 1].bitrate; stopB = bitRate + 1; /* Can be an arbitrary value greater than startB and bitrate. */ startPF = peTab[size - 1].bits2PeFactor[qualityIdx][nChannels - 1]; stopPF = peTab[size - 1].bits2PeFactor[qualityIdx][nChannels - 1]; } else { for (i = 0; i < size - 1; i++) { if ((peTab[i].bitrate <= bitRate) && (peTab[i + 1].bitrate > bitRate)) { startB = peTab[i].bitrate; stopB = peTab[i + 1].bitrate; startPF = peTab[i].bits2PeFactor[qualityIdx][nChannels - 1]; stopPF = peTab[i + 1].bits2PeFactor[qualityIdx][nChannels - 1]; break; } } } /*** 2.2) Configuration available? ***/ if (startB != -1) { /** 2.2.1) linear interpolate to actual PEfactor **/ FIXP_DBL bit2PE = 0; const FIXP_DBL maxBit2PE = FL2FXCONST_DBL(3.f / 4.f); /* bit2PE = ((stopPF-startPF)/(stopB-startB))*(bitRate-startB)+startPF; */ slope = fDivNorm(bitRate - startB, stopB - startB); bit2PE = fMult(slope, stopPF - startPF) + startPF; bit2PE = fMin(maxBit2PE, bit2PE); /** 2.2.2) sanity check if bits2pe value is high enough **/ if (bit2PE >= (FL2FXCONST_DBL(0.35f) >> 2)) { bit2PE_m = bit2PE; bit2PE_e = 2; /* table is fixed scaled */ } } /* br */ } /* sr */ } /* advancedBitsToPe */ if (dZoneQuantEnable) { if (bit2PE_m >= (FL2FXCONST_DBL(0.6f)) >> bit2PE_e) { /* Additional headroom for addition */ bit2PE_m >>= 1; bit2PE_e += 1; } /* the quantTendencyCompensator compensates a lower bit consumption due to * increasing the tendency to quantize low spectral values to the lower * quantizer border for bitrates below a certain bitrate threshold --> see * also function calcSfbDistLD in quantize.c */ if ((bitRate / nChannels > 32000) && (bitRate / nChannels <= 40000)) { bit2PE_m += (FL2FXCONST_DBL(0.4f)) >> bit2PE_e; } else if (bitRate / nChannels > 20000) { bit2PE_m += (FL2FXCONST_DBL(0.3f)) >> bit2PE_e; } else if (bitRate / nChannels >= 16000) { bit2PE_m += (FL2FXCONST_DBL(0.3f)) >> bit2PE_e; } else { bit2PE_m += (FL2FXCONST_DBL(0.0f)) >> bit2PE_e; } } /***** 3.) Return bits2pe factor *****/ *bits2PeFactor_m = bit2PE_m; *bits2PeFactor_e = bit2PE_e; } /***************************************************************************** functionname: FDKaacEnc_bits2pe2 description: convert from bits to pe *****************************************************************************/ FDK_INLINE INT FDKaacEnc_bits2pe2(const INT bits, const FIXP_DBL factor_m, const INT factor_e) { return (INT)(fMult(factor_m, (FIXP_DBL)(bits << Q_AVGBITS)) >> (Q_AVGBITS - factor_e)); } /***************************************************************************** functionname: FDKaacEnc_calcThreshExp description: loudness calculation (threshold to the power of redExp) *****************************************************************************/ static void FDKaacEnc_calcThreshExp( FIXP_DBL thrExp[(2)][MAX_GROUPED_SFB], const QC_OUT_CHANNEL *const qcOutChannel[(2)], const PSY_OUT_CHANNEL *const psyOutChannel[(2)], const INT nChannels) { INT ch, sfb, sfbGrp; FIXP_DBL thrExpLdData; for (ch = 0; ch < nChannels; ch++) { for (sfbGrp = 0; sfbGrp < psyOutChannel[ch]->sfbCnt; sfbGrp += psyOutChannel[ch]->sfbPerGroup) { for (sfb = 0; sfb < psyOutChannel[ch]->maxSfbPerGroup; sfb++) { thrExpLdData = psyOutChannel[ch]->sfbThresholdLdData[sfbGrp + sfb] >> 2; thrExp[ch][sfbGrp + sfb] = CalcInvLdData(thrExpLdData); } } } } /***************************************************************************** functionname: FDKaacEnc_adaptMinSnr description: reduce minSnr requirements for bands with relative low energies *****************************************************************************/ static void FDKaacEnc_adaptMinSnr( QC_OUT_CHANNEL *const qcOutChannel[(2)], const PSY_OUT_CHANNEL *const psyOutChannel[(2)], const MINSNR_ADAPT_PARAM *const msaParam, const INT nChannels) { INT ch, sfb, sfbGrp, nSfb; FIXP_DBL avgEnLD64, dbRatio, minSnrRed; FIXP_DBL minSnrLimitLD64 = FL2FXCONST_DBL(-0.00503012648262f); /* ld64(0.8f) */ FIXP_DBL nSfbLD64; FIXP_DBL accu; FIXP_DBL msaParam_maxRed = msaParam->maxRed; FIXP_DBL msaParam_startRatio = msaParam->startRatio; FIXP_DBL msaParam_redRatioFac = fMult(msaParam->redRatioFac, FL2FXCONST_DBL(0.3010299956f)); FIXP_DBL msaParam_redOffs = msaParam->redOffs; for (ch = 0; ch < nChannels; ch++) { /* calc average energy per scalefactor band */ nSfb = 0; accu = FL2FXCONST_DBL(0.0f); DWORD_ALIGNED(psyOutChannel[ch]->sfbEnergy); for (sfbGrp = 0; sfbGrp < psyOutChannel[ch]->sfbCnt; sfbGrp += psyOutChannel[ch]->sfbPerGroup) { int maxSfbPerGroup = psyOutChannel[ch]->maxSfbPerGroup; nSfb += maxSfbPerGroup; for (sfb = 0; sfb < maxSfbPerGroup; sfb++) { accu += psyOutChannel[ch]->sfbEnergy[sfbGrp + sfb] >> 6; } } if ((accu == FL2FXCONST_DBL(0.0f)) || (nSfb == 0)) { avgEnLD64 = FL2FXCONST_DBL(-1.0f); } else { nSfbLD64 = CalcLdInt(nSfb); avgEnLD64 = CalcLdData(accu); avgEnLD64 = avgEnLD64 + FL2FXCONST_DBL(0.09375f) - nSfbLD64; /* 0.09375f: compensate shift with 6 */ } /* reduce minSnr requirement by minSnr^minSnrRed dependent on avgEn/sfbEn */ int maxSfbPerGroup = psyOutChannel[ch]->maxSfbPerGroup; int sfbCnt = psyOutChannel[ch]->sfbCnt; int sfbPerGroup = psyOutChannel[ch]->sfbPerGroup; for (sfbGrp = 0; sfbGrp < sfbCnt; sfbGrp += sfbPerGroup) { FIXP_DBL *RESTRICT psfbEnergyLdData = &qcOutChannel[ch]->sfbEnergyLdData[sfbGrp]; FIXP_DBL *RESTRICT psfbMinSnrLdData = &qcOutChannel[ch]->sfbMinSnrLdData[sfbGrp]; for (sfb = 0; sfb < maxSfbPerGroup; sfb++) { FIXP_DBL sfbEnergyLdData = *psfbEnergyLdData++; FIXP_DBL sfbMinSnrLdData = *psfbMinSnrLdData; dbRatio = avgEnLD64 - sfbEnergyLdData; int update = (msaParam_startRatio < dbRatio) ? 1 : 0; minSnrRed = msaParam_redOffs + fMult(msaParam_redRatioFac, dbRatio); /* scaled by 1.0f/64.0f*/ minSnrRed = fixMax(minSnrRed, msaParam_maxRed); /* scaled by 1.0f/64.0f*/ minSnrRed = (fMult(sfbMinSnrLdData, minSnrRed)) << 6; minSnrRed = fixMin(minSnrLimitLD64, minSnrRed); *psfbMinSnrLdData++ = update ? minSnrRed : sfbMinSnrLdData; } } } } /***************************************************************************** functionname: FDKaacEnc_initAvoidHoleFlag description: determine bands where avoid hole is not necessary resp. possible *****************************************************************************/ static void FDKaacEnc_initAvoidHoleFlag( QC_OUT_CHANNEL *const qcOutChannel[(2)], const PSY_OUT_CHANNEL *const psyOutChannel[(2)], UCHAR ahFlag[(2)][MAX_GROUPED_SFB], const struct TOOLSINFO *const toolsInfo, const INT nChannels, const AH_PARAM *const ahParam) { INT ch, sfb, sfbGrp; FIXP_DBL sfbEn, sfbEnm1; FIXP_DBL sfbEnLdData; FIXP_DBL avgEnLdData; /* decrease spread energy by 3dB for long blocks, resp. 2dB for shorts (avoid more holes in long blocks) */ for (ch = 0; ch < nChannels; ch++) { QC_OUT_CHANNEL *const qcOutChan = qcOutChannel[ch]; if (psyOutChannel[ch]->lastWindowSequence != SHORT_WINDOW) { for (sfbGrp = 0; sfbGrp < psyOutChannel[ch]->sfbCnt; sfbGrp += psyOutChannel[ch]->sfbPerGroup) for (sfb = 0; sfb < psyOutChannel[ch]->maxSfbPerGroup; sfb++) qcOutChan->sfbSpreadEnergy[sfbGrp + sfb] >>= 1; } else { for (sfbGrp = 0; sfbGrp < psyOutChannel[ch]->sfbCnt; sfbGrp += psyOutChannel[ch]->sfbPerGroup) for (sfb = 0; sfb < psyOutChannel[ch]->maxSfbPerGroup; sfb++) qcOutChan->sfbSpreadEnergy[sfbGrp + sfb] = fMult( FL2FXCONST_DBL(0.63f), qcOutChan->sfbSpreadEnergy[sfbGrp + sfb]); } } /* increase minSnr for local peaks, decrease it for valleys */ if (ahParam->modifyMinSnr) { for (ch = 0; ch < nChannels; ch++) { QC_OUT_CHANNEL *const qcOutChan = qcOutChannel[ch]; for (sfbGrp = 0; sfbGrp < psyOutChannel[ch]->sfbCnt; sfbGrp += psyOutChannel[ch]->sfbPerGroup) { for (sfb = 0; sfb < psyOutChannel[ch]->maxSfbPerGroup; sfb++) { FIXP_DBL sfbEnp1, avgEn; if (sfb > 0) sfbEnm1 = qcOutChan->sfbEnergy[sfbGrp + sfb - 1]; else sfbEnm1 = qcOutChan->sfbEnergy[sfbGrp + sfb]; if (sfb < psyOutChannel[ch]->maxSfbPerGroup - 1) sfbEnp1 = qcOutChan->sfbEnergy[sfbGrp + sfb + 1]; else sfbEnp1 = qcOutChan->sfbEnergy[sfbGrp + sfb]; avgEn = (sfbEnm1 >> 1) + (sfbEnp1 >> 1); avgEnLdData = CalcLdData(avgEn); sfbEn = qcOutChan->sfbEnergy[sfbGrp + sfb]; sfbEnLdData = qcOutChan->sfbEnergyLdData[sfbGrp + sfb]; /* peak ? */ if (sfbEn > avgEn) { FIXP_DBL tmpMinSnrLdData; if (psyOutChannel[ch]->lastWindowSequence == LONG_WINDOW) tmpMinSnrLdData = SnrLdFac + fixMax(avgEnLdData - sfbEnLdData, SnrLdMin1 - SnrLdFac); else tmpMinSnrLdData = SnrLdFac + fixMax(avgEnLdData - sfbEnLdData, SnrLdMin3 - SnrLdFac); qcOutChan->sfbMinSnrLdData[sfbGrp + sfb] = fixMin( qcOutChan->sfbMinSnrLdData[sfbGrp + sfb], tmpMinSnrLdData); } /* valley ? */ if (((sfbEnLdData + (FIXP_DBL)SnrLdMin4) < (FIXP_DBL)avgEnLdData) && (sfbEn > FL2FXCONST_DBL(0.0))) { FIXP_DBL tmpMinSnrLdData = avgEnLdData - sfbEnLdData - (FIXP_DBL)SnrLdMin4 + qcOutChan->sfbMinSnrLdData[sfbGrp + sfb]; tmpMinSnrLdData = fixMin((FIXP_DBL)SnrLdFac, tmpMinSnrLdData); qcOutChan->sfbMinSnrLdData[sfbGrp + sfb] = fixMin(tmpMinSnrLdData, (FIXP_DBL)(qcOutChan->sfbMinSnrLdData[sfbGrp + sfb] + SnrLdMin2)); } } } } } /* stereo: adapt the minimum requirements sfbMinSnr of mid and side channels to avoid spending unnoticable bits */ if (nChannels == 2) { QC_OUT_CHANNEL *qcOutChanM = qcOutChannel[0]; QC_OUT_CHANNEL *qcOutChanS = qcOutChannel[1]; const PSY_OUT_CHANNEL *const psyOutChanM = psyOutChannel[0]; for (sfbGrp = 0; sfbGrp < psyOutChanM->sfbCnt; sfbGrp += psyOutChanM->sfbPerGroup) { for (sfb = 0; sfb < psyOutChanM->maxSfbPerGroup; sfb++) { if (toolsInfo->msMask[sfbGrp + sfb]) { FIXP_DBL maxSfbEnLd = fixMax(qcOutChanM->sfbEnergyLdData[sfbGrp + sfb], qcOutChanS->sfbEnergyLdData[sfbGrp + sfb]); FIXP_DBL maxThrLd, sfbMinSnrTmpLd; if (((SnrLdMin5 >> 1) + (maxSfbEnLd >> 1) + (qcOutChanM->sfbMinSnrLdData[sfbGrp + sfb] >> 1)) <= FL2FXCONST_DBL(-0.5f)) maxThrLd = FL2FXCONST_DBL(-1.0f); else maxThrLd = SnrLdMin5 + maxSfbEnLd + qcOutChanM->sfbMinSnrLdData[sfbGrp + sfb]; if (qcOutChanM->sfbEnergy[sfbGrp + sfb] > FL2FXCONST_DBL(0.0f)) sfbMinSnrTmpLd = maxThrLd - qcOutChanM->sfbEnergyLdData[sfbGrp + sfb]; else sfbMinSnrTmpLd = FL2FXCONST_DBL(0.0f); qcOutChanM->sfbMinSnrLdData[sfbGrp + sfb] = fixMax(qcOutChanM->sfbMinSnrLdData[sfbGrp + sfb], sfbMinSnrTmpLd); if (qcOutChanM->sfbMinSnrLdData[sfbGrp + sfb] <= FL2FXCONST_DBL(0.0f)) qcOutChanM->sfbMinSnrLdData[sfbGrp + sfb] = fixMin( qcOutChanM->sfbMinSnrLdData[sfbGrp + sfb], (FIXP_DBL)SnrLdFac); if (qcOutChanS->sfbEnergy[sfbGrp + sfb] > FL2FXCONST_DBL(0.0f)) sfbMinSnrTmpLd = maxThrLd - qcOutChanS->sfbEnergyLdData[sfbGrp + sfb]; else sfbMinSnrTmpLd = FL2FXCONST_DBL(0.0f); qcOutChanS->sfbMinSnrLdData[sfbGrp + sfb] = fixMax(qcOutChanS->sfbMinSnrLdData[sfbGrp + sfb], sfbMinSnrTmpLd); if (qcOutChanS->sfbMinSnrLdData[sfbGrp + sfb] <= FL2FXCONST_DBL(0.0f)) qcOutChanS->sfbMinSnrLdData[sfbGrp + sfb] = fixMin( qcOutChanS->sfbMinSnrLdData[sfbGrp + sfb], (FIXP_DBL)SnrLdFac); if (qcOutChanM->sfbEnergy[sfbGrp + sfb] > qcOutChanM->sfbSpreadEnergy[sfbGrp + sfb]) qcOutChanS->sfbSpreadEnergy[sfbGrp + sfb] = fMult( qcOutChanS->sfbEnergy[sfbGrp + sfb], FL2FXCONST_DBL(0.9f)); if (qcOutChanS->sfbEnergy[sfbGrp + sfb] > qcOutChanS->sfbSpreadEnergy[sfbGrp + sfb]) qcOutChanM->sfbSpreadEnergy[sfbGrp + sfb] = fMult( qcOutChanM->sfbEnergy[sfbGrp + sfb], FL2FXCONST_DBL(0.9f)); } /* if (toolsInfo->msMask[sfbGrp+sfb]) */ } /* sfb */ } /* sfbGrp */ } /* nChannels==2 */ /* init ahFlag (0: no ah necessary, 1: ah possible, 2: ah active */ for (ch = 0; ch < nChannels; ch++) { QC_OUT_CHANNEL *qcOutChan = qcOutChannel[ch]; const PSY_OUT_CHANNEL *const psyOutChan = psyOutChannel[ch]; for (sfbGrp = 0; sfbGrp < psyOutChan->sfbCnt; sfbGrp += psyOutChan->sfbPerGroup) { for (sfb = 0; sfb < psyOutChan->maxSfbPerGroup; sfb++) { if ((qcOutChan->sfbSpreadEnergy[sfbGrp + sfb] > qcOutChan->sfbEnergy[sfbGrp + sfb]) || (qcOutChan->sfbMinSnrLdData[sfbGrp + sfb] > FL2FXCONST_DBL(0.0f))) { ahFlag[ch][sfbGrp + sfb] = NO_AH; } else { ahFlag[ch][sfbGrp + sfb] = AH_INACTIVE; } } } } } /** * \brief Calculate constants that do not change during successive pe * calculations. * * \param peData Pointer to structure containing PE data of * current element. * \param psyOutChannel Pointer to PSY_OUT_CHANNEL struct holding * nChannels elements. * \param qcOutChannel Pointer to QC_OUT_CHANNEL struct holding * nChannels elements. * \param nChannels Number of channels in element. * \param peOffset Fixed PE offset defined while * FDKaacEnc_AdjThrInit() depending on bitrate. * * \return void */ static void FDKaacEnc_preparePe(PE_DATA *const peData, const PSY_OUT_CHANNEL *const psyOutChannel[(2)], const QC_OUT_CHANNEL *const qcOutChannel[(2)], const INT nChannels, const INT peOffset) { INT ch; for (ch = 0; ch < nChannels; ch++) { const PSY_OUT_CHANNEL *const psyOutChan = psyOutChannel[ch]; FDKaacEnc_prepareSfbPe( &peData->peChannelData[ch], psyOutChan->sfbEnergyLdData, psyOutChan->sfbThresholdLdData, qcOutChannel[ch]->sfbFormFactorLdData, psyOutChan->sfbOffsets, psyOutChan->sfbCnt, psyOutChan->sfbPerGroup, psyOutChan->maxSfbPerGroup); } peData->offset = peOffset; } /** * \brief Calculate weighting factor for threshold adjustment. * * Calculate weighting factor to be applied at energies and thresholds in ld64 * format. * * \param peData, Pointer to PE data in current element. * \param psyOutChannel Pointer to PSY_OUT_CHANNEL struct holding * nChannels elements. * \param qcOutChannel Pointer to QC_OUT_CHANNEL struct holding * nChannels elements. * \param toolsInfo Pointer to tools info struct of current element. * \param adjThrStateElement Pointer to ATS_ELEMENT holding enFacPatch * states. * \param nChannels Number of channels in element. * \param usePatchTool Apply the weighting tool 0 (no) else (yes). * * \return void */ static void FDKaacEnc_calcWeighting( const PE_DATA *const peData, const PSY_OUT_CHANNEL *const psyOutChannel[(2)], QC_OUT_CHANNEL *const qcOutChannel[(2)], const struct TOOLSINFO *const toolsInfo, ATS_ELEMENT *const adjThrStateElement, const INT nChannels, const INT usePatchTool) { int ch, noShortWindowInFrame = TRUE; INT exePatchM = 0; for (ch = 0; ch < nChannels; ch++) { if (psyOutChannel[ch]->lastWindowSequence == SHORT_WINDOW) { noShortWindowInFrame = FALSE; } FDKmemclear(qcOutChannel[ch]->sfbEnFacLd, MAX_GROUPED_SFB * sizeof(FIXP_DBL)); } if (usePatchTool == 0) { return; /* tool is disabled */ } for (ch = 0; ch < nChannels; ch++) { const PSY_OUT_CHANNEL *const psyOutChan = psyOutChannel[ch]; if (noShortWindowInFrame) { /* retain energy ratio between blocks of different length */ FIXP_DBL nrgSum14, nrgSum12, nrgSum34, nrgTotal; FIXP_DBL nrgFacLd_14, nrgFacLd_12, nrgFacLd_34; INT usePatch, exePatch; int sfb, sfbGrp, nLinesSum = 0; nrgSum14 = nrgSum12 = nrgSum34 = nrgTotal = FL2FXCONST_DBL(0.f); /* calculate flatness of audible spectrum, i.e. spectrum above masking * threshold. */ for (sfbGrp = 0; sfbGrp < psyOutChannel[ch]->sfbCnt; sfbGrp += psyOutChannel[ch]->sfbPerGroup) { for (sfb = 0; sfb < psyOutChannel[ch]->maxSfbPerGroup; sfb++) { FIXP_DBL nrgFac12 = CalcInvLdData( psyOutChan->sfbEnergyLdData[sfbGrp + sfb] >> 1); /* nrg^(1/2) */ FIXP_DBL nrgFac14 = CalcInvLdData( psyOutChan->sfbEnergyLdData[sfbGrp + sfb] >> 2); /* nrg^(1/4) */ /* maximal number of bands is 64, results scaling factor 6 */ nLinesSum += peData->peChannelData[ch] .sfbNLines[sfbGrp + sfb]; /* relevant lines */ nrgTotal += (psyOutChan->sfbEnergy[sfbGrp + sfb] >> 6); /* sum up nrg */ nrgSum12 += (nrgFac12 >> 6); /* sum up nrg^(2/4) */ nrgSum14 += (nrgFac14 >> 6); /* sum up nrg^(1/4) */ nrgSum34 += (fMult(nrgFac14, nrgFac12) >> 6); /* sum up nrg^(3/4) */ } } nrgTotal = CalcLdData(nrgTotal); /* get ld64 of total nrg */ nrgFacLd_14 = CalcLdData(nrgSum14) - nrgTotal; /* ld64(nrgSum14/nrgTotal) */ nrgFacLd_12 = CalcLdData(nrgSum12) - nrgTotal; /* ld64(nrgSum12/nrgTotal) */ nrgFacLd_34 = CalcLdData(nrgSum34) - nrgTotal; /* ld64(nrgSum34/nrgTotal) */ /* Note: nLinesSum cannot be larger than the number of total lines, thats * taken care of in line_pe.cpp FDKaacEnc_prepareSfbPe() */ adjThrStateElement->chaosMeasureEnFac[ch] = fMax(FL2FXCONST_DBL(0.1875f), fDivNorm(nLinesSum, psyOutChan->sfbOffsets[psyOutChan->sfbCnt])); usePatch = (adjThrStateElement->chaosMeasureEnFac[ch] > FL2FXCONST_DBL(0.78125f)); exePatch = ((usePatch) && (adjThrStateElement->lastEnFacPatch[ch])); for (sfbGrp = 0; sfbGrp < psyOutChannel[ch]->sfbCnt; sfbGrp += psyOutChannel[ch]->sfbPerGroup) { for (sfb = 0; sfb < psyOutChannel[ch]->maxSfbPerGroup; sfb++) { INT sfbExePatch; /* for MS coupled SFBs, also execute patch in side channel if done in * mid channel */ if ((ch == 1) && (toolsInfo->msMask[sfbGrp + sfb])) { sfbExePatch = exePatchM; } else { sfbExePatch = exePatch; } if ((sfbExePatch) && (psyOutChan->sfbEnergy[sfbGrp + sfb] > FL2FXCONST_DBL(0.f))) { /* execute patch based on spectral flatness calculated above */ if (adjThrStateElement->chaosMeasureEnFac[ch] > FL2FXCONST_DBL(0.8125f)) { qcOutChannel[ch]->sfbEnFacLd[sfbGrp + sfb] = ((nrgFacLd_14 + (psyOutChan->sfbEnergyLdData[sfbGrp + sfb] + (psyOutChan->sfbEnergyLdData[sfbGrp + sfb] >> 1))) >> 1); /* sfbEnergy^(3/4) */ } else if (adjThrStateElement->chaosMeasureEnFac[ch] > FL2FXCONST_DBL(0.796875f)) { qcOutChannel[ch]->sfbEnFacLd[sfbGrp + sfb] = ((nrgFacLd_12 + psyOutChan->sfbEnergyLdData[sfbGrp + sfb]) >> 1); /* sfbEnergy^(2/4) */ } else { qcOutChannel[ch]->sfbEnFacLd[sfbGrp + sfb] = ((nrgFacLd_34 + (psyOutChan->sfbEnergyLdData[sfbGrp + sfb] >> 1)) >> 1); /* sfbEnergy^(1/4) */ } qcOutChannel[ch]->sfbEnFacLd[sfbGrp + sfb] = fixMin(qcOutChannel[ch]->sfbEnFacLd[sfbGrp + sfb], (FIXP_DBL)0); } } } /* sfb loop */ adjThrStateElement->lastEnFacPatch[ch] = usePatch; exePatchM = exePatch; } else { /* !noShortWindowInFrame */ adjThrStateElement->chaosMeasureEnFac[ch] = FL2FXCONST_DBL(0.75f); adjThrStateElement->lastEnFacPatch[ch] = TRUE; /* allow use of sfbEnFac patch in upcoming frame */ } } /* ch loop */ } /***************************************************************************** functionname: FDKaacEnc_calcPe description: calculate pe for both channels *****************************************************************************/ static void FDKaacEnc_calcPe(const PSY_OUT_CHANNEL *const psyOutChannel[(2)], const QC_OUT_CHANNEL *const qcOutChannel[(2)], PE_DATA *const peData, const INT nChannels) { INT ch; peData->pe = peData->offset; peData->constPart = 0; peData->nActiveLines = 0; for (ch = 0; ch < nChannels; ch++) { PE_CHANNEL_DATA *peChanData = &peData->peChannelData[ch]; FDKaacEnc_calcSfbPe( peChanData, qcOutChannel[ch]->sfbWeightedEnergyLdData, qcOutChannel[ch]->sfbThresholdLdData, psyOutChannel[ch]->sfbCnt, psyOutChannel[ch]->sfbPerGroup, psyOutChannel[ch]->maxSfbPerGroup, psyOutChannel[ch]->isBook, psyOutChannel[ch]->isScale); peData->pe += peChanData->pe; peData->constPart += peChanData->constPart; peData->nActiveLines += peChanData->nActiveLines; } } void FDKaacEnc_peCalculation(PE_DATA *const peData, const PSY_OUT_CHANNEL *const psyOutChannel[(2)], QC_OUT_CHANNEL *const qcOutChannel[(2)], const struct TOOLSINFO *const toolsInfo, ATS_ELEMENT *const adjThrStateElement, const INT nChannels) { /* constants that will not change during successive pe calculations */ FDKaacEnc_preparePe(peData, psyOutChannel, qcOutChannel, nChannels, adjThrStateElement->peOffset); /* calculate weighting factor for threshold adjustment */ FDKaacEnc_calcWeighting(peData, psyOutChannel, qcOutChannel, toolsInfo, adjThrStateElement, nChannels, 1); { /* no weighting of threholds and energies for mlout */ /* weight energies and thresholds */ int ch; for (ch = 0; ch < nChannels; ch++) { int sfb, sfbGrp; QC_OUT_CHANNEL *pQcOutCh = qcOutChannel[ch]; for (sfbGrp = 0; sfbGrp < psyOutChannel[ch]->sfbCnt; sfbGrp += psyOutChannel[ch]->sfbPerGroup) { for (sfb = 0; sfb < psyOutChannel[ch]->maxSfbPerGroup; sfb++) { pQcOutCh->sfbWeightedEnergyLdData[sfb + sfbGrp] = pQcOutCh->sfbEnergyLdData[sfb + sfbGrp] - pQcOutCh->sfbEnFacLd[sfb + sfbGrp]; pQcOutCh->sfbThresholdLdData[sfb + sfbGrp] -= pQcOutCh->sfbEnFacLd[sfb + sfbGrp]; } } } } /* pe without reduction */ FDKaacEnc_calcPe(psyOutChannel, qcOutChannel, peData, nChannels); } /***************************************************************************** functionname: FDKaacEnc_FDKaacEnc_calcPeNoAH description: sum the pe data only for bands where avoid hole is inactive *****************************************************************************/ #define CONSTPART_HEADROOM 4 static void FDKaacEnc_FDKaacEnc_calcPeNoAH( INT *const pe, INT *const constPart, INT *const nActiveLines, const PE_DATA *const peData, const UCHAR ahFlag[(2)][MAX_GROUPED_SFB], const PSY_OUT_CHANNEL *const psyOutChannel[(2)], const INT nChannels) { INT ch, sfb, sfbGrp; INT pe_tmp = peData->offset; INT constPart_tmp = 0; INT nActiveLines_tmp = 0; for (ch = 0; ch < nChannels; ch++) { const PE_CHANNEL_DATA *const peChanData = &peData->peChannelData[ch]; for (sfbGrp = 0; sfbGrp < psyOutChannel[ch]->sfbCnt; sfbGrp += psyOutChannel[ch]->sfbPerGroup) { for (sfb = 0; sfb < psyOutChannel[ch]->maxSfbPerGroup; sfb++) { if (ahFlag[ch][sfbGrp + sfb] < AH_ACTIVE) { pe_tmp += peChanData->sfbPe[sfbGrp + sfb]; constPart_tmp += peChanData->sfbConstPart[sfbGrp + sfb] >> CONSTPART_HEADROOM; nActiveLines_tmp += peChanData->sfbNActiveLines[sfbGrp + sfb]; } } } } /* correct scaled pe and constPart values */ *pe = pe_tmp >> PE_CONSTPART_SHIFT; *constPart = constPart_tmp >> (PE_CONSTPART_SHIFT - CONSTPART_HEADROOM); *nActiveLines = nActiveLines_tmp; } /***************************************************************************** functionname: FDKaacEnc_reduceThresholdsCBR description: apply reduction formula *****************************************************************************/ static const FIXP_DBL limitThrReducedLdData = (FIXP_DBL)0x00008000; /*FL2FXCONST_DBL(FDKpow(2.0,-LD_DATA_SCALING/4.0));*/ static void FDKaacEnc_reduceThresholdsCBR( QC_OUT_CHANNEL *const qcOutChannel[(2)], const PSY_OUT_CHANNEL *const psyOutChannel[(2)], UCHAR ahFlag[(2)][MAX_GROUPED_SFB], const FIXP_DBL thrExp[(2)][MAX_GROUPED_SFB], const INT nChannels, const FIXP_DBL redVal_m, const SCHAR redVal_e) { INT ch, sfb, sfbGrp; FIXP_DBL sfbEnLdData, sfbThrLdData, sfbThrReducedLdData; FIXP_DBL sfbThrExp; for (ch = 0; ch < nChannels; ch++) { QC_OUT_CHANNEL *qcOutChan = qcOutChannel[ch]; for (sfbGrp = 0; sfbGrp < psyOutChannel[ch]->sfbCnt; sfbGrp += psyOutChannel[ch]->sfbPerGroup) { for (sfb = 0; sfb < psyOutChannel[ch]->maxSfbPerGroup; sfb++) { sfbEnLdData = qcOutChan->sfbWeightedEnergyLdData[sfbGrp + sfb]; sfbThrLdData = qcOutChan->sfbThresholdLdData[sfbGrp + sfb]; sfbThrExp = thrExp[ch][sfbGrp + sfb]; if ((sfbEnLdData > sfbThrLdData) && (ahFlag[ch][sfbGrp + sfb] != AH_ACTIVE)) { /* threshold reduction formula: float tmp = thrExp[ch][sfb]+redVal; tmp *= tmp; sfbThrReduced = tmp*tmp; */ int minScale = fixMin(CountLeadingBits(sfbThrExp), CountLeadingBits(redVal_m) - redVal_e) - 1; /* 4*log( sfbThrExp + redVal ) */ sfbThrReducedLdData = CalcLdData(fAbs(scaleValue(sfbThrExp, minScale) + scaleValue(redVal_m, redVal_e + minScale))) - (FIXP_DBL)(minScale << (DFRACT_BITS - 1 - LD_DATA_SHIFT)); sfbThrReducedLdData <<= 2; /* avoid holes */ if ((sfbThrReducedLdData > (qcOutChan->sfbMinSnrLdData[sfbGrp + sfb] + sfbEnLdData)) && (ahFlag[ch][sfbGrp + sfb] != NO_AH)) { if (qcOutChan->sfbMinSnrLdData[sfbGrp + sfb] > (FL2FXCONST_DBL(-1.0f) - sfbEnLdData)) { sfbThrReducedLdData = fixMax( (qcOutChan->sfbMinSnrLdData[sfbGrp + sfb] + sfbEnLdData), sfbThrLdData); } else sfbThrReducedLdData = sfbThrLdData; ahFlag[ch][sfbGrp + sfb] = AH_ACTIVE; } /* minimum of 29 dB Ratio for Thresholds */ if ((sfbEnLdData + (FIXP_DBL)MAXVAL_DBL) > FL2FXCONST_DBL(9.6336206 / LD_DATA_SCALING)) { sfbThrReducedLdData = fixMax( sfbThrReducedLdData, (sfbEnLdData - FL2FXCONST_DBL(9.6336206 / LD_DATA_SCALING))); } qcOutChan->sfbThresholdLdData[sfbGrp + sfb] = sfbThrReducedLdData; } } } } } /* similar to prepareSfbPe1() */ static FIXP_DBL FDKaacEnc_calcChaosMeasure( const PSY_OUT_CHANNEL *const psyOutChannel, const FIXP_DBL *const sfbFormFactorLdData) { #define SCALE_FORM_FAC \ (4) /* (SCALE_FORM_FAC+FORM_FAC_SHIFT) >= ld(FRAME_LENGTH)*/ #define SCALE_NRGS (8) #define SCALE_NLINES (16) #define SCALE_NRGS_SQRT4 (2) /* 0.25 * SCALE_NRGS */ #define SCALE_NLINES_P34 (12) /* 0.75 * SCALE_NLINES */ INT sfbGrp, sfb; FIXP_DBL chaosMeasure; INT frameNLines = 0; FIXP_DBL frameFormFactor = FL2FXCONST_DBL(0.f); FIXP_DBL frameEnergy = FL2FXCONST_DBL(0.f); for (sfbGrp = 0; sfbGrp < psyOutChannel->sfbCnt; sfbGrp += psyOutChannel->sfbPerGroup) { for (sfb = 0; sfb < psyOutChannel->maxSfbPerGroup; sfb++) { if (psyOutChannel->sfbEnergyLdData[sfbGrp + sfb] > psyOutChannel->sfbThresholdLdData[sfbGrp + sfb]) { frameFormFactor += (CalcInvLdData(sfbFormFactorLdData[sfbGrp + sfb]) >> SCALE_FORM_FAC); frameNLines += (psyOutChannel->sfbOffsets[sfbGrp + sfb + 1] - psyOutChannel->sfbOffsets[sfbGrp + sfb]); frameEnergy += (psyOutChannel->sfbEnergy[sfbGrp + sfb] >> SCALE_NRGS); } } } if (frameNLines > 0) { /* frameNActiveLines = frameFormFactor*2^FORM_FAC_SHIFT * ((frameEnergy *2^SCALE_NRGS)/frameNLines)^-0.25 chaosMeasure = frameNActiveLines / frameNLines */ chaosMeasure = CalcInvLdData( (((CalcLdData(frameFormFactor) >> 1) - (CalcLdData(frameEnergy) >> (2 + 1))) - (fMultDiv2(FL2FXCONST_DBL(0.75f), CalcLdData((FIXP_DBL)frameNLines << (DFRACT_BITS - 1 - SCALE_NLINES))) - (((FIXP_DBL)(-((-SCALE_FORM_FAC + SCALE_NRGS_SQRT4 - FORM_FAC_SHIFT + SCALE_NLINES_P34) << (DFRACT_BITS - 1 - LD_DATA_SHIFT)))) >> 1))) << 1); } else { /* assuming total chaos, if no sfb is above thresholds */ chaosMeasure = FL2FXCONST_DBL(1.f); } return chaosMeasure; } /* apply reduction formula for VBR-mode */ static void FDKaacEnc_reduceThresholdsVBR( QC_OUT_CHANNEL *const qcOutChannel[(2)], const PSY_OUT_CHANNEL *const psyOutChannel[(2)], UCHAR ahFlag[(2)][MAX_GROUPED_SFB], const FIXP_DBL thrExp[(2)][MAX_GROUPED_SFB], const INT nChannels, const FIXP_DBL vbrQualFactor, FIXP_DBL *const chaosMeasureOld) { INT ch, sfbGrp, sfb; FIXP_DBL chGroupEnergy[TRANS_FAC][2]; /*energy for each group and channel*/ FIXP_DBL chChaosMeasure[2]; FIXP_DBL frameEnergy = FL2FXCONST_DBL(1e-10f); FIXP_DBL chaosMeasure = FL2FXCONST_DBL(0.f); FIXP_DBL sfbEnLdData, sfbThrLdData, sfbThrExp; FIXP_DBL sfbThrReducedLdData; FIXP_DBL chaosMeasureAvg; INT groupCnt; /* loop counter */ FIXP_DBL redVal[TRANS_FAC]; /* reduction values; in short-block case one redVal for each group */ QC_OUT_CHANNEL *qcOutChan = NULL; const PSY_OUT_CHANNEL *psyOutChan = NULL; #define SCALE_GROUP_ENERGY (8) #define CONST_CHAOS_MEAS_AVG_FAC_0 (FL2FXCONST_DBL(0.25f)) #define CONST_CHAOS_MEAS_AVG_FAC_1 (FL2FXCONST_DBL(1.f - 0.25f)) #define MIN_LDTHRESH (FL2FXCONST_DBL(-0.515625f)) for (ch = 0; ch < nChannels; ch++) { psyOutChan = psyOutChannel[ch]; /* adding up energy for each channel and each group separately */ FIXP_DBL chEnergy = FL2FXCONST_DBL(0.f); groupCnt = 0; for (sfbGrp = 0; sfbGrp < psyOutChan->sfbCnt; sfbGrp += psyOutChan->sfbPerGroup, groupCnt++) { chGroupEnergy[groupCnt][ch] = FL2FXCONST_DBL(0.f); for (sfb = 0; sfb < psyOutChan->maxSfbPerGroup; sfb++) { chGroupEnergy[groupCnt][ch] += (psyOutChan->sfbEnergy[sfbGrp + sfb] >> SCALE_GROUP_ENERGY); } chEnergy += chGroupEnergy[groupCnt][ch]; } frameEnergy += chEnergy; /* chaosMeasure */ if (psyOutChannel[0]->lastWindowSequence == SHORT_WINDOW) { chChaosMeasure[ch] = FL2FXCONST_DBL( 0.5f); /* assume a constant chaos measure of 0.5f for short blocks */ } else { chChaosMeasure[ch] = FDKaacEnc_calcChaosMeasure( psyOutChannel[ch], qcOutChannel[ch]->sfbFormFactorLdData); } chaosMeasure += fMult(chChaosMeasure[ch], chEnergy); } if (frameEnergy > chaosMeasure) { INT scale = CntLeadingZeros(frameEnergy) - 1; FIXP_DBL num = chaosMeasure << scale; FIXP_DBL denum = frameEnergy << scale; chaosMeasure = schur_div(num, denum, 16); } else { chaosMeasure = FL2FXCONST_DBL(1.f); } chaosMeasureAvg = fMult(CONST_CHAOS_MEAS_AVG_FAC_0, chaosMeasure) + fMult(CONST_CHAOS_MEAS_AVG_FAC_1, *chaosMeasureOld); /* averaging chaos measure */ *chaosMeasureOld = chaosMeasure = (fixMin( chaosMeasure, chaosMeasureAvg)); /* use min-value, safe for next frame */ /* characteristic curve chaosMeasure = 0.2f + 0.7f/0.3f * (chaosMeasure - 0.2f); chaosMeasure = fixMin(1.0f, fixMax(0.1f, chaosMeasure)); constants scaled by 4.f */ chaosMeasure = ((FL2FXCONST_DBL(0.2f) >> 2) + fMult(FL2FXCONST_DBL(0.7f / (4.f * 0.3f)), (chaosMeasure - FL2FXCONST_DBL(0.2f)))); chaosMeasure = (fixMin((FIXP_DBL)(FL2FXCONST_DBL(1.0f) >> 2), fixMax((FIXP_DBL)(FL2FXCONST_DBL(0.1f) >> 2), chaosMeasure))) << 2; /* calculation of reduction value */ if (psyOutChannel[0]->lastWindowSequence == SHORT_WINDOW) { /* short-blocks */ FDK_ASSERT(TRANS_FAC == 8); #define WIN_TYPE_SCALE (3) groupCnt = 0; for (sfbGrp = 0; sfbGrp < psyOutChannel[0]->sfbCnt; sfbGrp += psyOutChannel[0]->sfbPerGroup, groupCnt++) { FIXP_DBL groupEnergy = FL2FXCONST_DBL(0.f); for (ch = 0; ch < nChannels; ch++) { groupEnergy += chGroupEnergy[groupCnt] [ch]; /* adding up the channels groupEnergy */ } FDK_ASSERT(psyOutChannel[0]->groupLen[groupCnt] <= INV_INT_TAB_SIZE); groupEnergy = fMult( groupEnergy, invInt[psyOutChannel[0]->groupLen[groupCnt]]); /* correction of group energy */ groupEnergy = fixMin(groupEnergy, frameEnergy >> WIN_TYPE_SCALE); /* do not allow an higher redVal as calculated framewise */ groupEnergy >>= 2; /* 2*WIN_TYPE_SCALE = 6 => 6+2 = 8 ==> 8/4 = int number */ redVal[groupCnt] = fMult(fMult(vbrQualFactor, chaosMeasure), CalcInvLdData(CalcLdData(groupEnergy) >> 2)) << (int)((2 + (2 * WIN_TYPE_SCALE) + SCALE_GROUP_ENERGY) >> 2); } } else { /* long-block */ redVal[0] = fMult(fMult(vbrQualFactor, chaosMeasure), CalcInvLdData(CalcLdData(frameEnergy) >> 2)) << (int)(SCALE_GROUP_ENERGY >> 2); } for (ch = 0; ch < nChannels; ch++) { qcOutChan = qcOutChannel[ch]; psyOutChan = psyOutChannel[ch]; for (sfbGrp = 0; sfbGrp < psyOutChan->sfbCnt; sfbGrp += psyOutChan->sfbPerGroup) { for (sfb = 0; sfb < psyOutChan->maxSfbPerGroup; sfb++) { sfbEnLdData = (qcOutChan->sfbWeightedEnergyLdData[sfbGrp + sfb]); sfbThrLdData = (qcOutChan->sfbThresholdLdData[sfbGrp + sfb]); sfbThrExp = thrExp[ch][sfbGrp + sfb]; if ((sfbThrLdData >= MIN_LDTHRESH) && (sfbEnLdData > sfbThrLdData) && (ahFlag[ch][sfbGrp + sfb] != AH_ACTIVE)) { /* Short-Window */ if (psyOutChannel[ch]->lastWindowSequence == SHORT_WINDOW) { const int groupNumber = (int)sfb / psyOutChan->sfbPerGroup; FDK_ASSERT(INV_SQRT4_TAB_SIZE > psyOutChan->groupLen[groupNumber]); sfbThrExp = fMult(sfbThrExp, fMult(FL2FXCONST_DBL(2.82f / 4.f), invSqrt4[psyOutChan->groupLen[groupNumber]])) << 2; if (sfbThrExp <= (limitThrReducedLdData - redVal[groupNumber])) { sfbThrReducedLdData = FL2FXCONST_DBL(-1.0f); } else { if ((FIXP_DBL)redVal[groupNumber] >= FL2FXCONST_DBL(1.0f) - sfbThrExp) sfbThrReducedLdData = FL2FXCONST_DBL(0.0f); else { /* threshold reduction formula */ sfbThrReducedLdData = CalcLdData(sfbThrExp + redVal[groupNumber]); sfbThrReducedLdData <<= 2; } } sfbThrReducedLdData += (CalcLdInt(psyOutChan->groupLen[groupNumber]) - ((FIXP_DBL)6 << (DFRACT_BITS - 1 - LD_DATA_SHIFT))); } /* Long-Window */ else { if ((FIXP_DBL)redVal[0] >= FL2FXCONST_DBL(1.0f) - sfbThrExp) { sfbThrReducedLdData = FL2FXCONST_DBL(0.0f); } else { /* threshold reduction formula */ sfbThrReducedLdData = CalcLdData(sfbThrExp + redVal[0]); sfbThrReducedLdData <<= 2; } } /* avoid holes */ if (((sfbThrReducedLdData - sfbEnLdData) > qcOutChan->sfbMinSnrLdData[sfbGrp + sfb]) && (ahFlag[ch][sfbGrp + sfb] != NO_AH)) { if (qcOutChan->sfbMinSnrLdData[sfbGrp + sfb] > (FL2FXCONST_DBL(-1.0f) - sfbEnLdData)) { sfbThrReducedLdData = fixMax( (qcOutChan->sfbMinSnrLdData[sfbGrp + sfb] + sfbEnLdData), sfbThrLdData); } else sfbThrReducedLdData = sfbThrLdData; ahFlag[ch][sfbGrp + sfb] = AH_ACTIVE; } if (sfbThrReducedLdData < FL2FXCONST_DBL(-0.5f)) sfbThrReducedLdData = FL2FXCONST_DBL(-1.f); sfbThrReducedLdData = fixMax(MIN_LDTHRESH, sfbThrReducedLdData); qcOutChan->sfbThresholdLdData[sfbGrp + sfb] = sfbThrReducedLdData; } } } } } /***************************************************************************** functionname: FDKaacEnc_correctThresh description: if pe difference deltaPe between desired pe and real pe is small enough, the difference can be distributed among the scale factor bands. New thresholds can be derived from this pe-difference *****************************************************************************/ static void FDKaacEnc_correctThresh( const CHANNEL_MAPPING *const cm, QC_OUT_ELEMENT *const qcElement[((8))], const PSY_OUT_ELEMENT *const psyOutElement[((8))], UCHAR ahFlag[((8))][(2)][MAX_GROUPED_SFB], const FIXP_DBL thrExp[((8))][(2)][MAX_GROUPED_SFB], const FIXP_DBL redVal_m, const SCHAR redVal_e, const INT deltaPe, const INT processElements, const INT elementOffset) { INT ch, sfb, sfbGrp; QC_OUT_CHANNEL *qcOutChan; PSY_OUT_CHANNEL *psyOutChan; PE_CHANNEL_DATA *peChanData; FIXP_DBL thrFactorLdData; FIXP_DBL sfbEnLdData, sfbThrLdData, sfbThrReducedLdData; FIXP_DBL *sfbPeFactorsLdData[((8))][(2)]; FIXP_DBL(*sfbNActiveLinesLdData)[(2)][MAX_GROUPED_SFB]; INT normFactorInt; FIXP_DBL normFactorLdData; INT nElements = elementOffset + processElements; INT elementId; /* scratch is empty; use temporal memory from quantSpec in QC_OUT_CHANNEL */ for (elementId = elementOffset; elementId < nElements; elementId++) { for (ch = 0; ch < cm->elInfo[elementId].nChannelsInEl; ch++) { /* The reinterpret_cast is used to suppress a compiler warning. We know * that qcElement[elementId]->qcOutChannel[ch]->quantSpec is sufficiently * aligned, so the cast is safe */ sfbPeFactorsLdData[elementId][ch] = reinterpret_cast(reinterpret_cast( qcElement[elementId]->qcOutChannel[ch]->quantSpec)); } } /* The reinterpret_cast is used to suppress a compiler warning. We know that * qcElement[0]->dynMem_SfbNActiveLinesLdData is sufficiently aligned, so the * cast is safe */ sfbNActiveLinesLdData = reinterpret_cast( reinterpret_cast(qcElement[0]->dynMem_SfbNActiveLinesLdData)); /* for each sfb calc relative factors for pe changes */ normFactorInt = 0; for (elementId = elementOffset; elementId < nElements; elementId++) { if (cm->elInfo[elementId].elType != ID_DSE) { for (ch = 0; ch < cm->elInfo[elementId].nChannelsInEl; ch++) { psyOutChan = psyOutElement[elementId]->psyOutChannel[ch]; peChanData = &qcElement[elementId]->peData.peChannelData[ch]; for (sfbGrp = 0; sfbGrp < psyOutChan->sfbCnt; sfbGrp += psyOutChan->sfbPerGroup) { for (sfb = 0; sfb < psyOutChan->maxSfbPerGroup; sfb++) { if (peChanData->sfbNActiveLines[sfbGrp + sfb] == 0) { sfbNActiveLinesLdData[elementId][ch][sfbGrp + sfb] = FL2FXCONST_DBL(-1.0f); } else { /* Both CalcLdInt and CalcLdData can be used! * No offset has to be subtracted, because sfbNActiveLinesLdData * is shorted while thrFactor calculation */ sfbNActiveLinesLdData[elementId][ch][sfbGrp + sfb] = CalcLdInt(peChanData->sfbNActiveLines[sfbGrp + sfb]); } if (((ahFlag[elementId][ch][sfbGrp + sfb] < AH_ACTIVE) || (deltaPe > 0)) && peChanData->sfbNActiveLines[sfbGrp + sfb] != 0) { if (thrExp[elementId][ch][sfbGrp + sfb] > -redVal_m) { /* sfbPeFactors[ch][sfbGrp+sfb] = peChanData->sfbNActiveLines[sfbGrp+sfb] / (thrExp[elementId][ch][sfbGrp+sfb] + redVal[elementId]); */ int minScale = fixMin( CountLeadingBits(thrExp[elementId][ch][sfbGrp + sfb]), CountLeadingBits(redVal_m) - redVal_e) - 1; /* sumld = ld64( sfbThrExp + redVal ) */ FIXP_DBL sumLd = CalcLdData(scaleValue(thrExp[elementId][ch][sfbGrp + sfb], minScale) + scaleValue(redVal_m, redVal_e + minScale)) - (FIXP_DBL)(minScale << (DFRACT_BITS - 1 - LD_DATA_SHIFT)); if (sumLd < FL2FXCONST_DBL(0.f)) { sfbPeFactorsLdData[elementId][ch][sfbGrp + sfb] = sfbNActiveLinesLdData[elementId][ch][sfbGrp + sfb] - sumLd; } else { if (sfbNActiveLinesLdData[elementId][ch][sfbGrp + sfb] > (FL2FXCONST_DBL(-1.f) + sumLd)) { sfbPeFactorsLdData[elementId][ch][sfbGrp + sfb] = sfbNActiveLinesLdData[elementId][ch][sfbGrp + sfb] - sumLd; } else { sfbPeFactorsLdData[elementId][ch][sfbGrp + sfb] = sfbNActiveLinesLdData[elementId][ch][sfbGrp + sfb]; } } normFactorInt += (INT)CalcInvLdData( sfbPeFactorsLdData[elementId][ch][sfbGrp + sfb]); } else sfbPeFactorsLdData[elementId][ch][sfbGrp + sfb] = FL2FXCONST_DBL(1.0f); } else sfbPeFactorsLdData[elementId][ch][sfbGrp + sfb] = FL2FXCONST_DBL(-1.0f); } } } } } /* normFactorLdData = ld64(deltaPe/normFactorInt) */ normFactorLdData = CalcLdData((FIXP_DBL)((deltaPe < 0) ? (-deltaPe) : (deltaPe))) - CalcLdData((FIXP_DBL)normFactorInt); /* distribute the pe difference to the scalefactors and calculate the according thresholds */ for (elementId = elementOffset; elementId < nElements; elementId++) { if (cm->elInfo[elementId].elType != ID_DSE) { for (ch = 0; ch < cm->elInfo[elementId].nChannelsInEl; ch++) { qcOutChan = qcElement[elementId]->qcOutChannel[ch]; psyOutChan = psyOutElement[elementId]->psyOutChannel[ch]; peChanData = &qcElement[elementId]->peData.peChannelData[ch]; for (sfbGrp = 0; sfbGrp < psyOutChan->sfbCnt; sfbGrp += psyOutChan->sfbPerGroup) { for (sfb = 0; sfb < psyOutChan->maxSfbPerGroup; sfb++) { if (peChanData->sfbNActiveLines[sfbGrp + sfb] > 0) { /* pe difference for this sfb */ if ((sfbPeFactorsLdData[elementId][ch][sfbGrp + sfb] == FL2FXCONST_DBL(-1.0f)) || (deltaPe == 0)) { thrFactorLdData = FL2FXCONST_DBL(0.f); } else { /* new threshold */ FIXP_DBL tmp = CalcInvLdData( sfbPeFactorsLdData[elementId][ch][sfbGrp + sfb] + normFactorLdData - sfbNActiveLinesLdData[elementId][ch][sfbGrp + sfb] - FL2FXCONST_DBL((float)LD_DATA_SHIFT / LD_DATA_SCALING)); /* limit thrFactor to 60dB */ tmp = (deltaPe < 0) ? tmp : (-tmp); thrFactorLdData = fMin(tmp, FL2FXCONST_DBL(20.f / LD_DATA_SCALING)); } /* new threshold */ sfbThrLdData = qcOutChan->sfbThresholdLdData[sfbGrp + sfb]; sfbEnLdData = qcOutChan->sfbWeightedEnergyLdData[sfbGrp + sfb]; if (thrFactorLdData < FL2FXCONST_DBL(0.f)) { if (sfbThrLdData > (FL2FXCONST_DBL(-1.f) - thrFactorLdData)) { sfbThrReducedLdData = sfbThrLdData + thrFactorLdData; } else { sfbThrReducedLdData = FL2FXCONST_DBL(-1.f); } } else { sfbThrReducedLdData = sfbThrLdData + thrFactorLdData; } /* avoid hole */ if ((sfbThrReducedLdData - sfbEnLdData > qcOutChan->sfbMinSnrLdData[sfbGrp + sfb]) && (ahFlag[elementId][ch][sfbGrp + sfb] == AH_INACTIVE)) { /* sfbThrReduced = max(psyOutChan[ch]->sfbMinSnr[i] * sfbEn, * sfbThr); */ if (sfbEnLdData > (sfbThrLdData - qcOutChan->sfbMinSnrLdData[sfbGrp + sfb])) { sfbThrReducedLdData = qcOutChan->sfbMinSnrLdData[sfbGrp + sfb] + sfbEnLdData; } else { sfbThrReducedLdData = sfbThrLdData; } ahFlag[elementId][ch][sfbGrp + sfb] = AH_ACTIVE; } qcOutChan->sfbThresholdLdData[sfbGrp + sfb] = sfbThrReducedLdData; } } } } } } } /***************************************************************************** functionname: FDKaacEnc_reduceMinSnr description: if the desired pe can not be reached, reduce pe by reducing minSnr *****************************************************************************/ static void FDKaacEnc_reduceMinSnr( const CHANNEL_MAPPING *const cm, QC_OUT_ELEMENT *const qcElement[((8))], const PSY_OUT_ELEMENT *const psyOutElement[((8))], const UCHAR ahFlag[((8))][(2)][MAX_GROUPED_SFB], const INT desiredPe, INT *const redPeGlobal, const INT processElements, const INT elementOffset) { INT ch, elementId, globalMaxSfb = 0; const INT nElements = elementOffset + processElements; INT newGlobalPe = *redPeGlobal; if (newGlobalPe <= desiredPe) { goto bail; } /* global maximum of maxSfbPerGroup */ for (elementId = elementOffset; elementId < nElements; elementId++) { if (cm->elInfo[elementId].elType != ID_DSE) { for (ch = 0; ch < cm->elInfo[elementId].nChannelsInEl; ch++) { globalMaxSfb = fMax(globalMaxSfb, psyOutElement[elementId]->psyOutChannel[ch]->maxSfbPerGroup); } } } /* as long as globalPE is above desirePE reduce SNR to 1.0 dB, starting at * highest SFB */ while ((newGlobalPe > desiredPe) && (--globalMaxSfb >= 0)) { for (elementId = elementOffset; elementId < nElements; elementId++) { if (cm->elInfo[elementId].elType != ID_DSE) { PE_DATA *peData = &qcElement[elementId]->peData; for (ch = 0; ch < cm->elInfo[elementId].nChannelsInEl; ch++) { QC_OUT_CHANNEL *qcOutChan = qcElement[elementId]->qcOutChannel[ch]; PSY_OUT_CHANNEL *psyOutChan = psyOutElement[elementId]->psyOutChannel[ch]; /* try to reduce SNR of channel's uppermost SFB(s) */ if (globalMaxSfb < psyOutChan->maxSfbPerGroup) { INT sfb, deltaPe = 0; for (sfb = globalMaxSfb; sfb < psyOutChan->sfbCnt; sfb += psyOutChan->sfbPerGroup) { if (ahFlag[elementId][ch][sfb] != NO_AH && qcOutChan->sfbMinSnrLdData[sfb] < SnrLdFac && (qcOutChan->sfbWeightedEnergyLdData[sfb] > qcOutChan->sfbThresholdLdData[sfb] - SnrLdFac)) { /* increase threshold to new minSnr of 1dB */ qcOutChan->sfbMinSnrLdData[sfb] = SnrLdFac; qcOutChan->sfbThresholdLdData[sfb] = qcOutChan->sfbWeightedEnergyLdData[sfb] + SnrLdFac; /* calc new pe */ /* C2 + C3*ld(1/0.8) = 1.5 */ deltaPe -= peData->peChannelData[ch].sfbPe[sfb]; /* sfbPe = 1.5 * sfbNLines */ peData->peChannelData[ch].sfbPe[sfb] = (3 * peData->peChannelData[ch].sfbNLines[sfb]) << (PE_CONSTPART_SHIFT - 1); deltaPe += peData->peChannelData[ch].sfbPe[sfb]; } } /* sfb loop */ deltaPe >>= PE_CONSTPART_SHIFT; peData->pe += deltaPe; peData->peChannelData[ch].pe += deltaPe; newGlobalPe += deltaPe; } /* if globalMaxSfb < maxSfbPerGroup */ /* stop if enough has been saved */ if (newGlobalPe <= desiredPe) { goto bail; } } /* ch loop */ } /* != ID_DSE */ } /* elementId loop */ } /* while ( newGlobalPe > desiredPe) && (--globalMaxSfb >= 0) ) */ bail: /* update global PE */ *redPeGlobal = newGlobalPe; } /***************************************************************************** functionname: FDKaacEnc_allowMoreHoles description: if the desired pe can not be reached, some more scalefactor bands have to be quantized to zero *****************************************************************************/ static void FDKaacEnc_allowMoreHoles( const CHANNEL_MAPPING *const cm, QC_OUT_ELEMENT *const qcElement[((8))], const PSY_OUT_ELEMENT *const psyOutElement[((8))], const ATS_ELEMENT *const AdjThrStateElement[((8))], UCHAR ahFlag[((8))][(2)][MAX_GROUPED_SFB], const INT desiredPe, const INT currentPe, const int processElements, const int elementOffset) { INT elementId; INT nElements = elementOffset + processElements; INT actPe = currentPe; if (actPe <= desiredPe) { return; /* nothing to do */ } for (elementId = elementOffset; elementId < nElements; elementId++) { if (cm->elInfo[elementId].elType != ID_DSE) { INT ch, sfb, sfbGrp; PE_DATA *peData = &qcElement[elementId]->peData; const INT nChannels = cm->elInfo[elementId].nChannelsInEl; QC_OUT_CHANNEL *qcOutChannel[(2)] = {NULL}; PSY_OUT_CHANNEL *psyOutChannel[(2)] = {NULL}; for (ch = 0; ch < nChannels; ch++) { /* init pointers */ qcOutChannel[ch] = qcElement[elementId]->qcOutChannel[ch]; psyOutChannel[ch] = psyOutElement[elementId]->psyOutChannel[ch]; for (sfbGrp = 0; sfbGrp < psyOutChannel[ch]->sfbCnt; sfbGrp += psyOutChannel[ch]->sfbPerGroup) { for (sfb = psyOutChannel[ch]->maxSfbPerGroup; sfb < psyOutChannel[ch]->sfbPerGroup; sfb++) { peData->peChannelData[ch].sfbPe[sfbGrp + sfb] = 0; } } } /* for MS allow hole in the channel with less energy */ if (nChannels == 2 && psyOutChannel[0]->lastWindowSequence == psyOutChannel[1]->lastWindowSequence) { for (sfb = psyOutChannel[0]->maxSfbPerGroup - 1; sfb >= 0; sfb--) { for (sfbGrp = 0; sfbGrp < psyOutChannel[0]->sfbCnt; sfbGrp += psyOutChannel[0]->sfbPerGroup) { if (psyOutElement[elementId]->toolsInfo.msMask[sfbGrp + sfb]) { FIXP_DBL EnergyLd_L = qcOutChannel[0]->sfbWeightedEnergyLdData[sfbGrp + sfb]; FIXP_DBL EnergyLd_R = qcOutChannel[1]->sfbWeightedEnergyLdData[sfbGrp + sfb]; /* allow hole in side channel ? */ if ((ahFlag[elementId][1][sfbGrp + sfb] != NO_AH) && (((FL2FXCONST_DBL(-0.02065512648f) >> 1) + (qcOutChannel[0]->sfbMinSnrLdData[sfbGrp + sfb] >> 1)) > ((EnergyLd_R >> 1) - (EnergyLd_L >> 1)))) { ahFlag[elementId][1][sfbGrp + sfb] = NO_AH; qcOutChannel[1]->sfbThresholdLdData[sfbGrp + sfb] = FL2FXCONST_DBL(0.015625f) + EnergyLd_R; actPe -= peData->peChannelData[1].sfbPe[sfbGrp + sfb] >> PE_CONSTPART_SHIFT; } /* allow hole in mid channel ? */ else if ((ahFlag[elementId][0][sfbGrp + sfb] != NO_AH) && (((FL2FXCONST_DBL(-0.02065512648f) >> 1) + (qcOutChannel[1]->sfbMinSnrLdData[sfbGrp + sfb] >> 1)) > ((EnergyLd_L >> 1) - (EnergyLd_R >> 1)))) { ahFlag[elementId][0][sfbGrp + sfb] = NO_AH; qcOutChannel[0]->sfbThresholdLdData[sfbGrp + sfb] = FL2FXCONST_DBL(0.015625f) + EnergyLd_L; actPe -= peData->peChannelData[0].sfbPe[sfbGrp + sfb] >> PE_CONSTPART_SHIFT; } /* if (ahFlag) */ } /* if MS */ } /* sfbGrp */ if (actPe <= desiredPe) { return; /* stop if enough has been saved */ } } /* sfb */ } /* MS possible ? */ } /* EOF DSE-suppression */ } /* EOF for all elements... */ if (actPe > desiredPe) { /* more holes necessary? subsequently erase bands starting with low energies */ INT ch, sfb, sfbGrp; INT minSfb, maxSfb; INT enIdx, ahCnt, done; INT startSfb[(8)]; INT sfbCnt[(8)]; INT sfbPerGroup[(8)]; INT maxSfbPerGroup[(8)]; FIXP_DBL avgEn; FIXP_DBL minEnLD64; FIXP_DBL avgEnLD64; FIXP_DBL enLD64[NUM_NRG_LEVS]; INT avgEn_e; /* get the scaling factor over all audio elements and channels */ maxSfb = 0; for (elementId = elementOffset; elementId < nElements; elementId++) { if (cm->elInfo[elementId].elType != ID_DSE) { for (ch = 0; ch < cm->elInfo[elementId].nChannelsInEl; ch++) { for (sfbGrp = 0; sfbGrp < psyOutElement[elementId]->psyOutChannel[ch]->sfbCnt; sfbGrp += psyOutElement[elementId]->psyOutChannel[ch]->sfbPerGroup) { maxSfb += psyOutElement[elementId]->psyOutChannel[ch]->maxSfbPerGroup; } } } } avgEn_e = (DFRACT_BITS - fixnormz_D((LONG)fMax(0, maxSfb - 1))); /* ilog2() */ ahCnt = 0; maxSfb = 0; minSfb = MAX_SFB; avgEn = FL2FXCONST_DBL(0.0f); minEnLD64 = FL2FXCONST_DBL(0.0f); for (elementId = elementOffset; elementId < nElements; elementId++) { if (cm->elInfo[elementId].elType != ID_DSE) { for (ch = 0; ch < cm->elInfo[elementId].nChannelsInEl; ch++) { const INT chIdx = cm->elInfo[elementId].ChannelIndex[ch]; QC_OUT_CHANNEL *qcOutChannel = qcElement[elementId]->qcOutChannel[ch]; PSY_OUT_CHANNEL *psyOutChannel = psyOutElement[elementId]->psyOutChannel[ch]; maxSfbPerGroup[chIdx] = psyOutChannel->maxSfbPerGroup; sfbCnt[chIdx] = psyOutChannel->sfbCnt; sfbPerGroup[chIdx] = psyOutChannel->sfbPerGroup; maxSfb = fMax(maxSfb, psyOutChannel->maxSfbPerGroup); if (psyOutChannel->lastWindowSequence != SHORT_WINDOW) { startSfb[chIdx] = AdjThrStateElement[elementId]->ahParam.startSfbL; } else { startSfb[chIdx] = AdjThrStateElement[elementId]->ahParam.startSfbS; } minSfb = fMin(minSfb, startSfb[chIdx]); sfbGrp = 0; sfb = startSfb[chIdx]; do { for (; sfb < psyOutChannel->maxSfbPerGroup; sfb++) { if ((ahFlag[elementId][ch][sfbGrp + sfb] != NO_AH) && (qcOutChannel->sfbWeightedEnergyLdData[sfbGrp + sfb] > qcOutChannel->sfbThresholdLdData[sfbGrp + sfb])) { minEnLD64 = fixMin(minEnLD64, qcOutChannel->sfbEnergyLdData[sfbGrp + sfb]); avgEn += qcOutChannel->sfbEnergy[sfbGrp + sfb] >> avgEn_e; ahCnt++; } } sfbGrp += psyOutChannel->sfbPerGroup; sfb = startSfb[chIdx]; } while (sfbGrp < psyOutChannel->sfbCnt); } } /* (cm->elInfo[elementId].elType != ID_DSE) */ } /* (elementId = elementOffset;elementIdelInfo[elementId].elType != ID_DSE) { PE_DATA *peData = &qcElement[elementId]->peData; for (ch = 0; ch < cm->elInfo[elementId].nChannelsInEl; ch++) { const INT chIdx = cm->elInfo[elementId].ChannelIndex[ch]; QC_OUT_CHANNEL *qcOutChannel = qcElement[elementId]->qcOutChannel[ch]; if (sfb >= startSfb[chIdx] && sfb < maxSfbPerGroup[chIdx]) { for (sfbGrp = 0; sfbGrp < sfbCnt[chIdx]; sfbGrp += sfbPerGroup[chIdx]) { /* sfb energy below border ? */ if (ahFlag[elementId][ch][sfbGrp + sfb] != NO_AH && qcOutChannel->sfbEnergyLdData[sfbGrp + sfb] < enLD64[enIdx]) { /* allow hole */ ahFlag[elementId][ch][sfbGrp + sfb] = NO_AH; qcOutChannel->sfbThresholdLdData[sfbGrp + sfb] = FL2FXCONST_DBL(0.015625f) + qcOutChannel->sfbWeightedEnergyLdData[sfbGrp + sfb]; actPe -= peData->peChannelData[ch].sfbPe[sfbGrp + sfb] >> PE_CONSTPART_SHIFT; } if (actPe <= desiredPe) { return; /* stop if enough has been saved */ } } /* sfbGrp */ } /* sfb */ } /* nChannelsInEl */ } /* ID_DSE */ } /* elementID */ sfb--; if (sfb < minSfb) { /* restart with next energy border */ sfb = maxSfb; enIdx++; if (enIdx >= NUM_NRG_LEVS) { done = 1; } } } /* done */ } /* (actPe <= desiredPe) */ } /* reset avoid hole flags from AH_ACTIVE to AH_INACTIVE */ static void FDKaacEnc_resetAHFlags( UCHAR ahFlag[(2)][MAX_GROUPED_SFB], const INT nChannels, const PSY_OUT_CHANNEL *const psyOutChannel[(2)]) { int ch, sfb, sfbGrp; for (ch = 0; ch < nChannels; ch++) { for (sfbGrp = 0; sfbGrp < psyOutChannel[ch]->sfbCnt; sfbGrp += psyOutChannel[ch]->sfbPerGroup) { for (sfb = 0; sfb < psyOutChannel[ch]->maxSfbPerGroup; sfb++) { if (ahFlag[ch][sfbGrp + sfb] == AH_ACTIVE) { ahFlag[ch][sfbGrp + sfb] = AH_INACTIVE; } } } } } static FIXP_DBL CalcRedValPower(FIXP_DBL num, FIXP_DBL denum, INT *scaling) { FIXP_DBL value = FL2FXCONST_DBL(0.f); if (num >= FL2FXCONST_DBL(0.f)) { value = fDivNorm(num, denum, scaling); } else { value = -fDivNorm(-num, denum, scaling); } value = f2Pow(value, *scaling, scaling); return value; } /***************************************************************************** functionname: FDKaacEnc_adaptThresholdsToPe description: two guesses for the reduction value and one final correction of the thresholds *****************************************************************************/ static void FDKaacEnc_adaptThresholdsToPe( const CHANNEL_MAPPING *const cm, ATS_ELEMENT *const AdjThrStateElement[((8))], QC_OUT_ELEMENT *const qcElement[((8))], const PSY_OUT_ELEMENT *const psyOutElement[((8))], const INT desiredPe, const INT maxIter2ndGuess, const INT processElements, const INT elementOffset) { FIXP_DBL reductionValue_m; SCHAR reductionValue_e; UCHAR(*pAhFlag)[(2)][MAX_GROUPED_SFB]; FIXP_DBL(*pThrExp)[(2)][MAX_GROUPED_SFB]; int iter; INT constPartGlobal, noRedPeGlobal, nActiveLinesGlobal, redPeGlobal; constPartGlobal = noRedPeGlobal = nActiveLinesGlobal = redPeGlobal = 0; int elementId; int nElements = elementOffset + processElements; if (nElements > cm->nElements) { nElements = cm->nElements; } /* The reinterpret_cast is used to suppress a compiler warning. We know that * qcElement[0]->dynMem_Ah_Flag is sufficiently aligned, so the cast is safe */ pAhFlag = reinterpret_cast( reinterpret_cast(qcElement[0]->dynMem_Ah_Flag)); /* The reinterpret_cast is used to suppress a compiler warning. We know that * qcElement[0]->dynMem_Thr_Exp is sufficiently aligned, so the cast is safe */ pThrExp = reinterpret_cast( reinterpret_cast(qcElement[0]->dynMem_Thr_Exp)); /* ------------------------------------------------------- */ /* Part I: Initialize data structures and variables... */ /* ------------------------------------------------------- */ for (elementId = elementOffset; elementId < nElements; elementId++) { if (cm->elInfo[elementId].elType != ID_DSE) { INT nChannels = cm->elInfo[elementId].nChannelsInEl; PE_DATA *peData = &qcElement[elementId]->peData; /* thresholds to the power of redExp */ FDKaacEnc_calcThreshExp( pThrExp[elementId], qcElement[elementId]->qcOutChannel, psyOutElement[elementId]->psyOutChannel, nChannels); /* lower the minSnr requirements for low energies compared to the average energy in this frame */ FDKaacEnc_adaptMinSnr(qcElement[elementId]->qcOutChannel, psyOutElement[elementId]->psyOutChannel, &AdjThrStateElement[elementId]->minSnrAdaptParam, nChannels); /* init ahFlag (0: no ah necessary, 1: ah possible, 2: ah active */ FDKaacEnc_initAvoidHoleFlag( qcElement[elementId]->qcOutChannel, psyOutElement[elementId]->psyOutChannel, pAhFlag[elementId], &psyOutElement[elementId]->toolsInfo, nChannels, &AdjThrStateElement[elementId]->ahParam); /* sum up */ constPartGlobal += peData->constPart; noRedPeGlobal += peData->pe; nActiveLinesGlobal += fixMax((INT)peData->nActiveLines, 1); } /* EOF DSE-suppression */ } /* EOF for all elements... */ /* First guess of reduction value: avgThrExp = (float)pow(2.0f, (constPartGlobal - noRedPeGlobal)/(4.0f * nActiveLinesGlobal)); redVal = (float)pow(2.0f, (constPartGlobal - desiredPe)/(4.0f * nActiveLinesGlobal)) - avgThrExp; redVal = max(0.f, redVal); */ int redVal_e, avgThrExp_e, result_e; FIXP_DBL redVal_m, avgThrExp_m; redVal_m = CalcRedValPower(constPartGlobal - desiredPe, 4 * nActiveLinesGlobal, &redVal_e); avgThrExp_m = CalcRedValPower(constPartGlobal - noRedPeGlobal, 4 * nActiveLinesGlobal, &avgThrExp_e); result_e = fMax(redVal_e, avgThrExp_e) + 1; reductionValue_m = fMax(FL2FXCONST_DBL(0.f), scaleValue(redVal_m, redVal_e - result_e) - scaleValue(avgThrExp_m, avgThrExp_e - result_e)); reductionValue_e = result_e; /* ----------------------------------------------------------------------- */ /* Part II: Calculate bit consumption of initial bit constraints setup */ /* ----------------------------------------------------------------------- */ for (elementId = elementOffset; elementId < nElements; elementId++) { if (cm->elInfo[elementId].elType != ID_DSE) { INT nChannels = cm->elInfo[elementId].nChannelsInEl; PE_DATA *peData = &qcElement[elementId]->peData; /* reduce thresholds */ FDKaacEnc_reduceThresholdsCBR( qcElement[elementId]->qcOutChannel, psyOutElement[elementId]->psyOutChannel, pAhFlag[elementId], pThrExp[elementId], nChannels, reductionValue_m, reductionValue_e); /* pe after first guess */ FDKaacEnc_calcPe(psyOutElement[elementId]->psyOutChannel, qcElement[elementId]->qcOutChannel, peData, nChannels); redPeGlobal += peData->pe; } /* EOF DSE-suppression */ } /* EOF for all elements... */ /* -------------------------------------------------- */ /* Part III: Iterate until bit constraints are met */ /* -------------------------------------------------- */ iter = 0; while ((fixp_abs(redPeGlobal - desiredPe) > fMultI(FL2FXCONST_DBL(0.05f), desiredPe)) && (iter < maxIter2ndGuess)) { INT desiredPeNoAHGlobal; INT redPeNoAHGlobal = 0; INT constPartNoAHGlobal = 0; INT nActiveLinesNoAHGlobal = 0; for (elementId = elementOffset; elementId < nElements; elementId++) { if (cm->elInfo[elementId].elType != ID_DSE) { INT redPeNoAH, constPartNoAH, nActiveLinesNoAH; INT nChannels = cm->elInfo[elementId].nChannelsInEl; PE_DATA *peData = &qcElement[elementId]->peData; /* pe for bands where avoid hole is inactive */ FDKaacEnc_FDKaacEnc_calcPeNoAH( &redPeNoAH, &constPartNoAH, &nActiveLinesNoAH, peData, pAhFlag[elementId], psyOutElement[elementId]->psyOutChannel, nChannels); redPeNoAHGlobal += redPeNoAH; constPartNoAHGlobal += constPartNoAH; nActiveLinesNoAHGlobal += nActiveLinesNoAH; } /* EOF DSE-suppression */ } /* EOF for all elements... */ /* Calculate new redVal ... */ if (desiredPe < redPeGlobal) { /* new desired pe without bands where avoid hole is active */ desiredPeNoAHGlobal = desiredPe - (redPeGlobal - redPeNoAHGlobal); /* limit desiredPeNoAH to positive values, as the PE can not become * negative */ desiredPeNoAHGlobal = fMax(0, desiredPeNoAHGlobal); /* second guess (only if there are bands left where avoid hole is * inactive)*/ if (nActiveLinesNoAHGlobal > 0) { /* avgThrExp = (float)pow(2.0f, (constPartNoAHGlobal - redPeNoAHGlobal) / (4.0f * nActiveLinesNoAHGlobal)); redVal += (float)pow(2.0f, (constPartNoAHGlobal - desiredPeNoAHGlobal) / (4.0f * nActiveLinesNoAHGlobal)) - avgThrExp; redVal = max(0.0f, redVal); */ redVal_m = CalcRedValPower(constPartNoAHGlobal - desiredPeNoAHGlobal, 4 * nActiveLinesNoAHGlobal, &redVal_e); avgThrExp_m = CalcRedValPower(constPartNoAHGlobal - redPeNoAHGlobal, 4 * nActiveLinesNoAHGlobal, &avgThrExp_e); result_e = fMax(reductionValue_e, fMax(redVal_e, avgThrExp_e) + 1) + 1; reductionValue_m = fMax(FL2FXCONST_DBL(0.f), scaleValue(reductionValue_m, reductionValue_e - result_e) + scaleValue(redVal_m, redVal_e - result_e) - scaleValue(avgThrExp_m, avgThrExp_e - result_e)); reductionValue_e = result_e; } /* nActiveLinesNoAHGlobal > 0 */ } else { /* redVal *= redPeGlobal/desiredPe; */ int sc0, sc1; reductionValue_m = fMultNorm( reductionValue_m, fDivNorm((FIXP_DBL)redPeGlobal, (FIXP_DBL)desiredPe, &sc0), &sc1); reductionValue_e += sc0 + sc1; for (elementId = elementOffset; elementId < nElements; elementId++) { if (cm->elInfo[elementId].elType != ID_DSE) { FDKaacEnc_resetAHFlags(pAhFlag[elementId], cm->elInfo[elementId].nChannelsInEl, psyOutElement[elementId]->psyOutChannel); } /* EOF DSE-suppression */ } /* EOF for all elements... */ } redPeGlobal = 0; /* Calculate new redVal's PE... */ for (elementId = elementOffset; elementId < nElements; elementId++) { if (cm->elInfo[elementId].elType != ID_DSE) { INT nChannels = cm->elInfo[elementId].nChannelsInEl; PE_DATA *peData = &qcElement[elementId]->peData; /* reduce thresholds */ FDKaacEnc_reduceThresholdsCBR( qcElement[elementId]->qcOutChannel, psyOutElement[elementId]->psyOutChannel, pAhFlag[elementId], pThrExp[elementId], nChannels, reductionValue_m, reductionValue_e); /* pe after second guess */ FDKaacEnc_calcPe(psyOutElement[elementId]->psyOutChannel, qcElement[elementId]->qcOutChannel, peData, nChannels); redPeGlobal += peData->pe; } /* EOF DSE-suppression */ } /* EOF for all elements... */ iter++; } /* EOF while */ /* ------------------------------------------------------- */ /* Part IV: if still required, further reduce constraints */ /* ------------------------------------------------------- */ /* 1.0* 1.15* 1.20* * desiredPe desiredPe desiredPe * | | | * ...XXXXXXXXXXXXXXXXXXXXXXXXXXX| | * | | |XXXXXXXXXXX... * | |XXXXXXXXXXX| * --- A --- | --- B --- | --- C --- * * (X): redPeGlobal * (A): FDKaacEnc_correctThresh() * (B): FDKaacEnc_allowMoreHoles() * (C): FDKaacEnc_reduceMinSnr() */ /* correct thresholds to get closer to the desired pe */ if (redPeGlobal > desiredPe) { FDKaacEnc_correctThresh(cm, qcElement, psyOutElement, pAhFlag, pThrExp, reductionValue_m, reductionValue_e, desiredPe - redPeGlobal, processElements, elementOffset); /* update PE */ redPeGlobal = 0; for (elementId = elementOffset; elementId < nElements; elementId++) { if (cm->elInfo[elementId].elType != ID_DSE) { INT nChannels = cm->elInfo[elementId].nChannelsInEl; PE_DATA *peData = &qcElement[elementId]->peData; /* pe after correctThresh */ FDKaacEnc_calcPe(psyOutElement[elementId]->psyOutChannel, qcElement[elementId]->qcOutChannel, peData, nChannels); redPeGlobal += peData->pe; } /* EOF DSE-suppression */ } /* EOF for all elements... */ } if (redPeGlobal > desiredPe) { /* reduce pe by reducing minSnr requirements */ FDKaacEnc_reduceMinSnr( cm, qcElement, psyOutElement, pAhFlag, (fMultI(FL2FXCONST_DBL(0.15f), desiredPe) + desiredPe), &redPeGlobal, processElements, elementOffset); /* reduce pe by allowing additional spectral holes */ FDKaacEnc_allowMoreHoles(cm, qcElement, psyOutElement, AdjThrStateElement, pAhFlag, desiredPe, redPeGlobal, processElements, elementOffset); } } /* similar to FDKaacEnc_adaptThresholdsToPe(), for VBR-mode */ static void FDKaacEnc_AdaptThresholdsVBR( QC_OUT_CHANNEL *const qcOutChannel[(2)], const PSY_OUT_CHANNEL *const psyOutChannel[(2)], ATS_ELEMENT *const AdjThrStateElement, const struct TOOLSINFO *const toolsInfo, const INT nChannels) { UCHAR(*pAhFlag)[MAX_GROUPED_SFB]; FIXP_DBL(*pThrExp)[MAX_GROUPED_SFB]; /* allocate scratch memory */ C_ALLOC_SCRATCH_START(_pAhFlag, UCHAR, (2) * MAX_GROUPED_SFB) C_ALLOC_SCRATCH_START(_pThrExp, FIXP_DBL, (2) * MAX_GROUPED_SFB) pAhFlag = (UCHAR(*)[MAX_GROUPED_SFB])_pAhFlag; pThrExp = (FIXP_DBL(*)[MAX_GROUPED_SFB])_pThrExp; /* thresholds to the power of redExp */ FDKaacEnc_calcThreshExp(pThrExp, qcOutChannel, psyOutChannel, nChannels); /* lower the minSnr requirements for low energies compared to the average energy in this frame */ FDKaacEnc_adaptMinSnr(qcOutChannel, psyOutChannel, &AdjThrStateElement->minSnrAdaptParam, nChannels); /* init ahFlag (0: no ah necessary, 1: ah possible, 2: ah active */ FDKaacEnc_initAvoidHoleFlag(qcOutChannel, psyOutChannel, pAhFlag, toolsInfo, nChannels, &AdjThrStateElement->ahParam); /* reduce thresholds */ FDKaacEnc_reduceThresholdsVBR(qcOutChannel, psyOutChannel, pAhFlag, pThrExp, nChannels, AdjThrStateElement->vbrQualFactor, &AdjThrStateElement->chaosMeasureOld); /* free scratch memory */ C_ALLOC_SCRATCH_END(_pThrExp, FIXP_DBL, (2) * MAX_GROUPED_SFB) C_ALLOC_SCRATCH_END(_pAhFlag, UCHAR, (2) * MAX_GROUPED_SFB) } /***************************************************************************** functionname: FDKaacEnc_calcBitSave description: Calculates percentage of bit save, see figure below returns: input: parameters and bitres-fullness output: percentage of bit save *****************************************************************************/ /* bitsave maxBitSave(%)| clipLow |---\ | \ | \ | \ | \ |--------\--------------> bitres | \ minBitSave(%)| \------------ clipHigh maxBitres */ static FIXP_DBL FDKaacEnc_calcBitSave(FIXP_DBL fillLevel, const FIXP_DBL clipLow, const FIXP_DBL clipHigh, const FIXP_DBL minBitSave, const FIXP_DBL maxBitSave, const FIXP_DBL bitsave_slope) { FIXP_DBL bitsave; fillLevel = fixMax(fillLevel, clipLow); fillLevel = fixMin(fillLevel, clipHigh); bitsave = maxBitSave - fMult((fillLevel - clipLow), bitsave_slope); return (bitsave); } /***************************************************************************** functionname: FDKaacEnc_calcBitSpend description: Calculates percentage of bit spend, see figure below returns: input: parameters and bitres-fullness output: percentage of bit spend *****************************************************************************/ /* bitspend clipHigh maxBitSpend(%)| /-----------maxBitres | / | / | / | / | / |----/-----------------> bitres | / minBitSpend(%)|--/ clipLow */ static FIXP_DBL FDKaacEnc_calcBitSpend(FIXP_DBL fillLevel, const FIXP_DBL clipLow, const FIXP_DBL clipHigh, const FIXP_DBL minBitSpend, const FIXP_DBL maxBitSpend, const FIXP_DBL bitspend_slope) { FIXP_DBL bitspend; fillLevel = fixMax(fillLevel, clipLow); fillLevel = fixMin(fillLevel, clipHigh); bitspend = minBitSpend + fMult(fillLevel - clipLow, bitspend_slope); return (bitspend); } /***************************************************************************** functionname: FDKaacEnc_adjustPeMinMax() description: adjusts peMin and peMax parameters over time returns: input: current pe, peMin, peMax, bitres size output: adjusted peMin/peMax *****************************************************************************/ static void FDKaacEnc_adjustPeMinMax(const INT currPe, INT *peMin, INT *peMax) { FIXP_DBL minFacHi = FL2FXCONST_DBL(0.3f), maxFacHi = (FIXP_DBL)MAXVAL_DBL, minFacLo = FL2FXCONST_DBL(0.14f), maxFacLo = FL2FXCONST_DBL(0.07f); INT diff; INT minDiff_fix = fMultI(FL2FXCONST_DBL(0.1666666667f), currPe); if (currPe > *peMax) { diff = (currPe - *peMax); *peMin += fMultI(minFacHi, diff); *peMax += fMultI(maxFacHi, diff); } else if (currPe < *peMin) { diff = (*peMin - currPe); *peMin -= fMultI(minFacLo, diff); *peMax -= fMultI(maxFacLo, diff); } else { *peMin += fMultI(minFacHi, (currPe - *peMin)); *peMax -= fMultI(maxFacLo, (*peMax - currPe)); } if ((*peMax - *peMin) < minDiff_fix) { INT peMax_fix = *peMax, peMin_fix = *peMin; FIXP_DBL partLo_fix, partHi_fix; partLo_fix = (FIXP_DBL)fixMax(0, currPe - peMin_fix); partHi_fix = (FIXP_DBL)fixMax(0, peMax_fix - currPe); peMax_fix = (INT)(currPe + fMultI(fDivNorm(partHi_fix, (partLo_fix + partHi_fix)), minDiff_fix)); peMin_fix = (INT)(currPe - fMultI(fDivNorm(partLo_fix, (partLo_fix + partHi_fix)), minDiff_fix)); peMin_fix = fixMax(0, peMin_fix); *peMax = peMax_fix; *peMin = peMin_fix; } } /***************************************************************************** functionname: BitresCalcBitFac description: calculates factor of spending bits for one frame 1.0 : take all frame dynpart bits >1.0 : take all frame dynpart bits + bitres <1.0 : put bits in bitreservoir returns: BitFac input: bitres-fullness, pe, blockType, parameter-settings output: *****************************************************************************/ /* bitfac(%) pemax bitspend(%) | /-----------maxBitres | / | / | / | / | / |----/-----------------> pe | / bitsave(%) |--/ pemin */ void FDKaacEnc_bitresCalcBitFac(const INT bitresBits, const INT maxBitresBits, const INT pe, const INT lastWindowSequence, const INT avgBits, const FIXP_DBL maxBitFac, const ADJ_THR_STATE *const AdjThr, ATS_ELEMENT *const adjThrChan, FIXP_DBL *const pBitresFac, INT *const pBitresFac_e) { const BRES_PARAM *bresParam; INT pex; FIXP_DBL fillLevel; INT fillLevel_e = 0; FIXP_DBL bitresFac; INT bitresFac_e; FIXP_DBL bitSave, bitSpend; FIXP_DBL bitsave_slope, bitspend_slope; FIXP_DBL fillLevel_fix = MAXVAL_DBL; FIXP_DBL slope = MAXVAL_DBL; if (lastWindowSequence != SHORT_WINDOW) { bresParam = &(AdjThr->bresParamLong); bitsave_slope = FL2FXCONST_DBL(0.466666666); bitspend_slope = FL2FXCONST_DBL(0.666666666); } else { bresParam = &(AdjThr->bresParamShort); bitsave_slope = (FIXP_DBL)0x2E8BA2E9; bitspend_slope = (FIXP_DBL)0x7fffffff; } // fillLevel = (float)(bitresBits+avgBits) / (float)(maxBitresBits + avgBits); if (bitresBits < maxBitresBits) { fillLevel_fix = fDivNorm(bitresBits, maxBitresBits); } pex = fMax(pe, adjThrChan->peMin); pex = fMin(pex, adjThrChan->peMax); bitSave = FDKaacEnc_calcBitSave( fillLevel_fix, bresParam->clipSaveLow, bresParam->clipSaveHigh, bresParam->minBitSave, bresParam->maxBitSave, bitsave_slope); bitSpend = FDKaacEnc_calcBitSpend( fillLevel_fix, bresParam->clipSpendLow, bresParam->clipSpendHigh, bresParam->minBitSpend, bresParam->maxBitSpend, bitspend_slope); slope = schur_div((pex - adjThrChan->peMin), (adjThrChan->peMax - adjThrChan->peMin), 31); /* scale down by 1 bit because the result of the following addition can be * bigger than 1 (though smaller than 2) */ bitresFac = ((FIXP_DBL)(MAXVAL_DBL >> 1) - (bitSave >> 1)); bitresFac_e = 1; /* exp=1 */ bitresFac = fMultAddDiv2(bitresFac, slope, bitSpend + bitSave); /* exp=1 */ /*** limit bitresFac for small bitreservoir ***/ fillLevel = fDivNorm(bitresBits, avgBits, &fillLevel_e); if (fillLevel_e < 0) { fillLevel = scaleValue(fillLevel, fillLevel_e); fillLevel_e = 0; } /* shift down value by 1 because of summation, ... */ fillLevel >>= 1; fillLevel_e += 1; /* ..., this summation: */ fillLevel += scaleValue(FL2FXCONST_DBL(0.7f), -fillLevel_e); /* set bitresfactor to same exponent as fillLevel */ if (scaleValue(bitresFac, -fillLevel_e + 1) > fillLevel) { bitresFac = fillLevel; bitresFac_e = fillLevel_e; } /* limit bitresFac for high bitrates */ if (scaleValue(bitresFac, bitresFac_e - (DFRACT_BITS - 1 - 24)) > maxBitFac) { bitresFac = maxBitFac; bitresFac_e = (DFRACT_BITS - 1 - 24); } FDKaacEnc_adjustPeMinMax(pe, &adjThrChan->peMin, &adjThrChan->peMax); /* output values */ *pBitresFac = bitresFac; *pBitresFac_e = bitresFac_e; } /***************************************************************************** functionname: FDKaacEnc_AdjThrNew description: allocate ADJ_THR_STATE *****************************************************************************/ INT FDKaacEnc_AdjThrNew(ADJ_THR_STATE **phAdjThr, INT nElements) { INT err = 0; INT i; ADJ_THR_STATE *hAdjThr = GetRam_aacEnc_AdjustThreshold(); if (hAdjThr == NULL) { err = 1; goto bail; } for (i = 0; i < nElements; i++) { hAdjThr->adjThrStateElem[i] = GetRam_aacEnc_AdjThrStateElement(i); if (hAdjThr->adjThrStateElem[i] == NULL) { err = 1; goto bail; } } bail: *phAdjThr = hAdjThr; return err; } /***************************************************************************** functionname: FDKaacEnc_AdjThrInit description: initialize ADJ_THR_STATE *****************************************************************************/ void FDKaacEnc_AdjThrInit( ADJ_THR_STATE *const hAdjThr, const INT meanPe, const INT invQuant, const CHANNEL_MAPPING *const channelMapping, const INT sampleRate, const INT totalBitrate, const INT isLowDelay, const AACENC_BITRES_MODE bitResMode, const INT dZoneQuantEnable, const INT bitDistributionMode, const FIXP_DBL vbrQualFactor) { INT i; FIXP_DBL POINT8 = FL2FXCONST_DBL(0.8f); FIXP_DBL POINT6 = FL2FXCONST_DBL(0.6f); if (bitDistributionMode == 1) { hAdjThr->bitDistributionMode = AACENC_BD_MODE_INTRA_ELEMENT; } else { hAdjThr->bitDistributionMode = AACENC_BD_MODE_INTER_ELEMENT; } /* Max number of iterations in second guess is 3 for lowdelay aot and for configurations with multiple audio elements in general, otherwise iteration value is always 1. */ hAdjThr->maxIter2ndGuess = (isLowDelay != 0 || channelMapping->nElements > 1) ? 3 : 1; /* common for all elements: */ /* parameters for bitres control */ hAdjThr->bresParamLong.clipSaveLow = (FIXP_DBL)0x1999999a; /* FL2FXCONST_DBL(0.2f); */ hAdjThr->bresParamLong.clipSaveHigh = (FIXP_DBL)0x7999999a; /* FL2FXCONST_DBL(0.95f); */ hAdjThr->bresParamLong.minBitSave = (FIXP_DBL)0xf999999a; /* FL2FXCONST_DBL(-0.05f); */ hAdjThr->bresParamLong.maxBitSave = (FIXP_DBL)0x26666666; /* FL2FXCONST_DBL(0.3f); */ hAdjThr->bresParamLong.clipSpendLow = (FIXP_DBL)0x1999999a; /* FL2FXCONST_DBL(0.2f); */ hAdjThr->bresParamLong.clipSpendHigh = (FIXP_DBL)0x7999999a; /* FL2FXCONST_DBL(0.95f); */ hAdjThr->bresParamLong.minBitSpend = (FIXP_DBL)0xf3333333; /* FL2FXCONST_DBL(-0.10f); */ hAdjThr->bresParamLong.maxBitSpend = (FIXP_DBL)0x33333333; /* FL2FXCONST_DBL(0.4f); */ hAdjThr->bresParamShort.clipSaveLow = (FIXP_DBL)0x199999a0; /* FL2FXCONST_DBL(0.2f); */ hAdjThr->bresParamShort.clipSaveHigh = (FIXP_DBL)0x5fffffff; /* FL2FXCONST_DBL(0.75f); */ hAdjThr->bresParamShort.minBitSave = (FIXP_DBL)0x00000000; /* FL2FXCONST_DBL(0.0f); */ hAdjThr->bresParamShort.maxBitSave = (FIXP_DBL)0x199999a0; /* FL2FXCONST_DBL(0.2f); */ hAdjThr->bresParamShort.clipSpendLow = (FIXP_DBL)0x199999a0; /* FL2FXCONST_DBL(0.2f); */ hAdjThr->bresParamShort.clipSpendHigh = (FIXP_DBL)0x5fffffff; /* FL2FXCONST_DBL(0.75f); */ hAdjThr->bresParamShort.minBitSpend = (FIXP_DBL)0xf9999998; /* FL2FXCONST_DBL(-0.05f); */ hAdjThr->bresParamShort.maxBitSpend = (FIXP_DBL)0x40000000; /* FL2FXCONST_DBL(0.5f); */ /* specific for each element: */ for (i = 0; i < channelMapping->nElements; i++) { const FIXP_DBL relativeBits = channelMapping->elInfo[i].relativeBits; const INT nChannelsInElement = channelMapping->elInfo[i].nChannelsInEl; const INT bitrateInElement = (relativeBits != (FIXP_DBL)MAXVAL_DBL) ? (INT)fMultNorm(relativeBits, (FIXP_DBL)totalBitrate) : totalBitrate; const INT chBitrate = bitrateInElement >> (nChannelsInElement == 1 ? 0 : 1); ATS_ELEMENT *atsElem = hAdjThr->adjThrStateElem[i]; MINSNR_ADAPT_PARAM *msaParam = &atsElem->minSnrAdaptParam; /* parameters for bitres control */ if (isLowDelay) { atsElem->peMin = fMultI(POINT8, meanPe); atsElem->peMax = fMultI(POINT6, meanPe) << 1; } else { atsElem->peMin = fMultI(POINT8, meanPe) >> 1; atsElem->peMax = fMultI(POINT6, meanPe); } /* for use in FDKaacEnc_reduceThresholdsVBR */ atsElem->chaosMeasureOld = FL2FXCONST_DBL(0.3f); /* additional pe offset to correct pe2bits for low bitrates */ /* ---- no longer necessary, set by table ----- */ atsElem->peOffset = 0; /* vbr initialisation */ atsElem->vbrQualFactor = vbrQualFactor; if (chBitrate < 32000) { atsElem->peOffset = fixMax(50, 100 - fMultI((FIXP_DBL)0x666667, chBitrate)); } /* avoid hole parameters */ if (chBitrate >= 20000) { atsElem->ahParam.modifyMinSnr = TRUE; atsElem->ahParam.startSfbL = 15; atsElem->ahParam.startSfbS = 3; } else { atsElem->ahParam.modifyMinSnr = FALSE; atsElem->ahParam.startSfbL = 0; atsElem->ahParam.startSfbS = 0; } /* minSnr adaptation */ msaParam->maxRed = FL2FXCONST_DBL(0.00390625f); /* 0.25f/64.0f */ /* start adaptation of minSnr for avgEn/sfbEn > startRatio */ msaParam->startRatio = FL2FXCONST_DBL(0.05190512648f); /* ld64(10.0f) */ /* maximum minSnr reduction to minSnr^maxRed is reached for avgEn/sfbEn >= maxRatio */ /* msaParam->maxRatio = 1000.0f; */ /*msaParam->redRatioFac = ((float)1.0f - msaParam->maxRed) / * ((float)10.0f*log10(msaParam->startRatio/msaParam->maxRatio)/log10(2.0f)*(float)0.3010299956f);*/ msaParam->redRatioFac = FL2FXCONST_DBL(-0.375f); /* -0.0375f * 10.0f */ /*msaParam->redOffs = (float)1.0f - msaParam->redRatioFac * (float)10.0f * * log10(msaParam->startRatio)/log10(2.0f) * (float)0.3010299956f;*/ msaParam->redOffs = FL2FXCONST_DBL(0.021484375); /* 1.375f/64.0f */ /* init pe correction */ atsElem->peCorrectionFactor_m = FL2FXCONST_DBL(0.5f); /* 1.0 */ atsElem->peCorrectionFactor_e = 1; atsElem->dynBitsLast = -1; atsElem->peLast = 0; /* init bits to pe factor */ /* init bits2PeFactor */ FDKaacEnc_InitBits2PeFactor( &atsElem->bits2PeFactor_m, &atsElem->bits2PeFactor_e, bitrateInElement, nChannelsInElement, sampleRate, isLowDelay, dZoneQuantEnable, invQuant); } /* for nElements */ } /***************************************************************************** functionname: FDKaacEnc_FDKaacEnc_calcPeCorrection description: calc desired pe *****************************************************************************/ static void FDKaacEnc_FDKaacEnc_calcPeCorrection( FIXP_DBL *const correctionFac_m, INT *const correctionFac_e, const INT peAct, const INT peLast, const INT bitsLast, const FIXP_DBL bits2PeFactor_m, const INT bits2PeFactor_e) { if ((bitsLast > 0) && (peAct < 1.5f * peLast) && (peAct > 0.7f * peLast) && (FDKaacEnc_bits2pe2(bitsLast, fMult(FL2FXCONST_DBL(1.2f / 2.f), bits2PeFactor_m), bits2PeFactor_e + 1) > peLast) && (FDKaacEnc_bits2pe2(bitsLast, fMult(FL2FXCONST_DBL(0.65f), bits2PeFactor_m), bits2PeFactor_e) < peLast)) { FIXP_DBL corrFac = *correctionFac_m; int scaling = 0; FIXP_DBL denum = (FIXP_DBL)FDKaacEnc_bits2pe2(bitsLast, bits2PeFactor_m, bits2PeFactor_e); FIXP_DBL newFac = fDivNorm((FIXP_DBL)peLast, denum, &scaling); /* dead zone, newFac and corrFac are scaled by 0.5 */ if ((FIXP_DBL)peLast <= denum) { /* ratio <= 1.f */ newFac = fixMax( scaleValue(fixMin(fMult(FL2FXCONST_DBL(1.1f / 2.f), newFac), scaleValue(FL2FXCONST_DBL(1.f / 2.f), -scaling)), scaling), FL2FXCONST_DBL(0.85f / 2.f)); } else { /* ratio < 1.f */ newFac = fixMax( fixMin(scaleValue(fMult(FL2FXCONST_DBL(0.9f / 2.f), newFac), scaling), FL2FXCONST_DBL(1.15f / 2.f)), FL2FXCONST_DBL(1.f / 2.f)); } if (((newFac > FL2FXCONST_DBL(1.f / 2.f)) && (corrFac < FL2FXCONST_DBL(1.f / 2.f))) || ((newFac < FL2FXCONST_DBL(1.f / 2.f)) && (corrFac > FL2FXCONST_DBL(1.f / 2.f)))) { corrFac = FL2FXCONST_DBL(1.f / 2.f); } /* faster adaptation towards 1.0, slower in the other direction */ if ((corrFac < FL2FXCONST_DBL(1.f / 2.f) && newFac < corrFac) || (corrFac > FL2FXCONST_DBL(1.f / 2.f) && newFac > corrFac)) { corrFac = fMult(FL2FXCONST_DBL(0.85f), corrFac) + fMult(FL2FXCONST_DBL(0.15f), newFac); } else { corrFac = fMult(FL2FXCONST_DBL(0.7f), corrFac) + fMult(FL2FXCONST_DBL(0.3f), newFac); } corrFac = fixMax(fixMin(corrFac, FL2FXCONST_DBL(1.15f / 2.f)), FL2FXCONST_DBL(0.85 / 2.f)); *correctionFac_m = corrFac; *correctionFac_e = 1; } else { *correctionFac_m = FL2FXCONST_DBL(1.f / 2.f); *correctionFac_e = 1; } } static void FDKaacEnc_calcPeCorrectionLowBitRes( FIXP_DBL *const correctionFac_m, INT *const correctionFac_e, const INT peLast, const INT bitsLast, const INT bitresLevel, const INT nChannels, const FIXP_DBL bits2PeFactor_m, const INT bits2PeFactor_e) { /* tuning params */ const FIXP_DBL amp = FL2FXCONST_DBL(0.005); const FIXP_DBL maxDiff = FL2FXCONST_DBL(0.25f); if (bitsLast > 0) { /* Estimate deviation of granted and used dynamic bits in previous frame, in * PE units */ const int bitsBalLast = peLast - FDKaacEnc_bits2pe2(bitsLast, bits2PeFactor_m, bits2PeFactor_e); /* reserve n bits per channel */ int headroom = (bitresLevel >= 50 * nChannels) ? 0 : (100 * nChannels); /* in PE units */ headroom = FDKaacEnc_bits2pe2(headroom, bits2PeFactor_m, bits2PeFactor_e); /* * diff = amp * ((bitsBalLast - headroom) / (bitresLevel + headroom) * diff = max ( min ( diff, maxDiff, -maxDiff)) / 2 */ FIXP_DBL denominator = (FIXP_DBL)FDKaacEnc_bits2pe2( bitresLevel, bits2PeFactor_m, bits2PeFactor_e) + (FIXP_DBL)headroom; int scaling = 0; FIXP_DBL diff = (bitsBalLast >= headroom) ? fMult(amp, fDivNorm((FIXP_DBL)(bitsBalLast - headroom), denominator, &scaling)) : -fMult(amp, fDivNorm(-(FIXP_DBL)(bitsBalLast - headroom), denominator, &scaling)); scaling -= 1; /* divide by 2 */ diff = (scaling <= 0) ? fMax(fMin(diff >> (-scaling), maxDiff >> 1), -maxDiff >> 1) : fMax(fMin(diff, maxDiff >> (1 + scaling)), -maxDiff >> (1 + scaling)) << scaling; /* * corrFac += diff * corrFac = max ( min ( corrFac/2.f, 1.f/2.f, 0.75f/2.f ) ) */ *correctionFac_m = fMax(fMin((*correctionFac_m) + diff, FL2FXCONST_DBL(1.0f / 2.f)), FL2FXCONST_DBL(0.75f / 2.f)); *correctionFac_e = 1; } else { *correctionFac_m = FL2FXCONST_DBL(0.75 / 2.f); *correctionFac_e = 1; } } void FDKaacEnc_DistributeBits( ADJ_THR_STATE *adjThrState, ATS_ELEMENT *AdjThrStateElement, PSY_OUT_CHANNEL *psyOutChannel[(2)], PE_DATA *peData, INT *grantedPe, INT *grantedPeCorr, const INT nChannels, const INT commonWindow, const INT grantedDynBits, const INT bitresBits, const INT maxBitresBits, const FIXP_DBL maxBitFac, const AACENC_BITRES_MODE bitResMode) { FIXP_DBL bitFactor; INT bitFactor_e; INT noRedPe = peData->pe; /* prefer short windows for calculation of bitFactor */ INT curWindowSequence = LONG_WINDOW; if (nChannels == 2) { if ((psyOutChannel[0]->lastWindowSequence == SHORT_WINDOW) || (psyOutChannel[1]->lastWindowSequence == SHORT_WINDOW)) { curWindowSequence = SHORT_WINDOW; } } else { curWindowSequence = psyOutChannel[0]->lastWindowSequence; } if (grantedDynBits >= 1) { if (bitResMode != AACENC_BR_MODE_FULL) { /* small or disabled bitreservoir */ *grantedPe = FDKaacEnc_bits2pe2(grantedDynBits, AdjThrStateElement->bits2PeFactor_m, AdjThrStateElement->bits2PeFactor_e); } else { /* factor dependend on current fill level and pe */ FDKaacEnc_bitresCalcBitFac( bitresBits, maxBitresBits, noRedPe, curWindowSequence, grantedDynBits, maxBitFac, adjThrState, AdjThrStateElement, &bitFactor, &bitFactor_e); /* desired pe for actual frame */ /* Worst case max of grantedDynBits is = 1024 * 5.27 * 2 */ *grantedPe = FDKaacEnc_bits2pe2( grantedDynBits, fMult(bitFactor, AdjThrStateElement->bits2PeFactor_m), AdjThrStateElement->bits2PeFactor_e + bitFactor_e); } } else { *grantedPe = 0; /* prevent divsion by 0 */ } /* correction of pe value */ switch (bitResMode) { case AACENC_BR_MODE_DISABLED: case AACENC_BR_MODE_REDUCED: /* correction of pe value for low bitres */ FDKaacEnc_calcPeCorrectionLowBitRes( &AdjThrStateElement->peCorrectionFactor_m, &AdjThrStateElement->peCorrectionFactor_e, AdjThrStateElement->peLast, AdjThrStateElement->dynBitsLast, bitresBits, nChannels, AdjThrStateElement->bits2PeFactor_m, AdjThrStateElement->bits2PeFactor_e); break; case AACENC_BR_MODE_FULL: default: /* correction of pe value for high bitres */ FDKaacEnc_FDKaacEnc_calcPeCorrection( &AdjThrStateElement->peCorrectionFactor_m, &AdjThrStateElement->peCorrectionFactor_e, fixMin(*grantedPe, noRedPe), AdjThrStateElement->peLast, AdjThrStateElement->dynBitsLast, AdjThrStateElement->bits2PeFactor_m, AdjThrStateElement->bits2PeFactor_e); break; } *grantedPeCorr = (INT)(fMult((FIXP_DBL)(*grantedPe << Q_AVGBITS), AdjThrStateElement->peCorrectionFactor_m) >> (Q_AVGBITS - AdjThrStateElement->peCorrectionFactor_e)); /* update last pe */ AdjThrStateElement->peLast = *grantedPe; AdjThrStateElement->dynBitsLast = -1; } /***************************************************************************** functionname: FDKaacEnc_AdjustThresholds description: adjust thresholds *****************************************************************************/ void FDKaacEnc_AdjustThresholds( ADJ_THR_STATE *const hAdjThr, QC_OUT_ELEMENT *const qcElement[((8))], QC_OUT *const qcOut, const PSY_OUT_ELEMENT *const psyOutElement[((8))], const INT CBRbitrateMode, const CHANNEL_MAPPING *const cm) { int i; if (CBRbitrateMode) { /* In case, no bits must be shifted between different elements, */ /* an element-wise execution of the pe-dependent threshold- */ /* adaption becomes necessary... */ if (hAdjThr->bitDistributionMode == AACENC_BD_MODE_INTRA_ELEMENT) { for (i = 0; i < cm->nElements; i++) { ELEMENT_INFO elInfo = cm->elInfo[i]; if ((elInfo.elType == ID_SCE) || (elInfo.elType == ID_CPE) || (elInfo.elType == ID_LFE)) { /* qcElement[i]->grantedPe = 2000; */ /* Use this only for debugging */ // if (totalGrantedPeCorr < totalNoRedPe) { if (qcElement[i]->grantedPeCorr < qcElement[i]->peData.pe) { /* calc threshold necessary for desired pe */ FDKaacEnc_adaptThresholdsToPe( cm, hAdjThr->adjThrStateElem, qcElement, psyOutElement, qcElement[i]->grantedPeCorr, hAdjThr->maxIter2ndGuess, 1, /* Process only 1 element */ i /* Process exactly THIS element */ ); } } /* -end- if(ID_SCE || ID_CPE || ID_LFE) */ } /* -end- element loop */ } /* AACENC_BD_MODE_INTRA_ELEMENT */ else if (hAdjThr->bitDistributionMode == AACENC_BD_MODE_INTER_ELEMENT) { /* Use global Pe to obtain the thresholds? */ if (qcOut->totalGrantedPeCorr < qcOut->totalNoRedPe) { /* add equal loadness quantization noise to match the */ /* desired pe calc threshold necessary for desired pe */ /* Now carried out globally to cover all(!) channels. */ FDKaacEnc_adaptThresholdsToPe(cm, hAdjThr->adjThrStateElem, qcElement, psyOutElement, qcOut->totalGrantedPeCorr, hAdjThr->maxIter2ndGuess, cm->nElements, /* Process all elements */ 0); /* Process exactly THIS element */ } else { /* In case global pe doesn't need to be reduced check each element to hold estimated bitrate below maximum element bitrate. */ for (i = 0; i < cm->nElements; i++) { if ((cm->elInfo[i].elType == ID_SCE) || (cm->elInfo[i].elType == ID_CPE) || (cm->elInfo[i].elType == ID_LFE)) { /* Element pe applies to dynamic bits of maximum element bitrate. */ const int maxElementPe = FDKaacEnc_bits2pe2( (cm->elInfo[i].nChannelsInEl * MIN_BUFSIZE_PER_EFF_CHAN) - qcElement[i]->staticBitsUsed - qcElement[i]->extBitsUsed, hAdjThr->adjThrStateElem[i]->bits2PeFactor_m, hAdjThr->adjThrStateElem[i]->bits2PeFactor_e); if (maxElementPe < qcElement[i]->peData.pe) { FDKaacEnc_adaptThresholdsToPe( cm, hAdjThr->adjThrStateElem, qcElement, psyOutElement, maxElementPe, hAdjThr->maxIter2ndGuess, 1, i); } } /* -end- if(ID_SCE || ID_CPE || ID_LFE) */ } /* -end- element loop */ } /* (qcOut->totalGrantedPeCorr < qcOut->totalNoRedPe) */ } /* AACENC_BD_MODE_INTER_ELEMENT */ } else { for (i = 0; i < cm->nElements; i++) { ELEMENT_INFO elInfo = cm->elInfo[i]; if ((elInfo.elType == ID_SCE) || (elInfo.elType == ID_CPE) || (elInfo.elType == ID_LFE)) { /* for VBR-mode */ FDKaacEnc_AdaptThresholdsVBR( qcElement[i]->qcOutChannel, psyOutElement[i]->psyOutChannel, hAdjThr->adjThrStateElem[i], &psyOutElement[i]->toolsInfo, cm->elInfo[i].nChannelsInEl); } /* -end- if(ID_SCE || ID_CPE || ID_LFE) */ } /* -end- element loop */ } for (i = 0; i < cm->nElements; i++) { int ch, sfb, sfbGrp; /* no weighting of threholds and energies for mlout */ /* weight energies and thresholds */ for (ch = 0; ch < cm->elInfo[i].nChannelsInEl; ch++) { QC_OUT_CHANNEL *pQcOutCh = qcElement[i]->qcOutChannel[ch]; for (sfbGrp = 0; sfbGrp < psyOutElement[i]->psyOutChannel[ch]->sfbCnt; sfbGrp += psyOutElement[i]->psyOutChannel[ch]->sfbPerGroup) { for (sfb = 0; sfb < psyOutElement[i]->psyOutChannel[ch]->maxSfbPerGroup; sfb++) { pQcOutCh->sfbThresholdLdData[sfb + sfbGrp] += pQcOutCh->sfbEnFacLd[sfb + sfbGrp]; } } } } } void FDKaacEnc_AdjThrClose(ADJ_THR_STATE **phAdjThr) { INT i; ADJ_THR_STATE *hAdjThr = *phAdjThr; if (hAdjThr != NULL) { for (i = 0; i < ((8)); i++) { if (hAdjThr->adjThrStateElem[i] != NULL) { FreeRam_aacEnc_AdjThrStateElement(&hAdjThr->adjThrStateElem[i]); } } FreeRam_aacEnc_AdjustThreshold(phAdjThr); } } fdk-aac-2.0.2/libAACenc/src/adj_thr.h000066400000000000000000000172131404225310700171170ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Werner Description: Threshold compensation *******************************************************************************/ #ifndef ADJ_THR_H #define ADJ_THR_H #include "common_fix.h" #include "adj_thr_data.h" #include "qc_data.h" #include "line_pe.h" #include "interface.h" /***************************************************************************** functionname: FDKaacEnc_peCalculation description: *****************************************************************************/ void FDKaacEnc_peCalculation(PE_DATA *const peData, const PSY_OUT_CHANNEL *const psyOutChannel[(2)], QC_OUT_CHANNEL *const qcOutChannel[(2)], const struct TOOLSINFO *const toolsInfo, ATS_ELEMENT *const adjThrStateElement, const INT nChannels); /***************************************************************************** functionname: FDKaacEnc_AdjThrNew description: allocate ADJ_THR_STATE *****************************************************************************/ INT FDKaacEnc_AdjThrNew(ADJ_THR_STATE **phAdjThr, INT nElements); /***************************************************************************** functionname: FDKaacEnc_AdjThrInit description: initialize ADJ_THR_STATE *****************************************************************************/ void FDKaacEnc_AdjThrInit( ADJ_THR_STATE *const hAdjThr, const INT meanPe, const INT invQuant, const CHANNEL_MAPPING *const channelMapping, const INT sampleRate, const INT totalBitrate, const INT isLowDelay, const AACENC_BITRES_MODE bitResMode, const INT dZoneQuantEnable, const INT bitDistributionMode, const FIXP_DBL vbrQualFactor); /***************************************************************************** functionname: FDKaacEnc_DistributeBits description: *****************************************************************************/ void FDKaacEnc_DistributeBits( ADJ_THR_STATE *adjThrState, ATS_ELEMENT *AdjThrStateElement, PSY_OUT_CHANNEL *psyOutChannel[(2)], PE_DATA *peData, INT *grantedPe, INT *grantedPeCorr, const INT nChannels, const INT commonWindow, const INT avgBits, const INT bitresBits, const INT maxBitresBits, const FIXP_DBL maxBitFac, const AACENC_BITRES_MODE bitResMode); /***************************************************************************** functionname: FDKaacEnc_AdjustThresholds description: adjust thresholds *****************************************************************************/ void FDKaacEnc_AdjustThresholds( ADJ_THR_STATE *const hAdjThr, QC_OUT_ELEMENT *const qcElement[((8))], QC_OUT *const qcOut, const PSY_OUT_ELEMENT *const psyOutElement[((8))], const INT CBRbitrateMode, const CHANNEL_MAPPING *const cm); /***************************************************************************** functionname: FDKaacEnc_AdjThrClose description: *****************************************************************************/ void FDKaacEnc_AdjThrClose(ADJ_THR_STATE **hAdjThr); #endif fdk-aac-2.0.2/libAACenc/src/adj_thr_data.h000066400000000000000000000144701404225310700201120ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Schug / A. Groeschel Description: threshold calculations *******************************************************************************/ #ifndef ADJ_THR_DATA_H #define ADJ_THR_DATA_H #include "psy_const.h" typedef enum { AACENC_BD_MODE_INTER_ELEMENT = 0, AACENC_BD_MODE_INTRA_ELEMENT = 1 } AACENC_BIT_DISTRIBUTION_MODE; typedef enum { AACENC_BR_MODE_FULL = 0, AACENC_BR_MODE_REDUCED = 1, AACENC_BR_MODE_DISABLED = 2 } AACENC_BITRES_MODE; typedef struct { FIXP_DBL clipSaveLow, clipSaveHigh; FIXP_DBL minBitSave, maxBitSave; FIXP_DBL clipSpendLow, clipSpendHigh; FIXP_DBL minBitSpend, maxBitSpend; } BRES_PARAM; typedef struct { INT modifyMinSnr; INT startSfbL, startSfbS; } AH_PARAM; typedef struct { FIXP_DBL maxRed; FIXP_DBL startRatio; FIXP_DBL maxRatio; FIXP_DBL redRatioFac; FIXP_DBL redOffs; } MINSNR_ADAPT_PARAM; typedef struct { /* parameters for bitreservoir control */ INT peMin, peMax; /* constant offset to pe */ INT peOffset; /* constant PeFactor */ FIXP_DBL bits2PeFactor_m; INT bits2PeFactor_e; /* avoid hole parameters */ AH_PARAM ahParam; /* parameters for adaptation of minSnr */ MINSNR_ADAPT_PARAM minSnrAdaptParam; /* values for correction of pe */ INT peLast; INT dynBitsLast; FIXP_DBL peCorrectionFactor_m; INT peCorrectionFactor_e; /* vbr encoding */ FIXP_DBL vbrQualFactor; FIXP_DBL chaosMeasureOld; /* threshold weighting */ FIXP_DBL chaosMeasureEnFac[(2)]; INT lastEnFacPatch[(2)]; } ATS_ELEMENT; typedef struct { BRES_PARAM bresParamLong, bresParamShort; ATS_ELEMENT* adjThrStateElem[((8))]; AACENC_BIT_DISTRIBUTION_MODE bitDistributionMode; INT maxIter2ndGuess; } ADJ_THR_STATE; #endif fdk-aac-2.0.2/libAACenc/src/band_nrg.cpp000066400000000000000000000336131404225310700176130ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Werner Description: Band/Line energy calculations *******************************************************************************/ #include "band_nrg.h" /***************************************************************************** functionname: FDKaacEnc_CalcSfbMaxScaleSpec description: input: output: *****************************************************************************/ void FDKaacEnc_CalcSfbMaxScaleSpec(const FIXP_DBL *RESTRICT mdctSpectrum, const INT *RESTRICT bandOffset, INT *RESTRICT sfbMaxScaleSpec, const INT numBands) { INT i, j; FIXP_DBL maxSpc, tmp; for (i = 0; i < numBands; i++) { maxSpc = (FIXP_DBL)0; DWORD_ALIGNED(mdctSpectrum); for (j = bandOffset[i]; j < bandOffset[i + 1]; j++) { tmp = fixp_abs(mdctSpectrum[j]); maxSpc = fixMax(maxSpc, tmp); } j = CntLeadingZeros(maxSpc) - 1; sfbMaxScaleSpec[i] = fixMin((DFRACT_BITS - 2), j); /* CountLeadingBits() is not necessary here since test value is always > 0 */ } } /***************************************************************************** functionname: FDKaacEnc_CheckBandEnergyOptim description: input: output: *****************************************************************************/ FIXP_DBL FDKaacEnc_CheckBandEnergyOptim(const FIXP_DBL *const RESTRICT mdctSpectrum, const INT *const RESTRICT sfbMaxScaleSpec, const INT *const RESTRICT bandOffset, const INT numBands, FIXP_DBL *RESTRICT bandEnergy, FIXP_DBL *RESTRICT bandEnergyLdData, const INT minSpecShift) { INT i, j, scale, nr = 0; FIXP_DBL maxNrgLd = FL2FXCONST_DBL(-1.0f); FIXP_DBL maxNrg = 0; FIXP_DBL spec; for (i = 0; i < numBands; i++) { scale = fixMax(0, sfbMaxScaleSpec[i] - 4); FIXP_DBL tmp = 0; DWORD_ALIGNED(mdctSpectrum); for (j = bandOffset[i]; j < bandOffset[i + 1]; j++) { spec = mdctSpectrum[j] << scale; tmp = fPow2AddDiv2(tmp, spec); } bandEnergy[i] = tmp << 1; /* calculate ld of bandNrg, subtract scaling */ bandEnergyLdData[i] = CalcLdData(bandEnergy[i]); if (bandEnergyLdData[i] != FL2FXCONST_DBL(-1.0f)) { bandEnergyLdData[i] -= scale * FL2FXCONST_DBL(2.0 / 64); } /* find index of maxNrg */ if (bandEnergyLdData[i] > maxNrgLd) { maxNrgLd = bandEnergyLdData[i]; nr = i; } } /* return unscaled maxNrg*/ scale = fixMax(0, sfbMaxScaleSpec[nr] - 4); scale = fixMax(2 * (minSpecShift - scale), -(DFRACT_BITS - 1)); maxNrg = scaleValue(bandEnergy[nr], scale); return maxNrg; } /***************************************************************************** functionname: FDKaacEnc_CalcBandEnergyOptimLong description: input: output: *****************************************************************************/ INT FDKaacEnc_CalcBandEnergyOptimLong(const FIXP_DBL *RESTRICT mdctSpectrum, INT *RESTRICT sfbMaxScaleSpec, const INT *RESTRICT bandOffset, const INT numBands, FIXP_DBL *RESTRICT bandEnergy, FIXP_DBL *RESTRICT bandEnergyLdData) { INT i, j, shiftBits = 0; FIXP_DBL maxNrgLd = FL2FXCONST_DBL(0.0f); FIXP_DBL spec; for (i = 0; i < numBands; i++) { INT leadingBits = sfbMaxScaleSpec[i] - 4; /* max sfbWidth = 96 ; 2^7=128 => 7/2 = 4 (spc*spc) */ FIXP_DBL tmp = FL2FXCONST_DBL(0.0); /* don't use scaleValue() here, it increases workload quite sufficiently... */ if (leadingBits >= 0) { for (j = bandOffset[i]; j < bandOffset[i + 1]; j++) { spec = mdctSpectrum[j] << leadingBits; tmp = fPow2AddDiv2(tmp, spec); } } else { INT shift = -leadingBits; for (j = bandOffset[i]; j < bandOffset[i + 1]; j++) { spec = mdctSpectrum[j] >> shift; tmp = fPow2AddDiv2(tmp, spec); } } bandEnergy[i] = tmp << 1; } /* calculate ld of bandNrg, subtract scaling */ LdDataVector(bandEnergy, bandEnergyLdData, numBands); for (i = numBands; i-- != 0;) { FIXP_DBL scaleDiff = (sfbMaxScaleSpec[i] - 4) * FL2FXCONST_DBL(2.0 / 64); bandEnergyLdData[i] = (bandEnergyLdData[i] >= ((FL2FXCONST_DBL(-1.f) >> 1) + (scaleDiff >> 1))) ? bandEnergyLdData[i] - scaleDiff : FL2FXCONST_DBL(-1.f); /* find maxNrgLd */ maxNrgLd = fixMax(maxNrgLd, bandEnergyLdData[i]); } if (maxNrgLd <= (FIXP_DBL)0) { for (i = numBands; i-- != 0;) { INT scale = fixMin((sfbMaxScaleSpec[i] - 4) << 1, (DFRACT_BITS - 1)); bandEnergy[i] = scaleValue(bandEnergy[i], -scale); } return 0; } else { /* scale down NRGs */ while (maxNrgLd > FL2FXCONST_DBL(0.0f)) { maxNrgLd -= FL2FXCONST_DBL(2.0 / 64); shiftBits++; } for (i = numBands; i-- != 0;) { INT scale = fixMin(((sfbMaxScaleSpec[i] - 4) + shiftBits) << 1, (DFRACT_BITS - 1)); bandEnergyLdData[i] -= shiftBits * FL2FXCONST_DBL(2.0 / 64); bandEnergy[i] = scaleValue(bandEnergy[i], -scale); } return shiftBits; } } /***************************************************************************** functionname: FDKaacEnc_CalcBandEnergyOptimShort description: input: output: *****************************************************************************/ void FDKaacEnc_CalcBandEnergyOptimShort(const FIXP_DBL *RESTRICT mdctSpectrum, INT *RESTRICT sfbMaxScaleSpec, const INT *RESTRICT bandOffset, const INT numBands, FIXP_DBL *RESTRICT bandEnergy) { INT i, j; for (i = 0; i < numBands; i++) { int leadingBits = sfbMaxScaleSpec[i] - 3; /* max sfbWidth = 36 ; 2^6=64 => 6/2 = 3 (spc*spc) */ FIXP_DBL tmp = FL2FXCONST_DBL(0.0); for (j = bandOffset[i]; j < bandOffset[i + 1]; j++) { FIXP_DBL spec = scaleValue(mdctSpectrum[j], leadingBits); tmp = fPow2AddDiv2(tmp, spec); } bandEnergy[i] = tmp; } for (i = 0; i < numBands; i++) { INT scale = (2 * (sfbMaxScaleSpec[i] - 3)) - 1; /* max sfbWidth = 36 ; 2^6=64 => 6/2 = 3 (spc*spc) */ scale = fixMax(fixMin(scale, (DFRACT_BITS - 1)), -(DFRACT_BITS - 1)); bandEnergy[i] = scaleValueSaturate(bandEnergy[i], -scale); } } /***************************************************************************** functionname: FDKaacEnc_CalcBandNrgMSOpt description: input: output: *****************************************************************************/ void FDKaacEnc_CalcBandNrgMSOpt( const FIXP_DBL *RESTRICT mdctSpectrumLeft, const FIXP_DBL *RESTRICT mdctSpectrumRight, INT *RESTRICT sfbMaxScaleSpecLeft, INT *RESTRICT sfbMaxScaleSpecRight, const INT *RESTRICT bandOffset, const INT numBands, FIXP_DBL *RESTRICT bandEnergyMid, FIXP_DBL *RESTRICT bandEnergySide, INT calcLdData, FIXP_DBL *RESTRICT bandEnergyMidLdData, FIXP_DBL *RESTRICT bandEnergySideLdData) { INT i, j, minScale; FIXP_DBL NrgMid, NrgSide, specm, specs; for (i = 0; i < numBands; i++) { NrgMid = NrgSide = FL2FXCONST_DBL(0.0); minScale = fixMin(sfbMaxScaleSpecLeft[i], sfbMaxScaleSpecRight[i]) - 4; minScale = fixMax(0, minScale); if (minScale > 0) { for (j = bandOffset[i]; j < bandOffset[i + 1]; j++) { FIXP_DBL specL = mdctSpectrumLeft[j] << (minScale - 1); FIXP_DBL specR = mdctSpectrumRight[j] << (minScale - 1); specm = specL + specR; specs = specL - specR; NrgMid = fPow2AddDiv2(NrgMid, specm); NrgSide = fPow2AddDiv2(NrgSide, specs); } } else { for (j = bandOffset[i]; j < bandOffset[i + 1]; j++) { FIXP_DBL specL = mdctSpectrumLeft[j] >> 1; FIXP_DBL specR = mdctSpectrumRight[j] >> 1; specm = specL + specR; specs = specL - specR; NrgMid = fPow2AddDiv2(NrgMid, specm); NrgSide = fPow2AddDiv2(NrgSide, specs); } } bandEnergyMid[i] = fMin(NrgMid, (FIXP_DBL)MAXVAL_DBL >> 1) << 1; bandEnergySide[i] = fMin(NrgSide, (FIXP_DBL)MAXVAL_DBL >> 1) << 1; } if (calcLdData) { LdDataVector(bandEnergyMid, bandEnergyMidLdData, numBands); LdDataVector(bandEnergySide, bandEnergySideLdData, numBands); } for (i = 0; i < numBands; i++) { minScale = fixMin(sfbMaxScaleSpecLeft[i], sfbMaxScaleSpecRight[i]); INT scale = fixMax(0, 2 * (minScale - 4)); if (calcLdData) { /* using the minimal scaling of left and right channel can cause very small energies; check ldNrg before subtract scaling multiplication: fract*INT we don't need fMult */ int minus = scale * FL2FXCONST_DBL(1.0 / 64); if (bandEnergyMidLdData[i] != FL2FXCONST_DBL(-1.0f)) bandEnergyMidLdData[i] -= minus; if (bandEnergySideLdData[i] != FL2FXCONST_DBL(-1.0f)) bandEnergySideLdData[i] -= minus; } scale = fixMin(scale, (DFRACT_BITS - 1)); bandEnergyMid[i] >>= scale; bandEnergySide[i] >>= scale; } } fdk-aac-2.0.2/libAACenc/src/band_nrg.h000066400000000000000000000150301404225310700172510ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Werner Description: Band/Line energy calculation *******************************************************************************/ #ifndef BAND_NRG_H #define BAND_NRG_H #include "common_fix.h" void FDKaacEnc_CalcSfbMaxScaleSpec(const FIXP_DBL *mdctSpectrum, const INT *bandOffset, INT *sfbMaxScaleSpec, const INT numBands); FIXP_DBL FDKaacEnc_CheckBandEnergyOptim(const FIXP_DBL *const RESTRICT mdctSpectrum, const INT *const RESTRICT sfbMaxScaleSpec, const INT *const RESTRICT bandOffset, const INT numBands, FIXP_DBL *RESTRICT bandEnergy, FIXP_DBL *RESTRICT bandEnergyLdData, const INT minSpecShift); INT FDKaacEnc_CalcBandEnergyOptimLong(const FIXP_DBL *mdctSpectrum, INT *sfbMaxScaleSpec, const INT *bandOffset, const INT numBands, FIXP_DBL *bandEnergy, FIXP_DBL *bandEnergyLdData); void FDKaacEnc_CalcBandEnergyOptimShort(const FIXP_DBL *mdctSpectrum, INT *sfbMaxScaleSpec, const INT *bandOffset, const INT numBands, FIXP_DBL *bandEnergy); void FDKaacEnc_CalcBandNrgMSOpt( const FIXP_DBL *RESTRICT mdctSpectrumLeft, const FIXP_DBL *RESTRICT mdctSpectrumRight, INT *RESTRICT sfbMaxScaleSpecLeft, INT *RESTRICT sfbMaxScaleSpecRight, const INT *RESTRICT bandOffset, const INT numBands, FIXP_DBL *RESTRICT bandEnergyMid, FIXP_DBL *RESTRICT bandEnergySide, INT calcLdData, FIXP_DBL *RESTRICT bandEnergyMidLdData, FIXP_DBL *RESTRICT bandEnergySideLdData); #endif fdk-aac-2.0.2/libAACenc/src/bandwidth.cpp000066400000000000000000000316651404225310700200120ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Schug / A. Groeschel Description: bandwidth expert *******************************************************************************/ #include "channel_map.h" #include "bandwidth.h" #include "aacEnc_ram.h" typedef struct { INT chanBitRate; INT bandWidthMono; INT bandWidth2AndMoreChan; } BANDWIDTH_TAB; static const BANDWIDTH_TAB bandWidthTable[] = { {0, 3700, 5000}, {12000, 5000, 6400}, {20000, 6900, 9640}, {28000, 9600, 13050}, {40000, 12060, 14260}, {56000, 13950, 15500}, {72000, 14200, 16120}, {96000, 17000, 17000}, {576001, 17000, 17000}}; static const BANDWIDTH_TAB bandWidthTable_LD_22050[] = { {8000, 2000, 2400}, {12000, 2500, 2700}, {16000, 3300, 3100}, {24000, 6250, 7200}, {32000, 9200, 10500}, {40000, 16000, 16000}, {48000, 16000, 16000}, {282241, 16000, 16000}}; static const BANDWIDTH_TAB bandWidthTable_LD_24000[] = { {8000, 2000, 2000}, {12000, 2000, 2300}, {16000, 2200, 2500}, {24000, 5650, 7200}, {32000, 11600, 12000}, {40000, 12000, 16000}, {48000, 16000, 16000}, {64000, 16000, 16000}, {307201, 16000, 16000}}; static const BANDWIDTH_TAB bandWidthTable_LD_32000[] = { {8000, 2000, 2000}, {12000, 2000, 2000}, {24000, 4250, 7200}, {32000, 8400, 9000}, {40000, 9400, 11300}, {48000, 11900, 14700}, {64000, 14800, 16000}, {76000, 16000, 16000}, {409601, 16000, 16000}}; static const BANDWIDTH_TAB bandWidthTable_LD_44100[] = { {8000, 2000, 2000}, {24000, 2000, 2000}, {32000, 4400, 5700}, {40000, 7400, 8800}, {48000, 9000, 10700}, {56000, 11000, 12900}, {64000, 14400, 15500}, {80000, 16000, 16200}, {96000, 16500, 16000}, {128000, 16000, 16000}, {564481, 16000, 16000}}; static const BANDWIDTH_TAB bandWidthTable_LD_48000[] = { {8000, 2000, 2000}, {24000, 2000, 2000}, {32000, 4400, 5700}, {40000, 7400, 8800}, {48000, 9000, 10700}, {56000, 11000, 12800}, {64000, 14300, 15400}, {80000, 16000, 16200}, {96000, 16500, 16000}, {128000, 16000, 16000}, {614401, 16000, 16000}}; typedef struct { AACENC_BITRATE_MODE bitrateMode; int bandWidthMono; int bandWidth2AndMoreChan; } BANDWIDTH_TAB_VBR; static const BANDWIDTH_TAB_VBR bandWidthTableVBR[] = { {AACENC_BR_MODE_CBR, 0, 0}, {AACENC_BR_MODE_VBR_1, 13000, 13000}, {AACENC_BR_MODE_VBR_2, 13000, 13000}, {AACENC_BR_MODE_VBR_3, 15750, 15750}, {AACENC_BR_MODE_VBR_4, 16500, 16500}, {AACENC_BR_MODE_VBR_5, 19293, 19293}, {AACENC_BR_MODE_SFR, 0, 0}, {AACENC_BR_MODE_FF, 0, 0} }; static INT GetBandwidthEntry(const INT frameLength, const INT sampleRate, const INT chanBitRate, const INT entryNo) { INT bandwidth = -1; const BANDWIDTH_TAB *pBwTab = NULL; INT bwTabSize = 0; switch (frameLength) { case 960: case 1024: pBwTab = bandWidthTable; bwTabSize = sizeof(bandWidthTable) / sizeof(BANDWIDTH_TAB); break; case 120: case 128: case 240: case 256: case 480: case 512: switch (sampleRate) { case 8000: case 11025: case 12000: case 16000: case 22050: pBwTab = bandWidthTable_LD_22050; bwTabSize = sizeof(bandWidthTable_LD_22050) / sizeof(BANDWIDTH_TAB); break; case 24000: pBwTab = bandWidthTable_LD_24000; bwTabSize = sizeof(bandWidthTable_LD_24000) / sizeof(BANDWIDTH_TAB); break; case 32000: pBwTab = bandWidthTable_LD_32000; bwTabSize = sizeof(bandWidthTable_LD_32000) / sizeof(BANDWIDTH_TAB); break; case 44100: pBwTab = bandWidthTable_LD_44100; bwTabSize = sizeof(bandWidthTable_LD_44100) / sizeof(BANDWIDTH_TAB); break; case 48000: case 64000: case 88200: case 96000: pBwTab = bandWidthTable_LD_48000; bwTabSize = sizeof(bandWidthTable_LD_48000) / sizeof(BANDWIDTH_TAB); break; } break; default: pBwTab = NULL; bwTabSize = 0; } if (pBwTab != NULL) { int i; for (i = 0; i < bwTabSize - 1; i++) { if (chanBitRate >= pBwTab[i].chanBitRate && chanBitRate < pBwTab[i + 1].chanBitRate) { switch (frameLength) { case 960: case 1024: bandwidth = (entryNo == 0) ? pBwTab[i].bandWidthMono : pBwTab[i].bandWidth2AndMoreChan; break; case 120: case 128: case 240: case 256: case 480: case 512: { INT q_res = 0; INT startBw = (entryNo == 0) ? pBwTab[i].bandWidthMono : pBwTab[i].bandWidth2AndMoreChan; INT endBw = (entryNo == 0) ? pBwTab[i + 1].bandWidthMono : pBwTab[i + 1].bandWidth2AndMoreChan; INT startBr = pBwTab[i].chanBitRate; INT endBr = pBwTab[i + 1].chanBitRate; FIXP_DBL bwFac_fix = fDivNorm(chanBitRate - startBr, endBr - startBr, &q_res); bandwidth = (INT)scaleValue(fMult(bwFac_fix, (FIXP_DBL)(endBw - startBw)), q_res) + startBw; } break; default: bandwidth = -1; } break; } /* within bitrate range */ } } /* pBwTab!=NULL */ return bandwidth; } AAC_ENCODER_ERROR FDKaacEnc_DetermineBandWidth( const INT proposedBandWidth, const INT bitrate, const AACENC_BITRATE_MODE bitrateMode, const INT sampleRate, const INT frameLength, const CHANNEL_MAPPING *const cm, const CHANNEL_MODE encoderMode, INT *const bandWidth) { AAC_ENCODER_ERROR ErrorStatus = AAC_ENC_OK; INT chanBitRate = bitrate / cm->nChannelsEff; switch (bitrateMode) { case AACENC_BR_MODE_VBR_1: case AACENC_BR_MODE_VBR_2: case AACENC_BR_MODE_VBR_3: case AACENC_BR_MODE_VBR_4: case AACENC_BR_MODE_VBR_5: if (proposedBandWidth != 0) { /* use given bw */ *bandWidth = proposedBandWidth; } else { /* take bw from table */ switch (encoderMode) { case MODE_1: *bandWidth = bandWidthTableVBR[bitrateMode].bandWidthMono; break; case MODE_2: case MODE_1_2: case MODE_1_2_1: case MODE_1_2_2: case MODE_1_2_2_1: case MODE_6_1: case MODE_1_2_2_2_1: case MODE_7_1_REAR_SURROUND: case MODE_7_1_FRONT_CENTER: case MODE_7_1_BACK: case MODE_7_1_TOP_FRONT: *bandWidth = bandWidthTableVBR[bitrateMode].bandWidth2AndMoreChan; break; default: return AAC_ENC_UNSUPPORTED_CHANNELCONFIG; } } break; case AACENC_BR_MODE_CBR: case AACENC_BR_MODE_SFR: case AACENC_BR_MODE_FF: /* bandwidth limiting */ if (proposedBandWidth != 0) { *bandWidth = fMin(proposedBandWidth, fMin(20000, sampleRate >> 1)); } else { /* search reasonable bandwidth */ int entryNo = 0; switch (encoderMode) { case MODE_1: /* mono */ entryNo = 0; /* use mono bandwidth settings */ break; case MODE_2: /* stereo */ case MODE_1_2: /* sce + cpe */ case MODE_1_2_1: /* sce + cpe + sce */ case MODE_1_2_2: /* sce + cpe + cpe */ case MODE_1_2_2_1: /* (5.1) sce + cpe + cpe + lfe */ case MODE_6_1: case MODE_1_2_2_2_1: case MODE_7_1_REAR_SURROUND: case MODE_7_1_FRONT_CENTER: case MODE_7_1_BACK: case MODE_7_1_TOP_FRONT: entryNo = 1; /* use stereo bandwidth settings */ break; default: return AAC_ENC_UNSUPPORTED_CHANNELCONFIG; } *bandWidth = GetBandwidthEntry(frameLength, sampleRate, chanBitRate, entryNo); if (*bandWidth == -1) { switch (frameLength) { case 120: case 128: case 240: case 256: *bandWidth = 16000; break; default: ErrorStatus = AAC_ENC_INVALID_CHANNEL_BITRATE; } } } break; default: *bandWidth = 0; return AAC_ENC_UNSUPPORTED_BITRATE_MODE; } *bandWidth = fMin(*bandWidth, sampleRate / 2); return ErrorStatus; } fdk-aac-2.0.2/libAACenc/src/bandwidth.h000066400000000000000000000121451404225310700174470ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Schug / A. Groeschel Description: bandwidth expert *******************************************************************************/ #ifndef BANDWIDTH_H #define BANDWIDTH_H #include "qc_data.h" AAC_ENCODER_ERROR FDKaacEnc_DetermineBandWidth( const INT proposedBandWidth, const INT bitrate, const AACENC_BITRATE_MODE bitrateMode, const INT sampleRate, const INT frameLength, const CHANNEL_MAPPING *const cm, const CHANNEL_MODE encoderMode, INT *const bandWidth); #endif /* BANDWIDTH_H */ fdk-aac-2.0.2/libAACenc/src/bit_cnt.cpp000066400000000000000000000722651404225310700174710ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M.Werner Description: Huffman Bitcounter & coder *******************************************************************************/ #include "bit_cnt.h" #include "aacEnc_ram.h" #define HI_LTAB(a) (a >> 16) #define LO_LTAB(a) (a & 0xffff) /***************************************************************************** functionname: FDKaacEnc_count1_2_3_4_5_6_7_8_9_10_11 description: counts tables 1-11 returns: input: quantized spectrum output: bitCount for tables 1-11 *****************************************************************************/ static void FDKaacEnc_count1_2_3_4_5_6_7_8_9_10_11(const SHORT *const values, const INT width, INT *RESTRICT bitCount) { INT i; INT bc1_2, bc3_4, bc5_6, bc7_8, bc9_10, bc11, sc; INT t0, t1, t2, t3; bc1_2 = 0; bc3_4 = 0; bc5_6 = 0; bc7_8 = 0; bc9_10 = 0; bc11 = 0; sc = 0; DWORD_ALIGNED(values); for (i = 0; i < width; i += 4) { t0 = values[i + 0]; t1 = values[i + 1]; t2 = values[i + 2]; t3 = values[i + 3]; bc1_2 += (INT)FDKaacEnc_huff_ltab1_2[t0 + 1][t1 + 1][t2 + 1][t3 + 1]; bc5_6 += (INT)FDKaacEnc_huff_ltab5_6[t0 + 4][t1 + 4] + (INT)FDKaacEnc_huff_ltab5_6[t2 + 4][t3 + 4]; t0 = fixp_abs(t0); sc += (t0 > 0); t1 = fixp_abs(t1); sc += (t1 > 0); t2 = fixp_abs(t2); sc += (t2 > 0); t3 = fixp_abs(t3); sc += (t3 > 0); bc3_4 += (INT)FDKaacEnc_huff_ltab3_4[t0][t1][t2][t3]; bc7_8 += (INT)FDKaacEnc_huff_ltab7_8[t0][t1] + (INT)FDKaacEnc_huff_ltab7_8[t2][t3]; bc9_10 += (INT)FDKaacEnc_huff_ltab9_10[t0][t1] + (INT)FDKaacEnc_huff_ltab9_10[t2][t3]; bc11 += (INT)FDKaacEnc_huff_ltab11[t0][t1] + (INT)FDKaacEnc_huff_ltab11[t2][t3]; } bitCount[1] = HI_LTAB(bc1_2); bitCount[2] = LO_LTAB(bc1_2); bitCount[3] = HI_LTAB(bc3_4) + sc; bitCount[4] = LO_LTAB(bc3_4) + sc; bitCount[5] = HI_LTAB(bc5_6); bitCount[6] = LO_LTAB(bc5_6); bitCount[7] = HI_LTAB(bc7_8) + sc; bitCount[8] = LO_LTAB(bc7_8) + sc; bitCount[9] = HI_LTAB(bc9_10) + sc; bitCount[10] = LO_LTAB(bc9_10) + sc; bitCount[11] = bc11 + sc; } /***************************************************************************** functionname: FDKaacEnc_count3_4_5_6_7_8_9_10_11 description: counts tables 3-11 returns: input: quantized spectrum output: bitCount for tables 3-11 *****************************************************************************/ static void FDKaacEnc_count3_4_5_6_7_8_9_10_11(const SHORT *const values, const INT width, INT *RESTRICT bitCount) { INT i; INT bc3_4, bc5_6, bc7_8, bc9_10, bc11, sc; INT t0, t1, t2, t3; bc3_4 = 0; bc5_6 = 0; bc7_8 = 0; bc9_10 = 0; bc11 = 0; sc = 0; DWORD_ALIGNED(values); for (i = 0; i < width; i += 4) { t0 = values[i + 0]; t1 = values[i + 1]; t2 = values[i + 2]; t3 = values[i + 3]; bc5_6 += (INT)FDKaacEnc_huff_ltab5_6[t0 + 4][t1 + 4] + (INT)FDKaacEnc_huff_ltab5_6[t2 + 4][t3 + 4]; t0 = fixp_abs(t0); sc += (t0 > 0); t1 = fixp_abs(t1); sc += (t1 > 0); t2 = fixp_abs(t2); sc += (t2 > 0); t3 = fixp_abs(t3); sc += (t3 > 0); bc3_4 += (INT)FDKaacEnc_huff_ltab3_4[t0][t1][t2][t3]; bc7_8 += (INT)FDKaacEnc_huff_ltab7_8[t0][t1] + (INT)FDKaacEnc_huff_ltab7_8[t2][t3]; bc9_10 += (INT)FDKaacEnc_huff_ltab9_10[t0][t1] + (INT)FDKaacEnc_huff_ltab9_10[t2][t3]; bc11 += (INT)FDKaacEnc_huff_ltab11[t0][t1] + (INT)FDKaacEnc_huff_ltab11[t2][t3]; } bitCount[1] = INVALID_BITCOUNT; bitCount[2] = INVALID_BITCOUNT; bitCount[3] = HI_LTAB(bc3_4) + sc; bitCount[4] = LO_LTAB(bc3_4) + sc; bitCount[5] = HI_LTAB(bc5_6); bitCount[6] = LO_LTAB(bc5_6); bitCount[7] = HI_LTAB(bc7_8) + sc; bitCount[8] = LO_LTAB(bc7_8) + sc; bitCount[9] = HI_LTAB(bc9_10) + sc; bitCount[10] = LO_LTAB(bc9_10) + sc; bitCount[11] = bc11 + sc; } /***************************************************************************** functionname: FDKaacEnc_count5_6_7_8_9_10_11 description: counts tables 5-11 returns: input: quantized spectrum output: bitCount for tables 5-11 *****************************************************************************/ static void FDKaacEnc_count5_6_7_8_9_10_11(const SHORT *const values, const INT width, INT *RESTRICT bitCount) { INT i; INT bc5_6, bc7_8, bc9_10, bc11, sc; INT t0, t1, t2, t3; bc5_6 = 0; bc7_8 = 0; bc9_10 = 0; bc11 = 0; sc = 0; DWORD_ALIGNED(values); for (i = 0; i < width; i += 4) { t0 = values[i + 0]; t1 = values[i + 1]; t2 = values[i + 2]; t3 = values[i + 3]; bc5_6 += (INT)FDKaacEnc_huff_ltab5_6[t0 + 4][t1 + 4] + (INT)FDKaacEnc_huff_ltab5_6[t2 + 4][t3 + 4]; t0 = fixp_abs(t0); sc += (t0 > 0); t1 = fixp_abs(t1); sc += (t1 > 0); t2 = fixp_abs(t2); sc += (t2 > 0); t3 = fixp_abs(t3); sc += (t3 > 0); bc7_8 += (INT)FDKaacEnc_huff_ltab7_8[t0][t1] + (INT)FDKaacEnc_huff_ltab7_8[t2][t3]; bc9_10 += (INT)FDKaacEnc_huff_ltab9_10[t0][t1] + (INT)FDKaacEnc_huff_ltab9_10[t2][t3]; bc11 += (INT)FDKaacEnc_huff_ltab11[t0][t1] + (INT)FDKaacEnc_huff_ltab11[t2][t3]; } bitCount[1] = INVALID_BITCOUNT; bitCount[2] = INVALID_BITCOUNT; bitCount[3] = INVALID_BITCOUNT; bitCount[4] = INVALID_BITCOUNT; bitCount[5] = HI_LTAB(bc5_6); bitCount[6] = LO_LTAB(bc5_6); bitCount[7] = HI_LTAB(bc7_8) + sc; bitCount[8] = LO_LTAB(bc7_8) + sc; bitCount[9] = HI_LTAB(bc9_10) + sc; bitCount[10] = LO_LTAB(bc9_10) + sc; bitCount[11] = bc11 + sc; } /***************************************************************************** functionname: FDKaacEnc_count7_8_9_10_11 description: counts tables 7-11 returns: input: quantized spectrum output: bitCount for tables 7-11 *****************************************************************************/ static void FDKaacEnc_count7_8_9_10_11(const SHORT *const values, const INT width, INT *RESTRICT bitCount) { INT i; INT bc7_8, bc9_10, bc11, sc; INT t0, t1, t2, t3; bc7_8 = 0; bc9_10 = 0; bc11 = 0; sc = 0; DWORD_ALIGNED(values); for (i = 0; i < width; i += 4) { t0 = values[i + 0]; t1 = values[i + 1]; t2 = values[i + 2]; t3 = values[i + 3]; t0 = fixp_abs(t0); sc += (t0 > 0); t1 = fixp_abs(t1); sc += (t1 > 0); t2 = fixp_abs(t2); sc += (t2 > 0); t3 = fixp_abs(t3); sc += (t3 > 0); bc7_8 += (INT)FDKaacEnc_huff_ltab7_8[t0][t1] + (INT)FDKaacEnc_huff_ltab7_8[t2][t3]; bc9_10 += (INT)FDKaacEnc_huff_ltab9_10[t0][t1] + (INT)FDKaacEnc_huff_ltab9_10[t2][t3]; bc11 += (INT)FDKaacEnc_huff_ltab11[t0][t1] + (INT)FDKaacEnc_huff_ltab11[t2][t3]; } bitCount[1] = INVALID_BITCOUNT; bitCount[2] = INVALID_BITCOUNT; bitCount[3] = INVALID_BITCOUNT; bitCount[4] = INVALID_BITCOUNT; bitCount[5] = INVALID_BITCOUNT; bitCount[6] = INVALID_BITCOUNT; bitCount[7] = HI_LTAB(bc7_8) + sc; bitCount[8] = LO_LTAB(bc7_8) + sc; bitCount[9] = HI_LTAB(bc9_10) + sc; bitCount[10] = LO_LTAB(bc9_10) + sc; bitCount[11] = bc11 + sc; } /***************************************************************************** functionname: FDKaacEnc_count9_10_11 description: counts tables 9-11 returns: input: quantized spectrum output: bitCount for tables 9-11 *****************************************************************************/ static void FDKaacEnc_count9_10_11(const SHORT *const values, const INT width, INT *RESTRICT bitCount) { INT i; INT bc9_10, bc11, sc; INT t0, t1, t2, t3; bc9_10 = 0; bc11 = 0; sc = 0; DWORD_ALIGNED(values); for (i = 0; i < width; i += 4) { t0 = values[i + 0]; t1 = values[i + 1]; t2 = values[i + 2]; t3 = values[i + 3]; t0 = fixp_abs(t0); sc += (t0 > 0); t1 = fixp_abs(t1); sc += (t1 > 0); t2 = fixp_abs(t2); sc += (t2 > 0); t3 = fixp_abs(t3); sc += (t3 > 0); bc9_10 += (INT)FDKaacEnc_huff_ltab9_10[t0][t1] + (INT)FDKaacEnc_huff_ltab9_10[t2][t3]; bc11 += (INT)FDKaacEnc_huff_ltab11[t0][t1] + (INT)FDKaacEnc_huff_ltab11[t2][t3]; } bitCount[1] = INVALID_BITCOUNT; bitCount[2] = INVALID_BITCOUNT; bitCount[3] = INVALID_BITCOUNT; bitCount[4] = INVALID_BITCOUNT; bitCount[5] = INVALID_BITCOUNT; bitCount[6] = INVALID_BITCOUNT; bitCount[7] = INVALID_BITCOUNT; bitCount[8] = INVALID_BITCOUNT; bitCount[9] = HI_LTAB(bc9_10) + sc; bitCount[10] = LO_LTAB(bc9_10) + sc; bitCount[11] = bc11 + sc; } /***************************************************************************** functionname: FDKaacEnc_count11 description: counts table 11 returns: input: quantized spectrum output: bitCount for table 11 *****************************************************************************/ static void FDKaacEnc_count11(const SHORT *const values, const INT width, INT *RESTRICT bitCount) { INT i; INT bc11, sc; INT t0, t1, t2, t3; bc11 = 0; sc = 0; DWORD_ALIGNED(values); for (i = 0; i < width; i += 4) { t0 = values[i + 0]; t1 = values[i + 1]; t2 = values[i + 2]; t3 = values[i + 3]; t0 = fixp_abs(t0); sc += (t0 > 0); t1 = fixp_abs(t1); sc += (t1 > 0); t2 = fixp_abs(t2); sc += (t2 > 0); t3 = fixp_abs(t3); sc += (t3 > 0); bc11 += (INT)FDKaacEnc_huff_ltab11[t0][t1] + (INT)FDKaacEnc_huff_ltab11[t2][t3]; } bitCount[1] = INVALID_BITCOUNT; bitCount[2] = INVALID_BITCOUNT; bitCount[3] = INVALID_BITCOUNT; bitCount[4] = INVALID_BITCOUNT; bitCount[5] = INVALID_BITCOUNT; bitCount[6] = INVALID_BITCOUNT; bitCount[7] = INVALID_BITCOUNT; bitCount[8] = INVALID_BITCOUNT; bitCount[9] = INVALID_BITCOUNT; bitCount[10] = INVALID_BITCOUNT; bitCount[11] = bc11 + sc; } /***************************************************************************** functionname: FDKaacEnc_countEsc description: counts table 11 (with Esc) returns: input: quantized spectrum output: bitCount for tables 11 (with Esc) *****************************************************************************/ static void FDKaacEnc_countEsc(const SHORT *const values, const INT width, INT *RESTRICT bitCount) { INT i; INT bc11, ec, sc; INT t0, t1, t00, t01; bc11 = 0; sc = 0; ec = 0; for (i = 0; i < width; i += 2) { t0 = fixp_abs(values[i + 0]); t1 = fixp_abs(values[i + 1]); sc += (t0 > 0) + (t1 > 0); t00 = fixMin(t0, 16); t01 = fixMin(t1, 16); bc11 += (INT)FDKaacEnc_huff_ltab11[t00][t01]; if (t0 >= 16) { ec += 5; while ((t0 >>= 1) >= 16) ec += 2; } if (t1 >= 16) { ec += 5; while ((t1 >>= 1) >= 16) ec += 2; } } for (i = 0; i < 11; i++) bitCount[i] = INVALID_BITCOUNT; bitCount[11] = bc11 + sc + ec; } typedef void (*COUNT_FUNCTION)(const SHORT *const values, const INT width, INT *RESTRICT bitCount); static const COUNT_FUNCTION countFuncTable[CODE_BOOK_ESC_LAV + 1] = { FDKaacEnc_count1_2_3_4_5_6_7_8_9_10_11, /* 0 */ FDKaacEnc_count1_2_3_4_5_6_7_8_9_10_11, /* 1 */ FDKaacEnc_count3_4_5_6_7_8_9_10_11, /* 2 */ FDKaacEnc_count5_6_7_8_9_10_11, /* 3 */ FDKaacEnc_count5_6_7_8_9_10_11, /* 4 */ FDKaacEnc_count7_8_9_10_11, /* 5 */ FDKaacEnc_count7_8_9_10_11, /* 6 */ FDKaacEnc_count7_8_9_10_11, /* 7 */ FDKaacEnc_count9_10_11, /* 8 */ FDKaacEnc_count9_10_11, /* 9 */ FDKaacEnc_count9_10_11, /* 10 */ FDKaacEnc_count9_10_11, /* 11 */ FDKaacEnc_count9_10_11, /* 12 */ FDKaacEnc_count11, /* 13 */ FDKaacEnc_count11, /* 14 */ FDKaacEnc_count11, /* 15 */ FDKaacEnc_countEsc /* 16 */ }; INT FDKaacEnc_bitCount(const SHORT *const values, const INT width, const INT maxVal, INT *const RESTRICT bitCount) { /* check if we can use codebook 0 */ bitCount[0] = (maxVal == 0) ? 0 : INVALID_BITCOUNT; countFuncTable[fixMin(maxVal, (INT)CODE_BOOK_ESC_LAV)](values, width, bitCount); return (0); } /* count difference between actual and zeroed lines */ INT FDKaacEnc_countValues(SHORT *RESTRICT values, INT width, INT codeBook) { INT i, t0, t1, t2, t3; INT bitCnt = 0; switch (codeBook) { case CODE_BOOK_ZERO_NO: break; case CODE_BOOK_1_NO: for (i = 0; i < width; i += 4) { t0 = values[i + 0]; t1 = values[i + 1]; t2 = values[i + 2]; t3 = values[i + 3]; bitCnt += HI_LTAB(FDKaacEnc_huff_ltab1_2[t0 + 1][t1 + 1][t2 + 1][t3 + 1]); } break; case CODE_BOOK_2_NO: for (i = 0; i < width; i += 4) { t0 = values[i + 0]; t1 = values[i + 1]; t2 = values[i + 2]; t3 = values[i + 3]; bitCnt += LO_LTAB(FDKaacEnc_huff_ltab1_2[t0 + 1][t1 + 1][t2 + 1][t3 + 1]); } break; case CODE_BOOK_3_NO: for (i = 0; i < width; i += 4) { t0 = fixp_abs(values[i + 0]); bitCnt += (t0 > 0); t1 = fixp_abs(values[i + 1]); bitCnt += (t1 > 0); t2 = fixp_abs(values[i + 2]); bitCnt += (t2 > 0); t3 = fixp_abs(values[i + 3]); bitCnt += (t3 > 0); bitCnt += HI_LTAB(FDKaacEnc_huff_ltab3_4[t0][t1][t2][t3]); } break; case CODE_BOOK_4_NO: for (i = 0; i < width; i += 4) { t0 = fixp_abs(values[i + 0]); bitCnt += (t0 > 0); t1 = fixp_abs(values[i + 1]); bitCnt += (t1 > 0); t2 = fixp_abs(values[i + 2]); bitCnt += (t2 > 0); t3 = fixp_abs(values[i + 3]); bitCnt += (t3 > 0); bitCnt += LO_LTAB(FDKaacEnc_huff_ltab3_4[t0][t1][t2][t3]); } break; case CODE_BOOK_5_NO: for (i = 0; i < width; i += 4) { t0 = values[i + 0]; t1 = values[i + 1]; t2 = values[i + 2]; t3 = values[i + 3]; bitCnt += HI_LTAB(FDKaacEnc_huff_ltab5_6[t0 + 4][t1 + 4]) + HI_LTAB(FDKaacEnc_huff_ltab5_6[t2 + 4][t3 + 4]); } break; case CODE_BOOK_6_NO: for (i = 0; i < width; i += 4) { t0 = values[i + 0]; t1 = values[i + 1]; t2 = values[i + 2]; t3 = values[i + 3]; bitCnt += LO_LTAB(FDKaacEnc_huff_ltab5_6[t0 + 4][t1 + 4]) + LO_LTAB(FDKaacEnc_huff_ltab5_6[t2 + 4][t3 + 4]); } break; case CODE_BOOK_7_NO: for (i = 0; i < width; i += 4) { t0 = fixp_abs(values[i + 0]); bitCnt += (t0 > 0); t1 = fixp_abs(values[i + 1]); bitCnt += (t1 > 0); t2 = fixp_abs(values[i + 2]); bitCnt += (t2 > 0); t3 = fixp_abs(values[i + 3]); bitCnt += (t3 > 0); bitCnt += HI_LTAB(FDKaacEnc_huff_ltab7_8[t0][t1]) + HI_LTAB(FDKaacEnc_huff_ltab7_8[t2][t3]); } break; case CODE_BOOK_8_NO: for (i = 0; i < width; i += 4) { t0 = fixp_abs(values[i + 0]); bitCnt += (t0 > 0); t1 = fixp_abs(values[i + 1]); bitCnt += (t1 > 0); t2 = fixp_abs(values[i + 2]); bitCnt += (t2 > 0); t3 = fixp_abs(values[i + 3]); bitCnt += (t3 > 0); bitCnt += LO_LTAB(FDKaacEnc_huff_ltab7_8[t0][t1]) + LO_LTAB(FDKaacEnc_huff_ltab7_8[t2][t3]); } break; case CODE_BOOK_9_NO: for (i = 0; i < width; i += 4) { t0 = fixp_abs(values[i + 0]); bitCnt += (t0 > 0); t1 = fixp_abs(values[i + 1]); bitCnt += (t1 > 0); t2 = fixp_abs(values[i + 2]); bitCnt += (t2 > 0); t3 = fixp_abs(values[i + 3]); bitCnt += (t3 > 0); bitCnt += HI_LTAB(FDKaacEnc_huff_ltab9_10[t0][t1]) + HI_LTAB(FDKaacEnc_huff_ltab9_10[t2][t3]); } break; case CODE_BOOK_10_NO: for (i = 0; i < width; i += 4) { t0 = fixp_abs(values[i + 0]); bitCnt += (t0 > 0); t1 = fixp_abs(values[i + 1]); bitCnt += (t1 > 0); t2 = fixp_abs(values[i + 2]); bitCnt += (t2 > 0); t3 = fixp_abs(values[i + 3]); bitCnt += (t3 > 0); bitCnt += LO_LTAB(FDKaacEnc_huff_ltab9_10[t0][t1]) + LO_LTAB(FDKaacEnc_huff_ltab9_10[t2][t3]); } break; case CODE_BOOK_ESC_NO: for (i = 0; i < width; i += 2) { t0 = fixp_abs(values[i + 0]); bitCnt += (t0 > 0); t1 = fixp_abs(values[i + 1]); bitCnt += (t1 > 0); bitCnt += (INT)FDKaacEnc_huff_ltab11[fixMin(t0, 16)][fixMin(t1, 16)]; if (t0 >= 16) { bitCnt += 5; while ((t0 >>= 1) >= 16) bitCnt += 2; } if (t1 >= 16) { bitCnt += 5; while ((t1 >>= 1) >= 16) bitCnt += 2; } } break; default: break; } return (bitCnt); } INT FDKaacEnc_codeValues(SHORT *RESTRICT values, INT width, INT codeBook, HANDLE_FDK_BITSTREAM hBitstream) { INT i, t0, t1, t2, t3, t00, t01; INT codeWord, codeLength; INT sign, signLength; DWORD_ALIGNED(values); switch (codeBook) { case CODE_BOOK_ZERO_NO: break; case CODE_BOOK_1_NO: for (i = 0; i < width; i += 4) { t0 = values[i + 0] + 1; t1 = values[i + 1] + 1; t2 = values[i + 2] + 1; t3 = values[i + 3] + 1; codeWord = FDKaacEnc_huff_ctab1[t0][t1][t2][t3]; codeLength = HI_LTAB(FDKaacEnc_huff_ltab1_2[t0][t1][t2][t3]); FDKwriteBits(hBitstream, codeWord, codeLength); } break; case CODE_BOOK_2_NO: for (i = 0; i < width; i += 4) { t0 = values[i + 0] + 1; t1 = values[i + 1] + 1; t2 = values[i + 2] + 1; t3 = values[i + 3] + 1; codeWord = FDKaacEnc_huff_ctab2[t0][t1][t2][t3]; codeLength = LO_LTAB(FDKaacEnc_huff_ltab1_2[t0][t1][t2][t3]); FDKwriteBits(hBitstream, codeWord, codeLength); } break; case CODE_BOOK_3_NO: for (i = 0; i < (width >> 2); i++) { sign = 0; signLength = 0; int index[4]; for (int j = 0; j < 4; j++) { int ti = *values++; int zero = (ti == 0) ? 0 : 1; signLength += zero; sign = (sign << zero) + ((UINT)ti >> 31); index[j] = fixp_abs(ti); } codeWord = FDKaacEnc_huff_ctab3[index[0]][index[1]][index[2]][index[3]]; codeLength = HI_LTAB( FDKaacEnc_huff_ltab3_4[index[0]][index[1]][index[2]][index[3]]); FDKwriteBits(hBitstream, (codeWord << signLength) | sign, codeLength + signLength); } break; case CODE_BOOK_4_NO: for (i = 0; i < width; i += 4) { sign = 0; signLength = 0; int index[4]; for (int j = 0; j < 4; j++) { int ti = *values++; int zero = (ti == 0) ? 0 : 1; signLength += zero; sign = (sign << zero) + ((UINT)ti >> 31); index[j] = fixp_abs(ti); } codeWord = FDKaacEnc_huff_ctab4[index[0]][index[1]][index[2]][index[3]]; codeLength = LO_LTAB( FDKaacEnc_huff_ltab3_4[index[0]][index[1]][index[2]][index[3]]); FDKwriteBits(hBitstream, (codeWord << signLength) | sign, codeLength + signLength); } break; case CODE_BOOK_5_NO: for (i = 0; i < (width >> 2); i++) { t0 = *values++ + 4; t1 = *values++ + 4; t2 = *values++ + 4; t3 = *values++ + 4; codeWord = FDKaacEnc_huff_ctab5[t0][t1]; codeLength = HI_LTAB(FDKaacEnc_huff_ltab5_6[t2][t3]); /* length of 2nd cw */ codeWord = (codeWord << codeLength) + FDKaacEnc_huff_ctab5[t2][t3]; codeLength += HI_LTAB(FDKaacEnc_huff_ltab5_6[t0][t1]); FDKwriteBits(hBitstream, codeWord, codeLength); } break; case CODE_BOOK_6_NO: for (i = 0; i < (width >> 2); i++) { t0 = *values++ + 4; t1 = *values++ + 4; t2 = *values++ + 4; t3 = *values++ + 4; codeWord = FDKaacEnc_huff_ctab6[t0][t1]; codeLength = LO_LTAB(FDKaacEnc_huff_ltab5_6[t2][t3]); /* length of 2nd cw */ codeWord = (codeWord << codeLength) + FDKaacEnc_huff_ctab6[t2][t3]; codeLength += LO_LTAB(FDKaacEnc_huff_ltab5_6[t0][t1]); FDKwriteBits(hBitstream, codeWord, codeLength); } break; case CODE_BOOK_7_NO: for (i = 0; i < (width >> 1); i++) { t0 = *values++; sign = ((UINT)t0 >> 31); t0 = fixp_abs(t0); signLength = (t0 == 0) ? 0 : 1; t1 = *values++; INT zero = (t1 == 0) ? 0 : 1; signLength += zero; sign = (sign << zero) + ((UINT)t1 >> 31); t1 = fixp_abs(t1); codeWord = FDKaacEnc_huff_ctab7[t0][t1]; codeLength = HI_LTAB(FDKaacEnc_huff_ltab7_8[t0][t1]); FDKwriteBits(hBitstream, (codeWord << signLength) | sign, codeLength + signLength); } break; case CODE_BOOK_8_NO: for (i = 0; i < (width >> 1); i++) { t0 = *values++; sign = ((UINT)t0 >> 31); t0 = fixp_abs(t0); signLength = (t0 == 0) ? 0 : 1; t1 = *values++; INT zero = (t1 == 0) ? 0 : 1; signLength += zero; sign = (sign << zero) + ((UINT)t1 >> 31); t1 = fixp_abs(t1); codeWord = FDKaacEnc_huff_ctab8[t0][t1]; codeLength = LO_LTAB(FDKaacEnc_huff_ltab7_8[t0][t1]); FDKwriteBits(hBitstream, (codeWord << signLength) | sign, codeLength + signLength); } break; case CODE_BOOK_9_NO: for (i = 0; i < (width >> 1); i++) { t0 = *values++; sign = ((UINT)t0 >> 31); t0 = fixp_abs(t0); signLength = (t0 == 0) ? 0 : 1; t1 = *values++; INT zero = (t1 == 0) ? 0 : 1; signLength += zero; sign = (sign << zero) + ((UINT)t1 >> 31); t1 = fixp_abs(t1); codeWord = FDKaacEnc_huff_ctab9[t0][t1]; codeLength = HI_LTAB(FDKaacEnc_huff_ltab9_10[t0][t1]); FDKwriteBits(hBitstream, (codeWord << signLength) | sign, codeLength + signLength); } break; case CODE_BOOK_10_NO: for (i = 0; i < (width >> 1); i++) { t0 = *values++; sign = ((UINT)t0 >> 31); t0 = fixp_abs(t0); signLength = (t0 == 0) ? 0 : 1; t1 = *values++; INT zero = (t1 == 0) ? 0 : 1; signLength += zero; sign = (sign << zero) + ((UINT)t1 >> 31); t1 = fixp_abs(t1); codeWord = FDKaacEnc_huff_ctab10[t0][t1]; codeLength = LO_LTAB(FDKaacEnc_huff_ltab9_10[t0][t1]); FDKwriteBits(hBitstream, (codeWord << signLength) | sign, codeLength + signLength); } break; case CODE_BOOK_ESC_NO: for (i = 0; i < (width >> 1); i++) { t0 = *values++; sign = ((UINT)t0 >> 31); t0 = fixp_abs(t0); signLength = (t0 == 0) ? 0 : 1; t1 = *values++; INT zero = (t1 == 0) ? 0 : 1; signLength += zero; sign = (sign << zero) + ((UINT)t1 >> 31); t1 = fixp_abs(t1); t00 = fixMin(t0, 16); t01 = fixMin(t1, 16); codeWord = FDKaacEnc_huff_ctab11[t00][t01]; codeLength = (INT)FDKaacEnc_huff_ltab11[t00][t01]; FDKwriteBits(hBitstream, (codeWord << signLength) | sign, codeLength + signLength); for (int j = 0; j < 2; j++) { if (t0 >= 16) { INT n = 4, p = t0; for (; (p >>= 1) >= 16;) n++; FDKwriteBits(hBitstream, (((1 << (n - 3)) - 2) << n) | (t0 - (1 << n)), n + n - 3); } t0 = t1; } } break; default: break; } return (0); } INT FDKaacEnc_codeScalefactorDelta(INT delta, HANDLE_FDK_BITSTREAM hBitstream) { INT codeWord, codeLength; if (fixp_abs(delta) > CODE_BOOK_SCF_LAV) return (1); codeWord = FDKaacEnc_huff_ctabscf[delta + CODE_BOOK_SCF_LAV]; codeLength = (INT)FDKaacEnc_huff_ltabscf[delta + CODE_BOOK_SCF_LAV]; FDKwriteBits(hBitstream, codeWord, codeLength); return (0); } fdk-aac-2.0.2/libAACenc/src/bit_cnt.h000066400000000000000000000156061404225310700171320ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M.Werner Description: Huffman Bitcounter & coder *******************************************************************************/ #ifndef BIT_CNT_H #define BIT_CNT_H #include "common_fix.h" #include "FDK_bitstream.h" #include "aacEnc_rom.h" #define INVALID_BITCOUNT (FDK_INT_MAX / 4) /* code book number table */ enum codeBookNo { CODE_BOOK_ZERO_NO = 0, CODE_BOOK_1_NO = 1, CODE_BOOK_2_NO = 2, CODE_BOOK_3_NO = 3, CODE_BOOK_4_NO = 4, CODE_BOOK_5_NO = 5, CODE_BOOK_6_NO = 6, CODE_BOOK_7_NO = 7, CODE_BOOK_8_NO = 8, CODE_BOOK_9_NO = 9, CODE_BOOK_10_NO = 10, CODE_BOOK_ESC_NO = 11, CODE_BOOK_RES_NO = 12, CODE_BOOK_PNS_NO = 13, CODE_BOOK_IS_OUT_OF_PHASE_NO = 14, CODE_BOOK_IS_IN_PHASE_NO = 15 }; /* code book index table */ enum codeBookNdx { CODE_BOOK_ZERO_NDX, CODE_BOOK_1_NDX, CODE_BOOK_2_NDX, CODE_BOOK_3_NDX, CODE_BOOK_4_NDX, CODE_BOOK_5_NDX, CODE_BOOK_6_NDX, CODE_BOOK_7_NDX, CODE_BOOK_8_NDX, CODE_BOOK_9_NDX, CODE_BOOK_10_NDX, CODE_BOOK_ESC_NDX, CODE_BOOK_RES_NDX, CODE_BOOK_PNS_NDX, CODE_BOOK_IS_OUT_OF_PHASE_NDX, CODE_BOOK_IS_IN_PHASE_NDX, NUMBER_OF_CODE_BOOKS }; /* code book lav table */ enum codeBookLav { CODE_BOOK_ZERO_LAV = 0, CODE_BOOK_1_LAV = 1, CODE_BOOK_2_LAV = 1, CODE_BOOK_3_LAV = 2, CODE_BOOK_4_LAV = 2, CODE_BOOK_5_LAV = 4, CODE_BOOK_6_LAV = 4, CODE_BOOK_7_LAV = 7, CODE_BOOK_8_LAV = 7, CODE_BOOK_9_LAV = 12, CODE_BOOK_10_LAV = 12, CODE_BOOK_ESC_LAV = 16, CODE_BOOK_SCF_LAV = 60, CODE_BOOK_PNS_LAV = 60 }; INT FDKaacEnc_bitCount(const SHORT *aQuantSpectrum, const INT noOfSpecLines, INT maxVal, INT *bitCountLut); INT FDKaacEnc_countValues(SHORT *values, INT width, INT codeBook); INT FDKaacEnc_codeValues(SHORT *values, INT width, INT codeBook, HANDLE_FDK_BITSTREAM hBitstream); INT FDKaacEnc_codeScalefactorDelta(INT scalefactor, HANDLE_FDK_BITSTREAM hBitstream); inline INT FDKaacEnc_bitCountScalefactorDelta(const INT delta) { FDK_ASSERT((0 <= (delta + CODE_BOOK_SCF_LAV)) && ((delta + CODE_BOOK_SCF_LAV) < (int)(sizeof(FDKaacEnc_huff_ltabscf) / sizeof((FDKaacEnc_huff_ltabscf[0]))))); return ((INT)FDKaacEnc_huff_ltabscf[delta + CODE_BOOK_SCF_LAV]); } #endif fdk-aac-2.0.2/libAACenc/src/bitenc.cpp000066400000000000000000001266001404225310700173040ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Werner Description: Bitstream encoder *******************************************************************************/ #include "bitenc.h" #include "bit_cnt.h" #include "dyn_bits.h" #include "qc_data.h" #include "interface.h" #include "aacEnc_ram.h" #include "tpenc_lib.h" #include "FDK_tools_rom.h" /* needed for the bitstream syntax tables */ static const int globalGainOffset = 100; static const int icsReservedBit = 0; static const int noiseOffset = 90; /***************************************************************************** functionname: FDKaacEnc_encodeSpectralData description: encode spectral data returns: the number of written bits input: output: *****************************************************************************/ static INT FDKaacEnc_encodeSpectralData(INT *sfbOffset, SECTION_DATA *sectionData, SHORT *quantSpectrum, HANDLE_FDK_BITSTREAM hBitStream) { INT i, sfb; INT dbgVal = FDKgetValidBits(hBitStream); for (i = 0; i < sectionData->noOfSections; i++) { if (sectionData->huffsection[i].codeBook != CODE_BOOK_PNS_NO) { /* huffencode spectral data for this huffsection */ INT tmp = sectionData->huffsection[i].sfbStart + sectionData->huffsection[i].sfbCnt; for (sfb = sectionData->huffsection[i].sfbStart; sfb < tmp; sfb++) { FDKaacEnc_codeValues(quantSpectrum + sfbOffset[sfb], sfbOffset[sfb + 1] - sfbOffset[sfb], sectionData->huffsection[i].codeBook, hBitStream); } } } return (FDKgetValidBits(hBitStream) - dbgVal); } /***************************************************************************** functionname:FDKaacEnc_encodeGlobalGain description: encodes Global Gain (common scale factor) returns: the number of static bits input: output: *****************************************************************************/ static INT FDKaacEnc_encodeGlobalGain(INT globalGain, INT scalefac, HANDLE_FDK_BITSTREAM hBitStream, INT mdctScale) { if (hBitStream != NULL) { FDKwriteBits(hBitStream, globalGain - scalefac + globalGainOffset - 4 * (LOG_NORM_PCM - mdctScale), 8); } return (8); } /***************************************************************************** functionname:FDKaacEnc_encodeIcsInfo description: encodes Ics Info returns: the number of static bits input: output: *****************************************************************************/ static INT FDKaacEnc_encodeIcsInfo(INT blockType, INT windowShape, INT groupingMask, INT maxSfbPerGroup, HANDLE_FDK_BITSTREAM hBitStream, UINT syntaxFlags) { INT statBits; if (blockType == SHORT_WINDOW) { statBits = 8 + TRANS_FAC - 1; } else { if (syntaxFlags & AC_ELD) { statBits = 6; } else { statBits = (!(syntaxFlags & AC_SCALABLE)) ? 11 : 10; } } if (hBitStream != NULL) { if (!(syntaxFlags & AC_ELD)) { FDKwriteBits(hBitStream, icsReservedBit, 1); FDKwriteBits(hBitStream, blockType, 2); FDKwriteBits(hBitStream, (windowShape == LOL_WINDOW) ? KBD_WINDOW : windowShape, 1); } switch (blockType) { case LONG_WINDOW: case START_WINDOW: case STOP_WINDOW: FDKwriteBits(hBitStream, maxSfbPerGroup, 6); if (!(syntaxFlags & (AC_SCALABLE | AC_ELD))) { /* If not scalable syntax then ... */ /* No predictor data present */ FDKwriteBits(hBitStream, 0, 1); } break; case SHORT_WINDOW: FDKwriteBits(hBitStream, maxSfbPerGroup, 4); /* Write grouping bits */ FDKwriteBits(hBitStream, groupingMask, TRANS_FAC - 1); break; } } return (statBits); } /***************************************************************************** functionname: FDKaacEnc_encodeSectionData description: encode section data (common Huffman codebooks for adjacent SFB's) returns: none input: output: *****************************************************************************/ static INT FDKaacEnc_encodeSectionData(SECTION_DATA *sectionData, HANDLE_FDK_BITSTREAM hBitStream, UINT useVCB11) { if (hBitStream != NULL) { INT sectEscapeVal = 0, sectLenBits = 0; INT sectLen; INT i; INT dbgVal = FDKgetValidBits(hBitStream); INT sectCbBits = 4; switch (sectionData->blockType) { case LONG_WINDOW: case START_WINDOW: case STOP_WINDOW: sectEscapeVal = SECT_ESC_VAL_LONG; sectLenBits = SECT_BITS_LONG; break; case SHORT_WINDOW: sectEscapeVal = SECT_ESC_VAL_SHORT; sectLenBits = SECT_BITS_SHORT; break; } for (i = 0; i < sectionData->noOfSections; i++) { INT codeBook = sectionData->huffsection[i].codeBook; FDKwriteBits(hBitStream, codeBook, sectCbBits); { sectLen = sectionData->huffsection[i].sfbCnt; while (sectLen >= sectEscapeVal) { FDKwriteBits(hBitStream, sectEscapeVal, sectLenBits); sectLen -= sectEscapeVal; } FDKwriteBits(hBitStream, sectLen, sectLenBits); } } return (FDKgetValidBits(hBitStream) - dbgVal); } return (0); } /***************************************************************************** functionname: FDKaacEnc_encodeScaleFactorData description: encode DPCM coded scale factors returns: none input: output: *****************************************************************************/ static INT FDKaacEnc_encodeScaleFactorData(UINT *maxValueInSfb, SECTION_DATA *sectionData, INT *scalefac, HANDLE_FDK_BITSTREAM hBitStream, INT *RESTRICT noiseNrg, const INT *isScale, INT globalGain) { if (hBitStream != NULL) { INT i, j, lastValScf, deltaScf; INT deltaPns; INT lastValPns = 0; INT noisePCMFlag = TRUE; INT lastValIs; INT dbgVal = FDKgetValidBits(hBitStream); lastValScf = scalefac[sectionData->firstScf]; lastValPns = globalGain - scalefac[sectionData->firstScf] + globalGainOffset - 4 * LOG_NORM_PCM - noiseOffset; lastValIs = 0; for (i = 0; i < sectionData->noOfSections; i++) { if (sectionData->huffsection[i].codeBook != CODE_BOOK_ZERO_NO) { if ((sectionData->huffsection[i].codeBook == CODE_BOOK_IS_OUT_OF_PHASE_NO) || (sectionData->huffsection[i].codeBook == CODE_BOOK_IS_IN_PHASE_NO)) { INT sfbStart = sectionData->huffsection[i].sfbStart; INT tmp = sfbStart + sectionData->huffsection[i].sfbCnt; for (j = sfbStart; j < tmp; j++) { INT deltaIs = isScale[j] - lastValIs; lastValIs = isScale[j]; if (FDKaacEnc_codeScalefactorDelta(deltaIs, hBitStream)) { return (1); } } /* sfb */ } else if (sectionData->huffsection[i].codeBook == CODE_BOOK_PNS_NO) { INT sfbStart = sectionData->huffsection[i].sfbStart; INT tmp = sfbStart + sectionData->huffsection[i].sfbCnt; for (j = sfbStart; j < tmp; j++) { deltaPns = noiseNrg[j] - lastValPns; lastValPns = noiseNrg[j]; if (noisePCMFlag) { FDKwriteBits(hBitStream, deltaPns + (1 << (PNS_PCM_BITS - 1)), PNS_PCM_BITS); noisePCMFlag = FALSE; } else { if (FDKaacEnc_codeScalefactorDelta(deltaPns, hBitStream)) { return (1); } } } /* sfb */ } else { INT tmp = sectionData->huffsection[i].sfbStart + sectionData->huffsection[i].sfbCnt; for (j = sectionData->huffsection[i].sfbStart; j < tmp; j++) { /* check if we can repeat the last value to save bits */ if (maxValueInSfb[j] == 0) deltaScf = 0; else { deltaScf = -(scalefac[j] - lastValScf); lastValScf = scalefac[j]; } if (FDKaacEnc_codeScalefactorDelta(deltaScf, hBitStream)) { return (1); } } /* sfb */ } /* code scalefactor */ } /* sectionData->huffsection[i].codeBook != CODE_BOOK_ZERO_NO */ } /* section loop */ return (FDKgetValidBits(hBitStream) - dbgVal); } /* if (hBitStream != NULL) */ return (0); } /***************************************************************************** functionname:encodeMsInfo description: encodes MS-Stereo Info returns: the number of static bits input: output: *****************************************************************************/ static INT FDKaacEnc_encodeMSInfo(INT sfbCnt, INT grpSfb, INT maxSfb, INT msDigest, INT *jsFlags, HANDLE_FDK_BITSTREAM hBitStream) { INT sfb, sfbOff, msBits = 0; if (hBitStream != NULL) { switch (msDigest) { case MS_NONE: FDKwriteBits(hBitStream, SI_MS_MASK_NONE, 2); msBits += 2; break; case MS_ALL: FDKwriteBits(hBitStream, SI_MS_MASK_ALL, 2); msBits += 2; break; case MS_SOME: FDKwriteBits(hBitStream, SI_MS_MASK_SOME, 2); msBits += 2; for (sfbOff = 0; sfbOff < sfbCnt; sfbOff += grpSfb) { for (sfb = 0; sfb < maxSfb; sfb++) { if (jsFlags[sfbOff + sfb] & MS_ON) { FDKwriteBits(hBitStream, 1, 1); } else { FDKwriteBits(hBitStream, 0, 1); } msBits += 1; } } break; } } else { msBits += 2; if (msDigest == MS_SOME) { for (sfbOff = 0; sfbOff < sfbCnt; sfbOff += grpSfb) { for (sfb = 0; sfb < maxSfb; sfb++) { msBits += 1; } } } } return (msBits); } /***************************************************************************** functionname: FDKaacEnc_encodeTnsDataPresent description: encode TNS data (filter order, coeffs, ..) returns: the number of static bits input: output: *****************************************************************************/ static INT FDKaacEnc_encodeTnsDataPresent(TNS_INFO *tnsInfo, INT blockType, HANDLE_FDK_BITSTREAM hBitStream) { if ((hBitStream != NULL) && (tnsInfo != NULL)) { INT i, tnsPresent = 0; INT numOfWindows = (blockType == SHORT_WINDOW ? TRANS_FAC : 1); for (i = 0; i < numOfWindows; i++) { if (tnsInfo->numOfFilters[i] != 0) { tnsPresent = 1; break; } } if (tnsPresent == 0) { FDKwriteBits(hBitStream, 0, 1); } else { FDKwriteBits(hBitStream, 1, 1); } } return (1); } /***************************************************************************** functionname: FDKaacEnc_encodeTnsData description: encode TNS data (filter order, coeffs, ..) returns: the number of static bits input: output: *****************************************************************************/ static INT FDKaacEnc_encodeTnsData(TNS_INFO *tnsInfo, INT blockType, HANDLE_FDK_BITSTREAM hBitStream) { INT tnsBits = 0; if (tnsInfo != NULL) { INT i, j, k; INT tnsPresent = 0; INT coefBits; INT numOfWindows = (blockType == SHORT_WINDOW ? TRANS_FAC : 1); for (i = 0; i < numOfWindows; i++) { if (tnsInfo->numOfFilters[i] != 0) { tnsPresent = 1; } } if (hBitStream != NULL) { if (tnsPresent == 1) { /* there is data to be written*/ for (i = 0; i < numOfWindows; i++) { FDKwriteBits(hBitStream, tnsInfo->numOfFilters[i], (blockType == SHORT_WINDOW ? 1 : 2)); tnsBits += (blockType == SHORT_WINDOW ? 1 : 2); if (tnsInfo->numOfFilters[i]) { FDKwriteBits(hBitStream, (tnsInfo->coefRes[i] == 4 ? 1 : 0), 1); tnsBits += 1; } for (j = 0; j < tnsInfo->numOfFilters[i]; j++) { FDKwriteBits(hBitStream, tnsInfo->length[i][j], (blockType == SHORT_WINDOW ? 4 : 6)); tnsBits += (blockType == SHORT_WINDOW ? 4 : 6); FDK_ASSERT(tnsInfo->order[i][j] <= 12); FDKwriteBits(hBitStream, tnsInfo->order[i][j], (blockType == SHORT_WINDOW ? 3 : 5)); tnsBits += (blockType == SHORT_WINDOW ? 3 : 5); if (tnsInfo->order[i][j]) { FDKwriteBits(hBitStream, tnsInfo->direction[i][j], 1); tnsBits += 1; /*direction*/ if (tnsInfo->coefRes[i] == 4) { coefBits = 3; for (k = 0; k < tnsInfo->order[i][j]; k++) { if (tnsInfo->coef[i][j][k] > 3 || tnsInfo->coef[i][j][k] < -4) { coefBits = 4; break; } } } else { coefBits = 2; for (k = 0; k < tnsInfo->order[i][j]; k++) { if (tnsInfo->coef[i][j][k] > 1 || tnsInfo->coef[i][j][k] < -2) { coefBits = 3; break; } } } FDKwriteBits(hBitStream, -(coefBits - tnsInfo->coefRes[i]), 1); /*coef_compres*/ tnsBits += 1; /*coef_compression */ for (k = 0; k < tnsInfo->order[i][j]; k++) { static const INT rmask[] = {0, 1, 3, 7, 15}; FDKwriteBits(hBitStream, tnsInfo->coef[i][j][k] & rmask[coefBits], coefBits); tnsBits += coefBits; } } } } } } else { if (tnsPresent != 0) { for (i = 0; i < numOfWindows; i++) { tnsBits += (blockType == SHORT_WINDOW ? 1 : 2); if (tnsInfo->numOfFilters[i]) { tnsBits += 1; for (j = 0; j < tnsInfo->numOfFilters[i]; j++) { tnsBits += (blockType == SHORT_WINDOW ? 4 : 6); tnsBits += (blockType == SHORT_WINDOW ? 3 : 5); if (tnsInfo->order[i][j]) { tnsBits += 1; /*direction*/ tnsBits += 1; /*coef_compression */ if (tnsInfo->coefRes[i] == 4) { coefBits = 3; for (k = 0; k < tnsInfo->order[i][j]; k++) { if (tnsInfo->coef[i][j][k] > 3 || tnsInfo->coef[i][j][k] < -4) { coefBits = 4; break; } } } else { coefBits = 2; for (k = 0; k < tnsInfo->order[i][j]; k++) { if (tnsInfo->coef[i][j][k] > 1 || tnsInfo->coef[i][j][k] < -2) { coefBits = 3; break; } } } for (k = 0; k < tnsInfo->order[i][j]; k++) { tnsBits += coefBits; } } } } } } } } /* (tnsInfo!=NULL) */ return (tnsBits); } /***************************************************************************** functionname: FDKaacEnc_encodeGainControlData description: unsupported returns: none input: output: *****************************************************************************/ static INT FDKaacEnc_encodeGainControlData(HANDLE_FDK_BITSTREAM hBitStream) { if (hBitStream != NULL) { FDKwriteBits(hBitStream, 0, 1); } return (1); } /***************************************************************************** functionname: FDKaacEnc_encodePulseData description: not supported yet (dummy) returns: none input: output: *****************************************************************************/ static INT FDKaacEnc_encodePulseData(HANDLE_FDK_BITSTREAM hBitStream) { if (hBitStream != NULL) { FDKwriteBits(hBitStream, 0, 1); } return (1); } /***************************************************************************** functionname: FDKaacEnc_writeExtensionPayload description: write extension payload to bitstream returns: number of written bits input: output: *****************************************************************************/ static INT FDKaacEnc_writeExtensionPayload(HANDLE_FDK_BITSTREAM hBitStream, EXT_PAYLOAD_TYPE extPayloadType, const UCHAR *extPayloadData, INT extPayloadBits) { #define EXT_TYPE_BITS (4) #define DATA_EL_VERSION_BITS (4) #define FILL_NIBBLE_BITS (4) INT extBitsUsed = 0; if (extPayloadBits >= EXT_TYPE_BITS) { UCHAR fillByte = 0x00; /* for EXT_FIL and EXT_FILL_DATA */ if (hBitStream != NULL) { FDKwriteBits(hBitStream, extPayloadType, EXT_TYPE_BITS); } extBitsUsed += EXT_TYPE_BITS; switch (extPayloadType) { /* case EXT_SAC_DATA: */ case EXT_LDSAC_DATA: if (hBitStream != NULL) { FDKwriteBits(hBitStream, *extPayloadData++, 4); /* nibble */ } extBitsUsed += 4; FDK_FALLTHROUGH; case EXT_DYNAMIC_RANGE: case EXT_SBR_DATA: case EXT_SBR_DATA_CRC: if (hBitStream != NULL) { int i, writeBits = extPayloadBits; for (i = 0; writeBits >= 8; i++) { FDKwriteBits(hBitStream, *extPayloadData++, 8); writeBits -= 8; } if (writeBits > 0) { FDKwriteBits(hBitStream, (*extPayloadData) >> (8 - writeBits), writeBits); } } extBitsUsed += extPayloadBits; break; case EXT_DATA_ELEMENT: { INT dataElementLength = (extPayloadBits + 7) >> 3; INT cnt = dataElementLength; int loopCounter = 1; while (dataElementLength >= 255) { loopCounter++; dataElementLength -= 255; } if (hBitStream != NULL) { int i; FDKwriteBits( hBitStream, 0x00, DATA_EL_VERSION_BITS); /* data_element_version = ANC_DATA */ for (i = 1; i < loopCounter; i++) { FDKwriteBits(hBitStream, 255, 8); } FDKwriteBits(hBitStream, dataElementLength, 8); for (i = 0; i < cnt; i++) { FDKwriteBits(hBitStream, extPayloadData[i], 8); } } extBitsUsed += DATA_EL_VERSION_BITS + (loopCounter * 8) + (cnt * 8); } break; case EXT_FILL_DATA: fillByte = 0xA5; FDK_FALLTHROUGH; case EXT_FIL: default: if (hBitStream != NULL) { int writeBits = extPayloadBits; FDKwriteBits(hBitStream, 0x00, FILL_NIBBLE_BITS); writeBits -= 8; /* acount for the extension type and the fill nibble */ while (writeBits >= 8) { FDKwriteBits(hBitStream, fillByte, 8); writeBits -= 8; } } extBitsUsed += FILL_NIBBLE_BITS + (extPayloadBits & ~0x7) - 8; break; } } return (extBitsUsed); } /***************************************************************************** functionname: FDKaacEnc_writeDataStreamElement description: write data stream elements like ancillary data ... returns: the amount of used bits input: output: ******************************************************************************/ static INT FDKaacEnc_writeDataStreamElement(HANDLE_TRANSPORTENC hTpEnc, INT elementInstanceTag, INT dataPayloadBytes, UCHAR *dataBuffer, UINT alignAnchor) { #define DATA_BYTE_ALIGN_FLAG (0) #define EL_INSTANCE_TAG_BITS (4) #define DATA_BYTE_ALIGN_FLAG_BITS (1) #define DATA_LEN_COUNT_BITS (8) #define DATA_LEN_ESC_COUNT_BITS (8) #define MAX_DATA_ALIGN_BITS (7) #define MAX_DSE_DATA_BYTES (510) INT dseBitsUsed = 0; while (dataPayloadBytes > 0) { int esc_count = -1; int cnt = 0; INT crcReg = -1; dseBitsUsed += EL_ID_BITS + EL_INSTANCE_TAG_BITS + DATA_BYTE_ALIGN_FLAG_BITS + DATA_LEN_COUNT_BITS; if (DATA_BYTE_ALIGN_FLAG) { dseBitsUsed += MAX_DATA_ALIGN_BITS; } cnt = fixMin(MAX_DSE_DATA_BYTES, dataPayloadBytes); if (cnt >= 255) { esc_count = cnt - 255; dseBitsUsed += DATA_LEN_ESC_COUNT_BITS; } dataPayloadBytes -= cnt; dseBitsUsed += cnt * 8; if (hTpEnc != NULL) { HANDLE_FDK_BITSTREAM hBitStream = transportEnc_GetBitstream(hTpEnc); int i; FDKwriteBits(hBitStream, ID_DSE, EL_ID_BITS); crcReg = transportEnc_CrcStartReg(hTpEnc, 0); FDKwriteBits(hBitStream, elementInstanceTag, EL_INSTANCE_TAG_BITS); FDKwriteBits(hBitStream, DATA_BYTE_ALIGN_FLAG, DATA_BYTE_ALIGN_FLAG_BITS); /* write length field(s) */ if (esc_count >= 0) { FDKwriteBits(hBitStream, 255, DATA_LEN_COUNT_BITS); FDKwriteBits(hBitStream, esc_count, DATA_LEN_ESC_COUNT_BITS); } else { FDKwriteBits(hBitStream, cnt, DATA_LEN_COUNT_BITS); } if (DATA_BYTE_ALIGN_FLAG) { INT tmp = (INT)FDKgetValidBits(hBitStream); FDKbyteAlign(hBitStream, alignAnchor); /* count actual bits */ dseBitsUsed += (INT)FDKgetValidBits(hBitStream) - tmp - MAX_DATA_ALIGN_BITS; } /* write payload */ for (i = 0; i < cnt; i++) { FDKwriteBits(hBitStream, dataBuffer[i], 8); } transportEnc_CrcEndReg(hTpEnc, crcReg); } } return (dseBitsUsed); } /***************************************************************************** functionname: FDKaacEnc_writeExtensionData description: write extension payload to bitstream returns: number of written bits input: output: *****************************************************************************/ INT FDKaacEnc_writeExtensionData(HANDLE_TRANSPORTENC hTpEnc, QC_OUT_EXTENSION *pExtension, INT elInstanceTag, /* for DSE only */ UINT alignAnchor, /* for DSE only */ UINT syntaxFlags, AUDIO_OBJECT_TYPE aot, SCHAR epConfig) { #define FILL_EL_COUNT_BITS (4) #define FILL_EL_ESC_COUNT_BITS (8) #define MAX_FILL_DATA_BYTES (269) HANDLE_FDK_BITSTREAM hBitStream = NULL; INT payloadBits = pExtension->nPayloadBits; INT extBitsUsed = 0; if (hTpEnc != NULL) { hBitStream = transportEnc_GetBitstream(hTpEnc); } if (syntaxFlags & (AC_SCALABLE | AC_ER)) { { if ((syntaxFlags & AC_ELD) && ((pExtension->type == EXT_SBR_DATA) || (pExtension->type == EXT_SBR_DATA_CRC))) { if (hBitStream != NULL) { int i, writeBits = payloadBits; UCHAR *extPayloadData = pExtension->pPayload; for (i = 0; writeBits >= 8; i++) { FDKwriteBits(hBitStream, extPayloadData[i], 8); writeBits -= 8; } if (writeBits > 0) { FDKwriteBits(hBitStream, extPayloadData[i] >> (8 - writeBits), writeBits); } } extBitsUsed += payloadBits; } else { /* ER or scalable syntax -> write extension en bloc */ extBitsUsed += FDKaacEnc_writeExtensionPayload( hBitStream, pExtension->type, pExtension->pPayload, payloadBits); } } } else { /* We have normal GA bitstream payload (AOT 2,5,29) so pack the data into a fill elements or DSEs */ if (pExtension->type == EXT_DATA_ELEMENT) { extBitsUsed += FDKaacEnc_writeDataStreamElement( hTpEnc, elInstanceTag, pExtension->nPayloadBits >> 3, pExtension->pPayload, alignAnchor); } else { while (payloadBits >= (EL_ID_BITS + FILL_EL_COUNT_BITS)) { INT cnt, esc_count = -1, alignBits = 7; if ((pExtension->type == EXT_FILL_DATA) || (pExtension->type == EXT_FIL)) { payloadBits -= EL_ID_BITS + FILL_EL_COUNT_BITS; if (payloadBits >= 15 * 8) { payloadBits -= FILL_EL_ESC_COUNT_BITS; esc_count = 0; /* write esc_count even if cnt becomes smaller 15 */ } alignBits = 0; } cnt = fixMin(MAX_FILL_DATA_BYTES, (payloadBits + alignBits) >> 3); if (cnt >= 15) { esc_count = cnt - 15 + 1; } if (hBitStream != NULL) { /* write bitstream */ FDKwriteBits(hBitStream, ID_FIL, EL_ID_BITS); if (esc_count >= 0) { FDKwriteBits(hBitStream, 15, FILL_EL_COUNT_BITS); FDKwriteBits(hBitStream, esc_count, FILL_EL_ESC_COUNT_BITS); } else { FDKwriteBits(hBitStream, cnt, FILL_EL_COUNT_BITS); } } extBitsUsed += EL_ID_BITS + FILL_EL_COUNT_BITS + ((esc_count >= 0) ? FILL_EL_ESC_COUNT_BITS : 0); cnt = fixMin(cnt * 8, payloadBits); /* convert back to bits */ extBitsUsed += FDKaacEnc_writeExtensionPayload( hBitStream, pExtension->type, pExtension->pPayload, cnt); payloadBits -= cnt; } } } return (extBitsUsed); } /***************************************************************************** functionname: FDKaacEnc_ByteAlignment description: returns: input: output: *****************************************************************************/ static void FDKaacEnc_ByteAlignment(HANDLE_FDK_BITSTREAM hBitStream, int alignBits) { FDKwriteBits(hBitStream, 0, alignBits); } AAC_ENCODER_ERROR FDKaacEnc_ChannelElementWrite( HANDLE_TRANSPORTENC hTpEnc, ELEMENT_INFO *pElInfo, QC_OUT_CHANNEL *qcOutChannel[(2)], PSY_OUT_ELEMENT *psyOutElement, PSY_OUT_CHANNEL *psyOutChannel[(2)], UINT syntaxFlags, AUDIO_OBJECT_TYPE aot, SCHAR epConfig, INT *pBitDemand, UCHAR minCnt) { AAC_ENCODER_ERROR error = AAC_ENC_OK; HANDLE_FDK_BITSTREAM hBitStream = NULL; INT bitDemand = 0; const element_list_t *list; int i, ch, decision_bit; INT crcReg1 = -1, crcReg2 = -1; UCHAR numberOfChannels; if (hTpEnc != NULL) { /* Get bitstream handle */ hBitStream = transportEnc_GetBitstream(hTpEnc); } if ((pElInfo->elType == ID_SCE) || (pElInfo->elType == ID_LFE)) { numberOfChannels = 1; } else { numberOfChannels = 2; } /* Get channel element sequence table */ list = getBitstreamElementList(aot, epConfig, numberOfChannels, 0, 0); if (list == NULL) { error = AAC_ENC_UNSUPPORTED_AOT; goto bail; } if (!(syntaxFlags & (AC_SCALABLE | AC_ER))) { if (hBitStream != NULL) { FDKwriteBits(hBitStream, pElInfo->elType, EL_ID_BITS); } bitDemand += EL_ID_BITS; } /* Iterate through sequence table */ i = 0; ch = 0; decision_bit = 0; do { /* some tmp values */ SECTION_DATA *pChSectionData = NULL; INT *pChScf = NULL; UINT *pChMaxValueInSfb = NULL; TNS_INFO *pTnsInfo = NULL; INT chGlobalGain = 0; INT chBlockType = 0; INT chMaxSfbPerGrp = 0; INT chSfbPerGrp = 0; INT chSfbCnt = 0; INT chFirstScf = 0; if (minCnt == 0) { if (qcOutChannel != NULL) { pChSectionData = &(qcOutChannel[ch]->sectionData); pChScf = qcOutChannel[ch]->scf; chGlobalGain = qcOutChannel[ch]->globalGain; pChMaxValueInSfb = qcOutChannel[ch]->maxValueInSfb; chBlockType = pChSectionData->blockType; chMaxSfbPerGrp = pChSectionData->maxSfbPerGroup; chSfbPerGrp = pChSectionData->sfbPerGroup; chSfbCnt = pChSectionData->sfbCnt; chFirstScf = pChScf[pChSectionData->firstScf]; } else { /* get values from PSY */ chSfbCnt = psyOutChannel[ch]->sfbCnt; chSfbPerGrp = psyOutChannel[ch]->sfbPerGroup; chMaxSfbPerGrp = psyOutChannel[ch]->maxSfbPerGroup; } pTnsInfo = &psyOutChannel[ch]->tnsInfo; } /* minCnt==0 */ if (qcOutChannel == NULL) { chBlockType = psyOutChannel[ch]->lastWindowSequence; } switch (list->id[i]) { case element_instance_tag: /* Write element instance tag */ if (hBitStream != NULL) { FDKwriteBits(hBitStream, pElInfo->instanceTag, 4); } bitDemand += 4; break; case common_window: /* Write common window flag */ decision_bit = psyOutElement->commonWindow; if (hBitStream != NULL) { FDKwriteBits(hBitStream, psyOutElement->commonWindow, 1); } bitDemand += 1; break; case ics_info: /* Write individual channel info */ bitDemand += FDKaacEnc_encodeIcsInfo(chBlockType, psyOutChannel[ch]->windowShape, psyOutChannel[ch]->groupingMask, chMaxSfbPerGrp, hBitStream, syntaxFlags); break; case ltp_data_present: /* Write LTP data present flag */ if (hBitStream != NULL) { FDKwriteBits(hBitStream, 0, 1); } bitDemand += 1; break; case ltp_data: /* Predictor data not supported. Nothing to do here. */ break; case ms: /* Write MS info */ bitDemand += FDKaacEnc_encodeMSInfo( chSfbCnt, chSfbPerGrp, chMaxSfbPerGrp, (minCnt == 0) ? psyOutElement->toolsInfo.msDigest : MS_NONE, psyOutElement->toolsInfo.msMask, hBitStream); break; case global_gain: bitDemand += FDKaacEnc_encodeGlobalGain( chGlobalGain, chFirstScf, hBitStream, psyOutChannel[ch]->mdctScale); break; case section_data: { INT siBits = FDKaacEnc_encodeSectionData( pChSectionData, hBitStream, (syntaxFlags & AC_ER_VCB11) ? 1 : 0); if (hBitStream != NULL) { if (siBits != qcOutChannel[ch]->sectionData.sideInfoBits) { error = AAC_ENC_WRITE_SEC_ERROR; } } bitDemand += siBits; } break; case scale_factor_data: { INT sfDataBits = FDKaacEnc_encodeScaleFactorData( pChMaxValueInSfb, pChSectionData, pChScf, hBitStream, psyOutChannel[ch]->noiseNrg, psyOutChannel[ch]->isScale, chGlobalGain); if ((hBitStream != NULL) && (sfDataBits != (qcOutChannel[ch]->sectionData.scalefacBits + qcOutChannel[ch]->sectionData.noiseNrgBits))) { error = AAC_ENC_WRITE_SCAL_ERROR; } bitDemand += sfDataBits; } break; case esc2_rvlc: if (syntaxFlags & AC_ER_RVLC) { /* write RVLC data into bitstream (error sens. cat. 2) */ error = AAC_ENC_UNSUPPORTED_AOT; } break; case pulse: /* Write pulse data */ bitDemand += FDKaacEnc_encodePulseData(hBitStream); break; case tns_data_present: /* Write TNS data present flag */ bitDemand += FDKaacEnc_encodeTnsDataPresent(pTnsInfo, chBlockType, hBitStream); break; case tns_data: /* Write TNS data */ bitDemand += FDKaacEnc_encodeTnsData(pTnsInfo, chBlockType, hBitStream); break; case gain_control_data: /* Nothing to do here */ break; case gain_control_data_present: bitDemand += FDKaacEnc_encodeGainControlData(hBitStream); break; case esc1_hcr: if (syntaxFlags & AC_ER_HCR) { error = AAC_ENC_UNKNOWN; } break; case spectral_data: if (hBitStream != NULL) { INT spectralBits = 0; spectralBits = FDKaacEnc_encodeSpectralData( psyOutChannel[ch]->sfbOffsets, pChSectionData, qcOutChannel[ch]->quantSpec, hBitStream); if (spectralBits != qcOutChannel[ch]->sectionData.huffmanBits) { return AAC_ENC_WRITE_SPEC_ERROR; } bitDemand += spectralBits; } break; /* Non data cases */ case adtscrc_start_reg1: if (hTpEnc != NULL) { crcReg1 = transportEnc_CrcStartReg(hTpEnc, 192); } break; case adtscrc_start_reg2: if (hTpEnc != NULL) { crcReg2 = transportEnc_CrcStartReg(hTpEnc, 128); } break; case adtscrc_end_reg1: case drmcrc_end_reg: if (hTpEnc != NULL) { transportEnc_CrcEndReg(hTpEnc, crcReg1); } break; case adtscrc_end_reg2: if (hTpEnc != NULL) { transportEnc_CrcEndReg(hTpEnc, crcReg2); } break; case drmcrc_start_reg: if (hTpEnc != NULL) { crcReg1 = transportEnc_CrcStartReg(hTpEnc, 0); } break; case next_channel: ch = (ch + 1) % numberOfChannels; break; case link_sequence: list = list->next[decision_bit]; i = -1; break; default: error = AAC_ENC_UNKNOWN; break; } if (error != AAC_ENC_OK) { return error; } i++; } while (list->id[i] != end_of_sequence); bail: if (pBitDemand != NULL) { *pBitDemand = bitDemand; } return error; } //----------------------------------------------------------------------------------------------- AAC_ENCODER_ERROR FDKaacEnc_WriteBitstream(HANDLE_TRANSPORTENC hTpEnc, CHANNEL_MAPPING *channelMapping, QC_OUT *qcOut, PSY_OUT *psyOut, QC_STATE *qcKernel, AUDIO_OBJECT_TYPE aot, UINT syntaxFlags, SCHAR epConfig) { HANDLE_FDK_BITSTREAM hBs = transportEnc_GetBitstream(hTpEnc); AAC_ENCODER_ERROR ErrorStatus = AAC_ENC_OK; int i, n, doByteAlign = 1; INT bitMarkUp; INT frameBits; /* Get first bit of raw data block. In case of ADTS+PCE, AU would start at PCE. This is okay because PCE assures alignment. */ UINT alignAnchor = FDKgetValidBits(hBs); frameBits = bitMarkUp = alignAnchor; /* Channel element loop */ for (i = 0; i < channelMapping->nElements; i++) { ELEMENT_INFO elInfo = channelMapping->elInfo[i]; INT elementUsedBits = 0; switch (elInfo.elType) { case ID_SCE: /* single channel */ case ID_CPE: /* channel pair */ case ID_LFE: /* low freq effects channel */ { if (AAC_ENC_OK != (ErrorStatus = FDKaacEnc_ChannelElementWrite( hTpEnc, &elInfo, qcOut->qcElement[i]->qcOutChannel, psyOut->psyOutElement[i], psyOut->psyOutElement[i]->psyOutChannel, syntaxFlags, /* syntaxFlags (ER tools ...) */ aot, /* aot: AOT_AAC_LC, AOT_SBR, AOT_PS */ epConfig, /* epConfig -1, 0, 1 */ NULL, 0))) { return ErrorStatus; } if (!(syntaxFlags & AC_ER)) { /* Write associated extension payload */ for (n = 0; n < qcOut->qcElement[i]->nExtensions; n++) { FDKaacEnc_writeExtensionData( hTpEnc, &qcOut->qcElement[i]->extension[n], 0, alignAnchor, syntaxFlags, aot, epConfig); } } } break; /* In FDK, DSE signalling explicit done in elDSE. See channel_map.cpp */ default: return AAC_ENC_INVALID_ELEMENTINFO_TYPE; } /* switch */ if (elInfo.elType != ID_DSE) { elementUsedBits -= bitMarkUp; bitMarkUp = FDKgetValidBits(hBs); elementUsedBits += bitMarkUp; frameBits += elementUsedBits; } } /* for (i=0; inElements; i++) { for (n = 0; n < qcOut->qcElement[i]->nExtensions; n++) { if ((qcOut->qcElement[i]->extension[n].type == EXT_SBR_DATA) || (qcOut->qcElement[i]->extension[n].type == EXT_SBR_DATA_CRC)) { /* Write sbr extension payload */ FDKaacEnc_writeExtensionData( hTpEnc, &qcOut->qcElement[i]->extension[n], 0, alignAnchor, syntaxFlags, aot, epConfig); channelElementExtensionWritten[i][n] = 1; } /* SBR */ } /* n */ } /* i */ } /* AC_ELD */ for (i = 0; i < channelMapping->nElements; i++) { for (n = 0; n < qcOut->qcElement[i]->nExtensions; n++) { if (channelElementExtensionWritten[i][n] == 0) { /* Write all ramaining extension payloads in element */ FDKaacEnc_writeExtensionData(hTpEnc, &qcOut->qcElement[i]->extension[n], 0, alignAnchor, syntaxFlags, aot, epConfig); } } /* n */ } /* i */ } /* if AC_ER */ /* Extend global extension payload table with fill bits */ n = qcOut->nExtensions; /* Add fill data / stuffing bits */ qcOut->extension[n].type = EXT_FILL_DATA; qcOut->extension[n].nPayloadBits = qcOut->totFillBits; qcOut->nExtensions++; /* Write global extension payload and fill data */ for (n = 0; (n < qcOut->nExtensions) && (n < (2 + 2)); n++) { FDKaacEnc_writeExtensionData(hTpEnc, &qcOut->extension[n], 0, alignAnchor, syntaxFlags, aot, epConfig); /* For EXT_FIL or EXT_FILL_DATA we could do an additional sanity check here */ } if (!(syntaxFlags & (AC_SCALABLE | AC_ER))) { FDKwriteBits(hBs, ID_END, EL_ID_BITS); } if (doByteAlign) { /* Assure byte alignment*/ if (((FDKgetValidBits(hBs) - alignAnchor + qcOut->alignBits) & 0x7) != 0) { return AAC_ENC_WRITTEN_BITS_ERROR; } FDKaacEnc_ByteAlignment(hBs, qcOut->alignBits); } frameBits -= bitMarkUp; frameBits += FDKgetValidBits(hBs); transportEnc_EndAccessUnit(hTpEnc, &frameBits); if (frameBits != qcOut->totalBits + qcKernel->globHdrBits) { return AAC_ENC_WRITTEN_BITS_ERROR; } return ErrorStatus; } fdk-aac-2.0.2/libAACenc/src/bitenc.h000066400000000000000000000177631404225310700167620ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Werner Description: Bitstream encoder *******************************************************************************/ #ifndef BITENC_H #define BITENC_H #include "qc_data.h" #include "aacenc_tns.h" #include "channel_map.h" #include "interface.h" /* obsolete, when PSY_OUT is thrown out of the WritBS-call! */ #include "FDK_audio.h" #include "aacenc.h" #include "tpenc_lib.h" typedef enum { MAX_ENCODER_CHANNELS = 9, MAX_BLOCK_TYPES = 4, MAX_AAC_LAYERS = 9, MAX_LAYERS = MAX_AAC_LAYERS, /* only one core layer if present */ FIRST_LAY = 1 /* default layer number for AAC nonscalable */ } _MAX_CONST; #define BUFFER_MX_HUFFCB_SIZE \ (32 * sizeof(INT)) /* our FDK_bitbuffer needs size of power 2 */ #define EL_ID_BITS (3) /** * \brief Arbitrary order bitstream writer. This function can either assemble a * bit stream and write into the bit buffer of hTpEnc or calculate the number of * static bits (signal independent) TpEnc handle must be NULL in this * case. Or also Calculate the minimum possible number of static bits * which by disabling all tools e.g. MS, TNS and sbfCnt=0. The minCnt * parameter has to be 1 in this latter case. * \param hTpEnc Transport encoder handle. If NULL, the number of static bits * will be returned into *pBitDemand. * \param pElInfo * \param qcOutChannel * \param hReorderInfo * \param psyOutElement * \param psyOutChannel * \param syntaxFlags Bit stream syntax flags as defined in FDK_audio.h (Audio * Codec flags). * \param aot * \param epConfig * \param pBitDemand Pointer to an int where the amount of bits is returned * into. The returned value depends on if hTpEnc is NULL and minCnt. * \param minCnt If non-zero the value returned into *pBitDemand is the absolute * minimum required amount of static bits in order to write a valid bit stream. * \return AAC_ENCODER_ERROR error code */ AAC_ENCODER_ERROR FDKaacEnc_ChannelElementWrite( HANDLE_TRANSPORTENC hTpEnc, ELEMENT_INFO *pElInfo, QC_OUT_CHANNEL *qcOutChannel[(2)], PSY_OUT_ELEMENT *psyOutElement, PSY_OUT_CHANNEL *psyOutChannel[(2)], UINT syntaxFlags, AUDIO_OBJECT_TYPE aot, SCHAR epConfig, INT *pBitDemand, UCHAR minCnt); /** * \brief Write bit stream or account static bits * \param hTpEnc transport encoder handle. If NULL, the function will * not write any bit stream data but only count the amount * of static (signal independent) bits * \param channelMapping Channel mapping info * \param qcOut * \param psyOut * \param qcKernel * \param hBSE * \param aot Audio Object Type being encoded * \param syntaxFlags Flags indicating format specific detail * \param epConfig Error protection config */ AAC_ENCODER_ERROR FDKaacEnc_WriteBitstream(HANDLE_TRANSPORTENC hTpEnc, CHANNEL_MAPPING *channelMapping, QC_OUT *qcOut, PSY_OUT *psyOut, QC_STATE *qcKernel, AUDIO_OBJECT_TYPE aot, UINT syntaxFlags, SCHAR epConfig); INT FDKaacEnc_writeExtensionData(HANDLE_TRANSPORTENC hTpEnc, QC_OUT_EXTENSION *pExtension, INT elInstanceTag, UINT alignAnchor, UINT syntaxFlags, AUDIO_OBJECT_TYPE aot, SCHAR epConfig); #endif /* BITENC_H */ fdk-aac-2.0.2/libAACenc/src/block_switch.cpp000066400000000000000000000575061404225310700205230ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Werner, Tobias Chalupka Description: Block switching *******************************************************************************/ /****************** Includes *****************************/ #include "block_switch.h" #include "genericStds.h" #define LOWOV_WINDOW _LOWOV_WINDOW /**************** internal function prototypes ***********/ static FIXP_DBL FDKaacEnc_GetWindowEnergy(const FIXP_DBL in[], const INT blSwWndIdx); static void FDKaacEnc_CalcWindowEnergy( BLOCK_SWITCHING_CONTROL *RESTRICT blockSwitchingControl, INT windowLen, const INT_PCM *pTimeSignal); /****************** Constants *****************************/ /* LONG START * SHORT STOP LOWOV */ static const INT blockType2windowShape[2][5] = { {SINE_WINDOW, KBD_WINDOW, WRONG_WINDOW, SINE_WINDOW, KBD_WINDOW}, /* LD */ {KBD_WINDOW, SINE_WINDOW, SINE_WINDOW, KBD_WINDOW, WRONG_WINDOW}}; /* LC */ /* IIR high pass coeffs */ #ifndef SINETABLE_16BIT static const FIXP_DBL hiPassCoeff[BLOCK_SWITCHING_IIR_LEN] = { FL2FXCONST_DBL(-0.5095), FL2FXCONST_DBL(0.7548)}; static const FIXP_DBL accWindowNrgFac = FL2FXCONST_DBL(0.3f); /* factor for accumulating filtered window energies */ static const FIXP_DBL oneMinusAccWindowNrgFac = FL2FXCONST_DBL(0.7f); /* static const float attackRatio = 10.0; */ /* lower ratio limit for attacks */ static const FIXP_DBL invAttackRatio = FL2FXCONST_DBL(0.1f); /* inverted lower ratio limit for attacks */ /* The next constants are scaled, because they are used for comparison with * scaled values*/ /* minimum energy for attacks */ static const FIXP_DBL minAttackNrg = (FL2FXCONST_DBL(1e+6f * NORM_PCM_ENERGY) >> BLOCK_SWITCH_ENERGY_SHIFT); /* minimum energy for attacks */ #else static const FIXP_SGL hiPassCoeff[BLOCK_SWITCHING_IIR_LEN] = { FL2FXCONST_SGL(-0.5095), FL2FXCONST_SGL(0.7548)}; static const FIXP_DBL accWindowNrgFac = FL2FXCONST_DBL(0.3f); /* factor for accumulating filtered window energies */ static const FIXP_SGL oneMinusAccWindowNrgFac = FL2FXCONST_SGL(0.7f); /* static const float attackRatio = 10.0; */ /* lower ratio limit for attacks */ static const FIXP_SGL invAttackRatio = FL2FXCONST_SGL(0.1f); /* inverted lower ratio limit for attacks */ /* minimum energy for attacks */ static const FIXP_DBL minAttackNrg = (FL2FXCONST_DBL(1e+6f * NORM_PCM_ENERGY) >> BLOCK_SWITCH_ENERGY_SHIFT); /* minimum energy for attacks */ #endif /**************** internal function prototypes ***********/ /****************** Routines ****************************/ void FDKaacEnc_InitBlockSwitching( BLOCK_SWITCHING_CONTROL *blockSwitchingControl, INT isLowDelay) { FDKmemclear(blockSwitchingControl, sizeof(BLOCK_SWITCHING_CONTROL)); if (isLowDelay) { blockSwitchingControl->nBlockSwitchWindows = 4; blockSwitchingControl->allowShortFrames = 0; blockSwitchingControl->allowLookAhead = 0; } else { blockSwitchingControl->nBlockSwitchWindows = 8; blockSwitchingControl->allowShortFrames = 1; blockSwitchingControl->allowLookAhead = 1; } blockSwitchingControl->noOfGroups = MAX_NO_OF_GROUPS; /* Initialize startvalue for blocktype */ blockSwitchingControl->lastWindowSequence = LONG_WINDOW; blockSwitchingControl->windowShape = blockType2windowShape[blockSwitchingControl->allowShortFrames] [blockSwitchingControl->lastWindowSequence]; } static const INT suggestedGroupingTable[TRANS_FAC][MAX_NO_OF_GROUPS] = { /* Attack in Window 0 */ {1, 3, 3, 1}, /* Attack in Window 1 */ {1, 1, 3, 3}, /* Attack in Window 2 */ {2, 1, 3, 2}, /* Attack in Window 3 */ {3, 1, 3, 1}, /* Attack in Window 4 */ {3, 1, 1, 3}, /* Attack in Window 5 */ {3, 2, 1, 2}, /* Attack in Window 6 */ {3, 3, 1, 1}, /* Attack in Window 7 */ {3, 3, 1, 1}}; /* change block type depending on current blocktype and whether there's an * attack */ /* assume no look-ahead */ static const INT chgWndSq[2][N_BLOCKTYPES] = { /* LONG WINDOW START_WINDOW SHORT_WINDOW STOP_WINDOW, LOWOV_WINDOW, WRONG_WINDOW */ /*no attack*/ {LONG_WINDOW, STOP_WINDOW, WRONG_WINDOW, LONG_WINDOW, STOP_WINDOW, WRONG_WINDOW}, /*attack */ {START_WINDOW, LOWOV_WINDOW, WRONG_WINDOW, START_WINDOW, LOWOV_WINDOW, WRONG_WINDOW}}; /* change block type depending on current blocktype and whether there's an * attack */ /* assume look-ahead */ static const INT chgWndSqLkAhd[2][2][N_BLOCKTYPES] = { /*attack LONG WINDOW START_WINDOW SHORT_WINDOW STOP_WINDOW LOWOV_WINDOW, WRONG_WINDOW */ /* last attack */ /*no attack*/ { {LONG_WINDOW, SHORT_WINDOW, STOP_WINDOW, LONG_WINDOW, WRONG_WINDOW, WRONG_WINDOW}, /* no attack */ /*attack */ {START_WINDOW, SHORT_WINDOW, SHORT_WINDOW, START_WINDOW, WRONG_WINDOW, WRONG_WINDOW}}, /* no attack */ /*no attack*/ {{LONG_WINDOW, SHORT_WINDOW, SHORT_WINDOW, LONG_WINDOW, WRONG_WINDOW, WRONG_WINDOW}, /* attack */ /*attack */ {START_WINDOW, SHORT_WINDOW, SHORT_WINDOW, START_WINDOW, WRONG_WINDOW, WRONG_WINDOW}} /* attack */ }; int FDKaacEnc_BlockSwitching(BLOCK_SWITCHING_CONTROL *blockSwitchingControl, const INT granuleLength, const int isLFE, const INT_PCM *pTimeSignal) { UINT i; FIXP_DBL enM1, enMax; UINT nBlockSwitchWindows = blockSwitchingControl->nBlockSwitchWindows; /* for LFE : only LONG window allowed */ if (isLFE) { /* case LFE: */ /* only long blocks, always use sine windows (MPEG2 AAC, MPEG4 AAC) */ blockSwitchingControl->lastWindowSequence = LONG_WINDOW; blockSwitchingControl->windowShape = SINE_WINDOW; blockSwitchingControl->noOfGroups = 1; blockSwitchingControl->groupLen[0] = 1; return (0); }; /* Save current attack index as last attack index */ blockSwitchingControl->lastattack = blockSwitchingControl->attack; blockSwitchingControl->lastAttackIndex = blockSwitchingControl->attackIndex; /* Save current window energy as last window energy */ FDKmemcpy(blockSwitchingControl->windowNrg[0], blockSwitchingControl->windowNrg[1], sizeof(blockSwitchingControl->windowNrg[0])); FDKmemcpy(blockSwitchingControl->windowNrgF[0], blockSwitchingControl->windowNrgF[1], sizeof(blockSwitchingControl->windowNrgF[0])); if (blockSwitchingControl->allowShortFrames) { /* Calculate suggested grouping info for the last frame */ /* Reset grouping info */ FDKmemclear(blockSwitchingControl->groupLen, sizeof(blockSwitchingControl->groupLen)); /* Set grouping info */ blockSwitchingControl->noOfGroups = MAX_NO_OF_GROUPS; FDKmemcpy(blockSwitchingControl->groupLen, suggestedGroupingTable[blockSwitchingControl->lastAttackIndex], sizeof(blockSwitchingControl->groupLen)); if (blockSwitchingControl->attack == TRUE) blockSwitchingControl->maxWindowNrg = FDKaacEnc_GetWindowEnergy(blockSwitchingControl->windowNrg[0], blockSwitchingControl->lastAttackIndex); else blockSwitchingControl->maxWindowNrg = FL2FXCONST_DBL(0.0); } /* Calculate unfiltered and filtered energies in subwindows and combine to * segments */ FDKaacEnc_CalcWindowEnergy( blockSwitchingControl, granuleLength >> (nBlockSwitchWindows == 4 ? 2 : 3), pTimeSignal); /* now calculate if there is an attack */ /* reset attack */ blockSwitchingControl->attack = FALSE; /* look for attack */ enMax = FL2FXCONST_DBL(0.0f); enM1 = blockSwitchingControl->windowNrgF[0][nBlockSwitchWindows - 1]; for (i = 0; i < nBlockSwitchWindows; i++) { FIXP_DBL tmp = fMultDiv2(oneMinusAccWindowNrgFac, blockSwitchingControl->accWindowNrg); blockSwitchingControl->accWindowNrg = fMultAdd(tmp, accWindowNrgFac, enM1); if (fMult(blockSwitchingControl->windowNrgF[1][i], invAttackRatio) > blockSwitchingControl->accWindowNrg) { blockSwitchingControl->attack = TRUE; blockSwitchingControl->attackIndex = i; } enM1 = blockSwitchingControl->windowNrgF[1][i]; enMax = fixMax(enMax, enM1); } if (enMax < minAttackNrg) blockSwitchingControl->attack = FALSE; /* Check if attack spreads over frame border */ if ((blockSwitchingControl->attack == FALSE) && (blockSwitchingControl->lastattack == TRUE)) { /* if attack is in last window repeat SHORT_WINDOW */ if (((blockSwitchingControl->windowNrgF[0][nBlockSwitchWindows - 1] >> 4) > fMult((FIXP_DBL)(10 << (DFRACT_BITS - 1 - 4)), blockSwitchingControl->windowNrgF[1][1])) && (blockSwitchingControl->lastAttackIndex == (INT)nBlockSwitchWindows - 1)) { blockSwitchingControl->attack = TRUE; blockSwitchingControl->attackIndex = 0; } } if (blockSwitchingControl->allowLookAhead) { blockSwitchingControl->lastWindowSequence = chgWndSqLkAhd[blockSwitchingControl->lastattack] [blockSwitchingControl->attack] [blockSwitchingControl->lastWindowSequence]; } else { /* Low Delay */ blockSwitchingControl->lastWindowSequence = chgWndSq[blockSwitchingControl->attack] [blockSwitchingControl->lastWindowSequence]; } /* update window shape */ blockSwitchingControl->windowShape = blockType2windowShape[blockSwitchingControl->allowShortFrames] [blockSwitchingControl->lastWindowSequence]; return (0); } static FIXP_DBL FDKaacEnc_GetWindowEnergy(const FIXP_DBL in[], const INT blSwWndIdx) { /* For coherency, change FDKaacEnc_GetWindowEnergy() to calcluate the energy for a block switching analysis windows, not for a short block. The same is done FDKaacEnc_CalcWindowEnergy(). The result of FDKaacEnc_GetWindowEnergy() is used for a comparision of the max energy of left/right channel. */ return in[blSwWndIdx]; } static void FDKaacEnc_CalcWindowEnergy( BLOCK_SWITCHING_CONTROL *RESTRICT blockSwitchingControl, INT windowLen, const INT_PCM *pTimeSignal) { INT i; UINT w; #ifndef SINETABLE_16BIT const FIXP_DBL hiPassCoeff0 = hiPassCoeff[0]; const FIXP_DBL hiPassCoeff1 = hiPassCoeff[1]; #else const FIXP_SGL hiPassCoeff0 = hiPassCoeff[0]; const FIXP_SGL hiPassCoeff1 = hiPassCoeff[1]; #endif FIXP_DBL temp_iirState0 = blockSwitchingControl->iirStates[0]; FIXP_DBL temp_iirState1 = blockSwitchingControl->iirStates[1]; /* sum up scalarproduct of timesignal as windowed Energies */ for (w = 0; w < blockSwitchingControl->nBlockSwitchWindows; w++) { ULONG temp_windowNrg = 0x0; ULONG temp_windowNrgF = 0x0; /* windowNrg = sum(timesample^2) */ for (i = 0; i < windowLen; i++) { FIXP_DBL tempUnfiltered, t1, t2; /* tempUnfiltered is scaled with 1 to prevent overflows during calculation * of tempFiltred */ #if SAMPLE_BITS == DFRACT_BITS tempUnfiltered = (FIXP_DBL)*pTimeSignal++ >> 1; #else tempUnfiltered = (FIXP_DBL)*pTimeSignal++ << (DFRACT_BITS - SAMPLE_BITS - 1); #endif t1 = fMultDiv2(hiPassCoeff1, tempUnfiltered - temp_iirState0); t2 = fMultDiv2(hiPassCoeff0, temp_iirState1); temp_iirState0 = tempUnfiltered; temp_iirState1 = (t1 - t2) << 1; temp_windowNrg += (LONG)fPow2Div2(temp_iirState0) >> (BLOCK_SWITCH_ENERGY_SHIFT - 1 - 2); temp_windowNrgF += (LONG)fPow2Div2(temp_iirState1) >> (BLOCK_SWITCH_ENERGY_SHIFT - 1 - 2); } blockSwitchingControl->windowNrg[1][w] = (LONG)fMin(temp_windowNrg, (UINT)MAXVAL_DBL); blockSwitchingControl->windowNrgF[1][w] = (LONG)fMin(temp_windowNrgF, (UINT)MAXVAL_DBL); } blockSwitchingControl->iirStates[0] = temp_iirState0; blockSwitchingControl->iirStates[1] = temp_iirState1; } static const UCHAR synchronizedBlockTypeTable[5][5] = { /* LONG_WINDOW START_WINDOW SHORT_WINDOW STOP_WINDOW LOWOV_WINDOW*/ /* LONG_WINDOW */ {LONG_WINDOW, START_WINDOW, SHORT_WINDOW, STOP_WINDOW, LOWOV_WINDOW}, /* START_WINDOW */ {START_WINDOW, START_WINDOW, SHORT_WINDOW, SHORT_WINDOW, LOWOV_WINDOW}, /* SHORT_WINDOW */ {SHORT_WINDOW, SHORT_WINDOW, SHORT_WINDOW, SHORT_WINDOW, WRONG_WINDOW}, /* STOP_WINDOW */ {STOP_WINDOW, SHORT_WINDOW, SHORT_WINDOW, STOP_WINDOW, LOWOV_WINDOW}, /* LOWOV_WINDOW */ {LOWOV_WINDOW, LOWOV_WINDOW, WRONG_WINDOW, LOWOV_WINDOW, LOWOV_WINDOW}, }; int FDKaacEnc_SyncBlockSwitching( BLOCK_SWITCHING_CONTROL *blockSwitchingControlLeft, BLOCK_SWITCHING_CONTROL *blockSwitchingControlRight, const INT nChannels, const INT commonWindow) { UCHAR patchType = LONG_WINDOW; if (nChannels == 2 && commonWindow == TRUE) { /* could be better with a channel loop (need a handle to psy_data) */ /* get suggested Block Types and synchronize */ patchType = synchronizedBlockTypeTable[patchType][blockSwitchingControlLeft ->lastWindowSequence]; patchType = synchronizedBlockTypeTable[patchType][blockSwitchingControlRight ->lastWindowSequence]; /* sanity check (no change from low overlap window to short winow and vice * versa) */ if (patchType == WRONG_WINDOW) return -1; /* mixed up AAC-LC and AAC-LD */ /* Set synchronized Blocktype */ blockSwitchingControlLeft->lastWindowSequence = patchType; blockSwitchingControlRight->lastWindowSequence = patchType; /* update window shape */ blockSwitchingControlLeft->windowShape = blockType2windowShape[blockSwitchingControlLeft->allowShortFrames] [blockSwitchingControlLeft->lastWindowSequence]; blockSwitchingControlRight->windowShape = blockType2windowShape[blockSwitchingControlLeft->allowShortFrames] [blockSwitchingControlRight->lastWindowSequence]; } if (blockSwitchingControlLeft->allowShortFrames) { int i; if (nChannels == 2) { if (commonWindow == TRUE) { /* Synchronize grouping info */ int windowSequenceLeftOld = blockSwitchingControlLeft->lastWindowSequence; int windowSequenceRightOld = blockSwitchingControlRight->lastWindowSequence; /* Long Blocks */ if (patchType != SHORT_WINDOW) { /* Set grouping info */ blockSwitchingControlLeft->noOfGroups = 1; blockSwitchingControlRight->noOfGroups = 1; blockSwitchingControlLeft->groupLen[0] = 1; blockSwitchingControlRight->groupLen[0] = 1; for (i = 1; i < MAX_NO_OF_GROUPS; i++) { blockSwitchingControlLeft->groupLen[i] = 0; blockSwitchingControlRight->groupLen[i] = 0; } } /* Short Blocks */ else { /* in case all two channels were detected as short-blocks before * syncing, use the grouping of channel with higher maxWindowNrg */ if ((windowSequenceLeftOld == SHORT_WINDOW) && (windowSequenceRightOld == SHORT_WINDOW)) { if (blockSwitchingControlLeft->maxWindowNrg > blockSwitchingControlRight->maxWindowNrg) { /* Left Channel wins */ blockSwitchingControlRight->noOfGroups = blockSwitchingControlLeft->noOfGroups; for (i = 0; i < MAX_NO_OF_GROUPS; i++) { blockSwitchingControlRight->groupLen[i] = blockSwitchingControlLeft->groupLen[i]; } } else { /* Right Channel wins */ blockSwitchingControlLeft->noOfGroups = blockSwitchingControlRight->noOfGroups; for (i = 0; i < MAX_NO_OF_GROUPS; i++) { blockSwitchingControlLeft->groupLen[i] = blockSwitchingControlRight->groupLen[i]; } } } else if ((windowSequenceLeftOld == SHORT_WINDOW) && (windowSequenceRightOld != SHORT_WINDOW)) { /* else use grouping of short-block channel */ blockSwitchingControlRight->noOfGroups = blockSwitchingControlLeft->noOfGroups; for (i = 0; i < MAX_NO_OF_GROUPS; i++) { blockSwitchingControlRight->groupLen[i] = blockSwitchingControlLeft->groupLen[i]; } } else if ((windowSequenceRightOld == SHORT_WINDOW) && (windowSequenceLeftOld != SHORT_WINDOW)) { blockSwitchingControlLeft->noOfGroups = blockSwitchingControlRight->noOfGroups; for (i = 0; i < MAX_NO_OF_GROUPS; i++) { blockSwitchingControlLeft->groupLen[i] = blockSwitchingControlRight->groupLen[i]; } } else { /* syncing a start and stop window ... */ blockSwitchingControlLeft->noOfGroups = blockSwitchingControlRight->noOfGroups = 2; blockSwitchingControlLeft->groupLen[0] = blockSwitchingControlRight->groupLen[0] = 4; blockSwitchingControlLeft->groupLen[1] = blockSwitchingControlRight->groupLen[1] = 4; } } /* Short Blocks */ } else { /* stereo, no common window */ if (blockSwitchingControlLeft->lastWindowSequence != SHORT_WINDOW) { blockSwitchingControlLeft->noOfGroups = 1; blockSwitchingControlLeft->groupLen[0] = 1; for (i = 1; i < MAX_NO_OF_GROUPS; i++) { blockSwitchingControlLeft->groupLen[i] = 0; } } if (blockSwitchingControlRight->lastWindowSequence != SHORT_WINDOW) { blockSwitchingControlRight->noOfGroups = 1; blockSwitchingControlRight->groupLen[0] = 1; for (i = 1; i < MAX_NO_OF_GROUPS; i++) { blockSwitchingControlRight->groupLen[i] = 0; } } } /* common window */ } else { /* Mono */ if (blockSwitchingControlLeft->lastWindowSequence != SHORT_WINDOW) { blockSwitchingControlLeft->noOfGroups = 1; blockSwitchingControlLeft->groupLen[0] = 1; for (i = 1; i < MAX_NO_OF_GROUPS; i++) { blockSwitchingControlLeft->groupLen[i] = 0; } } } } /* allowShortFrames */ /* Translate LOWOV_WINDOW block type to a meaningful window shape. */ if (!blockSwitchingControlLeft->allowShortFrames) { if (blockSwitchingControlLeft->lastWindowSequence != LONG_WINDOW && blockSwitchingControlLeft->lastWindowSequence != STOP_WINDOW) { blockSwitchingControlLeft->lastWindowSequence = LONG_WINDOW; blockSwitchingControlLeft->windowShape = LOL_WINDOW; } } if (nChannels == 2) { if (!blockSwitchingControlRight->allowShortFrames) { if (blockSwitchingControlRight->lastWindowSequence != LONG_WINDOW && blockSwitchingControlRight->lastWindowSequence != STOP_WINDOW) { blockSwitchingControlRight->lastWindowSequence = LONG_WINDOW; blockSwitchingControlRight->windowShape = LOL_WINDOW; } } } return 0; } fdk-aac-2.0.2/libAACenc/src/block_switch.h000066400000000000000000000155261404225310700201640ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Werner Description: Block switching *******************************************************************************/ #ifndef BLOCK_SWITCH_H #define BLOCK_SWITCH_H #include "common_fix.h" #include "psy_const.h" /****************** Defines ******************************/ #define BLOCK_SWITCH_WINDOWS 8 /* number of windows for energy calculation */ #define BLOCK_SWITCHING_IIR_LEN \ 2 /* Length of HighPass-IIR-Filter for Attack-Detection */ #define BLOCK_SWITCH_ENERGY_SHIFT \ 7 /* should be logDualis(BLOCK_SWITCH_WINDOW_LEN) to avoid overflow in \ windowNrgs. */ #define LAST_WINDOW 0 #define THIS_WINDOW 1 /****************** Structures ***************************/ typedef struct { INT lastWindowSequence; INT windowShape; INT lastWindowShape; UINT nBlockSwitchWindows; /* number of windows for energy calculation */ INT attack; INT lastattack; INT attackIndex; INT lastAttackIndex; INT allowShortFrames; /* for Low Delay, don't allow short frames */ INT allowLookAhead; /* for Low Delay, don't do look-ahead */ INT noOfGroups; INT groupLen[MAX_NO_OF_GROUPS]; FIXP_DBL maxWindowNrg; /* max energy in subwindows */ FIXP_DBL windowNrg[2][BLOCK_SWITCH_WINDOWS]; /* time signal energy in Subwindows (last and current) */ FIXP_DBL windowNrgF[2][BLOCK_SWITCH_WINDOWS]; /* filtered time signal energy in segments (last and current) */ FIXP_DBL accWindowNrg; /* recursively accumulated windowNrgF */ FIXP_DBL iirStates[BLOCK_SWITCHING_IIR_LEN]; /* filter delay-line */ } BLOCK_SWITCHING_CONTROL; void FDKaacEnc_InitBlockSwitching( BLOCK_SWITCHING_CONTROL *blockSwitchingControl, INT isLowDelay); int FDKaacEnc_BlockSwitching(BLOCK_SWITCHING_CONTROL *blockSwitchingControl, const INT granuleLength, const int isLFE, const INT_PCM *pTimeSignal); int FDKaacEnc_SyncBlockSwitching( BLOCK_SWITCHING_CONTROL *blockSwitchingControlLeft, BLOCK_SWITCHING_CONTROL *blockSwitchingControlRight, const INT noOfChannels, const INT commonWindow); #endif /* #ifndef BLOCK_SWITCH_H */ fdk-aac-2.0.2/libAACenc/src/channel_map.cpp000066400000000000000000000653221404225310700203100ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): A. Groeschel Description: channel mapping functionality *******************************************************************************/ #include "channel_map.h" #include "bitenc.h" #include "psy_const.h" #include "qc_data.h" #include "aacEnc_ram.h" #include "FDK_tools_rom.h" /* channel_assignment treats the relationship of Input file channels to the encoder channels. This is necessary because the usual order in RIFF files (.wav) is different from the elements order in the coder given by Table 8.1 (implicit speaker mapping) of the AAC standard. In mono and stereo case, this is trivial. In mc case, it looks like this: Channel Input file coder chan 5ch: front center 2 0 (SCE channel) left center 0 1 (1st of 1st CPE) right center 1 2 (2nd of 1st CPE) left surround 3 3 (1st of 2nd CPE) right surround 4 4 (2nd of 2nd CPE) 5.1ch: front center 2 0 (SCE channel) left center 0 1 (1st of 1st CPE) right center 1 2 (2nd of 1st CPE) left surround 4 3 (1st of 2nd CPE) right surround 5 4 (2nd of 2nd CPE) LFE 3 5 (LFE) */ /* Channel mode configuration tab provides, corresponding number of channels and elements */ static const CHANNEL_MODE_CONFIG_TAB channelModeConfig[] = { {MODE_1, 1, 1, 1}, /* chCfg 1, SCE */ {MODE_2, 2, 2, 1}, /* chCfg 2, CPE */ {MODE_1_2, 3, 3, 2}, /* chCfg 3, SCE,CPE */ {MODE_1_2_1, 4, 4, 3}, /* chCfg 4, SCE,CPE,SCE */ {MODE_1_2_2, 5, 5, 3}, /* chCfg 5, SCE,CPE,CPE */ {MODE_1_2_2_1, 6, 5, 4}, /* chCfg 6, SCE,CPE,CPE,LFE */ {MODE_1_2_2_2_1, 8, 7, 5}, /* chCfg 7, SCE,CPE,CPE,CPE,LFE */ {MODE_6_1, 7, 6, 5}, /* chCfg 11, SCE,CPE,CPE,SCE,LFE */ {MODE_7_1_BACK, 8, 7, 5}, /* chCfg 12, SCE,CPE,CPE,CPE,LFE */ {MODE_7_1_TOP_FRONT, 8, 7, 5}, /* chCfg 14, SCE,CPE,CPE,LFE,CPE */ {MODE_7_1_REAR_SURROUND, 8, 7, 5}, /* same as MODE_7_1_BACK, SCE,CPE,CPE,CPE,LFE */ {MODE_7_1_FRONT_CENTER, 8, 7, 5}, /* same as MODE_1_2_2_2_1, SCE,CPE,CPE,CPE,LFE */ }; AAC_ENCODER_ERROR FDKaacEnc_DetermineEncoderMode(CHANNEL_MODE* mode, INT nChannels) { INT i; CHANNEL_MODE encMode = MODE_INVALID; if (*mode == MODE_UNKNOWN) { for (i = 0; i < (INT)sizeof(channelModeConfig) / (INT)sizeof(CHANNEL_MODE_CONFIG_TAB); i++) { if (channelModeConfig[i].nChannels == nChannels) { encMode = channelModeConfig[i].encMode; break; } } *mode = encMode; } else { /* check if valid channel configuration */ if (FDKaacEnc_GetChannelModeConfiguration(*mode)->nChannels == nChannels) { encMode = *mode; } } if (encMode == MODE_INVALID) { return AAC_ENC_UNSUPPORTED_CHANNELCONFIG; } return AAC_ENC_OK; } static INT FDKaacEnc_initElement(ELEMENT_INFO* elInfo, MP4_ELEMENT_ID elType, INT* cnt, FDK_channelMapDescr* mapDescr, UINT mapIdx, INT* it_cnt, const FIXP_DBL relBits) { INT error = 0; INT counter = *cnt; elInfo->elType = elType; elInfo->relativeBits = relBits; switch (elInfo->elType) { case ID_SCE: case ID_LFE: case ID_CCE: elInfo->nChannelsInEl = 1; elInfo->ChannelIndex[0] = FDK_chMapDescr_getMapValue(mapDescr, counter++, mapIdx); elInfo->instanceTag = it_cnt[elType]++; break; case ID_CPE: elInfo->nChannelsInEl = 2; elInfo->ChannelIndex[0] = FDK_chMapDescr_getMapValue(mapDescr, counter++, mapIdx); elInfo->ChannelIndex[1] = FDK_chMapDescr_getMapValue(mapDescr, counter++, mapIdx); elInfo->instanceTag = it_cnt[elType]++; break; case ID_DSE: elInfo->nChannelsInEl = 0; elInfo->ChannelIndex[0] = 0; elInfo->ChannelIndex[1] = 0; elInfo->instanceTag = it_cnt[elType]++; break; default: error = 1; }; *cnt = counter; return error; } AAC_ENCODER_ERROR FDKaacEnc_InitChannelMapping(CHANNEL_MODE mode, CHANNEL_ORDER co, CHANNEL_MAPPING* cm) { INT count = 0; /* count through coder channels */ INT it_cnt[ID_END + 1]; INT i; UINT mapIdx; FDK_channelMapDescr mapDescr; for (i = 0; i < ID_END; i++) it_cnt[i] = 0; FDKmemclear(cm, sizeof(CHANNEL_MAPPING)); /* init channel mapping*/ for (i = 0; i < (INT)sizeof(channelModeConfig) / (INT)sizeof(CHANNEL_MODE_CONFIG_TAB); i++) { if (channelModeConfig[i].encMode == mode) { cm->encMode = channelModeConfig[i].encMode; cm->nChannels = channelModeConfig[i].nChannels; cm->nChannelsEff = channelModeConfig[i].nChannelsEff; cm->nElements = channelModeConfig[i].nElements; break; } } /* init map descriptor */ FDK_chMapDescr_init(&mapDescr, NULL, 0, (co == CH_ORDER_MPEG) ? 1 : 0); switch (mode) { case MODE_7_1_REAR_SURROUND: /* MODE_7_1_REAR_SURROUND is equivalent to MODE_7_1_BACK */ mapIdx = (INT)MODE_7_1_BACK; break; case MODE_7_1_FRONT_CENTER: /* MODE_7_1_FRONT_CENTER is equivalent to MODE_1_2_2_2_1 */ mapIdx = (INT)MODE_1_2_2_2_1; break; default: mapIdx = (INT)mode > 14 ? 0 : (INT) mode; /* if channel config > 14 MPEG mapping will be used */ } /* init element info struct */ switch (mode) { case MODE_1: /* (mono) sce */ FDKaacEnc_initElement(&cm->elInfo[0], ID_SCE, &count, &mapDescr, mapIdx, it_cnt, (FIXP_DBL)MAXVAL_DBL); break; case MODE_2: /* (stereo) cpe */ FDKaacEnc_initElement(&cm->elInfo[0], ID_CPE, &count, &mapDescr, mapIdx, it_cnt, (FIXP_DBL)MAXVAL_DBL); break; case MODE_1_2: /* sce + cpe */ FDKaacEnc_initElement(&cm->elInfo[0], ID_SCE, &count, &mapDescr, mapIdx, it_cnt, FL2FXCONST_DBL(0.4f)); FDKaacEnc_initElement(&cm->elInfo[1], ID_CPE, &count, &mapDescr, mapIdx, it_cnt, FL2FXCONST_DBL(0.6f)); break; case MODE_1_2_1: /* sce + cpe + sce */ FDKaacEnc_initElement(&cm->elInfo[0], ID_SCE, &count, &mapDescr, mapIdx, it_cnt, FL2FXCONST_DBL(0.3f)); FDKaacEnc_initElement(&cm->elInfo[1], ID_CPE, &count, &mapDescr, mapIdx, it_cnt, FL2FXCONST_DBL(0.4f)); FDKaacEnc_initElement(&cm->elInfo[2], ID_SCE, &count, &mapDescr, mapIdx, it_cnt, FL2FXCONST_DBL(0.3f)); break; case MODE_1_2_2: /* sce + cpe + cpe */ FDKaacEnc_initElement(&cm->elInfo[0], ID_SCE, &count, &mapDescr, mapIdx, it_cnt, FL2FXCONST_DBL(0.26f)); FDKaacEnc_initElement(&cm->elInfo[1], ID_CPE, &count, &mapDescr, mapIdx, it_cnt, FL2FXCONST_DBL(0.37f)); FDKaacEnc_initElement(&cm->elInfo[2], ID_CPE, &count, &mapDescr, mapIdx, it_cnt, FL2FXCONST_DBL(0.37f)); break; case MODE_1_2_2_1: /* (5.1) sce + cpe + cpe + lfe */ FDKaacEnc_initElement(&cm->elInfo[0], ID_SCE, &count, &mapDescr, mapIdx, it_cnt, FL2FXCONST_DBL(0.24f)); FDKaacEnc_initElement(&cm->elInfo[1], ID_CPE, &count, &mapDescr, mapIdx, it_cnt, FL2FXCONST_DBL(0.35f)); FDKaacEnc_initElement(&cm->elInfo[2], ID_CPE, &count, &mapDescr, mapIdx, it_cnt, FL2FXCONST_DBL(0.35f)); FDKaacEnc_initElement(&cm->elInfo[3], ID_LFE, &count, &mapDescr, mapIdx, it_cnt, FL2FXCONST_DBL(0.06f)); break; case MODE_6_1: /* (6.1) sce + cpe + cpe + sce + lfe */ FDKaacEnc_initElement(&cm->elInfo[0], ID_SCE, &count, &mapDescr, mapIdx, it_cnt, FL2FXCONST_DBL(0.2f)); FDKaacEnc_initElement(&cm->elInfo[1], ID_CPE, &count, &mapDescr, mapIdx, it_cnt, FL2FXCONST_DBL(0.275f)); FDKaacEnc_initElement(&cm->elInfo[2], ID_CPE, &count, &mapDescr, mapIdx, it_cnt, FL2FXCONST_DBL(0.275f)); FDKaacEnc_initElement(&cm->elInfo[3], ID_SCE, &count, &mapDescr, mapIdx, it_cnt, FL2FXCONST_DBL(0.2f)); FDKaacEnc_initElement(&cm->elInfo[4], ID_LFE, &count, &mapDescr, mapIdx, it_cnt, FL2FXCONST_DBL(0.05f)); break; case MODE_1_2_2_2_1: case MODE_7_1_BACK: case MODE_7_1_TOP_FRONT: case MODE_7_1_REAR_SURROUND: case MODE_7_1_FRONT_CENTER: { /* (7.1) sce + cpe + cpe + cpe + lfe */ /* (7.1 top) sce + cpe + cpe + lfe + cpe */ FDKaacEnc_initElement(&cm->elInfo[0], ID_SCE, &count, &mapDescr, mapIdx, it_cnt, FL2FXCONST_DBL(0.18f)); FDKaacEnc_initElement(&cm->elInfo[1], ID_CPE, &count, &mapDescr, mapIdx, it_cnt, FL2FXCONST_DBL(0.26f)); FDKaacEnc_initElement(&cm->elInfo[2], ID_CPE, &count, &mapDescr, mapIdx, it_cnt, FL2FXCONST_DBL(0.26f)); if (mode != MODE_7_1_TOP_FRONT) { FDKaacEnc_initElement(&cm->elInfo[3], ID_CPE, &count, &mapDescr, mapIdx, it_cnt, FL2FXCONST_DBL(0.26f)); FDKaacEnc_initElement(&cm->elInfo[4], ID_LFE, &count, &mapDescr, mapIdx, it_cnt, FL2FXCONST_DBL(0.04f)); } else { FDKaacEnc_initElement(&cm->elInfo[3], ID_LFE, &count, &mapDescr, mapIdx, it_cnt, FL2FXCONST_DBL(0.04f)); FDKaacEnc_initElement(&cm->elInfo[4], ID_CPE, &count, &mapDescr, mapIdx, it_cnt, FL2FXCONST_DBL(0.26f)); } break; } default: //*chMap=0; return AAC_ENC_UNSUPPORTED_CHANNELCONFIG; }; FDK_ASSERT(cm->nElements <= ((8))); return AAC_ENC_OK; } AAC_ENCODER_ERROR FDKaacEnc_InitElementBits(QC_STATE* hQC, CHANNEL_MAPPING* cm, INT bitrateTot, INT averageBitsTot, INT maxChannelBits) { int sc_brTot = CountLeadingBits(bitrateTot); switch (cm->encMode) { case MODE_1: hQC->elementBits[0]->chBitrateEl = bitrateTot; hQC->elementBits[0]->maxBitsEl = maxChannelBits; hQC->elementBits[0]->relativeBitsEl = cm->elInfo[0].relativeBits; break; case MODE_2: hQC->elementBits[0]->chBitrateEl = bitrateTot >> 1; hQC->elementBits[0]->maxBitsEl = 2 * maxChannelBits; hQC->elementBits[0]->relativeBitsEl = cm->elInfo[0].relativeBits; break; case MODE_1_2: { hQC->elementBits[0]->relativeBitsEl = cm->elInfo[0].relativeBits; hQC->elementBits[1]->relativeBitsEl = cm->elInfo[1].relativeBits; FIXP_DBL sceRate = cm->elInfo[0].relativeBits; FIXP_DBL cpeRate = cm->elInfo[1].relativeBits; hQC->elementBits[0]->chBitrateEl = fMult(sceRate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> sc_brTot; hQC->elementBits[1]->chBitrateEl = fMult(cpeRate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> (sc_brTot + 1); hQC->elementBits[0]->maxBitsEl = maxChannelBits; hQC->elementBits[1]->maxBitsEl = 2 * maxChannelBits; break; } case MODE_1_2_1: { /* sce + cpe + sce */ hQC->elementBits[0]->relativeBitsEl = cm->elInfo[0].relativeBits; hQC->elementBits[1]->relativeBitsEl = cm->elInfo[1].relativeBits; hQC->elementBits[2]->relativeBitsEl = cm->elInfo[2].relativeBits; FIXP_DBL sce1Rate = cm->elInfo[0].relativeBits; FIXP_DBL cpeRate = cm->elInfo[1].relativeBits; FIXP_DBL sce2Rate = cm->elInfo[2].relativeBits; hQC->elementBits[0]->chBitrateEl = fMult(sce1Rate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> sc_brTot; hQC->elementBits[1]->chBitrateEl = fMult(cpeRate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> (sc_brTot + 1); hQC->elementBits[2]->chBitrateEl = fMult(sce2Rate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> sc_brTot; hQC->elementBits[0]->maxBitsEl = maxChannelBits; hQC->elementBits[1]->maxBitsEl = 2 * maxChannelBits; hQC->elementBits[2]->maxBitsEl = maxChannelBits; break; } case MODE_1_2_2: { /* sce + cpe + cpe */ hQC->elementBits[0]->relativeBitsEl = cm->elInfo[0].relativeBits; hQC->elementBits[1]->relativeBitsEl = cm->elInfo[1].relativeBits; hQC->elementBits[2]->relativeBitsEl = cm->elInfo[2].relativeBits; FIXP_DBL sceRate = cm->elInfo[0].relativeBits; FIXP_DBL cpe1Rate = cm->elInfo[1].relativeBits; FIXP_DBL cpe2Rate = cm->elInfo[2].relativeBits; hQC->elementBits[0]->chBitrateEl = fMult(sceRate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> sc_brTot; hQC->elementBits[1]->chBitrateEl = fMult(cpe1Rate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> (sc_brTot + 1); hQC->elementBits[2]->chBitrateEl = fMult(cpe2Rate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> (sc_brTot + 1); hQC->elementBits[0]->maxBitsEl = maxChannelBits; hQC->elementBits[1]->maxBitsEl = 2 * maxChannelBits; hQC->elementBits[2]->maxBitsEl = 2 * maxChannelBits; break; } case MODE_1_2_2_1: { /* (5.1) sce + cpe + cpe + lfe */ hQC->elementBits[0]->relativeBitsEl = cm->elInfo[0].relativeBits; hQC->elementBits[1]->relativeBitsEl = cm->elInfo[1].relativeBits; hQC->elementBits[2]->relativeBitsEl = cm->elInfo[2].relativeBits; hQC->elementBits[3]->relativeBitsEl = cm->elInfo[3].relativeBits; FIXP_DBL sceRate = cm->elInfo[0].relativeBits; FIXP_DBL cpe1Rate = cm->elInfo[1].relativeBits; FIXP_DBL cpe2Rate = cm->elInfo[2].relativeBits; FIXP_DBL lfeRate = cm->elInfo[3].relativeBits; int maxBitsTot = maxChannelBits * 5; /* LFE does not add to bit reservoir */ int sc = CountLeadingBits(fixMax(maxChannelBits, averageBitsTot)); int maxLfeBits = (int)fMax( (INT)((fMult(lfeRate, (FIXP_DBL)(maxChannelBits << sc)) >> sc) << 1), (INT)((fMult(FL2FXCONST_DBL(1.1f / 2.f), fMult(lfeRate, (FIXP_DBL)(averageBitsTot << sc))) << 1) >> sc)); maxChannelBits = (maxBitsTot - maxLfeBits); sc = CountLeadingBits(maxChannelBits); maxChannelBits = fMult((FIXP_DBL)maxChannelBits << sc, GetInvInt(5)) >> sc; hQC->elementBits[0]->chBitrateEl = fMult(sceRate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> sc_brTot; hQC->elementBits[1]->chBitrateEl = fMult(cpe1Rate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> (sc_brTot + 1); hQC->elementBits[2]->chBitrateEl = fMult(cpe2Rate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> (sc_brTot + 1); hQC->elementBits[3]->chBitrateEl = fMult(lfeRate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> sc_brTot; hQC->elementBits[0]->maxBitsEl = maxChannelBits; hQC->elementBits[1]->maxBitsEl = 2 * maxChannelBits; hQC->elementBits[2]->maxBitsEl = 2 * maxChannelBits; hQC->elementBits[3]->maxBitsEl = maxLfeBits; break; } case MODE_6_1: { /* (6.1) sce + cpe + cpe + sce + lfe */ FIXP_DBL sceRate = hQC->elementBits[0]->relativeBitsEl = cm->elInfo[0].relativeBits; FIXP_DBL cpe1Rate = hQC->elementBits[1]->relativeBitsEl = cm->elInfo[1].relativeBits; FIXP_DBL cpe2Rate = hQC->elementBits[2]->relativeBitsEl = cm->elInfo[2].relativeBits; FIXP_DBL sce2Rate = hQC->elementBits[3]->relativeBitsEl = cm->elInfo[3].relativeBits; FIXP_DBL lfeRate = hQC->elementBits[4]->relativeBitsEl = cm->elInfo[4].relativeBits; int maxBitsTot = maxChannelBits * 6; /* LFE does not add to bit reservoir */ int sc = CountLeadingBits(fixMax(maxChannelBits, averageBitsTot)); int maxLfeBits = (int)fMax( (INT)((fMult(lfeRate, (FIXP_DBL)(maxChannelBits << sc)) >> sc) << 1), (INT)((fMult(FL2FXCONST_DBL(1.1f / 2.f), fMult(lfeRate, (FIXP_DBL)(averageBitsTot << sc))) << 1) >> sc)); maxChannelBits = (maxBitsTot - maxLfeBits) / 6; hQC->elementBits[0]->chBitrateEl = fMult(sceRate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> sc_brTot; hQC->elementBits[1]->chBitrateEl = fMult(cpe1Rate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> (sc_brTot + 1); hQC->elementBits[2]->chBitrateEl = fMult(cpe2Rate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> (sc_brTot + 1); hQC->elementBits[3]->chBitrateEl = fMult(sce2Rate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> (sc_brTot + 1); hQC->elementBits[4]->chBitrateEl = fMult(lfeRate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> sc_brTot; hQC->elementBits[0]->maxBitsEl = maxChannelBits; hQC->elementBits[1]->maxBitsEl = 2 * maxChannelBits; hQC->elementBits[2]->maxBitsEl = 2 * maxChannelBits; hQC->elementBits[3]->maxBitsEl = maxChannelBits; hQC->elementBits[4]->maxBitsEl = maxLfeBits; break; } case MODE_7_1_TOP_FRONT: case MODE_7_1_BACK: case MODE_7_1_REAR_SURROUND: case MODE_7_1_FRONT_CENTER: case MODE_1_2_2_2_1: { int cpe3Idx = (cm->encMode != MODE_7_1_TOP_FRONT) ? 3 : 4; int lfeIdx = (cm->encMode != MODE_7_1_TOP_FRONT) ? 4 : 3; /* (7.1) sce + cpe + cpe + cpe + lfe */ FIXP_DBL sceRate = hQC->elementBits[0]->relativeBitsEl = cm->elInfo[0].relativeBits; FIXP_DBL cpe1Rate = hQC->elementBits[1]->relativeBitsEl = cm->elInfo[1].relativeBits; FIXP_DBL cpe2Rate = hQC->elementBits[2]->relativeBitsEl = cm->elInfo[2].relativeBits; FIXP_DBL cpe3Rate = hQC->elementBits[cpe3Idx]->relativeBitsEl = cm->elInfo[cpe3Idx].relativeBits; FIXP_DBL lfeRate = hQC->elementBits[lfeIdx]->relativeBitsEl = cm->elInfo[lfeIdx].relativeBits; int maxBitsTot = maxChannelBits * 7; /* LFE does not add to bit reservoir */ int sc = CountLeadingBits(fixMax(maxChannelBits, averageBitsTot)); int maxLfeBits = (int)fMax( (INT)((fMult(lfeRate, (FIXP_DBL)(maxChannelBits << sc)) >> sc) << 1), (INT)((fMult(FL2FXCONST_DBL(1.1f / 2.f), fMult(lfeRate, (FIXP_DBL)(averageBitsTot << sc))) << 1) >> sc)); maxChannelBits = (maxBitsTot - maxLfeBits) / 7; hQC->elementBits[0]->chBitrateEl = fMult(sceRate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> sc_brTot; hQC->elementBits[1]->chBitrateEl = fMult(cpe1Rate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> (sc_brTot + 1); hQC->elementBits[2]->chBitrateEl = fMult(cpe2Rate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> (sc_brTot + 1); hQC->elementBits[cpe3Idx]->chBitrateEl = fMult(cpe3Rate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> (sc_brTot + 1); hQC->elementBits[lfeIdx]->chBitrateEl = fMult(lfeRate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> sc_brTot; hQC->elementBits[0]->maxBitsEl = maxChannelBits; hQC->elementBits[1]->maxBitsEl = 2 * maxChannelBits; hQC->elementBits[2]->maxBitsEl = 2 * maxChannelBits; hQC->elementBits[cpe3Idx]->maxBitsEl = 2 * maxChannelBits; hQC->elementBits[lfeIdx]->maxBitsEl = maxLfeBits; break; } default: return AAC_ENC_UNSUPPORTED_CHANNELCONFIG; } return AAC_ENC_OK; } /********************************************************************************/ /* */ /* function: GetMonoStereoMODE(const CHANNEL_MODE mode) */ /* */ /* description: Determines encoder setting from channel mode. */ /* Multichannel modes are mapped to mono or stereo modes */ /* returns MODE_MONO in case of mono, */ /* MODE_STEREO in case of stereo */ /* MODE_INVALID in case of error */ /* */ /* input: CHANNEL_MODE mode: Encoder mode (see qc_data.h). */ /* output: return: CM_STEREO_MODE monoStereoSetting */ /* (MODE_INVALID: error, */ /* MODE_MONO: mono */ /* MODE_STEREO: stereo). */ /* */ /* misc: No memory is allocated. */ /* */ /********************************************************************************/ ELEMENT_MODE FDKaacEnc_GetMonoStereoMode(const CHANNEL_MODE mode) { ELEMENT_MODE monoStereoSetting = EL_MODE_INVALID; switch (mode) { case MODE_1: /* mono setups */ monoStereoSetting = EL_MODE_MONO; break; case MODE_2: /* stereo setups */ case MODE_1_2: case MODE_1_2_1: case MODE_1_2_2: case MODE_1_2_2_1: case MODE_6_1: case MODE_1_2_2_2_1: case MODE_7_1_REAR_SURROUND: case MODE_7_1_FRONT_CENTER: case MODE_7_1_BACK: case MODE_7_1_TOP_FRONT: monoStereoSetting = EL_MODE_STEREO; break; default: /* error */ monoStereoSetting = EL_MODE_INVALID; break; } return monoStereoSetting; } const CHANNEL_MODE_CONFIG_TAB* FDKaacEnc_GetChannelModeConfiguration( const CHANNEL_MODE mode) { INT i; const CHANNEL_MODE_CONFIG_TAB* cm_config = NULL; /* get channel mode config */ for (i = 0; i < (INT)sizeof(channelModeConfig) / (INT)sizeof(CHANNEL_MODE_CONFIG_TAB); i++) { if (channelModeConfig[i].encMode == mode) { cm_config = &channelModeConfig[i]; break; } } return cm_config; } fdk-aac-2.0.2/libAACenc/src/channel_map.h000066400000000000000000000134711404225310700177530ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): A. Groeschel Description: channel mapping functionality *******************************************************************************/ #ifndef CHANNEL_MAP_H #define CHANNEL_MAP_H #include "aacenc.h" #include "psy_const.h" #include "qc_data.h" typedef struct { CHANNEL_MODE encMode; INT nChannels; INT nChannelsEff; INT nElements; } CHANNEL_MODE_CONFIG_TAB; /* Element mode */ typedef enum { EL_MODE_INVALID = 0, EL_MODE_MONO, EL_MODE_STEREO } ELEMENT_MODE; AAC_ENCODER_ERROR FDKaacEnc_DetermineEncoderMode(CHANNEL_MODE* mode, INT nChannels); AAC_ENCODER_ERROR FDKaacEnc_InitChannelMapping(CHANNEL_MODE mode, CHANNEL_ORDER co, CHANNEL_MAPPING* chMap); AAC_ENCODER_ERROR FDKaacEnc_InitElementBits(QC_STATE* hQC, CHANNEL_MAPPING* cm, INT bitrateTot, INT averageBitsTot, INT maxChannelBits); ELEMENT_MODE FDKaacEnc_GetMonoStereoMode(const CHANNEL_MODE mode); const CHANNEL_MODE_CONFIG_TAB* FDKaacEnc_GetChannelModeConfiguration( const CHANNEL_MODE mode); #endif /* CHANNEL_MAP_H */ fdk-aac-2.0.2/libAACenc/src/chaosmeasure.cpp000066400000000000000000000203261404225310700205150ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Werner Description: Chaos measure calculation *******************************************************************************/ #include "chaosmeasure.h" /***************************************************************************** functionname: FDKaacEnc_FDKaacEnc_CalculateChaosMeasurePeakFast description: Eberlein method of chaos measure calculation by high-pass filtering amplitude spectrum A special case of FDKaacEnc_CalculateChaosMeasureTonalGeneric -- highly optimized *****************************************************************************/ static void FDKaacEnc_FDKaacEnc_CalculateChaosMeasurePeakFast( FIXP_DBL *RESTRICT paMDCTDataNM0, INT numberOfLines, FIXP_DBL *RESTRICT chaosMeasure) { INT i, j; /* calculate chaos measure by "peak filter" */ /* make even and odd pass through data */ FIXP_DBL left_0_div2, center_0; /* left, center tap of filter, even numbered */ FIXP_DBL left_1_div2, center_1; /* left, center tap of filter, odd numbered */ left_0_div2 = (FIXP_DBL)(((LONG)paMDCTDataNM0[0] ^ ((LONG)paMDCTDataNM0[0] >> (DFRACT_BITS - 1))) >> 1); left_1_div2 = (FIXP_DBL)(((LONG)paMDCTDataNM0[1] ^ ((LONG)paMDCTDataNM0[1] >> (DFRACT_BITS - 1))) >> 1); center_0 = (FIXP_DBL)((LONG)paMDCTDataNM0[2] ^ ((LONG)paMDCTDataNM0[2] >> (DFRACT_BITS - 1))); center_1 = (FIXP_DBL)((LONG)paMDCTDataNM0[3] ^ ((LONG)paMDCTDataNM0[3] >> (DFRACT_BITS - 1))); for (j = 2; j < numberOfLines - 2; j += 2) { FIXP_DBL right_0 = (FIXP_DBL)((LONG)paMDCTDataNM0[j + 2] ^ ((LONG)paMDCTDataNM0[j + 2] >> (DFRACT_BITS - 1))); FIXP_DBL tmp_0 = left_0_div2 + (right_0 >> 1); FIXP_DBL right_1 = (FIXP_DBL)((LONG)paMDCTDataNM0[j + 3] ^ ((LONG)paMDCTDataNM0[j + 3] >> (DFRACT_BITS - 1))); FIXP_DBL tmp_1 = left_1_div2 + (right_1 >> 1); if (tmp_0 < center_0) { INT leadingBits = CntLeadingZeros(center_0) - 1; tmp_0 = schur_div(tmp_0 << leadingBits, center_0 << leadingBits, 8); tmp_0 = fMult(tmp_0, tmp_0); } else { tmp_0 = (FIXP_DBL)MAXVAL_DBL; } chaosMeasure[j + 0] = tmp_0; left_0_div2 = center_0 >> 1; center_0 = right_0; if (tmp_1 < center_1) { INT leadingBits = CntLeadingZeros(center_1) - 1; tmp_1 = schur_div(tmp_1 << leadingBits, center_1 << leadingBits, 8); tmp_1 = fMult(tmp_1, tmp_1); } else { tmp_1 = (FIXP_DBL)MAXVAL_DBL; } left_1_div2 = center_1 >> 1; center_1 = right_1; chaosMeasure[j + 1] = tmp_1; } /* provide chaos measure for first few lines */ chaosMeasure[0] = chaosMeasure[2]; chaosMeasure[1] = chaosMeasure[2]; /* provide chaos measure for last few lines */ for (i = (numberOfLines - 3); i < numberOfLines; i++) chaosMeasure[i] = FL2FXCONST_DBL(0.5); } /***************************************************************************** functionname: FDKaacEnc_CalculateChaosMeasure description: calculates a chaosmeasure for every line, different methods are available. 0 means tonal, 1 means noiselike returns: input: MDCT data, number of lines output: chaosMeasure *****************************************************************************/ void FDKaacEnc_CalculateChaosMeasure(FIXP_DBL *paMDCTDataNM0, INT numberOfLines, FIXP_DBL *chaosMeasure) { FDKaacEnc_FDKaacEnc_CalculateChaosMeasurePeakFast( paMDCTDataNM0, numberOfLines, chaosMeasure); } fdk-aac-2.0.2/libAACenc/src/chaosmeasure.h000066400000000000000000000117661404225310700201720ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Werner Description: Chaos measure calculation *******************************************************************************/ #ifndef CHAOSMEASURE_H #define CHAOSMEASURE_H #include "common_fix.h" #include "psy_const.h" void FDKaacEnc_CalculateChaosMeasure(FIXP_DBL *paMDCTDataNM0, INT numberOfLines, FIXP_DBL *chaosMeasure); #endif /* CHAOSMEASURE_H */ fdk-aac-2.0.2/libAACenc/src/dyn_bits.cpp000066400000000000000000000606221404225310700176540ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M.Werner Description: Noiseless coder module *******************************************************************************/ #include "dyn_bits.h" #include "bit_cnt.h" #include "psy_const.h" #include "aacenc_pns.h" #include "aacEnc_ram.h" #include "aacEnc_rom.h" typedef INT (*lookUpTable)[CODE_BOOK_ESC_NDX + 1]; static INT FDKaacEnc_getSideInfoBits(const SECTION_INFO* const huffsection, const SHORT* const sideInfoTab, const INT useHCR) { INT sideInfoBits; if (useHCR && ((huffsection->codeBook == 11) || (huffsection->codeBook >= 16))) { sideInfoBits = 5; } else { sideInfoBits = sideInfoTab[huffsection->sfbCnt]; } return (sideInfoBits); } /* count bits using all possible tables */ static void FDKaacEnc_buildBitLookUp( const SHORT* const quantSpectrum, const INT maxSfb, const INT* const sfbOffset, const UINT* const sfbMax, INT bitLookUp[MAX_SFB_LONG][CODE_BOOK_ESC_NDX + 1], SECTION_INFO* const huffsection) { INT i, sfbWidth; for (i = 0; i < maxSfb; i++) { huffsection[i].sfbCnt = 1; huffsection[i].sfbStart = i; huffsection[i].sectionBits = INVALID_BITCOUNT; huffsection[i].codeBook = -1; sfbWidth = sfbOffset[i + 1] - sfbOffset[i]; FDKaacEnc_bitCount(quantSpectrum + sfbOffset[i], sfbWidth, sfbMax[i], bitLookUp[i]); } } /* essential helper functions */ static inline INT FDKaacEnc_findBestBook(const INT* const bc, INT* const book, const INT useVCB11) { INT minBits = INVALID_BITCOUNT, j; int end = CODE_BOOK_ESC_NDX; for (j = 0; j <= end; j++) { if (bc[j] < minBits) { minBits = bc[j]; *book = j; } } return (minBits); } static inline INT FDKaacEnc_findMinMergeBits(const INT* const bc1, const INT* const bc2, const INT useVCB11) { INT minBits = INVALID_BITCOUNT, j; DWORD_ALIGNED(bc1); DWORD_ALIGNED(bc2); for (j = 0; j <= CODE_BOOK_ESC_NDX; j++) { minBits = fixMin(minBits, bc1[j] + bc2[j]); } return (minBits); } static inline void FDKaacEnc_mergeBitLookUp(INT* const RESTRICT bc1, const INT* const RESTRICT bc2) { int j; for (j = 0; j <= CODE_BOOK_ESC_NDX; j++) { FDK_ASSERT(INVALID_BITCOUNT == 0x1FFFFFFF); bc1[j] = fixMin(bc1[j] + bc2[j], INVALID_BITCOUNT); } } static inline INT FDKaacEnc_findMaxMerge(const INT* const mergeGainLookUp, const SECTION_INFO* const huffsection, const INT maxSfb, INT* const maxNdx) { INT i, maxMergeGain = 0; int lastMaxNdx = 0; for (i = 0; i + huffsection[i].sfbCnt < maxSfb; i += huffsection[i].sfbCnt) { if (mergeGainLookUp[i] > maxMergeGain) { maxMergeGain = mergeGainLookUp[i]; lastMaxNdx = i; } } *maxNdx = lastMaxNdx; return (maxMergeGain); } static inline INT FDKaacEnc_CalcMergeGain( const SECTION_INFO* const huffsection, const INT bitLookUp[MAX_SFB_LONG][CODE_BOOK_ESC_NDX + 1], const SHORT* const sideInfoTab, const INT ndx1, const INT ndx2, const INT useVCB11) { INT MergeGain, MergeBits, SplitBits; MergeBits = sideInfoTab[huffsection[ndx1].sfbCnt + huffsection[ndx2].sfbCnt] + FDKaacEnc_findMinMergeBits(bitLookUp[ndx1], bitLookUp[ndx2], useVCB11); SplitBits = huffsection[ndx1].sectionBits + huffsection[ndx2].sectionBits; /* Bit amount for splitted huffsections */ MergeGain = SplitBits - MergeBits; if ((huffsection[ndx1].codeBook == CODE_BOOK_PNS_NO) || (huffsection[ndx2].codeBook == CODE_BOOK_PNS_NO) || (huffsection[ndx1].codeBook == CODE_BOOK_IS_OUT_OF_PHASE_NO) || (huffsection[ndx2].codeBook == CODE_BOOK_IS_OUT_OF_PHASE_NO) || (huffsection[ndx1].codeBook == CODE_BOOK_IS_IN_PHASE_NO) || (huffsection[ndx2].codeBook == CODE_BOOK_IS_IN_PHASE_NO)) { MergeGain = -1; } return (MergeGain); } /* sectioning Stage 0:find minimum codbooks */ static void FDKaacEnc_gmStage0( SECTION_INFO* const RESTRICT huffsection, const INT bitLookUp[MAX_SFB_LONG][CODE_BOOK_ESC_NDX + 1], const INT maxSfb, const INT* const noiseNrg, const INT* const isBook) { INT i; for (i = 0; i < maxSfb; i++) { /* Side-Info bits will be calculated in Stage 1! */ if (huffsection[i].sectionBits == INVALID_BITCOUNT) { /* intensity and pns codebooks are already allocated in bitcount.c */ if (noiseNrg[i] != NO_NOISE_PNS) { huffsection[i].codeBook = CODE_BOOK_PNS_NO; huffsection[i].sectionBits = 0; } else if (isBook[i]) { huffsection[i].codeBook = isBook[i]; huffsection[i].sectionBits = 0; } else { huffsection[i].sectionBits = FDKaacEnc_findBestBook(bitLookUp[i], &(huffsection[i].codeBook), 0); /* useVCB11 must be 0!!! */ } } } } /* sectioning Stage 1:merge all connected regions with the same code book and calculate side info */ static void FDKaacEnc_gmStage1( SECTION_INFO* const RESTRICT huffsection, INT bitLookUp[MAX_SFB_LONG][CODE_BOOK_ESC_NDX + 1], const INT maxSfb, const SHORT* const sideInfoTab, const INT useVCB11) { INT mergeStart = 0, mergeEnd; do { for (mergeEnd = mergeStart + 1; mergeEnd < maxSfb; mergeEnd++) { if (huffsection[mergeStart].codeBook != huffsection[mergeEnd].codeBook) break; /* we can merge. update tables, side info bits will be updated outside of * this loop */ huffsection[mergeStart].sfbCnt++; huffsection[mergeStart].sectionBits += huffsection[mergeEnd].sectionBits; /* update bit look up for all code books */ FDKaacEnc_mergeBitLookUp(bitLookUp[mergeStart], bitLookUp[mergeEnd]); } /* add side info info bits */ huffsection[mergeStart].sectionBits += FDKaacEnc_getSideInfoBits( &huffsection[mergeStart], sideInfoTab, useVCB11); huffsection[mergeEnd - 1].sfbStart = huffsection[mergeStart].sfbStart; /* speed up prev search */ mergeStart = mergeEnd; } while (mergeStart < maxSfb); } /* sectioning Stage 2:greedy merge algorithm, merge connected sections with maximum bit gain until no more gain is possible */ static inline void FDKaacEnc_gmStage2( SECTION_INFO* const RESTRICT huffsection, INT* const RESTRICT mergeGainLookUp, INT bitLookUp[MAX_SFB_LONG][CODE_BOOK_ESC_NDX + 1], const INT maxSfb, const SHORT* const sideInfoTab, const INT useVCB11) { INT i; for (i = 0; i + huffsection[i].sfbCnt < maxSfb; i += huffsection[i].sfbCnt) { mergeGainLookUp[i] = FDKaacEnc_CalcMergeGain(huffsection, bitLookUp, sideInfoTab, i, i + huffsection[i].sfbCnt, useVCB11); } while (TRUE) { INT maxMergeGain, maxNdx, maxNdxNext, maxNdxLast; maxMergeGain = FDKaacEnc_findMaxMerge(mergeGainLookUp, huffsection, maxSfb, &maxNdx); /* exit while loop if no more gain is possible */ if (maxMergeGain <= 0) break; maxNdxNext = maxNdx + huffsection[maxNdx].sfbCnt; /* merge sections with maximum bit gain */ huffsection[maxNdx].sfbCnt += huffsection[maxNdxNext].sfbCnt; huffsection[maxNdx].sectionBits += huffsection[maxNdxNext].sectionBits - maxMergeGain; /* update bit look up table for merged huffsection */ FDKaacEnc_mergeBitLookUp(bitLookUp[maxNdx], bitLookUp[maxNdxNext]); /* update mergeLookUpTable */ if (maxNdx != 0) { maxNdxLast = huffsection[maxNdx - 1].sfbStart; mergeGainLookUp[maxNdxLast] = FDKaacEnc_CalcMergeGain( huffsection, bitLookUp, sideInfoTab, maxNdxLast, maxNdx, useVCB11); } maxNdxNext = maxNdx + huffsection[maxNdx].sfbCnt; huffsection[maxNdxNext - 1].sfbStart = huffsection[maxNdx].sfbStart; if (maxNdxNext < maxSfb) mergeGainLookUp[maxNdx] = FDKaacEnc_CalcMergeGain( huffsection, bitLookUp, sideInfoTab, maxNdx, maxNdxNext, useVCB11); } } /* count bits used by the noiseless coder */ static void FDKaacEnc_noiselessCounter( SECTION_DATA* const RESTRICT sectionData, INT mergeGainLookUp[MAX_SFB_LONG], INT bitLookUp[MAX_SFB_LONG][CODE_BOOK_ESC_NDX + 1], const SHORT* const quantSpectrum, const UINT* const maxValueInSfb, const INT* const sfbOffset, const INT blockType, const INT* const noiseNrg, const INT* const isBook, const INT useVCB11) { INT grpNdx, i; const SHORT* sideInfoTab = NULL; SECTION_INFO* huffsection; /* use appropriate side info table */ switch (blockType) { case LONG_WINDOW: case START_WINDOW: case STOP_WINDOW: default: sideInfoTab = FDKaacEnc_sideInfoTabLong; break; case SHORT_WINDOW: sideInfoTab = FDKaacEnc_sideInfoTabShort; break; } sectionData->noOfSections = 0; sectionData->huffmanBits = 0; sectionData->sideInfoBits = 0; if (sectionData->maxSfbPerGroup == 0) return; /* loop trough groups */ for (grpNdx = 0; grpNdx < sectionData->sfbCnt; grpNdx += sectionData->sfbPerGroup) { huffsection = sectionData->huffsection + sectionData->noOfSections; /* count bits in this group */ FDKaacEnc_buildBitLookUp(quantSpectrum, sectionData->maxSfbPerGroup, sfbOffset + grpNdx, maxValueInSfb + grpNdx, bitLookUp, huffsection); /* 0.Stage :Find minimum Codebooks */ FDKaacEnc_gmStage0(huffsection, bitLookUp, sectionData->maxSfbPerGroup, noiseNrg + grpNdx, isBook + grpNdx); /* 1.Stage :Merge all connected regions with the same code book */ FDKaacEnc_gmStage1(huffsection, bitLookUp, sectionData->maxSfbPerGroup, sideInfoTab, useVCB11); /* 2.Stage greedy merge algorithm, merge connected huffsections with maximum bit gain until no more gain is possible */ FDKaacEnc_gmStage2(huffsection, mergeGainLookUp, bitLookUp, sectionData->maxSfbPerGroup, sideInfoTab, useVCB11); /* compress output, calculate total huff and side bits since we did not update the actual codebook in stage 2 to save time, we must set it here for later use in bitenc */ for (i = 0; i < sectionData->maxSfbPerGroup; i += huffsection[i].sfbCnt) { if ((huffsection[i].codeBook == CODE_BOOK_PNS_NO) || (huffsection[i].codeBook == CODE_BOOK_IS_OUT_OF_PHASE_NO) || (huffsection[i].codeBook == CODE_BOOK_IS_IN_PHASE_NO)) { huffsection[i].sectionBits = 0; } else { /* the sections in the sectionData are now marked with the optimal code * book */ FDKaacEnc_findBestBook(bitLookUp[i], &(huffsection[i].codeBook), useVCB11); sectionData->huffmanBits += huffsection[i].sectionBits - FDKaacEnc_getSideInfoBits(&huffsection[i], sideInfoTab, useVCB11); } huffsection[i].sfbStart += grpNdx; /* sum up side info bits (section data bits) */ sectionData->sideInfoBits += FDKaacEnc_getSideInfoBits(&huffsection[i], sideInfoTab, useVCB11); sectionData->huffsection[sectionData->noOfSections++] = huffsection[i]; } } } /******************************************************************************* functionname: FDKaacEnc_scfCount returns : --- description : count bits used by scalefactors. not in all cases if maxValueInSfb[] == 0 we set deltaScf to zero. only if the difference of the last and future scalefacGain is not greater then CODE_BOOK_SCF_LAV (60). example: ^ scalefacGain | | | last 75 | | | | | | | | current 50 | | | | | | | | | | | | | | | future 5 | | | | --- ... ---------------------------- ... ---------> sfb if maxValueInSfb[] of current is zero because of a notfallstrategie, we do not save bits and transmit a deltaScf of 25. otherwise the deltaScf between the last scalfacGain (75) and the future scalefacGain (5) is 70. ********************************************************************************/ static void FDKaacEnc_scfCount(const INT* const scalefacGain, const UINT* const maxValueInSfb, SECTION_DATA* const RESTRICT sectionData, const INT* const isScale) { INT i, j, k, m, n; INT lastValScf = 0; INT deltaScf = 0; INT found = 0; INT scfSkipCounter = 0; INT lastValIs = 0; sectionData->scalefacBits = 0; if (scalefacGain == NULL) return; sectionData->firstScf = 0; for (i = 0; i < sectionData->noOfSections; i++) { if (sectionData->huffsection[i].codeBook != CODE_BOOK_ZERO_NO) { sectionData->firstScf = sectionData->huffsection[i].sfbStart; lastValScf = scalefacGain[sectionData->firstScf]; break; } } for (i = 0; i < sectionData->noOfSections; i++) { if ((sectionData->huffsection[i].codeBook == CODE_BOOK_IS_OUT_OF_PHASE_NO) || (sectionData->huffsection[i].codeBook == CODE_BOOK_IS_IN_PHASE_NO)) { for (j = sectionData->huffsection[i].sfbStart; j < sectionData->huffsection[i].sfbStart + sectionData->huffsection[i].sfbCnt; j++) { INT deltaIs = isScale[j] - lastValIs; lastValIs = isScale[j]; sectionData->scalefacBits += FDKaacEnc_bitCountScalefactorDelta(deltaIs); } } /* Intensity */ else if ((sectionData->huffsection[i].codeBook != CODE_BOOK_ZERO_NO) && (sectionData->huffsection[i].codeBook != CODE_BOOK_PNS_NO)) { INT tmp = sectionData->huffsection[i].sfbStart + sectionData->huffsection[i].sfbCnt; for (j = sectionData->huffsection[i].sfbStart; j < tmp; j++) { /* check if we can repeat the last value to save bits */ if (maxValueInSfb[j] == 0) { found = 0; /* are scalefactors skipped? */ if (scfSkipCounter == 0) { /* end of section */ if (j == (tmp - 1)) found = 0; /* search in other sections for maxValueInSfb != 0 */ else { /* search in this section for the next maxValueInSfb[] != 0 */ for (k = (j + 1); k < tmp; k++) { if (maxValueInSfb[k] != 0) { found = 1; if ((fixp_abs(scalefacGain[k] - lastValScf)) <= CODE_BOOK_SCF_LAV) deltaScf = 0; /* save bits */ else { /* do not save bits */ deltaScf = lastValScf - scalefacGain[j]; lastValScf = scalefacGain[j]; scfSkipCounter = 0; } break; } /* count scalefactor skip */ scfSkipCounter++; } } /* search for the next maxValueInSfb[] != 0 in all other sections */ for (m = (i + 1); (m < sectionData->noOfSections) && (found == 0); m++) { if ((sectionData->huffsection[m].codeBook != CODE_BOOK_ZERO_NO) && (sectionData->huffsection[m].codeBook != CODE_BOOK_PNS_NO)) { INT end = sectionData->huffsection[m].sfbStart + sectionData->huffsection[m].sfbCnt; for (n = sectionData->huffsection[m].sfbStart; n < end; n++) { if (maxValueInSfb[n] != 0) { found = 1; if (fixp_abs(scalefacGain[n] - lastValScf) <= CODE_BOOK_SCF_LAV) deltaScf = 0; /* save bits */ else { /* do not save bits */ deltaScf = lastValScf - scalefacGain[j]; lastValScf = scalefacGain[j]; scfSkipCounter = 0; } break; } /* count scalefactor skip */ scfSkipCounter++; } } } /* no maxValueInSfb[] != 0 found */ if (found == 0) { deltaScf = 0; scfSkipCounter = 0; } } else { /* consider skipped scalefactors */ deltaScf = 0; scfSkipCounter--; } } else { deltaScf = lastValScf - scalefacGain[j]; lastValScf = scalefacGain[j]; } sectionData->scalefacBits += FDKaacEnc_bitCountScalefactorDelta(deltaScf); } } } /* for (i=0; inoOfSections; i++) */ } /* count bits used by pns */ static void FDKaacEnc_noiseCount(SECTION_DATA* const RESTRICT sectionData, const INT* const noiseNrg) { INT noisePCMFlag = TRUE; INT lastValPns = 0, deltaPns; int i, j; sectionData->noiseNrgBits = 0; for (i = 0; i < sectionData->noOfSections; i++) { if (sectionData->huffsection[i].codeBook == CODE_BOOK_PNS_NO) { int sfbStart = sectionData->huffsection[i].sfbStart; int sfbEnd = sfbStart + sectionData->huffsection[i].sfbCnt; for (j = sfbStart; j < sfbEnd; j++) { if (noisePCMFlag) { sectionData->noiseNrgBits += PNS_PCM_BITS; lastValPns = noiseNrg[j]; noisePCMFlag = FALSE; } else { deltaPns = noiseNrg[j] - lastValPns; lastValPns = noiseNrg[j]; sectionData->noiseNrgBits += FDKaacEnc_bitCountScalefactorDelta(deltaPns); } } } } } INT FDKaacEnc_dynBitCount(BITCNTR_STATE* const hBC, const SHORT* const quantSpectrum, const UINT* const maxValueInSfb, const INT* const scalefac, const INT blockType, const INT sfbCnt, const INT maxSfbPerGroup, const INT sfbPerGroup, const INT* const sfbOffset, SECTION_DATA* const RESTRICT sectionData, const INT* const noiseNrg, const INT* const isBook, const INT* const isScale, const UINT syntaxFlags) { sectionData->blockType = blockType; sectionData->sfbCnt = sfbCnt; sectionData->sfbPerGroup = sfbPerGroup; sectionData->noOfGroups = sfbCnt / sfbPerGroup; sectionData->maxSfbPerGroup = maxSfbPerGroup; FDKaacEnc_noiselessCounter(sectionData, hBC->mergeGainLookUp, (lookUpTable)hBC->bitLookUp, quantSpectrum, maxValueInSfb, sfbOffset, blockType, noiseNrg, isBook, (syntaxFlags & AC_ER_VCB11) ? 1 : 0); FDKaacEnc_scfCount(scalefac, maxValueInSfb, sectionData, isScale); FDKaacEnc_noiseCount(sectionData, noiseNrg); return (sectionData->huffmanBits + sectionData->sideInfoBits + sectionData->scalefacBits + sectionData->noiseNrgBits); } INT FDKaacEnc_BCNew(BITCNTR_STATE** phBC, UCHAR* dynamic_RAM) { BITCNTR_STATE* hBC = GetRam_aacEnc_BitCntrState(); if (hBC) { *phBC = hBC; hBC->bitLookUp = GetRam_aacEnc_BitLookUp(0, dynamic_RAM); hBC->mergeGainLookUp = GetRam_aacEnc_MergeGainLookUp(0, dynamic_RAM); if (hBC->bitLookUp == 0 || hBC->mergeGainLookUp == 0) { return 1; } } return (hBC == 0) ? 1 : 0; } void FDKaacEnc_BCClose(BITCNTR_STATE** phBC) { if (*phBC != NULL) { FreeRam_aacEnc_BitCntrState(phBC); } } fdk-aac-2.0.2/libAACenc/src/dyn_bits.h000066400000000000000000000145351404225310700173230ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M.Werner Description: Noiseless coder module *******************************************************************************/ #ifndef DYN_BITS_H #define DYN_BITS_H #include "common_fix.h" #include "psy_const.h" #include "aacenc_tns.h" #define MAX_SECTIONS MAX_GROUPED_SFB #define SECT_ESC_VAL_LONG 31 #define SECT_ESC_VAL_SHORT 7 #define CODE_BOOK_BITS 4 #define SECT_BITS_LONG 5 #define SECT_BITS_SHORT 3 #define PNS_PCM_BITS 9 typedef struct { INT codeBook; INT sfbStart; INT sfbCnt; INT sectionBits; /* huff + si ! */ } SECTION_INFO; typedef struct { INT blockType; INT noOfGroups; INT sfbCnt; INT maxSfbPerGroup; INT sfbPerGroup; INT noOfSections; SECTION_INFO huffsection[MAX_SECTIONS]; INT sideInfoBits; /* sectioning bits */ INT huffmanBits; /* huffman coded bits */ INT scalefacBits; /* scalefac coded bits */ INT noiseNrgBits; /* noiseEnergy coded bits */ INT firstScf; /* first scf to be coded */ } SECTION_DATA; struct BITCNTR_STATE { INT* bitLookUp; INT* mergeGainLookUp; }; INT FDKaacEnc_BCNew(BITCNTR_STATE** phBC, UCHAR* dynamic_RAM); void FDKaacEnc_BCClose(BITCNTR_STATE** phBC); INT FDKaacEnc_dynBitCount(BITCNTR_STATE* const hBC, const SHORT* const quantSpectrum, const UINT* const maxValueInSfb, const INT* const scalefac, const INT blockType, const INT sfbCnt, const INT maxSfbPerGroup, const INT sfbPerGroup, const INT* const sfbOffset, SECTION_DATA* const RESTRICT sectionData, const INT* const noiseNrg, const INT* const isBook, const INT* const isScale, const UINT syntaxFlags); #endif fdk-aac-2.0.2/libAACenc/src/grp_data.cpp000066400000000000000000000236671404225310700176320ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M.Werner Description: Short block grouping *******************************************************************************/ #include "psy_const.h" #include "interface.h" /* * this routine does not work in-place */ /* * Don't use fAddSaturate2() because it looses one bit accuracy which is * usefull for quality. */ static inline FIXP_DBL nrgAddSaturate(const FIXP_DBL a, const FIXP_DBL b) { return ((a >= (FIXP_DBL)MAXVAL_DBL - b) ? (FIXP_DBL)MAXVAL_DBL : (a + b)); } void FDKaacEnc_groupShortData(FIXP_DBL *mdctSpectrum, /* in-out */ SFB_THRESHOLD *sfbThreshold, /* in-out */ SFB_ENERGY *sfbEnergy, /* in-out */ SFB_ENERGY *sfbEnergyMS, /* in-out */ SFB_ENERGY *sfbSpreadEnergy, const INT sfbCnt, const INT sfbActive, const INT *sfbOffset, const FIXP_DBL *sfbMinSnrLdData, INT *groupedSfbOffset, /* out */ INT *maxSfbPerGroup, /* out */ FIXP_DBL *groupedSfbMinSnrLdData, const INT noOfGroups, const INT *groupLen, const INT granuleLength) { INT i, j; INT line; /* counts through lines */ INT sfb; /* counts through scalefactor bands */ INT grp; /* counts through groups */ INT wnd; /* counts through windows in a group */ INT offset; /* needed in sfbOffset grouping */ INT highestSfb; INT granuleLength_short = granuleLength / TRANS_FAC; C_ALLOC_SCRATCH_START(tmpSpectrum, FIXP_DBL, (1024)) /* for short blocks: regroup spectrum and */ /* group energies and thresholds according to grouping */ /* calculate maxSfbPerGroup */ highestSfb = 0; for (wnd = 0; wnd < TRANS_FAC; wnd++) { for (sfb = sfbActive - 1; sfb >= highestSfb; sfb--) { for (line = sfbOffset[sfb + 1] - 1; line >= sfbOffset[sfb]; line--) { if (mdctSpectrum[wnd * granuleLength_short + line] != FL2FXCONST_SPC(0.0)) break; /* this band is not completely zero */ } if (line >= sfbOffset[sfb]) break; /* this band was not completely zero */ } highestSfb = fixMax(highestSfb, sfb); } highestSfb = highestSfb > 0 ? highestSfb : 0; *maxSfbPerGroup = highestSfb + 1; /* calculate groupedSfbOffset */ i = 0; offset = 0; for (grp = 0; grp < noOfGroups; grp++) { for (sfb = 0; sfb < sfbActive + 1; sfb++) { groupedSfbOffset[i++] = offset + sfbOffset[sfb] * groupLen[grp]; } i += sfbCnt - sfb; offset += groupLen[grp] * granuleLength_short; } groupedSfbOffset[i++] = granuleLength; /* calculate groupedSfbMinSnr */ i = 0; for (grp = 0; grp < noOfGroups; grp++) { for (sfb = 0; sfb < sfbActive; sfb++) { groupedSfbMinSnrLdData[i++] = sfbMinSnrLdData[sfb]; } i += sfbCnt - sfb; } /* sum up sfbThresholds */ wnd = 0; i = 0; for (grp = 0; grp < noOfGroups; grp++) { for (sfb = 0; sfb < sfbActive; sfb++) { FIXP_DBL thresh = sfbThreshold->Short[wnd][sfb]; for (j = 1; j < groupLen[grp]; j++) { thresh = nrgAddSaturate(thresh, sfbThreshold->Short[wnd + j][sfb]); } sfbThreshold->Long[i++] = thresh; } i += sfbCnt - sfb; wnd += groupLen[grp]; } /* sum up sfbEnergies left/right */ wnd = 0; i = 0; for (grp = 0; grp < noOfGroups; grp++) { for (sfb = 0; sfb < sfbActive; sfb++) { FIXP_DBL energy = sfbEnergy->Short[wnd][sfb]; for (j = 1; j < groupLen[grp]; j++) { energy = nrgAddSaturate(energy, sfbEnergy->Short[wnd + j][sfb]); } sfbEnergy->Long[i++] = energy; } i += sfbCnt - sfb; wnd += groupLen[grp]; } /* sum up sfbEnergies mid/side */ wnd = 0; i = 0; for (grp = 0; grp < noOfGroups; grp++) { for (sfb = 0; sfb < sfbActive; sfb++) { FIXP_DBL energy = sfbEnergyMS->Short[wnd][sfb]; for (j = 1; j < groupLen[grp]; j++) { energy = nrgAddSaturate(energy, sfbEnergyMS->Short[wnd + j][sfb]); } sfbEnergyMS->Long[i++] = energy; } i += sfbCnt - sfb; wnd += groupLen[grp]; } /* sum up sfbSpreadEnergies */ wnd = 0; i = 0; for (grp = 0; grp < noOfGroups; grp++) { for (sfb = 0; sfb < sfbActive; sfb++) { FIXP_DBL energy = sfbSpreadEnergy->Short[wnd][sfb]; for (j = 1; j < groupLen[grp]; j++) { energy = nrgAddSaturate(energy, sfbSpreadEnergy->Short[wnd + j][sfb]); } sfbSpreadEnergy->Long[i++] = energy; } i += sfbCnt - sfb; wnd += groupLen[grp]; } /* re-group spectrum */ wnd = 0; i = 0; for (grp = 0; grp < noOfGroups; grp++) { for (sfb = 0; sfb < sfbActive; sfb++) { int width = sfbOffset[sfb + 1] - sfbOffset[sfb]; FIXP_DBL *pMdctSpectrum = &mdctSpectrum[sfbOffset[sfb]] + wnd * granuleLength_short; for (j = 0; j < groupLen[grp]; j++) { FIXP_DBL *pTmp = pMdctSpectrum; for (line = width; line > 0; line--) { tmpSpectrum[i++] = *pTmp++; } pMdctSpectrum += granuleLength_short; } } i += (groupLen[grp] * (sfbOffset[sfbCnt] - sfbOffset[sfb])); wnd += groupLen[grp]; } FDKmemcpy(mdctSpectrum, tmpSpectrum, granuleLength * sizeof(FIXP_DBL)); C_ALLOC_SCRATCH_END(tmpSpectrum, FIXP_DBL, (1024)) } fdk-aac-2.0.2/libAACenc/src/grp_data.h000066400000000000000000000131771404225310700172720ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M.Werner Description: Short block grouping *******************************************************************************/ #ifndef GRP_DATA_H #define GRP_DATA_H #include "common_fix.h" #include "psy_data.h" void FDKaacEnc_groupShortData(FIXP_DBL *mdctSpectrum, /* in-out */ SFB_THRESHOLD *sfbThreshold, /* in-out */ SFB_ENERGY *sfbEnergy, /* in-out */ SFB_ENERGY *sfbEnergyMS, /* in-out */ SFB_ENERGY *sfbSpreadEnergy, const INT sfbCnt, const INT sfbActive, const INT *sfbOffset, const FIXP_DBL *sfbMinSnrLdData, INT *groupedSfbOffset, /* out */ INT *maxSfbPerGroup, FIXP_DBL *groupedSfbMinSnrLdData, const INT noOfGroups, const INT *groupLen, const INT granuleLength); #endif /* _INTERFACE_H */ fdk-aac-2.0.2/libAACenc/src/intensity.cpp000066400000000000000000000774171404225310700201010ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): A. Horndasch (code originally from lwr) / Josef Hoepfl (FDK) Description: intensity stereo processing *******************************************************************************/ #include "intensity.h" #include "interface.h" #include "psy_configuration.h" #include "psy_const.h" #include "qc_main.h" #include "bit_cnt.h" /* only set an IS seed it left/right channel correlation is above IS_CORR_THRESH */ #define IS_CORR_THRESH FL2FXCONST_DBL(0.95f) /* when expanding the IS region to more SFBs only accept an error that is * not more than IS_TOTAL_ERROR_THRESH overall and * not more than IS_LOCAL_ERROR_THRESH for the current SFB */ #define IS_TOTAL_ERROR_THRESH FL2FXCONST_DBL(0.04f) #define IS_LOCAL_ERROR_THRESH FL2FXCONST_DBL(0.01f) /* the maximum allowed change of the intensity direction (unit: IS scale) - * scaled with factor 0.25 - */ #define IS_DIRECTION_DEVIATION_THRESH_SF 2 #define IS_DIRECTION_DEVIATION_THRESH \ FL2FXCONST_DBL(2.0f / (1 << IS_DIRECTION_DEVIATION_THRESH_SF)) /* IS regions need to have a minimal percentage of the overall loudness, e.g. * 0.06 == 6% */ #define IS_REGION_MIN_LOUDNESS FL2FXCONST_DBL(0.1f) /* only perform IS if IS_MIN_SFBS neighboring SFBs can be processed */ #define IS_MIN_SFBS 6 /* only do IS if * if IS_LEFT_RIGHT_RATIO_THRESH < sfbEnergyLeft[sfb]/sfbEnergyRight[sfb] < 1 / * IS_LEFT_RIGHT_RATIO_THRESH * -> no IS if the panning angle is not far from the middle, MS will do */ /* this is equivalent to a scale of +/-1.02914634566 */ #define IS_LEFT_RIGHT_RATIO_THRESH FL2FXCONST_DBL(0.7f) /* scalefactor of realScale */ #define REAL_SCALE_SF 1 /* scalefactor overallLoudness */ #define OVERALL_LOUDNESS_SF 6 /* scalefactor for sum over max samples per goup */ #define MAX_SFB_PER_GROUP_SF 6 /* scalefactor for sum of mdct spectrum */ #define MDCT_SPEC_SF 6 typedef struct { FIXP_DBL corr_thresh; /*!< Only set an IS seed it left/right channel correlation is above corr_thresh */ FIXP_DBL total_error_thresh; /*!< When expanding the IS region to more SFBs only accept an error that is not more than 'total_error_thresh' overall. */ FIXP_DBL local_error_thresh; /*!< When expanding the IS region to more SFBs only accept an error that is not more than 'local_error_thresh' for the current SFB. */ FIXP_DBL direction_deviation_thresh; /*!< The maximum allowed change of the intensity direction (unit: IS scale) */ FIXP_DBL is_region_min_loudness; /*!< IS regions need to have a minimal percentage of the overall loudness, e.g. 0.06 == 6% */ INT min_is_sfbs; /*!< Only perform IS if 'min_is_sfbs' neighboring SFBs can be processed */ FIXP_DBL left_right_ratio_threshold; /*!< No IS if the panning angle is not far from the middle, MS will do */ } INTENSITY_PARAMETERS; /***************************************************************************** functionname: calcSfbMaxScale description: Calc max value in scalefactor band input: *mdctSpectrum l1 l2 output: none returns: scalefactor *****************************************************************************/ static INT calcSfbMaxScale(const FIXP_DBL *mdctSpectrum, const INT l1, const INT l2) { INT i; INT sfbMaxScale; FIXP_DBL maxSpc; maxSpc = FL2FXCONST_DBL(0.0); for (i = l1; i < l2; i++) { FIXP_DBL tmp = fixp_abs((FIXP_DBL)mdctSpectrum[i]); maxSpc = fixMax(maxSpc, tmp); } sfbMaxScale = (maxSpc == FL2FXCONST_DBL(0.0)) ? (DFRACT_BITS - 2) : CntLeadingZeros(maxSpc) - 1; return sfbMaxScale; } /***************************************************************************** functionname: FDKaacEnc_initIsParams description: Initialization of intensity parameters input: isParams output: isParams returns: none *****************************************************************************/ static void FDKaacEnc_initIsParams(INTENSITY_PARAMETERS *isParams) { isParams->corr_thresh = IS_CORR_THRESH; isParams->total_error_thresh = IS_TOTAL_ERROR_THRESH; isParams->local_error_thresh = IS_LOCAL_ERROR_THRESH; isParams->direction_deviation_thresh = IS_DIRECTION_DEVIATION_THRESH; isParams->is_region_min_loudness = IS_REGION_MIN_LOUDNESS; isParams->min_is_sfbs = IS_MIN_SFBS; isParams->left_right_ratio_threshold = IS_LEFT_RIGHT_RATIO_THRESH; } /***************************************************************************** functionname: FDKaacEnc_prepareIntensityDecision description: Prepares intensity decision input: sfbEnergyLeft sfbEnergyRight sfbEnergyLdDataLeft sfbEnergyLdDataRight mdctSpectrumLeft sfbEnergyLdDataRight isParams output: hrrErr scale: none isMask scale: none realScale scale: LD_DATA_SHIFT + REAL_SCALE_SF normSfbLoudness scale: none returns: none *****************************************************************************/ static void FDKaacEnc_prepareIntensityDecision( const FIXP_DBL *sfbEnergyLeft, const FIXP_DBL *sfbEnergyRight, const FIXP_DBL *sfbEnergyLdDataLeft, const FIXP_DBL *sfbEnergyLdDataRight, const FIXP_DBL *mdctSpectrumLeft, const FIXP_DBL *mdctSpectrumRight, const INTENSITY_PARAMETERS *isParams, FIXP_DBL *hrrErr, INT *isMask, FIXP_DBL *realScale, FIXP_DBL *normSfbLoudness, const INT sfbCnt, const INT sfbPerGroup, const INT maxSfbPerGroup, const INT *sfbOffset) { INT j, sfb, sfboffs; INT grpCounter; /* temporary variables to compute loudness */ FIXP_DBL overallLoudness[MAX_NO_OF_GROUPS]; /* temporary variables to compute correlation */ FIXP_DBL channelCorr[MAX_GROUPED_SFB]; FIXP_DBL ml, mr; FIXP_DBL prod_lr; FIXP_DBL square_l, square_r; FIXP_DBL tmp_l, tmp_r; FIXP_DBL inv_n; FDKmemclear(channelCorr, MAX_GROUPED_SFB * sizeof(FIXP_DBL)); FDKmemclear(normSfbLoudness, MAX_GROUPED_SFB * sizeof(FIXP_DBL)); FDKmemclear(overallLoudness, MAX_NO_OF_GROUPS * sizeof(FIXP_DBL)); FDKmemclear(realScale, MAX_GROUPED_SFB * sizeof(FIXP_DBL)); for (grpCounter = 0, sfboffs = 0; sfboffs < sfbCnt; sfboffs += sfbPerGroup, grpCounter++) { overallLoudness[grpCounter] = FL2FXCONST_DBL(0.0f); for (sfb = 0; sfb < maxSfbPerGroup; sfb++) { INT sL, sR, s; FIXP_DBL isValue = sfbEnergyLdDataLeft[sfb + sfboffs] - sfbEnergyLdDataRight[sfb + sfboffs]; /* delimitate intensity scale value to representable range */ realScale[sfb + sfboffs] = fixMin( FL2FXCONST_DBL(60.f / (1 << (REAL_SCALE_SF + LD_DATA_SHIFT))), fixMax(FL2FXCONST_DBL(-60.f / (1 << (REAL_SCALE_SF + LD_DATA_SHIFT))), isValue)); sL = fixMax(0, (CntLeadingZeros(sfbEnergyLeft[sfb + sfboffs]) - 1)); sR = fixMax(0, (CntLeadingZeros(sfbEnergyRight[sfb + sfboffs]) - 1)); s = (fixMin(sL, sR) >> 2) << 2; normSfbLoudness[sfb + sfboffs] = sqrtFixp(sqrtFixp(((sfbEnergyLeft[sfb + sfboffs] << s) >> 1) + ((sfbEnergyRight[sfb + sfboffs] << s) >> 1))) >> (s >> 2); overallLoudness[grpCounter] += normSfbLoudness[sfb + sfboffs] >> OVERALL_LOUDNESS_SF; /* don't do intensity if * - panning angle is too close to the middle or * - one channel is non-existent or * - if it is dual mono */ if ((sfbEnergyLeft[sfb + sfboffs] >= fMult(isParams->left_right_ratio_threshold, sfbEnergyRight[sfb + sfboffs])) && (fMult(isParams->left_right_ratio_threshold, sfbEnergyLeft[sfb + sfboffs]) <= sfbEnergyRight[sfb + sfboffs])) { /* this will prevent post processing from considering this SFB for * merging */ hrrErr[sfb + sfboffs] = FL2FXCONST_DBL(1.0 / 8.0); } } } for (grpCounter = 0, sfboffs = 0; sfboffs < sfbCnt; sfboffs += sfbPerGroup, grpCounter++) { INT invOverallLoudnessSF; FIXP_DBL invOverallLoudness; if (overallLoudness[grpCounter] == FL2FXCONST_DBL(0.0)) { invOverallLoudness = FL2FXCONST_DBL(0.0); invOverallLoudnessSF = 0; } else { invOverallLoudness = fDivNorm((FIXP_DBL)MAXVAL_DBL, overallLoudness[grpCounter], &invOverallLoudnessSF); invOverallLoudnessSF = invOverallLoudnessSF - OVERALL_LOUDNESS_SF + 1; /* +1: compensate fMultDiv2() in subsequent loop */ } invOverallLoudnessSF = fixMin( fixMax(invOverallLoudnessSF, -(DFRACT_BITS - 1)), DFRACT_BITS - 1); for (sfb = 0; sfb < maxSfbPerGroup; sfb++) { FIXP_DBL tmp; tmp = fMultDiv2((normSfbLoudness[sfb + sfboffs] >> OVERALL_LOUDNESS_SF) << OVERALL_LOUDNESS_SF, invOverallLoudness); normSfbLoudness[sfb + sfboffs] = scaleValue(tmp, invOverallLoudnessSF); channelCorr[sfb + sfboffs] = FL2FXCONST_DBL(0.0f); /* max width of scalefactorband is 96; width's are always even */ /* inv_n is scaled with factor 2 to compensate fMultDiv2() in subsequent * loops */ inv_n = GetInvInt( (sfbOffset[sfb + sfboffs + 1] - sfbOffset[sfb + sfboffs]) >> 1); if (inv_n > FL2FXCONST_DBL(0.0f)) { INT s, sL, sR; /* correlation := Pearson's product-moment coefficient */ /* compute correlation between channels and check if it is over * threshold */ ml = FL2FXCONST_DBL(0.0f); mr = FL2FXCONST_DBL(0.0f); prod_lr = FL2FXCONST_DBL(0.0f); square_l = FL2FXCONST_DBL(0.0f); square_r = FL2FXCONST_DBL(0.0f); sL = calcSfbMaxScale(mdctSpectrumLeft, sfbOffset[sfb + sfboffs], sfbOffset[sfb + sfboffs + 1]); sR = calcSfbMaxScale(mdctSpectrumRight, sfbOffset[sfb + sfboffs], sfbOffset[sfb + sfboffs + 1]); s = fixMin(sL, sR); for (j = sfbOffset[sfb + sfboffs]; j < sfbOffset[sfb + sfboffs + 1]; j++) { ml += fMultDiv2((mdctSpectrumLeft[j] << s), inv_n); // scaled with mdctScale - s + inv_n mr += fMultDiv2((mdctSpectrumRight[j] << s), inv_n); // scaled with mdctScale - s + inv_n } ml = fMultDiv2(ml, inv_n); // scaled with mdctScale - s + inv_n mr = fMultDiv2(mr, inv_n); // scaled with mdctScale - s + inv_n for (j = sfbOffset[sfb + sfboffs]; j < sfbOffset[sfb + sfboffs + 1]; j++) { tmp_l = fMultDiv2((mdctSpectrumLeft[j] << s), inv_n) - ml; // scaled with mdctScale - s + inv_n tmp_r = fMultDiv2((mdctSpectrumRight[j] << s), inv_n) - mr; // scaled with mdctScale - s + inv_n prod_lr += fMultDiv2( tmp_l, tmp_r); // scaled with 2*(mdctScale - s + inv_n) + 1 square_l += fPow2Div2(tmp_l); // scaled with 2*(mdctScale - s + inv_n) + 1 square_r += fPow2Div2(tmp_r); // scaled with 2*(mdctScale - s + inv_n) + 1 } prod_lr = prod_lr << 1; // scaled with 2*(mdctScale - s + inv_n) square_l = square_l << 1; // scaled with 2*(mdctScale - s + inv_n) square_r = square_r << 1; // scaled with 2*(mdctScale - s + inv_n) if (square_l > FL2FXCONST_DBL(0.0f) && square_r > FL2FXCONST_DBL(0.0f)) { INT channelCorrSF = 0; /* local scaling of square_l and square_r is compensated after sqrt * calculation */ sL = fixMax(0, (CntLeadingZeros(square_l) - 1)); sR = fixMax(0, (CntLeadingZeros(square_r) - 1)); s = ((sL + sR) >> 1) << 1; sL = fixMin(sL, s); sR = s - sL; tmp = fMult(square_l << sL, square_r << sR); tmp = sqrtFixp(tmp); FDK_ASSERT(tmp > FL2FXCONST_DBL(0.0f)); /* numerator and denominator have the same scaling */ if (prod_lr < FL2FXCONST_DBL(0.0f)) { channelCorr[sfb + sfboffs] = -(fDivNorm(-prod_lr, tmp, &channelCorrSF)); } else { channelCorr[sfb + sfboffs] = (fDivNorm(prod_lr, tmp, &channelCorrSF)); } channelCorrSF = fixMin( fixMax((channelCorrSF + ((sL + sR) >> 1)), -(DFRACT_BITS - 1)), DFRACT_BITS - 1); if (channelCorrSF < 0) { channelCorr[sfb + sfboffs] = channelCorr[sfb + sfboffs] >> (-channelCorrSF); } else { /* avoid overflows due to limited computational accuracy */ if (fAbs(channelCorr[sfb + sfboffs]) > (((FIXP_DBL)MAXVAL_DBL) >> channelCorrSF)) { if (channelCorr[sfb + sfboffs] < FL2FXCONST_DBL(0.0f)) channelCorr[sfb + sfboffs] = -(FIXP_DBL)MAXVAL_DBL; else channelCorr[sfb + sfboffs] = (FIXP_DBL)MAXVAL_DBL; } else { channelCorr[sfb + sfboffs] = channelCorr[sfb + sfboffs] << channelCorrSF; } } } } /* for post processing: hrrErr is the error in terms of (too little) * correlation weighted with the loudness of the SFB; SFBs with small * hrrErr can be merged */ if (hrrErr[sfb + sfboffs] == FL2FXCONST_DBL(1.0 / 8.0)) { continue; } hrrErr[sfb + sfboffs] = fMultDiv2((FL2FXCONST_DBL(0.25f) - (channelCorr[sfb + sfboffs] >> 2)), normSfbLoudness[sfb + sfboffs]); /* set IS mask/vector to 1, if correlation is high enough */ if (fAbs(channelCorr[sfb + sfboffs]) >= isParams->corr_thresh) { isMask[sfb + sfboffs] = 1; } } } } /***************************************************************************** functionname: FDKaacEnc_finalizeIntensityDecision description: Finalizes intensity decision input: isParams scale: none hrrErr scale: none realIsScale scale: LD_DATA_SHIFT + REAL_SCALE_SF normSfbLoudness scale: none output: isMask scale: none returns: none *****************************************************************************/ static void FDKaacEnc_finalizeIntensityDecision( const FIXP_DBL *hrrErr, INT *isMask, const FIXP_DBL *realIsScale, const FIXP_DBL *normSfbLoudness, const INTENSITY_PARAMETERS *isParams, const INT sfbCnt, const INT sfbPerGroup, const INT maxSfbPerGroup) { INT sfb, sfboffs, j; FIXP_DBL isScaleLast = FL2FXCONST_DBL(0.0f); INT isStartValueFound = 0; for (sfboffs = 0; sfboffs < sfbCnt; sfboffs += sfbPerGroup) { INT startIsSfb = 0; INT inIsBlock = 0; INT currentIsSfbCount = 0; FIXP_DBL overallHrrError = FL2FXCONST_DBL(0.0f); FIXP_DBL isRegionLoudness = FL2FXCONST_DBL(0.0f); for (sfb = 0; sfb < maxSfbPerGroup; sfb++) { if (isMask[sfboffs + sfb] == 1) { if (currentIsSfbCount == 0) { startIsSfb = sfboffs + sfb; } if (isStartValueFound == 0) { isScaleLast = realIsScale[sfboffs + sfb]; isStartValueFound = 1; } inIsBlock = 1; currentIsSfbCount++; overallHrrError += hrrErr[sfboffs + sfb] >> (MAX_SFB_PER_GROUP_SF - 3); isRegionLoudness += normSfbLoudness[sfboffs + sfb] >> MAX_SFB_PER_GROUP_SF; } else { /* based on correlation, IS should not be used * -> use it anyway, if overall error is below threshold * and if local error does not exceed threshold * otherwise: check if there are enough IS SFBs */ if (inIsBlock) { overallHrrError += hrrErr[sfboffs + sfb] >> (MAX_SFB_PER_GROUP_SF - 3); isRegionLoudness += normSfbLoudness[sfboffs + sfb] >> MAX_SFB_PER_GROUP_SF; if ((hrrErr[sfboffs + sfb] < (isParams->local_error_thresh >> 3)) && (overallHrrError < (isParams->total_error_thresh >> MAX_SFB_PER_GROUP_SF))) { currentIsSfbCount++; /* overwrite correlation based decision */ isMask[sfboffs + sfb] = 1; } else { inIsBlock = 0; } } } /* check for large direction deviation */ if (inIsBlock) { if (fAbs(isScaleLast - realIsScale[sfboffs + sfb]) < (isParams->direction_deviation_thresh >> (REAL_SCALE_SF + LD_DATA_SHIFT - IS_DIRECTION_DEVIATION_THRESH_SF))) { isScaleLast = realIsScale[sfboffs + sfb]; } else { isMask[sfboffs + sfb] = 0; inIsBlock = 0; currentIsSfbCount--; } } if (currentIsSfbCount > 0 && (!inIsBlock || sfb == maxSfbPerGroup - 1)) { /* not enough SFBs -> do not use IS */ if (currentIsSfbCount < isParams->min_is_sfbs || (isRegionLoudnessis_region_min_loudness>> MAX_SFB_PER_GROUP_SF)) { for (j = startIsSfb; j <= sfboffs + sfb; j++) { isMask[j] = 0; } isScaleLast = FL2FXCONST_DBL(0.0f); isStartValueFound = 0; for (j = 0; j < startIsSfb; j++) { if (isMask[j] != 0) { isScaleLast = realIsScale[j]; isStartValueFound = 1; } } } currentIsSfbCount = 0; overallHrrError = FL2FXCONST_DBL(0.0f); isRegionLoudness = FL2FXCONST_DBL(0.0f); } } } } /***************************************************************************** functionname: FDKaacEnc_IntensityStereoProcessing description: Intensity stereo processing tool input: sfbEnergyLeft sfbEnergyRight mdctSpectrumLeft mdctSpectrumRight sfbThresholdLeft sfbThresholdRight sfbSpreadEnLeft sfbSpreadEnRight sfbEnergyLdDataLeft sfbEnergyLdDataRight output: isBook isScale pnsData->pnsFlag msDigest zeroed from start to sfbCnt msMask zeroed from start to sfbCnt mdctSpectrumRight zeroed where isBook!=0 sfbEnergyRight zeroed where isBook!=0 sfbSpreadEnRight zeroed where isBook!=0 sfbThresholdRight zeroed where isBook!=0 sfbEnergyLdDataRight FL2FXCONST_DBL(-1.0) where isBook!=0 sfbThresholdLdDataRight FL2FXCONST_DBL(-0.515625f) where isBook!=0 returns: none *****************************************************************************/ void FDKaacEnc_IntensityStereoProcessing( FIXP_DBL *sfbEnergyLeft, FIXP_DBL *sfbEnergyRight, FIXP_DBL *mdctSpectrumLeft, FIXP_DBL *mdctSpectrumRight, FIXP_DBL *sfbThresholdLeft, FIXP_DBL *sfbThresholdRight, FIXP_DBL *sfbThresholdLdDataRight, FIXP_DBL *sfbSpreadEnLeft, FIXP_DBL *sfbSpreadEnRight, FIXP_DBL *sfbEnergyLdDataLeft, FIXP_DBL *sfbEnergyLdDataRight, INT *msDigest, INT *msMask, const INT sfbCnt, const INT sfbPerGroup, const INT maxSfbPerGroup, const INT *sfbOffset, const INT allowIS, INT *isBook, INT *isScale, PNS_DATA *RESTRICT pnsData[2]) { INT sfb, sfboffs, j; FIXP_DBL scale; FIXP_DBL lr; FIXP_DBL hrrErr[MAX_GROUPED_SFB]; FIXP_DBL normSfbLoudness[MAX_GROUPED_SFB]; FIXP_DBL realIsScale[MAX_GROUPED_SFB]; INTENSITY_PARAMETERS isParams; INT isMask[MAX_GROUPED_SFB]; FDKmemclear((void *)isBook, sfbCnt * sizeof(INT)); FDKmemclear((void *)isMask, sfbCnt * sizeof(INT)); FDKmemclear((void *)realIsScale, sfbCnt * sizeof(FIXP_DBL)); FDKmemclear((void *)isScale, sfbCnt * sizeof(INT)); FDKmemclear((void *)hrrErr, sfbCnt * sizeof(FIXP_DBL)); if (!allowIS) return; FDKaacEnc_initIsParams(&isParams); /* compute / set the following values per SFB: * - left/right ratio between channels * - normalized loudness * + loudness == average of energy in channels to 0.25 * + normalization: division by sum of all SFB loudnesses * - isMask (is set to 0 if channels are the same or one is 0) */ FDKaacEnc_prepareIntensityDecision( sfbEnergyLeft, sfbEnergyRight, sfbEnergyLdDataLeft, sfbEnergyLdDataRight, mdctSpectrumLeft, mdctSpectrumRight, &isParams, hrrErr, isMask, realIsScale, normSfbLoudness, sfbCnt, sfbPerGroup, maxSfbPerGroup, sfbOffset); FDKaacEnc_finalizeIntensityDecision(hrrErr, isMask, realIsScale, normSfbLoudness, &isParams, sfbCnt, sfbPerGroup, maxSfbPerGroup); for (sfb = 0; sfb < sfbCnt; sfb += sfbPerGroup) { for (sfboffs = 0; sfboffs < maxSfbPerGroup; sfboffs++) { INT sL, sR; FIXP_DBL inv_n; INT mdct_spec_sf = MDCT_SPEC_SF; msMask[sfb + sfboffs] = 0; if (isMask[sfb + sfboffs] == 0) { continue; } if ((sfbEnergyLeft[sfb + sfboffs] < sfbThresholdLeft[sfb + sfboffs]) && (fMult(FL2FXCONST_DBL(1.0f / 1.5f), sfbEnergyRight[sfb + sfboffs]) > sfbThresholdRight[sfb + sfboffs])) { continue; } /* NEW: if there is a big-enough IS region, switch off PNS */ if (pnsData[0]) { if (pnsData[0]->pnsFlag[sfb + sfboffs]) { pnsData[0]->pnsFlag[sfb + sfboffs] = 0; } if (pnsData[1]->pnsFlag[sfb + sfboffs]) { pnsData[1]->pnsFlag[sfb + sfboffs] = 0; } } if (sfbOffset[sfb + sfboffs + 1] - sfbOffset[sfb + sfboffs] > 1 << mdct_spec_sf) { mdct_spec_sf++; /* This is for rare cases where the number of bins in a scale factor band is > 64 */ } inv_n = GetInvInt( (sfbOffset[sfb + sfboffs + 1] - sfbOffset[sfb + sfboffs]) >> 1); // scaled with 2 to compensate fMultDiv2() in subsequent loop sL = calcSfbMaxScale(mdctSpectrumLeft, sfbOffset[sfb + sfboffs], sfbOffset[sfb + sfboffs + 1]); sR = calcSfbMaxScale(mdctSpectrumRight, sfbOffset[sfb + sfboffs], sfbOffset[sfb + sfboffs + 1]); lr = FL2FXCONST_DBL(0.0f); for (j = sfbOffset[sfb + sfboffs]; j < sfbOffset[sfb + sfboffs + 1]; j++) lr += fMultDiv2( fMultDiv2(mdctSpectrumLeft[j] << sL, mdctSpectrumRight[j] << sR), inv_n); lr = lr << 1; if (lr < FL2FXCONST_DBL(0.0f)) { /* This means OUT OF phase intensity stereo, cf. standard */ INT s0, s1, s2; FIXP_DBL tmp, d, ed = FL2FXCONST_DBL(0.0f); s0 = fixMin(sL, sR); for (j = sfbOffset[sfb + sfboffs]; j < sfbOffset[sfb + sfboffs + 1]; j++) { d = ((mdctSpectrumLeft[j] << s0) >> 1) - ((mdctSpectrumRight[j] << s0) >> 1); ed += fMultDiv2(d, d) >> (mdct_spec_sf - 1); } msMask[sfb + sfboffs] = 1; tmp = fDivNorm(sfbEnergyLeft[sfb + sfboffs], ed, &s1); s2 = (s1) + (2 * s0) - 2 - mdct_spec_sf; if (s2 & 1) { tmp = tmp >> 1; s2 = s2 + 1; } s2 = (s2 >> 1) + 1; // +1 compensate fMultDiv2() in subsequent loop s2 = fixMin(fixMax(s2, -(DFRACT_BITS - 1)), (DFRACT_BITS - 1)); scale = sqrtFixp(tmp); if (s2 < 0) { s2 = -s2; for (j = sfbOffset[sfb + sfboffs]; j < sfbOffset[sfb + sfboffs + 1]; j++) { mdctSpectrumLeft[j] = (fMultDiv2(mdctSpectrumLeft[j], scale) - fMultDiv2(mdctSpectrumRight[j], scale)) >> s2; mdctSpectrumRight[j] = FL2FXCONST_DBL(0.0f); } } else { for (j = sfbOffset[sfb + sfboffs]; j < sfbOffset[sfb + sfboffs + 1]; j++) { mdctSpectrumLeft[j] = (fMultDiv2(mdctSpectrumLeft[j], scale) - fMultDiv2(mdctSpectrumRight[j], scale)) << s2; mdctSpectrumRight[j] = FL2FXCONST_DBL(0.0f); } } } else { /* This means IN phase intensity stereo, cf. standard */ INT s0, s1, s2; FIXP_DBL tmp, s, es = FL2FXCONST_DBL(0.0f); s0 = fixMin(sL, sR); for (j = sfbOffset[sfb + sfboffs]; j < sfbOffset[sfb + sfboffs + 1]; j++) { s = ((mdctSpectrumLeft[j] << s0) >> 1) + ((mdctSpectrumRight[j] << s0) >> 1); es += fMultDiv2(s, s) >> (mdct_spec_sf - 1); // scaled 2*(mdctScale - s0 + 1) + mdct_spec_sf } msMask[sfb + sfboffs] = 0; tmp = fDivNorm(sfbEnergyLeft[sfb + sfboffs], es, &s1); s2 = (s1) + (2 * s0) - 2 - mdct_spec_sf; if (s2 & 1) { tmp = tmp >> 1; s2 = s2 + 1; } s2 = (s2 >> 1) + 1; // +1 compensate fMultDiv2() in subsequent loop s2 = fixMin(fixMax(s2, -(DFRACT_BITS - 1)), (DFRACT_BITS - 1)); scale = sqrtFixp(tmp); if (s2 < 0) { s2 = -s2; for (j = sfbOffset[sfb + sfboffs]; j < sfbOffset[sfb + sfboffs + 1]; j++) { mdctSpectrumLeft[j] = (fMultDiv2(mdctSpectrumLeft[j], scale) + fMultDiv2(mdctSpectrumRight[j], scale)) >> s2; mdctSpectrumRight[j] = FL2FXCONST_DBL(0.0f); } } else { for (j = sfbOffset[sfb + sfboffs]; j < sfbOffset[sfb + sfboffs + 1]; j++) { mdctSpectrumLeft[j] = (fMultDiv2(mdctSpectrumLeft[j], scale) + fMultDiv2(mdctSpectrumRight[j], scale)) << s2; mdctSpectrumRight[j] = FL2FXCONST_DBL(0.0f); } } } isBook[sfb + sfboffs] = CODE_BOOK_IS_IN_PHASE_NO; if (realIsScale[sfb + sfboffs] < FL2FXCONST_DBL(0.0f)) { isScale[sfb + sfboffs] = (INT)(((realIsScale[sfb + sfboffs] >> 1) - FL2FXCONST_DBL( 0.5f / (1 << (REAL_SCALE_SF + LD_DATA_SHIFT + 1)))) >> (DFRACT_BITS - 1 - REAL_SCALE_SF - LD_DATA_SHIFT - 1)) + 1; } else { isScale[sfb + sfboffs] = (INT)(((realIsScale[sfb + sfboffs] >> 1) + FL2FXCONST_DBL( 0.5f / (1 << (REAL_SCALE_SF + LD_DATA_SHIFT + 1)))) >> (DFRACT_BITS - 1 - REAL_SCALE_SF - LD_DATA_SHIFT - 1)); } sfbEnergyRight[sfb + sfboffs] = FL2FXCONST_DBL(0.0f); sfbEnergyLdDataRight[sfb + sfboffs] = FL2FXCONST_DBL(-1.0f); sfbThresholdRight[sfb + sfboffs] = FL2FXCONST_DBL(0.0f); sfbThresholdLdDataRight[sfb + sfboffs] = FL2FXCONST_DBL(-0.515625f); sfbSpreadEnRight[sfb + sfboffs] = FL2FXCONST_DBL(0.0f); *msDigest = MS_SOME; } } } fdk-aac-2.0.2/libAACenc/src/intensity.h000066400000000000000000000127521404225310700175350ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): A. Horndasch (code originally from lwr and rtb) / Josef Hoepfl (FDK) Description: intensity stereo prototype *******************************************************************************/ #ifndef INTENSITY_H #define INTENSITY_H #include "aacenc_pns.h" #include "common_fix.h" void FDKaacEnc_IntensityStereoProcessing( FIXP_DBL *sfbEnergyLeft, FIXP_DBL *sfbEnergyRight, FIXP_DBL *mdctSpectrumLeft, FIXP_DBL *mdctSpectrumRight, FIXP_DBL *sfbThresholdLeft, FIXP_DBL *sfbThresholdRight, FIXP_DBL *sfbThresholdLdDataRight, FIXP_DBL *sfbSpreadEnLeft, FIXP_DBL *sfbSpreadEnRight, FIXP_DBL *sfbEnergyLdDataLeft, FIXP_DBL *sfbEnergyLdDataRight, INT *msDigest, INT *msMask, const INT sfbCnt, const INT sfbPerGroup, const INT maxSfbPerGroup, const INT *sfbOffset, const INT allowIS, INT *isBook, INT *isScale, PNS_DATA *RESTRICT pnsData[2]); #endif /* INTENSITY_H */ fdk-aac-2.0.2/libAACenc/src/interface.h000066400000000000000000000141401404225310700174400ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M.Werner Description: Interface psychoaccoustic/quantizer *******************************************************************************/ #ifndef INTERFACE_H #define INTERFACE_H #include "common_fix.h" #include "FDK_audio.h" #include "psy_data.h" #include "aacenc_tns.h" enum { MS_NONE = 0, MS_SOME = 1, MS_ALL = 2 }; enum { MS_ON = 1 }; struct TOOLSINFO { INT msDigest; /* 0 = no MS; 1 = some MS, 2 = all MS */ INT msMask[MAX_GROUPED_SFB]; }; typedef struct { INT sfbCnt; INT sfbPerGroup; INT maxSfbPerGroup; INT lastWindowSequence; INT windowShape; INT groupingMask; INT sfbOffsets[MAX_GROUPED_SFB + 1]; INT mdctScale; /* number of transform shifts */ INT groupLen[MAX_NO_OF_GROUPS]; TNS_INFO tnsInfo; INT noiseNrg[MAX_GROUPED_SFB]; INT isBook[MAX_GROUPED_SFB]; INT isScale[MAX_GROUPED_SFB]; /* memory located in QC_OUT_CHANNEL */ FIXP_DBL *mdctSpectrum; FIXP_DBL *sfbEnergy; FIXP_DBL *sfbSpreadEnergy; FIXP_DBL *sfbThresholdLdData; FIXP_DBL *sfbMinSnrLdData; FIXP_DBL *sfbEnergyLdData; } PSY_OUT_CHANNEL; typedef struct { /* information specific to each channel */ PSY_OUT_CHANNEL *psyOutChannel[(2)]; /* information shared by both channels */ INT commonWindow; struct TOOLSINFO toolsInfo; } PSY_OUT_ELEMENT; typedef struct { PSY_OUT_ELEMENT *psyOutElement[((8))]; PSY_OUT_CHANNEL *pPsyOutChannels[(8)]; } PSY_OUT; inline int isLowDelay(AUDIO_OBJECT_TYPE aot) { return (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD); } #endif /* INTERFACE_H */ fdk-aac-2.0.2/libAACenc/src/line_pe.cpp000066400000000000000000000236241404225310700174550ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Werner Description: Perceptual entropie module *******************************************************************************/ #include "line_pe.h" #include "sf_estim.h" #include "bit_cnt.h" #include "genericStds.h" static const FIXP_DBL C1LdData = FL2FXCONST_DBL(3.0 / LD_DATA_SCALING); /* C1 = 3.0 = log(8.0)/log(2) */ static const FIXP_DBL C2LdData = FL2FXCONST_DBL( 1.3219281 / LD_DATA_SCALING); /* C2 = 1.3219281 = log(2.5)/log(2) */ static const FIXP_DBL C3LdData = FL2FXCONST_DBL(0.5593573); /* 1-C2/C1 */ /* constants that do not change during successive pe calculations */ void FDKaacEnc_prepareSfbPe(PE_CHANNEL_DATA *RESTRICT const peChanData, const FIXP_DBL *RESTRICT const sfbEnergyLdData, const FIXP_DBL *RESTRICT const sfbThresholdLdData, const FIXP_DBL *RESTRICT const sfbFormFactorLdData, const INT *RESTRICT const sfbOffset, const INT sfbCnt, const INT sfbPerGroup, const INT maxSfbPerGroup) { INT sfbGrp, sfb; INT sfbWidth; FIXP_DBL avgFormFactorLdData; const FIXP_DBL formFacScaling = FL2FXCONST_DBL((float)FORM_FAC_SHIFT / LD_DATA_SCALING); for (sfbGrp = 0; sfbGrp < sfbCnt; sfbGrp += sfbPerGroup) { for (sfb = 0; sfb < maxSfbPerGroup; sfb++) { if ((FIXP_DBL)sfbEnergyLdData[sfbGrp + sfb] > (FIXP_DBL)sfbThresholdLdData[sfbGrp + sfb]) { sfbWidth = sfbOffset[sfbGrp + sfb + 1] - sfbOffset[sfbGrp + sfb]; /* estimate number of active lines */ avgFormFactorLdData = ((-sfbEnergyLdData[sfbGrp + sfb] >> 1) + (CalcLdInt(sfbWidth) >> 1)) >> 1; peChanData->sfbNLines[sfbGrp + sfb] = (INT)CalcInvLdData( (sfbFormFactorLdData[sfbGrp + sfb] + formFacScaling) + avgFormFactorLdData); /* Make sure sfbNLines is never greater than sfbWidth due to * unaccuracies (e.g. sfbEnergyLdData[sfbGrp+sfb] = 0x80000000) */ peChanData->sfbNLines[sfbGrp + sfb] = fMin(sfbWidth, peChanData->sfbNLines[sfbGrp + sfb]); } else { peChanData->sfbNLines[sfbGrp + sfb] = 0; } } } } /* formula for one sfb: pe = n * ld(en/thr), if ld(en/thr) >= C1 pe = n * (C2 + C3 * ld(en/thr)), if ld(en/thr) < C1 n: estimated number of lines in sfb, ld(x) = log(x)/log(2) constPart is sfbPe without the threshold part n*ld(thr) or n*C3*ld(thr) */ void FDKaacEnc_calcSfbPe(PE_CHANNEL_DATA *RESTRICT const peChanData, const FIXP_DBL *RESTRICT const sfbEnergyLdData, const FIXP_DBL *RESTRICT const sfbThresholdLdData, const INT sfbCnt, const INT sfbPerGroup, const INT maxSfbPerGroup, const INT *RESTRICT const isBook, const INT *RESTRICT const isScale) { INT sfbGrp, sfb, thisSfb; INT nLines; FIXP_DBL logDataRatio; FIXP_DBL scaleLd = (FIXP_DBL)0; INT lastValIs = 0; FIXP_DBL pe = 0; FIXP_DBL constPart = 0; FIXP_DBL nActiveLines = 0; FIXP_DBL tmpPe, tmpConstPart, tmpNActiveLines; for (sfbGrp = 0; sfbGrp < sfbCnt; sfbGrp += sfbPerGroup) { for (sfb = 0; sfb < maxSfbPerGroup; sfb++) { tmpPe = (FIXP_DBL)0; tmpConstPart = (FIXP_DBL)0; tmpNActiveLines = (FIXP_DBL)0; thisSfb = sfbGrp + sfb; if (sfbEnergyLdData[thisSfb] > sfbThresholdLdData[thisSfb]) { logDataRatio = sfbEnergyLdData[thisSfb] - sfbThresholdLdData[thisSfb]; nLines = peChanData->sfbNLines[thisSfb]; FIXP_DBL factor = nLines << (LD_DATA_SHIFT + PE_CONSTPART_SHIFT + 1); if (logDataRatio >= C1LdData) { /* scale sfbPe and sfbConstPart with PE_CONSTPART_SHIFT */ tmpPe = fMultDiv2(logDataRatio, factor); tmpConstPart = fMultDiv2(sfbEnergyLdData[thisSfb] + scaleLd, factor); } else { /* scale sfbPe and sfbConstPart with PE_CONSTPART_SHIFT */ tmpPe = fMultDiv2( ((FIXP_DBL)C2LdData + fMult(C3LdData, logDataRatio)), factor); tmpConstPart = fMultDiv2(((FIXP_DBL)C2LdData + fMult(C3LdData, sfbEnergyLdData[thisSfb] + scaleLd)), factor); nLines = fMultI(C3LdData, nLines); } tmpNActiveLines = (FIXP_DBL)nLines; } else if (isBook[thisSfb]) { /* provide for cost of scale factor for Intensity */ INT delta = isScale[thisSfb] - lastValIs; lastValIs = isScale[thisSfb]; peChanData->sfbPe[thisSfb] = FDKaacEnc_bitCountScalefactorDelta(delta) << PE_CONSTPART_SHIFT; peChanData->sfbConstPart[thisSfb] = 0; peChanData->sfbNActiveLines[thisSfb] = 0; } peChanData->sfbPe[thisSfb] = tmpPe; peChanData->sfbConstPart[thisSfb] = tmpConstPart; peChanData->sfbNActiveLines[thisSfb] = tmpNActiveLines; /* sum up peChanData values */ pe += tmpPe; constPart += tmpConstPart; nActiveLines += tmpNActiveLines; } } /* correct scaled pe and constPart values */ peChanData->pe = pe >> PE_CONSTPART_SHIFT; peChanData->constPart = constPart >> PE_CONSTPART_SHIFT; peChanData->nActiveLines = nActiveLines; } fdk-aac-2.0.2/libAACenc/src/line_pe.h000066400000000000000000000150161404225310700171160ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Werner Description: Perceptual entropie module *******************************************************************************/ #ifndef LINE_PE_H #define LINE_PE_H #include "common_fix.h" #include "psy_const.h" #define PE_CONSTPART_SHIFT FRACT_BITS typedef struct { /* calculated by FDKaacEnc_prepareSfbPe */ INT sfbNLines[MAX_GROUPED_SFB]; /* number of relevant lines in sfb */ /* the rest is calculated by FDKaacEnc_calcSfbPe */ INT sfbPe[MAX_GROUPED_SFB]; /* pe for each sfb */ INT sfbConstPart[MAX_GROUPED_SFB]; /* constant part for each sfb */ INT sfbNActiveLines[MAX_GROUPED_SFB]; /* number of active lines in sfb */ INT pe; /* sum of sfbPe */ INT constPart; /* sum of sfbConstPart */ INT nActiveLines; /* sum of sfbNActiveLines */ } PE_CHANNEL_DATA; typedef struct { PE_CHANNEL_DATA peChannelData[(2)]; INT pe; INT constPart; INT nActiveLines; INT offset; } PE_DATA; void FDKaacEnc_prepareSfbPe(PE_CHANNEL_DATA *RESTRICT const peChanData, const FIXP_DBL *RESTRICT const sfbEnergyLdData, const FIXP_DBL *RESTRICT const sfbThresholdLdData, const FIXP_DBL *RESTRICT const sfbFormFactorLdData, const INT *RESTRICT const sfbOffset, const INT sfbCnt, const INT sfbPerGroup, const INT maxSfbPerGroup); void FDKaacEnc_calcSfbPe(PE_CHANNEL_DATA *RESTRICT const peChanData, const FIXP_DBL *RESTRICT const sfbEnergyLdData, const FIXP_DBL *RESTRICT const sfbThresholdLdData, const INT sfbCnt, const INT sfbPerGroup, const INT maxSfbPerGroup, const INT *RESTRICT const isBook, const INT *RESTRICT const isScale); #endif fdk-aac-2.0.2/libAACenc/src/metadata_compressor.cpp000066400000000000000000002023171404225310700220740ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Neusinger Description: Compressor for AAC Metadata Generator *******************************************************************************/ #include "metadata_compressor.h" #include "channel_map.h" #define LOG2 0.69314718056f /* natural logarithm of 2 */ #define ILOG2 1.442695041f /* 1/LOG2 */ #define FIXP_ILOG2_DIV2 (FL2FXCONST_DBL(ILOG2 / 2)) /*----------------- defines ----------------------*/ #define MAX_DRC_CHANNELS (8) /*!< Max number of audio input channels. */ #define DOWNMIX_SHIFT (3) /*!< Max 8 channel. */ #define WEIGHTING_FILTER_SHIFT (2) /*!< Scaling used in weighting filter. */ #define METADATA_INT_BITS 10 #define METADATA_LINT_BITS 20 #define METADATA_INT_SCALE (INT64(1) << (METADATA_INT_BITS)) #define METADATA_FRACT_BITS (DFRACT_BITS - 1 - METADATA_INT_BITS) #define METADATA_FRACT_SCALE (INT64(1) << (METADATA_FRACT_BITS)) /** * Enum for channel assignment. */ enum { L = 0, R = 1, C = 2, LFE = 3, LS = 4, RS = 5, S = 6, LS2 = 7, RS2 = 8 }; /*--------------- structure definitions --------------------*/ /** * Structure holds weighting filter filter states. */ struct WEIGHTING_STATES { FIXP_DBL x1; FIXP_DBL x2; FIXP_DBL y1; FIXP_DBL y2; }; /** * Dynamic Range Control compressor structure. */ struct DRC_COMP { FIXP_DBL maxBoostThr[2]; /*!< Max boost threshold. */ FIXP_DBL boostThr[2]; /*!< Boost threshold. */ FIXP_DBL earlyCutThr[2]; /*!< Early cut threshold. */ FIXP_DBL cutThr[2]; /*!< Cut threshold. */ FIXP_DBL maxCutThr[2]; /*!< Max cut threshold. */ FIXP_DBL boostFac[2]; /*!< Precalculated factor for boost compression. */ FIXP_DBL earlyCutFac[2]; /*!< Precalculated factor for early cut compression. */ FIXP_DBL cutFac[2]; /*!< Precalculated factor for cut compression. */ FIXP_DBL maxBoost[2]; /*!< Maximum boost. */ FIXP_DBL maxCut[2]; /*!< Maximum cut. */ FIXP_DBL maxEarlyCut[2]; /*!< Maximum early cut. */ FIXP_DBL fastAttack[2]; /*!< Fast attack coefficient. */ FIXP_DBL fastDecay[2]; /*!< Fast release coefficient. */ FIXP_DBL slowAttack[2]; /*!< Slow attack coefficient. */ FIXP_DBL slowDecay[2]; /*!< Slow release coefficient. */ UINT holdOff[2]; /*!< Hold time in blocks. */ FIXP_DBL attackThr[2]; /*!< Slow/fast attack threshold. */ FIXP_DBL decayThr[2]; /*!< Slow/fast release threshold. */ DRC_PROFILE profile[2]; /*!< DRC profile. */ INT blockLength; /*!< Block length in samples. */ UINT sampleRate; /*!< Sample rate. */ CHANNEL_MODE chanConfig; /*!< Channel configuration. */ UCHAR useWeighting; /*!< Use weighting filter. */ UINT channels; /*!< Number of channels. */ UINT fullChannels; /*!< Number of full range channels. */ INT channelIdx[9]; /*!< Offsets of interleaved channel samples (L, R, C, LFE, Ls, Rs, S, Ls2, Rs2). */ FIXP_DBL smoothLevel[2]; /*!< level smoothing states */ FIXP_DBL smoothGain[2]; /*!< gain smoothing states */ UINT holdCnt[2]; /*!< hold counter */ FIXP_DBL limGain[2]; /*!< limiter gain */ FIXP_DBL limDecay; /*!< limiter decay (linear) */ FIXP_DBL prevPeak[2]; /*!< max peak of previous block (stereo/mono)*/ WEIGHTING_STATES filter[MAX_DRC_CHANNELS]; /*!< array holds weighting filter states */ }; /*---------------- constants -----------------------*/ /** * Profile tables. */ static const FIXP_DBL tabMaxBoostThr[] = { (FIXP_DBL)(-(43 << METADATA_FRACT_BITS)), (FIXP_DBL)(-(53 << METADATA_FRACT_BITS)), (FIXP_DBL)(-(55 << METADATA_FRACT_BITS)), (FIXP_DBL)(-(65 << METADATA_FRACT_BITS)), (FIXP_DBL)(-(50 << METADATA_FRACT_BITS)), (FIXP_DBL)(-(40 << METADATA_FRACT_BITS))}; static const FIXP_DBL tabBoostThr[] = { (FIXP_DBL)(-(31 << METADATA_FRACT_BITS)), (FIXP_DBL)(-(41 << METADATA_FRACT_BITS)), (FIXP_DBL)(-(31 << METADATA_FRACT_BITS)), (FIXP_DBL)(-(41 << METADATA_FRACT_BITS)), (FIXP_DBL)(-(31 << METADATA_FRACT_BITS)), (FIXP_DBL)(-(31 << METADATA_FRACT_BITS))}; static const FIXP_DBL tabEarlyCutThr[] = { (FIXP_DBL)(-(26 << METADATA_FRACT_BITS)), (FIXP_DBL)(-(21 << METADATA_FRACT_BITS)), (FIXP_DBL)(-(26 << METADATA_FRACT_BITS)), (FIXP_DBL)(-(21 << METADATA_FRACT_BITS)), (FIXP_DBL)(-(26 << METADATA_FRACT_BITS)), (FIXP_DBL)(-(20 << METADATA_FRACT_BITS))}; static const FIXP_DBL tabCutThr[] = {(FIXP_DBL)(-(16 << METADATA_FRACT_BITS)), (FIXP_DBL)(-(11 << METADATA_FRACT_BITS)), (FIXP_DBL)(-(16 << METADATA_FRACT_BITS)), (FIXP_DBL)(-(21 << METADATA_FRACT_BITS)), (FIXP_DBL)(-(16 << METADATA_FRACT_BITS)), (FIXP_DBL)(-(10 << METADATA_FRACT_BITS))}; static const FIXP_DBL tabMaxCutThr[] = { (FIXP_DBL)(4 << METADATA_FRACT_BITS), (FIXP_DBL)(9 << METADATA_FRACT_BITS), (FIXP_DBL)(4 << METADATA_FRACT_BITS), (FIXP_DBL)(9 << METADATA_FRACT_BITS), (FIXP_DBL)(4 << METADATA_FRACT_BITS), (FIXP_DBL)(4 << METADATA_FRACT_BITS)}; static const FIXP_DBL tabBoostRatio[] = { FL2FXCONST_DBL(((1.f / 2.f) - 1.f)), FL2FXCONST_DBL(((1.f / 2.f) - 1.f)), FL2FXCONST_DBL(((1.f / 2.f) - 1.f)), FL2FXCONST_DBL(((1.f / 2.f) - 1.f)), FL2FXCONST_DBL(((1.f / 5.f) - 1.f)), FL2FXCONST_DBL(((1.f / 5.f) - 1.f))}; static const FIXP_DBL tabEarlyCutRatio[] = { FL2FXCONST_DBL(((1.f / 2.f) - 1.f)), FL2FXCONST_DBL(((1.f / 2.f) - 1.f)), FL2FXCONST_DBL(((1.f / 2.f) - 1.f)), FL2FXCONST_DBL(((1.f / 1.f) - 1.f)), FL2FXCONST_DBL(((1.f / 2.f) - 1.f)), FL2FXCONST_DBL(((1.f / 2.f) - 1.f))}; static const FIXP_DBL tabCutRatio[] = { FL2FXCONST_DBL(((1.f / 20.f) - 1.f)), FL2FXCONST_DBL(((1.f / 20.f) - 1.f)), FL2FXCONST_DBL(((1.f / 20.f) - 1.f)), FL2FXCONST_DBL(((1.f / 2.f) - 1.f)), FL2FXCONST_DBL(((1.f / 20.f) - 1.f)), FL2FXCONST_DBL(((1.f / 20.f) - 1.f))}; static const FIXP_DBL tabMaxBoost[] = {(FIXP_DBL)(6 << METADATA_FRACT_BITS), (FIXP_DBL)(6 << METADATA_FRACT_BITS), (FIXP_DBL)(12 << METADATA_FRACT_BITS), (FIXP_DBL)(12 << METADATA_FRACT_BITS), (FIXP_DBL)(15 << METADATA_FRACT_BITS), (FIXP_DBL)(15 << METADATA_FRACT_BITS)}; static const FIXP_DBL tabMaxCut[] = {(FIXP_DBL)(24 << METADATA_FRACT_BITS), (FIXP_DBL)(24 << METADATA_FRACT_BITS), (FIXP_DBL)(24 << METADATA_FRACT_BITS), (FIXP_DBL)(15 << METADATA_FRACT_BITS), (FIXP_DBL)(24 << METADATA_FRACT_BITS), (FIXP_DBL)(24 << METADATA_FRACT_BITS)}; static const FIXP_DBL tabFastAttack[] = { FL2FXCONST_DBL((10.f / 1000.f) / METADATA_INT_SCALE), FL2FXCONST_DBL((10.f / 1000.f) / METADATA_INT_SCALE), FL2FXCONST_DBL((10.f / 1000.f) / METADATA_INT_SCALE), FL2FXCONST_DBL((10.f / 1000.f) / METADATA_INT_SCALE), FL2FXCONST_DBL((10.f / 1000.f) / METADATA_INT_SCALE), FL2FXCONST_DBL((0.f / 1000.f) / METADATA_INT_SCALE)}; static const FIXP_DBL tabFastDecay[] = { FL2FXCONST_DBL((1000.f / 1000.f) / METADATA_INT_SCALE), FL2FXCONST_DBL((1000.f / 1000.f) / METADATA_INT_SCALE), FL2FXCONST_DBL((1000.f / 1000.f) / METADATA_INT_SCALE), FL2FXCONST_DBL((1000.f / 1000.f) / METADATA_INT_SCALE), FL2FXCONST_DBL((200.f / 1000.f) / METADATA_INT_SCALE), FL2FXCONST_DBL((0.f / 1000.f) / METADATA_INT_SCALE)}; static const FIXP_DBL tabSlowAttack[] = { FL2FXCONST_DBL((100.f / 1000.f) / METADATA_INT_SCALE), FL2FXCONST_DBL((100.f / 1000.f) / METADATA_INT_SCALE), FL2FXCONST_DBL((100.f / 1000.f) / METADATA_INT_SCALE), FL2FXCONST_DBL((100.f / 1000.f) / METADATA_INT_SCALE), FL2FXCONST_DBL((100.f / 1000.f) / METADATA_INT_SCALE), FL2FXCONST_DBL((0.f / 1000.f) / METADATA_INT_SCALE)}; static const FIXP_DBL tabSlowDecay[] = { FL2FXCONST_DBL((3000.f / 1000.f) / METADATA_INT_SCALE), FL2FXCONST_DBL((3000.f / 1000.f) / METADATA_INT_SCALE), FL2FXCONST_DBL((10000.f / 1000.f) / METADATA_INT_SCALE), FL2FXCONST_DBL((3000.f / 1000.f) / METADATA_INT_SCALE), FL2FXCONST_DBL((1000.f / 1000.f) / METADATA_INT_SCALE), FL2FXCONST_DBL((0.f / 1000.f) / METADATA_INT_SCALE)}; static const INT tabHoldOff[] = {10, 10, 10, 10, 10, 0}; static const FIXP_DBL tabAttackThr[] = {(FIXP_DBL)(15 << METADATA_FRACT_BITS), (FIXP_DBL)(15 << METADATA_FRACT_BITS), (FIXP_DBL)(15 << METADATA_FRACT_BITS), (FIXP_DBL)(15 << METADATA_FRACT_BITS), (FIXP_DBL)(10 << METADATA_FRACT_BITS), (FIXP_DBL)(0 << METADATA_FRACT_BITS)}; static const FIXP_DBL tabDecayThr[] = {(FIXP_DBL)(20 << METADATA_FRACT_BITS), (FIXP_DBL)(20 << METADATA_FRACT_BITS), (FIXP_DBL)(20 << METADATA_FRACT_BITS), (FIXP_DBL)(20 << METADATA_FRACT_BITS), (FIXP_DBL)(10 << METADATA_FRACT_BITS), (FIXP_DBL)(0 << METADATA_FRACT_BITS)}; /** * Weighting filter coefficients (biquad bandpass). */ static const FIXP_DBL b0 = FL2FXCONST_DBL(0.53050662f); /* b1 = 0, b2 = -b0 */ static const FIXP_DBL a1 = FL2FXCONST_DBL(-0.95237983f), a2 = FL2FXCONST_DBL(-0.02248836f); /* a0 = 1 */ /*------------- function definitions ----------------*/ /** * \brief Calculate scaling factor for denoted processing block. * * \param blockLength Length of processing block. * * \return shiftFactor */ static UINT getShiftFactor(const UINT length) { UINT ldN; for (ldN = 1; (((UINT)1) << ldN) < length; ldN++) ; return ldN; } /** * \brief Sum up fixpoint values with best possible accuracy. * * \param value1 First input value. * \param q1 Scaling factor of first input value. * \param pValue2 Pointer to second input value, will be modified on * return. * \param pQ2 Pointer to second scaling factor, will be modified on * return. * * \return void */ static void fixpAdd(const FIXP_DBL value1, const int q1, FIXP_DBL* const pValue2, int* const pQ2) { const int headroom1 = fNormz(fixp_abs(value1)) - 1; const int headroom2 = fNormz(fixp_abs(*pValue2)) - 1; int resultScale = fixMax(q1 - headroom1, (*pQ2) - headroom2); if ((value1 != FL2FXCONST_DBL(0.f)) && (*pValue2 != FL2FXCONST_DBL(0.f))) { resultScale++; } *pValue2 = scaleValue(value1, q1 - resultScale) + scaleValue(*pValue2, (*pQ2) - resultScale); *pQ2 = (*pValue2 != (FIXP_DBL)0) ? resultScale : DFRACT_BITS - 1; } /** * \brief Function for converting time constant to filter coefficient. * * \param t Time constant. * \param sampleRate Sampling rate in Hz. * \param blockLength Length of processing block in samples per channel. * * \return result = 1.0 - exp(-1.0/((t) * (f))) */ static FIXP_DBL tc2Coeff(const FIXP_DBL t, const INT sampleRate, const INT blockLength) { FIXP_DBL sampleRateFract; FIXP_DBL blockLengthFract; FIXP_DBL f, product; FIXP_DBL exponent, result; INT e_res; /* f = sampleRate/blockLength */ sampleRateFract = (FIXP_DBL)(sampleRate << (DFRACT_BITS - 1 - METADATA_LINT_BITS)); blockLengthFract = (FIXP_DBL)(blockLength << (DFRACT_BITS - 1 - METADATA_LINT_BITS)); f = fDivNorm(sampleRateFract, blockLengthFract, &e_res); f = scaleValue(f, e_res - METADATA_INT_BITS); /* convert to METADATA_FRACT */ /* product = t*f */ product = fMultNorm(t, f, &e_res); product = scaleValue( product, e_res + METADATA_INT_BITS); /* convert to METADATA_FRACT */ /* exponent = (-1.0/((t) * (f))) */ exponent = fDivNorm(METADATA_FRACT_SCALE, product, &e_res); exponent = scaleValue( exponent, e_res - METADATA_INT_BITS); /* convert to METADATA_FRACT */ /* exponent * ld(e) */ exponent = fMult(exponent, FIXP_ILOG2_DIV2) << 1; /* e^(x) = 2^(x*ld(e)) */ /* exp(-1.0/((t) * (f))) */ result = f2Pow(-exponent, DFRACT_BITS - 1 - METADATA_FRACT_BITS, &e_res); /* result = 1.0 - exp(-1.0/((t) * (f))) */ result = (FIXP_DBL)MAXVAL_DBL - scaleValue(result, e_res); return result; } static void findPeakLevels(HDRC_COMP drcComp, const INT_PCM* const inSamples, const FIXP_DBL clev, const FIXP_DBL slev, const FIXP_DBL ext_leva, const FIXP_DBL ext_levb, const FIXP_DBL lfe_lev, const FIXP_DBL dmxGain5, const FIXP_DBL dmxGain2, FIXP_DBL peak[2]) { int i, c; FIXP_DBL tmp = FL2FXCONST_DBL(0.f); INT_PCM maxSample = 0; /* find peak level */ peak[0] = peak[1] = FL2FXCONST_DBL(0.f); for (i = 0; i < drcComp->blockLength; i++) { const INT_PCM* pSamples = &inSamples[i * drcComp->channels]; /* single channels */ for (c = 0; c < (int)drcComp->channels; c++) { maxSample = fMax(maxSample, (INT_PCM)fAbs(pSamples[c])); } } peak[0] = fixMax(peak[0], FX_PCM2FX_DBL(maxSample) >> DOWNMIX_SHIFT); /* 7.1/6.1 to 5.1 downmixes */ if (drcComp->fullChannels > 5) { for (i = 0; i < drcComp->blockLength; i++) { const INT_PCM* pSamples = &inSamples[i * drcComp->channels]; /* channel 1 (L, Ls,...) */ tmp = FL2FXCONST_DBL(0.f); switch (drcComp->chanConfig) { case MODE_6_1: tmp += fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >> (DOWNMIX_SHIFT - 1); /* Ls */ tmp += fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[6]]) >> (DOWNMIX_SHIFT - 1); /* Cs */ break; case MODE_7_1_BACK: case MODE_7_1_REAR_SURROUND: tmp += fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >> (DOWNMIX_SHIFT - 1); /* Ls */ tmp += fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> (DOWNMIX_SHIFT - 1); /* Lrs / Lss */ break; case MODE_1_2_2_2_1: case MODE_7_1_FRONT_CENTER: tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >> DOWNMIX_SHIFT); /* L */ tmp += fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> (DOWNMIX_SHIFT - 1); /* Lc */ break; case MODE_7_1_TOP_FRONT: tmp += fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >> (DOWNMIX_SHIFT - 1); /* L */ tmp += fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> (DOWNMIX_SHIFT - 1); /* Lvh */ break; default: break; } peak[0] = fixMax(peak[0], fixp_abs(tmp)); /* channel 2 (R, Rs,...) */ tmp = FL2FXCONST_DBL(0.f); switch (drcComp->chanConfig) { case MODE_6_1: tmp += fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >> (DOWNMIX_SHIFT - 1); /* Rs */ tmp += fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[6]]) >> (DOWNMIX_SHIFT - 1); /* Cs */ break; case MODE_7_1_BACK: case MODE_7_1_REAR_SURROUND: tmp += fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >> (DOWNMIX_SHIFT - 1); /* Rs */ tmp += fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> (DOWNMIX_SHIFT - 1); /* Rrs / Rss */ break; case MODE_1_2_2_2_1: case MODE_7_1_FRONT_CENTER: tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >> DOWNMIX_SHIFT); /* R */ tmp += fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> (DOWNMIX_SHIFT - 1); /* Rc */ break; case MODE_7_1_TOP_FRONT: tmp += fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >> (DOWNMIX_SHIFT - 1); /* R */ tmp += fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> (DOWNMIX_SHIFT - 1); /* Rvh */ break; default: break; } peak[0] = fixMax(peak[0], fixp_abs(tmp)); /* channel 3 (C) */ tmp = FL2FXCONST_DBL(0.f); switch (drcComp->chanConfig) { case MODE_1_2_2_2_1: case MODE_7_1_FRONT_CENTER: tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> DOWNMIX_SHIFT); /* C */ tmp += fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> (DOWNMIX_SHIFT - 1); /* Lc */ tmp += fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> (DOWNMIX_SHIFT - 1); /* Rc */ break; default: break; } peak[0] = fixMax(peak[0], fixp_abs(tmp)); } /* for (blocklength) */ /* take downmix gain into accout */ peak[0] = fMult(dmxGain5, peak[0]) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS); } /* 7.1 / 5.1 to stereo downmixes */ if (drcComp->fullChannels > 2) { /* Lt/Rt downmix */ for (i = 0; i < drcComp->blockLength; i++) { const INT_PCM* pSamples = &inSamples[i * drcComp->channels]; /* Lt */ tmp = FL2FXCONST_DBL(0.f); if (drcComp->channelIdx[LS] >= 0) tmp -= fMultDiv2(FL2FXCONST_DBL(0.707f), (FIXP_PCM)pSamples[drcComp->channelIdx[LS]]) >> (DOWNMIX_SHIFT - 1); /* Ls */ if (drcComp->channelIdx[LS2] >= 0) tmp -= fMultDiv2(FL2FXCONST_DBL(0.707f), (FIXP_PCM)pSamples[drcComp->channelIdx[LS2]]) >> (DOWNMIX_SHIFT - 1); /* Ls2 */ if (drcComp->channelIdx[RS] >= 0) tmp -= fMultDiv2(FL2FXCONST_DBL(0.707f), (FIXP_PCM)pSamples[drcComp->channelIdx[RS]]) >> (DOWNMIX_SHIFT - 1); /* Rs */ if (drcComp->channelIdx[RS2] >= 0) tmp -= fMultDiv2(FL2FXCONST_DBL(0.707f), (FIXP_PCM)pSamples[drcComp->channelIdx[RS2]]) >> (DOWNMIX_SHIFT - 1); /* Rs2 */ if ((drcComp->channelIdx[LS] >= 0) && (drcComp->channelIdx[LS2] >= 0)) tmp = fMult(FL2FXCONST_DBL(0.707f), tmp); /* 7.1ch */ if (drcComp->channelIdx[S] >= 0) tmp -= fMultDiv2(FL2FXCONST_DBL(0.707f), (FIXP_PCM)pSamples[drcComp->channelIdx[S]]) >> (DOWNMIX_SHIFT - 1); /* S */ if (drcComp->channelIdx[C] >= 0) tmp += fMultDiv2(FL2FXCONST_DBL(0.707f), (FIXP_PCM)pSamples[drcComp->channelIdx[C]]) >> (DOWNMIX_SHIFT - 1); /* C */ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[L]]) >> DOWNMIX_SHIFT); /* L */ /* apply scaling of downmix gains */ /* only for positive values only, as legacy decoders might not know this * parameter */ if (dmxGain2 > FL2FXCONST_DBL(0.f)) { if (drcComp->fullChannels > 5) { tmp = fMult(dmxGain5, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS); } tmp = fMult(dmxGain2, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS); } peak[0] = fixMax(peak[0], fixp_abs(tmp)); /* Rt */ tmp = FL2FXCONST_DBL(0.f); if (drcComp->channelIdx[LS] >= 0) tmp += fMultDiv2(FL2FXCONST_DBL(0.707f), (FIXP_PCM)pSamples[drcComp->channelIdx[LS]]) >> (DOWNMIX_SHIFT - 1); /* Ls */ if (drcComp->channelIdx[LS2] >= 0) tmp += fMultDiv2(FL2FXCONST_DBL(0.707f), (FIXP_PCM)pSamples[drcComp->channelIdx[LS2]]) >> (DOWNMIX_SHIFT - 1); /* Ls2 */ if (drcComp->channelIdx[RS] >= 0) tmp += fMultDiv2(FL2FXCONST_DBL(0.707f), (FIXP_PCM)pSamples[drcComp->channelIdx[RS]]) >> (DOWNMIX_SHIFT - 1); /* Rs */ if (drcComp->channelIdx[RS2] >= 0) tmp += fMultDiv2(FL2FXCONST_DBL(0.707f), (FIXP_PCM)pSamples[drcComp->channelIdx[RS2]]) >> (DOWNMIX_SHIFT - 1); /* Rs2 */ if ((drcComp->channelIdx[RS] >= 0) && (drcComp->channelIdx[RS2] >= 0)) tmp = fMult(FL2FXCONST_DBL(0.707f), tmp); /* 7.1ch */ if (drcComp->channelIdx[S] >= 0) tmp += fMultDiv2(FL2FXCONST_DBL(0.707f), (FIXP_PCM)pSamples[drcComp->channelIdx[S]]) >> (DOWNMIX_SHIFT - 1); /* S */ if (drcComp->channelIdx[C] >= 0) tmp += fMultDiv2(FL2FXCONST_DBL(0.707f), (FIXP_PCM)pSamples[drcComp->channelIdx[C]]) >> (DOWNMIX_SHIFT - 1); /* C */ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[R]]) >> DOWNMIX_SHIFT); /* R */ /* apply scaling of downmix gains */ /* only for positive values only, as legacy decoders might not know this * parameter */ if (dmxGain2 > FL2FXCONST_DBL(0.f)) { if (drcComp->fullChannels > 5) { tmp = fMult(dmxGain5, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS); } tmp = fMult(dmxGain2, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS); } peak[0] = fixMax(peak[0], fixp_abs(tmp)); } /* Lo/Ro downmix */ for (i = 0; i < drcComp->blockLength; i++) { const INT_PCM* pSamples = &inSamples[i * drcComp->channels]; /* Lo */ tmp = FL2FXCONST_DBL(0.f); switch (drcComp->chanConfig) { case MODE_6_1: tmp += fMultDiv2(fMult(slev, ext_leva), (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >> (DOWNMIX_SHIFT - 1); /* Ls */ tmp += fMultDiv2(fMult(slev, ext_levb), (FIXP_PCM)pSamples[drcComp->channelIdx[6]]) >> (DOWNMIX_SHIFT - 1); /* Cs */ tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> (DOWNMIX_SHIFT - 1); /* C */ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >> DOWNMIX_SHIFT); /* L */ tmp += fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ break; case MODE_7_1_BACK: case MODE_7_1_REAR_SURROUND: tmp += fMultDiv2(fMult(slev, ext_leva), (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >> (DOWNMIX_SHIFT - 1); /* Ls */ tmp += fMultDiv2(fMult(slev, ext_levb), (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> (DOWNMIX_SHIFT - 1); /* Lrs / Lss*/ tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> (DOWNMIX_SHIFT - 1); /* C */ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >> DOWNMIX_SHIFT); /* L */ tmp += fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ break; case MODE_1_2_2_2_1: case MODE_7_1_FRONT_CENTER: tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >> DOWNMIX_SHIFT); /* L */ tmp += fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> (DOWNMIX_SHIFT - 1); /* Lc */ tmp += fMultDiv2(fMult(ext_leva, clev), (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> (DOWNMIX_SHIFT - 1); /* Lc - second path*/ tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> (DOWNMIX_SHIFT - 1); /* C */ tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >> (DOWNMIX_SHIFT - 1); /* Ls */ tmp += fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ break; case MODE_7_1_TOP_FRONT: tmp += fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >> (DOWNMIX_SHIFT - 1); /* L */ tmp += fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> (DOWNMIX_SHIFT - 1); /* Lvh */ tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> (DOWNMIX_SHIFT - 1); /* C */ tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >> (DOWNMIX_SHIFT - 1); /* Ls */ tmp += fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ break; default: if (drcComp->channelIdx[LS] >= 0) tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[LS]]) >> (DOWNMIX_SHIFT - 1); /* Ls */ if (drcComp->channelIdx[LS2] >= 0) tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[LS2]]) >> (DOWNMIX_SHIFT - 1); /* Ls2 */ if ((drcComp->channelIdx[LS] >= 0) && (drcComp->channelIdx[LS2] >= 0)) tmp = fMult(FL2FXCONST_DBL(0.707f), tmp); /* 7.1ch */ if (drcComp->channelIdx[S] >= 0) tmp += fMultDiv2(slev, fMult(FL2FXCONST_DBL(0.7f), (FIXP_PCM)pSamples[drcComp->channelIdx[S]])) >> (DOWNMIX_SHIFT - 1); /* S */ if (drcComp->channelIdx[C] >= 0) tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[C]]) >> (DOWNMIX_SHIFT - 1); /* C */ if (drcComp->channelIdx[3] >= 0) tmp += fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[L]]) >> DOWNMIX_SHIFT); /* L */ break; } /* apply scaling of downmix gains */ /* only for positive values only, as legacy decoders might not know this * parameter */ if (dmxGain2 > FL2FXCONST_DBL(0.f)) { if (drcComp->fullChannels > 5) { tmp = fMult(dmxGain5, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS); } tmp = fMult(dmxGain2, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS); } peak[0] = fixMax(peak[0], fixp_abs(tmp)); /* Ro */ tmp = FL2FXCONST_DBL(0.f); switch (drcComp->chanConfig) { case MODE_6_1: tmp += fMultDiv2(fMult(slev, ext_leva), (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >> (DOWNMIX_SHIFT - 1); /* Rs */ tmp += fMultDiv2(fMult(slev, ext_levb), (FIXP_PCM)pSamples[drcComp->channelIdx[6]]) >> (DOWNMIX_SHIFT - 1); /* Cs */ tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> (DOWNMIX_SHIFT - 1); /* C */ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >> DOWNMIX_SHIFT); /* R */ tmp += fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ break; case MODE_7_1_BACK: case MODE_7_1_REAR_SURROUND: tmp += fMultDiv2(fMult(slev, ext_leva), (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >> (DOWNMIX_SHIFT - 1); /* Rs */ tmp += fMultDiv2(fMult(slev, ext_levb), (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> (DOWNMIX_SHIFT - 1); /* Rrs / Rss*/ tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> (DOWNMIX_SHIFT - 1); /* C */ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >> DOWNMIX_SHIFT); /* R */ tmp += fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ break; case MODE_1_2_2_2_1: case MODE_7_1_FRONT_CENTER: tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >> DOWNMIX_SHIFT); /* R */ tmp += fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> (DOWNMIX_SHIFT - 1); /* Rc */ tmp += fMultDiv2(fMult(ext_leva, clev), (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> (DOWNMIX_SHIFT - 1); /* Rc - second path*/ tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> (DOWNMIX_SHIFT - 1); /* C */ tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >> (DOWNMIX_SHIFT - 1); /* Rs */ tmp += fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ break; case MODE_7_1_TOP_FRONT: tmp += fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >> (DOWNMIX_SHIFT - 1); /* R */ tmp += fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> (DOWNMIX_SHIFT - 1); /* Rvh */ tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> (DOWNMIX_SHIFT - 1); /* C */ tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >> (DOWNMIX_SHIFT - 1); /* Rs */ tmp += fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ break; default: if (drcComp->channelIdx[RS] >= 0) tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[RS]]) >> (DOWNMIX_SHIFT - 1); /* Rs */ if (drcComp->channelIdx[RS2] >= 0) tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[RS2]]) >> (DOWNMIX_SHIFT - 1); /* Rs2 */ if ((drcComp->channelIdx[RS] >= 0) && (drcComp->channelIdx[RS2] >= 0)) tmp = fMult(FL2FXCONST_DBL(0.707f), tmp); /* 7.1ch */ if (drcComp->channelIdx[S] >= 0) tmp += fMultDiv2(slev, fMult(FL2FXCONST_DBL(0.7f), (FIXP_PCM)pSamples[drcComp->channelIdx[S]])) >> (DOWNMIX_SHIFT - 1); /* S */ if (drcComp->channelIdx[C] >= 0) tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[C]]) >> (DOWNMIX_SHIFT - 1); /* C */ if (drcComp->channelIdx[3] >= 0) tmp += fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[R]]) >> DOWNMIX_SHIFT); /* R */ } /* apply scaling of downmix gains */ /* only for positive values only, as legacy decoders might not know this * parameter */ if (dmxGain2 > FL2FXCONST_DBL(0.f)) { if (drcComp->fullChannels > 5) { tmp = fMult(dmxGain5, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS); } tmp = fMult(dmxGain2, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS); } peak[0] = fixMax(peak[0], fixp_abs(tmp)); } } peak[1] = fixMax(peak[0], peak[1]); /* Mono Downmix - for comp_val only */ if (drcComp->fullChannels > 1) { for (i = 0; i < drcComp->blockLength; i++) { const INT_PCM* pSamples = &inSamples[i * drcComp->channels]; tmp = FL2FXCONST_DBL(0.f); switch (drcComp->chanConfig) { case MODE_6_1: tmp += fMultDiv2(fMult(slev, ext_leva), (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >> (DOWNMIX_SHIFT - 1); /* Ls */ tmp += fMultDiv2(fMult(slev, ext_leva), (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >> (DOWNMIX_SHIFT - 1); /* Rs */ tmp += fMult(fMult(slev, ext_levb), (FIXP_PCM)pSamples[drcComp->channelIdx[6]]) >> (DOWNMIX_SHIFT - 1); /* Cs */ tmp += fMult(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> (DOWNMIX_SHIFT - 1); /* C */ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >> DOWNMIX_SHIFT); /* L */ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >> DOWNMIX_SHIFT); /* R */ tmp += fMult(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ break; case MODE_7_1_BACK: case MODE_7_1_REAR_SURROUND: tmp += fMultDiv2(fMult(slev, ext_leva), (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >> (DOWNMIX_SHIFT - 1); /* Ls */ tmp += fMultDiv2(fMult(slev, ext_leva), (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >> (DOWNMIX_SHIFT - 1); /* Rs */ tmp += fMultDiv2(fMult(slev, ext_levb), (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> (DOWNMIX_SHIFT - 1); /* Lrs / Lss*/ tmp += fMultDiv2(fMult(slev, ext_levb), (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> (DOWNMIX_SHIFT - 1); /* Rrs / Rss*/ tmp += fMult(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> (DOWNMIX_SHIFT - 1); /* C */ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >> DOWNMIX_SHIFT); /* L */ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >> DOWNMIX_SHIFT); /* R */ tmp += fMult(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ break; case MODE_1_2_2_2_1: case MODE_7_1_FRONT_CENTER: tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >> DOWNMIX_SHIFT); /* L */ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >> DOWNMIX_SHIFT); /* R */ tmp += fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> (DOWNMIX_SHIFT - 1); /* Lc */ tmp += fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> (DOWNMIX_SHIFT - 1); /* Rc */ tmp += fMultDiv2(fMult(ext_leva, clev), (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> (DOWNMIX_SHIFT - 1); /* Lc - second path*/ tmp += fMultDiv2(fMult(ext_leva, clev), (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> (DOWNMIX_SHIFT - 1); /* Rc - second path*/ tmp += fMult(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> (DOWNMIX_SHIFT - 1); /* C */ tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >> (DOWNMIX_SHIFT - 1); /* Ls */ tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >> (DOWNMIX_SHIFT - 1); /* Rs */ tmp += fMult(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ break; case MODE_7_1_TOP_FRONT: tmp += fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >> (DOWNMIX_SHIFT - 1); /* L */ tmp += fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >> (DOWNMIX_SHIFT - 1); /* R */ tmp += fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> (DOWNMIX_SHIFT - 1); /* Lvh */ tmp += fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> (DOWNMIX_SHIFT - 1); /* Rvh */ tmp += fMult(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> (DOWNMIX_SHIFT - 1); /* C */ tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >> (DOWNMIX_SHIFT - 1); /* Ls */ tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >> (DOWNMIX_SHIFT - 1); /* Rs */ tmp += fMult(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ break; default: if (drcComp->channelIdx[LS] >= 0) tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[LS]]) >> (DOWNMIX_SHIFT - 1); /* Ls */ if (drcComp->channelIdx[LS2] >= 0) tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[LS2]]) >> (DOWNMIX_SHIFT - 1); /* Ls2 */ if (drcComp->channelIdx[RS] >= 0) tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[RS]]) >> (DOWNMIX_SHIFT - 1); /* Rs */ if (drcComp->channelIdx[RS2] >= 0) tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[RS2]]) >> (DOWNMIX_SHIFT - 1); /* Rs2 */ if ((drcComp->channelIdx[LS] >= 0) && (drcComp->channelIdx[LS2] >= 0)) tmp = fMult(FL2FXCONST_DBL(0.707f), tmp); /* 7.1ch */ /*if ((drcComp->channelIdx[RS] >= 0) && (drcComp->channelIdx[RS2] >= 0)) tmp *=0.707f;*/ /* 7.1ch */ if (drcComp->channelIdx[S] >= 0) tmp += fMultDiv2(slev, fMult(FL2FXCONST_DBL(0.7f), (FIXP_PCM)pSamples[drcComp->channelIdx[S]])) >> (DOWNMIX_SHIFT - 1); /* S */ if (drcComp->channelIdx[C] >= 0) tmp += fMult(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[C]]) >> (DOWNMIX_SHIFT - 1); /* C (2*clev) */ if (drcComp->channelIdx[3] >= 0) tmp += fMult(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[L]]) >> DOWNMIX_SHIFT); /* L */ tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[R]]) >> DOWNMIX_SHIFT); /* R */ } /* apply scaling of downmix gains */ /* only for positive values only, as legacy decoders might not know this * parameter */ if (dmxGain2 > FL2FXCONST_DBL(0.f)) { if (drcComp->fullChannels > 5) { tmp = fMult(dmxGain5, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS); } tmp = fMult(dmxGain2, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS); } peak[1] = fixMax(peak[1], fixp_abs(tmp)); } } } INT FDK_DRC_Generator_Open(HDRC_COMP* phDrcComp) { INT err = 0; HDRC_COMP hDcComp = NULL; if (phDrcComp == NULL) { err = -1; goto bail; } /* allocate memory */ hDcComp = (HDRC_COMP)FDKcalloc(1, sizeof(DRC_COMP)); if (hDcComp == NULL) { err = -1; goto bail; } FDKmemclear(hDcComp, sizeof(DRC_COMP)); /* Return drc compressor instance */ *phDrcComp = hDcComp; return err; bail: FDK_DRC_Generator_Close(&hDcComp); return err; } INT FDK_DRC_Generator_Close(HDRC_COMP* phDrcComp) { if (phDrcComp == NULL) { return -1; } if (*phDrcComp != NULL) { FDKfree(*phDrcComp); *phDrcComp = NULL; } return 0; } INT FDK_DRC_Generator_Initialize(HDRC_COMP drcComp, const DRC_PROFILE profileLine, const DRC_PROFILE profileRF, const INT blockLength, const UINT sampleRate, const CHANNEL_MODE channelMode, const CHANNEL_ORDER channelOrder, const UCHAR useWeighting) { int i; CHANNEL_MAPPING channelMapping; drcComp->limDecay = FL2FXCONST_DBL(((0.006f / 256) * blockLength) / METADATA_INT_SCALE); /* Save parameters. */ drcComp->blockLength = blockLength; drcComp->sampleRate = sampleRate; drcComp->chanConfig = channelMode; drcComp->useWeighting = useWeighting; if (FDK_DRC_Generator_setDrcProfile(drcComp, profileLine, profileRF) != 0) { /* expects initialized blockLength and sampleRate */ return (-1); } /* Set number of channels and channel offsets. */ if (FDKaacEnc_InitChannelMapping(channelMode, channelOrder, &channelMapping) != AAC_ENC_OK) { return (-2); } for (i = 0; i < 9; i++) drcComp->channelIdx[i] = -1; switch (channelMode) { case MODE_1: /* mono */ drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; break; case MODE_2: /* stereo */ drcComp->channelIdx[L] = channelMapping.elInfo[0].ChannelIndex[0]; drcComp->channelIdx[R] = channelMapping.elInfo[0].ChannelIndex[1]; break; case MODE_1_2: /* 3ch */ drcComp->channelIdx[L] = channelMapping.elInfo[1].ChannelIndex[0]; drcComp->channelIdx[R] = channelMapping.elInfo[1].ChannelIndex[1]; drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; break; case MODE_1_2_1: /* 4ch */ drcComp->channelIdx[L] = channelMapping.elInfo[1].ChannelIndex[0]; drcComp->channelIdx[R] = channelMapping.elInfo[1].ChannelIndex[1]; drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; drcComp->channelIdx[S] = channelMapping.elInfo[2].ChannelIndex[0]; break; case MODE_1_2_2: /* 5ch */ drcComp->channelIdx[L] = channelMapping.elInfo[1].ChannelIndex[0]; drcComp->channelIdx[R] = channelMapping.elInfo[1].ChannelIndex[1]; drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; drcComp->channelIdx[LS] = channelMapping.elInfo[2].ChannelIndex[0]; drcComp->channelIdx[RS] = channelMapping.elInfo[2].ChannelIndex[1]; break; case MODE_1_2_2_1: /* 5.1 ch */ drcComp->channelIdx[L] = channelMapping.elInfo[1].ChannelIndex[0]; drcComp->channelIdx[R] = channelMapping.elInfo[1].ChannelIndex[1]; drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; drcComp->channelIdx[LFE] = channelMapping.elInfo[3].ChannelIndex[0]; drcComp->channelIdx[LS] = channelMapping.elInfo[2].ChannelIndex[0]; drcComp->channelIdx[RS] = channelMapping.elInfo[2].ChannelIndex[1]; break; case MODE_1_2_2_2_1: /* 7.1 ch */ case MODE_7_1_FRONT_CENTER: drcComp->channelIdx[L] = channelMapping.elInfo[2].ChannelIndex[0]; /* l */ drcComp->channelIdx[R] = channelMapping.elInfo[2].ChannelIndex[1]; /* r */ drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; /* c */ drcComp->channelIdx[LFE] = channelMapping.elInfo[4].ChannelIndex[0]; /* lfe */ drcComp->channelIdx[LS] = channelMapping.elInfo[3].ChannelIndex[0]; /* ls */ drcComp->channelIdx[RS] = channelMapping.elInfo[3].ChannelIndex[1]; /* rs */ drcComp->channelIdx[LS2] = channelMapping.elInfo[1].ChannelIndex[0]; /* lc */ drcComp->channelIdx[RS2] = channelMapping.elInfo[1].ChannelIndex[1]; /* rc */ break; case MODE_7_1_BACK: case MODE_7_1_REAR_SURROUND: drcComp->channelIdx[L] = channelMapping.elInfo[1].ChannelIndex[0]; /* l */ drcComp->channelIdx[R] = channelMapping.elInfo[1].ChannelIndex[1]; /* r */ drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; /* c */ drcComp->channelIdx[LFE] = channelMapping.elInfo[4].ChannelIndex[0]; /* lfe */ drcComp->channelIdx[LS] = channelMapping.elInfo[3].ChannelIndex[0]; /* lrear */ drcComp->channelIdx[RS] = channelMapping.elInfo[3].ChannelIndex[1]; /* rrear */ drcComp->channelIdx[LS2] = channelMapping.elInfo[2].ChannelIndex[0]; /* ls */ drcComp->channelIdx[RS2] = channelMapping.elInfo[2].ChannelIndex[1]; /* rs */ break; case MODE_6_1: drcComp->channelIdx[L] = channelMapping.elInfo[1].ChannelIndex[0]; /* l */ drcComp->channelIdx[R] = channelMapping.elInfo[1].ChannelIndex[1]; /* r */ drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; /* c */ drcComp->channelIdx[LFE] = channelMapping.elInfo[4].ChannelIndex[0]; /* lfe */ drcComp->channelIdx[LS] = channelMapping.elInfo[2].ChannelIndex[0]; /* ls */ drcComp->channelIdx[RS] = channelMapping.elInfo[2].ChannelIndex[1]; /* rs */ drcComp->channelIdx[S] = channelMapping.elInfo[3].ChannelIndex[0]; /* s */ break; case MODE_7_1_TOP_FRONT: drcComp->channelIdx[L] = channelMapping.elInfo[1].ChannelIndex[0]; /* l */ drcComp->channelIdx[R] = channelMapping.elInfo[1].ChannelIndex[1]; /* r */ drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; /* c */ drcComp->channelIdx[LFE] = channelMapping.elInfo[3].ChannelIndex[0]; /* lfe */ drcComp->channelIdx[LS] = channelMapping.elInfo[2].ChannelIndex[0]; /* ls */ drcComp->channelIdx[RS] = channelMapping.elInfo[2].ChannelIndex[1]; /* rs */ drcComp->channelIdx[LS2] = channelMapping.elInfo[4].ChannelIndex[0]; /* lvh2 */ drcComp->channelIdx[RS2] = channelMapping.elInfo[4].ChannelIndex[1]; /* rvh2 */ break; default: return (-1); } drcComp->fullChannels = channelMapping.nChannelsEff; drcComp->channels = channelMapping.nChannels; /* Init states. */ drcComp->smoothLevel[0] = drcComp->smoothLevel[1] = (FIXP_DBL)(-(135 << METADATA_FRACT_BITS)); FDKmemclear(drcComp->smoothGain, sizeof(drcComp->smoothGain)); FDKmemclear(drcComp->holdCnt, sizeof(drcComp->holdCnt)); FDKmemclear(drcComp->limGain, sizeof(drcComp->limGain)); FDKmemclear(drcComp->prevPeak, sizeof(drcComp->prevPeak)); FDKmemclear(drcComp->filter, sizeof(drcComp->filter)); return (0); } INT FDK_DRC_Generator_setDrcProfile(HDRC_COMP drcComp, const DRC_PROFILE profileLine, const DRC_PROFILE profileRF) { int profileIdx, i; drcComp->profile[0] = profileLine; drcComp->profile[1] = profileRF; for (i = 0; i < 2; i++) { /* get profile index */ switch (drcComp->profile[i]) { case DRC_NONE: case DRC_NOT_PRESENT: case DRC_FILMSTANDARD: profileIdx = 0; break; case DRC_FILMLIGHT: profileIdx = 1; break; case DRC_MUSICSTANDARD: profileIdx = 2; break; case DRC_MUSICLIGHT: profileIdx = 3; break; case DRC_SPEECH: profileIdx = 4; break; case DRC_DELAY_TEST: profileIdx = 5; break; default: return (-1); } /* get parameters for selected profile */ if (profileIdx >= 0) { drcComp->maxBoostThr[i] = tabMaxBoostThr[profileIdx]; drcComp->boostThr[i] = tabBoostThr[profileIdx]; drcComp->earlyCutThr[i] = tabEarlyCutThr[profileIdx]; drcComp->cutThr[i] = tabCutThr[profileIdx]; drcComp->maxCutThr[i] = tabMaxCutThr[profileIdx]; drcComp->boostFac[i] = tabBoostRatio[profileIdx]; drcComp->earlyCutFac[i] = tabEarlyCutRatio[profileIdx]; drcComp->cutFac[i] = tabCutRatio[profileIdx]; drcComp->maxBoost[i] = tabMaxBoost[profileIdx]; drcComp->maxCut[i] = tabMaxCut[profileIdx]; drcComp->maxEarlyCut[i] = -fMult((drcComp->cutThr[i] - drcComp->earlyCutThr[i]), drcComp->earlyCutFac[i]); /* no scaling after mult needed, earlyCutFac is in FIXP_DBL */ drcComp->fastAttack[i] = tc2Coeff( tabFastAttack[profileIdx], drcComp->sampleRate, drcComp->blockLength); drcComp->fastDecay[i] = tc2Coeff( tabFastDecay[profileIdx], drcComp->sampleRate, drcComp->blockLength); drcComp->slowAttack[i] = tc2Coeff( tabSlowAttack[profileIdx], drcComp->sampleRate, drcComp->blockLength); drcComp->slowDecay[i] = tc2Coeff( tabSlowDecay[profileIdx], drcComp->sampleRate, drcComp->blockLength); drcComp->holdOff[i] = tabHoldOff[profileIdx] * 256 / drcComp->blockLength; drcComp->attackThr[i] = tabAttackThr[profileIdx]; drcComp->decayThr[i] = tabDecayThr[profileIdx]; } drcComp->smoothGain[i] = FL2FXCONST_DBL(0.f); } return (0); } INT FDK_DRC_Generator_Calc(HDRC_COMP drcComp, const INT_PCM* const inSamples, const UINT inSamplesBufSize, const INT dialnorm, const INT drc_TargetRefLevel, const INT comp_TargetRefLevel, const FIXP_DBL clev, const FIXP_DBL slev, const FIXP_DBL ext_leva, const FIXP_DBL ext_levb, const FIXP_DBL lfe_lev, const INT dmxGain5, const INT dmxGain2, INT* const pDynrng, INT* const pCompr) { int i, c; FIXP_DBL peak[2]; /************************************************************************** * compressor **************************************************************************/ if ((drcComp->profile[0] != DRC_NONE) || (drcComp->profile[1] != DRC_NONE)) { /* Calc loudness level */ FIXP_DBL level_b = FL2FXCONST_DBL(0.f); int level_e = DFRACT_BITS - 1; /* Increase energy time resolution with shorter processing blocks. 16 is an * empiric value. */ const int granuleLength = fixMin(16, drcComp->blockLength); if (drcComp->useWeighting) { FIXP_DBL x1, x2, y, y1, y2; /* sum of filter coefficients about 2.5 -> squared value is 6.25 WEIGHTING_FILTER_SHIFT is 2 -> scaling about 16, therefore reduce granuleShift by 1. */ const int granuleShift = getShiftFactor(granuleLength) - 1; for (c = 0; c < (int)drcComp->channels; c++) { const INT_PCM* pSamples = inSamples + c * inSamplesBufSize; if (c == drcComp->channelIdx[LFE]) { continue; /* skip LFE */ } /* get filter states */ x1 = drcComp->filter[c].x1; x2 = drcComp->filter[c].x2; y1 = drcComp->filter[c].y1; y2 = drcComp->filter[c].y2; i = 0; do { int offset = i; FIXP_DBL accu = FL2FXCONST_DBL(0.f); for (i = offset; i < fixMin(offset + granuleLength, drcComp->blockLength); i++) { /* apply weighting filter */ FIXP_DBL x = FX_PCM2FX_DBL((FIXP_PCM)pSamples[i]) >> WEIGHTING_FILTER_SHIFT; /* y = b0 * (x - x2) - a1 * y1 - a2 * y2; */ y = fMult(b0, x - x2) - fMult(a1, y1) - fMult(a2, y2); x2 = x1; x1 = x; y2 = y1; y1 = y; accu += fPow2Div2(y) >> (granuleShift - 1); /* partial energy */ } /* i */ fixpAdd(accu, granuleShift + 2 * WEIGHTING_FILTER_SHIFT, &level_b, &level_e); /* sup up partial energies */ } while (i < drcComp->blockLength); /* save filter states */ drcComp->filter[c].x1 = x1; drcComp->filter[c].x2 = x2; drcComp->filter[c].y1 = y1; drcComp->filter[c].y2 = y2; } /* c */ } /* weighting */ else { const int granuleShift = getShiftFactor(granuleLength); for (c = 0; c < (int)drcComp->channels; c++) { const INT_PCM* pSamples = inSamples + c * inSamplesBufSize; if ((int)c == drcComp->channelIdx[LFE]) { continue; /* skip LFE */ } i = 0; do { int offset = i; FIXP_DBL accu = FL2FXCONST_DBL(0.f); for (i = offset; i < fixMin(offset + granuleLength, drcComp->blockLength); i++) { /* partial energy */ accu += fPow2Div2((FIXP_PCM)pSamples[i]) >> (granuleShift - 1); } /* i */ fixpAdd(accu, granuleShift, &level_b, &level_e); /* sup up partial energies */ } while (i < drcComp->blockLength); } } /* weighting */ /* * Convert to dBFS, apply dialnorm */ /* level scaling */ /* descaled level in ld64 representation */ FIXP_DBL ldLevel = CalcLdData(level_b) + (FIXP_DBL)((level_e - 12) << (DFRACT_BITS - 1 - LD_DATA_SHIFT)) - CalcLdData((FIXP_DBL)(drcComp->blockLength << (DFRACT_BITS - 1 - 12))); /* if (level < 1e-10) level = 1e-10f; */ ldLevel = fMax(ldLevel, FL2FXCONST_DBL(-0.51905126482615036685473741085772f)); /* level = 10 * log(level)/log(10) + 3; * = 10*log(2)/log(10) * ld(level) + 3; * = 10 * 0.30102999566398119521373889472449 * ld(level) + 3 * = 10 * (0.30102999566398119521373889472449 * ld(level) + 0.3) * = 10 * (0.30102999566398119521373889472449 * ld64(level) + 0.3/64) * * 64 * * additional scaling with METADATA_FRACT_BITS: * = 10 * (0.30102999566398119521373889472449 * ld64(level) + 0.3/64) * * 64 * 2^(METADATA_FRACT_BITS) = 10 * (0.30102999566398119521373889472449 * * ld64(level) + 0.3/64) * 2^(METADATA_FRACT_BITS+LD_DATA_SHIFT) = * 10*2^(METADATA_FRACT_BITS+LD_DATA_SHIFT) * ( * 0.30102999566398119521373889472449 * ld64(level) + 0.3/64 ) * */ FIXP_DBL level = fMult( (FIXP_DBL)(10 << (METADATA_FRACT_BITS + LD_DATA_SHIFT)), fMult(FL2FXCONST_DBL(0.30102999566398119521373889472449f), ldLevel) + (FIXP_DBL)(FL2FXCONST_DBL(0.3f) >> LD_DATA_SHIFT)); /* level -= dialnorm + 31 */ /* this is fixed to Dolby-ReferenceLevel as compressor profiles are defined relative to this */ level -= ((FIXP_DBL)(dialnorm << (METADATA_FRACT_BITS - 16)) + (FIXP_DBL)(31 << METADATA_FRACT_BITS)); for (i = 0; i < 2; i++) { if (drcComp->profile[i] == DRC_NONE) { /* no compression */ drcComp->smoothGain[i] = FL2FXCONST_DBL(0.f); } else { FIXP_DBL gain, alpha, lvl2smthlvl; /* calc static gain */ if (level <= drcComp->maxBoostThr[i]) { /* max boost */ gain = drcComp->maxBoost[i]; } else if (level < drcComp->boostThr[i]) { /* boost range */ gain = fMult((level - drcComp->boostThr[i]), drcComp->boostFac[i]); } else if (level <= drcComp->earlyCutThr[i]) { /* null band */ gain = FL2FXCONST_DBL(0.f); } else if (level <= drcComp->cutThr[i]) { /* early cut range */ gain = fMult((level - drcComp->earlyCutThr[i]), drcComp->earlyCutFac[i]); } else if (level < drcComp->maxCutThr[i]) { /* cut range */ gain = fMult((level - drcComp->cutThr[i]), drcComp->cutFac[i]) - drcComp->maxEarlyCut[i]; } else { /* max cut */ gain = -drcComp->maxCut[i]; } /* choose time constant */ lvl2smthlvl = level - drcComp->smoothLevel[i]; if (gain < drcComp->smoothGain[i]) { /* attack */ if (lvl2smthlvl > drcComp->attackThr[i]) { /* fast attack */ alpha = drcComp->fastAttack[i]; } else { /* slow attack */ alpha = drcComp->slowAttack[i]; } } else { /* release */ if (lvl2smthlvl < -drcComp->decayThr[i]) { /* fast release */ alpha = drcComp->fastDecay[i]; } else { /* slow release */ alpha = drcComp->slowDecay[i]; } } /* smooth gain & level */ if ((gain < drcComp->smoothGain[i]) || (drcComp->holdCnt[i] == 0)) { /* hold gain unless we have an attack or hold period is over */ FIXP_DBL accu; /* drcComp->smoothLevel[i] = (1-alpha) * drcComp->smoothLevel[i] + * alpha * level; */ accu = fMult(((FIXP_DBL)MAXVAL_DBL - alpha), drcComp->smoothLevel[i]); accu += fMult(alpha, level); drcComp->smoothLevel[i] = accu; /* drcComp->smoothGain[i] = (1-alpha) * drcComp->smoothGain[i] + * alpha * gain; */ accu = fMult(((FIXP_DBL)MAXVAL_DBL - alpha), drcComp->smoothGain[i]); accu += fMult(alpha, gain); drcComp->smoothGain[i] = accu; } /* hold counter */ if (drcComp->holdCnt[i]) { drcComp->holdCnt[i]--; } if (gain < drcComp->smoothGain[i]) { drcComp->holdCnt[i] = drcComp->holdOff[i]; } } /* profile != DRC_NONE */ } /* for i=1..2 */ } else { /* no compression */ drcComp->smoothGain[0] = FL2FXCONST_DBL(0.f); drcComp->smoothGain[1] = FL2FXCONST_DBL(0.f); } /************************************************************************** * limiter **************************************************************************/ findPeakLevels(drcComp, inSamples, clev, slev, ext_leva, ext_levb, lfe_lev, (FIXP_DBL)((LONG)(dmxGain5) << (METADATA_FRACT_BITS - 16)), (FIXP_DBL)((LONG)(dmxGain2) << (METADATA_FRACT_BITS - 16)), peak); for (i = 0; i < 2; i++) { FIXP_DBL tmp = drcComp->prevPeak[i]; drcComp->prevPeak[i] = peak[i]; peak[i] = fixMax(peak[i], tmp); /* * Convert to dBFS, apply dialnorm */ /* descaled peak in ld64 representation */ FIXP_DBL ld_peak = CalcLdData(peak[i]) + (FIXP_DBL)((LONG)DOWNMIX_SHIFT << (DFRACT_BITS - 1 - LD_DATA_SHIFT)); /* if (peak < 1e-6) level = 1e-6f; */ ld_peak = fMax(ld_peak, FL2FXCONST_DBL(-0.31143075889569022011284244651463f)); /* peak[i] = 20 * log(peak[i])/log(10) + 0.2f + * (drcComp->smoothGain[i]*2^METADATA_FRACT_BITS) peak[i] = 20 * * log(2)/log(10) * ld(peak[i]) + 0.2f + * (drcComp->smoothGain[i]*2^METADATA_FRACT_BITS) peak[i] = 10 * * 2*0.30102999566398119521373889472449 * ld(peak[i]) + 0.2f + * (drcComp->smoothGain[i]*2^METADATA_FRACT_BITS) * * additional scaling with METADATA_FRACT_BITS: * peak[i] = (10 * 2*0.30102999566398119521373889472449 * ld64(peak[i]) * 64 * + 0.2f + * (drcComp->smoothGain[i]*2^METADATA_FRACT_BITS))*2^(-METADATA_FRACT_BITS) * peak[i] = 10*2^(METADATA_FRACT_BITS+LD_DATA_SHIFT) * * 2*0.30102999566398119521373889472449 * ld64(peak[i]) * + 0.2f*2^(-METADATA_FRACT_BITS) + drcComp->smoothGain[i] */ peak[i] = fMult( (FIXP_DBL)(10 << (METADATA_FRACT_BITS + LD_DATA_SHIFT)), fMult(FL2FX_DBL(2 * 0.30102999566398119521373889472449f), ld_peak)); peak[i] += (FL2FX_DBL(0.5f) >> METADATA_INT_BITS); /* add a little bit headroom */ peak[i] += drcComp->smoothGain[i]; } /* peak -= dialnorm + 31; */ /* this is Dolby style only */ peak[0] -= (FIXP_DBL)((dialnorm - drc_TargetRefLevel) << (METADATA_FRACT_BITS - 16)); /* peak[0] -= dialnorm - drc_TargetRefLevel */ /* peak += 11; */ /* this is Dolby style only */ /* RF mode output is 11dB higher */ /*peak += comp_TargetRefLevel - drc_TargetRefLevel;*/ peak[1] -= (FIXP_DBL)((dialnorm - comp_TargetRefLevel) << (METADATA_FRACT_BITS - 16)); /* peak[1] -= dialnorm - comp_TargetRefLevel */ /* limiter gain */ drcComp->limGain[0] += drcComp->limDecay; /* linear limiter release */ drcComp->limGain[0] = fixMin(drcComp->limGain[0], -peak[0]); drcComp->limGain[1] += 2 * drcComp->limDecay; /* linear limiter release */ drcComp->limGain[1] = fixMin(drcComp->limGain[1], -peak[1]); /*************************************************************************/ /* apply limiting, return DRC gains*/ { FIXP_DBL tmp; tmp = drcComp->smoothGain[0]; if (drcComp->limGain[0] < FL2FXCONST_DBL(0.f)) { tmp += drcComp->limGain[0]; } *pDynrng = (LONG)scaleValue(tmp, -(METADATA_FRACT_BITS - 16)); tmp = drcComp->smoothGain[1]; if (drcComp->limGain[1] < FL2FXCONST_DBL(0.f)) { tmp += drcComp->limGain[1]; } *pCompr = (LONG)scaleValue(tmp, -(METADATA_FRACT_BITS - 16)); } return 0; } DRC_PROFILE FDK_DRC_Generator_getDrcProfile(const HDRC_COMP drcComp) { return drcComp->profile[0]; } DRC_PROFILE FDK_DRC_Generator_getCompProfile(const HDRC_COMP drcComp) { return drcComp->profile[1]; } fdk-aac-2.0.2/libAACenc/src/metadata_compressor.h000066400000000000000000000227451404225310700215460ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Neusinger Description: Compressor for AAC Metadata Generator *******************************************************************************/ #ifndef METADATA_COMPRESSOR_H #define METADATA_COMPRESSOR_H #include "FDK_audio.h" #include "common_fix.h" #include "aacenc.h" #define LFE_LEV_SCALE 2 /** * DRC compression profiles. */ typedef enum DRC_PROFILE { DRC_NONE = 0, DRC_FILMSTANDARD = 1, DRC_FILMLIGHT = 2, DRC_MUSICSTANDARD = 3, DRC_MUSICLIGHT = 4, DRC_SPEECH = 5, DRC_DELAY_TEST = 6, DRC_NOT_PRESENT = -2 } DRC_PROFILE; /** * DRC Compressor handle. */ typedef struct DRC_COMP DRC_COMP, *HDRC_COMP; /** * \brief Open a DRC Compressor instance. * * Allocate memory for a compressor instance. * * \param phDrcComp A pointer to a compressor handle. Initialized on * return. * * \return * - 0, on succes. * - unequal 0, on failure. */ INT FDK_DRC_Generator_Open(HDRC_COMP *phDrcComp); /** * \brief Close the DRC Compressor instance. * * Deallocate instance and free whole memory. * * \param phDrcComp Pointer to the compressor handle to be * deallocated. * * \return * - 0, on succes. * - unequal 0, on failure. */ INT FDK_DRC_Generator_Close(HDRC_COMP *phDrcComp); /** * \brief Configure DRC Compressor. * * \param drcComp Compressor handle. * \param profileLine DRC profile for line mode. * \param profileRF DRC profile for RF mode. * \param blockLength Length of processing block in samples per * channel. * \param sampleRate Sampling rate in Hz. * \param channelMode Channel configuration. * \param channelOrder Channel order, MPEG or WAV. * \param useWeighting Use weighting filter for loudness calculation * * \return * - 0, on success, * - unequal 0, on failure */ INT FDK_DRC_Generator_Initialize(HDRC_COMP drcComp, const DRC_PROFILE profileLine, const DRC_PROFILE profileRF, const INT blockLength, const UINT sampleRate, const CHANNEL_MODE channelMode, const CHANNEL_ORDER channelOrder, const UCHAR useWeighting); /** * \brief Calculate DRC Compressor Gain. * * \param drcComp Compressor handle. * \param inSamples Pointer to interleaved input audio samples. * \param inSamplesBufSize Size of inSamples for one channel. * \param dialnorm Dialog Level in dB (typically -31...-1). * \param drc_TargetRefLevel * \param comp_TargetRefLevel * \param clev Downmix center mix factor (typically 0.707, * 0.595 or 0.5) * \param slev Downmix surround mix factor (typically 0.707, * 0.5, or 0) * \param ext_leva Downmix gain factor A * \param ext_levb Downmix gain factor B * \param lfe_lev LFE gain factor * \param dmxGain5 Gain factor for downmix to 5 channels * \param dmxGain2 Gain factor for downmix to 2 channels * \param dynrng Pointer to variable receiving line mode DRC gain * in dB * \param compr Pointer to variable receiving RF mode DRC gain * in dB * * \return * - 0, on success, * - unequal 0, on failure */ INT FDK_DRC_Generator_Calc(HDRC_COMP drcComp, const INT_PCM *const inSamples, const UINT inSamplesBufSize, const INT dialnorm, const INT drc_TargetRefLevel, const INT comp_TargetRefLevel, const FIXP_DBL clev, const FIXP_DBL slev, const FIXP_DBL ext_leva, const FIXP_DBL ext_levb, const FIXP_DBL lfe_lev, const INT dmxGain5, const INT dmxGain2, INT *const dynrng, INT *const compr); /** * \brief Configure DRC Compressor Profile. * * \param drcComp Compressor handle. * \param profileLine DRC profile for line mode. * \param profileRF DRC profile for RF mode. * * \return * - 0, on success, * - unequal 0, on failure */ INT FDK_DRC_Generator_setDrcProfile(HDRC_COMP drcComp, const DRC_PROFILE profileLine, const DRC_PROFILE profileRF); /** * \brief Get DRC profile for line mode. * * \param drcComp Compressor handle. * * \return Current Profile. */ DRC_PROFILE FDK_DRC_Generator_getDrcProfile(const HDRC_COMP drcComp); /** * \brief Get DRC profile for RF mode. * * \param drcComp Compressor handle. * * \return Current Profile. */ DRC_PROFILE FDK_DRC_Generator_getCompProfile(const HDRC_COMP drcComp); #endif /* METADATA_COMPRESSOR_H */ fdk-aac-2.0.2/libAACenc/src/metadata_main.cpp000066400000000000000000001310351404225310700206220ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): V. Bacigalupo Description: Metadata Encoder library interface functions *******************************************************************************/ #include "metadata_main.h" #include "metadata_compressor.h" #include "FDK_bitstream.h" #include "FDK_audio.h" #include "genericStds.h" /*----------------- defines ----------------------*/ #define MAX_DRC_BANDS (1 << 4) #define MAX_DRC_FRAMELEN (2 * 1024) #define MAX_DELAY_FRAMES (3) /*--------------- structure definitions --------------------*/ typedef struct AAC_METADATA { /* MPEG: Dynamic Range Control */ struct { UCHAR prog_ref_level_present; SCHAR prog_ref_level; UCHAR dyn_rng_sgn[MAX_DRC_BANDS]; UCHAR dyn_rng_ctl[MAX_DRC_BANDS]; UCHAR drc_bands_present; UCHAR drc_band_incr; UCHAR drc_band_top[MAX_DRC_BANDS]; UCHAR drc_interpolation_scheme; AACENC_METADATA_DRC_PROFILE drc_profile; INT drc_TargetRefLevel; /* used for Limiter */ /* excluded channels */ UCHAR excluded_chns_present; UCHAR exclude_mask[2]; /* MAX_NUMBER_CHANNELS/8 */ } mpegDrc; /* ETSI: addtl ancillary data */ struct { /* Heavy Compression */ UCHAR compression_on; /* flag, if compression value should be written */ UCHAR compression_value; /* compression value */ AACENC_METADATA_DRC_PROFILE comp_profile; INT comp_TargetRefLevel; /* used for Limiter */ INT timecode_coarse_status; INT timecode_fine_status; UCHAR extAncDataStatus; struct { UCHAR ext_downmix_lvl_status; UCHAR ext_downmix_gain_status; UCHAR ext_lfe_downmix_status; UCHAR ext_dmix_a_idx; /* extended downmix level (0..7, according to table) */ UCHAR ext_dmix_b_idx; /* extended downmix level (0..7, according to table) */ UCHAR dmx_gain_5_sgn; UCHAR dmx_gain_5_idx; UCHAR dmx_gain_2_sgn; UCHAR dmx_gain_2_idx; UCHAR ext_dmix_lfe_idx; /* extended downmix level for lfe (0..15, according to table) */ } extAncData; } etsiAncData; SCHAR centerMixLevel; /* center downmix level (0...7, according to table) */ SCHAR surroundMixLevel; /* surround downmix level (0...7, according to table) */ UCHAR WritePCEMixDwnIdx; /* flag */ UCHAR DmxLvl_On; /* flag */ UCHAR dolbySurroundMode; UCHAR drcPresentationMode; UCHAR metadataMode; /* indicate meta data mode in current frame (delay line) */ } AAC_METADATA; typedef struct FDK_METADATA_ENCODER { INT metadataMode; HDRC_COMP hDrcComp; AACENC_MetaData submittedMetaData; INT nAudioDataDelay; /* Additional delay to round up to next frame border (in samples) */ INT nMetaDataDelay; /* Meta data delay (in frames) */ INT nChannels; CHANNEL_MODE channelMode; INT_PCM* pAudioDelayBuffer; AAC_METADATA metaDataBuffer[MAX_DELAY_FRAMES]; INT metaDataDelayIdx; UCHAR drcInfoPayload[12]; UCHAR drcDsePayload[8]; INT matrix_mixdown_idx; AACENC_EXT_PAYLOAD exPayload[2]; INT nExtensions; UINT maxChannels; /* Maximum number of audio channels to be supported. */ INT finalizeMetaData; /* Delay switch off by one frame and write default configuration to finalize the metadata setup. */ INT initializeMetaData; /* Fill up delay line with first meta data info. This is required to have meta data already in first frame. */ } FDK_METADATA_ENCODER; /*---------------- constants -----------------------*/ static const AACENC_MetaData defaultMetaDataSetup = { AACENC_METADATA_DRC_NONE, /* drc_profile */ AACENC_METADATA_DRC_NOT_PRESENT, /* comp_profile */ -(31 << 16), /* drc_TargetRefLevel */ -(23 << 16), /* comp_TargetRefLevel */ 0, /* prog_ref_level_present */ -(23 << 16), /* prog_ref_level */ 0, /* PCE_mixdown_idx_present */ 0, /* ETSI_DmxLvl_present */ 0, /* centerMixLevel */ 0, /* surroundMixLevel */ 0, /* dolbySurroundMode */ 0, /* drcPresentationMode */ {0, 0, 0, 0, 0, 0, 0, 0, 0} /* ExtMetaData */ }; static const FIXP_DBL dmxTable[8] = { ((FIXP_DBL)MAXVAL_DBL), FL2FXCONST_DBL(0.841f), FL2FXCONST_DBL(0.707f), FL2FXCONST_DBL(0.596f), FL2FXCONST_DBL(0.500f), FL2FXCONST_DBL(0.422f), FL2FXCONST_DBL(0.355f), FL2FXCONST_DBL(0.000f)}; #define FL2DMXLFE(a) FL2FXCONST_DBL((a) / (1 << LFE_LEV_SCALE)) static const FIXP_DBL dmxLfeTable[16] = { FL2DMXLFE(3.162f), FL2DMXLFE(2.000f), FL2DMXLFE(1.679f), FL2DMXLFE(1.413f), FL2DMXLFE(1.189f), FL2DMXLFE(1.000f), FL2DMXLFE(0.841f), FL2DMXLFE(0.707f), FL2DMXLFE(0.596f), FL2DMXLFE(0.500f), FL2DMXLFE(0.316f), FL2DMXLFE(0.178f), FL2DMXLFE(0.100f), FL2DMXLFE(0.032f), FL2DMXLFE(0.010f), FL2DMXLFE(0.000f)}; static const UCHAR surmix2matrix_mixdown_idx[8] = {0, 0, 0, 1, 1, 2, 2, 3}; /*--------------- function declarations --------------------*/ static FDK_METADATA_ERROR WriteMetadataPayload( const HANDLE_FDK_METADATA_ENCODER hMetaData, const AAC_METADATA* const pMetadata); static INT WriteDynamicRangeInfoPayload(const AAC_METADATA* const pMetadata, UCHAR* const pExtensionPayload); static INT WriteEtsiAncillaryDataPayload(const AAC_METADATA* const pMetadata, UCHAR* const pExtensionPayload); static FDK_METADATA_ERROR CompensateAudioDelay( HANDLE_FDK_METADATA_ENCODER hMetaDataEnc, INT_PCM* const pAudioSamples, const UINT audioSamplesBufSize, const INT nAudioSamples); static FDK_METADATA_ERROR LoadSubmittedMetadata( const AACENC_MetaData* const hMetadata, const INT nChannels, const INT metadataMode, AAC_METADATA* const pAacMetaData); static FDK_METADATA_ERROR ProcessCompressor(AAC_METADATA* pMetadata, HDRC_COMP hDrcComp, const INT_PCM* const pSamples, const UINT samplesBufSize, const INT nSamples); /*------------- function definitions ----------------*/ static DRC_PROFILE convertProfile(AACENC_METADATA_DRC_PROFILE aacProfile) { DRC_PROFILE drcProfile = DRC_NONE; switch (aacProfile) { case AACENC_METADATA_DRC_NONE: drcProfile = DRC_NONE; break; case AACENC_METADATA_DRC_FILMSTANDARD: drcProfile = DRC_FILMSTANDARD; break; case AACENC_METADATA_DRC_FILMLIGHT: drcProfile = DRC_FILMLIGHT; break; case AACENC_METADATA_DRC_MUSICSTANDARD: drcProfile = DRC_MUSICSTANDARD; break; case AACENC_METADATA_DRC_MUSICLIGHT: drcProfile = DRC_MUSICLIGHT; break; case AACENC_METADATA_DRC_SPEECH: drcProfile = DRC_SPEECH; break; case AACENC_METADATA_DRC_NOT_PRESENT: drcProfile = DRC_NOT_PRESENT; break; default: drcProfile = DRC_NONE; break; } return drcProfile; } /* convert dialog normalization to program reference level */ /* NOTE: this only is correct, if the decoder target level is set to -31dB for * line mode / -20dB for RF mode */ static UCHAR dialnorm2progreflvl(const INT d) { return ((UCHAR)fMax(0, fMin((-d + (1 << 13)) >> 14, 127))); } /* convert program reference level to dialog normalization */ static INT progreflvl2dialnorm(const UCHAR p) { return -((INT)(p << (16 - 2))); } /* encode downmix levels to Downmixing_levels_MPEG4 */ static SCHAR encodeDmxLvls(const SCHAR cmixlev, const SCHAR surmixlev) { SCHAR dmxLvls = 0; dmxLvls |= 0x80 | (cmixlev << 4); /* center_mix_level_on */ dmxLvls |= 0x08 | surmixlev; /* surround_mix_level_on */ return dmxLvls; } /* encode AAC DRC gain (ISO/IEC 14496-3:2005 4.5.2.7) */ static void encodeDynrng(INT gain, UCHAR* const dyn_rng_ctl, UCHAR* const dyn_rng_sgn) { if (gain < 0) { *dyn_rng_sgn = 1; gain = -gain; } else { *dyn_rng_sgn = 0; } gain = fMin(gain, (127 << 14)); *dyn_rng_ctl = (UCHAR)((gain + (1 << 13)) >> 14); } /* decode AAC DRC gain (ISO/IEC 14496-3:2005 4.5.2.7) */ static INT decodeDynrng(const UCHAR dyn_rng_ctl, const UCHAR dyn_rng_sgn) { INT tmp = ((INT)dyn_rng_ctl << (16 - 2)); if (dyn_rng_sgn) tmp = -tmp; return tmp; } /* encode AAC compression value (ETSI TS 101 154 page 99) */ static UCHAR encodeCompr(INT gain) { UCHAR x, y; INT tmp; /* tmp = (int)((48.164f - gain) / 6.0206f * 15 + 0.5f); */ tmp = ((3156476 - gain) * 15 + 197283) / 394566; if (tmp >= 240) { return 0xFF; } else if (tmp < 0) { return 0; } else { x = tmp / 15; y = tmp % 15; } return (x << 4) | y; } /* decode AAC compression value (ETSI TS 101 154 page 99) */ static INT decodeCompr(const UCHAR compr) { INT gain; SCHAR x = compr >> 4; /* 4 MSB of compr */ UCHAR y = (compr & 0x0F); /* 4 LSB of compr */ /* gain = (INT)((48.164f - 6.0206f * x - 0.4014f * y) ); */ gain = (INT)( scaleValue((FIXP_DBL)(((LONG)FL2FXCONST_DBL(6.0206f / 128.f) * (8 - x) - (LONG)FL2FXCONST_DBL(0.4014f / 128.f) * y)), -(DFRACT_BITS - 1 - 7 - 16))); return gain; } FDK_METADATA_ERROR FDK_MetadataEnc_Open(HANDLE_FDK_METADATA_ENCODER* phMetaData, const UINT maxChannels) { FDK_METADATA_ERROR err = METADATA_OK; HANDLE_FDK_METADATA_ENCODER hMetaData = NULL; if (phMetaData == NULL) { err = METADATA_INVALID_HANDLE; goto bail; } /* allocate memory */ if (NULL == (hMetaData = (HANDLE_FDK_METADATA_ENCODER)FDKcalloc( 1, sizeof(FDK_METADATA_ENCODER)))) { err = METADATA_MEMORY_ERROR; goto bail; } FDKmemclear(hMetaData, sizeof(FDK_METADATA_ENCODER)); if (NULL == (hMetaData->pAudioDelayBuffer = (INT_PCM*)FDKcalloc( maxChannels * MAX_DRC_FRAMELEN, sizeof(INT_PCM)))) { err = METADATA_MEMORY_ERROR; goto bail; } FDKmemclear(hMetaData->pAudioDelayBuffer, maxChannels * MAX_DRC_FRAMELEN * sizeof(INT_PCM)); hMetaData->maxChannels = maxChannels; /* Allocate DRC Compressor. */ if (FDK_DRC_Generator_Open(&hMetaData->hDrcComp) != 0) { err = METADATA_MEMORY_ERROR; goto bail; } hMetaData->channelMode = MODE_UNKNOWN; /* Return metadata instance */ *phMetaData = hMetaData; return err; bail: FDK_MetadataEnc_Close(&hMetaData); return err; } FDK_METADATA_ERROR FDK_MetadataEnc_Close( HANDLE_FDK_METADATA_ENCODER* phMetaData) { FDK_METADATA_ERROR err = METADATA_OK; if (phMetaData == NULL) { err = METADATA_INVALID_HANDLE; goto bail; } if (*phMetaData != NULL) { FDK_DRC_Generator_Close(&(*phMetaData)->hDrcComp); FDKfree((*phMetaData)->pAudioDelayBuffer); FDKfree(*phMetaData); *phMetaData = NULL; } bail: return err; } FDK_METADATA_ERROR FDK_MetadataEnc_Init( HANDLE_FDK_METADATA_ENCODER hMetaData, const INT resetStates, const INT metadataMode, const INT audioDelay, const UINT frameLength, const UINT sampleRate, const UINT nChannels, const CHANNEL_MODE channelMode, const CHANNEL_ORDER channelOrder) { FDK_METADATA_ERROR err = METADATA_OK; int i, nFrames, delay; if (hMetaData == NULL) { err = METADATA_INVALID_HANDLE; goto bail; } /* Determine values for delay compensation. */ for (nFrames = 0, delay = audioDelay - (INT)frameLength; delay > 0; delay -= (INT)frameLength, nFrames++) ; if ((nChannels > (8)) || (nChannels > hMetaData->maxChannels) || ((-delay) > MAX_DRC_FRAMELEN) || nFrames >= MAX_DELAY_FRAMES) { err = METADATA_INIT_ERROR; goto bail; } /* Initialize with default setup. */ FDKmemcpy(&hMetaData->submittedMetaData, &defaultMetaDataSetup, sizeof(AACENC_MetaData)); hMetaData->finalizeMetaData = 0; /* finalize meta data only while on/off switching, else disabled */ hMetaData->initializeMetaData = 0; /* fill up meta data delay line only at a reset otherwise disabled */ /* Reset delay lines. */ if (resetStates || (hMetaData->nAudioDataDelay != -delay) || (hMetaData->channelMode != channelMode)) { if (resetStates || (hMetaData->channelMode == MODE_UNKNOWN)) { /* clear delay buffer */ FDKmemclear(hMetaData->pAudioDelayBuffer, hMetaData->maxChannels * MAX_DRC_FRAMELEN * sizeof(INT_PCM)); } else { /* if possible, keep static audio channels for seamless channel * reconfiguration */ FDK_channelMapDescr mapDescrPrev, mapDescr; int c, src[2] = {-1, -1}, dst[2] = {-1, -1}; if (channelOrder == CH_ORDER_WG4) { FDK_chMapDescr_init(&mapDescrPrev, FDK_mapInfoTabWg4, FDK_mapInfoTabLenWg4, 0); FDK_chMapDescr_init(&mapDescr, FDK_mapInfoTabWg4, FDK_mapInfoTabLenWg4, 0); } else { FDK_chMapDescr_init(&mapDescrPrev, NULL, 0, (channelOrder == CH_ORDER_MPEG) ? 1 : 0); FDK_chMapDescr_init(&mapDescr, NULL, 0, (channelOrder == CH_ORDER_MPEG) ? 1 : 0); } switch (channelMode) { case MODE_1: if ((INT)nChannels != 2) { /* preserve center channel */ src[0] = FDK_chMapDescr_getMapValue(&mapDescrPrev, 0, hMetaData->channelMode); dst[0] = FDK_chMapDescr_getMapValue(&mapDescr, 0, channelMode); } break; case MODE_2: case MODE_1_2: case MODE_1_2_1: case MODE_1_2_2: case MODE_1_2_2_1: if (hMetaData->nChannels >= 2) { /* preserve left/right channel */ src[0] = FDK_chMapDescr_getMapValue( &mapDescrPrev, ((hMetaData->channelMode == 2) ? 0 : 1), hMetaData->channelMode); src[1] = FDK_chMapDescr_getMapValue( &mapDescrPrev, ((hMetaData->channelMode == 2) ? 1 : 2), hMetaData->channelMode); dst[0] = FDK_chMapDescr_getMapValue( &mapDescr, ((channelMode == 2) ? 0 : 1), channelMode); dst[1] = FDK_chMapDescr_getMapValue( &mapDescr, ((channelMode == 2) ? 1 : 2), channelMode); } break; default:; } C_ALLOC_SCRATCH_START(scratch_audioDelayBuffer, INT_PCM, (8)); FDKmemclear(scratch_audioDelayBuffer, (8) * sizeof(INT_PCM)); i = (hMetaData->nChannels > (INT)nChannels) ? 0 : hMetaData->nAudioDataDelay - 1; do { for (c = 0; c < 2; c++) { if (src[c] != -1 && dst[c] != -1) { scratch_audioDelayBuffer[dst[c]] = hMetaData->pAudioDelayBuffer[i * hMetaData->nChannels + src[c]]; } } FDKmemcpy(&hMetaData->pAudioDelayBuffer[i * nChannels], scratch_audioDelayBuffer, nChannels * sizeof(INT_PCM)); i += (hMetaData->nChannels > (INT)nChannels) ? 1 : -1; } while ((i < hMetaData->nAudioDataDelay) && (i >= 0)); C_ALLOC_SCRATCH_END(scratch_audioDelayBuffer, INT_PCM, (8)); } FDKmemclear(hMetaData->metaDataBuffer, sizeof(hMetaData->metaDataBuffer)); hMetaData->metaDataDelayIdx = 0; hMetaData->initializeMetaData = 1; /* fill up delay line with first meta data info */ } else { /* Enable meta data. */ if ((hMetaData->metadataMode == 0) && (metadataMode != 0)) { /* disable meta data in all delay lines */ for (i = 0; i < (int)(sizeof(hMetaData->metaDataBuffer) / sizeof(AAC_METADATA)); i++) { LoadSubmittedMetadata(&hMetaData->submittedMetaData, nChannels, 0, &hMetaData->metaDataBuffer[i]); } } /* Disable meta data.*/ if ((hMetaData->metadataMode != 0) && (metadataMode == 0)) { hMetaData->finalizeMetaData = hMetaData->metadataMode; } } /* Initialize delay. */ hMetaData->nAudioDataDelay = -delay; hMetaData->nMetaDataDelay = nFrames; hMetaData->nChannels = nChannels; hMetaData->channelMode = channelMode; hMetaData->metadataMode = metadataMode; /* Initialize compressor. */ if ((metadataMode == 1) || (metadataMode == 2)) { if (FDK_DRC_Generator_Initialize(hMetaData->hDrcComp, DRC_NONE, DRC_NONE, frameLength, sampleRate, channelMode, channelOrder, 1) != 0) { err = METADATA_INIT_ERROR; } } bail: return err; } static FDK_METADATA_ERROR ProcessCompressor(AAC_METADATA* pMetadata, HDRC_COMP hDrcComp, const INT_PCM* const pSamples, const UINT samplesBufSize, const INT nSamples) { FDK_METADATA_ERROR err = METADATA_OK; INT dynrng, compr; INT dmxGain5, dmxGain2; DRC_PROFILE profileDrc; DRC_PROFILE profileComp; if ((pMetadata == NULL) || (hDrcComp == NULL)) { err = METADATA_INVALID_HANDLE; return err; } profileDrc = convertProfile(pMetadata->mpegDrc.drc_profile); profileComp = convertProfile(pMetadata->etsiAncData.comp_profile); /* first, check if profile is same as last frame * otherwise, update setup */ if ((profileDrc != FDK_DRC_Generator_getDrcProfile(hDrcComp)) || (profileComp != FDK_DRC_Generator_getCompProfile(hDrcComp))) { FDK_DRC_Generator_setDrcProfile(hDrcComp, profileDrc, profileComp); } /* Sanity check */ if (profileComp == DRC_NONE) { pMetadata->etsiAncData.compression_value = 0x80; /* to ensure no external values will be written if not configured */ } /* in case of embedding external values, copy this now (limiter may overwrite * them) */ dynrng = decodeDynrng(pMetadata->mpegDrc.dyn_rng_ctl[0], pMetadata->mpegDrc.dyn_rng_sgn[0]); compr = decodeCompr(pMetadata->etsiAncData.compression_value); dmxGain5 = decodeDynrng(pMetadata->etsiAncData.extAncData.dmx_gain_5_idx, pMetadata->etsiAncData.extAncData.dmx_gain_5_sgn); dmxGain2 = decodeDynrng(pMetadata->etsiAncData.extAncData.dmx_gain_2_idx, pMetadata->etsiAncData.extAncData.dmx_gain_2_sgn); /* Call compressor */ if (FDK_DRC_Generator_Calc( hDrcComp, pSamples, samplesBufSize, progreflvl2dialnorm(pMetadata->mpegDrc.prog_ref_level), pMetadata->mpegDrc.drc_TargetRefLevel, pMetadata->etsiAncData.comp_TargetRefLevel, dmxTable[pMetadata->centerMixLevel], dmxTable[pMetadata->surroundMixLevel], dmxTable[pMetadata->etsiAncData.extAncData.ext_dmix_a_idx], dmxTable[pMetadata->etsiAncData.extAncData.ext_dmix_b_idx], pMetadata->etsiAncData.extAncData.ext_lfe_downmix_status ? dmxLfeTable[pMetadata->etsiAncData.extAncData.ext_dmix_lfe_idx] : (FIXP_DBL)0, dmxGain5, dmxGain2, &dynrng, &compr) != 0) { err = METADATA_ENCODE_ERROR; goto bail; } /* Write DRC values */ pMetadata->mpegDrc.drc_band_incr = 0; encodeDynrng(dynrng, pMetadata->mpegDrc.dyn_rng_ctl, pMetadata->mpegDrc.dyn_rng_sgn); pMetadata->etsiAncData.compression_value = encodeCompr(compr); bail: return err; } FDK_METADATA_ERROR FDK_MetadataEnc_Process( HANDLE_FDK_METADATA_ENCODER hMetaDataEnc, INT_PCM* const pAudioSamples, const UINT audioSamplesBufSize, const INT nAudioSamples, const AACENC_MetaData* const pMetadata, AACENC_EXT_PAYLOAD** ppMetaDataExtPayload, UINT* nMetaDataExtensions, INT* matrix_mixdown_idx) { FDK_METADATA_ERROR err = METADATA_OK; int metaDataDelayWriteIdx, metaDataDelayReadIdx, metadataMode; /* Where to write new meta data info */ metaDataDelayWriteIdx = hMetaDataEnc->metaDataDelayIdx; /* How to write the data */ metadataMode = hMetaDataEnc->metadataMode; /* Compensate meta data delay. */ hMetaDataEnc->metaDataDelayIdx++; if (hMetaDataEnc->metaDataDelayIdx > hMetaDataEnc->nMetaDataDelay) hMetaDataEnc->metaDataDelayIdx = 0; /* Where to read pending meta data info from. */ metaDataDelayReadIdx = hMetaDataEnc->metaDataDelayIdx; /* Submit new data if available. */ if (pMetadata != NULL) { FDKmemcpy(&hMetaDataEnc->submittedMetaData, pMetadata, sizeof(AACENC_MetaData)); } /* Write one additional frame with default configuration of meta data. Ensure * defined behaviour on decoder side. */ if ((hMetaDataEnc->finalizeMetaData != 0) && (hMetaDataEnc->metadataMode == 0)) { FDKmemcpy(&hMetaDataEnc->submittedMetaData, &defaultMetaDataSetup, sizeof(AACENC_MetaData)); metadataMode = hMetaDataEnc->finalizeMetaData; hMetaDataEnc->finalizeMetaData = 0; } /* Get last submitted data. */ if ((err = LoadSubmittedMetadata( &hMetaDataEnc->submittedMetaData, hMetaDataEnc->nChannels, metadataMode, &hMetaDataEnc->metaDataBuffer[metaDataDelayWriteIdx])) != METADATA_OK) { goto bail; } /* Calculate compressor if necessary and updata meta data info */ if ((hMetaDataEnc->metaDataBuffer[metaDataDelayWriteIdx].metadataMode == 1) || (hMetaDataEnc->metaDataBuffer[metaDataDelayWriteIdx].metadataMode == 2)) { if ((err = ProcessCompressor( &hMetaDataEnc->metaDataBuffer[metaDataDelayWriteIdx], hMetaDataEnc->hDrcComp, pAudioSamples, audioSamplesBufSize, nAudioSamples)) != METADATA_OK) { /* Get last submitted data again. */ LoadSubmittedMetadata( &hMetaDataEnc->submittedMetaData, hMetaDataEnc->nChannels, metadataMode, &hMetaDataEnc->metaDataBuffer[metaDataDelayWriteIdx]); } } /* Fill up delay line with initial meta data info.*/ if ((hMetaDataEnc->initializeMetaData != 0) && (hMetaDataEnc->metadataMode != 0)) { int i; for (i = 0; i < (int)(sizeof(hMetaDataEnc->metaDataBuffer) / sizeof(AAC_METADATA)); i++) { if (i != metaDataDelayWriteIdx) { FDKmemcpy(&hMetaDataEnc->metaDataBuffer[i], &hMetaDataEnc->metaDataBuffer[metaDataDelayWriteIdx], sizeof(hMetaDataEnc->metaDataBuffer[metaDataDelayWriteIdx])); } } hMetaDataEnc->initializeMetaData = 0; } /* Convert Meta Data side info to bitstream data. */ FDK_ASSERT(metaDataDelayReadIdx < MAX_DELAY_FRAMES); if ((err = WriteMetadataPayload( hMetaDataEnc, &hMetaDataEnc->metaDataBuffer[metaDataDelayReadIdx])) != METADATA_OK) { goto bail; } /* Assign meta data to output */ *ppMetaDataExtPayload = hMetaDataEnc->exPayload; *nMetaDataExtensions = hMetaDataEnc->nExtensions; *matrix_mixdown_idx = hMetaDataEnc->matrix_mixdown_idx; bail: /* Compensate audio delay, reset err status. */ err = CompensateAudioDelay(hMetaDataEnc, pAudioSamples, audioSamplesBufSize, nAudioSamples / hMetaDataEnc->nChannels); return err; } static FDK_METADATA_ERROR CompensateAudioDelay( HANDLE_FDK_METADATA_ENCODER hMetaDataEnc, INT_PCM* const pAudioSamples, const UINT audioSamplesBufSize, const INT nAudioSamples) { FDK_METADATA_ERROR err = METADATA_OK; if (hMetaDataEnc->nAudioDataDelay) { C_ALLOC_SCRATCH_START(scratch_audioDelayBuffer, INT_PCM, 1024); for (int c = 0; c < hMetaDataEnc->nChannels; c++) { int M = 1024; INT_PCM* pAudioSamples2 = pAudioSamples + c * audioSamplesBufSize; int delayIdx = hMetaDataEnc->nAudioDataDelay; do { M = fMin(M, delayIdx); delayIdx -= M; FDKmemcpy(&scratch_audioDelayBuffer[0], &pAudioSamples2[(nAudioSamples - M)], sizeof(INT_PCM) * M); FDKmemmove(&pAudioSamples2[M], &pAudioSamples2[0], sizeof(INT_PCM) * (nAudioSamples - M)); FDKmemcpy( &pAudioSamples2[0], &hMetaDataEnc->pAudioDelayBuffer[delayIdx + c * hMetaDataEnc->nAudioDataDelay], sizeof(INT_PCM) * M); FDKmemcpy( &hMetaDataEnc->pAudioDelayBuffer[delayIdx + c * hMetaDataEnc->nAudioDataDelay], &scratch_audioDelayBuffer[0], sizeof(INT_PCM) * M); } while (delayIdx > 0); } C_ALLOC_SCRATCH_END(scratch_audioDelayBuffer, INT_PCM, 1024); } return err; } /*----------------------------------------------------------------------------- functionname: WriteMetadataPayload description: fills anc data and extension payload returns: Error status ------------------------------------------------------------------------------*/ static FDK_METADATA_ERROR WriteMetadataPayload( const HANDLE_FDK_METADATA_ENCODER hMetaData, const AAC_METADATA* const pMetadata) { FDK_METADATA_ERROR err = METADATA_OK; if ((hMetaData == NULL) || (pMetadata == NULL)) { err = METADATA_INVALID_HANDLE; goto bail; } hMetaData->nExtensions = 0; hMetaData->matrix_mixdown_idx = -1; if (pMetadata->metadataMode != 0) { /* AAC-DRC */ if ((pMetadata->metadataMode == 1) || (pMetadata->metadataMode == 2)) { hMetaData->exPayload[hMetaData->nExtensions].pData = hMetaData->drcInfoPayload; hMetaData->exPayload[hMetaData->nExtensions].dataType = EXT_DYNAMIC_RANGE; hMetaData->exPayload[hMetaData->nExtensions].associatedChElement = -1; hMetaData->exPayload[hMetaData->nExtensions].dataSize = WriteDynamicRangeInfoPayload( pMetadata, hMetaData->exPayload[hMetaData->nExtensions].pData); hMetaData->nExtensions++; } /* pMetadata->metadataMode==1 || pMetadata->metadataMode==2 */ /* Matrix Mixdown Coefficient in PCE */ if (pMetadata->WritePCEMixDwnIdx) { hMetaData->matrix_mixdown_idx = surmix2matrix_mixdown_idx[pMetadata->surroundMixLevel]; } /* ETSI TS 101 154 (DVB) - MPEG4 ancillary_data() */ if ((pMetadata->metadataMode == 2) || (pMetadata->metadataMode == 3)) /* MP4_METADATA_MPEG_ETSI */ { hMetaData->exPayload[hMetaData->nExtensions].pData = hMetaData->drcDsePayload; hMetaData->exPayload[hMetaData->nExtensions].dataType = EXT_DATA_ELEMENT; hMetaData->exPayload[hMetaData->nExtensions].associatedChElement = -1; hMetaData->exPayload[hMetaData->nExtensions].dataSize = WriteEtsiAncillaryDataPayload( pMetadata, hMetaData->exPayload[hMetaData->nExtensions].pData); hMetaData->nExtensions++; } /* metadataMode==2 || metadataMode==3 */ } /* metadataMode != 0 */ bail: return err; } static INT WriteDynamicRangeInfoPayload(const AAC_METADATA* const pMetadata, UCHAR* const pExtensionPayload) { const INT pce_tag_present = 0; /* yet fixed setting! */ const INT prog_ref_lev_res_bits = 0; INT i, drc_num_bands = 1; FDK_BITSTREAM bsWriter; FDKinitBitStream(&bsWriter, pExtensionPayload, 16, 0, BS_WRITER); /* dynamic_range_info() */ FDKwriteBits(&bsWriter, pce_tag_present, 1); /* pce_tag_present */ if (pce_tag_present) { FDKwriteBits(&bsWriter, 0x0, 4); /* pce_instance_tag */ FDKwriteBits(&bsWriter, 0x0, 4); /* drc_tag_reserved_bits */ } /* Exclude channels */ FDKwriteBits(&bsWriter, (pMetadata->mpegDrc.excluded_chns_present) ? 1 : 0, 1); /* excluded_chns_present*/ /* Multiband DRC */ FDKwriteBits(&bsWriter, (pMetadata->mpegDrc.drc_bands_present) ? 1 : 0, 1); /* drc_bands_present */ if (pMetadata->mpegDrc.drc_bands_present) { FDKwriteBits(&bsWriter, pMetadata->mpegDrc.drc_band_incr, 4); /* drc_band_incr */ FDKwriteBits(&bsWriter, pMetadata->mpegDrc.drc_interpolation_scheme, 4); /* drc_interpolation_scheme */ drc_num_bands += pMetadata->mpegDrc.drc_band_incr; for (i = 0; i < drc_num_bands; i++) { FDKwriteBits(&bsWriter, pMetadata->mpegDrc.drc_band_top[i], 8); /* drc_band_top */ } } /* Program Reference Level */ FDKwriteBits(&bsWriter, pMetadata->mpegDrc.prog_ref_level_present, 1); /* prog_ref_level_present */ if (pMetadata->mpegDrc.prog_ref_level_present) { FDKwriteBits(&bsWriter, pMetadata->mpegDrc.prog_ref_level, 7); /* prog_ref_level */ FDKwriteBits(&bsWriter, prog_ref_lev_res_bits, 1); /* prog_ref_level_reserved_bits */ } /* DRC Values */ for (i = 0; i < drc_num_bands; i++) { FDKwriteBits(&bsWriter, (pMetadata->mpegDrc.dyn_rng_sgn[i]) ? 1 : 0, 1); /* dyn_rng_sgn[ */ FDKwriteBits(&bsWriter, pMetadata->mpegDrc.dyn_rng_ctl[i], 7); /* dyn_rng_ctl */ } /* return number of valid bits in extension payload. */ return FDKgetValidBits(&bsWriter); } static INT WriteEtsiAncillaryDataPayload(const AAC_METADATA* const pMetadata, UCHAR* const pExtensionPayload) { FDK_BITSTREAM bsWriter; FDKinitBitStream(&bsWriter, pExtensionPayload, 16, 0, BS_WRITER); /* ancillary_data_sync */ FDKwriteBits(&bsWriter, 0xBC, 8); /* bs_info */ FDKwriteBits(&bsWriter, 0x3, 2); /* mpeg_audio_type */ FDKwriteBits(&bsWriter, pMetadata->dolbySurroundMode, 2); /* dolby_surround_mode */ FDKwriteBits(&bsWriter, pMetadata->drcPresentationMode, 2); /* DRC presentation mode */ FDKwriteBits(&bsWriter, 0x0, 1); /* stereo_downmix_mode */ FDKwriteBits(&bsWriter, 0x0, 1); /* reserved */ /* ancillary_data_status */ FDKwriteBits(&bsWriter, 0, 3); /* 3 bit Reserved, set to "0" */ FDKwriteBits(&bsWriter, (pMetadata->DmxLvl_On) ? 1 : 0, 1); /* downmixing_levels_MPEG4_status */ FDKwriteBits(&bsWriter, pMetadata->etsiAncData.extAncDataStatus, 1); /* ext_anc_data_status */ FDKwriteBits(&bsWriter, (pMetadata->etsiAncData.compression_on) ? 1 : 0, 1); /* audio_coding_mode_and_compression status */ FDKwriteBits(&bsWriter, (pMetadata->etsiAncData.timecode_coarse_status) ? 1 : 0, 1); /* coarse_grain_timecode_status */ FDKwriteBits(&bsWriter, (pMetadata->etsiAncData.timecode_fine_status) ? 1 : 0, 1); /* fine_grain_timecode_status */ /* downmixing_levels_MPEG4_status */ if (pMetadata->DmxLvl_On) { FDKwriteBits( &bsWriter, encodeDmxLvls(pMetadata->centerMixLevel, pMetadata->surroundMixLevel), 8); } /* audio_coding_mode_and_compression_status */ if (pMetadata->etsiAncData.compression_on) { FDKwriteBits(&bsWriter, 0x01, 8); /* audio coding mode */ FDKwriteBits(&bsWriter, pMetadata->etsiAncData.compression_value, 8); /* compression value */ } /* grain-timecode coarse/fine */ if (pMetadata->etsiAncData.timecode_coarse_status) { FDKwriteBits(&bsWriter, 0x0, 16); /* not yet supported */ } if (pMetadata->etsiAncData.timecode_fine_status) { FDKwriteBits(&bsWriter, 0x0, 16); /* not yet supported */ } /* extended ancillary data structure */ if (pMetadata->etsiAncData.extAncDataStatus) { /* ext_ancillary_data_status */ FDKwriteBits(&bsWriter, 0, 1); /* Reserved, set to "0" */ FDKwriteBits(&bsWriter, pMetadata->etsiAncData.extAncData.ext_downmix_lvl_status, 1); /* ext_downmixing_levels_status */ FDKwriteBits(&bsWriter, pMetadata->etsiAncData.extAncData.ext_downmix_gain_status, 1); /* ext_downmixing_global_gains_status */ FDKwriteBits(&bsWriter, pMetadata->etsiAncData.extAncData.ext_lfe_downmix_status, 1); /* ext_downmixing_lfe_level_status" */ FDKwriteBits(&bsWriter, 0, 4); /* Reserved, set to "0" */ /* ext_downmixing_levels */ if (pMetadata->etsiAncData.extAncData.ext_downmix_lvl_status) { FDKwriteBits(&bsWriter, pMetadata->etsiAncData.extAncData.ext_dmix_a_idx, 3); /* dmix_a_idx */ FDKwriteBits(&bsWriter, pMetadata->etsiAncData.extAncData.ext_dmix_b_idx, 3); /* dmix_b_idx */ FDKwriteBits(&bsWriter, 0, 2); /* Reserved, set to "0" */ } /* ext_downmixing_gains */ if (pMetadata->etsiAncData.extAncData.ext_downmix_gain_status) { FDKwriteBits(&bsWriter, pMetadata->etsiAncData.extAncData.dmx_gain_5_sgn, 1); /* dmx_gain_5_sign */ FDKwriteBits(&bsWriter, pMetadata->etsiAncData.extAncData.dmx_gain_5_idx, 6); /* dmx_gain_5_idx */ FDKwriteBits(&bsWriter, 0, 1); /* Reserved, set to "0" */ FDKwriteBits(&bsWriter, pMetadata->etsiAncData.extAncData.dmx_gain_2_sgn, 1); /* dmx_gain_2_sign */ FDKwriteBits(&bsWriter, pMetadata->etsiAncData.extAncData.dmx_gain_2_idx, 6); /* dmx_gain_2_idx */ FDKwriteBits(&bsWriter, 0, 1); /* Reserved, set to "0" */ } if (pMetadata->etsiAncData.extAncData.ext_lfe_downmix_status) { FDKwriteBits(&bsWriter, pMetadata->etsiAncData.extAncData.ext_dmix_lfe_idx, 4); /* dmix_lfe_idx */ FDKwriteBits(&bsWriter, 0, 4); /* Reserved, set to "0" */ } } return FDKgetValidBits(&bsWriter); } static FDK_METADATA_ERROR LoadSubmittedMetadata( const AACENC_MetaData* const hMetadata, const INT nChannels, const INT metadataMode, AAC_METADATA* const pAacMetaData) { FDK_METADATA_ERROR err = METADATA_OK; if (pAacMetaData == NULL) { err = METADATA_INVALID_HANDLE; } else { /* init struct */ FDKmemclear(pAacMetaData, sizeof(AAC_METADATA)); if (hMetadata != NULL) { /* convert data */ pAacMetaData->mpegDrc.drc_profile = hMetadata->drc_profile; pAacMetaData->etsiAncData.comp_profile = hMetadata->comp_profile; pAacMetaData->mpegDrc.drc_TargetRefLevel = hMetadata->drc_TargetRefLevel; pAacMetaData->etsiAncData.comp_TargetRefLevel = hMetadata->comp_TargetRefLevel; pAacMetaData->mpegDrc.prog_ref_level_present = hMetadata->prog_ref_level_present; pAacMetaData->mpegDrc.prog_ref_level = dialnorm2progreflvl(hMetadata->prog_ref_level); pAacMetaData->centerMixLevel = hMetadata->centerMixLevel; pAacMetaData->surroundMixLevel = hMetadata->surroundMixLevel; pAacMetaData->WritePCEMixDwnIdx = hMetadata->PCE_mixdown_idx_present; pAacMetaData->DmxLvl_On = hMetadata->ETSI_DmxLvl_present; pAacMetaData->etsiAncData.compression_on = (hMetadata->comp_profile == AACENC_METADATA_DRC_NOT_PRESENT ? 0 : 1); if (pAacMetaData->mpegDrc.drc_profile == AACENC_METADATA_DRC_NOT_PRESENT) { pAacMetaData->mpegDrc.drc_profile = AACENC_METADATA_DRC_NONE; /* MPEG DRC gains are always present in BS syntax */ /* we should give a warning, but ErrorHandler does not support this */ } if (nChannels == 2) { pAacMetaData->dolbySurroundMode = hMetadata->dolbySurroundMode; /* dolby_surround_mode */ } else { pAacMetaData->dolbySurroundMode = 0; } pAacMetaData->drcPresentationMode = hMetadata->drcPresentationMode; /* override external values if DVB DRC presentation mode is given */ if (pAacMetaData->drcPresentationMode == 1) { pAacMetaData->mpegDrc.drc_TargetRefLevel = fMax(-(31 << 16), pAacMetaData->mpegDrc.drc_TargetRefLevel); pAacMetaData->etsiAncData.comp_TargetRefLevel = fMax( -(20 << 16), pAacMetaData->etsiAncData.comp_TargetRefLevel); /* implies -23dB */ } if (pAacMetaData->drcPresentationMode == 2) { pAacMetaData->mpegDrc.drc_TargetRefLevel = fMax(-(23 << 16), pAacMetaData->mpegDrc.drc_TargetRefLevel); pAacMetaData->etsiAncData.comp_TargetRefLevel = fMax(-(23 << 16), pAacMetaData->etsiAncData.comp_TargetRefLevel); } if (pAacMetaData->etsiAncData.comp_profile == AACENC_METADATA_DRC_NOT_PRESENT) { /* DVB defines to revert to Light DRC if heavy is not present */ if (pAacMetaData->drcPresentationMode != 0) { /* we exclude the "not indicated" mode as this requires the user to * define desired levels anyway */ pAacMetaData->mpegDrc.drc_TargetRefLevel = fMax(pAacMetaData->etsiAncData.comp_TargetRefLevel, pAacMetaData->mpegDrc.drc_TargetRefLevel); } } pAacMetaData->etsiAncData.timecode_coarse_status = 0; /* not yet supported - attention: Update GetEstMetadataBytesPerFrame() if enable this! */ pAacMetaData->etsiAncData.timecode_fine_status = 0; /* not yet supported - attention: Update GetEstMetadataBytesPerFrame() if enable this! */ /* extended ancillary data */ pAacMetaData->etsiAncData.extAncDataStatus = ((hMetadata->ExtMetaData.extAncDataEnable == 1) ? 1 : 0); if (pAacMetaData->etsiAncData.extAncDataStatus) { pAacMetaData->etsiAncData.extAncData.ext_downmix_lvl_status = (hMetadata->ExtMetaData.extDownmixLevelEnable ? 1 : 0); pAacMetaData->etsiAncData.extAncData.ext_downmix_gain_status = (hMetadata->ExtMetaData.dmxGainEnable ? 1 : 0); pAacMetaData->etsiAncData.extAncData.ext_lfe_downmix_status = (hMetadata->ExtMetaData.lfeDmxEnable ? 1 : 0); pAacMetaData->etsiAncData.extAncData.ext_dmix_a_idx = hMetadata->ExtMetaData.extDownmixLevel_A; pAacMetaData->etsiAncData.extAncData.ext_dmix_b_idx = hMetadata->ExtMetaData.extDownmixLevel_B; if (pAacMetaData->etsiAncData.extAncData.ext_downmix_gain_status) { encodeDynrng(hMetadata->ExtMetaData.dmxGain5, &(pAacMetaData->etsiAncData.extAncData.dmx_gain_5_idx), &(pAacMetaData->etsiAncData.extAncData.dmx_gain_5_sgn)); encodeDynrng(hMetadata->ExtMetaData.dmxGain2, &(pAacMetaData->etsiAncData.extAncData.dmx_gain_2_idx), &(pAacMetaData->etsiAncData.extAncData.dmx_gain_2_sgn)); } else { encodeDynrng(1 << 16, &(pAacMetaData->etsiAncData.extAncData.dmx_gain_5_idx), &(pAacMetaData->etsiAncData.extAncData.dmx_gain_5_sgn)); encodeDynrng(1 << 16, &(pAacMetaData->etsiAncData.extAncData.dmx_gain_2_idx), &(pAacMetaData->etsiAncData.extAncData.dmx_gain_2_sgn)); } if (pAacMetaData->etsiAncData.extAncData.ext_lfe_downmix_status) { pAacMetaData->etsiAncData.extAncData.ext_dmix_lfe_idx = hMetadata->ExtMetaData.lfeDmxLevel; } else { pAacMetaData->etsiAncData.extAncData.ext_dmix_lfe_idx = 15; /* -inf dB */ } } else { pAacMetaData->etsiAncData.extAncData.ext_downmix_lvl_status = 0; pAacMetaData->etsiAncData.extAncData.ext_downmix_gain_status = 0; pAacMetaData->etsiAncData.extAncData.ext_lfe_downmix_status = 0; pAacMetaData->etsiAncData.extAncData.ext_dmix_a_idx = 7; /* -inf dB */ pAacMetaData->etsiAncData.extAncData.ext_dmix_b_idx = 7; /* -inf dB */ encodeDynrng(1 << 16, &(pAacMetaData->etsiAncData.extAncData.dmx_gain_5_idx), &(pAacMetaData->etsiAncData.extAncData.dmx_gain_5_sgn)); encodeDynrng(1 << 16, &(pAacMetaData->etsiAncData.extAncData.dmx_gain_2_idx), &(pAacMetaData->etsiAncData.extAncData.dmx_gain_2_sgn)); pAacMetaData->etsiAncData.extAncData.ext_dmix_lfe_idx = 15; /* -inf dB */ } pAacMetaData->metadataMode = metadataMode; } else { pAacMetaData->metadataMode = 0; /* there is no configuration available */ } } return err; } INT FDK_MetadataEnc_GetDelay(HANDLE_FDK_METADATA_ENCODER hMetadataEnc) { INT delay = 0; if (hMetadataEnc != NULL) { delay = hMetadataEnc->nAudioDataDelay; } return delay; } fdk-aac-2.0.2/libAACenc/src/metadata_main.h000066400000000000000000000223311404225310700202650ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): V. Bacigalupo Description: Metadata Encoder library interface functions *******************************************************************************/ #ifndef METADATA_MAIN_H #define METADATA_MAIN_H /* Includes ******************************************************************/ #include "aacenc_lib.h" #include "aacenc.h" /* Defines *******************************************************************/ /* Data Types ****************************************************************/ typedef enum { METADATA_OK = 0x0000, /*!< No error happened. All fine. */ METADATA_INVALID_HANDLE = 0x0020, /*!< Handle passed to function call was invalid. */ METADATA_MEMORY_ERROR = 0x0021, /*!< Memory allocation failed. */ METADATA_INIT_ERROR = 0x0040, /*!< General initialization error. */ METADATA_ENCODE_ERROR = 0x0060 /*!< The encoding process was interrupted by an unexpected error. */ } FDK_METADATA_ERROR; /** * Meta Data handle. */ typedef struct FDK_METADATA_ENCODER *HANDLE_FDK_METADATA_ENCODER; /** * \brief Open a Meta Data instance. * * \param phMetadataEnc A pointer to a Meta Data handle to be allocated. * Initialized on return. * \param maxChannels Maximum number of supported audio channels. * * \return * - METADATA_OK, on succes. * - METADATA_INVALID_HANDLE, METADATA_MEMORY_ERROR, on failure. */ FDK_METADATA_ERROR FDK_MetadataEnc_Open( HANDLE_FDK_METADATA_ENCODER *phMetadataEnc, const UINT maxChannels); /** * \brief Initialize a Meta Data instance. * * \param hMetadataEnc Meta Data handle. * \param resetStates Indication for full reset of all states. * \param metadataMode Configures meta data output format (0,1,2,3). * \param audioDelay Delay cause by the audio encoder. * \param frameLength Number of samples to be processes within one * frame. * \param sampleRate Sampling rat in Hz of audio input signal. * \param nChannels Number of audio input channels. * \param channelMode Channel configuration which is used by the * encoder. * \param channelOrder Channel order of the input data. (WAV, MPEG) * * \return * - METADATA_OK, on succes. * - METADATA_INVALID_HANDLE, METADATA_INIT_ERROR, on failure. */ FDK_METADATA_ERROR FDK_MetadataEnc_Init( HANDLE_FDK_METADATA_ENCODER hMetadataEnc, const INT resetStates, const INT metadataMode, const INT audioDelay, const UINT frameLength, const UINT sampleRate, const UINT nChannels, const CHANNEL_MODE channelMode, const CHANNEL_ORDER channelOrder); /** * \brief Calculate Meta Data processing. * * This function treats all step necessary for meta data processing. * - Receive new meta data and make usable. * - Calculate DRC compressor and extract meta data info. * - Make meta data available for extern use. * - Apply audio data and meta data delay compensation. * * \param hMetadataEnc Meta Data handle. * \param pAudioSamples Pointer to audio input data. Existing function * overwrites audio data with delayed audio samples. * \param nAudioSamples Number of input audio samples to be prcessed. * \param pMetadata Pointer to Metat Data input. * \param ppMetaDataExtPayload Pointer to extension payload array. Filled on * return. * \param nMetaDataExtensions Pointer to variable to describe number of * available extension payloads. Filled on return. * \param matrix_mixdown_idx Pointer to variable for matrix mixdown * coefficient. Filled on return. * * \return * - METADATA_OK, on succes. * - METADATA_INVALID_HANDLE, METADATA_ENCODE_ERROR, on failure. */ FDK_METADATA_ERROR FDK_MetadataEnc_Process( HANDLE_FDK_METADATA_ENCODER hMetadataEnc, INT_PCM *const pAudioSamples, const UINT audioSamplesBufSize, const INT nAudioSamples, const AACENC_MetaData *const pMetadata, AACENC_EXT_PAYLOAD **ppMetaDataExtPayload, UINT *nMetaDataExtensions, INT *matrix_mixdown_idx); /** * \brief Close the Meta Data instance. * * Deallocate instance and free whole memory. * * \param phMetaData Pointer to the Meta Data handle to be * deallocated. * * \return * - METADATA_OK, on succes. * - METADATA_INVALID_HANDLE, on failure. */ FDK_METADATA_ERROR FDK_MetadataEnc_Close( HANDLE_FDK_METADATA_ENCODER *phMetaData); /** * \brief Get Meta Data Encoder delay. * * \param hMetadataEnc Meta Data Encoder handle. * * \return Delay caused by Meta Data module. */ INT FDK_MetadataEnc_GetDelay(HANDLE_FDK_METADATA_ENCODER hMetadataEnc); #endif /* METADATA_MAIN_H */ fdk-aac-2.0.2/libAACenc/src/mps_main.cpp000066400000000000000000000451031404225310700176410ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): Markus Lohwasser Description: Mpeg Surround library interface functions *******************************************************************************/ /* Includes ******************************************************************/ #include "mps_main.h" #include "sacenc_lib.h" /* Data Types ****************************************************************/ struct MPS_ENCODER { HANDLE_MP4SPACE_ENCODER hSacEncoder; AUDIO_OBJECT_TYPE audioObjectType; FDK_bufDescr inBufDesc; FDK_bufDescr outBufDesc; SACENC_InArgs inargs; SACENC_OutArgs outargs; void *pInBuffer[1]; UINT pInBufferSize[1]; UINT pInBufferElSize[1]; UINT pInBufferType[1]; void *pOutBuffer[2]; UINT pOutBufferSize[2]; UINT pOutBufferElSize[2]; UINT pOutBufferType[2]; UCHAR sacOutBuffer[1024]; /* Worst case memory consumption for ELDv2: 768 bytes => 6144 bits (Core + SBR + MPS) */ }; struct MPS_CONFIG_TAB { AUDIO_OBJECT_TYPE audio_object_type; CHANNEL_MODE channel_mode; ULONG sbr_ratio; ULONG sampling_rate; ULONG bitrate_min; ULONG bitrate_max; }; /* Constants *****************************************************************/ static const MPS_CONFIG_TAB mpsConfigTab[] = { {AOT_ER_AAC_ELD, MODE_212, 0, 16000, 16000, 39999}, {AOT_ER_AAC_ELD, MODE_212, 0, 22050, 16000, 49999}, {AOT_ER_AAC_ELD, MODE_212, 0, 24000, 16000, 61999}, {AOT_ER_AAC_ELD, MODE_212, 0, 32000, 20000, 84999}, {AOT_ER_AAC_ELD, MODE_212, 0, 44100, 50000, 192000}, {AOT_ER_AAC_ELD, MODE_212, 0, 48000, 62000, 192000}, {AOT_ER_AAC_ELD, MODE_212, 1, 16000, 18000, 31999}, {AOT_ER_AAC_ELD, MODE_212, 1, 22050, 18000, 31999}, {AOT_ER_AAC_ELD, MODE_212, 1, 24000, 20000, 64000}, {AOT_ER_AAC_ELD, MODE_212, 2, 32000, 18000, 64000}, {AOT_ER_AAC_ELD, MODE_212, 2, 44100, 21000, 64000}, {AOT_ER_AAC_ELD, MODE_212, 2, 48000, 26000, 64000} }; /* Function / Class Declarations *********************************************/ /* Function / Class Definition ***********************************************/ static INT FDK_MpegsEnc_WriteFrameHeader(HANDLE_MPS_ENCODER hMpsEnc, UCHAR *const pOutputBuffer, const int outputBufferSize); MPS_ENCODER_ERROR FDK_MpegsEnc_Open(HANDLE_MPS_ENCODER *phMpsEnc) { MPS_ENCODER_ERROR error = MPS_ENCODER_OK; HANDLE_MPS_ENCODER hMpsEnc = NULL; if (phMpsEnc == NULL) { error = MPS_ENCODER_INVALID_HANDLE; goto bail; } if (NULL == (hMpsEnc = (HANDLE_MPS_ENCODER)FDKcalloc(1, sizeof(MPS_ENCODER)))) { error = MPS_ENCODER_MEMORY_ERROR; goto bail; } FDKmemclear(hMpsEnc, sizeof(MPS_ENCODER)); if (SACENC_OK != FDK_sacenc_open(&hMpsEnc->hSacEncoder)) { error = MPS_ENCODER_MEMORY_ERROR; goto bail; } /* Return mps encoder instance */ *phMpsEnc = hMpsEnc; bail: if (error != MPS_ENCODER_OK) { FDK_MpegsEnc_Close(&hMpsEnc); } return error; } MPS_ENCODER_ERROR FDK_MpegsEnc_Close(HANDLE_MPS_ENCODER *phMpsEnc) { MPS_ENCODER_ERROR error = MPS_ENCODER_OK; if (phMpsEnc == NULL) { error = MPS_ENCODER_INVALID_HANDLE; goto bail; } if (*phMpsEnc != NULL) { FDK_sacenc_close(&(*phMpsEnc)->hSacEncoder); FDKfree(*phMpsEnc); *phMpsEnc = NULL; } bail: return error; } MPS_ENCODER_ERROR FDK_MpegsEnc_Init(HANDLE_MPS_ENCODER hMpsEnc, const AUDIO_OBJECT_TYPE audioObjectType, const UINT samplingrate, const UINT bitrate, const UINT sbrRatio, const UINT framelength, const UINT inputBufferSizePerChannel, const UINT coreCoderDelay) { MPS_ENCODER_ERROR error = MPS_ENCODER_OK; const UINT fs_low = 27713; /* low MPS sampling frequencies */ const UINT fs_high = 55426; /* high MPS sampling frequencies */ UINT nTimeSlots = 0, nQmfBandsLd = 0; if (hMpsEnc == NULL) { error = MPS_ENCODER_INVALID_HANDLE; goto bail; } /* Combine MPS with SBR only if the number of QMF band fits together.*/ switch (sbrRatio) { case 1: /* downsampled sbr - 32 QMF bands required */ if (!(samplingrate < fs_low)) { error = MPS_ENCODER_INIT_ERROR; goto bail; } break; case 2: /* dualrate - 64 QMF bands required */ if (!((samplingrate >= fs_low) && (samplingrate < fs_high))) { error = MPS_ENCODER_INIT_ERROR; goto bail; } break; case 0: default:; /* time interface - no samplingrate restriction */ } /* 32 QMF-Bands ( fs < 27713 ) * 64 QMF-Bands ( 27713 >= fs <= 55426 ) * 128 QMF-Bands ( fs > 55426 ) */ nQmfBandsLd = (samplingrate < fs_low) ? 5 : ((samplingrate > fs_high) ? 7 : 6); nTimeSlots = framelength >> nQmfBandsLd; /* check if number of qmf bands is usable for given framelength */ if (framelength != (nTimeSlots << nQmfBandsLd)) { error = MPS_ENCODER_INIT_ERROR; goto bail; } /* is given bitrate intended to be supported */ if ((INT)bitrate != FDK_MpegsEnc_GetClosestBitRate(audioObjectType, MODE_212, samplingrate, sbrRatio, bitrate)) { error = MPS_ENCODER_INIT_ERROR; goto bail; } /* init SAC library */ switch (audioObjectType) { case AOT_ER_AAC_ELD: { const UINT noInterFrameCoding = 0; if ((SACENC_OK != FDK_sacenc_setParam(hMpsEnc->hSacEncoder, SACENC_LOWDELAY, (noInterFrameCoding == 1) ? 1 : 2)) || (SACENC_OK != FDK_sacenc_setParam(hMpsEnc->hSacEncoder, SACENC_ENC_MODE, SACENC_212)) || (SACENC_OK != FDK_sacenc_setParam(hMpsEnc->hSacEncoder, SACENC_SAMPLERATE, samplingrate)) || (SACENC_OK != FDK_sacenc_setParam(hMpsEnc->hSacEncoder, SACENC_FRAME_TIME_SLOTS, nTimeSlots)) || (SACENC_OK != FDK_sacenc_setParam(hMpsEnc->hSacEncoder, SACENC_PARAM_BANDS, SACENC_BANDS_15)) || (SACENC_OK != FDK_sacenc_setParam(hMpsEnc->hSacEncoder, SACENC_TIME_DOM_DMX, 2)) || (SACENC_OK != FDK_sacenc_setParam(hMpsEnc->hSacEncoder, SACENC_COARSE_QUANT, 0)) || (SACENC_OK != FDK_sacenc_setParam(hMpsEnc->hSacEncoder, SACENC_QUANT_MODE, SACENC_QUANTMODE_FINE)) || (SACENC_OK != FDK_sacenc_setParam(hMpsEnc->hSacEncoder, SACENC_TIME_ALIGNMENT, 0)) || (SACENC_OK != FDK_sacenc_setParam(hMpsEnc->hSacEncoder, SACENC_INDEPENDENCY_FACTOR, 20))) { error = MPS_ENCODER_INIT_ERROR; goto bail; } break; } default: error = MPS_ENCODER_INIT_ERROR; goto bail; } if (SACENC_OK != FDK_sacenc_init(hMpsEnc->hSacEncoder, coreCoderDelay)) { error = MPS_ENCODER_INIT_ERROR; } hMpsEnc->audioObjectType = audioObjectType; hMpsEnc->inBufDesc.ppBase = (void **)&hMpsEnc->pInBuffer; hMpsEnc->inBufDesc.pBufSize = hMpsEnc->pInBufferSize; hMpsEnc->inBufDesc.pEleSize = hMpsEnc->pInBufferElSize; hMpsEnc->inBufDesc.pBufType = hMpsEnc->pInBufferType; hMpsEnc->inBufDesc.numBufs = 1; hMpsEnc->outBufDesc.ppBase = (void **)&hMpsEnc->pOutBuffer; hMpsEnc->outBufDesc.pBufSize = hMpsEnc->pOutBufferSize; hMpsEnc->outBufDesc.pEleSize = hMpsEnc->pOutBufferElSize; hMpsEnc->outBufDesc.pBufType = hMpsEnc->pOutBufferType; hMpsEnc->outBufDesc.numBufs = 2; hMpsEnc->pInBuffer[0] = NULL; hMpsEnc->pInBufferSize[0] = 0; hMpsEnc->pInBufferElSize[0] = sizeof(INT_PCM); hMpsEnc->pInBufferType[0] = (FDK_BUF_TYPE_INPUT | FDK_BUF_TYPE_PCM_DATA); hMpsEnc->pOutBuffer[0] = NULL; hMpsEnc->pOutBufferSize[0] = 0; hMpsEnc->pOutBufferElSize[0] = sizeof(INT_PCM); hMpsEnc->pOutBufferType[0] = (FDK_BUF_TYPE_OUTPUT | FDK_BUF_TYPE_PCM_DATA); hMpsEnc->pOutBuffer[1] = NULL; hMpsEnc->pOutBufferSize[1] = 0; hMpsEnc->pOutBufferElSize[1] = sizeof(UCHAR); hMpsEnc->pOutBufferType[1] = (FDK_BUF_TYPE_OUTPUT | FDK_BUF_TYPE_BS_DATA); hMpsEnc->inargs.isInputInterleaved = 0; hMpsEnc->inargs.inputBufferSizePerChannel = inputBufferSizePerChannel; bail: return error; } MPS_ENCODER_ERROR FDK_MpegsEnc_Process(HANDLE_MPS_ENCODER hMpsEnc, INT_PCM *const pAudioSamples, const INT nAudioSamples, AACENC_EXT_PAYLOAD *pMpsExtPayload) { MPS_ENCODER_ERROR error = MPS_ENCODER_OK; if (hMpsEnc == NULL) { error = MPS_ENCODER_INVALID_HANDLE; } else { int sacHeaderFlag = 1; int sacOutBufferOffset = 0; /* In case of eld the ssc is explicit and doesn't need to be inband */ if (hMpsEnc->audioObjectType == AOT_ER_AAC_ELD) { sacHeaderFlag = 0; } /* 4 bits nibble after extension type */ hMpsEnc->sacOutBuffer[0] = (sacHeaderFlag == 0) ? 0x3 : 0x7; sacOutBufferOffset += 1; if (sacHeaderFlag) { sacOutBufferOffset += FDK_MpegsEnc_WriteFrameHeader( hMpsEnc, &hMpsEnc->sacOutBuffer[sacOutBufferOffset], sizeof(hMpsEnc->sacOutBuffer) - sacOutBufferOffset); } /* Register input and output buffer. */ hMpsEnc->pInBuffer[0] = (void *)pAudioSamples; hMpsEnc->inargs.nInputSamples = nAudioSamples; hMpsEnc->pOutBuffer[0] = (void *)pAudioSamples; hMpsEnc->pOutBufferSize[0] = sizeof(INT_PCM) * nAudioSamples / 2; hMpsEnc->pOutBuffer[1] = (void *)&hMpsEnc->sacOutBuffer[sacOutBufferOffset]; hMpsEnc->pOutBufferSize[1] = sizeof(hMpsEnc->sacOutBuffer) - sacOutBufferOffset; /* encode SAC frame */ if (SACENC_OK != FDK_sacenc_encode(hMpsEnc->hSacEncoder, &hMpsEnc->inBufDesc, &hMpsEnc->outBufDesc, &hMpsEnc->inargs, &hMpsEnc->outargs)) { error = MPS_ENCODER_ENCODE_ERROR; goto bail; } /* export MPS payload */ pMpsExtPayload->pData = (UCHAR *)hMpsEnc->sacOutBuffer; pMpsExtPayload->dataSize = hMpsEnc->outargs.nOutputBits + 8 * (sacOutBufferOffset - 1); pMpsExtPayload->dataType = EXT_LDSAC_DATA; pMpsExtPayload->associatedChElement = -1; } bail: return error; } INT FDK_MpegsEnc_WriteSpatialSpecificConfig(HANDLE_MPS_ENCODER hMpsEnc, HANDLE_FDK_BITSTREAM hBs) { INT sscBits = 0; if (NULL != hMpsEnc) { MP4SPACEENC_INFO mp4SpaceEncoderInfo; FDK_sacenc_getInfo(hMpsEnc->hSacEncoder, &mp4SpaceEncoderInfo); if (hBs != NULL) { int i; int writtenBits = 0; for (i = 0; inSscSizeBits>> 3; i++) { FDKwriteBits(hBs, mp4SpaceEncoderInfo.pSscBuf->pSsc[i], 8); writtenBits += 8; } FDKwriteBits(hBs, mp4SpaceEncoderInfo.pSscBuf->pSsc[i], mp4SpaceEncoderInfo.pSscBuf->nSscSizeBits - writtenBits); } /* hBS */ sscBits = mp4SpaceEncoderInfo.pSscBuf->nSscSizeBits; } /* valid hMpsEnc */ return sscBits; } static INT FDK_MpegsEnc_WriteFrameHeader(HANDLE_MPS_ENCODER hMpsEnc, UCHAR *const pOutputBuffer, const int outputBufferSize) { const int sacTimeAlignFlag = 0; /* Initialize variables */ int numBits = 0; if ((NULL != hMpsEnc) && (NULL != pOutputBuffer)) { UINT alignAnchor, cnt; FDK_BITSTREAM Bs; FDKinitBitStream(&Bs, pOutputBuffer, outputBufferSize, 0, BS_WRITER); /* Calculate SSC length information */ cnt = (FDK_MpegsEnc_WriteSpatialSpecificConfig(hMpsEnc, NULL) + 7) >> 3; /* Write SSC */ FDKwriteBits(&Bs, sacTimeAlignFlag, 1); if (cnt < 127) { FDKwriteBits(&Bs, cnt, 7); } else { FDKwriteBits(&Bs, 127, 7); FDKwriteBits(&Bs, cnt - 127, 16); } alignAnchor = FDKgetValidBits(&Bs); FDK_MpegsEnc_WriteSpatialSpecificConfig(hMpsEnc, &Bs); FDKbyteAlign(&Bs, alignAnchor); /* bsFillBits */ if (sacTimeAlignFlag) { FDK_ASSERT(1); /* time alignment not supported */ } numBits = FDKgetValidBits(&Bs); } /* valid handle */ return ((numBits + 7) >> 3); } INT FDK_MpegsEnc_GetClosestBitRate(const AUDIO_OBJECT_TYPE audioObjectType, const CHANNEL_MODE channelMode, const UINT samplingrate, const UINT sbrRatio, const UINT bitrate) { unsigned int i; int targetBitrate = -1; for (i = 0; i < sizeof(mpsConfigTab) / sizeof(MPS_CONFIG_TAB); i++) { if ((mpsConfigTab[i].audio_object_type == audioObjectType) && (mpsConfigTab[i].channel_mode == channelMode) && (mpsConfigTab[i].sbr_ratio == sbrRatio) && (mpsConfigTab[i].sampling_rate == samplingrate)) { targetBitrate = fMin(fMax(bitrate, mpsConfigTab[i].bitrate_min), mpsConfigTab[i].bitrate_max); } } return targetBitrate; } INT FDK_MpegsEnc_GetDelay(HANDLE_MPS_ENCODER hMpsEnc) { INT delay = 0; if (NULL != hMpsEnc) { MP4SPACEENC_INFO mp4SpaceEncoderInfo; FDK_sacenc_getInfo(hMpsEnc->hSacEncoder, &mp4SpaceEncoderInfo); delay = mp4SpaceEncoderInfo.nCodecDelay; } return delay; } INT FDK_MpegsEnc_GetDecDelay(HANDLE_MPS_ENCODER hMpsEnc) { INT delay = 0; if (NULL != hMpsEnc) { MP4SPACEENC_INFO mp4SpaceEncoderInfo; FDK_sacenc_getInfo(hMpsEnc->hSacEncoder, &mp4SpaceEncoderInfo); delay = mp4SpaceEncoderInfo.nDecoderDelay; } return delay; } MPS_ENCODER_ERROR FDK_MpegsEnc_GetLibInfo(LIB_INFO *info) { MPS_ENCODER_ERROR error = MPS_ENCODER_OK; if (NULL == info) { error = MPS_ENCODER_INVALID_HANDLE; } else if (SACENC_OK != FDK_sacenc_getLibInfo(info)) { error = MPS_ENCODER_INIT_ERROR; } return error; } fdk-aac-2.0.2/libAACenc/src/mps_main.h000066400000000000000000000253351404225310700173130ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): Markus Lohwasser Description: Mpeg Surround library interface functions *******************************************************************************/ #ifndef MPS_MAIN_H #define MPS_MAIN_H /* Includes ******************************************************************/ #include "aacenc.h" #include "FDK_audio.h" #include "machine_type.h" /* Defines *******************************************************************/ typedef enum { MPS_ENCODER_OK = 0x0000, /*!< No error happened. All fine. */ MPS_ENCODER_INVALID_HANDLE = 0x0020, /*!< Handle passed to function call was invalid. */ MPS_ENCODER_MEMORY_ERROR = 0x0021, /*!< Memory allocation failed. */ MPS_ENCODER_INIT_ERROR = 0x0040, /*!< General initialization error. */ MPS_ENCODER_ENCODE_ERROR = 0x0060 /*!< The encoding process was interrupted by an unexpected error. */ } MPS_ENCODER_ERROR; /* Data Types ****************************************************************/ /** * MPEG Surround Encoder interface handle. */ typedef struct MPS_ENCODER MPS_ENCODER, *HANDLE_MPS_ENCODER; /* Function / Class Declarations *********************************************/ /** * \brief Open a Mpeg Surround Encoder instance. * * \phMpsEnc A pointer to a MPS handle to be allocated. * Initialized on return. * * \return * - MPS_ENCODER_OK, on succes. * - MPS_ENCODER_INVALID_HANDLE, MPS_ENCODER_MEMORY_ERROR, on failure. */ MPS_ENCODER_ERROR FDK_MpegsEnc_Open(HANDLE_MPS_ENCODER *phMpsEnc); /** * \brief Close the Mpeg Surround Encoder instance. * * Deallocate instance and free whole memory. * * \param phMpsEnc Pointer to the MPS handle to be deallocated. * * \return * - MPS_ENCODER_OK, on succes. * - MPS_ENCODER_INVALID_HANDLE, on failure. */ MPS_ENCODER_ERROR FDK_MpegsEnc_Close(HANDLE_MPS_ENCODER *phMpsEnc); /** * \brief Initialize a Mpeg Surround Encoder instance. * * \param hMpsEnc MPS Encoder handle. * \param audioObjectType Audio object type. * \param samplingrate Sampling rate in Hz of audio input signal. * \param bitrate Encder target bitrate. * \param sbrRatio SBR sampling rate ratio. * \param framelength Number of samples to be processes within one * frame. * \param inputBufferSizePerChannel Size of input buffer per channel. * \param coreCoderDelay Core coder delay. * * \return * - MPS_ENCODER_OK, on succes. * - MPS_ENCODER_INVALID_HANDLE, MPS_ENCODER_ENCODE_ERROR, on failure. */ MPS_ENCODER_ERROR FDK_MpegsEnc_Init(HANDLE_MPS_ENCODER hMpsEnc, const AUDIO_OBJECT_TYPE audioObjectType, const UINT samplingrate, const UINT bitrate, const UINT sbrRatio, const UINT framelength, const UINT inputBufferSizePerChannel, const UINT coreCoderDelay); /** * \brief Calculate Mpeg Surround processing. * * This fuction applies the MPS processing. The MPS side info will be written to * extension payload. The input audio data will be overwritten by the calculated * downmix. * * \param hMpsEnc MPS Encoder handle. * \param pAudioSamples Pointer to audio input/output data. * \param nAudioSamples Number of input audio samples to be prcessed. * \param pMpsExtPayload Pointer to extension payload to be filled on * return. * * \return * - MPS_ENCODER_OK, on succes. * - MPS_ENCODER_INVALID_HANDLE, MPS_ENCODER_ENCODE_ERROR, on failure. */ MPS_ENCODER_ERROR FDK_MpegsEnc_Process(HANDLE_MPS_ENCODER hMpsEnc, INT_PCM *const pAudioSamples, const INT nAudioSamples, AACENC_EXT_PAYLOAD *pMpsExtPayload); /** * \brief Write Spatial Specific Config. * * This function can be called via call back from the transport library to write * the Spatial Specific Config to given bitstream buffer. * * \param hMpsEnc MPS Encoder handle. * \param hBs Bitstream buffer handle. * * \return Number of written bits. */ INT FDK_MpegsEnc_WriteSpatialSpecificConfig(HANDLE_MPS_ENCODER hMpsEnc, HANDLE_FDK_BITSTREAM hBs); /** * \brief Get closest valid bitrate supported by given config. * * \param audioObjectType Audio object type. * \param channelMode Encoder channel mode. * \param samplingrate Sampling rate in Hz of audio input signal. * \param sbrRatio SBR sampling rate ratio. * \param bitrate The desired target bitrate. * * \return Closest valid bitrate to given bitrate.. */ INT FDK_MpegsEnc_GetClosestBitRate(const AUDIO_OBJECT_TYPE audioObjectType, const CHANNEL_MODE channelMode, const UINT samplingrate, const UINT sbrRatio, const UINT bitrate); /** * \brief Get codec delay. * * This function returns delay of the whole en-/decoded signal, including * corecoder delay. * * \param hMpsEnc MPS Encoder handle. * * \return Codec delay in samples. */ INT FDK_MpegsEnc_GetDelay(HANDLE_MPS_ENCODER hMpsEnc); /** * \brief Get Mpeg Surround Decoder delay. * * This function returns delay of the Mpeg Surround decoder. * * \param hMpsEnc MPS Encoder handle. * * \return Mpeg Surround Decoder delay in samples. */ INT FDK_MpegsEnc_GetDecDelay(HANDLE_MPS_ENCODER hMpsEnc); /** * \brief Get information about encoder library build. * * Fill a given LIB_INFO structure with library version information. * * \param info Pointer to an allocated LIB_INFO struct. * * \return * - MPS_ENCODER_OK, on succes. * - MPS_ENCODER_INVALID_HANDLE, MPS_ENCODER_INIT_ERROR, on failure. */ MPS_ENCODER_ERROR FDK_MpegsEnc_GetLibInfo(LIB_INFO *info); #endif /* MPS_MAIN_H */ fdk-aac-2.0.2/libAACenc/src/ms_stereo.cpp000066400000000000000000000323511404225310700200370ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M.Werner Description: MS stereo processing *******************************************************************************/ #include "ms_stereo.h" #include "psy_const.h" /* static const float scaleMinThres = 1.0f; */ /* 0.75f for 3db boost */ void FDKaacEnc_MsStereoProcessing(PSY_DATA *RESTRICT psyData[(2)], PSY_OUT_CHANNEL *psyOutChannel[2], const INT *isBook, INT *msDigest, /* output */ INT *msMask, /* output */ const INT allowMS, const INT sfbCnt, const INT sfbPerGroup, const INT maxSfbPerGroup, const INT *sfbOffset) { FIXP_DBL *sfbEnergyLeft = psyData[0]->sfbEnergy.Long; /* modified where msMask==1 */ FIXP_DBL *sfbEnergyRight = psyData[1]->sfbEnergy.Long; /* modified where msMask==1 */ const FIXP_DBL *sfbEnergyMid = psyData[0]->sfbEnergyMS.Long; const FIXP_DBL *sfbEnergySide = psyData[1]->sfbEnergyMS.Long; FIXP_DBL *sfbThresholdLeft = psyData[0]->sfbThreshold.Long; /* modified where msMask==1 */ FIXP_DBL *sfbThresholdRight = psyData[1]->sfbThreshold.Long; /* modified where msMask==1 */ FIXP_DBL *sfbSpreadEnLeft = psyData[0]->sfbSpreadEnergy.Long; FIXP_DBL *sfbSpreadEnRight = psyData[1]->sfbSpreadEnergy.Long; FIXP_DBL *sfbEnergyLeftLdData = psyOutChannel[0]->sfbEnergyLdData; /* modified where msMask==1 */ FIXP_DBL *sfbEnergyRightLdData = psyOutChannel[1]->sfbEnergyLdData; /* modified where msMask==1 */ FIXP_DBL *sfbEnergyMidLdData = psyData[0]->sfbEnergyMSLdData; FIXP_DBL *sfbEnergySideLdData = psyData[1]->sfbEnergyMSLdData; FIXP_DBL *sfbThresholdLeftLdData = psyOutChannel[0]->sfbThresholdLdData; /* modified where msMask==1 */ FIXP_DBL *sfbThresholdRightLdData = psyOutChannel[1]->sfbThresholdLdData; /* modified where msMask==1 */ FIXP_DBL *mdctSpectrumLeft = psyData[0]->mdctSpectrum; /* modified where msMask==1 */ FIXP_DBL *mdctSpectrumRight = psyData[1]->mdctSpectrum; /* modified where msMask==1 */ INT sfb, sfboffs, j; /* loop counters */ FIXP_DBL pnlrLdData, pnmsLdData; FIXP_DBL minThresholdLdData; FIXP_DBL minThreshold; INT useMS; INT msMaskTrueSomewhere = 0; /* to determine msDigest */ INT numMsMaskFalse = 0; /* number of non-intensity bands where L/R coding is used */ for (sfb = 0; sfb < sfbCnt; sfb += sfbPerGroup) { for (sfboffs = 0; sfboffs < maxSfbPerGroup; sfboffs++) { if ((isBook == NULL) ? 1 : (isBook[sfb + sfboffs] == 0)) { FIXP_DBL tmp; /* minThreshold=min(sfbThresholdLeft[sfb+sfboffs], sfbThresholdRight[sfb+sfboffs])*scaleMinThres; pnlr = (sfbThresholdLeft[sfb+sfboffs]/ max(sfbEnergyLeft[sfb+sfboffs],sfbThresholdLeft[sfb+sfboffs]))* (sfbThresholdRight[sfb+sfboffs]/ max(sfbEnergyRight[sfb+sfboffs],sfbThresholdRight[sfb+sfboffs])); pnms = (minThreshold/max(sfbEnergyMid[sfb+sfboffs],minThreshold))* (minThreshold/max(sfbEnergySide[sfb+sfboffs],minThreshold)); useMS = (pnms > pnlr); */ /* we assume that scaleMinThres == 1.0f and we can drop it */ minThresholdLdData = fixMin(sfbThresholdLeftLdData[sfb + sfboffs], sfbThresholdRightLdData[sfb + sfboffs]); /* pnlrLdData = sfbThresholdLeftLdData[sfb+sfboffs] - max(sfbEnergyLeftLdData[sfb+sfboffs], sfbThresholdLeftLdData[sfb+sfboffs]) + sfbThresholdRightLdData[sfb+sfboffs] - max(sfbEnergyRightLdData[sfb+sfboffs], sfbThresholdRightLdData[sfb+sfboffs]); */ tmp = fixMax(sfbEnergyLeftLdData[sfb + sfboffs], sfbThresholdLeftLdData[sfb + sfboffs]); pnlrLdData = (sfbThresholdLeftLdData[sfb + sfboffs] >> 1) - (tmp >> 1); pnlrLdData = pnlrLdData + (sfbThresholdRightLdData[sfb + sfboffs] >> 1); tmp = fixMax(sfbEnergyRightLdData[sfb + sfboffs], sfbThresholdRightLdData[sfb + sfboffs]); pnlrLdData = pnlrLdData - (tmp >> 1); /* pnmsLdData = minThresholdLdData - max(sfbEnergyMidLdData[sfb+sfboffs], minThresholdLdData) + minThresholdLdData - max(sfbEnergySideLdData[sfb+sfboffs], minThresholdLdData); */ tmp = fixMax(sfbEnergyMidLdData[sfb + sfboffs], minThresholdLdData); pnmsLdData = minThresholdLdData - (tmp >> 1); tmp = fixMax(sfbEnergySideLdData[sfb + sfboffs], minThresholdLdData); pnmsLdData = pnmsLdData - (tmp >> 1); useMS = ((allowMS != 0) && (pnmsLdData > pnlrLdData)) ? 1 : 0; if (useMS) { msMask[sfb + sfboffs] = 1; msMaskTrueSomewhere = 1; for (j = sfbOffset[sfb + sfboffs]; j < sfbOffset[sfb + sfboffs + 1]; j++) { FIXP_DBL specL, specR; specL = mdctSpectrumLeft[j] >> 1; specR = mdctSpectrumRight[j] >> 1; mdctSpectrumLeft[j] = specL + specR; mdctSpectrumRight[j] = specL - specR; } minThreshold = fixMin(sfbThresholdLeft[sfb + sfboffs], sfbThresholdRight[sfb + sfboffs]); sfbThresholdLeft[sfb + sfboffs] = sfbThresholdRight[sfb + sfboffs] = minThreshold; sfbThresholdLeftLdData[sfb + sfboffs] = sfbThresholdRightLdData[sfb + sfboffs] = minThresholdLdData; sfbEnergyLeft[sfb + sfboffs] = sfbEnergyMid[sfb + sfboffs]; sfbEnergyRight[sfb + sfboffs] = sfbEnergySide[sfb + sfboffs]; sfbEnergyLeftLdData[sfb + sfboffs] = sfbEnergyMidLdData[sfb + sfboffs]; sfbEnergyRightLdData[sfb + sfboffs] = sfbEnergySideLdData[sfb + sfboffs]; sfbSpreadEnLeft[sfb + sfboffs] = sfbSpreadEnRight[sfb + sfboffs] = fixMin(sfbSpreadEnLeft[sfb + sfboffs], sfbSpreadEnRight[sfb + sfboffs]) >> 1; } else { msMask[sfb + sfboffs] = 0; numMsMaskFalse++; } /* useMS */ } /* isBook */ else { /* keep mDigest from IS module */ if (msMask[sfb + sfboffs]) { msMaskTrueSomewhere = 1; } /* prohibit MS_MASK_ALL in combination with IS */ numMsMaskFalse = 9; } /* isBook */ } /* sfboffs */ } /* sfb */ if (msMaskTrueSomewhere == 1) { if ((numMsMaskFalse == 0) || ((numMsMaskFalse < maxSfbPerGroup) && (numMsMaskFalse < 9))) { *msDigest = SI_MS_MASK_ALL; /* loop through M/S bands; if msMask==0, set it to 1 and apply M/S */ for (sfb = 0; sfb < sfbCnt; sfb += sfbPerGroup) { for (sfboffs = 0; sfboffs < maxSfbPerGroup; sfboffs++) { if (((isBook == NULL) ? 1 : (isBook[sfb + sfboffs] == 0)) && (msMask[sfb + sfboffs] == 0)) { msMask[sfb + sfboffs] = 1; /* apply M/S coding */ for (j = sfbOffset[sfb + sfboffs]; j < sfbOffset[sfb + sfboffs + 1]; j++) { FIXP_DBL specL, specR; specL = mdctSpectrumLeft[j] >> 1; specR = mdctSpectrumRight[j] >> 1; mdctSpectrumLeft[j] = specL + specR; mdctSpectrumRight[j] = specL - specR; } minThreshold = fixMin(sfbThresholdLeft[sfb + sfboffs], sfbThresholdRight[sfb + sfboffs]); sfbThresholdLeft[sfb + sfboffs] = sfbThresholdRight[sfb + sfboffs] = minThreshold; minThresholdLdData = fixMin(sfbThresholdLeftLdData[sfb + sfboffs], sfbThresholdRightLdData[sfb + sfboffs]); sfbThresholdLeftLdData[sfb + sfboffs] = sfbThresholdRightLdData[sfb + sfboffs] = minThresholdLdData; sfbEnergyLeft[sfb + sfboffs] = sfbEnergyMid[sfb + sfboffs]; sfbEnergyRight[sfb + sfboffs] = sfbEnergySide[sfb + sfboffs]; sfbEnergyLeftLdData[sfb + sfboffs] = sfbEnergyMidLdData[sfb + sfboffs]; sfbEnergyRightLdData[sfb + sfboffs] = sfbEnergySideLdData[sfb + sfboffs]; sfbSpreadEnLeft[sfb + sfboffs] = sfbSpreadEnRight[sfb + sfboffs] = fixMin(sfbSpreadEnLeft[sfb + sfboffs], sfbSpreadEnRight[sfb + sfboffs]) >> 1; } } } } else { *msDigest = SI_MS_MASK_SOME; } } else { *msDigest = SI_MS_MASK_NONE; } } fdk-aac-2.0.2/libAACenc/src/ms_stereo.h000066400000000000000000000125371404225310700175100ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M.Werner Description: MS stereo processing *******************************************************************************/ #ifndef MS_STEREO_H #define MS_STEREO_H #include "interface.h" void FDKaacEnc_MsStereoProcessing(PSY_DATA *RESTRICT psyData[(2)], PSY_OUT_CHANNEL *psyOutChannel[2], const INT *isBook, INT *msDigest, /* output */ INT *msMask, /* output */ const INT allowMS, const INT sfbCnt, const INT sfbPerGroup, const INT maxSfbPerGroup, const INT *sfbOffset); #endif /* MS_STEREO_H */ fdk-aac-2.0.2/libAACenc/src/noisedet.cpp000066400000000000000000000232521404225310700176510ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Lohwasser Description: noisedet.c Routines for Noise Detection *******************************************************************************/ #include "noisedet.h" #include "aacenc_pns.h" #include "pnsparam.h" /***************************************************************************** functionname: FDKaacEnc_fuzzyIsSmaller description: Fuzzy value calculation for "testVal is smaller than refVal" returns: fuzzy value input: test and ref Value, low and high Lim output: return fuzzy value *****************************************************************************/ static FIXP_SGL FDKaacEnc_fuzzyIsSmaller(FIXP_DBL testVal, FIXP_DBL refVal, FIXP_DBL loLim, FIXP_DBL hiLim) { if (refVal <= FL2FXCONST_DBL(0.0)) return (FL2FXCONST_SGL(0.0f)); else if (testVal >= fMult((hiLim >> 1) + (loLim >> 1), refVal)) return (FL2FXCONST_SGL(0.0f)); else return ((FIXP_SGL)MAXVAL_SGL); } /***************************************************************************** functionname: FDKaacEnc_noiseDetect description: detect tonal sfb's; two tests Powerdistribution: sfb splittet in four regions, compare the energy in all sections PsychTonality: compare tonality from chaosmeasure with reftonality returns: input: spectrum of one large mdct number of sfb's pointer to offset of sfb's pointer to noiseFuzzyMeasure (modified) noiseparams struct pointer to sfb energies pointer to tonality calculated in chaosmeasure output: noiseFuzzy Measure *****************************************************************************/ void FDKaacEnc_noiseDetect(FIXP_DBL *RESTRICT mdctSpectrum, INT *RESTRICT sfbMaxScaleSpec, INT sfbActive, const INT *RESTRICT sfbOffset, FIXP_SGL *RESTRICT noiseFuzzyMeasure, NOISEPARAMS *np, FIXP_SGL *RESTRICT sfbtonality) { int i, k, sfb, sfbWidth; FIXP_SGL fuzzy, fuzzyTotal; FIXP_DBL refVal, testVal; /***** Start detection phase *****/ /* Start noise detection for each band based on a number of checks */ for (sfb = 0; sfb < sfbActive; sfb++) { fuzzyTotal = (FIXP_SGL)MAXVAL_SGL; sfbWidth = sfbOffset[sfb + 1] - sfbOffset[sfb]; /* Reset output for lower bands or too small bands */ if (sfb < np->startSfb || sfbWidth < np->minSfbWidth) { noiseFuzzyMeasure[sfb] = FL2FXCONST_SGL(0.0f); continue; } if ((np->detectionAlgorithmFlags & USE_POWER_DISTRIBUTION) && (fuzzyTotal > FL2FXCONST_SGL(0.5f))) { FIXP_DBL fhelp1, fhelp2, fhelp3, fhelp4, maxVal, minVal; INT leadingBits = fixMax( 0, (sfbMaxScaleSpec[sfb] - 3)); /* max sfbWidth = 96/4 ; 2^5=32 => 5/2 = 3 (spc*spc) */ /* check power distribution in four regions */ fhelp1 = fhelp2 = fhelp3 = fhelp4 = FL2FXCONST_DBL(0.0f); k = sfbWidth >> 2; /* Width of a quarter band */ for (i = sfbOffset[sfb]; i < sfbOffset[sfb] + k; i++) { fhelp1 = fPow2AddDiv2(fhelp1, mdctSpectrum[i] << leadingBits); fhelp2 = fPow2AddDiv2(fhelp2, mdctSpectrum[i + k] << leadingBits); fhelp3 = fPow2AddDiv2(fhelp3, mdctSpectrum[i + 2 * k] << leadingBits); fhelp4 = fPow2AddDiv2(fhelp4, mdctSpectrum[i + 3 * k] << leadingBits); } /* get max into fhelp: */ maxVal = fixMax(fhelp1, fhelp2); maxVal = fixMax(maxVal, fhelp3); maxVal = fixMax(maxVal, fhelp4); /* get min into fhelp1: */ minVal = fixMin(fhelp1, fhelp2); minVal = fixMin(minVal, fhelp3); minVal = fixMin(minVal, fhelp4); /* Normalize min and max Val */ leadingBits = CountLeadingBits(maxVal); testVal = maxVal << leadingBits; refVal = minVal << leadingBits; /* calculate fuzzy value for power distribution */ testVal = fMultDiv2(testVal, np->powDistPSDcurve[sfb]); fuzzy = FDKaacEnc_fuzzyIsSmaller( testVal, /* 1/2 * maxValue * PSDcurve */ refVal, /* 1 * minValue */ FL2FXCONST_DBL(0.495), /* 1/2 * loLim (0.99f/2) */ FL2FXCONST_DBL(0.505)); /* 1/2 * hiLim (1.01f/2) */ fuzzyTotal = fixMin(fuzzyTotal, fuzzy); } if ((np->detectionAlgorithmFlags & USE_PSYCH_TONALITY) && (fuzzyTotal > FL2FXCONST_SGL(0.5f))) { /* Detection with tonality-value of psych. acoustic (here: 1 is tonal!)*/ testVal = FX_SGL2FX_DBL(sfbtonality[sfb]) >> 1; /* 1/2 * sfbTonality */ refVal = np->refTonality; fuzzy = FDKaacEnc_fuzzyIsSmaller( testVal, refVal, FL2FXCONST_DBL(0.45f), /* 1/2 * loLim (0.9f/2) */ FL2FXCONST_DBL(0.55f)); /* 1/2 * hiLim (1.1f/2) */ fuzzyTotal = fixMin(fuzzyTotal, fuzzy); } /* Output of final result */ noiseFuzzyMeasure[sfb] = fuzzyTotal; } } fdk-aac-2.0.2/libAACenc/src/noisedet.h000066400000000000000000000121531404225310700173140ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Lohwasser Description: noisedet.h *******************************************************************************/ #ifndef NOISEDET_H #define NOISEDET_H #include "common_fix.h" #include "pnsparam.h" #include "psy_data.h" void FDKaacEnc_noiseDetect(FIXP_DBL *mdctSpectrum, INT *sfbMaxScaleSpec, INT sfbActive, const INT *sfbOffset, FIXP_SGL noiseFuzzyMeasure[], NOISEPARAMS *np, FIXP_SGL *sfbtonality); #endif /* NOISEDET_H */ fdk-aac-2.0.2/libAACenc/src/pns_func.h000066400000000000000000000145171404225310700173230ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Lohwasser Description: pns_func.h *******************************************************************************/ #ifndef PNS_FUNC_H #define PNS_FUNC_H #include "common_fix.h" #include "aacenc_pns.h" #include "psy_data.h" AAC_ENCODER_ERROR FDKaacEnc_InitPnsConfiguration( PNS_CONFIG *pnsConf, INT bitRate, INT sampleRate, INT usePns, INT sfbCnt, const INT *sfbOffset, const INT numChan, const INT isLC); void FDKaacEnc_PnsDetect(PNS_CONFIG *pnsConf, PNS_DATA *pnsData, const INT lastWindowSequence, const INT sfbActive, const INT maxSfbPerGroup, FIXP_DBL *sfbThresholdLdData, const INT *sfbOffset, FIXP_DBL *mdctSpectrum, INT *sfbMaxScaleSpec, FIXP_SGL *sfbtonality, int tnsOrder, INT tnsPredictionGain, INT tnsActive, FIXP_DBL *sfbEnergyLdData, INT *noiseNrg); void FDKaacEnc_CodePnsChannel(const INT sfbActive, PNS_CONFIG *pnsConf, INT *pnsFlag, FIXP_DBL *sfbEnergy, INT *noiseNrg, FIXP_DBL *sfbThreshold); void FDKaacEnc_PreProcessPnsChannelPair( const INT sfbActive, FIXP_DBL *sfbEnergyLeft, FIXP_DBL *sfbEnergyRight, FIXP_DBL *sfbEnergyLeftLD, FIXP_DBL *sfbEnergyRightLD, FIXP_DBL *sfbEnergyMid, PNS_CONFIG *pnsConfLeft, PNS_DATA *pnsDataLeft, PNS_DATA *pnsDataRight); void FDKaacEnc_PostProcessPnsChannelPair(const INT sfbActive, PNS_CONFIG *pnsConf, PNS_DATA *pnsDataLeft, PNS_DATA *pnsDataRight, INT *msMask, INT *msDigest); #endif /* PNS_FUNC_H */ fdk-aac-2.0.2/libAACenc/src/pnsparam.cpp000066400000000000000000000363011404225310700176570ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M.Lohwasser Description: PNS parameters depending on bitrate and bandwidth *******************************************************************************/ #include "pnsparam.h" #include "psy_configuration.h" typedef struct { SHORT startFreq; /* Parameters for detection */ FIXP_SGL refPower; FIXP_SGL refTonality; SHORT tnsGainThreshold; /* scaled by TNS_PREDGAIN_SCALE (=1000) */ SHORT tnsPNSGainThreshold; /* scaled by TNS_PREDGAIN_SCALE (=1000) */ FIXP_SGL gapFillThr; SHORT minSfbWidth; USHORT detectionAlgorithmFlags; } PNS_INFO_TAB; typedef struct { ULONG brFrom; ULONG brTo; UCHAR S16000; UCHAR S22050; UCHAR S24000; UCHAR S32000; UCHAR S44100; UCHAR S48000; } AUTO_PNS_TAB; static const AUTO_PNS_TAB levelTable_mono[] = { { 0, 11999, 0, 1, 1, 1, 1, 1, }, { 12000, 19999, 0, 1, 1, 1, 1, 1, }, { 20000, 28999, 0, 2, 1, 1, 1, 1, }, { 29000, 40999, 0, 4, 4, 4, 2, 2, }, { 41000, 55999, 0, 9, 9, 7, 7, 7, }, { 56000, 61999, 0, 0, 0, 0, 9, 9, }, { 62000, 75999, 0, 0, 0, 0, 0, 0, }, { 76000, 92999, 0, 0, 0, 0, 0, 0, }, { 93000, 999999, 0, 0, 0, 0, 0, 0, }, }; static const AUTO_PNS_TAB levelTable_stereo[] = { { 0, 11999, 0, 1, 1, 1, 1, 1, }, { 12000, 19999, 0, 3, 1, 1, 1, 1, }, { 20000, 28999, 0, 3, 3, 3, 2, 2, }, { 29000, 40999, 0, 7, 6, 6, 5, 5, }, { 41000, 55999, 0, 9, 9, 7, 7, 7, }, { 56000, 79999, 0, 0, 0, 0, 0, 0, }, { 80000, 99999, 0, 0, 0, 0, 0, 0, }, { 100000, 999999, 0, 0, 0, 0, 0, 0, }, }; static const PNS_INFO_TAB pnsInfoTab[] = { /*0 pns off */ /*1*/ {4000, FL2FXCONST_SGL(0.04), FL2FXCONST_SGL(0.06), 1150, 1200, FL2FXCONST_SGL(0.02), 8, USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | USE_TNS_PNS /*| JUST_LONG_WINDOW*/}, /*2*/ {4000, FL2FXCONST_SGL(0.04), FL2FXCONST_SGL(0.07), 1130, 1300, FL2FXCONST_SGL(0.05), 8, USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | USE_TNS_PNS /*| JUST_LONG_WINDOW*/}, /*3*/ {4100, FL2FXCONST_SGL(0.04), FL2FXCONST_SGL(0.07), 1100, 1400, FL2FXCONST_SGL(0.10), 8, USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | USE_TNS_PNS /*| JUST_LONG_WINDOW*/}, /*4*/ {4100, FL2FXCONST_SGL(0.03), FL2FXCONST_SGL(0.10), 1100, 1400, FL2FXCONST_SGL(0.15), 8, USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | USE_TNS_PNS /*| JUST_LONG_WINDOW*/}, /*5*/ {4300, FL2FXCONST_SGL(0.03), FL2FXCONST_SGL(0.10), 1100, 1400, FL2FXCONST_SGL(0.15), 8, USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | USE_TNS_PNS | JUST_LONG_WINDOW}, /*6*/ {5000, FL2FXCONST_SGL(0.03), FL2FXCONST_SGL(0.10), 1100, 1400, FL2FXCONST_SGL(0.25), 8, USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | USE_TNS_PNS | JUST_LONG_WINDOW}, /*7*/ {5500, FL2FXCONST_SGL(0.03), FL2FXCONST_SGL(0.12), 1100, 1400, FL2FXCONST_SGL(0.35), 8, USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | USE_TNS_PNS | JUST_LONG_WINDOW}, /*8*/ {6000, FL2FXCONST_SGL(0.03), FL2FXCONST_SGL(0.12), 1080, 1400, FL2FXCONST_SGL(0.40), 8, USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | USE_TNS_PNS | JUST_LONG_WINDOW}, /*9*/ {6000, FL2FXCONST_SGL(0.03), FL2FXCONST_SGL(0.14), 1070, 1400, FL2FXCONST_SGL(0.45), 8, USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | USE_TNS_PNS | JUST_LONG_WINDOW}, }; static const AUTO_PNS_TAB levelTable_lowComplexity[] = { { 0, 27999, 0, 0, 0, 0, 0, 0, }, { 28000, 31999, 0, 2, 2, 2, 2, 2, }, { 32000, 47999, 0, 3, 3, 3, 3, 3, }, { 48000, 48000, 0, 4, 4, 4, 4, 4, }, { 48001, 999999, 0, 0, 0, 0, 0, 0, }, }; /* conversion of old LC tuning tables to new (LD enc) structure (only entries * which are actually used were converted) */ static const PNS_INFO_TAB pnsInfoTab_lowComplexity[] = { /*0 pns off */ /* DEFAULT parameter set */ /*1*/ {4100, FL2FXCONST_SGL(0.03), FL2FXCONST_SGL(0.16), 1100, 1400, FL2FXCONST_SGL(0.5), 16, USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | USE_TNS_PNS | JUST_LONG_WINDOW}, /*2*/ {4100, FL2FXCONST_SGL(0.05), FL2FXCONST_SGL(0.10), 1410, 1400, FL2FXCONST_SGL(0.5), 16, USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | USE_TNS_PNS | JUST_LONG_WINDOW}, /*3*/ {4100, FL2FXCONST_SGL(0.05), FL2FXCONST_SGL(0.10), 1100, 1400, FL2FXCONST_SGL(0.5), 16, USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | USE_TNS_PNS | JUST_LONG_WINDOW}, /* LOWSUBST -> PNS is used less often than with DEFAULT parameter set (for br: 48000 - 79999) */ /*4*/ {4100, FL2FXCONST_SGL(0.20), FL2FXCONST_SGL(0.10), 1410, 1400, FL2FXCONST_SGL(0.5), 16, USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | USE_TNS_PNS | JUST_LONG_WINDOW}, }; /**************************************************************************** function to look up used pns level ****************************************************************************/ int FDKaacEnc_lookUpPnsUse(int bitRate, int sampleRate, int numChan, const int isLC) { int hUsePns = 0, size, i; const AUTO_PNS_TAB *levelTable; if (isLC) { levelTable = &levelTable_lowComplexity[0]; size = sizeof(levelTable_lowComplexity); } else { /* (E)LD */ levelTable = (numChan > 1) ? &levelTable_stereo[0] : &levelTable_mono[0]; size = (numChan > 1) ? sizeof(levelTable_stereo) : sizeof(levelTable_mono); } for (i = 0; i < (int)(size / sizeof(AUTO_PNS_TAB)); i++) { if (((ULONG)bitRate >= levelTable[i].brFrom) && ((ULONG)bitRate <= levelTable[i].brTo)) break; } /* sanity check */ if ((int)(sizeof(pnsInfoTab) / sizeof(PNS_INFO_TAB)) < i) { return (PNS_TABLE_ERROR); } switch (sampleRate) { case 16000: hUsePns = levelTable[i].S16000; break; case 22050: hUsePns = levelTable[i].S22050; break; case 24000: hUsePns = levelTable[i].S24000; break; case 32000: hUsePns = levelTable[i].S32000; break; case 44100: hUsePns = levelTable[i].S44100; break; case 48000: hUsePns = levelTable[i].S48000; break; default: if (isLC) { hUsePns = levelTable[i].S48000; } break; } return (hUsePns); } /***************************************************************************** functionname: FDKaacEnc_GetPnsParam description: Gets PNS parameters depending on bitrate and bandwidth or bitsPerLine returns: error status input: Noiseparams struct, bitrate, sampling rate, number of sfb's, pointer to sfb offset output: PNS parameters *****************************************************************************/ AAC_ENCODER_ERROR FDKaacEnc_GetPnsParam(NOISEPARAMS *np, INT bitRate, INT sampleRate, INT sfbCnt, const INT *sfbOffset, INT *usePns, INT numChan, const INT isLC) { int i, hUsePns; const PNS_INFO_TAB *pnsInfo; if (*usePns <= 0) return AAC_ENC_OK; if (isLC) { np->detectionAlgorithmFlags = IS_LOW_COMPLEXITY; pnsInfo = pnsInfoTab_lowComplexity; /* new pns params */ hUsePns = FDKaacEnc_lookUpPnsUse(bitRate, sampleRate, numChan, isLC); if (hUsePns == 0) { *usePns = 0; return AAC_ENC_OK; } if (hUsePns == PNS_TABLE_ERROR) { return AAC_ENC_PNS_TABLE_ERROR; } /* select correct row of tuning table */ pnsInfo += hUsePns - 1; } else { np->detectionAlgorithmFlags = 0; pnsInfo = pnsInfoTab; /* new pns params */ hUsePns = FDKaacEnc_lookUpPnsUse(bitRate, sampleRate, numChan, isLC); if (hUsePns == 0) { *usePns = 0; return AAC_ENC_OK; } if (hUsePns == PNS_TABLE_ERROR) return AAC_ENC_PNS_TABLE_ERROR; /* select correct row of tuning table */ pnsInfo += hUsePns - 1; } np->startSfb = FDKaacEnc_FreqToBandWidthRounding( pnsInfo->startFreq, sampleRate, sfbCnt, sfbOffset); np->detectionAlgorithmFlags |= pnsInfo->detectionAlgorithmFlags; np->refPower = FX_SGL2FX_DBL(pnsInfo->refPower); np->refTonality = FX_SGL2FX_DBL(pnsInfo->refTonality); np->tnsGainThreshold = pnsInfo->tnsGainThreshold; np->tnsPNSGainThreshold = pnsInfo->tnsPNSGainThreshold; np->minSfbWidth = pnsInfo->minSfbWidth; np->gapFillThr = pnsInfo->gapFillThr; /* for LC it is always FL2FXCONST_SGL(0.5) */ /* assuming a constant dB/Hz slope in the signal's PSD curve, the detection threshold needs to be corrected for the width of the band */ for (i = 0; i < (sfbCnt - 1); i++) { INT qtmp, sfbWidth; FIXP_DBL tmp; sfbWidth = sfbOffset[i + 1] - sfbOffset[i]; tmp = fPow(np->refPower, 0, sfbWidth, DFRACT_BITS - 1 - 5, &qtmp); np->powDistPSDcurve[i] = (FIXP_SGL)((LONG)(scaleValue(tmp, qtmp) >> 16)); } np->powDistPSDcurve[sfbCnt] = np->powDistPSDcurve[sfbCnt - 1]; return AAC_ENC_OK; } fdk-aac-2.0.2/libAACenc/src/pnsparam.h000066400000000000000000000137601404225310700173300ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Lohwasser Description: PNS parameters depending on bitrate and bandwidth *******************************************************************************/ #ifndef PNSPARAM_H #define PNSPARAM_H #include "aacenc.h" #include "common_fix.h" #include "psy_const.h" #define NUM_PNSINFOTAB 4 #define PNS_TABLE_ERROR -1 /* detection algorithm flags */ #define USE_POWER_DISTRIBUTION (1 << 0) #define USE_PSYCH_TONALITY (1 << 1) #define USE_TNS_GAIN_THR (1 << 2) #define USE_TNS_PNS (1 << 3) #define JUST_LONG_WINDOW (1 << 4) /* additional algorithm flags */ #define IS_LOW_COMPLEXITY (1 << 5) typedef struct { /* PNS start band */ short startSfb; /* detection algorithm flags */ USHORT detectionAlgorithmFlags; /* Parameters for detection */ FIXP_DBL refPower; FIXP_DBL refTonality; INT tnsGainThreshold; INT tnsPNSGainThreshold; INT minSfbWidth; FIXP_SGL powDistPSDcurve[MAX_GROUPED_SFB]; FIXP_SGL gapFillThr; } NOISEPARAMS; int FDKaacEnc_lookUpPnsUse(int bitRate, int sampleRate, int numChan, const int isLC); /****** Definition of prototypes ******/ AAC_ENCODER_ERROR FDKaacEnc_GetPnsParam(NOISEPARAMS *np, INT bitRate, INT sampleRate, INT sfbCnt, const INT *sfbOffset, INT *usePns, INT numChan, const INT isLC); #endif /* PNSPARAM_H */ fdk-aac-2.0.2/libAACenc/src/pre_echo_control.cpp000066400000000000000000000165621404225310700213710ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M.Werner Description: Pre echo control *******************************************************************************/ #include "pre_echo_control.h" #include "psy_configuration.h" void FDKaacEnc_InitPreEchoControl(FIXP_DBL *RESTRICT pbThresholdNm1, INT *calcPreEcho, INT numPb, FIXP_DBL *RESTRICT sfbPcmQuantThreshold, INT *mdctScalenm1) { *mdctScalenm1 = PCM_QUANT_THR_SCALE >> 1; FDKmemcpy(pbThresholdNm1, sfbPcmQuantThreshold, numPb * sizeof(FIXP_DBL)); *calcPreEcho = 1; } void FDKaacEnc_PreEchoControl(FIXP_DBL *RESTRICT pbThresholdNm1, INT calcPreEcho, INT numPb, INT maxAllowedIncreaseFactor, FIXP_SGL minRemainingThresholdFactor, FIXP_DBL *RESTRICT pbThreshold, INT mdctScale, INT *mdctScalenm1) { int i; FIXP_DBL tmpThreshold1, tmpThreshold2; int scaling; /* If lastWindowSequence in previous frame was start- or stop-window, skip preechocontrol calculation */ if (calcPreEcho == 0) { /* copy thresholds to internal memory */ FDKmemcpy(pbThresholdNm1, pbThreshold, numPb * sizeof(FIXP_DBL)); *mdctScalenm1 = mdctScale; return; } if (mdctScale > *mdctScalenm1) { /* if current thresholds are downscaled more than the ones from the last * block */ scaling = 2 * (mdctScale - *mdctScalenm1); for (i = 0; i < numPb; i++) { /* multiplication with return data type fract ist equivalent to int * multiplication */ FDK_ASSERT(scaling >= 0); tmpThreshold1 = maxAllowedIncreaseFactor * (pbThresholdNm1[i] >> scaling); tmpThreshold2 = fMult(minRemainingThresholdFactor, pbThreshold[i]); FIXP_DBL tmp = pbThreshold[i]; /* copy thresholds to internal memory */ pbThresholdNm1[i] = tmp; tmp = fixMin(tmp, tmpThreshold1); pbThreshold[i] = fixMax(tmp, tmpThreshold2); } } else { /* if thresholds of last block are more downscaled than the current ones */ scaling = 2 * (*mdctScalenm1 - mdctScale); for (i = 0; i < numPb; i++) { /* multiplication with return data type fract ist equivalent to int * multiplication */ tmpThreshold1 = (maxAllowedIncreaseFactor >> 1) * pbThresholdNm1[i]; tmpThreshold2 = fMult(minRemainingThresholdFactor, pbThreshold[i]); /* copy thresholds to internal memory */ pbThresholdNm1[i] = pbThreshold[i]; FDK_ASSERT(scaling >= 0); if ((pbThreshold[i] >> (scaling + 1)) > tmpThreshold1) { pbThreshold[i] = tmpThreshold1 << (scaling + 1); } pbThreshold[i] = fixMax(pbThreshold[i], tmpThreshold2); } } *mdctScalenm1 = mdctScale; } fdk-aac-2.0.2/libAACenc/src/pre_echo_control.h000066400000000000000000000125251404225310700210310ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M.Werner Description: Pre echo control *******************************************************************************/ #ifndef PRE_ECHO_CONTROL_H #define PRE_ECHO_CONTROL_H #include "common_fix.h" void FDKaacEnc_InitPreEchoControl(FIXP_DBL *pbThresholdnm1, INT *calcPreEcho, INT numPb, FIXP_DBL *sfbPcmQuantThreshold, INT *mdctScalenm1); void FDKaacEnc_PreEchoControl(FIXP_DBL *pbThresholdNm1, INT calcPreEcho, INT numPb, INT maxAllowedIncreaseFactor, FIXP_SGL minRemainingThresholdFactor, FIXP_DBL *pbThreshold, INT mdctScale, INT *mdctScalenm1); #endif fdk-aac-2.0.2/libAACenc/src/psy_configuration.cpp000066400000000000000000000561631404225310700216100ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M.Werner Description: Psychoaccoustic configuration *******************************************************************************/ #include "psy_configuration.h" #include "adj_thr.h" #include "aacEnc_rom.h" #include "genericStds.h" #include "FDK_trigFcts.h" typedef struct { LONG sampleRate; const SFB_PARAM_LONG *paramLong; const SFB_PARAM_SHORT *paramShort; } SFB_INFO_TAB; static const SFB_INFO_TAB sfbInfoTab[] = { {8000, &p_FDKaacEnc_8000_long_1024, &p_FDKaacEnc_8000_short_128}, {11025, &p_FDKaacEnc_11025_long_1024, &p_FDKaacEnc_11025_short_128}, {12000, &p_FDKaacEnc_12000_long_1024, &p_FDKaacEnc_12000_short_128}, {16000, &p_FDKaacEnc_16000_long_1024, &p_FDKaacEnc_16000_short_128}, {22050, &p_FDKaacEnc_22050_long_1024, &p_FDKaacEnc_22050_short_128}, {24000, &p_FDKaacEnc_24000_long_1024, &p_FDKaacEnc_24000_short_128}, {32000, &p_FDKaacEnc_32000_long_1024, &p_FDKaacEnc_32000_short_128}, {44100, &p_FDKaacEnc_44100_long_1024, &p_FDKaacEnc_44100_short_128}, {48000, &p_FDKaacEnc_48000_long_1024, &p_FDKaacEnc_48000_short_128}, {64000, &p_FDKaacEnc_64000_long_1024, &p_FDKaacEnc_64000_short_128}, {88200, &p_FDKaacEnc_88200_long_1024, &p_FDKaacEnc_88200_short_128}, {96000, &p_FDKaacEnc_96000_long_1024, &p_FDKaacEnc_96000_short_128} }; /* 22050 and 24000 Hz */ static const SFB_PARAM_LONG p_22050_long_512 = { 31, {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 20, 24, 28, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32}}; /* 32000 Hz */ static const SFB_PARAM_LONG p_32000_long_512 = { 37, {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 12, 12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 32, 32, 32, 32, 32, 32, 32}}; /* 44100 Hz */ static const SFB_PARAM_LONG p_44100_long_512 = { 36, {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 12, 12, 12, 12, 16, 20, 24, 28, 32, 32, 32, 32, 32, 32, 32, 52}}; static const SFB_INFO_TAB sfbInfoTabLD512[] = { {8000, &p_22050_long_512, NULL}, {11025, &p_22050_long_512, NULL}, {12000, &p_22050_long_512, NULL}, {16000, &p_22050_long_512, NULL}, {22050, &p_22050_long_512, NULL}, {24000, &p_22050_long_512, NULL}, {32000, &p_32000_long_512, NULL}, {44100, &p_44100_long_512, NULL}, {48000, &p_44100_long_512, NULL}, {64000, &p_44100_long_512, NULL}, {88200, &p_44100_long_512, NULL}, {96000, &p_44100_long_512, NULL}, {128000, &p_44100_long_512, NULL}, {176400, &p_44100_long_512, NULL}, {192000, &p_44100_long_512, NULL}, {256000, &p_44100_long_512, NULL}, {352800, &p_44100_long_512, NULL}, {384000, &p_44100_long_512, NULL}, }; /* 22050 and 24000 Hz */ static const SFB_PARAM_LONG p_22050_long_480 = { 30, {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 20, 24, 28, 32, 32, 32, 32, 32, 32, 32, 32, 32}}; /* 32000 Hz */ static const SFB_PARAM_LONG p_32000_long_480 = { 37, {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 12, 12, 12, 16, 16, 20, 24, 32, 32, 32, 32, 32, 32, 32, 32}}; /* 44100 Hz */ static const SFB_PARAM_LONG p_44100_long_480 = { 35, {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 12, 12, 12, 12, 12, 16, 16, 24, 28, 32, 32, 32, 32, 32, 32, 48}}; static const SFB_INFO_TAB sfbInfoTabLD480[] = { {8000, &p_22050_long_480, NULL}, {11025, &p_22050_long_480, NULL}, {12000, &p_22050_long_480, NULL}, {16000, &p_22050_long_480, NULL}, {22050, &p_22050_long_480, NULL}, {24000, &p_22050_long_480, NULL}, {32000, &p_32000_long_480, NULL}, {44100, &p_44100_long_480, NULL}, {48000, &p_44100_long_480, NULL}, {64000, &p_44100_long_480, NULL}, {88200, &p_44100_long_480, NULL}, {96000, &p_44100_long_480, NULL}, {128000, &p_44100_long_480, NULL}, {176400, &p_44100_long_480, NULL}, {192000, &p_44100_long_480, NULL}, {256000, &p_44100_long_480, NULL}, {352800, &p_44100_long_480, NULL}, {384000, &p_44100_long_480, NULL}, }; /* Fixed point precision definitions */ #define Q_BARCVAL (25) AAC_ENCODER_ERROR FDKaacEnc_initSfbTable(const LONG sampleRate, const INT blockType, const INT granuleLength, INT *const sfbOffset, INT *const sfbCnt) { INT i, specStartOffset = 0; INT granuleLengthWindow = granuleLength; const UCHAR *sfbWidth = NULL; const SFB_INFO_TAB *sfbInfo = NULL; int size; /* select table */ switch (granuleLength) { case 1024: case 960: sfbInfo = sfbInfoTab; size = (INT)(sizeof(sfbInfoTab) / sizeof(SFB_INFO_TAB)); break; case 512: sfbInfo = sfbInfoTabLD512; size = sizeof(sfbInfoTabLD512); break; case 480: sfbInfo = sfbInfoTabLD480; size = sizeof(sfbInfoTabLD480); break; default: return AAC_ENC_INVALID_FRAME_LENGTH; } for (i = 0; i < size; i++) { if (sfbInfo[i].sampleRate == sampleRate) { switch (blockType) { case LONG_WINDOW: case START_WINDOW: case STOP_WINDOW: sfbWidth = sfbInfo[i].paramLong->sfbWidth; *sfbCnt = sfbInfo[i].paramLong->sfbCnt; break; case SHORT_WINDOW: sfbWidth = sfbInfo[i].paramShort->sfbWidth; *sfbCnt = sfbInfo[i].paramShort->sfbCnt; granuleLengthWindow /= TRANS_FAC; break; } break; } } if (i == size) { return AAC_ENC_UNSUPPORTED_SAMPLINGRATE; } /* calc sfb offsets */ for (i = 0; i < *sfbCnt; i++) { sfbOffset[i] = specStartOffset; specStartOffset += sfbWidth[i]; if (specStartOffset >= granuleLengthWindow) { i++; break; } } *sfbCnt = fixMin(i, *sfbCnt); sfbOffset[*sfbCnt] = fixMin(specStartOffset, granuleLengthWindow); return AAC_ENC_OK; } /***************************************************************************** functionname: FDKaacEnc_BarcLineValue description: Calculates barc value for one frequency line returns: barc value of line input: number of lines in transform, index of line to check, Fs output: *****************************************************************************/ static FIXP_DBL FDKaacEnc_BarcLineValue(INT noOfLines, INT fftLine, LONG samplingFreq) { FIXP_DBL FOURBY3EM4 = (FIXP_DBL)0x45e7b273; /* 4.0/3 * 0.0001 in q43 */ FIXP_DBL PZZZ76 = (FIXP_DBL)0x639d5e4a; /* 0.00076 in q41 */ FIXP_DBL ONE3P3 = (FIXP_DBL)0x35333333; /* 13.3 in q26 */ FIXP_DBL THREEP5 = (FIXP_DBL)0x1c000000; /* 3.5 in q27 */ FIXP_DBL INV480 = (FIXP_DBL)0x44444444; // 1/480 in q39 FIXP_DBL center_freq, x1, x2; FIXP_DBL bvalFFTLine, atan1, atan2; /* Theoritical maximum of center_freq (samp_freq*0.5) is 96khz * 0.5 = 48000 */ /* Theoritical maximum of x1 is 1.3333333e-4f * center_freq = 6.4, can keep in * q28 */ /* Theoritical maximum of x2 is 0.00076f * center_freq = 36.48, can keep in * q25 */ center_freq = fftLine * samplingFreq; /* q11 or q8 */ switch (noOfLines) { case 1024: center_freq = center_freq << 2; /* q13 */ break; case 128: center_freq = center_freq << 5; /* q13 */ break; case 512: center_freq = (fftLine * samplingFreq) << 3; // q13 break; case 480: center_freq = fMult(center_freq, INV480) << 4; // q13 break; default: center_freq = (FIXP_DBL)0; } x1 = fMult(center_freq, FOURBY3EM4); /* q13 * q43 - (DFRACT_BITS-1) = q25 */ x2 = fMult(center_freq, PZZZ76) << 2; /* q13 * q41 - (DFRACT_BITS-1) + 2 = q25 */ atan1 = fixp_atan(x1); atan2 = fixp_atan(x2); /* q25 (q26 * q30 - (DFRACT_BITS-1)) + q25 (q27 * q30 * q30) */ bvalFFTLine = fMult(ONE3P3, atan2) + fMult(THREEP5, fMult(atan1, atan1)); return (bvalFFTLine); } /* do not consider energies below a certain input signal level, i.e. of -96dB or 1 bit at 16 bit PCM resolution, might need to be configurable to e.g. 24 bit PCM Input or a lower resolution for low bit rates */ static void FDKaacEnc_InitMinPCMResolution(int numPb, int *pbOffset, FIXP_DBL *sfbPCMquantThreshold) { /* PCM_QUANT_NOISE = FDKpow(10.0f, - 20.f / 10.0f) * ABS_LOW * NORM_PCM_ENERGY * * FDKpow(2,PCM_QUANT_THR_SCALE) */ #define PCM_QUANT_NOISE ((FIXP_DBL)0x00547062) for (int i = 0; i < numPb; i++) { sfbPCMquantThreshold[i] = (pbOffset[i + 1] - pbOffset[i]) * PCM_QUANT_NOISE; } } static FIXP_DBL getMaskFactor(const FIXP_DBL dbVal_fix, const INT dbVal_e, const FIXP_DBL ten_fix, const INT ten_e) { INT q_msk; FIXP_DBL mask_factor; mask_factor = fPow(ten_fix, DFRACT_BITS - 1 - ten_e, -dbVal_fix, DFRACT_BITS - 1 - dbVal_e, &q_msk); q_msk = fixMin(DFRACT_BITS - 1, fixMax(-(DFRACT_BITS - 1), q_msk)); if ((q_msk > 0) && (mask_factor > (FIXP_DBL)MAXVAL_DBL >> q_msk)) { mask_factor = (FIXP_DBL)MAXVAL_DBL; } else { mask_factor = scaleValue(mask_factor, q_msk); } return (mask_factor); } static void FDKaacEnc_initSpreading(INT numPb, FIXP_DBL *pbBarcValue, FIXP_DBL *pbMaskLoFactor, FIXP_DBL *pbMaskHiFactor, FIXP_DBL *pbMaskLoFactorSprEn, FIXP_DBL *pbMaskHiFactorSprEn, const LONG bitrate, const INT blockType) { INT i; FIXP_DBL MASKLOWSPREN, MASKHIGHSPREN; FIXP_DBL MASKHIGH = (FIXP_DBL)0x30000000; /* 1.5 in q29 */ FIXP_DBL MASKLOW = (FIXP_DBL)0x60000000; /* 3.0 in q29 */ FIXP_DBL MASKLOWSPRENLONG = (FIXP_DBL)0x60000000; /* 3.0 in q29 */ FIXP_DBL MASKHIGHSPRENLONG = (FIXP_DBL)0x40000000; /* 2.0 in q29 */ FIXP_DBL MASKHIGHSPRENLONGLOWBR = (FIXP_DBL)0x30000000; /* 1.5 in q29 */ FIXP_DBL MASKLOWSPRENSHORT = (FIXP_DBL)0x40000000; /* 2.0 in q29 */ FIXP_DBL MASKHIGHSPRENSHORT = (FIXP_DBL)0x30000000; /* 1.5 in q29 */ FIXP_DBL TEN = (FIXP_DBL)0x50000000; /* 10.0 in q27 */ if (blockType != SHORT_WINDOW) { MASKLOWSPREN = MASKLOWSPRENLONG; MASKHIGHSPREN = (bitrate > 20000) ? MASKHIGHSPRENLONG : MASKHIGHSPRENLONGLOWBR; } else { MASKLOWSPREN = MASKLOWSPRENSHORT; MASKHIGHSPREN = MASKHIGHSPRENSHORT; } for (i = 0; i < numPb; i++) { if (i > 0) { pbMaskHiFactor[i] = getMaskFactor( fMult(MASKHIGH, (pbBarcValue[i] - pbBarcValue[i - 1])), 23, TEN, 27); pbMaskLoFactor[i - 1] = getMaskFactor( fMult(MASKLOW, (pbBarcValue[i] - pbBarcValue[i - 1])), 23, TEN, 27); pbMaskHiFactorSprEn[i] = getMaskFactor( fMult(MASKHIGHSPREN, (pbBarcValue[i] - pbBarcValue[i - 1])), 23, TEN, 27); pbMaskLoFactorSprEn[i - 1] = getMaskFactor( fMult(MASKLOWSPREN, (pbBarcValue[i] - pbBarcValue[i - 1])), 23, TEN, 27); } else { pbMaskHiFactor[i] = (FIXP_DBL)0; pbMaskLoFactor[numPb - 1] = (FIXP_DBL)0; pbMaskHiFactorSprEn[i] = (FIXP_DBL)0; pbMaskLoFactorSprEn[numPb - 1] = (FIXP_DBL)0; } } } static void FDKaacEnc_initBarcValues(INT numPb, INT *pbOffset, INT numLines, INT samplingFrequency, FIXP_DBL *pbBval) { INT i; FIXP_DBL MAX_BARC = (FIXP_DBL)0x30000000; /* 24.0 in q25 */ for (i = 0; i < numPb; i++) { FIXP_DBL v1, v2, cur_bark; v1 = FDKaacEnc_BarcLineValue(numLines, pbOffset[i], samplingFrequency); v2 = FDKaacEnc_BarcLineValue(numLines, pbOffset[i + 1], samplingFrequency); cur_bark = (v1 >> 1) + (v2 >> 1); pbBval[i] = fixMin(cur_bark, MAX_BARC); } } static void FDKaacEnc_initMinSnr(const LONG bitrate, const LONG samplerate, const INT numLines, const INT *sfbOffset, const INT sfbActive, const INT blockType, FIXP_DBL *sfbMinSnrLdData) { INT sfb; /* Fix conversion variables */ INT qbfac, qperwin, qdiv, qpeprt_const, qpeprt; INT qtmp, qsnr, sfbWidth; FIXP_DBL MAX_BARC = (FIXP_DBL)0x30000000; /* 24.0 in q25 */ FIXP_DBL MAX_BARCP1 = (FIXP_DBL)0x32000000; /* 25.0 in q25 */ FIXP_DBL BITS2PEFAC = (FIXP_DBL)0x4b851eb8; /* 1.18 in q30 */ FIXP_DBL PERS2P4 = (FIXP_DBL)0x624dd2f2; /* 0.024 in q36 */ FIXP_DBL ONEP5 = (FIXP_DBL)0x60000000; /* 1.5 in q30 */ FIXP_DBL MAX_SNR = (FIXP_DBL)0x33333333; /* 0.8 in q30 */ FIXP_DBL MIN_SNR = (FIXP_DBL)0x003126e9; /* 0.003 in q30 */ FIXP_DBL barcFactor, pePerWindow, pePart, barcWidth; FIXP_DBL pePart_const, tmp, snr, one_qsnr, one_point5; /* relative number of active barks */ barcFactor = fDivNorm(fixMin(FDKaacEnc_BarcLineValue( numLines, sfbOffset[sfbActive], samplerate), MAX_BARC), MAX_BARCP1, &qbfac); qbfac = DFRACT_BITS - 1 - qbfac; pePerWindow = fDivNorm(bitrate, samplerate, &qperwin); qperwin = DFRACT_BITS - 1 - qperwin; pePerWindow = fMult(pePerWindow, BITS2PEFAC); qperwin = qperwin + 30 - (DFRACT_BITS - 1); pePerWindow = fMult(pePerWindow, PERS2P4); qperwin = qperwin + 36 - (DFRACT_BITS - 1); switch (numLines) { case 1024: qperwin = qperwin - 10; break; case 128: qperwin = qperwin - 7; break; case 512: qperwin = qperwin - 9; break; case 480: qperwin = qperwin - 9; pePerWindow = fMult(pePerWindow, FL2FXCONST_DBL(480.f / 512.f)); break; } /* for short blocks it is assumed that more bits are available */ if (blockType == SHORT_WINDOW) { pePerWindow = fMult(pePerWindow, ONEP5); qperwin = qperwin + 30 - (DFRACT_BITS - 1); } pePart_const = fDivNorm(pePerWindow, barcFactor, &qdiv); qpeprt_const = qperwin - qbfac + DFRACT_BITS - 1 - qdiv; for (sfb = 0; sfb < sfbActive; sfb++) { barcWidth = FDKaacEnc_BarcLineValue(numLines, sfbOffset[sfb + 1], samplerate) - FDKaacEnc_BarcLineValue(numLines, sfbOffset[sfb], samplerate); /* adapt to sfb bands */ pePart = fMult(pePart_const, barcWidth); qpeprt = qpeprt_const + 25 - (DFRACT_BITS - 1); /* pe -> snr calculation */ sfbWidth = (sfbOffset[sfb + 1] - sfbOffset[sfb]); pePart = fDivNorm(pePart, sfbWidth, &qdiv); qpeprt += DFRACT_BITS - 1 - qdiv; tmp = f2Pow(pePart, DFRACT_BITS - 1 - qpeprt, &qtmp); qtmp = DFRACT_BITS - 1 - qtmp; /* Subtract 1.5 */ qsnr = fixMin(qtmp, 30); tmp = tmp >> (qtmp - qsnr); if ((30 + 1 - qsnr) > (DFRACT_BITS - 1)) one_point5 = (FIXP_DBL)0; else one_point5 = (FIXP_DBL)(ONEP5 >> (30 + 1 - qsnr)); snr = (tmp >> 1) - (one_point5); qsnr -= 1; /* max(snr, 1.0) */ if (qsnr > 0) one_qsnr = (FIXP_DBL)(1 << qsnr); else one_qsnr = (FIXP_DBL)0; snr = fixMax(one_qsnr, snr); /* 1/snr */ snr = fDivNorm(one_qsnr, snr, &qsnr); qsnr = DFRACT_BITS - 1 - qsnr; snr = (qsnr > 30) ? (snr >> (qsnr - 30)) : snr; /* upper limit is -1 dB */ snr = (snr > MAX_SNR) ? MAX_SNR : snr; /* lower limit is -25 dB */ snr = (snr < MIN_SNR) ? MIN_SNR : snr; snr = snr << 1; sfbMinSnrLdData[sfb] = CalcLdData(snr); } } AAC_ENCODER_ERROR FDKaacEnc_InitPsyConfiguration(INT bitrate, INT samplerate, INT bandwidth, INT blocktype, INT granuleLength, INT useIS, INT useMS, PSY_CONFIGURATION *psyConf, FB_TYPE filterbank) { AAC_ENCODER_ERROR ErrorStatus; INT sfb; FIXP_DBL sfbBarcVal[MAX_SFB]; const INT frameLengthLong = granuleLength; const INT frameLengthShort = granuleLength / TRANS_FAC; INT downscaleFactor = 1; switch (granuleLength) { case 256: case 240: downscaleFactor = 2; break; case 128: case 120: downscaleFactor = 4; break; default: downscaleFactor = 1; break; } FDKmemclear(psyConf, sizeof(PSY_CONFIGURATION)); psyConf->granuleLength = granuleLength; psyConf->filterbank = filterbank; psyConf->allowIS = (useIS) && ((bitrate / bandwidth) < 5); psyConf->allowMS = useMS; /* init sfb table */ ErrorStatus = FDKaacEnc_initSfbTable(samplerate * downscaleFactor, blocktype, granuleLength * downscaleFactor, psyConf->sfbOffset, &psyConf->sfbCnt); if (ErrorStatus != AAC_ENC_OK) return ErrorStatus; /* calculate barc values for each pb */ FDKaacEnc_initBarcValues(psyConf->sfbCnt, psyConf->sfbOffset, psyConf->sfbOffset[psyConf->sfbCnt], samplerate, sfbBarcVal); FDKaacEnc_InitMinPCMResolution(psyConf->sfbCnt, psyConf->sfbOffset, psyConf->sfbPcmQuantThreshold); /* calculate spreading function */ FDKaacEnc_initSpreading(psyConf->sfbCnt, sfbBarcVal, psyConf->sfbMaskLowFactor, psyConf->sfbMaskHighFactor, psyConf->sfbMaskLowFactorSprEn, psyConf->sfbMaskHighFactorSprEn, bitrate, blocktype); /* init ratio */ psyConf->maxAllowedIncreaseFactor = 2; /* integer */ psyConf->minRemainingThresholdFactor = (FIXP_SGL)0x0148; /* FL2FXCONST_SGL(0.01f); */ /* fract */ psyConf->clipEnergy = (FIXP_DBL)0x773593ff; /* FL2FXCONST_DBL(1.0e9*NORM_PCM_ENERGY); */ if (blocktype != SHORT_WINDOW) { psyConf->lowpassLine = (INT)((2 * bandwidth * frameLengthLong) / samplerate); psyConf->lowpassLineLFE = LFE_LOWPASS_LINE; } else { psyConf->lowpassLine = (INT)((2 * bandwidth * frameLengthShort) / samplerate); psyConf->lowpassLineLFE = 0; /* LFE only in lonf blocks */ /* psyConf->clipEnergy /= (TRANS_FAC * TRANS_FAC); */ psyConf->clipEnergy >>= 6; } for (sfb = 0; sfb < psyConf->sfbCnt; sfb++) { if (psyConf->sfbOffset[sfb] >= psyConf->lowpassLine) break; } psyConf->sfbActive = fMax(sfb, 1); for (sfb = 0; sfb < psyConf->sfbCnt; sfb++) { if (psyConf->sfbOffset[sfb] >= psyConf->lowpassLineLFE) break; } psyConf->sfbActiveLFE = sfb; psyConf->sfbActive = fMax(psyConf->sfbActive, psyConf->sfbActiveLFE); /* calculate minSnr */ FDKaacEnc_initMinSnr(bitrate, samplerate * downscaleFactor, psyConf->sfbOffset[psyConf->sfbCnt], psyConf->sfbOffset, psyConf->sfbActive, blocktype, psyConf->sfbMinSnrLdData); return AAC_ENC_OK; } fdk-aac-2.0.2/libAACenc/src/psy_configuration.h000066400000000000000000000153601404225310700212470ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M.Werner Description: Psychoaccoustic configuration *******************************************************************************/ #ifndef PSY_CONFIGURATION_H #define PSY_CONFIGURATION_H #include "aacenc.h" #include "common_fix.h" #include "psy_const.h" #include "aacenc_tns.h" #include "aacenc_pns.h" #define THR_SHIFTBITS 4 #define PCM_QUANT_THR_SCALE 16 #define BITS_PER_LINE_SHIFT 3 #define C_RATIO \ (FIXP_DBL)0x02940a10 /* FL2FXCONST_DBL(0.001258925f) << THR_SHIFTBITS; */ /* pow(10.0f, -(29.0f/10.0f)) */ typedef struct { INT sfbCnt; /* number of existing sf bands */ INT sfbActive; /* number of sf bands containing energy after lowpass */ INT sfbActiveLFE; INT sfbOffset[MAX_SFB + 1]; INT filterbank; /* LC, LD or ELD */ FIXP_DBL sfbPcmQuantThreshold[MAX_SFB]; INT maxAllowedIncreaseFactor; /* preecho control */ FIXP_SGL minRemainingThresholdFactor; INT lowpassLine; INT lowpassLineLFE; FIXP_DBL clipEnergy; /* for level dependend tmn */ FIXP_DBL sfbMaskLowFactor[MAX_SFB]; FIXP_DBL sfbMaskHighFactor[MAX_SFB]; FIXP_DBL sfbMaskLowFactorSprEn[MAX_SFB]; FIXP_DBL sfbMaskHighFactorSprEn[MAX_SFB]; FIXP_DBL sfbMinSnrLdData[MAX_SFB]; /* minimum snr (formerly known as bmax) */ TNS_CONFIG tnsConf; PNS_CONFIG pnsConf; INT granuleLength; INT allowIS; INT allowMS; } PSY_CONFIGURATION; typedef struct { UCHAR sfbCnt; /* Number of scalefactor bands */ UCHAR sfbWidth[MAX_SFB_LONG]; /* Width of scalefactor bands for long blocks */ } SFB_PARAM_LONG; typedef struct { UCHAR sfbCnt; /* Number of scalefactor bands */ UCHAR sfbWidth[MAX_SFB_SHORT]; /* Width of scalefactor bands for short blocks */ } SFB_PARAM_SHORT; AAC_ENCODER_ERROR FDKaacEnc_InitPsyConfiguration(INT bitrate, INT samplerate, INT bandwidth, INT blocktype, INT granuleLength, INT useIS, INT useMS, PSY_CONFIGURATION *psyConf, FB_TYPE filterbank); #endif /* PSY_CONFIGURATION_H */ fdk-aac-2.0.2/libAACenc/src/psy_const.h000066400000000000000000000147151404225310700175310ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M.Werner Description: Global psychoaccoustic constants *******************************************************************************/ #ifndef PSY_CONST_H #define PSY_CONST_H #define TRUE 1 #define FALSE 0 #define TRANS_FAC 8 /* encoder short long ratio */ #define FRAME_MAXLEN_SHORT ((1024) / TRANS_FAC) #define FRAME_LEN_SHORT_128 ((1024) / TRANS_FAC) #define FRAME_LEN_SHORT_120 (FRAME_LEN_LONG_960 / TRANS_FAC) /* Filterbank type*/ enum FB_TYPE { FB_LC = 0, FB_LD = 1, FB_ELD = 2 }; /* Block types */ #define N_BLOCKTYPES 6 enum { LONG_WINDOW = 0, START_WINDOW, SHORT_WINDOW, STOP_WINDOW, _LOWOV_WINDOW, /* Do not use this block type out side of block_switch.cpp */ WRONG_WINDOW }; /* Window shapes */ enum { SINE_WINDOW = 0, KBD_WINDOW = 1, LOL_WINDOW = 2 /* Low OverLap window shape for AAC-LD */ }; /* MS stuff */ enum { SI_MS_MASK_NONE = 0, SI_MS_MASK_SOME = 1, SI_MS_MASK_ALL = 2 }; #define MAX_NO_OF_GROUPS 4 #define MAX_SFB_LONG \ 51 /* 51 for a memory optimized implementation, maybe 64 for convenient \ debugging */ #define MAX_SFB_SHORT \ 15 /* 15 for a memory optimized implementation, maybe 16 for convenient \ debugging */ #define MAX_SFB \ (MAX_SFB_SHORT > MAX_SFB_LONG ? MAX_SFB_SHORT : MAX_SFB_LONG) /* = 51 */ #define MAX_GROUPED_SFB \ (MAX_NO_OF_GROUPS * MAX_SFB_SHORT > MAX_SFB_LONG \ ? MAX_NO_OF_GROUPS * MAX_SFB_SHORT \ : MAX_SFB_LONG) /* = 60 */ #define MAX_INPUT_BUFFER_SIZE (2 * (1024)) /* 2048 */ #define PCM_LEVEL 1.0f #define NORM_PCM (PCM_LEVEL / 32768.0f) #define NORM_PCM_ENERGY (NORM_PCM * NORM_PCM) #define LOG_NORM_PCM -15 #define TNS_PREDGAIN_SCALE (1000) #define LFE_LOWPASS_LINE 12 #define LFE_LOWPASS_LINE_MIN 4 #endif /* PSY_CONST_H */ fdk-aac-2.0.2/libAACenc/src/psy_data.h000066400000000000000000000155751404225310700173210ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M.Werner Description: Psychoaccoustic data *******************************************************************************/ #ifndef PSY_DATA_H #define PSY_DATA_H #include "block_switch.h" #include "mdct.h" /* Be careful with MAX_SFB_LONG as length of the .Long arrays. * sfbEnergy.Long and sfbEnergyMS.Long and sfbThreshold.Long are used as a * temporary storage for the regrouped short energies and thresholds between * FDKaacEnc_groupShortData() and BuildInterface() in FDKaacEnc_psyMain(). The * space required for this is MAX_GROUPED_SFB ( = MAX_NO_OF_GROUPS*MAX_SFB_SHORT * ). However, this is not important if unions are used (which is not possible * with pfloat). */ typedef shouldBeUnion { FIXP_DBL Long[MAX_GROUPED_SFB]; FIXP_DBL Short[TRANS_FAC][MAX_SFB_SHORT]; } SFB_THRESHOLD; typedef shouldBeUnion { FIXP_DBL Long[MAX_GROUPED_SFB]; FIXP_DBL Short[TRANS_FAC][MAX_SFB_SHORT]; } SFB_ENERGY; typedef shouldBeUnion { FIXP_DBL Long[MAX_GROUPED_SFB]; FIXP_DBL Short[TRANS_FAC][MAX_SFB_SHORT]; } SFB_LD_ENERGY; typedef shouldBeUnion { INT Long[MAX_GROUPED_SFB]; INT Short[TRANS_FAC][MAX_SFB_SHORT]; } SFB_MAX_SCALE; typedef struct { INT_PCM* psyInputBuffer; FIXP_DBL overlapAddBuffer[3 * 512 / 2]; mdct_t mdctPers; /* MDCT persistent data */ BLOCK_SWITCHING_CONTROL blockSwitchingControl; /* block switching */ FIXP_DBL sfbThresholdnm1[MAX_SFB]; /* FDKaacEnc_PreEchoControl */ INT mdctScalenm1; /* scale of last block's mdct (FDKaacEnc_PreEchoControl) */ INT calcPreEcho; INT isLFE; } PSY_STATIC; typedef struct { FIXP_DBL* mdctSpectrum; SFB_THRESHOLD sfbThreshold; /* adapt */ SFB_ENERGY sfbEnergy; /* sfb energies */ SFB_LD_ENERGY sfbEnergyLdData; /* sfb energies in ldData format */ SFB_MAX_SCALE sfbMaxScaleSpec; SFB_ENERGY sfbEnergyMS; /* mid/side sfb energies */ FIXP_DBL sfbEnergyMSLdData[MAX_GROUPED_SFB]; /* mid/side sfb energies in ldData format */ SFB_ENERGY sfbSpreadEnergy; INT mdctScale; /* exponent of data in mdctSpectrum */ INT groupedSfbOffset[MAX_GROUPED_SFB + 1]; INT sfbActive; INT lowpassLine; } PSY_DATA; #endif /* PSY_DATA_H */ fdk-aac-2.0.2/libAACenc/src/psy_main.cpp000066400000000000000000001454571404225310700176720ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M.Werner Description: Psychoaccoustic major function block *******************************************************************************/ #include "psy_const.h" #include "block_switch.h" #include "transform.h" #include "spreading.h" #include "pre_echo_control.h" #include "band_nrg.h" #include "psy_configuration.h" #include "psy_data.h" #include "ms_stereo.h" #include "interface.h" #include "psy_main.h" #include "grp_data.h" #include "tns_func.h" #include "pns_func.h" #include "tonality.h" #include "aacEnc_ram.h" #include "intensity.h" /* blending to reduce gibbs artifacts */ #define FADE_OUT_LEN 6 static const FIXP_DBL fadeOutFactor[FADE_OUT_LEN] = { 1840644096, 1533870080, 1227096064, 920322048, 613548032, 306774016}; /* forward definitions */ /***************************************************************************** functionname: FDKaacEnc_PsyNew description: allocates memory for psychoacoustic returns: an error code input: pointer to a psych handle *****************************************************************************/ AAC_ENCODER_ERROR FDKaacEnc_PsyNew(PSY_INTERNAL **phpsy, const INT nElements, const INT nChannels, UCHAR *dynamic_RAM) { AAC_ENCODER_ERROR ErrorStatus; PSY_INTERNAL *hPsy; INT i; hPsy = GetRam_aacEnc_PsyInternal(); *phpsy = hPsy; if (hPsy == NULL) { ErrorStatus = AAC_ENC_NO_MEMORY; goto bail; } for (i = 0; i < nElements; i++) { /* PSY_ELEMENT */ hPsy->psyElement[i] = GetRam_aacEnc_PsyElement(i); if (hPsy->psyElement[i] == NULL) { ErrorStatus = AAC_ENC_NO_MEMORY; goto bail; } } for (i = 0; i < nChannels; i++) { /* PSY_STATIC */ hPsy->pStaticChannels[i] = GetRam_aacEnc_PsyStatic(i); if (hPsy->pStaticChannels[i] == NULL) { ErrorStatus = AAC_ENC_NO_MEMORY; goto bail; } /* AUDIO INPUT BUFFER */ hPsy->pStaticChannels[i]->psyInputBuffer = GetRam_aacEnc_PsyInputBuffer(i); if (hPsy->pStaticChannels[i]->psyInputBuffer == NULL) { ErrorStatus = AAC_ENC_NO_MEMORY; goto bail; } } /* reusable psych memory */ hPsy->psyDynamic = GetRam_aacEnc_PsyDynamic(0, dynamic_RAM); return AAC_ENC_OK; bail: FDKaacEnc_PsyClose(phpsy, NULL); return ErrorStatus; } /***************************************************************************** functionname: FDKaacEnc_PsyOutNew description: allocates memory for psyOut struc returns: an error code input: pointer to a psych handle *****************************************************************************/ AAC_ENCODER_ERROR FDKaacEnc_PsyOutNew(PSY_OUT **phpsyOut, const INT nElements, const INT nChannels, const INT nSubFrames, UCHAR *dynamic_RAM) { AAC_ENCODER_ERROR ErrorStatus; int n, i; int elInc = 0, chInc = 0; for (n = 0; n < nSubFrames; n++) { phpsyOut[n] = GetRam_aacEnc_PsyOut(n); if (phpsyOut[n] == NULL) { ErrorStatus = AAC_ENC_NO_MEMORY; goto bail; } for (i = 0; i < nChannels; i++) { phpsyOut[n]->pPsyOutChannels[i] = GetRam_aacEnc_PsyOutChannel(chInc++); if (NULL == phpsyOut[n]->pPsyOutChannels[i]) { ErrorStatus = AAC_ENC_NO_MEMORY; goto bail; } } for (i = 0; i < nElements; i++) { phpsyOut[n]->psyOutElement[i] = GetRam_aacEnc_PsyOutElements(elInc++); if (phpsyOut[n]->psyOutElement[i] == NULL) { ErrorStatus = AAC_ENC_NO_MEMORY; goto bail; } } } /* nSubFrames */ return AAC_ENC_OK; bail: FDKaacEnc_PsyClose(NULL, phpsyOut); return ErrorStatus; } AAC_ENCODER_ERROR FDKaacEnc_psyInitStates(PSY_INTERNAL *hPsy, PSY_STATIC *psyStatic, AUDIO_OBJECT_TYPE audioObjectType) { /* init input buffer */ FDKmemclear(psyStatic->psyInputBuffer, MAX_INPUT_BUFFER_SIZE * sizeof(INT_PCM)); FDKaacEnc_InitBlockSwitching(&psyStatic->blockSwitchingControl, isLowDelay(audioObjectType)); return AAC_ENC_OK; } AAC_ENCODER_ERROR FDKaacEnc_psyInit(PSY_INTERNAL *hPsy, PSY_OUT **phpsyOut, const INT nSubFrames, const INT nMaxChannels, const AUDIO_OBJECT_TYPE audioObjectType, CHANNEL_MAPPING *cm) { AAC_ENCODER_ERROR ErrorStatus = AAC_ENC_OK; int i, ch, n, chInc = 0, resetChannels = 3; if ((nMaxChannels > 2) && (cm->nChannels == 2)) { chInc = 1; FDKaacEnc_psyInitStates(hPsy, hPsy->pStaticChannels[0], audioObjectType); } if ((nMaxChannels == 2)) { resetChannels = 0; } for (i = 0; i < cm->nElements; i++) { for (ch = 0; ch < cm->elInfo[i].nChannelsInEl; ch++) { hPsy->psyElement[i]->psyStatic[ch] = hPsy->pStaticChannels[chInc]; if (cm->elInfo[i].elType != ID_LFE) { if (chInc >= resetChannels) { FDKaacEnc_psyInitStates(hPsy, hPsy->psyElement[i]->psyStatic[ch], audioObjectType); } mdct_init(&(hPsy->psyElement[i]->psyStatic[ch]->mdctPers), NULL, 0); hPsy->psyElement[i]->psyStatic[ch]->isLFE = 0; } else { hPsy->psyElement[i]->psyStatic[ch]->isLFE = 1; } chInc++; } } for (n = 0; n < nSubFrames; n++) { chInc = 0; for (i = 0; i < cm->nElements; i++) { for (ch = 0; ch < cm->elInfo[i].nChannelsInEl; ch++) { phpsyOut[n]->psyOutElement[i]->psyOutChannel[ch] = phpsyOut[n]->pPsyOutChannels[chInc++]; } } } return ErrorStatus; } /***************************************************************************** functionname: FDKaacEnc_psyMainInit description: initializes psychoacoustic returns: an error code *****************************************************************************/ AAC_ENCODER_ERROR FDKaacEnc_psyMainInit( PSY_INTERNAL *hPsy, AUDIO_OBJECT_TYPE audioObjectType, CHANNEL_MAPPING *cm, INT sampleRate, INT granuleLength, INT bitRate, INT tnsMask, INT bandwidth, INT usePns, INT useIS, INT useMS, UINT syntaxFlags, ULONG initFlags) { AAC_ENCODER_ERROR ErrorStatus; int i, ch; int channelsEff = cm->nChannelsEff; int tnsChannels = 0; FB_TYPE filterBank; switch (FDKaacEnc_GetMonoStereoMode(cm->encMode)) { /* ... and map to tnsChannels */ case EL_MODE_MONO: tnsChannels = 1; break; case EL_MODE_STEREO: tnsChannels = 2; break; default: tnsChannels = 0; } switch (audioObjectType) { default: filterBank = FB_LC; break; case AOT_ER_AAC_LD: filterBank = FB_LD; break; case AOT_ER_AAC_ELD: filterBank = FB_ELD; break; } hPsy->granuleLength = granuleLength; ErrorStatus = FDKaacEnc_InitPsyConfiguration( bitRate / channelsEff, sampleRate, bandwidth, LONG_WINDOW, hPsy->granuleLength, useIS, useMS, &(hPsy->psyConf[0]), filterBank); if (ErrorStatus != AAC_ENC_OK) return ErrorStatus; ErrorStatus = FDKaacEnc_InitTnsConfiguration( (bitRate * tnsChannels) / channelsEff, sampleRate, tnsChannels, LONG_WINDOW, hPsy->granuleLength, isLowDelay(audioObjectType), (syntaxFlags & AC_SBR_PRESENT) ? 1 : 0, &(hPsy->psyConf[0].tnsConf), &hPsy->psyConf[0], (INT)(tnsMask & 2), (INT)(tnsMask & 8)); if (ErrorStatus != AAC_ENC_OK) return ErrorStatus; if (granuleLength > 512) { ErrorStatus = FDKaacEnc_InitPsyConfiguration( bitRate / channelsEff, sampleRate, bandwidth, SHORT_WINDOW, hPsy->granuleLength, useIS, useMS, &hPsy->psyConf[1], filterBank); if (ErrorStatus != AAC_ENC_OK) return ErrorStatus; ErrorStatus = FDKaacEnc_InitTnsConfiguration( (bitRate * tnsChannels) / channelsEff, sampleRate, tnsChannels, SHORT_WINDOW, hPsy->granuleLength, isLowDelay(audioObjectType), (syntaxFlags & AC_SBR_PRESENT) ? 1 : 0, &hPsy->psyConf[1].tnsConf, &hPsy->psyConf[1], (INT)(tnsMask & 1), (INT)(tnsMask & 4)); if (ErrorStatus != AAC_ENC_OK) return ErrorStatus; } for (i = 0; i < cm->nElements; i++) { for (ch = 0; ch < cm->elInfo[i].nChannelsInEl; ch++) { if (initFlags) { /* reset states */ FDKaacEnc_psyInitStates(hPsy, hPsy->psyElement[i]->psyStatic[ch], audioObjectType); } FDKaacEnc_InitPreEchoControl( hPsy->psyElement[i]->psyStatic[ch]->sfbThresholdnm1, &hPsy->psyElement[i]->psyStatic[ch]->calcPreEcho, hPsy->psyConf[0].sfbCnt, hPsy->psyConf[0].sfbPcmQuantThreshold, &hPsy->psyElement[i]->psyStatic[ch]->mdctScalenm1); } } ErrorStatus = FDKaacEnc_InitPnsConfiguration( &hPsy->psyConf[0].pnsConf, bitRate / channelsEff, sampleRate, usePns, hPsy->psyConf[0].sfbCnt, hPsy->psyConf[0].sfbOffset, cm->elInfo[0].nChannelsInEl, (hPsy->psyConf[0].filterbank == FB_LC)); if (ErrorStatus != AAC_ENC_OK) return ErrorStatus; if (granuleLength > 512) { ErrorStatus = FDKaacEnc_InitPnsConfiguration( &hPsy->psyConf[1].pnsConf, bitRate / channelsEff, sampleRate, usePns, hPsy->psyConf[1].sfbCnt, hPsy->psyConf[1].sfbOffset, cm->elInfo[1].nChannelsInEl, (hPsy->psyConf[1].filterbank == FB_LC)); if (ErrorStatus != AAC_ENC_OK) return ErrorStatus; } return ErrorStatus; } /***************************************************************************** functionname: FDKaacEnc_psyMain description: psychoacoustic returns: an error code This function assumes that enough input data is in the modulo buffer. *****************************************************************************/ AAC_ENCODER_ERROR FDKaacEnc_psyMain(INT channels, PSY_ELEMENT *psyElement, PSY_DYNAMIC *psyDynamic, PSY_CONFIGURATION *psyConf, PSY_OUT_ELEMENT *RESTRICT psyOutElement, INT_PCM *pInput, const UINT inputBufSize, INT *chIdx, INT totalChannels) { const INT commonWindow = 1; INT maxSfbPerGroup[(2)]; INT mdctSpectrum_e; INT ch; /* counts through channels */ INT w; /* counts through windows */ INT sfb; /* counts through scalefactor bands */ INT line; /* counts through lines */ PSY_CONFIGURATION *RESTRICT hPsyConfLong = &psyConf[0]; PSY_CONFIGURATION *RESTRICT hPsyConfShort = &psyConf[1]; PSY_OUT_CHANNEL **RESTRICT psyOutChannel = psyOutElement->psyOutChannel; FIXP_SGL sfbTonality[(2)][MAX_SFB_LONG]; PSY_STATIC **RESTRICT psyStatic = psyElement->psyStatic; PSY_DATA *RESTRICT psyData[(2)]; TNS_DATA *RESTRICT tnsData[(2)]; PNS_DATA *RESTRICT pnsData[(2)]; INT zeroSpec = TRUE; /* means all spectral lines are zero */ INT blockSwitchingOffset; PSY_CONFIGURATION *RESTRICT hThisPsyConf[(2)]; INT windowLength[(2)]; INT nWindows[(2)]; INT wOffset; INT maxSfb[(2)]; INT *pSfbMaxScaleSpec[(2)]; FIXP_DBL *pSfbEnergy[(2)]; FIXP_DBL *pSfbSpreadEnergy[(2)]; FIXP_DBL *pSfbEnergyLdData[(2)]; FIXP_DBL *pSfbEnergyMS[(2)]; FIXP_DBL *pSfbThreshold[(2)]; INT isShortWindow[(2)]; /* number of incoming time samples to be processed */ const INT nTimeSamples = psyConf->granuleLength; switch (hPsyConfLong->filterbank) { case FB_LC: blockSwitchingOffset = nTimeSamples + (9 * nTimeSamples / (2 * TRANS_FAC)); break; case FB_LD: case FB_ELD: blockSwitchingOffset = nTimeSamples; break; default: return AAC_ENC_UNSUPPORTED_FILTERBANK; } for (ch = 0; ch < channels; ch++) { psyData[ch] = &psyDynamic->psyData[ch]; tnsData[ch] = &psyDynamic->tnsData[ch]; pnsData[ch] = &psyDynamic->pnsData[ch]; psyData[ch]->mdctSpectrum = psyOutChannel[ch]->mdctSpectrum; } /* block switching */ if (hPsyConfLong->filterbank != FB_ELD) { int err; for (ch = 0; ch < channels; ch++) { C_ALLOC_SCRATCH_START(pTimeSignal, INT_PCM, (1024)) /* copy input data and use for block switching */ FDKmemcpy(pTimeSignal, pInput + chIdx[ch] * inputBufSize, nTimeSamples * sizeof(INT_PCM)); FDKaacEnc_BlockSwitching(&psyStatic[ch]->blockSwitchingControl, nTimeSamples, psyStatic[ch]->isLFE, pTimeSignal); /* fill up internal input buffer, to 2xframelength samples */ FDKmemcpy(psyStatic[ch]->psyInputBuffer + blockSwitchingOffset, pTimeSignal, (2 * nTimeSamples - blockSwitchingOffset) * sizeof(INT_PCM)); C_ALLOC_SCRATCH_END(pTimeSignal, INT_PCM, (1024)) } /* synch left and right block type */ err = FDKaacEnc_SyncBlockSwitching( &psyStatic[0]->blockSwitchingControl, (channels > 1) ? &psyStatic[1]->blockSwitchingControl : NULL, channels, commonWindow); if (err) { return AAC_ENC_UNSUPPORTED_AOT; /* mixed up LC and LD */ } } else { for (ch = 0; ch < channels; ch++) { /* copy input data and use for block switching */ FDKmemcpy(psyStatic[ch]->psyInputBuffer + blockSwitchingOffset, pInput + chIdx[ch] * inputBufSize, nTimeSamples * sizeof(INT_PCM)); } } for (ch = 0; ch < channels; ch++) isShortWindow[ch] = (psyStatic[ch]->blockSwitchingControl.lastWindowSequence == SHORT_WINDOW); /* set parameters according to window length */ for (ch = 0; ch < channels; ch++) { if (isShortWindow[ch]) { hThisPsyConf[ch] = hPsyConfShort; windowLength[ch] = psyConf->granuleLength / TRANS_FAC; nWindows[ch] = TRANS_FAC; maxSfb[ch] = MAX_SFB_SHORT; pSfbMaxScaleSpec[ch] = psyData[ch]->sfbMaxScaleSpec.Short[0]; pSfbEnergy[ch] = psyData[ch]->sfbEnergy.Short[0]; pSfbSpreadEnergy[ch] = psyData[ch]->sfbSpreadEnergy.Short[0]; pSfbEnergyLdData[ch] = psyData[ch]->sfbEnergyLdData.Short[0]; pSfbEnergyMS[ch] = psyData[ch]->sfbEnergyMS.Short[0]; pSfbThreshold[ch] = psyData[ch]->sfbThreshold.Short[0]; } else { hThisPsyConf[ch] = hPsyConfLong; windowLength[ch] = psyConf->granuleLength; nWindows[ch] = 1; maxSfb[ch] = MAX_GROUPED_SFB; pSfbMaxScaleSpec[ch] = psyData[ch]->sfbMaxScaleSpec.Long; pSfbEnergy[ch] = psyData[ch]->sfbEnergy.Long; pSfbSpreadEnergy[ch] = psyData[ch]->sfbSpreadEnergy.Long; pSfbEnergyLdData[ch] = psyData[ch]->sfbEnergyLdData.Long; pSfbEnergyMS[ch] = psyData[ch]->sfbEnergyMS.Long; pSfbThreshold[ch] = psyData[ch]->sfbThreshold.Long; } } /* Transform and get mdctScaling for all channels and windows. */ for (ch = 0; ch < channels; ch++) { /* update number of active bands */ if (psyStatic[ch]->isLFE) { psyData[ch]->sfbActive = hThisPsyConf[ch]->sfbActiveLFE; psyData[ch]->lowpassLine = hThisPsyConf[ch]->lowpassLineLFE; } else { psyData[ch]->sfbActive = hThisPsyConf[ch]->sfbActive; psyData[ch]->lowpassLine = hThisPsyConf[ch]->lowpassLine; } if (hThisPsyConf[ch]->filterbank == FB_ELD) { if (FDKaacEnc_Transform_Real_Eld( psyStatic[ch]->psyInputBuffer, psyData[ch]->mdctSpectrum, psyStatic[ch]->blockSwitchingControl.lastWindowSequence, psyStatic[ch]->blockSwitchingControl.windowShape, &psyStatic[ch]->blockSwitchingControl.lastWindowShape, nTimeSamples, &mdctSpectrum_e, hThisPsyConf[ch]->filterbank, psyStatic[ch]->overlapAddBuffer) != 0) { return AAC_ENC_UNSUPPORTED_FILTERBANK; } } else { if (FDKaacEnc_Transform_Real( psyStatic[ch]->psyInputBuffer, psyData[ch]->mdctSpectrum, psyStatic[ch]->blockSwitchingControl.lastWindowSequence, psyStatic[ch]->blockSwitchingControl.windowShape, &psyStatic[ch]->blockSwitchingControl.lastWindowShape, &psyStatic[ch]->mdctPers, nTimeSamples, &mdctSpectrum_e, hThisPsyConf[ch]->filterbank) != 0) { return AAC_ENC_UNSUPPORTED_FILTERBANK; } } for (w = 0; w < nWindows[ch]; w++) { wOffset = w * windowLength[ch]; /* Low pass / highest sfb */ FDKmemclear( &psyData[ch]->mdctSpectrum[psyData[ch]->lowpassLine + wOffset], (windowLength[ch] - psyData[ch]->lowpassLine) * sizeof(FIXP_DBL)); if ((hPsyConfLong->filterbank != FB_LC) && (psyData[ch]->lowpassLine >= FADE_OUT_LEN)) { /* Do blending to reduce gibbs artifacts */ for (int i = 0; i < FADE_OUT_LEN; i++) { psyData[ch]->mdctSpectrum[psyData[ch]->lowpassLine + wOffset - FADE_OUT_LEN + i] = fMult(psyData[ch]->mdctSpectrum[psyData[ch]->lowpassLine + wOffset - FADE_OUT_LEN + i], fadeOutFactor[i]); } } /* Check for zero spectrum. These loops will usually terminate very, very * early. */ for (line = 0; (line < psyData[ch]->lowpassLine) && (zeroSpec == TRUE); line++) { if (psyData[ch]->mdctSpectrum[line + wOffset] != (FIXP_DBL)0) { zeroSpec = FALSE; break; } } } /* w loop */ psyData[ch]->mdctScale = mdctSpectrum_e; /* rotate internal time samples */ FDKmemmove(psyStatic[ch]->psyInputBuffer, psyStatic[ch]->psyInputBuffer + nTimeSamples, nTimeSamples * sizeof(INT_PCM)); /* ... and get remaining samples from input buffer */ FDKmemcpy(psyStatic[ch]->psyInputBuffer + nTimeSamples, pInput + (2 * nTimeSamples - blockSwitchingOffset) + chIdx[ch] * inputBufSize, (blockSwitchingOffset - nTimeSamples) * sizeof(INT_PCM)); } /* ch */ /* Do some rescaling to get maximum possible accuracy for energies */ if (zeroSpec == FALSE) { /* Calc possible spectrum leftshift for each sfb (1 means: 1 bit left shift * is possible without overflow) */ INT minSpecShift = MAX_SHIFT_DBL; INT nrgShift = MAX_SHIFT_DBL; INT finalShift = MAX_SHIFT_DBL; FIXP_DBL currNrg = 0; FIXP_DBL maxNrg = 0; for (ch = 0; ch < channels; ch++) { for (w = 0; w < nWindows[ch]; w++) { wOffset = w * windowLength[ch]; FDKaacEnc_CalcSfbMaxScaleSpec( psyData[ch]->mdctSpectrum + wOffset, hThisPsyConf[ch]->sfbOffset, pSfbMaxScaleSpec[ch] + w * maxSfb[ch], psyData[ch]->sfbActive); for (sfb = 0; sfb < psyData[ch]->sfbActive; sfb++) minSpecShift = fixMin(minSpecShift, (pSfbMaxScaleSpec[ch] + w * maxSfb[ch])[sfb]); } } /* Calc possible energy leftshift for each sfb (1 means: 1 bit left shift is * possible without overflow) */ for (ch = 0; ch < channels; ch++) { for (w = 0; w < nWindows[ch]; w++) { wOffset = w * windowLength[ch]; currNrg = FDKaacEnc_CheckBandEnergyOptim( psyData[ch]->mdctSpectrum + wOffset, pSfbMaxScaleSpec[ch] + w * maxSfb[ch], hThisPsyConf[ch]->sfbOffset, psyData[ch]->sfbActive, pSfbEnergy[ch] + w * maxSfb[ch], pSfbEnergyLdData[ch] + w * maxSfb[ch], minSpecShift - 4); maxNrg = fixMax(maxNrg, currNrg); } } if (maxNrg != (FIXP_DBL)0) { nrgShift = (CountLeadingBits(maxNrg) >> 1) + (minSpecShift - 4); } /* 2check: Hasn't this decision to be made for both channels? */ /* For short windows 1 additional bit headroom is necessary to prevent * overflows when summing up energies in FDKaacEnc_groupShortData() */ if (isShortWindow[0]) nrgShift--; /* both spectrum and energies mustn't overflow */ finalShift = fixMin(minSpecShift, nrgShift); /* do not shift more than 3 bits more to the left than signal without * blockfloating point would be to avoid overflow of scaled PCM quantization * thresholds */ if (finalShift > psyData[0]->mdctScale + 3) finalShift = psyData[0]->mdctScale + 3; FDK_ASSERT(finalShift >= 0); /* right shift is not allowed */ /* correct sfbEnergy and sfbEnergyLdData with new finalShift */ FIXP_DBL ldShift = finalShift * FL2FXCONST_DBL(2.0 / 64); for (ch = 0; ch < channels; ch++) { INT maxSfb_ch = maxSfb[ch]; INT w_maxSfb_ch = 0; for (w = 0; w < nWindows[ch]; w++) { for (sfb = 0; sfb < psyData[ch]->sfbActive; sfb++) { INT scale = fixMax(0, (pSfbMaxScaleSpec[ch] + w_maxSfb_ch)[sfb] - 4); scale = fixMin((scale - finalShift) << 1, DFRACT_BITS - 1); if (scale >= 0) (pSfbEnergy[ch] + w_maxSfb_ch)[sfb] >>= (scale); else (pSfbEnergy[ch] + w_maxSfb_ch)[sfb] <<= (-scale); (pSfbThreshold[ch] + w_maxSfb_ch)[sfb] = fMult((pSfbEnergy[ch] + w_maxSfb_ch)[sfb], C_RATIO); (pSfbEnergyLdData[ch] + w_maxSfb_ch)[sfb] += ldShift; } w_maxSfb_ch += maxSfb_ch; } } if (finalShift != 0) { for (ch = 0; ch < channels; ch++) { INT wLen = windowLength[ch]; INT lowpassLine = psyData[ch]->lowpassLine; wOffset = 0; FIXP_DBL *mdctSpectrum = &psyData[ch]->mdctSpectrum[0]; for (w = 0; w < nWindows[ch]; w++) { FIXP_DBL *spectrum = &mdctSpectrum[wOffset]; for (line = 0; line < lowpassLine; line++) { spectrum[line] <<= finalShift; } wOffset += wLen; /* update sfbMaxScaleSpec */ for (sfb = 0; sfb < psyData[ch]->sfbActive; sfb++) (pSfbMaxScaleSpec[ch] + w * maxSfb[ch])[sfb] -= finalShift; } /* update mdctScale */ psyData[ch]->mdctScale -= finalShift; } } } else { /* all spectral lines are zero */ for (ch = 0; ch < channels; ch++) { psyData[ch]->mdctScale = 0; /* otherwise mdctScale would be for example 7 and PCM quantization * thresholds would be shifted 14 bits to the right causing some of * them to become 0 (which causes problems later) */ /* clear sfbMaxScaleSpec */ for (w = 0; w < nWindows[ch]; w++) { for (sfb = 0; sfb < psyData[ch]->sfbActive; sfb++) { (pSfbMaxScaleSpec[ch] + w * maxSfb[ch])[sfb] = 0; (pSfbEnergy[ch] + w * maxSfb[ch])[sfb] = (FIXP_DBL)0; (pSfbEnergyLdData[ch] + w * maxSfb[ch])[sfb] = FL2FXCONST_DBL(-1.0f); (pSfbThreshold[ch] + w * maxSfb[ch])[sfb] = (FIXP_DBL)0; } } } } /* Advance psychoacoustics: Tonality and TNS */ if ((channels >= 1) && (psyStatic[0]->isLFE)) { tnsData[0]->dataRaw.Long.subBlockInfo.tnsActive[HIFILT] = 0; tnsData[0]->dataRaw.Long.subBlockInfo.tnsActive[LOFILT] = 0; } else { for (ch = 0; ch < channels; ch++) { if (!isShortWindow[ch]) { /* tonality */ FDKaacEnc_CalculateFullTonality( psyData[ch]->mdctSpectrum, pSfbMaxScaleSpec[ch], pSfbEnergyLdData[ch], sfbTonality[ch], psyData[ch]->sfbActive, hThisPsyConf[ch]->sfbOffset, hThisPsyConf[ch]->pnsConf.usePns); } } /* ch */ if (hPsyConfLong->tnsConf.tnsActive || hPsyConfShort->tnsConf.tnsActive) { INT tnsActive[TRANS_FAC] = {0}; INT nrgScaling[2] = {0, 0}; INT tnsSpecShift = 0; for (ch = 0; ch < channels; ch++) { for (w = 0; w < nWindows[ch]; w++) { wOffset = w * windowLength[ch]; /* TNS */ FDKaacEnc_TnsDetect( tnsData[ch], &hThisPsyConf[ch]->tnsConf, &psyOutChannel[ch]->tnsInfo, hThisPsyConf[ch]->sfbCnt, psyData[ch]->mdctSpectrum + wOffset, w, psyStatic[ch]->blockSwitchingControl.lastWindowSequence); } } if (channels == 2) { FDKaacEnc_TnsSync( tnsData[1], tnsData[0], &psyOutChannel[1]->tnsInfo, &psyOutChannel[0]->tnsInfo, psyStatic[1]->blockSwitchingControl.lastWindowSequence, psyStatic[0]->blockSwitchingControl.lastWindowSequence, &hThisPsyConf[1]->tnsConf); } if (channels >= 1) { FDK_ASSERT(1 == commonWindow); /* all checks for TNS do only work for common windows (which is always set)*/ for (w = 0; w < nWindows[0]; w++) { if (isShortWindow[0]) tnsActive[w] = tnsData[0]->dataRaw.Short.subBlockInfo[w].tnsActive[HIFILT] || tnsData[0]->dataRaw.Short.subBlockInfo[w].tnsActive[LOFILT] || tnsData[channels - 1] ->dataRaw.Short.subBlockInfo[w] .tnsActive[HIFILT] || tnsData[channels - 1] ->dataRaw.Short.subBlockInfo[w] .tnsActive[LOFILT]; else tnsActive[w] = tnsData[0]->dataRaw.Long.subBlockInfo.tnsActive[HIFILT] || tnsData[0]->dataRaw.Long.subBlockInfo.tnsActive[LOFILT] || tnsData[channels - 1] ->dataRaw.Long.subBlockInfo.tnsActive[HIFILT] || tnsData[channels - 1] ->dataRaw.Long.subBlockInfo.tnsActive[LOFILT]; } } for (ch = 0; ch < channels; ch++) { if (tnsActive[0] && !isShortWindow[ch]) { /* Scale down spectrum if tns is active in one of the two channels * with same lastWindowSequence */ /* first part of threshold calculation; it's not necessary to update * sfbMaxScaleSpec */ INT shift = 1; for (sfb = 0; sfb < hThisPsyConf[ch]->lowpassLine; sfb++) { psyData[ch]->mdctSpectrum[sfb] = psyData[ch]->mdctSpectrum[sfb] >> shift; } /* update thresholds */ for (sfb = 0; sfb < psyData[ch]->sfbActive; sfb++) { pSfbThreshold[ch][sfb] >>= (2 * shift); } psyData[ch]->mdctScale += shift; /* update mdctScale */ /* calc sfbEnergies after tnsEncode again ! */ } } for (ch = 0; ch < channels; ch++) { for (w = 0; w < nWindows[ch]; w++) { wOffset = w * windowLength[ch]; FDKaacEnc_TnsEncode( &psyOutChannel[ch]->tnsInfo, tnsData[ch], hThisPsyConf[ch]->sfbCnt, &hThisPsyConf[ch]->tnsConf, hThisPsyConf[ch]->sfbOffset[psyData[ch]->sfbActive], /*hThisPsyConf[ch]->lowpassLine*/ /* filter stops before that line ! */ psyData[ch]->mdctSpectrum + wOffset, w, psyStatic[ch]->blockSwitchingControl.lastWindowSequence); if (tnsActive[w]) { /* Calc sfb-bandwise mdct-energies for left and right channel again, */ /* if tns active in current channel or in one channel with same * lastWindowSequence left and right */ FDKaacEnc_CalcSfbMaxScaleSpec(psyData[ch]->mdctSpectrum + wOffset, hThisPsyConf[ch]->sfbOffset, pSfbMaxScaleSpec[ch] + w * maxSfb[ch], psyData[ch]->sfbActive); } } } for (ch = 0; ch < channels; ch++) { for (w = 0; w < nWindows[ch]; w++) { if (tnsActive[w]) { if (isShortWindow[ch]) { FDKaacEnc_CalcBandEnergyOptimShort( psyData[ch]->mdctSpectrum + w * windowLength[ch], pSfbMaxScaleSpec[ch] + w * maxSfb[ch], hThisPsyConf[ch]->sfbOffset, psyData[ch]->sfbActive, pSfbEnergy[ch] + w * maxSfb[ch]); } else { nrgScaling[ch] = /* with tns, energy calculation can overflow; -> scaling */ FDKaacEnc_CalcBandEnergyOptimLong( psyData[ch]->mdctSpectrum, pSfbMaxScaleSpec[ch], hThisPsyConf[ch]->sfbOffset, psyData[ch]->sfbActive, pSfbEnergy[ch], pSfbEnergyLdData[ch]); tnsSpecShift = fixMax(tnsSpecShift, nrgScaling[ch]); /* nrgScaling is set only if nrg would have an overflow */ } } /* if tnsActive */ } } /* end channel loop */ /* adapt scaling to prevent nrg overflow, only for long blocks */ for (ch = 0; ch < channels; ch++) { if ((tnsSpecShift != 0) && !isShortWindow[ch]) { /* scale down spectrum, nrg's and thresholds, if there was an overflow * in sfbNrg calculation after tns */ for (line = 0; line < hThisPsyConf[ch]->lowpassLine; line++) { psyData[ch]->mdctSpectrum[line] >>= tnsSpecShift; } INT scale = (tnsSpecShift - nrgScaling[ch]) << 1; for (sfb = 0; sfb < psyData[ch]->sfbActive; sfb++) { pSfbEnergyLdData[ch][sfb] -= scale * FL2FXCONST_DBL(1.0 / LD_DATA_SCALING); pSfbEnergy[ch][sfb] >>= scale; pSfbThreshold[ch][sfb] >>= (tnsSpecShift << 1); } psyData[ch]->mdctScale += tnsSpecShift; /* update mdctScale; not necessary to update sfbMaxScaleSpec */ } } /* end channel loop */ } /* TNS active */ else { /* In case of disable TNS, reset its dynamic data. Some of its elements is * required in PNS detection below. */ FDKmemclear(psyDynamic->tnsData, sizeof(psyDynamic->tnsData)); } } /* !isLFE */ /* Advance thresholds */ for (ch = 0; ch < channels; ch++) { INT headroom; FIXP_DBL clipEnergy; INT energyShift = psyData[ch]->mdctScale * 2; INT clipNrgShift = energyShift - THR_SHIFTBITS; if (isShortWindow[ch]) headroom = 6; else headroom = 0; if (clipNrgShift >= 0) clipEnergy = hThisPsyConf[ch]->clipEnergy >> clipNrgShift; else if (clipNrgShift >= -headroom) clipEnergy = hThisPsyConf[ch]->clipEnergy << -clipNrgShift; else clipEnergy = (FIXP_DBL)MAXVAL_DBL; for (w = 0; w < nWindows[ch]; w++) { INT i; /* limit threshold to avoid clipping */ for (i = 0; i < psyData[ch]->sfbActive; i++) { *(pSfbThreshold[ch] + w * maxSfb[ch] + i) = fixMin(*(pSfbThreshold[ch] + w * maxSfb[ch] + i), clipEnergy); } /* spreading */ FDKaacEnc_SpreadingMax(psyData[ch]->sfbActive, hThisPsyConf[ch]->sfbMaskLowFactor, hThisPsyConf[ch]->sfbMaskHighFactor, pSfbThreshold[ch] + w * maxSfb[ch]); /* PCM quantization threshold */ energyShift += PCM_QUANT_THR_SCALE; if (energyShift >= 0) { energyShift = fixMin(DFRACT_BITS - 1, energyShift); for (i = 0; i < psyData[ch]->sfbActive; i++) { *(pSfbThreshold[ch] + w * maxSfb[ch] + i) = fixMax( *(pSfbThreshold[ch] + w * maxSfb[ch] + i) >> THR_SHIFTBITS, (hThisPsyConf[ch]->sfbPcmQuantThreshold[i] >> energyShift)); } } else { energyShift = fixMin(DFRACT_BITS - 1, -energyShift); for (i = 0; i < psyData[ch]->sfbActive; i++) { *(pSfbThreshold[ch] + w * maxSfb[ch] + i) = fixMax( *(pSfbThreshold[ch] + w * maxSfb[ch] + i) >> THR_SHIFTBITS, (hThisPsyConf[ch]->sfbPcmQuantThreshold[i] << energyShift)); } } if (!psyStatic[ch]->isLFE) { /* preecho control */ if (psyStatic[ch]->blockSwitchingControl.lastWindowSequence == STOP_WINDOW) { /* prevent FDKaacEnc_PreEchoControl from comparing stop thresholds with short thresholds */ for (i = 0; i < psyData[ch]->sfbActive; i++) { psyStatic[ch]->sfbThresholdnm1[i] = (FIXP_DBL)MAXVAL_DBL; } psyStatic[ch]->mdctScalenm1 = 0; psyStatic[ch]->calcPreEcho = 0; } FDKaacEnc_PreEchoControl( psyStatic[ch]->sfbThresholdnm1, psyStatic[ch]->calcPreEcho, psyData[ch]->sfbActive, hThisPsyConf[ch]->maxAllowedIncreaseFactor, hThisPsyConf[ch]->minRemainingThresholdFactor, pSfbThreshold[ch] + w * maxSfb[ch], psyData[ch]->mdctScale, &psyStatic[ch]->mdctScalenm1); psyStatic[ch]->calcPreEcho = 1; if (psyStatic[ch]->blockSwitchingControl.lastWindowSequence == START_WINDOW) { /* prevent FDKaacEnc_PreEchoControl in next frame to compare start thresholds with short thresholds */ for (i = 0; i < psyData[ch]->sfbActive; i++) { psyStatic[ch]->sfbThresholdnm1[i] = (FIXP_DBL)MAXVAL_DBL; } psyStatic[ch]->mdctScalenm1 = 0; psyStatic[ch]->calcPreEcho = 0; } } /* spread energy to avoid hole detection */ FDKmemcpy(pSfbSpreadEnergy[ch] + w * maxSfb[ch], pSfbEnergy[ch] + w * maxSfb[ch], psyData[ch]->sfbActive * sizeof(FIXP_DBL)); FDKaacEnc_SpreadingMax(psyData[ch]->sfbActive, hThisPsyConf[ch]->sfbMaskLowFactorSprEn, hThisPsyConf[ch]->sfbMaskHighFactorSprEn, pSfbSpreadEnergy[ch] + w * maxSfb[ch]); } } /* Calc bandwise energies for mid and side channel. Do it only if 2 channels * exist */ if (channels == 2) { for (w = 0; w < nWindows[1]; w++) { wOffset = w * windowLength[1]; FDKaacEnc_CalcBandNrgMSOpt( psyData[0]->mdctSpectrum + wOffset, psyData[1]->mdctSpectrum + wOffset, pSfbMaxScaleSpec[0] + w * maxSfb[0], pSfbMaxScaleSpec[1] + w * maxSfb[1], hThisPsyConf[1]->sfbOffset, psyData[0]->sfbActive, pSfbEnergyMS[0] + w * maxSfb[0], pSfbEnergyMS[1] + w * maxSfb[1], (psyStatic[1]->blockSwitchingControl.lastWindowSequence != SHORT_WINDOW), psyData[0]->sfbEnergyMSLdData, psyData[1]->sfbEnergyMSLdData); } } /* group short data (maxSfb[ch] for short blocks is determined here) */ for (ch = 0; ch < channels; ch++) { if (isShortWindow[ch]) { int sfbGrp; int noSfb = psyStatic[ch]->blockSwitchingControl.noOfGroups * hPsyConfShort->sfbCnt; /* At this point, energies and thresholds are copied/regrouped from the * ".Short" to the ".Long" arrays */ FDKaacEnc_groupShortData( psyData[ch]->mdctSpectrum, &psyData[ch]->sfbThreshold, &psyData[ch]->sfbEnergy, &psyData[ch]->sfbEnergyMS, &psyData[ch]->sfbSpreadEnergy, hPsyConfShort->sfbCnt, psyData[ch]->sfbActive, hPsyConfShort->sfbOffset, hPsyConfShort->sfbMinSnrLdData, psyData[ch]->groupedSfbOffset, &maxSfbPerGroup[ch], psyOutChannel[ch]->sfbMinSnrLdData, psyStatic[ch]->blockSwitchingControl.noOfGroups, psyStatic[ch]->blockSwitchingControl.groupLen, psyConf[1].granuleLength); /* calculate ldData arrays (short values are in .Long-arrays after * FDKaacEnc_groupShortData) */ for (sfbGrp = 0; sfbGrp < noSfb; sfbGrp += hPsyConfShort->sfbCnt) { LdDataVector(&psyData[ch]->sfbEnergy.Long[sfbGrp], &psyOutChannel[ch]->sfbEnergyLdData[sfbGrp], psyData[ch]->sfbActive); } /* calc sfbThrld and set Values smaller 2^-31 to 2^-33*/ for (sfbGrp = 0; sfbGrp < noSfb; sfbGrp += hPsyConfShort->sfbCnt) { LdDataVector(&psyData[ch]->sfbThreshold.Long[sfbGrp], &psyOutChannel[ch]->sfbThresholdLdData[sfbGrp], psyData[ch]->sfbActive); for (sfb = 0; sfb < psyData[ch]->sfbActive; sfb++) { psyOutChannel[ch]->sfbThresholdLdData[sfbGrp + sfb] = fixMax(psyOutChannel[ch]->sfbThresholdLdData[sfbGrp + sfb], FL2FXCONST_DBL(-0.515625f)); } } if (channels == 2) { for (sfbGrp = 0; sfbGrp < noSfb; sfbGrp += hPsyConfShort->sfbCnt) { LdDataVector(&psyData[ch]->sfbEnergyMS.Long[sfbGrp], &psyData[ch]->sfbEnergyMSLdData[sfbGrp], psyData[ch]->sfbActive); } } FDKmemcpy(psyOutChannel[ch]->sfbOffsets, psyData[ch]->groupedSfbOffset, (MAX_GROUPED_SFB + 1) * sizeof(INT)); } else { int i; /* maxSfb[ch] for long blocks */ for (sfb = psyData[ch]->sfbActive - 1; sfb >= 0; sfb--) { for (line = hPsyConfLong->sfbOffset[sfb + 1] - 1; line >= hPsyConfLong->sfbOffset[sfb]; line--) { if (psyData[ch]->mdctSpectrum[line] != FL2FXCONST_SGL(0.0f)) break; } if (line > hPsyConfLong->sfbOffset[sfb]) break; } maxSfbPerGroup[ch] = sfb + 1; maxSfbPerGroup[ch] = fixMax(fixMin(5, psyData[ch]->sfbActive), maxSfbPerGroup[ch]); /* sfbNrgLdData is calculated in FDKaacEnc_advancePsychLong, copy in * psyOut structure */ FDKmemcpy(psyOutChannel[ch]->sfbEnergyLdData, psyData[ch]->sfbEnergyLdData.Long, psyData[ch]->sfbActive * sizeof(FIXP_DBL)); FDKmemcpy(psyOutChannel[ch]->sfbOffsets, hPsyConfLong->sfbOffset, (MAX_GROUPED_SFB + 1) * sizeof(INT)); /* sfbMinSnrLdData modified in adjust threshold, copy necessary */ FDKmemcpy(psyOutChannel[ch]->sfbMinSnrLdData, hPsyConfLong->sfbMinSnrLdData, psyData[ch]->sfbActive * sizeof(FIXP_DBL)); /* sfbEnergyMSLdData ist already calculated in FDKaacEnc_CalcBandNrgMSOpt; * only in long case */ /* calc sfbThrld and set Values smaller 2^-31 to 2^-33*/ LdDataVector(psyData[ch]->sfbThreshold.Long, psyOutChannel[ch]->sfbThresholdLdData, psyData[ch]->sfbActive); for (i = 0; i < psyData[ch]->sfbActive; i++) { psyOutChannel[ch]->sfbThresholdLdData[i] = fixMax(psyOutChannel[ch]->sfbThresholdLdData[i], FL2FXCONST_DBL(-0.515625f)); } } } /* Intensity parameter intialization. */ for (ch = 0; ch < channels; ch++) { FDKmemclear(psyOutChannel[ch]->isBook, MAX_GROUPED_SFB * sizeof(INT)); FDKmemclear(psyOutChannel[ch]->isScale, MAX_GROUPED_SFB * sizeof(INT)); } for (ch = 0; ch < channels; ch++) { INT win = (isShortWindow[ch] ? 1 : 0); if (!psyStatic[ch]->isLFE) { /* PNS Decision */ FDKaacEnc_PnsDetect( &(psyConf[0].pnsConf), pnsData[ch], psyStatic[ch]->blockSwitchingControl.lastWindowSequence, psyData[ch]->sfbActive, maxSfbPerGroup[ch], /* count of Sfb which are not zero. */ psyOutChannel[ch]->sfbThresholdLdData, psyConf[win].sfbOffset, psyData[ch]->mdctSpectrum, psyData[ch]->sfbMaxScaleSpec.Long, sfbTonality[ch], psyOutChannel[ch]->tnsInfo.order[0][0], tnsData[ch]->dataRaw.Long.subBlockInfo.predictionGain[HIFILT], tnsData[ch]->dataRaw.Long.subBlockInfo.tnsActive[HIFILT], psyOutChannel[ch]->sfbEnergyLdData, psyOutChannel[ch]->noiseNrg); } /* !isLFE */ } /* ch */ /* stereo Processing */ if (channels == 2) { psyOutElement->toolsInfo.msDigest = MS_NONE; psyOutElement->commonWindow = commonWindow; if (psyOutElement->commonWindow) maxSfbPerGroup[0] = maxSfbPerGroup[1] = fixMax(maxSfbPerGroup[0], maxSfbPerGroup[1]); if (psyStatic[0]->blockSwitchingControl.lastWindowSequence != SHORT_WINDOW) { /* PNS preprocessing depending on ms processing: PNS not in Short Window! */ FDKaacEnc_PreProcessPnsChannelPair( psyData[0]->sfbActive, (&psyData[0]->sfbEnergy)->Long, (&psyData[1]->sfbEnergy)->Long, psyOutChannel[0]->sfbEnergyLdData, psyOutChannel[1]->sfbEnergyLdData, psyData[0]->sfbEnergyMS.Long, &(psyConf[0].pnsConf), pnsData[0], pnsData[1]); FDKaacEnc_IntensityStereoProcessing( psyData[0]->sfbEnergy.Long, psyData[1]->sfbEnergy.Long, psyData[0]->mdctSpectrum, psyData[1]->mdctSpectrum, psyData[0]->sfbThreshold.Long, psyData[1]->sfbThreshold.Long, psyOutChannel[1]->sfbThresholdLdData, psyData[0]->sfbSpreadEnergy.Long, psyData[1]->sfbSpreadEnergy.Long, psyOutChannel[0]->sfbEnergyLdData, psyOutChannel[1]->sfbEnergyLdData, &psyOutElement->toolsInfo.msDigest, psyOutElement->toolsInfo.msMask, psyConf[0].sfbCnt, psyConf[0].sfbCnt, maxSfbPerGroup[0], psyConf[0].sfbOffset, psyConf[0].allowIS && psyOutElement->commonWindow, psyOutChannel[1]->isBook, psyOutChannel[1]->isScale, pnsData); FDKaacEnc_MsStereoProcessing( psyData, psyOutChannel, psyOutChannel[1]->isBook, &psyOutElement->toolsInfo.msDigest, psyOutElement->toolsInfo.msMask, psyConf[0].allowMS, psyData[0]->sfbActive, psyData[0]->sfbActive, maxSfbPerGroup[0], psyOutChannel[0]->sfbOffsets); /* PNS postprocessing */ FDKaacEnc_PostProcessPnsChannelPair( psyData[0]->sfbActive, &(psyConf[0].pnsConf), pnsData[0], pnsData[1], psyOutElement->toolsInfo.msMask, &psyOutElement->toolsInfo.msDigest); } else { FDKaacEnc_IntensityStereoProcessing( psyData[0]->sfbEnergy.Long, psyData[1]->sfbEnergy.Long, psyData[0]->mdctSpectrum, psyData[1]->mdctSpectrum, psyData[0]->sfbThreshold.Long, psyData[1]->sfbThreshold.Long, psyOutChannel[1]->sfbThresholdLdData, psyData[0]->sfbSpreadEnergy.Long, psyData[1]->sfbSpreadEnergy.Long, psyOutChannel[0]->sfbEnergyLdData, psyOutChannel[1]->sfbEnergyLdData, &psyOutElement->toolsInfo.msDigest, psyOutElement->toolsInfo.msMask, psyStatic[0]->blockSwitchingControl.noOfGroups * hPsyConfShort->sfbCnt, psyConf[1].sfbCnt, maxSfbPerGroup[0], psyData[0]->groupedSfbOffset, psyConf[0].allowIS && psyOutElement->commonWindow, psyOutChannel[1]->isBook, psyOutChannel[1]->isScale, pnsData); /* it's OK to pass the ".Long" arrays here. They contain grouped short * data since FDKaacEnc_groupShortData() */ FDKaacEnc_MsStereoProcessing( psyData, psyOutChannel, psyOutChannel[1]->isBook, &psyOutElement->toolsInfo.msDigest, psyOutElement->toolsInfo.msMask, psyConf[1].allowMS, psyStatic[0]->blockSwitchingControl.noOfGroups * hPsyConfShort->sfbCnt, hPsyConfShort->sfbCnt, maxSfbPerGroup[0], psyOutChannel[0]->sfbOffsets); } } /* (channels == 2) */ /* PNS Coding */ for (ch = 0; ch < channels; ch++) { if (psyStatic[ch]->isLFE) { /* no PNS coding */ for (sfb = 0; sfb < psyData[ch]->sfbActive; sfb++) { psyOutChannel[ch]->noiseNrg[sfb] = NO_NOISE_PNS; } } else { FDKaacEnc_CodePnsChannel( psyData[ch]->sfbActive, &(hThisPsyConf[ch]->pnsConf), pnsData[ch]->pnsFlag, psyData[ch]->sfbEnergyLdData.Long, psyOutChannel[ch]->noiseNrg, /* this is the energy that will be written to the bitstream */ psyOutChannel[ch]->sfbThresholdLdData); } } /* build output */ for (ch = 0; ch < channels; ch++) { INT mask; int grp; psyOutChannel[ch]->maxSfbPerGroup = maxSfbPerGroup[ch]; psyOutChannel[ch]->mdctScale = psyData[ch]->mdctScale; if (isShortWindow[ch] == 0) { psyOutChannel[ch]->sfbCnt = hPsyConfLong->sfbActive; psyOutChannel[ch]->sfbPerGroup = hPsyConfLong->sfbActive; psyOutChannel[ch]->lastWindowSequence = psyStatic[ch]->blockSwitchingControl.lastWindowSequence; psyOutChannel[ch]->windowShape = psyStatic[ch]->blockSwitchingControl.windowShape; } else { INT sfbCnt = psyStatic[ch]->blockSwitchingControl.noOfGroups * hPsyConfShort->sfbCnt; psyOutChannel[ch]->sfbCnt = sfbCnt; psyOutChannel[ch]->sfbPerGroup = hPsyConfShort->sfbCnt; psyOutChannel[ch]->lastWindowSequence = SHORT_WINDOW; psyOutChannel[ch]->windowShape = SINE_WINDOW; } /* generate grouping mask */ mask = 0; for (grp = 0; grp < psyStatic[ch]->blockSwitchingControl.noOfGroups; grp++) { int j; mask <<= 1; for (j = 1; j < psyStatic[ch]->blockSwitchingControl.groupLen[grp]; j++) { mask = (mask << 1) | 1; } } psyOutChannel[ch]->groupingMask = mask; /* build interface */ FDKmemcpy(psyOutChannel[ch]->groupLen, psyStatic[ch]->blockSwitchingControl.groupLen, MAX_NO_OF_GROUPS * sizeof(INT)); FDKmemcpy(psyOutChannel[ch]->sfbEnergy, (&psyData[ch]->sfbEnergy)->Long, MAX_GROUPED_SFB * sizeof(FIXP_DBL)); FDKmemcpy(psyOutChannel[ch]->sfbSpreadEnergy, (&psyData[ch]->sfbSpreadEnergy)->Long, MAX_GROUPED_SFB * sizeof(FIXP_DBL)); // FDKmemcpy(psyOutChannel[ch]->mdctSpectrum, // psyData[ch]->mdctSpectrum, (1024)*sizeof(FIXP_DBL)); } return AAC_ENC_OK; } void FDKaacEnc_PsyClose(PSY_INTERNAL **phPsyInternal, PSY_OUT **phPsyOut) { int n, i; if (phPsyInternal != NULL) { PSY_INTERNAL *hPsyInternal = *phPsyInternal; if (hPsyInternal) { for (i = 0; i < (8); i++) { if (hPsyInternal->pStaticChannels[i]) { if (hPsyInternal->pStaticChannels[i]->psyInputBuffer) FreeRam_aacEnc_PsyInputBuffer( &hPsyInternal->pStaticChannels[i] ->psyInputBuffer); /* AUDIO INPUT BUFFER */ FreeRam_aacEnc_PsyStatic( &hPsyInternal->pStaticChannels[i]); /* PSY_STATIC */ } } for (i = 0; i < ((8)); i++) { if (hPsyInternal->psyElement[i]) FreeRam_aacEnc_PsyElement( &hPsyInternal->psyElement[i]); /* PSY_ELEMENT */ } FreeRam_aacEnc_PsyInternal(phPsyInternal); } } if (phPsyOut != NULL) { for (n = 0; n < (1); n++) { if (phPsyOut[n]) { for (i = 0; i < (8); i++) { if (phPsyOut[n]->pPsyOutChannels[i]) FreeRam_aacEnc_PsyOutChannel( &phPsyOut[n]->pPsyOutChannels[i]); /* PSY_OUT_CHANNEL */ } for (i = 0; i < ((8)); i++) { if (phPsyOut[n]->psyOutElement[i]) FreeRam_aacEnc_PsyOutElements( &phPsyOut[n]->psyOutElement[i]); /* PSY_OUT_ELEMENTS */ } FreeRam_aacEnc_PsyOut(&phPsyOut[n]); } } } } fdk-aac-2.0.2/libAACenc/src/psy_main.h000066400000000000000000000153221404225310700173220ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M.Werner Description: Psychoaccoustic major function block *******************************************************************************/ #ifndef PSY_MAIN_H #define PSY_MAIN_H #include "psy_configuration.h" #include "qc_data.h" #include "aacenc_pns.h" /* psych internal */ typedef struct { PSY_STATIC *psyStatic[(2)]; } PSY_ELEMENT; typedef struct { PSY_DATA psyData[(2)]; TNS_DATA tnsData[(2)]; PNS_DATA pnsData[(2)]; } PSY_DYNAMIC; typedef struct { PSY_CONFIGURATION psyConf[2]; /* LONG / SHORT */ PSY_ELEMENT *psyElement[((8))]; PSY_STATIC *pStaticChannels[(8)]; PSY_DYNAMIC *psyDynamic; INT granuleLength; } PSY_INTERNAL; AAC_ENCODER_ERROR FDKaacEnc_PsyNew(PSY_INTERNAL **phpsy, const INT nElements, const INT nChannels, UCHAR *dynamic_RAM); AAC_ENCODER_ERROR FDKaacEnc_PsyOutNew(PSY_OUT **phpsyOut, const INT nElements, const INT nChannels, const INT nSubFrames, UCHAR *dynamic_RAM); AAC_ENCODER_ERROR FDKaacEnc_psyInit(PSY_INTERNAL *hPsy, PSY_OUT **phpsyOut, const INT nSubFrames, const INT nMaxChannels, const AUDIO_OBJECT_TYPE audioObjectType, CHANNEL_MAPPING *cm); AAC_ENCODER_ERROR FDKaacEnc_psyMainInit( PSY_INTERNAL *hPsy, AUDIO_OBJECT_TYPE audioObjectType, CHANNEL_MAPPING *cm, INT sampleRate, INT granuleLength, INT bitRate, INT tnsMask, INT bandwidth, INT usePns, INT useIS, INT useMS, UINT syntaxFlags, ULONG initFlags); AAC_ENCODER_ERROR FDKaacEnc_psyMain(INT channels, PSY_ELEMENT *psyElement, PSY_DYNAMIC *psyDynamic, PSY_CONFIGURATION *psyConf, PSY_OUT_ELEMENT *psyOutElement, INT_PCM *pInput, const UINT inputBufSize, INT *chIdx, INT totalChannels); void FDKaacEnc_PsyClose(PSY_INTERNAL **phPsyInternal, PSY_OUT **phPsyOut); #endif /* PSY_MAIN_H */ fdk-aac-2.0.2/libAACenc/src/qc_data.h000066400000000000000000000256571404225310700171130ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Werner Description: Quantizing & coding data *******************************************************************************/ #ifndef QC_DATA_H #define QC_DATA_H #include "aacenc.h" #include "psy_const.h" #include "dyn_bits.h" #include "adj_thr_data.h" #include "line_pe.h" #include "FDK_audio.h" #include "interface.h" typedef enum { QCDATA_BR_MODE_INVALID = -1, QCDATA_BR_MODE_CBR = 0, /* Constant bit rate, given average bitrate */ QCDATA_BR_MODE_VBR_1 = 1, /* Variable bit rate, very low */ QCDATA_BR_MODE_VBR_2 = 2, /* Variable bit rate, low */ QCDATA_BR_MODE_VBR_3 = 3, /* Variable bit rate, medium */ QCDATA_BR_MODE_VBR_4 = 4, /* Variable bit rate, high */ QCDATA_BR_MODE_VBR_5 = 5, /* Variable bit rate, very high */ QCDATA_BR_MODE_FF = 6, /* Fixed frame mode. */ QCDATA_BR_MODE_SFR = 7 /* Superframe mode. */ } QCDATA_BR_MODE; typedef struct { MP4_ELEMENT_ID elType; INT instanceTag; INT nChannelsInEl; INT ChannelIndex[2]; FIXP_DBL relativeBits; } ELEMENT_INFO; typedef struct { CHANNEL_MODE encMode; INT nChannels; INT nChannelsEff; INT nElements; ELEMENT_INFO elInfo[((8))]; } CHANNEL_MAPPING; typedef struct { INT paddingRest; } PADDING; /* Quantizing & coding stage */ struct QC_INIT { CHANNEL_MAPPING *channelMapping; INT sceCpe; /* not used yet */ INT maxBits; /* maximum number of bits in reservoir */ INT averageBits; /* average number of bits we should use */ INT bitRes; INT sampleRate; /* output sample rate */ INT isLowDelay; /* if set, calc bits2PE factor depending on samplerate */ INT staticBits; /* Bits per frame consumed by transport layers. */ QCDATA_BR_MODE bitrateMode; INT meanPe; INT chBitrate; /* Bitrate/channel */ INT invQuant; INT maxIterations; /* Maximum number of allowed iterations before FDKaacEnc_crashRecovery() is applied. */ FIXP_DBL maxBitFac; INT bitrate; INT nSubFrames; /* helper variable */ INT minBits; /* minimal number of bits in one frame*/ AACENC_BITRES_MODE bitResMode; /* 0: full bitreservoir, 1: reduced bitreservoir, 2: disabled bitreservoir */ INT bitDistributionMode; /* Configure element-wise execution or execution over all elements for the pe-dependent threshold-adaption */ PADDING padding; }; typedef struct { FIXP_DBL mdctSpectrum[(1024)]; SHORT quantSpec[(1024)]; UINT maxValueInSfb[MAX_GROUPED_SFB]; INT scf[MAX_GROUPED_SFB]; INT globalGain; SECTION_DATA sectionData; FIXP_DBL sfbFormFactorLdData[MAX_GROUPED_SFB]; FIXP_DBL sfbThresholdLdData[MAX_GROUPED_SFB]; FIXP_DBL sfbMinSnrLdData[MAX_GROUPED_SFB]; FIXP_DBL sfbEnergyLdData[MAX_GROUPED_SFB]; FIXP_DBL sfbEnergy[MAX_GROUPED_SFB]; FIXP_DBL sfbWeightedEnergyLdData[MAX_GROUPED_SFB]; FIXP_DBL sfbEnFacLd[MAX_GROUPED_SFB]; FIXP_DBL sfbSpreadEnergy[MAX_GROUPED_SFB]; } QC_OUT_CHANNEL; typedef struct { EXT_PAYLOAD_TYPE type; /* type of the extension payload */ INT nPayloadBits; /* size of the payload */ UCHAR *pPayload; /* pointer to payload */ } QC_OUT_EXTENSION; typedef struct { INT staticBitsUsed; /* for verification purposes */ INT dynBitsUsed; /* for verification purposes */ INT extBitsUsed; /* bit consumption of extended fill elements */ INT nExtensions; /* number of extension payloads for this element */ QC_OUT_EXTENSION extension[(1)]; /* reffering extension payload */ INT grantedDynBits; INT grantedPe; INT grantedPeCorr; PE_DATA peData; QC_OUT_CHANNEL *qcOutChannel[(2)]; UCHAR *dynMem_Ah_Flag; /* pointer to dynamic buffer used by AhFlag in function FDKaacEnc_adaptThresholdsToPe() */ UCHAR *dynMem_Thr_Exp; /* pointer to dynamic buffer used by ThrExp in function FDKaacEnc_adaptThresholdsToPe() */ UCHAR *dynMem_SfbNActiveLinesLdData; /* pointer to dynamic buffer used by sfbNActiveLinesLdData in function FDKaacEnc_correctThresh() */ } QC_OUT_ELEMENT; typedef struct { QC_OUT_ELEMENT *qcElement[((8))]; QC_OUT_CHANNEL *pQcOutChannels[(8)]; QC_OUT_EXTENSION extension[(2 + 2)]; /* global extension payload */ INT nExtensions; /* number of extension payloads for this AU */ INT maxDynBits; /* maximal allowed dynamic bits in frame */ INT grantedDynBits; /* granted dynamic bits in frame */ INT totFillBits; /* fill bits */ INT elementExtBits; /* element associated extension payload bits, e.g. sbr, drc ... */ INT globalExtBits; /* frame/au associated extension payload bits (anc data ...) */ INT staticBits; /* aac side info bits */ INT totalNoRedPe; INT totalGrantedPeCorr; INT usedDynBits; /* number of dynamic bits in use */ INT alignBits; /* AU alignment bits */ INT totalBits; /* sum of static, dyn, sbr, fill, align and dse bits */ } QC_OUT; typedef struct { INT chBitrateEl; /* channel bitrate in element (totalbitrate*el_relativeBits/el_channels) */ INT maxBitsEl; /* used in crash recovery */ INT bitResLevelEl; /* update bitreservoir level in each call of FDKaacEnc_QCMain */ INT maxBitResBitsEl; /* nEffChannels*6144 - averageBitsInFrame */ FIXP_DBL relativeBitsEl; /* Bits relative to total Bits*/ } ELEMENT_BITS; typedef struct { /* this is basically struct QC_INIT */ INT globHdrBits; INT maxBitsPerFrame; /* maximal allowed bits per frame, 6144*nChannelsEff */ INT minBitsPerFrame; /* minimal allowd bits per fram, superframing - DRM */ INT nElements; QCDATA_BR_MODE bitrateMode; AACENC_BITRES_MODE bitResMode; /* 0: full bitreservoir, 1: reduced bitreservoir, 2: disabled bitreservoir */ INT bitResTot; INT bitResTotMax; INT maxIterations; /* Maximum number of allowed iterations before FDKaacEnc_crashRecovery() is applied. */ INT invQuant; FIXP_DBL vbrQualFactor; FIXP_DBL maxBitFac; PADDING padding; ELEMENT_BITS *elementBits[((8))]; BITCNTR_STATE *hBitCounter; ADJ_THR_STATE *hAdjThr; INT dZoneQuantEnable; /* enable dead zone quantizer */ } QC_STATE; #endif /* QC_DATA_H */ fdk-aac-2.0.2/libAACenc/src/qc_main.cpp000066400000000000000000001564741404225310700174630ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Werner Description: Quantizing & coding *******************************************************************************/ #include "qc_main.h" #include "quantize.h" #include "interface.h" #include "adj_thr.h" #include "sf_estim.h" #include "bit_cnt.h" #include "dyn_bits.h" #include "channel_map.h" #include "aacEnc_ram.h" #include "genericStds.h" #define AACENC_DZQ_BR_THR 32000 /* Dead zone quantizer bitrate threshold */ typedef struct { QCDATA_BR_MODE bitrateMode; LONG vbrQualFactor; } TAB_VBR_QUAL_FACTOR; static const TAB_VBR_QUAL_FACTOR tableVbrQualFactor[] = { {QCDATA_BR_MODE_VBR_1, FL2FXCONST_DBL(0.150f)}, /* Approx. 32 kbps mono AAC-LC + SBR + PS */ {QCDATA_BR_MODE_VBR_2, FL2FXCONST_DBL(0.162f)}, /* Approx. 64 kbps stereo AAC-LC + SBR */ {QCDATA_BR_MODE_VBR_3, FL2FXCONST_DBL(0.176f)}, /* Approx. 96 kbps stereo AAC-LC */ {QCDATA_BR_MODE_VBR_4, FL2FXCONST_DBL(0.120f)}, /* Approx. 128 kbps stereo AAC-LC */ {QCDATA_BR_MODE_VBR_5, FL2FXCONST_DBL(0.070f)} /* Approx. 192 kbps stereo AAC-LC */ }; static INT isConstantBitrateMode(const QCDATA_BR_MODE bitrateMode) { return (((bitrateMode == QCDATA_BR_MODE_CBR) || (bitrateMode == QCDATA_BR_MODE_SFR) || (bitrateMode == QCDATA_BR_MODE_FF)) ? 1 : 0); } typedef enum { FRAME_LEN_BYTES_MODULO = 1, FRAME_LEN_BYTES_INT = 2 } FRAME_LEN_RESULT_MODE; /* forward declarations */ static INT FDKaacEnc_calcMaxValueInSfb(INT sfbCnt, INT maxSfbPerGroup, INT sfbPerGroup, INT* RESTRICT sfbOffset, SHORT* RESTRICT quantSpectrum, UINT* RESTRICT maxValue); static void FDKaacEnc_crashRecovery(INT nChannels, PSY_OUT_ELEMENT* psyOutElement, QC_OUT* qcOut, QC_OUT_ELEMENT* qcElement, INT bitsToSave, AUDIO_OBJECT_TYPE aot, UINT syntaxFlags, SCHAR epConfig); static AAC_ENCODER_ERROR FDKaacEnc_reduceBitConsumption( int* iterations, const int maxIterations, int gainAdjustment, int* chConstraintsFulfilled, int* calculateQuant, int nChannels, PSY_OUT_ELEMENT* psyOutElement, QC_OUT* qcOut, QC_OUT_ELEMENT* qcOutElement, ELEMENT_BITS* elBits, AUDIO_OBJECT_TYPE aot, UINT syntaxFlags, SCHAR epConfig); void FDKaacEnc_QCClose(QC_STATE** phQCstate, QC_OUT** phQC); /***************************************************************************** functionname: FDKaacEnc_calcFrameLen description: returns: input: output: *****************************************************************************/ static INT FDKaacEnc_calcFrameLen(INT bitRate, INT sampleRate, INT granuleLength, FRAME_LEN_RESULT_MODE mode) { INT result; result = ((granuleLength) >> 3) * (bitRate); switch (mode) { case FRAME_LEN_BYTES_MODULO: result %= sampleRate; break; case FRAME_LEN_BYTES_INT: result /= sampleRate; break; } return (result); } /***************************************************************************** functionname:FDKaacEnc_framePadding description: Calculates if padding is needed for actual frame returns: input: output: *****************************************************************************/ static INT FDKaacEnc_framePadding(INT bitRate, INT sampleRate, INT granuleLength, INT* paddingRest) { INT paddingOn; INT difference; paddingOn = 0; difference = FDKaacEnc_calcFrameLen(bitRate, sampleRate, granuleLength, FRAME_LEN_BYTES_MODULO); *paddingRest -= difference; if (*paddingRest <= 0) { paddingOn = 1; *paddingRest += sampleRate; } return (paddingOn); } /********************************************************************************* functionname: FDKaacEnc_QCOutNew description: return: **********************************************************************************/ AAC_ENCODER_ERROR FDKaacEnc_QCOutNew(QC_OUT** phQC, const INT nElements, const INT nChannels, const INT nSubFrames, UCHAR* dynamic_RAM) { AAC_ENCODER_ERROR ErrorStatus; int n, i; int elInc = 0, chInc = 0; for (n = 0; n < nSubFrames; n++) { phQC[n] = GetRam_aacEnc_QCout(n); if (phQC[n] == NULL) { ErrorStatus = AAC_ENC_NO_MEMORY; goto QCOutNew_bail; } for (i = 0; i < nChannels; i++) { phQC[n]->pQcOutChannels[i] = GetRam_aacEnc_QCchannel(chInc, dynamic_RAM); if (phQC[n]->pQcOutChannels[i] == NULL) { ErrorStatus = AAC_ENC_NO_MEMORY; goto QCOutNew_bail; } chInc++; } /* nChannels */ for (i = 0; i < nElements; i++) { phQC[n]->qcElement[i] = GetRam_aacEnc_QCelement(elInc); if (phQC[n]->qcElement[i] == NULL) { ErrorStatus = AAC_ENC_NO_MEMORY; goto QCOutNew_bail; } elInc++; /* initialize pointer to dynamic buffer which are used in adjust * thresholds */ phQC[n]->qcElement[i]->dynMem_Ah_Flag = dynamic_RAM + (P_BUF_1); phQC[n]->qcElement[i]->dynMem_Thr_Exp = dynamic_RAM + (P_BUF_1) + ADJ_THR_AH_FLAG_SIZE; phQC[n]->qcElement[i]->dynMem_SfbNActiveLinesLdData = dynamic_RAM + (P_BUF_1) + ADJ_THR_AH_FLAG_SIZE + ADJ_THR_THR_EXP_SIZE; } /* nElements */ } /* nSubFrames */ return AAC_ENC_OK; QCOutNew_bail: return ErrorStatus; } /********************************************************************************* functionname: FDKaacEnc_QCOutInit description: return: **********************************************************************************/ AAC_ENCODER_ERROR FDKaacEnc_QCOutInit(QC_OUT* phQC[(1)], const INT nSubFrames, const CHANNEL_MAPPING* cm) { INT n, i, ch; for (n = 0; n < nSubFrames; n++) { INT chInc = 0; for (i = 0; i < cm->nElements; i++) { for (ch = 0; ch < cm->elInfo[i].nChannelsInEl; ch++) { phQC[n]->qcElement[i]->qcOutChannel[ch] = phQC[n]->pQcOutChannels[chInc]; chInc++; } /* chInEl */ } /* nElements */ } /* nSubFrames */ return AAC_ENC_OK; } /********************************************************************************* functionname: FDKaacEnc_QCNew description: return: **********************************************************************************/ AAC_ENCODER_ERROR FDKaacEnc_QCNew(QC_STATE** phQC, INT nElements, UCHAR* dynamic_RAM) { AAC_ENCODER_ERROR ErrorStatus; int i; QC_STATE* hQC = GetRam_aacEnc_QCstate(); *phQC = hQC; if (hQC == NULL) { ErrorStatus = AAC_ENC_NO_MEMORY; goto QCNew_bail; } if (FDKaacEnc_AdjThrNew(&hQC->hAdjThr, nElements)) { ErrorStatus = AAC_ENC_NO_MEMORY; goto QCNew_bail; } if (FDKaacEnc_BCNew(&(hQC->hBitCounter), dynamic_RAM)) { ErrorStatus = AAC_ENC_NO_MEMORY; goto QCNew_bail; } for (i = 0; i < nElements; i++) { hQC->elementBits[i] = GetRam_aacEnc_ElementBits(i); if (hQC->elementBits[i] == NULL) { ErrorStatus = AAC_ENC_NO_MEMORY; goto QCNew_bail; } } return AAC_ENC_OK; QCNew_bail: FDKaacEnc_QCClose(phQC, NULL); return ErrorStatus; } /********************************************************************************* functionname: FDKaacEnc_QCInit description: return: **********************************************************************************/ AAC_ENCODER_ERROR FDKaacEnc_QCInit(QC_STATE* hQC, struct QC_INIT* init, const ULONG initFlags) { AAC_ENCODER_ERROR err = AAC_ENC_OK; int i; hQC->maxBitsPerFrame = init->maxBits; hQC->minBitsPerFrame = init->minBits; hQC->nElements = init->channelMapping->nElements; if ((initFlags != 0) || ((init->bitrateMode != QCDATA_BR_MODE_FF) && (hQC->bitResTotMax != init->bitRes))) { hQC->bitResTot = init->bitRes; } hQC->bitResTotMax = init->bitRes; hQC->maxBitFac = init->maxBitFac; hQC->bitrateMode = init->bitrateMode; hQC->invQuant = init->invQuant; hQC->maxIterations = init->maxIterations; /* 0: full bitreservoir, 1: reduced bitreservoir, 2: disabled bitreservoir */ hQC->bitResMode = init->bitResMode; hQC->padding.paddingRest = init->padding.paddingRest; hQC->globHdrBits = init->staticBits; /* Bit overhead due to transport */ err = FDKaacEnc_InitElementBits( hQC, init->channelMapping, init->bitrate, (init->averageBits / init->nSubFrames) - hQC->globHdrBits, hQC->maxBitsPerFrame / init->channelMapping->nChannelsEff); if (err != AAC_ENC_OK) goto bail; hQC->vbrQualFactor = FL2FXCONST_DBL(0.f); for (i = 0; i < (int)(sizeof(tableVbrQualFactor) / sizeof(TAB_VBR_QUAL_FACTOR)); i++) { if (hQC->bitrateMode == tableVbrQualFactor[i].bitrateMode) { hQC->vbrQualFactor = (FIXP_DBL)tableVbrQualFactor[i].vbrQualFactor; break; } } if (init->channelMapping->nChannelsEff == 1 && (init->bitrate / init->channelMapping->nChannelsEff) < AACENC_DZQ_BR_THR && init->isLowDelay != 0) /* watch out here: init->bitrate is the bitrate "minus" the standard SBR bitrate (=2500kbps) --> for the FDK the OFFSTE tuning should start somewhere below 32000kbps-2500kbps ... so everything is fine here */ { hQC->dZoneQuantEnable = 1; } else { hQC->dZoneQuantEnable = 0; } FDKaacEnc_AdjThrInit( hQC->hAdjThr, init->meanPe, hQC->invQuant, init->channelMapping, init->sampleRate, /* output sample rate */ init->bitrate, /* total bitrate */ init->isLowDelay, /* if set, calc bits2PE factor depending on samplerate */ init->bitResMode /* for a small bitreservoir, the pe correction is calc'd differently */ , hQC->dZoneQuantEnable, init->bitDistributionMode, hQC->vbrQualFactor); bail: return err; } /********************************************************************************* functionname: FDKaacEnc_QCMainPrepare description: return: **********************************************************************************/ AAC_ENCODER_ERROR FDKaacEnc_QCMainPrepare( ELEMENT_INFO* elInfo, ATS_ELEMENT* RESTRICT adjThrStateElement, PSY_OUT_ELEMENT* RESTRICT psyOutElement, QC_OUT_ELEMENT* RESTRICT qcOutElement, AUDIO_OBJECT_TYPE aot, UINT syntaxFlags, SCHAR epConfig) { AAC_ENCODER_ERROR ErrorStatus = AAC_ENC_OK; INT nChannels = elInfo->nChannelsInEl; PSY_OUT_CHANNEL** RESTRICT psyOutChannel = psyOutElement->psyOutChannel; /* may be modified in-place */ FDKaacEnc_CalcFormFactor(qcOutElement->qcOutChannel, psyOutChannel, nChannels); /* prepare and calculate PE without reduction */ FDKaacEnc_peCalculation(&qcOutElement->peData, psyOutChannel, qcOutElement->qcOutChannel, &psyOutElement->toolsInfo, adjThrStateElement, nChannels); ErrorStatus = FDKaacEnc_ChannelElementWrite( NULL, elInfo, NULL, psyOutElement, psyOutElement->psyOutChannel, syntaxFlags, aot, epConfig, &qcOutElement->staticBitsUsed, 0); return ErrorStatus; } /********************************************************************************* functionname: FDKaacEnc_AdjustBitrate description: adjusts framelength via padding on a frame to frame basis, to achieve a bitrate that demands a non byte aligned framelength return: errorcode **********************************************************************************/ AAC_ENCODER_ERROR FDKaacEnc_AdjustBitrate( QC_STATE* RESTRICT hQC, CHANNEL_MAPPING* RESTRICT cm, INT* avgTotalBits, INT bitRate, /* total bitrate */ INT sampleRate, /* output sampling rate */ INT granuleLength) /* frame length */ { INT paddingOn; INT frameLen; /* Do we need an extra padding byte? */ paddingOn = FDKaacEnc_framePadding(bitRate, sampleRate, granuleLength, &hQC->padding.paddingRest); frameLen = paddingOn + FDKaacEnc_calcFrameLen(bitRate, sampleRate, granuleLength, FRAME_LEN_BYTES_INT); *avgTotalBits = frameLen << 3; return AAC_ENC_OK; } #define isAudioElement(elType) \ ((elType == ID_SCE) || (elType == ID_CPE) || (elType == ID_LFE)) /********************************************************************************* functionname: FDKaacEnc_distributeElementDynBits description: distributes all bits over all elements. The relative bit distibution is described in the ELEMENT_INFO of the appropriate element. The bit distribution table is initialized in FDKaacEnc_InitChannelMapping(). return: errorcode **********************************************************************************/ static AAC_ENCODER_ERROR FDKaacEnc_distributeElementDynBits( QC_STATE* hQC, QC_OUT_ELEMENT* qcElement[((8))], CHANNEL_MAPPING* cm, INT codeBits) { INT i; /* counter variable */ INT totalBits = 0; /* sum of bits over all elements */ for (i = (cm->nElements - 1); i >= 0; i--) { if (isAudioElement(cm->elInfo[i].elType)) { qcElement[i]->grantedDynBits = fMax(0, fMultI(hQC->elementBits[i]->relativeBitsEl, codeBits)); totalBits += qcElement[i]->grantedDynBits; } } /* Due to inaccuracies with the multiplication, codeBits may differ from totalBits. For that case, the difference must be added/substracted again to/from one element, i.e: Negative differences are substracted from the element with the most bits. Positive differences are added to the element with the least bits. */ if (codeBits != totalBits) { INT elMaxBits = cm->nElements - 1; /* element with the most bits */ INT elMinBits = cm->nElements - 1; /* element with the least bits */ /* Search for biggest and smallest audio element */ for (i = (cm->nElements - 1); i >= 0; i--) { if (isAudioElement(cm->elInfo[i].elType)) { if (qcElement[i]->grantedDynBits > qcElement[elMaxBits]->grantedDynBits) { elMaxBits = i; } if (qcElement[i]->grantedDynBits < qcElement[elMinBits]->grantedDynBits) { elMinBits = i; } } } /* Compensate for bit distibution difference */ if (codeBits - totalBits > 0) { qcElement[elMinBits]->grantedDynBits += codeBits - totalBits; } else { qcElement[elMaxBits]->grantedDynBits += codeBits - totalBits; } } return AAC_ENC_OK; } /** * \brief Verify whether minBitsPerFrame criterion can be satisfied. * * This function evaluates the bit consumption only if minBitsPerFrame parameter * is not 0. In hyperframing mode the difference between grantedDynBits and * usedDynBits of all sub frames results the number of fillbits to be written. * This bits can be distrubitued in superframe to reach minBitsPerFrame bit * consumption in single AU's. The return value denotes if enough desired fill * bits are available to achieve minBitsPerFrame in all frames. This check can * only be used within superframes. * * \param qcOut Pointer to coding data struct. * \param minBitsPerFrame Minimal number of bits to be consumed in each frame. * \param nSubFrames Number of frames in superframe * * \return * - 1: all fine * - 0: criterion not fulfilled */ static int checkMinFrameBitsDemand(QC_OUT** qcOut, const INT minBitsPerFrame, const INT nSubFrames) { int result = 1; /* all fine*/ return result; } //////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// //////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// /********************************************************************************* functionname: FDKaacEnc_getMinimalStaticBitdemand description: calculate minmal size of static bits by reduction , to zero spectrum and deactivating tns and MS return: number of static bits **********************************************************************************/ static int FDKaacEnc_getMinimalStaticBitdemand(CHANNEL_MAPPING* cm, PSY_OUT** psyOut) { AUDIO_OBJECT_TYPE aot = AOT_AAC_LC; UINT syntaxFlags = 0; SCHAR epConfig = -1; int i, bitcount = 0; for (i = 0; i < cm->nElements; i++) { ELEMENT_INFO elInfo = cm->elInfo[i]; if ((elInfo.elType == ID_SCE) || (elInfo.elType == ID_CPE) || (elInfo.elType == ID_LFE)) { INT minElBits = 0; FDKaacEnc_ChannelElementWrite(NULL, &elInfo, NULL, psyOut[0]->psyOutElement[i], psyOut[0]->psyOutElement[i]->psyOutChannel, syntaxFlags, aot, epConfig, &minElBits, 1); bitcount += minElBits; } } return bitcount; } //////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// static AAC_ENCODER_ERROR FDKaacEnc_prepareBitDistribution( QC_STATE* hQC, PSY_OUT** psyOut, QC_OUT** qcOut, CHANNEL_MAPPING* cm, QC_OUT_ELEMENT* qcElement[(1)][((8))], INT avgTotalBits, INT* totalAvailableBits, INT* avgTotalDynBits) { int i; /* get maximal allowed dynamic bits */ qcOut[0]->grantedDynBits = (fixMin(hQC->maxBitsPerFrame, avgTotalBits) - hQC->globHdrBits) & ~7; qcOut[0]->grantedDynBits -= (qcOut[0]->globalExtBits + qcOut[0]->staticBits + qcOut[0]->elementExtBits); qcOut[0]->maxDynBits = ((hQC->maxBitsPerFrame) & ~7) - (qcOut[0]->globalExtBits + qcOut[0]->staticBits + qcOut[0]->elementExtBits); /* assure that enough bits are available */ if ((qcOut[0]->grantedDynBits + hQC->bitResTot) < 0) { /* crash recovery allows to reduce static bits to a minimum */ if ((qcOut[0]->grantedDynBits + hQC->bitResTot) < (FDKaacEnc_getMinimalStaticBitdemand(cm, psyOut) - qcOut[0]->staticBits)) return AAC_ENC_BITRES_TOO_LOW; } /* distribute dynamic bits to each element */ FDKaacEnc_distributeElementDynBits(hQC, qcElement[0], cm, qcOut[0]->grantedDynBits); *avgTotalDynBits = 0; /*frameDynBits;*/ *totalAvailableBits = avgTotalBits; /* sum up corrected granted PE */ qcOut[0]->totalGrantedPeCorr = 0; for (i = 0; i < cm->nElements; i++) { ELEMENT_INFO elInfo = cm->elInfo[i]; int nChannels = elInfo.nChannelsInEl; if ((elInfo.elType == ID_SCE) || (elInfo.elType == ID_CPE) || (elInfo.elType == ID_LFE)) { /* for ( all sub frames ) ... */ FDKaacEnc_DistributeBits( hQC->hAdjThr, hQC->hAdjThr->adjThrStateElem[i], psyOut[0]->psyOutElement[i]->psyOutChannel, &qcElement[0][i]->peData, &qcElement[0][i]->grantedPe, &qcElement[0][i]->grantedPeCorr, nChannels, psyOut[0]->psyOutElement[i]->commonWindow, qcElement[0][i]->grantedDynBits, hQC->elementBits[i]->bitResLevelEl, hQC->elementBits[i]->maxBitResBitsEl, hQC->maxBitFac, hQC->bitResMode); *totalAvailableBits += hQC->elementBits[i]->bitResLevelEl; /* get total corrected granted PE */ qcOut[0]->totalGrantedPeCorr += qcElement[0][i]->grantedPeCorr; } /* -end- if(ID_SCE || ID_CPE || ID_LFE) */ } /* -end- element loop */ *totalAvailableBits = fMin(hQC->maxBitsPerFrame, (*totalAvailableBits)); return AAC_ENC_OK; } //////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// static AAC_ENCODER_ERROR FDKaacEnc_updateUsedDynBits( INT* sumDynBitsConsumed, QC_OUT_ELEMENT* qcElement[((8))], CHANNEL_MAPPING* cm) { INT i; *sumDynBitsConsumed = 0; for (i = 0; i < cm->nElements; i++) { ELEMENT_INFO elInfo = cm->elInfo[i]; if ((elInfo.elType == ID_SCE) || (elInfo.elType == ID_CPE) || (elInfo.elType == ID_LFE)) { /* sum up bits consumed */ *sumDynBitsConsumed += qcElement[i]->dynBitsUsed; } /* -end- if(ID_SCE || ID_CPE || ID_LFE) */ } /* -end- element loop */ return AAC_ENC_OK; } static INT FDKaacEnc_getTotalConsumedDynBits(QC_OUT** qcOut, INT nSubFrames) { INT c, totalBits = 0; /* sum up bit consumption for all sub frames */ for (c = 0; c < nSubFrames; c++) { /* bit consumption not valid if dynamic bits not available in one sub frame */ if (qcOut[c]->usedDynBits == -1) return -1; totalBits += qcOut[c]->usedDynBits; } return totalBits; } static INT FDKaacEnc_getTotalConsumedBits(QC_OUT** qcOut, QC_OUT_ELEMENT* qcElement[(1)][((8))], CHANNEL_MAPPING* cm, INT globHdrBits, INT nSubFrames) { int c, i; int totalUsedBits = 0; for (c = 0; c < nSubFrames; c++) { int dataBits = 0; for (i = 0; i < cm->nElements; i++) { if ((cm->elInfo[i].elType == ID_SCE) || (cm->elInfo[i].elType == ID_CPE) || (cm->elInfo[i].elType == ID_LFE)) { dataBits += qcElement[c][i]->dynBitsUsed + qcElement[c][i]->staticBitsUsed + qcElement[c][i]->extBitsUsed; } } dataBits += qcOut[c]->globalExtBits; totalUsedBits += (8 - (dataBits) % 8) % 8; totalUsedBits += dataBits + globHdrBits; /* header bits for every frame */ } return totalUsedBits; } static AAC_ENCODER_ERROR FDKaacEnc_BitResRedistribution( QC_STATE* const hQC, const CHANNEL_MAPPING* const cm, const INT avgTotalBits) { /* check bitreservoir fill level */ if (hQC->bitResTot < 0) { return AAC_ENC_BITRES_TOO_LOW; } else if (hQC->bitResTot > hQC->bitResTotMax) { return AAC_ENC_BITRES_TOO_HIGH; } else { INT i; INT totalBits = 0, totalBits_max = 0; const int totalBitreservoir = fMin(hQC->bitResTot, (hQC->maxBitsPerFrame - avgTotalBits)); const int totalBitreservoirMax = fMin(hQC->bitResTotMax, (hQC->maxBitsPerFrame - avgTotalBits)); for (i = (cm->nElements - 1); i >= 0; i--) { if ((cm->elInfo[i].elType == ID_SCE) || (cm->elInfo[i].elType == ID_CPE) || (cm->elInfo[i].elType == ID_LFE)) { hQC->elementBits[i]->bitResLevelEl = fMultI(hQC->elementBits[i]->relativeBitsEl, totalBitreservoir); totalBits += hQC->elementBits[i]->bitResLevelEl; hQC->elementBits[i]->maxBitResBitsEl = fMultI(hQC->elementBits[i]->relativeBitsEl, totalBitreservoirMax); totalBits_max += hQC->elementBits[i]->maxBitResBitsEl; } } for (i = 0; i < cm->nElements; i++) { if ((cm->elInfo[i].elType == ID_SCE) || (cm->elInfo[i].elType == ID_CPE) || (cm->elInfo[i].elType == ID_LFE)) { int deltaBits = fMax(totalBitreservoir - totalBits, -hQC->elementBits[i]->bitResLevelEl); hQC->elementBits[i]->bitResLevelEl += deltaBits; totalBits += deltaBits; deltaBits = fMax(totalBitreservoirMax - totalBits_max, -hQC->elementBits[i]->maxBitResBitsEl); hQC->elementBits[i]->maxBitResBitsEl += deltaBits; totalBits_max += deltaBits; } } } return AAC_ENC_OK; } AAC_ENCODER_ERROR FDKaacEnc_QCMain(QC_STATE* RESTRICT hQC, PSY_OUT** psyOut, QC_OUT** qcOut, INT avgTotalBits, CHANNEL_MAPPING* cm, const AUDIO_OBJECT_TYPE aot, UINT syntaxFlags, SCHAR epConfig) { int i, c; AAC_ENCODER_ERROR ErrorStatus = AAC_ENC_OK; INT avgTotalDynBits = 0; /* maximal allowed dynamic bits for all frames */ INT totalAvailableBits = 0; INT nSubFrames = 1; const INT isCBRAdjustment = (isConstantBitrateMode(hQC->bitrateMode) || (hQC->bitResMode != AACENC_BR_MODE_FULL)) ? 1 : 0; /*-------------------------------------------- */ /* redistribute total bitreservoir to elements */ ErrorStatus = FDKaacEnc_BitResRedistribution( hQC, cm, (isCBRAdjustment == 0) ? hQC->maxBitsPerFrame : avgTotalBits); if (ErrorStatus != AAC_ENC_OK) { return ErrorStatus; } /*-------------------------------------------- */ /* fastenc needs one time threshold simulation, in case of multiple frames, one more guess has to be calculated */ /*-------------------------------------------- */ /* helper pointer */ QC_OUT_ELEMENT* qcElement[(1)][((8))]; /* work on a copy of qcChannel and qcElement */ for (i = 0; i < cm->nElements; i++) { ELEMENT_INFO elInfo = cm->elInfo[i]; if ((elInfo.elType == ID_SCE) || (elInfo.elType == ID_CPE) || (elInfo.elType == ID_LFE)) { /* for ( all sub frames ) ... */ for (c = 0; c < nSubFrames; c++) { { qcElement[c][i] = qcOut[c]->qcElement[i]; } } } } /*-------------------------------------------- */ /*-------------------------------------------- */ /* calc granted dynamic bits for sub frame and distribute it to each element */ ErrorStatus = FDKaacEnc_prepareBitDistribution( hQC, psyOut, qcOut, cm, qcElement, (isCBRAdjustment == 0) ? hQC->maxBitsPerFrame : avgTotalBits, &totalAvailableBits, &avgTotalDynBits); if (ErrorStatus != AAC_ENC_OK) { return ErrorStatus; } /* for ( all sub frames ) ... */ for (c = 0; c < nSubFrames; c++) { /* for CBR and VBR mode */ FDKaacEnc_AdjustThresholds(hQC->hAdjThr, qcElement[c], qcOut[c], psyOut[c]->psyOutElement, isCBRAdjustment, cm); } /* -end- sub frame counter */ /*-------------------------------------------- */ INT iterations[(1)][((8))]; INT chConstraintsFulfilled[(1)][((8))][(2)]; INT calculateQuant[(1)][((8))][(2)]; INT constraintsFulfilled[(1)][((8))]; /*-------------------------------------------- */ /* for ( all sub frames ) ... */ for (c = 0; c < nSubFrames; c++) { for (i = 0; i < cm->nElements; i++) { ELEMENT_INFO elInfo = cm->elInfo[i]; INT ch, nChannels = elInfo.nChannelsInEl; if ((elInfo.elType == ID_SCE) || (elInfo.elType == ID_CPE) || (elInfo.elType == ID_LFE)) { /* Turn thresholds into scalefactors, optimize bit consumption and * verify conformance */ FDKaacEnc_EstimateScaleFactors( psyOut[c]->psyOutElement[i]->psyOutChannel, qcElement[c][i]->qcOutChannel, hQC->invQuant, hQC->dZoneQuantEnable, cm->elInfo[i].nChannelsInEl); /*-------------------------------------------- */ constraintsFulfilled[c][i] = 1; iterations[c][i] = 0; for (ch = 0; ch < nChannels; ch++) { chConstraintsFulfilled[c][i][ch] = 1; calculateQuant[c][i][ch] = 1; } /*-------------------------------------------- */ } /* -end- if(ID_SCE || ID_CPE || ID_LFE) */ } /* -end- element loop */ qcOut[c]->usedDynBits = -1; } /* -end- sub frame counter */ INT quantizationDone = 0; INT sumDynBitsConsumedTotal = 0; INT decreaseBitConsumption = -1; /* no direction yet! */ /*-------------------------------------------- */ /* -start- Quantization loop ... */ /*-------------------------------------------- */ do /* until max allowed bits per frame and maxDynBits!=-1*/ { quantizationDone = 0; c = 0; /* get frame to process */ for (i = 0; i < cm->nElements; i++) { ELEMENT_INFO elInfo = cm->elInfo[i]; INT ch, nChannels = elInfo.nChannelsInEl; if ((elInfo.elType == ID_SCE) || (elInfo.elType == ID_CPE) || (elInfo.elType == ID_LFE)) { do /* until element bits < nChannels*MIN_BUFSIZE_PER_EFF_CHAN */ { do /* until spectral values < MAX_QUANT */ { /*-------------------------------------------- */ if (!constraintsFulfilled[c][i]) { if ((ErrorStatus = FDKaacEnc_reduceBitConsumption( &iterations[c][i], hQC->maxIterations, (decreaseBitConsumption) ? 1 : -1, chConstraintsFulfilled[c][i], calculateQuant[c][i], nChannels, psyOut[c]->psyOutElement[i], qcOut[c], qcElement[c][i], hQC->elementBits[i], aot, syntaxFlags, epConfig)) != AAC_ENC_OK) { return ErrorStatus; } } /*-------------------------------------------- */ /*-------------------------------------------- */ constraintsFulfilled[c][i] = 1; /*-------------------------------------------- */ /* quantize spectrum (per each channel) */ for (ch = 0; ch < nChannels; ch++) { /*-------------------------------------------- */ chConstraintsFulfilled[c][i][ch] = 1; /*-------------------------------------------- */ if (calculateQuant[c][i][ch]) { QC_OUT_CHANNEL* qcOutCh = qcElement[c][i]->qcOutChannel[ch]; PSY_OUT_CHANNEL* psyOutCh = psyOut[c]->psyOutElement[i]->psyOutChannel[ch]; calculateQuant[c][i][ch] = 0; /* calculate quantization only if necessary */ /*-------------------------------------------- */ FDKaacEnc_QuantizeSpectrum( psyOutCh->sfbCnt, psyOutCh->maxSfbPerGroup, psyOutCh->sfbPerGroup, psyOutCh->sfbOffsets, qcOutCh->mdctSpectrum, qcOutCh->globalGain, qcOutCh->scf, qcOutCh->quantSpec, hQC->dZoneQuantEnable); /*-------------------------------------------- */ if (FDKaacEnc_calcMaxValueInSfb( psyOutCh->sfbCnt, psyOutCh->maxSfbPerGroup, psyOutCh->sfbPerGroup, psyOutCh->sfbOffsets, qcOutCh->quantSpec, qcOutCh->maxValueInSfb) > MAX_QUANT) { chConstraintsFulfilled[c][i][ch] = 0; constraintsFulfilled[c][i] = 0; /* if quanizted value out of range; increase global gain! */ decreaseBitConsumption = 1; } /*-------------------------------------------- */ } /* if calculateQuant[c][i][ch] */ } /* channel loop */ /*-------------------------------------------- */ /* quantize spectrum (per each channel) */ /*-------------------------------------------- */ } while (!constraintsFulfilled[c][i]); /* does not regard bit consumption */ /*-------------------------------------------- */ /*-------------------------------------------- */ qcElement[c][i]->dynBitsUsed = 0; /* reset dynamic bits */ /* quantization valid in current channel! */ for (ch = 0; ch < nChannels; ch++) { QC_OUT_CHANNEL* qcOutCh = qcElement[c][i]->qcOutChannel[ch]; PSY_OUT_CHANNEL* psyOutCh = psyOut[c]->psyOutElement[i]->psyOutChannel[ch]; /* count dynamic bits */ INT chDynBits = FDKaacEnc_dynBitCount( hQC->hBitCounter, qcOutCh->quantSpec, qcOutCh->maxValueInSfb, qcOutCh->scf, psyOutCh->lastWindowSequence, psyOutCh->sfbCnt, psyOutCh->maxSfbPerGroup, psyOutCh->sfbPerGroup, psyOutCh->sfbOffsets, &qcOutCh->sectionData, psyOutCh->noiseNrg, psyOutCh->isBook, psyOutCh->isScale, syntaxFlags); /* sum up dynamic channel bits */ qcElement[c][i]->dynBitsUsed += chDynBits; } /* save dynBitsUsed for correction of bits2pe relation */ if (hQC->hAdjThr->adjThrStateElem[i]->dynBitsLast == -1) { hQC->hAdjThr->adjThrStateElem[i]->dynBitsLast = qcElement[c][i]->dynBitsUsed; } /* hold total bit consumption in present element below maximum allowed */ if (qcElement[c][i]->dynBitsUsed > ((nChannels * MIN_BUFSIZE_PER_EFF_CHAN) - qcElement[c][i]->staticBitsUsed - qcElement[c][i]->extBitsUsed)) { constraintsFulfilled[c][i] = 0; } } while (!constraintsFulfilled[c][i]); } /* -end- if(ID_SCE || ID_CPE || ID_LFE) */ } /* -end- element loop */ /* update dynBits of current subFrame */ FDKaacEnc_updateUsedDynBits(&qcOut[c]->usedDynBits, qcElement[c], cm); /* get total consumed bits, dyn bits in all sub frames have to be valid */ sumDynBitsConsumedTotal = FDKaacEnc_getTotalConsumedDynBits(qcOut, nSubFrames); if (sumDynBitsConsumedTotal == -1) { quantizationDone = 0; /* bit consumption not valid in all sub frames */ } else { int sumBitsConsumedTotal = FDKaacEnc_getTotalConsumedBits( qcOut, qcElement, cm, hQC->globHdrBits, nSubFrames); /* in all frames are valid dynamic bits */ if (((sumBitsConsumedTotal < totalAvailableBits) || sumDynBitsConsumedTotal == 0) && (decreaseBitConsumption == 1) && checkMinFrameBitsDemand(qcOut, hQC->minBitsPerFrame, nSubFrames) /*()*/) { quantizationDone = 1; /* exit bit adjustment */ } if (sumBitsConsumedTotal > totalAvailableBits && (decreaseBitConsumption == 0)) { quantizationDone = 0; /* reset! */ } } /*-------------------------------------------- */ int emergencyIterations = 1; int dynBitsOvershoot = 0; for (c = 0; c < nSubFrames; c++) { for (i = 0; i < cm->nElements; i++) { ELEMENT_INFO elInfo = cm->elInfo[i]; if ((elInfo.elType == ID_SCE) || (elInfo.elType == ID_CPE) || (elInfo.elType == ID_LFE)) { /* iteration limitation */ emergencyIterations &= ((iterations[c][i] < hQC->maxIterations) ? 0 : 1); } } /* detection if used dyn bits exceeds the maximal allowed criterion */ dynBitsOvershoot |= ((qcOut[c]->usedDynBits > qcOut[c]->maxDynBits) ? 1 : 0); } if (quantizationDone == 0 || dynBitsOvershoot) { int sumBitsConsumedTotal = FDKaacEnc_getTotalConsumedBits( qcOut, qcElement, cm, hQC->globHdrBits, nSubFrames); if ((sumDynBitsConsumedTotal >= avgTotalDynBits) || (sumDynBitsConsumedTotal == 0)) { quantizationDone = 1; } if (emergencyIterations && (sumBitsConsumedTotal < totalAvailableBits)) { quantizationDone = 1; } if ((sumBitsConsumedTotal > totalAvailableBits) || !checkMinFrameBitsDemand(qcOut, hQC->minBitsPerFrame, nSubFrames)) { quantizationDone = 0; } if ((sumBitsConsumedTotal < totalAvailableBits) && checkMinFrameBitsDemand(qcOut, hQC->minBitsPerFrame, nSubFrames)) { decreaseBitConsumption = 0; } else { decreaseBitConsumption = 1; } if (dynBitsOvershoot) { quantizationDone = 0; decreaseBitConsumption = 1; } /* reset constraints fullfilled flags */ FDKmemclear(constraintsFulfilled, sizeof(constraintsFulfilled)); FDKmemclear(chConstraintsFulfilled, sizeof(chConstraintsFulfilled)); } /* quantizationDone */ } while (!quantizationDone); /*-------------------------------------------- */ /* ... -end- Quantization loop */ /*-------------------------------------------- */ /*-------------------------------------------- */ /*-------------------------------------------- */ return AAC_ENC_OK; } static AAC_ENCODER_ERROR FDKaacEnc_reduceBitConsumption( int* iterations, const int maxIterations, int gainAdjustment, int* chConstraintsFulfilled, int* calculateQuant, int nChannels, PSY_OUT_ELEMENT* psyOutElement, QC_OUT* qcOut, QC_OUT_ELEMENT* qcOutElement, ELEMENT_BITS* elBits, AUDIO_OBJECT_TYPE aot, UINT syntaxFlags, SCHAR epConfig) { int ch; /** SOLVING PROBLEM **/ if ((*iterations) < maxIterations) { /* increase gain (+ next iteration) */ for (ch = 0; ch < nChannels; ch++) { if (!chConstraintsFulfilled[ch]) { qcOutElement->qcOutChannel[ch]->globalGain += gainAdjustment; calculateQuant[ch] = 1; /* global gain has changed, recalculate quantization in next iteration! */ } } } else if ((*iterations) == maxIterations) { if (qcOutElement->dynBitsUsed == 0) { return AAC_ENC_QUANT_ERROR; } else { /* crash recovery */ INT bitsToSave = 0; if ((bitsToSave = fixMax( (qcOutElement->dynBitsUsed + 8) - (elBits->bitResLevelEl + qcOutElement->grantedDynBits), (qcOutElement->dynBitsUsed + qcOutElement->staticBitsUsed + 8) - (elBits->maxBitsEl))) > 0) { FDKaacEnc_crashRecovery(nChannels, psyOutElement, qcOut, qcOutElement, bitsToSave, aot, syntaxFlags, epConfig); } else { for (ch = 0; ch < nChannels; ch++) { qcOutElement->qcOutChannel[ch]->globalGain += 1; } } for (ch = 0; ch < nChannels; ch++) { calculateQuant[ch] = 1; } } } else { /* (*iterations) > maxIterations */ return AAC_ENC_QUANT_ERROR; } (*iterations)++; return AAC_ENC_OK; } AAC_ENCODER_ERROR FDKaacEnc_updateFillBits(CHANNEL_MAPPING* cm, QC_STATE* qcKernel, ELEMENT_BITS* RESTRICT elBits[((8))], QC_OUT** qcOut) { switch (qcKernel->bitrateMode) { case QCDATA_BR_MODE_SFR: break; case QCDATA_BR_MODE_FF: break; case QCDATA_BR_MODE_VBR_1: case QCDATA_BR_MODE_VBR_2: case QCDATA_BR_MODE_VBR_3: case QCDATA_BR_MODE_VBR_4: case QCDATA_BR_MODE_VBR_5: qcOut[0]->totFillBits = (qcOut[0]->grantedDynBits - qcOut[0]->usedDynBits) & 7; /* precalculate alignment bits */ qcOut[0]->totalBits = qcOut[0]->staticBits + qcOut[0]->usedDynBits + qcOut[0]->totFillBits + qcOut[0]->elementExtBits + qcOut[0]->globalExtBits; qcOut[0]->totFillBits += (fixMax(0, qcKernel->minBitsPerFrame - qcOut[0]->totalBits) + 7) & ~7; break; case QCDATA_BR_MODE_CBR: case QCDATA_BR_MODE_INVALID: default: INT bitResSpace = qcKernel->bitResTotMax - qcKernel->bitResTot; /* processing fill-bits */ INT deltaBitRes = qcOut[0]->grantedDynBits - qcOut[0]->usedDynBits; qcOut[0]->totFillBits = fixMax( (deltaBitRes & 7), (deltaBitRes - (fixMax(0, bitResSpace - 7) & ~7))); qcOut[0]->totalBits = qcOut[0]->staticBits + qcOut[0]->usedDynBits + qcOut[0]->totFillBits + qcOut[0]->elementExtBits + qcOut[0]->globalExtBits; qcOut[0]->totFillBits += (fixMax(0, qcKernel->minBitsPerFrame - qcOut[0]->totalBits) + 7) & ~7; break; } /* switch (qcKernel->bitrateMode) */ return AAC_ENC_OK; } /********************************************************************************* functionname: FDKaacEnc_calcMaxValueInSfb description: return: **********************************************************************************/ static INT FDKaacEnc_calcMaxValueInSfb(INT sfbCnt, INT maxSfbPerGroup, INT sfbPerGroup, INT* RESTRICT sfbOffset, SHORT* RESTRICT quantSpectrum, UINT* RESTRICT maxValue) { INT sfbOffs, sfb; INT maxValueAll = 0; for (sfbOffs = 0; sfbOffs < sfbCnt; sfbOffs += sfbPerGroup) for (sfb = 0; sfb < maxSfbPerGroup; sfb++) { INT line; INT maxThisSfb = 0; for (line = sfbOffset[sfbOffs + sfb]; line < sfbOffset[sfbOffs + sfb + 1]; line++) { INT tmp = fixp_abs(quantSpectrum[line]); maxThisSfb = fixMax(tmp, maxThisSfb); } maxValue[sfbOffs + sfb] = maxThisSfb; maxValueAll = fixMax(maxThisSfb, maxValueAll); } return maxValueAll; } /********************************************************************************* functionname: FDKaacEnc_updateBitres description: return: **********************************************************************************/ void FDKaacEnc_updateBitres(CHANNEL_MAPPING* cm, QC_STATE* qcKernel, QC_OUT** qcOut) { switch (qcKernel->bitrateMode) { case QCDATA_BR_MODE_VBR_1: case QCDATA_BR_MODE_VBR_2: case QCDATA_BR_MODE_VBR_3: case QCDATA_BR_MODE_VBR_4: case QCDATA_BR_MODE_VBR_5: /* variable bitrate */ qcKernel->bitResTot = fMin(qcKernel->maxBitsPerFrame, qcKernel->bitResTotMax); break; case QCDATA_BR_MODE_CBR: case QCDATA_BR_MODE_SFR: case QCDATA_BR_MODE_INVALID: default: int c = 0; /* constant bitrate */ { qcKernel->bitResTot += qcOut[c]->grantedDynBits - (qcOut[c]->usedDynBits + qcOut[c]->totFillBits + qcOut[c]->alignBits); } break; } } /********************************************************************************* functionname: FDKaacEnc_FinalizeBitConsumption description: return: **********************************************************************************/ AAC_ENCODER_ERROR FDKaacEnc_FinalizeBitConsumption( CHANNEL_MAPPING* cm, QC_STATE* qcKernel, QC_OUT* qcOut, QC_OUT_ELEMENT** qcElement, HANDLE_TRANSPORTENC hTpEnc, AUDIO_OBJECT_TYPE aot, UINT syntaxFlags, SCHAR epConfig) { QC_OUT_EXTENSION fillExtPayload; INT totFillBits, alignBits; /* Get total consumed bits in AU */ qcOut->totalBits = qcOut->staticBits + qcOut->usedDynBits + qcOut->totFillBits + qcOut->elementExtBits + qcOut->globalExtBits; if (qcKernel->bitrateMode == QCDATA_BR_MODE_CBR) { /* Now we can get the exact transport bit amount, and hopefully it is equal * to the estimated value */ INT exactTpBits = transportEnc_GetStaticBits(hTpEnc, qcOut->totalBits); if (exactTpBits != qcKernel->globHdrBits) { INT diffFillBits = 0; /* How many bits can be take by bitreservoir */ const INT bitresSpace = qcKernel->bitResTotMax - (qcKernel->bitResTot + (qcOut->grantedDynBits - (qcOut->usedDynBits + qcOut->totFillBits))); /* Number of bits which can be moved to bitreservoir. */ const INT bitsToBitres = qcKernel->globHdrBits - exactTpBits; FDK_ASSERT(bitsToBitres >= 0); /* is always positive */ /* If bitreservoir can not take all bits, move ramaining bits to fillbits */ diffFillBits = fMax(0, bitsToBitres - bitresSpace); /* Assure previous alignment */ diffFillBits = (diffFillBits + 7) & ~7; /* Move as many bits as possible to bitreservoir */ qcKernel->bitResTot += (bitsToBitres - diffFillBits); /* Write remaing bits as fill bits */ qcOut->totFillBits += diffFillBits; qcOut->totalBits += diffFillBits; qcOut->grantedDynBits += diffFillBits; /* Get new header bits */ qcKernel->globHdrBits = transportEnc_GetStaticBits(hTpEnc, qcOut->totalBits); if (qcKernel->globHdrBits != exactTpBits) { /* In previous step, fill bits and corresponding total bits were changed when bitreservoir was completely filled. Now we can take the too much taken bits caused by header overhead from bitreservoir. */ qcKernel->bitResTot -= (qcKernel->globHdrBits - exactTpBits); } } } /* MODE_CBR */ /* Update exact number of consumed header bits. */ qcKernel->globHdrBits = transportEnc_GetStaticBits(hTpEnc, qcOut->totalBits); /* Save total fill bits and distribut to alignment and fill bits */ totFillBits = qcOut->totFillBits; /* fake a fill extension payload */ FDKmemclear(&fillExtPayload, sizeof(QC_OUT_EXTENSION)); fillExtPayload.type = EXT_FILL_DATA; fillExtPayload.nPayloadBits = totFillBits; /* ask bitstream encoder how many of that bits can be written in a fill * extension data entity */ qcOut->totFillBits = FDKaacEnc_writeExtensionData(NULL, &fillExtPayload, 0, 0, syntaxFlags, aot, epConfig); /* now distribute extra fillbits and alignbits */ alignBits = 7 - (qcOut->staticBits + qcOut->usedDynBits + qcOut->elementExtBits + qcOut->totFillBits + qcOut->globalExtBits - 1) % 8; /* Maybe we could remove this */ if (((alignBits + qcOut->totFillBits - totFillBits) == 8) && (qcOut->totFillBits > 8)) qcOut->totFillBits -= 8; qcOut->totalBits = qcOut->staticBits + qcOut->usedDynBits + qcOut->totFillBits + alignBits + qcOut->elementExtBits + qcOut->globalExtBits; if ((qcOut->totalBits > qcKernel->maxBitsPerFrame) || (qcOut->totalBits < qcKernel->minBitsPerFrame)) { return AAC_ENC_QUANT_ERROR; } qcOut->alignBits = alignBits; return AAC_ENC_OK; } /********************************************************************************* functionname: FDKaacEnc_crashRecovery description: fulfills constraints by means of brute force... => bits are saved by cancelling out spectral lines!! (beginning at the highest frequencies) return: errorcode **********************************************************************************/ static void FDKaacEnc_crashRecovery(INT nChannels, PSY_OUT_ELEMENT* psyOutElement, QC_OUT* qcOut, QC_OUT_ELEMENT* qcElement, INT bitsToSave, AUDIO_OBJECT_TYPE aot, UINT syntaxFlags, SCHAR epConfig) { INT ch; INT savedBits = 0; INT sfb, sfbGrp; INT bitsPerScf[(2)][MAX_GROUPED_SFB]; INT sectionToScf[(2)][MAX_GROUPED_SFB]; INT* sfbOffset; INT sect, statBitsNew; QC_OUT_CHANNEL** qcChannel = qcElement->qcOutChannel; PSY_OUT_CHANNEL** psyChannel = psyOutElement->psyOutChannel; /* create a table which converts frq-bins to bit-demand... [bitsPerScf] */ /* ...and another one which holds the corresponding sections [sectionToScf] */ for (ch = 0; ch < nChannels; ch++) { sfbOffset = psyChannel[ch]->sfbOffsets; for (sect = 0; sect < qcChannel[ch]->sectionData.noOfSections; sect++) { INT codeBook = qcChannel[ch]->sectionData.huffsection[sect].codeBook; for (sfb = qcChannel[ch]->sectionData.huffsection[sect].sfbStart; sfb < qcChannel[ch]->sectionData.huffsection[sect].sfbStart + qcChannel[ch]->sectionData.huffsection[sect].sfbCnt; sfb++) { bitsPerScf[ch][sfb] = 0; if ((codeBook != CODE_BOOK_PNS_NO) /*&& (sfb < (qcChannel[ch]->sectionData.noOfGroups*qcChannel[ch]->sectionData.maxSfbPerGroup))*/) { INT sfbStartLine = sfbOffset[sfb]; INT noOfLines = sfbOffset[sfb + 1] - sfbStartLine; bitsPerScf[ch][sfb] = FDKaacEnc_countValues( &(qcChannel[ch]->quantSpec[sfbStartLine]), noOfLines, codeBook); } sectionToScf[ch][sfb] = sect; } } } /* LOWER [maxSfb] IN BOTH CHANNELS!! */ /* Attention: in case of stereo: maxSfbL == maxSfbR, GroupingL == GroupingR ; */ for (sfb = qcChannel[0]->sectionData.maxSfbPerGroup - 1; sfb >= 0; sfb--) { for (sfbGrp = 0; sfbGrp < psyChannel[0]->sfbCnt; sfbGrp += psyChannel[0]->sfbPerGroup) { for (ch = 0; ch < nChannels; ch++) { sect = sectionToScf[ch][sfbGrp + sfb]; qcChannel[ch]->sectionData.huffsection[sect].sfbCnt--; savedBits += bitsPerScf[ch][sfbGrp + sfb]; if (qcChannel[ch]->sectionData.huffsection[sect].sfbCnt == 0) { savedBits += (psyChannel[ch]->lastWindowSequence != SHORT_WINDOW) ? FDKaacEnc_sideInfoTabLong[0] : FDKaacEnc_sideInfoTabShort[0]; } } } /* ...have enough bits been saved? */ if (savedBits >= bitsToSave) break; } /* sfb loop */ /* if not enough bits saved, clean whole spectrum and remove side info overhead */ if (sfb == -1) { sfb = 0; } for (ch = 0; ch < nChannels; ch++) { qcChannel[ch]->sectionData.maxSfbPerGroup = sfb; psyChannel[ch]->maxSfbPerGroup = sfb; /* when no spectrum is coded save tools info in bitstream */ if (sfb == 0) { FDKmemclear(&psyChannel[ch]->tnsInfo, sizeof(TNS_INFO)); FDKmemclear(&psyOutElement->toolsInfo, sizeof(TOOLSINFO)); } } /* dynamic bits will be updated in iteration loop */ { /* if stop sfb has changed save bits in side info, e.g. MS or TNS coding */ ELEMENT_INFO elInfo; FDKmemclear(&elInfo, sizeof(ELEMENT_INFO)); elInfo.nChannelsInEl = nChannels; elInfo.elType = (nChannels == 2) ? ID_CPE : ID_SCE; FDKaacEnc_ChannelElementWrite(NULL, &elInfo, NULL, psyOutElement, psyChannel, syntaxFlags, aot, epConfig, &statBitsNew, 0); } savedBits = qcElement->staticBitsUsed - statBitsNew; /* update static and dynamic bits */ qcElement->staticBitsUsed -= savedBits; qcElement->grantedDynBits += savedBits; qcOut->staticBits -= savedBits; qcOut->grantedDynBits += savedBits; qcOut->maxDynBits += savedBits; } void FDKaacEnc_QCClose(QC_STATE** phQCstate, QC_OUT** phQC) { int n, i; if (phQC != NULL) { for (n = 0; n < (1); n++) { if (phQC[n] != NULL) { QC_OUT* hQC = phQC[n]; for (i = 0; i < (8); i++) { } for (i = 0; i < ((8)); i++) { if (hQC->qcElement[i]) FreeRam_aacEnc_QCelement(&hQC->qcElement[i]); } FreeRam_aacEnc_QCout(&phQC[n]); } } } if (phQCstate != NULL) { if (*phQCstate != NULL) { QC_STATE* hQCstate = *phQCstate; if (hQCstate->hAdjThr != NULL) FDKaacEnc_AdjThrClose(&hQCstate->hAdjThr); if (hQCstate->hBitCounter != NULL) FDKaacEnc_BCClose(&hQCstate->hBitCounter); for (i = 0; i < ((8)); i++) { if (hQCstate->elementBits[i] != NULL) { FreeRam_aacEnc_ElementBits(&hQCstate->elementBits[i]); } } FreeRam_aacEnc_QCstate(phQCstate); } } } fdk-aac-2.0.2/libAACenc/src/qc_main.h000066400000000000000000000160501404225310700171110ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Werner Description: Quantizing & coding *******************************************************************************/ #ifndef QC_MAIN_H #define QC_MAIN_H #include "aacenc.h" #include "qc_data.h" #include "interface.h" #include "psy_main.h" #include "tpenc_lib.h" /* Quantizing & coding stage */ AAC_ENCODER_ERROR FDKaacEnc_QCOutNew(QC_OUT **phQC, const INT nElements, const INT nChannels, const INT nSubFrames, UCHAR *dynamic_RAM); AAC_ENCODER_ERROR FDKaacEnc_QCOutInit(QC_OUT *phQC[(1)], const INT nSubFrames, const CHANNEL_MAPPING *cm); AAC_ENCODER_ERROR FDKaacEnc_QCNew(QC_STATE **phQC, INT nElements, UCHAR *dynamic_RAM); AAC_ENCODER_ERROR FDKaacEnc_QCInit(QC_STATE *hQC, struct QC_INIT *init, const ULONG initFlags); AAC_ENCODER_ERROR FDKaacEnc_QCMainPrepare( ELEMENT_INFO *elInfo, ATS_ELEMENT *RESTRICT adjThrStateElement, PSY_OUT_ELEMENT *RESTRICT psyOutElement, QC_OUT_ELEMENT *RESTRICT qcOutElement, /* returns error code */ AUDIO_OBJECT_TYPE aot, UINT syntaxFlags, SCHAR epConfig); AAC_ENCODER_ERROR FDKaacEnc_QCMain(QC_STATE *RESTRICT hQC, PSY_OUT **psyOut, QC_OUT **qcOut, INT avgTotalBits, CHANNEL_MAPPING *cm, AUDIO_OBJECT_TYPE aot, UINT syntaxFlags, SCHAR epConfig); AAC_ENCODER_ERROR FDKaacEnc_updateFillBits(CHANNEL_MAPPING *cm, QC_STATE *qcKernel, ELEMENT_BITS *RESTRICT elBits[((8))], QC_OUT **qcOut); void FDKaacEnc_updateBitres(CHANNEL_MAPPING *cm, QC_STATE *qcKernel, QC_OUT **qcOut); AAC_ENCODER_ERROR FDKaacEnc_FinalizeBitConsumption( CHANNEL_MAPPING *cm, QC_STATE *hQC, QC_OUT *qcOut, QC_OUT_ELEMENT **qcElement, HANDLE_TRANSPORTENC hTpEnc, AUDIO_OBJECT_TYPE aot, UINT syntaxFlags, SCHAR epConfig); AAC_ENCODER_ERROR FDKaacEnc_AdjustBitrate(QC_STATE *RESTRICT hQC, CHANNEL_MAPPING *RESTRICT cm, INT *avgTotalBits, INT bitRate, INT sampleRate, INT granuleLength); void FDKaacEnc_QCClose(QC_STATE **phQCstate, QC_OUT **phQC); #endif /* QC_MAIN_H */ fdk-aac-2.0.2/libAACenc/src/quantize.cpp000066400000000000000000000352651404225310700177060ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M.Werner Description: Quantization *******************************************************************************/ #include "quantize.h" #include "aacEnc_rom.h" /***************************************************************************** functionname: FDKaacEnc_quantizeLines description: quantizes spectrum lines returns: input: global gain, number of lines to process, spectral data output: quantized spectrum *****************************************************************************/ static void FDKaacEnc_quantizeLines(INT gain, INT noOfLines, const FIXP_DBL *mdctSpectrum, SHORT *quaSpectrum, INT dZoneQuantEnable) { int line; FIXP_DBL k = FL2FXCONST_DBL(0.0f); FIXP_QTD quantizer = FDKaacEnc_quantTableQ[(-gain) & 3]; INT quantizershift = ((-gain) >> 2) + 1; const INT kShift = 16; if (dZoneQuantEnable) k = FL2FXCONST_DBL(0.23f) >> kShift; else k = FL2FXCONST_DBL(-0.0946f + 0.5f) >> kShift; for (line = 0; line < noOfLines; line++) { FIXP_DBL accu = fMultDiv2(mdctSpectrum[line], quantizer); if (accu < FL2FXCONST_DBL(0.0f)) { accu = -accu; /* normalize */ INT accuShift = CntLeadingZeros(accu) - 1; /* CountLeadingBits() is not necessary here since test value is always > 0 */ accu <<= accuShift; INT tabIndex = (INT)(accu >> (DFRACT_BITS - 2 - MANT_DIGITS)) & (~MANT_SIZE); INT totalShift = quantizershift - accuShift + 1; accu = fMultDiv2(FDKaacEnc_mTab_3_4[tabIndex], FDKaacEnc_quantTableE[totalShift & 3]); totalShift = (16 - 4) - (3 * (totalShift >> 2)); FDK_ASSERT(totalShift >= 0); /* MAX_QUANT_VIOLATION */ accu >>= fixMin(totalShift, DFRACT_BITS - 1); quaSpectrum[line] = (SHORT)(-((LONG)(k + accu) >> (DFRACT_BITS - 1 - 16))); } else if (accu > FL2FXCONST_DBL(0.0f)) { /* normalize */ INT accuShift = CntLeadingZeros(accu) - 1; /* CountLeadingBits() is not necessary here since test value is always > 0 */ accu <<= accuShift; INT tabIndex = (INT)(accu >> (DFRACT_BITS - 2 - MANT_DIGITS)) & (~MANT_SIZE); INT totalShift = quantizershift - accuShift + 1; accu = fMultDiv2(FDKaacEnc_mTab_3_4[tabIndex], FDKaacEnc_quantTableE[totalShift & 3]); totalShift = (16 - 4) - (3 * (totalShift >> 2)); FDK_ASSERT(totalShift >= 0); /* MAX_QUANT_VIOLATION */ accu >>= fixMin(totalShift, DFRACT_BITS - 1); quaSpectrum[line] = (SHORT)((LONG)(k + accu) >> (DFRACT_BITS - 1 - 16)); } else { quaSpectrum[line] = 0; } } } /***************************************************************************** functionname:iFDKaacEnc_quantizeLines description: iquantizes spectrum lines mdctSpectrum = iquaSpectrum^4/3 *2^(0.25*gain) input: global gain, number of lines to process,quantized spectrum output: spectral data *****************************************************************************/ static void FDKaacEnc_invQuantizeLines(INT gain, INT noOfLines, SHORT *quantSpectrum, FIXP_DBL *mdctSpectrum) { INT iquantizermod; INT iquantizershift; INT line; iquantizermod = gain & 3; iquantizershift = gain >> 2; for (line = 0; line < noOfLines; line++) { if (quantSpectrum[line] < 0) { FIXP_DBL accu; INT ex, specExp, tabIndex; FIXP_DBL s, t; accu = (FIXP_DBL)-quantSpectrum[line]; ex = CountLeadingBits(accu); accu <<= ex; specExp = (DFRACT_BITS - 1) - ex; FDK_ASSERT(specExp < 14); /* this fails if abs(value) > 8191 */ tabIndex = (INT)(accu >> (DFRACT_BITS - 2 - MANT_DIGITS)) & (~MANT_SIZE); /* calculate "mantissa" ^4/3 */ s = FDKaacEnc_mTab_4_3Elc[tabIndex]; /* get approperiate exponent multiplier for specExp^3/4 combined with * scfMod */ t = FDKaacEnc_specExpMantTableCombElc[iquantizermod][specExp]; /* multiply "mantissa" ^4/3 with exponent multiplier */ accu = fMult(s, t); /* get approperiate exponent shifter */ specExp = FDKaacEnc_specExpTableComb[iquantizermod][specExp] - 1; /* -1 to avoid overflows in accu */ if ((-iquantizershift - specExp) < 0) accu <<= -(-iquantizershift - specExp); else accu >>= -iquantizershift - specExp; mdctSpectrum[line] = -accu; } else if (quantSpectrum[line] > 0) { FIXP_DBL accu; INT ex, specExp, tabIndex; FIXP_DBL s, t; accu = (FIXP_DBL)(INT)quantSpectrum[line]; ex = CountLeadingBits(accu); accu <<= ex; specExp = (DFRACT_BITS - 1) - ex; FDK_ASSERT(specExp < 14); /* this fails if abs(value) > 8191 */ tabIndex = (INT)(accu >> (DFRACT_BITS - 2 - MANT_DIGITS)) & (~MANT_SIZE); /* calculate "mantissa" ^4/3 */ s = FDKaacEnc_mTab_4_3Elc[tabIndex]; /* get approperiate exponent multiplier for specExp^3/4 combined with * scfMod */ t = FDKaacEnc_specExpMantTableCombElc[iquantizermod][specExp]; /* multiply "mantissa" ^4/3 with exponent multiplier */ accu = fMult(s, t); /* get approperiate exponent shifter */ specExp = FDKaacEnc_specExpTableComb[iquantizermod][specExp] - 1; /* -1 to avoid overflows in accu */ if ((-iquantizershift - specExp) < 0) accu <<= -(-iquantizershift - specExp); else accu >>= -iquantizershift - specExp; mdctSpectrum[line] = accu; } else { mdctSpectrum[line] = FL2FXCONST_DBL(0.0f); } } } /***************************************************************************** functionname: FDKaacEnc_QuantizeSpectrum description: quantizes the entire spectrum returns: input: number of scalefactor bands to be quantized, ... output: quantized spectrum *****************************************************************************/ void FDKaacEnc_QuantizeSpectrum(INT sfbCnt, INT maxSfbPerGroup, INT sfbPerGroup, const INT *sfbOffset, const FIXP_DBL *mdctSpectrum, INT globalGain, const INT *scalefactors, SHORT *quantizedSpectrum, INT dZoneQuantEnable) { INT sfbOffs, sfb; /* in FDKaacEnc_quantizeLines quaSpectrum is calculated with: spec^(3/4) * 2^(-3/16*QSS) * 2^(3/4*scale) + k simplify scaling calculation and reduce QSS before: spec^(3/4) * 2^(-3/16*(QSS - 4*scale)) */ for (sfbOffs = 0; sfbOffs < sfbCnt; sfbOffs += sfbPerGroup) for (sfb = 0; sfb < maxSfbPerGroup; sfb++) { INT scalefactor = scalefactors[sfbOffs + sfb]; FDKaacEnc_quantizeLines( globalGain - scalefactor, /* QSS */ sfbOffset[sfbOffs + sfb + 1] - sfbOffset[sfbOffs + sfb], mdctSpectrum + sfbOffset[sfbOffs + sfb], quantizedSpectrum + sfbOffset[sfbOffs + sfb], dZoneQuantEnable); } } /***************************************************************************** functionname: FDKaacEnc_calcSfbDist description: calculates distortion of quantized values returns: distortion input: gain, number of lines to process, spectral data output: *****************************************************************************/ FIXP_DBL FDKaacEnc_calcSfbDist(const FIXP_DBL *mdctSpectrum, SHORT *quantSpectrum, INT noOfLines, INT gain, INT dZoneQuantEnable) { INT i, scale; FIXP_DBL xfsf; FIXP_DBL diff; FIXP_DBL invQuantSpec; xfsf = FL2FXCONST_DBL(0.0f); for (i = 0; i < noOfLines; i++) { /* quantization */ FDKaacEnc_quantizeLines(gain, 1, &mdctSpectrum[i], &quantSpectrum[i], dZoneQuantEnable); if (fAbs(quantSpectrum[i]) > MAX_QUANT) { return FL2FXCONST_DBL(0.0f); } /* inverse quantization */ FDKaacEnc_invQuantizeLines(gain, 1, &quantSpectrum[i], &invQuantSpec); /* dist */ diff = fixp_abs(fixp_abs(invQuantSpec) - fixp_abs(mdctSpectrum[i] >> 1)); scale = CountLeadingBits(diff); diff = scaleValue(diff, scale); diff = fPow2(diff); scale = fixMin(2 * (scale - 1), DFRACT_BITS - 1); diff = scaleValue(diff, -scale); xfsf = xfsf + diff; } xfsf = CalcLdData(xfsf); return xfsf; } /***************************************************************************** functionname: FDKaacEnc_calcSfbQuantEnergyAndDist description: calculates energy and distortion of quantized values returns: input: gain, number of lines to process, quantized spectral data, spectral data output: energy, distortion *****************************************************************************/ void FDKaacEnc_calcSfbQuantEnergyAndDist(FIXP_DBL *mdctSpectrum, SHORT *quantSpectrum, INT noOfLines, INT gain, FIXP_DBL *en, FIXP_DBL *dist) { INT i, scale; FIXP_DBL invQuantSpec; FIXP_DBL diff; FIXP_DBL energy = FL2FXCONST_DBL(0.0f); FIXP_DBL distortion = FL2FXCONST_DBL(0.0f); for (i = 0; i < noOfLines; i++) { if (fAbs(quantSpectrum[i]) > MAX_QUANT) { *en = FL2FXCONST_DBL(0.0f); *dist = FL2FXCONST_DBL(0.0f); return; } /* inverse quantization */ FDKaacEnc_invQuantizeLines(gain, 1, &quantSpectrum[i], &invQuantSpec); /* energy */ energy += fPow2(invQuantSpec); /* dist */ diff = fixp_abs(fixp_abs(invQuantSpec) - fixp_abs(mdctSpectrum[i] >> 1)); scale = CountLeadingBits(diff); diff = scaleValue(diff, scale); diff = fPow2(diff); scale = fixMin(2 * (scale - 1), DFRACT_BITS - 1); diff = scaleValue(diff, -scale); distortion += diff; } *en = CalcLdData(energy) + FL2FXCONST_DBL(0.03125f); *dist = CalcLdData(distortion); } fdk-aac-2.0.2/libAACenc/src/quantize.h000066400000000000000000000132141404225310700173410ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M.Werner Description: Quantization *******************************************************************************/ #ifndef QUANTIZE_H #define QUANTIZE_H #include "common_fix.h" /* quantizing */ #define MAX_QUANT 8191 void FDKaacEnc_QuantizeSpectrum(INT sfbCnt, INT maxSfbPerGroup, INT sfbPerGroup, const INT *sfbOffset, const FIXP_DBL *mdctSpectrum, INT globalGain, const INT *scalefactors, SHORT *quantizedSpectrum, INT dZoneQuantEnable); FIXP_DBL FDKaacEnc_calcSfbDist(const FIXP_DBL *mdctSpectrum, SHORT *quantSpectrum, INT noOfLines, INT gain, INT dZoneQuantEnable); void FDKaacEnc_calcSfbQuantEnergyAndDist(FIXP_DBL *mdctSpectrum, SHORT *quantSpectrum, INT noOfLines, INT gain, FIXP_DBL *en, FIXP_DBL *dist); #endif /* QUANTIZE_H */ fdk-aac-2.0.2/libAACenc/src/sf_estim.cpp000066400000000000000000001374631404225310700176620ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Werner Description: Scale factor estimation *******************************************************************************/ #include "sf_estim.h" #include "aacEnc_rom.h" #include "quantize.h" #include "bit_cnt.h" #ifdef __arm__ #endif #define UPCOUNT_LIMIT 1 #define AS_PE_FAC_SHIFT 7 #define DIST_FAC_SHIFT 3 #define AS_PE_FAC_FLOAT (float)(1 << AS_PE_FAC_SHIFT) static const INT MAX_SCF_DELTA = 60; static const FIXP_DBL PE_C1 = FL2FXCONST_DBL( 3.0f / AS_PE_FAC_FLOAT); /* (log(8.0)/log(2)) >> AS_PE_FAC_SHIFT */ static const FIXP_DBL PE_C2 = FL2FXCONST_DBL( 1.3219281f / AS_PE_FAC_FLOAT); /* (log(2.5)/log(2)) >> AS_PE_FAC_SHIFT */ static const FIXP_DBL PE_C3 = FL2FXCONST_DBL(0.5593573f); /* 1-C2/C1 */ /* Function; FDKaacEnc_FDKaacEnc_CalcFormFactorChannel Description: Calculates the formfactor sf: scale factor of the mdct spectrum sfbFormFactorLdData is scaled with the factor 1/(((2^sf)^0.5) * (2^FORM_FAC_SHIFT)) */ static void FDKaacEnc_FDKaacEnc_CalcFormFactorChannel( FIXP_DBL *RESTRICT sfbFormFactorLdData, PSY_OUT_CHANNEL *RESTRICT psyOutChan) { INT j, sfb, sfbGrp; FIXP_DBL formFactor; int tmp0 = psyOutChan->sfbCnt; int tmp1 = psyOutChan->maxSfbPerGroup; int step = psyOutChan->sfbPerGroup; for (sfbGrp = 0; sfbGrp < tmp0; sfbGrp += step) { for (sfb = 0; sfb < tmp1; sfb++) { formFactor = FL2FXCONST_DBL(0.0f); /* calc sum of sqrt(spec) */ for (j = psyOutChan->sfbOffsets[sfbGrp + sfb]; j < psyOutChan->sfbOffsets[sfbGrp + sfb + 1]; j++) { formFactor += sqrtFixp(fixp_abs(psyOutChan->mdctSpectrum[j])) >> FORM_FAC_SHIFT; } sfbFormFactorLdData[sfbGrp + sfb] = CalcLdData(formFactor); } /* set sfbFormFactor for sfbs with zero spec to zero. Just for debugging. */ for (; sfb < psyOutChan->sfbPerGroup; sfb++) { sfbFormFactorLdData[sfbGrp + sfb] = FL2FXCONST_DBL(-1.0f); } } } /* Function: FDKaacEnc_CalcFormFactor Description: Calls FDKaacEnc_FDKaacEnc_CalcFormFactorChannel() for each channel */ void FDKaacEnc_CalcFormFactor(QC_OUT_CHANNEL *qcOutChannel[(2)], PSY_OUT_CHANNEL *psyOutChannel[(2)], const INT nChannels) { INT j; for (j = 0; j < nChannels; j++) { FDKaacEnc_FDKaacEnc_CalcFormFactorChannel( qcOutChannel[j]->sfbFormFactorLdData, psyOutChannel[j]); } } /* Function: FDKaacEnc_calcSfbRelevantLines Description: Calculates sfbNRelevantLines sfbNRelevantLines is scaled with the factor 1/((2^FORM_FAC_SHIFT) * 2.0) */ static void FDKaacEnc_calcSfbRelevantLines( const FIXP_DBL *const sfbFormFactorLdData, const FIXP_DBL *const sfbEnergyLdData, const FIXP_DBL *const sfbThresholdLdData, const INT *const sfbOffsets, const INT sfbCnt, const INT sfbPerGroup, const INT maxSfbPerGroup, FIXP_DBL *sfbNRelevantLines) { INT sfbOffs, sfb; FIXP_DBL sfbWidthLdData; FIXP_DBL asPeFacLdData = FL2FXCONST_DBL(0.109375); /* AS_PE_FAC_SHIFT*ld64(2) */ FIXP_DBL accu; /* sfbNRelevantLines[i] = 2^( (sfbFormFactorLdData[i] - 0.25 * * (sfbEnergyLdData[i] - ld64(sfbWidth[i]/(2^7)) - AS_PE_FAC_SHIFT*ld64(2)) * * 64); */ FDKmemclear(sfbNRelevantLines, sfbCnt * sizeof(FIXP_DBL)); for (sfbOffs = 0; sfbOffs < sfbCnt; sfbOffs += sfbPerGroup) { for (sfb = 0; sfb < maxSfbPerGroup; sfb++) { /* calc sum of sqrt(spec) */ if ((FIXP_DBL)sfbEnergyLdData[sfbOffs + sfb] > (FIXP_DBL)sfbThresholdLdData[sfbOffs + sfb]) { INT sfbWidth = sfbOffsets[sfbOffs + sfb + 1] - sfbOffsets[sfbOffs + sfb]; /* avgFormFactorLdData = * sqrtFixp(sqrtFixp(sfbEnergyLdData[sfbOffs+sfb]/sfbWidth)); */ /* sfbNRelevantLines[sfbOffs+sfb] = sfbFormFactor[sfbOffs+sfb] / * avgFormFactorLdData; */ sfbWidthLdData = (FIXP_DBL)(sfbWidth << (DFRACT_BITS - 1 - AS_PE_FAC_SHIFT)); sfbWidthLdData = CalcLdData(sfbWidthLdData); accu = sfbEnergyLdData[sfbOffs + sfb] - sfbWidthLdData - asPeFacLdData; accu = sfbFormFactorLdData[sfbOffs + sfb] - (accu >> 2); sfbNRelevantLines[sfbOffs + sfb] = CalcInvLdData(accu) >> 1; } } } } /* Function: FDKaacEnc_countSingleScfBits Description: scfBitsFract is scaled by 1/(2^(2*AS_PE_FAC_SHIFT)) */ static FIXP_DBL FDKaacEnc_countSingleScfBits(INT scf, INT scfLeft, INT scfRight) { FIXP_DBL scfBitsFract; scfBitsFract = (FIXP_DBL)(FDKaacEnc_bitCountScalefactorDelta(scfLeft - scf) + FDKaacEnc_bitCountScalefactorDelta(scf - scfRight)); scfBitsFract = scfBitsFract << (DFRACT_BITS - 1 - (2 * AS_PE_FAC_SHIFT)); return scfBitsFract; /* output scaled by 1/(2^(2*AS_PE_FAC)) */ } /* Function: FDKaacEnc_calcSingleSpecPe specPe is scaled by 1/(2^(2*AS_PE_FAC_SHIFT)) */ static FIXP_DBL FDKaacEnc_calcSingleSpecPe(INT scf, FIXP_DBL sfbConstPePart, FIXP_DBL nLines) { FIXP_DBL specPe = FL2FXCONST_DBL(0.0f); FIXP_DBL ldRatio; FIXP_DBL scfFract; scfFract = (FIXP_DBL)(scf << (DFRACT_BITS - 1 - AS_PE_FAC_SHIFT)); ldRatio = sfbConstPePart - fMult(FL2FXCONST_DBL(0.375f), scfFract); if (ldRatio >= PE_C1) { specPe = fMult(FL2FXCONST_DBL(0.7f), fMult(nLines, ldRatio)); } else { specPe = fMult(FL2FXCONST_DBL(0.7f), fMult(nLines, (PE_C2 + fMult(PE_C3, ldRatio)))); } return specPe; /* output scaled by 1/(2^(2*AS_PE_FAC)) */ } /* Function: FDKaacEnc_countScfBitsDiff scfBitsDiff is scaled by 1/(2^(2*AS_PE_FAC_SHIFT)) */ static FIXP_DBL FDKaacEnc_countScfBitsDiff(INT *scfOld, INT *scfNew, INT sfbCnt, INT startSfb, INT stopSfb) { FIXP_DBL scfBitsFract; INT scfBitsDiff = 0; INT sfb = 0, sfbLast; INT sfbPrev, sfbNext; /* search for first relevant sfb */ sfbLast = startSfb; while ((sfbLast < stopSfb) && (scfOld[sfbLast] == FDK_INT_MIN)) sfbLast++; /* search for previous relevant sfb and count diff */ sfbPrev = startSfb - 1; while ((sfbPrev >= 0) && (scfOld[sfbPrev] == FDK_INT_MIN)) sfbPrev--; if (sfbPrev >= 0) scfBitsDiff += FDKaacEnc_bitCountScalefactorDelta(scfNew[sfbPrev] - scfNew[sfbLast]) - FDKaacEnc_bitCountScalefactorDelta(scfOld[sfbPrev] - scfOld[sfbLast]); /* now loop through all sfbs and count diffs of relevant sfbs */ for (sfb = sfbLast + 1; sfb < stopSfb; sfb++) { if (scfOld[sfb] != FDK_INT_MIN) { scfBitsDiff += FDKaacEnc_bitCountScalefactorDelta(scfNew[sfbLast] - scfNew[sfb]) - FDKaacEnc_bitCountScalefactorDelta(scfOld[sfbLast] - scfOld[sfb]); sfbLast = sfb; } } /* search for next relevant sfb and count diff */ sfbNext = stopSfb; while ((sfbNext < sfbCnt) && (scfOld[sfbNext] == FDK_INT_MIN)) sfbNext++; if (sfbNext < sfbCnt) scfBitsDiff += FDKaacEnc_bitCountScalefactorDelta(scfNew[sfbLast] - scfNew[sfbNext]) - FDKaacEnc_bitCountScalefactorDelta(scfOld[sfbLast] - scfOld[sfbNext]); scfBitsFract = (FIXP_DBL)(scfBitsDiff << (DFRACT_BITS - 1 - (2 * AS_PE_FAC_SHIFT))); return scfBitsFract; } /* Function: FDKaacEnc_calcSpecPeDiff specPeDiff is scaled by 1/(2^(2*AS_PE_FAC_SHIFT)) */ static FIXP_DBL FDKaacEnc_calcSpecPeDiff( PSY_OUT_CHANNEL *psyOutChan, QC_OUT_CHANNEL *qcOutChannel, INT *scfOld, INT *scfNew, FIXP_DBL *sfbConstPePart, FIXP_DBL *sfbFormFactorLdData, FIXP_DBL *sfbNRelevantLines, INT startSfb, INT stopSfb) { FIXP_DBL specPeDiff = FL2FXCONST_DBL(0.0f); FIXP_DBL scfFract = FL2FXCONST_DBL(0.0f); INT sfb; /* loop through all sfbs and count pe difference */ for (sfb = startSfb; sfb < stopSfb; sfb++) { if (scfOld[sfb] != FDK_INT_MIN) { FIXP_DBL ldRatioOld, ldRatioNew, pOld, pNew; /* sfbConstPePart[sfb] = (float)log(psyOutChan->sfbEnergy[sfb] * 6.75f / * sfbFormFactor[sfb]) * LOG2_1; */ /* 0.02152255861f = log(6.75)/log(2)/AS_PE_FAC_FLOAT; LOG2_1 is 1.0 for * log2 */ /* 0.09375f = log(64.0)/log(2.0)/64.0 = scale of sfbFormFactorLdData */ if (sfbConstPePart[sfb] == (FIXP_DBL)FDK_INT_MIN) sfbConstPePart[sfb] = ((psyOutChan->sfbEnergyLdData[sfb] - sfbFormFactorLdData[sfb] - FL2FXCONST_DBL(0.09375f)) >> 1) + FL2FXCONST_DBL(0.02152255861f); scfFract = (FIXP_DBL)(scfOld[sfb] << (DFRACT_BITS - 1 - AS_PE_FAC_SHIFT)); ldRatioOld = sfbConstPePart[sfb] - fMult(FL2FXCONST_DBL(0.375f), scfFract); scfFract = (FIXP_DBL)(scfNew[sfb] << (DFRACT_BITS - 1 - AS_PE_FAC_SHIFT)); ldRatioNew = sfbConstPePart[sfb] - fMult(FL2FXCONST_DBL(0.375f), scfFract); if (ldRatioOld >= PE_C1) pOld = ldRatioOld; else pOld = PE_C2 + fMult(PE_C3, ldRatioOld); if (ldRatioNew >= PE_C1) pNew = ldRatioNew; else pNew = PE_C2 + fMult(PE_C3, ldRatioNew); specPeDiff += fMult(FL2FXCONST_DBL(0.7f), fMult(sfbNRelevantLines[sfb], (pNew - pOld))); } } return specPeDiff; } /* Function: FDKaacEnc_improveScf Description: Calculate the distortion by quantization and inverse quantization of the spectrum with various scalefactors. The scalefactor which provides the best results will be used. */ static INT FDKaacEnc_improveScf(const FIXP_DBL *spec, SHORT *quantSpec, SHORT *quantSpecTmp, INT sfbWidth, FIXP_DBL threshLdData, INT scf, INT minScf, FIXP_DBL *distLdData, INT *minScfCalculated, INT dZoneQuantEnable) { FIXP_DBL sfbDistLdData; INT scfBest = scf; INT k; FIXP_DBL distFactorLdData = FL2FXCONST_DBL(-0.0050301265); /* ld64(1/1.25) */ /* calc real distortion */ sfbDistLdData = FDKaacEnc_calcSfbDist(spec, quantSpec, sfbWidth, scf, dZoneQuantEnable); *minScfCalculated = scf; /* nmr > 1.25 -> try to improve nmr */ if (sfbDistLdData > (threshLdData - distFactorLdData)) { INT scfEstimated = scf; FIXP_DBL sfbDistBestLdData = sfbDistLdData; INT cnt; /* improve by bigger scf ? */ cnt = 0; while ((sfbDistLdData > (threshLdData - distFactorLdData)) && (cnt++ < UPCOUNT_LIMIT)) { scf++; sfbDistLdData = FDKaacEnc_calcSfbDist(spec, quantSpecTmp, sfbWidth, scf, dZoneQuantEnable); if (sfbDistLdData < sfbDistBestLdData) { scfBest = scf; sfbDistBestLdData = sfbDistLdData; for (k = 0; k < sfbWidth; k++) quantSpec[k] = quantSpecTmp[k]; } } /* improve by smaller scf ? */ cnt = 0; scf = scfEstimated; sfbDistLdData = sfbDistBestLdData; while ((sfbDistLdData > (threshLdData - distFactorLdData)) && (cnt++ < 1) && (scf > minScf)) { scf--; sfbDistLdData = FDKaacEnc_calcSfbDist(spec, quantSpecTmp, sfbWidth, scf, dZoneQuantEnable); if (sfbDistLdData < sfbDistBestLdData) { scfBest = scf; sfbDistBestLdData = sfbDistLdData; for (k = 0; k < sfbWidth; k++) quantSpec[k] = quantSpecTmp[k]; } *minScfCalculated = scf; } *distLdData = sfbDistBestLdData; } else { /* nmr <= 1.25 -> try to find bigger scf to use less bits */ FIXP_DBL sfbDistBestLdData = sfbDistLdData; FIXP_DBL sfbDistAllowedLdData = fixMin(sfbDistLdData - distFactorLdData, threshLdData); int cnt; for (cnt = 0; cnt < UPCOUNT_LIMIT; cnt++) { scf++; sfbDistLdData = FDKaacEnc_calcSfbDist(spec, quantSpecTmp, sfbWidth, scf, dZoneQuantEnable); if (sfbDistLdData < sfbDistAllowedLdData) { *minScfCalculated = scfBest + 1; scfBest = scf; sfbDistBestLdData = sfbDistLdData; for (k = 0; k < sfbWidth; k++) quantSpec[k] = quantSpecTmp[k]; } } *distLdData = sfbDistBestLdData; } /* return best scalefactor */ return scfBest; } /* Function: FDKaacEnc_assimilateSingleScf */ static void FDKaacEnc_assimilateSingleScf( const PSY_OUT_CHANNEL *psyOutChan, const QC_OUT_CHANNEL *qcOutChannel, SHORT *quantSpec, SHORT *quantSpecTmp, INT dZoneQuantEnable, INT *scf, const INT *minScf, FIXP_DBL *sfbDist, FIXP_DBL *sfbConstPePart, const FIXP_DBL *sfbFormFactorLdData, const FIXP_DBL *sfbNRelevantLines, INT *minScfCalculated, INT restartOnSuccess) { INT sfbLast, sfbAct, sfbNext; INT scfAct, *scfLast, *scfNext, scfMin, scfMax; INT sfbWidth, sfbOffs; FIXP_DBL enLdData; FIXP_DBL sfbPeOld, sfbPeNew; FIXP_DBL sfbDistNew; INT i, k; INT success = 0; FIXP_DBL deltaPe = FL2FXCONST_DBL(0.0f); FIXP_DBL deltaPeNew, deltaPeTmp; INT prevScfLast[MAX_GROUPED_SFB], prevScfNext[MAX_GROUPED_SFB]; FIXP_DBL deltaPeLast[MAX_GROUPED_SFB]; INT updateMinScfCalculated; for (i = 0; i < psyOutChan->sfbCnt; i++) { prevScfLast[i] = FDK_INT_MAX; prevScfNext[i] = FDK_INT_MAX; deltaPeLast[i] = (FIXP_DBL)FDK_INT_MAX; } sfbLast = -1; sfbAct = -1; sfbNext = -1; scfLast = 0; scfNext = 0; scfMin = FDK_INT_MAX; scfMax = FDK_INT_MAX; do { /* search for new relevant sfb */ sfbNext++; while ((sfbNext < psyOutChan->sfbCnt) && (scf[sfbNext] == FDK_INT_MIN)) sfbNext++; if ((sfbLast >= 0) && (sfbAct >= 0) && (sfbNext < psyOutChan->sfbCnt)) { /* relevant scfs to the left and to the right */ scfAct = scf[sfbAct]; scfLast = scf + sfbLast; scfNext = scf + sfbNext; scfMin = fixMin(*scfLast, *scfNext); scfMax = fixMax(*scfLast, *scfNext); } else if ((sfbLast == -1) && (sfbAct >= 0) && (sfbNext < psyOutChan->sfbCnt)) { /* first relevant scf */ scfAct = scf[sfbAct]; scfLast = &scfAct; scfNext = scf + sfbNext; scfMin = *scfNext; scfMax = *scfNext; } else if ((sfbLast >= 0) && (sfbAct >= 0) && (sfbNext == psyOutChan->sfbCnt)) { /* last relevant scf */ scfAct = scf[sfbAct]; scfLast = scf + sfbLast; scfNext = &scfAct; scfMin = *scfLast; scfMax = *scfLast; } if (sfbAct >= 0) scfMin = fixMax(scfMin, minScf[sfbAct]); if ((sfbAct >= 0) && (sfbLast >= 0 || sfbNext < psyOutChan->sfbCnt) && (scfAct > scfMin) && (scfAct <= scfMin + MAX_SCF_DELTA) && (scfAct >= scfMax - MAX_SCF_DELTA) && (scfAct <= fixMin(scfMin, fixMin(*scfLast, *scfNext)) + MAX_SCF_DELTA) && (*scfLast != prevScfLast[sfbAct] || *scfNext != prevScfNext[sfbAct] || deltaPe < deltaPeLast[sfbAct])) { /* bigger than neighbouring scf found, try to use smaller scf */ success = 0; sfbWidth = psyOutChan->sfbOffsets[sfbAct + 1] - psyOutChan->sfbOffsets[sfbAct]; sfbOffs = psyOutChan->sfbOffsets[sfbAct]; /* estimate required bits for actual scf */ enLdData = qcOutChannel->sfbEnergyLdData[sfbAct]; /* sfbConstPePart[sfbAct] = (float)log(6.75f*en/sfbFormFactor[sfbAct]) * * LOG2_1; */ /* 0.02152255861f = log(6.75)/log(2)/AS_PE_FAC_FLOAT; LOG2_1 is 1.0 for * log2 */ /* 0.09375f = log(64.0)/log(2.0)/64.0 = scale of sfbFormFactorLdData */ if (sfbConstPePart[sfbAct] == (FIXP_DBL)FDK_INT_MIN) { sfbConstPePart[sfbAct] = ((enLdData - sfbFormFactorLdData[sfbAct] - FL2FXCONST_DBL(0.09375f)) >> 1) + FL2FXCONST_DBL(0.02152255861f); } sfbPeOld = FDKaacEnc_calcSingleSpecPe(scfAct, sfbConstPePart[sfbAct], sfbNRelevantLines[sfbAct]) + FDKaacEnc_countSingleScfBits(scfAct, *scfLast, *scfNext); deltaPeNew = deltaPe; updateMinScfCalculated = 1; do { /* estimate required bits for smaller scf */ scfAct--; /* check only if the same check was not done before */ if (scfAct < minScfCalculated[sfbAct] && scfAct >= scfMax - MAX_SCF_DELTA) { /* estimate required bits for new scf */ sfbPeNew = FDKaacEnc_calcSingleSpecPe(scfAct, sfbConstPePart[sfbAct], sfbNRelevantLines[sfbAct]) + FDKaacEnc_countSingleScfBits(scfAct, *scfLast, *scfNext); /* use new scf if no increase in pe and quantization error is smaller */ deltaPeTmp = deltaPe + sfbPeNew - sfbPeOld; /* 0.0006103515625f = 10.0f/(2^(2*AS_PE_FAC_SHIFT)) */ if (deltaPeTmp < FL2FXCONST_DBL(0.0006103515625f)) { /* distortion of new scf */ sfbDistNew = FDKaacEnc_calcSfbDist( qcOutChannel->mdctSpectrum + sfbOffs, quantSpecTmp + sfbOffs, sfbWidth, scfAct, dZoneQuantEnable); if (sfbDistNew < sfbDist[sfbAct]) { /* success, replace scf by new one */ scf[sfbAct] = scfAct; sfbDist[sfbAct] = sfbDistNew; for (k = 0; k < sfbWidth; k++) quantSpec[sfbOffs + k] = quantSpecTmp[sfbOffs + k]; deltaPeNew = deltaPeTmp; success = 1; } /* mark as already checked */ if (updateMinScfCalculated) minScfCalculated[sfbAct] = scfAct; } else { /* from this scf value on not all new values have been checked */ updateMinScfCalculated = 0; } } } while (scfAct > scfMin); deltaPe = deltaPeNew; /* save parameters to avoid multiple computations of the same sfb */ prevScfLast[sfbAct] = *scfLast; prevScfNext[sfbAct] = *scfNext; deltaPeLast[sfbAct] = deltaPe; } if (success && restartOnSuccess) { /* start again at first sfb */ sfbLast = -1; sfbAct = -1; sfbNext = -1; scfLast = 0; scfNext = 0; scfMin = FDK_INT_MAX; scfMax = FDK_INT_MAX; success = 0; } else { /* shift sfbs for next band */ sfbLast = sfbAct; sfbAct = sfbNext; } } while (sfbNext < psyOutChan->sfbCnt); } /* Function: FDKaacEnc_assimilateMultipleScf */ static void FDKaacEnc_assimilateMultipleScf( PSY_OUT_CHANNEL *psyOutChan, QC_OUT_CHANNEL *qcOutChannel, SHORT *quantSpec, SHORT *quantSpecTmp, INT dZoneQuantEnable, INT *scf, const INT *minScf, FIXP_DBL *sfbDist, FIXP_DBL *sfbConstPePart, FIXP_DBL *sfbFormFactorLdData, FIXP_DBL *sfbNRelevantLines) { INT sfb, startSfb, stopSfb; INT scfTmp[MAX_GROUPED_SFB], scfMin, scfMax, scfAct; INT possibleRegionFound; INT sfbWidth, sfbOffs, i, k; FIXP_DBL sfbDistNew[MAX_GROUPED_SFB], distOldSum, distNewSum; INT deltaScfBits; FIXP_DBL deltaSpecPe; FIXP_DBL deltaPe = FL2FXCONST_DBL(0.0f); FIXP_DBL deltaPeNew; INT sfbCnt = psyOutChan->sfbCnt; /* calc min and max scalfactors */ scfMin = FDK_INT_MAX; scfMax = FDK_INT_MIN; for (sfb = 0; sfb < sfbCnt; sfb++) { if (scf[sfb] != FDK_INT_MIN) { scfMin = fixMin(scfMin, scf[sfb]); scfMax = fixMax(scfMax, scf[sfb]); } } if (scfMax != FDK_INT_MIN && scfMax <= scfMin + MAX_SCF_DELTA) { scfAct = scfMax; do { /* try smaller scf */ scfAct--; for (i = 0; i < MAX_GROUPED_SFB; i++) scfTmp[i] = scf[i]; stopSfb = 0; do { /* search for region where all scfs are bigger than scfAct */ sfb = stopSfb; while (sfb < sfbCnt && (scf[sfb] == FDK_INT_MIN || scf[sfb] <= scfAct)) sfb++; startSfb = sfb; sfb++; while (sfb < sfbCnt && (scf[sfb] == FDK_INT_MIN || scf[sfb] > scfAct)) sfb++; stopSfb = sfb; /* check if in all sfb of a valid region scfAct >= minScf[sfb] */ possibleRegionFound = 0; if (startSfb < sfbCnt) { possibleRegionFound = 1; for (sfb = startSfb; sfb < stopSfb; sfb++) { if (scf[sfb] != FDK_INT_MIN) if (scfAct < minScf[sfb]) { possibleRegionFound = 0; break; } } } if (possibleRegionFound) { /* region found */ /* replace scfs in region by scfAct */ for (sfb = startSfb; sfb < stopSfb; sfb++) { if (scfTmp[sfb] != FDK_INT_MIN) scfTmp[sfb] = scfAct; } /* estimate change in bit demand for new scfs */ deltaScfBits = FDKaacEnc_countScfBitsDiff(scf, scfTmp, sfbCnt, startSfb, stopSfb); deltaSpecPe = FDKaacEnc_calcSpecPeDiff( psyOutChan, qcOutChannel, scf, scfTmp, sfbConstPePart, sfbFormFactorLdData, sfbNRelevantLines, startSfb, stopSfb); deltaPeNew = deltaPe + (FIXP_DBL)deltaScfBits + deltaSpecPe; /* new bit demand small enough ? */ /* 0.0006103515625f = 10.0f/(2^(2*AS_PE_FAC_SHIFT)) */ if (deltaPeNew < FL2FXCONST_DBL(0.0006103515625f)) { /* quantize and calc sum of new distortion */ distOldSum = distNewSum = FL2FXCONST_DBL(0.0f); for (sfb = startSfb; sfb < stopSfb; sfb++) { if (scfTmp[sfb] != FDK_INT_MIN) { distOldSum += CalcInvLdData(sfbDist[sfb]) >> DIST_FAC_SHIFT; sfbWidth = psyOutChan->sfbOffsets[sfb + 1] - psyOutChan->sfbOffsets[sfb]; sfbOffs = psyOutChan->sfbOffsets[sfb]; sfbDistNew[sfb] = FDKaacEnc_calcSfbDist( qcOutChannel->mdctSpectrum + sfbOffs, quantSpecTmp + sfbOffs, sfbWidth, scfAct, dZoneQuantEnable); if (sfbDistNew[sfb] > qcOutChannel->sfbThresholdLdData[sfb]) { /* no improvement, skip further dist. calculations */ distNewSum = distOldSum << 1; break; } distNewSum += CalcInvLdData(sfbDistNew[sfb]) >> DIST_FAC_SHIFT; } } /* distortion smaller ? -> use new scalefactors */ if (distNewSum < distOldSum) { deltaPe = deltaPeNew; for (sfb = startSfb; sfb < stopSfb; sfb++) { if (scf[sfb] != FDK_INT_MIN) { sfbWidth = psyOutChan->sfbOffsets[sfb + 1] - psyOutChan->sfbOffsets[sfb]; sfbOffs = psyOutChan->sfbOffsets[sfb]; scf[sfb] = scfAct; sfbDist[sfb] = sfbDistNew[sfb]; for (k = 0; k < sfbWidth; k++) quantSpec[sfbOffs + k] = quantSpecTmp[sfbOffs + k]; } } } } } } while (stopSfb <= sfbCnt); } while (scfAct > scfMin); } } /* Function: FDKaacEnc_FDKaacEnc_assimilateMultipleScf2 */ static void FDKaacEnc_FDKaacEnc_assimilateMultipleScf2( PSY_OUT_CHANNEL *psyOutChan, QC_OUT_CHANNEL *qcOutChannel, SHORT *quantSpec, SHORT *quantSpecTmp, INT dZoneQuantEnable, INT *scf, const INT *minScf, FIXP_DBL *sfbDist, FIXP_DBL *sfbConstPePart, FIXP_DBL *sfbFormFactorLdData, FIXP_DBL *sfbNRelevantLines) { INT sfb, startSfb, stopSfb; INT scfTmp[MAX_GROUPED_SFB], scfAct, scfNew; INT scfPrev, scfNext, scfPrevNextMin, scfPrevNextMax, scfLo, scfHi; INT scfMin, scfMax; INT *sfbOffs = psyOutChan->sfbOffsets; FIXP_DBL sfbDistNew[MAX_GROUPED_SFB], sfbDistMax[MAX_GROUPED_SFB]; FIXP_DBL distOldSum, distNewSum; INT deltaScfBits; FIXP_DBL deltaSpecPe; FIXP_DBL deltaPe = FL2FXCONST_DBL(0.0f); FIXP_DBL deltaPeNew = FL2FXCONST_DBL(0.0f); INT sfbCnt = psyOutChan->sfbCnt; INT bSuccess, bCheckScf; INT i, k; /* calc min and max scalfactors */ scfMin = FDK_INT_MAX; scfMax = FDK_INT_MIN; for (sfb = 0; sfb < sfbCnt; sfb++) { if (scf[sfb] != FDK_INT_MIN) { scfMin = fixMin(scfMin, scf[sfb]); scfMax = fixMax(scfMax, scf[sfb]); } } stopSfb = 0; scfAct = FDK_INT_MIN; do { /* search for region with same scf values scfAct */ scfPrev = scfAct; sfb = stopSfb; while (sfb < sfbCnt && (scf[sfb] == FDK_INT_MIN)) sfb++; startSfb = sfb; scfAct = scf[startSfb]; sfb++; while (sfb < sfbCnt && ((scf[sfb] == FDK_INT_MIN) || (scf[sfb] == scf[startSfb]))) sfb++; stopSfb = sfb; if (stopSfb < sfbCnt) scfNext = scf[stopSfb]; else scfNext = scfAct; if (scfPrev == FDK_INT_MIN) scfPrev = scfAct; scfPrevNextMax = fixMax(scfPrev, scfNext); scfPrevNextMin = fixMin(scfPrev, scfNext); /* try to reduce bits by checking scf values in the range scf[startSfb]...scfHi */ scfHi = fixMax(scfPrevNextMax, scfAct); /* try to find a better solution by reducing the scf difference to the nearest possible lower scf */ if (scfPrevNextMax >= scfAct) scfLo = fixMin(scfAct, scfPrevNextMin); else scfLo = scfPrevNextMax; if (startSfb < sfbCnt && scfHi - scfLo <= MAX_SCF_DELTA) { /* region found */ /* 1. try to save bits by coarser quantization */ if (scfHi > scf[startSfb]) { /* calculate the allowed distortion */ for (sfb = startSfb; sfb < stopSfb; sfb++) { if (scf[sfb] != FDK_INT_MIN) { /* sfbDistMax[sfb] = * (float)pow(qcOutChannel->sfbThreshold[sfb]*sfbDist[sfb]*sfbDist[sfb],1.0f/3.0f); */ /* sfbDistMax[sfb] = * fixMax(sfbDistMax[sfb],qcOutChannel->sfbEnergy[sfb]*FL2FXCONST_DBL(1.e-3f)); */ /* -0.15571537944 = ld64(1.e-3f)*/ sfbDistMax[sfb] = fMult(FL2FXCONST_DBL(1.0f / 3.0f), qcOutChannel->sfbThresholdLdData[sfb]) + fMult(FL2FXCONST_DBL(1.0f / 3.0f), sfbDist[sfb]) + fMult(FL2FXCONST_DBL(1.0f / 3.0f), sfbDist[sfb]); sfbDistMax[sfb] = fixMax(sfbDistMax[sfb], qcOutChannel->sfbEnergyLdData[sfb] - FL2FXCONST_DBL(0.15571537944)); sfbDistMax[sfb] = fixMin(sfbDistMax[sfb], qcOutChannel->sfbThresholdLdData[sfb]); } } /* loop over all possible scf values for this region */ bCheckScf = 1; for (scfNew = scf[startSfb] + 1; scfNew <= scfHi; scfNew++) { for (k = 0; k < MAX_GROUPED_SFB; k++) scfTmp[k] = scf[k]; /* replace scfs in region by scfNew */ for (sfb = startSfb; sfb < stopSfb; sfb++) { if (scfTmp[sfb] != FDK_INT_MIN) scfTmp[sfb] = scfNew; } /* estimate change in bit demand for new scfs */ deltaScfBits = FDKaacEnc_countScfBitsDiff(scf, scfTmp, sfbCnt, startSfb, stopSfb); deltaSpecPe = FDKaacEnc_calcSpecPeDiff( psyOutChan, qcOutChannel, scf, scfTmp, sfbConstPePart, sfbFormFactorLdData, sfbNRelevantLines, startSfb, stopSfb); deltaPeNew = deltaPe + (FIXP_DBL)deltaScfBits + deltaSpecPe; /* new bit demand small enough ? */ if (deltaPeNew < FL2FXCONST_DBL(0.0f)) { bSuccess = 1; /* quantize and calc sum of new distortion */ for (sfb = startSfb; sfb < stopSfb; sfb++) { if (scfTmp[sfb] != FDK_INT_MIN) { sfbDistNew[sfb] = FDKaacEnc_calcSfbDist( qcOutChannel->mdctSpectrum + sfbOffs[sfb], quantSpecTmp + sfbOffs[sfb], sfbOffs[sfb + 1] - sfbOffs[sfb], scfNew, dZoneQuantEnable); if (sfbDistNew[sfb] > sfbDistMax[sfb]) { /* no improvement, skip further dist. calculations */ bSuccess = 0; if (sfbDistNew[sfb] == qcOutChannel->sfbEnergyLdData[sfb]) { /* if whole sfb is already quantized to 0, further checks with even coarser quant. are useless*/ bCheckScf = 0; } break; } } } if (bCheckScf == 0) /* further calculations useless ? */ break; /* distortion small enough ? -> use new scalefactors */ if (bSuccess) { deltaPe = deltaPeNew; for (sfb = startSfb; sfb < stopSfb; sfb++) { if (scf[sfb] != FDK_INT_MIN) { scf[sfb] = scfNew; sfbDist[sfb] = sfbDistNew[sfb]; for (k = 0; k < sfbOffs[sfb + 1] - sfbOffs[sfb]; k++) quantSpec[sfbOffs[sfb] + k] = quantSpecTmp[sfbOffs[sfb] + k]; } } } } } } /* 2. only if coarser quantization was not successful, try to find a better solution by finer quantization and reducing bits for scalefactor coding */ if (scfAct == scf[startSfb] && scfLo < scfAct && scfMax - scfMin <= MAX_SCF_DELTA) { int bminScfViolation = 0; for (k = 0; k < MAX_GROUPED_SFB; k++) scfTmp[k] = scf[k]; scfNew = scfLo; /* replace scfs in region by scfNew and check if in all sfb scfNew >= minScf[sfb] */ for (sfb = startSfb; sfb < stopSfb; sfb++) { if (scfTmp[sfb] != FDK_INT_MIN) { scfTmp[sfb] = scfNew; if (scfNew < minScf[sfb]) bminScfViolation = 1; } } if (!bminScfViolation) { /* estimate change in bit demand for new scfs */ deltaScfBits = FDKaacEnc_countScfBitsDiff(scf, scfTmp, sfbCnt, startSfb, stopSfb); deltaSpecPe = FDKaacEnc_calcSpecPeDiff( psyOutChan, qcOutChannel, scf, scfTmp, sfbConstPePart, sfbFormFactorLdData, sfbNRelevantLines, startSfb, stopSfb); deltaPeNew = deltaPe + (FIXP_DBL)deltaScfBits + deltaSpecPe; } /* new bit demand small enough ? */ if (!bminScfViolation && deltaPeNew < FL2FXCONST_DBL(0.0f)) { /* quantize and calc sum of new distortion */ distOldSum = distNewSum = FL2FXCONST_DBL(0.0f); for (sfb = startSfb; sfb < stopSfb; sfb++) { if (scfTmp[sfb] != FDK_INT_MIN) { distOldSum += CalcInvLdData(sfbDist[sfb]) >> DIST_FAC_SHIFT; sfbDistNew[sfb] = FDKaacEnc_calcSfbDist( qcOutChannel->mdctSpectrum + sfbOffs[sfb], quantSpecTmp + sfbOffs[sfb], sfbOffs[sfb + 1] - sfbOffs[sfb], scfNew, dZoneQuantEnable); if (sfbDistNew[sfb] > qcOutChannel->sfbThresholdLdData[sfb]) { /* no improvement, skip further dist. calculations */ distNewSum = distOldSum << 1; break; } distNewSum += CalcInvLdData(sfbDistNew[sfb]) >> DIST_FAC_SHIFT; } } /* distortion smaller ? -> use new scalefactors */ if (distNewSum < fMult(FL2FXCONST_DBL(0.8f), distOldSum)) { deltaPe = deltaPeNew; for (sfb = startSfb; sfb < stopSfb; sfb++) { if (scf[sfb] != FDK_INT_MIN) { scf[sfb] = scfNew; sfbDist[sfb] = sfbDistNew[sfb]; for (k = 0; k < sfbOffs[sfb + 1] - sfbOffs[sfb]; k++) quantSpec[sfbOffs[sfb] + k] = quantSpecTmp[sfbOffs[sfb] + k]; } } } } } /* 3. try to find a better solution (save bits) by only reducing the scalefactor without new quantization */ if (scfMax - scfMin <= MAX_SCF_DELTA - 3) { /* 3 bec. scf is reduced 3 times, see for loop below */ for (k = 0; k < sfbCnt; k++) scfTmp[k] = scf[k]; for (i = 0; i < 3; i++) { scfNew = scfTmp[startSfb] - 1; /* replace scfs in region by scfNew */ for (sfb = startSfb; sfb < stopSfb; sfb++) { if (scfTmp[sfb] != FDK_INT_MIN) scfTmp[sfb] = scfNew; } /* estimate change in bit demand for new scfs */ deltaScfBits = FDKaacEnc_countScfBitsDiff(scf, scfTmp, sfbCnt, startSfb, stopSfb); deltaPeNew = deltaPe + (FIXP_DBL)deltaScfBits; /* new bit demand small enough ? */ if (deltaPeNew <= FL2FXCONST_DBL(0.0f)) { bSuccess = 1; distOldSum = distNewSum = FL2FXCONST_DBL(0.0f); for (sfb = startSfb; sfb < stopSfb; sfb++) { if (scfTmp[sfb] != FDK_INT_MIN) { FIXP_DBL sfbEnQ; /* calc the energy and distortion of the quantized spectrum for a smaller scf */ FDKaacEnc_calcSfbQuantEnergyAndDist( qcOutChannel->mdctSpectrum + sfbOffs[sfb], quantSpec + sfbOffs[sfb], sfbOffs[sfb + 1] - sfbOffs[sfb], scfNew, &sfbEnQ, &sfbDistNew[sfb]); distOldSum += CalcInvLdData(sfbDist[sfb]) >> DIST_FAC_SHIFT; distNewSum += CalcInvLdData(sfbDistNew[sfb]) >> DIST_FAC_SHIFT; /* 0.00259488556167 = ld64(1.122f) */ /* -0.00778722686652 = ld64(0.7079f) */ if ((sfbDistNew[sfb] > (sfbDist[sfb] + FL2FXCONST_DBL(0.00259488556167f))) || (sfbEnQ < (qcOutChannel->sfbEnergyLdData[sfb] - FL2FXCONST_DBL(0.00778722686652f)))) { bSuccess = 0; break; } } } /* distortion smaller ? -> use new scalefactors */ if (distNewSum < distOldSum && bSuccess) { deltaPe = deltaPeNew; for (sfb = startSfb; sfb < stopSfb; sfb++) { if (scf[sfb] != FDK_INT_MIN) { scf[sfb] = scfNew; sfbDist[sfb] = sfbDistNew[sfb]; } } } } } } } } while (stopSfb <= sfbCnt); } static void FDKaacEnc_EstimateScaleFactorsChannel( QC_OUT_CHANNEL *qcOutChannel, PSY_OUT_CHANNEL *psyOutChannel, INT *RESTRICT scf, INT *RESTRICT globalGain, FIXP_DBL *RESTRICT sfbFormFactorLdData, const INT invQuant, SHORT *RESTRICT quantSpec, const INT dZoneQuantEnable) { INT i, j, sfb, sfbOffs; INT scfInt; INT maxSf; INT minSf; FIXP_DBL threshLdData; FIXP_DBL energyLdData; FIXP_DBL energyPartLdData; FIXP_DBL thresholdPartLdData; FIXP_DBL scfFract; FIXP_DBL maxSpec; INT minScfCalculated[MAX_GROUPED_SFB]; FIXP_DBL sfbDistLdData[MAX_GROUPED_SFB]; C_ALLOC_SCRATCH_START(quantSpecTmp, SHORT, (1024)) INT minSfMaxQuant[MAX_GROUPED_SFB]; FIXP_DBL threshConstLdData = FL2FXCONST_DBL(0.04304511722f); /* log10(6.75)/log10(2.0)/64.0 */ FIXP_DBL convConst = FL2FXCONST_DBL(0.30102999566f); /* log10(2.0) */ FIXP_DBL c1Const = FL2FXCONST_DBL(-0.27083183594f); /* C1 = -69.33295 => C1/2^8 */ if (invQuant > 0) { FDKmemclear(quantSpec, (1024) * sizeof(SHORT)); } /* scfs without energy or with thresh>energy are marked with FDK_INT_MIN */ for (i = 0; i < psyOutChannel->sfbCnt; i++) { scf[i] = FDK_INT_MIN; } for (i = 0; i < MAX_GROUPED_SFB; i++) { minSfMaxQuant[i] = FDK_INT_MIN; } for (sfbOffs = 0; sfbOffs < psyOutChannel->sfbCnt; sfbOffs += psyOutChannel->sfbPerGroup) { for (sfb = 0; sfb < psyOutChannel->maxSfbPerGroup; sfb++) { threshLdData = qcOutChannel->sfbThresholdLdData[sfbOffs + sfb]; energyLdData = qcOutChannel->sfbEnergyLdData[sfbOffs + sfb]; sfbDistLdData[sfbOffs + sfb] = energyLdData; if (energyLdData > threshLdData) { FIXP_DBL tmp; /* energyPart = (float)log10(sfbFormFactor[sfbOffs+sfb]); */ /* 0.09375f = log(64.0)/log(2.0)/64.0 = scale of sfbFormFactorLdData */ energyPartLdData = sfbFormFactorLdData[sfbOffs + sfb] + FL2FXCONST_DBL(0.09375f); /* influence of allowed distortion */ /* thresholdPart = (float)log10(6.75*thresh+FLT_MIN); */ thresholdPartLdData = threshConstLdData + threshLdData; /* scf calc */ /* scfFloat = 8.8585f * (thresholdPart - energyPart); */ scfFract = thresholdPartLdData - energyPartLdData; /* conversion from log2 to log10 */ scfFract = fMult(convConst, scfFract); /* (8.8585f * scfFract)/8 = 8/8 * scfFract + 0.8585 * scfFract/8 */ scfFract = scfFract + fMult(FL2FXCONST_DBL(0.8585f), scfFract >> 3); /* integer scalefactor */ /* scfInt = (int)floor(scfFloat); */ scfInt = (INT)(scfFract >> ((DFRACT_BITS - 1) - 3 - LD_DATA_SHIFT)); /* 3 bits => scfFract/8.0; 6 bits => ld64 */ /* maximum of spectrum */ maxSpec = FL2FXCONST_DBL(0.0f); /* Unroll by 4, allow dual memory access */ DWORD_ALIGNED(qcOutChannel->mdctSpectrum); for (j = psyOutChannel->sfbOffsets[sfbOffs + sfb]; j < psyOutChannel->sfbOffsets[sfbOffs + sfb + 1]; j += 4) { maxSpec = fMax(maxSpec, fMax(fMax(fAbs(qcOutChannel->mdctSpectrum[j + 0]), fAbs(qcOutChannel->mdctSpectrum[j + 1])), fMax(fAbs(qcOutChannel->mdctSpectrum[j + 2]), fAbs(qcOutChannel->mdctSpectrum[j + 3])))); } /* lower scf limit to avoid quantized values bigger than MAX_QUANT */ /* C1 = -69.33295f, C2 = 5.77078f = 4/log(2) */ /* minSfMaxQuant[sfbOffs+sfb] = (int)ceil(C1 + C2*log(maxSpec)); */ /* C1/2^8 + 4/log(2.0)*log(maxSpec)/2^8 => C1/2^8 + * log(maxSpec)/log(2.0)*4/2^8 => C1/2^8 + log(maxSpec)/log(2.0)/64.0 */ // minSfMaxQuant[sfbOffs+sfb] = ((INT) ((c1Const + CalcLdData(maxSpec)) // >> ((DFRACT_BITS-1)-8))) + 1; tmp = CalcLdData(maxSpec); if (c1Const > FL2FXCONST_DBL(-1.f) - tmp) { minSfMaxQuant[sfbOffs + sfb] = ((INT)((c1Const + tmp) >> ((DFRACT_BITS - 1) - 8))) + 1; } else { minSfMaxQuant[sfbOffs + sfb] = ((INT)(FL2FXCONST_DBL(-1.f) >> ((DFRACT_BITS - 1) - 8))) + 1; } scfInt = fixMax(scfInt, minSfMaxQuant[sfbOffs + sfb]); /* find better scalefactor with analysis by synthesis */ if (invQuant > 0) { scfInt = FDKaacEnc_improveScf( qcOutChannel->mdctSpectrum + psyOutChannel->sfbOffsets[sfbOffs + sfb], quantSpec + psyOutChannel->sfbOffsets[sfbOffs + sfb], quantSpecTmp + psyOutChannel->sfbOffsets[sfbOffs + sfb], psyOutChannel->sfbOffsets[sfbOffs + sfb + 1] - psyOutChannel->sfbOffsets[sfbOffs + sfb], threshLdData, scfInt, minSfMaxQuant[sfbOffs + sfb], &sfbDistLdData[sfbOffs + sfb], &minScfCalculated[sfbOffs + sfb], dZoneQuantEnable); } scf[sfbOffs + sfb] = scfInt; } } } if (invQuant > 0) { /* try to decrease scf differences */ FIXP_DBL sfbConstPePart[MAX_GROUPED_SFB]; FIXP_DBL sfbNRelevantLines[MAX_GROUPED_SFB]; for (i = 0; i < psyOutChannel->sfbCnt; i++) sfbConstPePart[i] = (FIXP_DBL)FDK_INT_MIN; FDKaacEnc_calcSfbRelevantLines( sfbFormFactorLdData, qcOutChannel->sfbEnergyLdData, qcOutChannel->sfbThresholdLdData, psyOutChannel->sfbOffsets, psyOutChannel->sfbCnt, psyOutChannel->sfbPerGroup, psyOutChannel->maxSfbPerGroup, sfbNRelevantLines); FDKaacEnc_assimilateSingleScf( psyOutChannel, qcOutChannel, quantSpec, quantSpecTmp, dZoneQuantEnable, scf, minSfMaxQuant, sfbDistLdData, sfbConstPePart, sfbFormFactorLdData, sfbNRelevantLines, minScfCalculated, 1); if (invQuant > 1) { FDKaacEnc_assimilateMultipleScf( psyOutChannel, qcOutChannel, quantSpec, quantSpecTmp, dZoneQuantEnable, scf, minSfMaxQuant, sfbDistLdData, sfbConstPePart, sfbFormFactorLdData, sfbNRelevantLines); FDKaacEnc_FDKaacEnc_assimilateMultipleScf2( psyOutChannel, qcOutChannel, quantSpec, quantSpecTmp, dZoneQuantEnable, scf, minSfMaxQuant, sfbDistLdData, sfbConstPePart, sfbFormFactorLdData, sfbNRelevantLines); } } /* get min scalefac */ minSf = FDK_INT_MAX; for (sfbOffs = 0; sfbOffs < psyOutChannel->sfbCnt; sfbOffs += psyOutChannel->sfbPerGroup) { for (sfb = 0; sfb < psyOutChannel->maxSfbPerGroup; sfb++) { if (scf[sfbOffs + sfb] != FDK_INT_MIN) minSf = fixMin(minSf, scf[sfbOffs + sfb]); } } /* limit scf delta */ for (sfbOffs = 0; sfbOffs < psyOutChannel->sfbCnt; sfbOffs += psyOutChannel->sfbPerGroup) { for (sfb = 0; sfb < psyOutChannel->maxSfbPerGroup; sfb++) { if ((scf[sfbOffs + sfb] != FDK_INT_MIN) && (minSf + MAX_SCF_DELTA) < scf[sfbOffs + sfb]) { scf[sfbOffs + sfb] = minSf + MAX_SCF_DELTA; if (invQuant > 0) { /* changed bands need to be quantized again */ sfbDistLdData[sfbOffs + sfb] = FDKaacEnc_calcSfbDist( qcOutChannel->mdctSpectrum + psyOutChannel->sfbOffsets[sfbOffs + sfb], quantSpec + psyOutChannel->sfbOffsets[sfbOffs + sfb], psyOutChannel->sfbOffsets[sfbOffs + sfb + 1] - psyOutChannel->sfbOffsets[sfbOffs + sfb], scf[sfbOffs + sfb], dZoneQuantEnable); } } } } /* get max scalefac for global gain */ maxSf = FDK_INT_MIN; for (sfbOffs = 0; sfbOffs < psyOutChannel->sfbCnt; sfbOffs += psyOutChannel->sfbPerGroup) { for (sfb = 0; sfb < psyOutChannel->maxSfbPerGroup; sfb++) { maxSf = fixMax(maxSf, scf[sfbOffs + sfb]); } } /* calc loop scalefactors, if spec is not all zero (i.e. maxSf == -99) */ if (maxSf > FDK_INT_MIN) { *globalGain = maxSf; for (sfbOffs = 0; sfbOffs < psyOutChannel->sfbCnt; sfbOffs += psyOutChannel->sfbPerGroup) { for (sfb = 0; sfb < psyOutChannel->maxSfbPerGroup; sfb++) { if (scf[sfbOffs + sfb] == FDK_INT_MIN) { scf[sfbOffs + sfb] = 0; /* set band explicitely to zero */ for (j = psyOutChannel->sfbOffsets[sfbOffs + sfb]; j < psyOutChannel->sfbOffsets[sfbOffs + sfb + 1]; j++) { qcOutChannel->mdctSpectrum[j] = FL2FXCONST_DBL(0.0f); } } else { scf[sfbOffs + sfb] = maxSf - scf[sfbOffs + sfb]; } } } } else { *globalGain = 0; /* set spectrum explicitely to zero */ for (sfbOffs = 0; sfbOffs < psyOutChannel->sfbCnt; sfbOffs += psyOutChannel->sfbPerGroup) { for (sfb = 0; sfb < psyOutChannel->maxSfbPerGroup; sfb++) { scf[sfbOffs + sfb] = 0; /* set band explicitely to zero */ for (j = psyOutChannel->sfbOffsets[sfbOffs + sfb]; j < psyOutChannel->sfbOffsets[sfbOffs + sfb + 1]; j++) { qcOutChannel->mdctSpectrum[j] = FL2FXCONST_DBL(0.0f); } } } } /* free quantSpecTmp from scratch */ C_ALLOC_SCRATCH_END(quantSpecTmp, SHORT, (1024)) } void FDKaacEnc_EstimateScaleFactors(PSY_OUT_CHANNEL *psyOutChannel[], QC_OUT_CHANNEL *qcOutChannel[], const INT invQuant, const INT dZoneQuantEnable, const INT nChannels) { int ch; for (ch = 0; ch < nChannels; ch++) { FDKaacEnc_EstimateScaleFactorsChannel( qcOutChannel[ch], psyOutChannel[ch], qcOutChannel[ch]->scf, &qcOutChannel[ch]->globalGain, qcOutChannel[ch]->sfbFormFactorLdData, invQuant, qcOutChannel[ch]->quantSpec, dZoneQuantEnable); } } fdk-aac-2.0.2/libAACenc/src/sf_estim.h000066400000000000000000000126041404225310700173140ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Werner Description: Scale factor estimation *******************************************************************************/ #ifndef SF_ESTIM_H #define SF_ESTIM_H #include "common_fix.h" #include "psy_const.h" #include "qc_data.h" #include "interface.h" #define FORM_FAC_SHIFT 6 void FDKaacEnc_CalcFormFactor(QC_OUT_CHANNEL *qcOutChannel[(2)], PSY_OUT_CHANNEL *psyOutChannel[(2)], const INT nChannels); void FDKaacEnc_EstimateScaleFactors(PSY_OUT_CHANNEL *psyOutChannel[], QC_OUT_CHANNEL *qcOutChannel[], const INT invQuant, const INT dZoneQuantEnable, const INT nChannels); #endif fdk-aac-2.0.2/libAACenc/src/spreading.cpp000066400000000000000000000126651404225310700200210ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M.Werner Description: Spreading of energy *******************************************************************************/ #include "spreading.h" void FDKaacEnc_SpreadingMax(const INT pbCnt, const FIXP_DBL *RESTRICT maskLowFactor, const FIXP_DBL *RESTRICT maskHighFactor, FIXP_DBL *RESTRICT pbSpreadEnergy) { int i; FIXP_DBL delay; /* slope to higher frequencies */ delay = pbSpreadEnergy[0]; for (i = 1; i < pbCnt; i++) { delay = fixMax(pbSpreadEnergy[i], fMult(maskHighFactor[i], delay)); pbSpreadEnergy[i] = delay; } /* slope to lower frequencies */ delay = pbSpreadEnergy[pbCnt - 1]; for (i = pbCnt - 2; i >= 0; i--) { delay = fixMax(pbSpreadEnergy[i], fMult(maskLowFactor[i], delay)); pbSpreadEnergy[i] = delay; } } fdk-aac-2.0.2/libAACenc/src/spreading.h000066400000000000000000000121241404225310700174540ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M.Werner Description: Spreading of energy and weighted tonality *******************************************************************************/ #ifndef SPREADING_H #define SPREADING_H #include "common_fix.h" void FDKaacEnc_SpreadingMax(const INT pbCnt, const FIXP_DBL *RESTRICT maskLowFactor, const FIXP_DBL *RESTRICT maskHighFactor, FIXP_DBL *RESTRICT pbSpreadEnergy); #endif /* #ifndef SPREADING_H */ fdk-aac-2.0.2/libAACenc/src/tns_func.h000066400000000000000000000135041404225310700173220ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): Alex Goeschel Description: Temporal noise shaping *******************************************************************************/ #ifndef TNS_FUNC_H #define TNS_FUNC_H #include "common_fix.h" #include "psy_configuration.h" AAC_ENCODER_ERROR FDKaacEnc_InitTnsConfiguration( INT bitrate, INT samplerate, INT channels, INT blocktype, INT granuleLength, INT isLowDelay, INT ldSbrPresent, TNS_CONFIG *tnsConfig, PSY_CONFIGURATION *psyConfig, INT active, INT useTnsPeak); INT FDKaacEnc_TnsDetect(TNS_DATA *tnsData, const TNS_CONFIG *tC, TNS_INFO *tnsInfo, INT sfbCnt, const FIXP_DBL *spectrum, INT subBlockNumber, INT blockType); void FDKaacEnc_TnsSync(TNS_DATA *tnsDataDest, const TNS_DATA *tnsDataSrc, TNS_INFO *tnsInfoDest, TNS_INFO *tnsInfoSrc, const INT blockTypeDest, const INT blockTypeSrc, const TNS_CONFIG *tC); INT FDKaacEnc_TnsEncode(TNS_INFO *tnsInfo, TNS_DATA *tnsData, const INT numOfSfb, const TNS_CONFIG *tC, const INT lowPassLine, FIXP_DBL *spectrum, const INT subBlockNumber, const INT blockType); #endif /* TNS_FUNC_H */ fdk-aac-2.0.2/libAACenc/src/tonality.cpp000066400000000000000000000221641404225310700177030ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Werner Description: Convert chaos measure to the tonality index *******************************************************************************/ #include "tonality.h" #include "chaosmeasure.h" #if defined(__arm__) #endif static const FIXP_DBL normlog = (FIXP_DBL)0xd977d949; /*FL2FXCONST_DBL(-0.4342944819f * FDKlog(2.0)/FDKlog(2.7182818)); */ static void FDKaacEnc_CalcSfbTonality(FIXP_DBL *RESTRICT spectrum, INT *RESTRICT sfbMaxScaleSpec, FIXP_DBL *RESTRICT chaosMeasure, FIXP_SGL *RESTRICT sfbTonality, INT sfbCnt, const INT *RESTRICT sfbOffset, FIXP_DBL *RESTRICT sfbEnergyLD64); void FDKaacEnc_CalculateFullTonality(FIXP_DBL *RESTRICT spectrum, INT *RESTRICT sfbMaxScaleSpec, FIXP_DBL *RESTRICT sfbEnergyLD64, FIXP_SGL *RESTRICT sfbTonality, INT sfbCnt, const INT *sfbOffset, INT usePns) { INT j; INT numberOfLines = sfbOffset[sfbCnt]; if (usePns) { C_ALLOC_SCRATCH_START(chaosMeasurePerLine, FIXP_DBL, (1024)) /* calculate chaos measure */ FDKaacEnc_CalculateChaosMeasure(spectrum, numberOfLines, chaosMeasurePerLine); /* smooth ChaosMeasure */ FIXP_DBL left = chaosMeasurePerLine[0]; FIXP_DBL right; for (j = 1; j < (numberOfLines - 1); j += 2) { right = chaosMeasurePerLine[j]; right = right - (right >> 2); left = right + (left >> 2); chaosMeasurePerLine[j] = left; /* 0.25 left + 0.75 right */ right = chaosMeasurePerLine[j + 1]; right = right - (right >> 2); left = right + (left >> 2); chaosMeasurePerLine[j + 1] = left; } if (j == (numberOfLines - 1)) { right = chaosMeasurePerLine[j]; right = right - (right >> 2); left = right + (left >> 2); chaosMeasurePerLine[j] = left; } FDKaacEnc_CalcSfbTonality(spectrum, sfbMaxScaleSpec, chaosMeasurePerLine, sfbTonality, sfbCnt, sfbOffset, sfbEnergyLD64); C_ALLOC_SCRATCH_END(chaosMeasurePerLine, FIXP_DBL, (1024)) } } /***************************************************************************** functionname: CalculateTonalityIndex description: computes tonality values out of unpredictability values limits range and computes log() returns: input: ptr to energies, ptr to chaos measure values, number of sfb output: sfb wise tonality values *****************************************************************************/ static void FDKaacEnc_CalcSfbTonality(FIXP_DBL *RESTRICT spectrum, INT *RESTRICT sfbMaxScaleSpec, FIXP_DBL *RESTRICT chaosMeasure, FIXP_SGL *RESTRICT sfbTonality, INT sfbCnt, const INT *RESTRICT sfbOffset, FIXP_DBL *RESTRICT sfbEnergyLD64) { INT i; for (i = 0; i < sfbCnt; i++) { FIXP_DBL chaosMeasureSfbLD64; INT shiftBits = fixMax(0, sfbMaxScaleSpec[i] - 4); /* max sfbWidth = 96 ; 2^7=128 => 7/2 = 4 (spc*spc) */ INT j; FIXP_DBL chaosMeasureSfb = FL2FXCONST_DBL(0.0); /* calc chaosMeasurePerSfb */ for (j = (sfbOffset[i + 1] - sfbOffset[i]) - 1; j >= 0; j--) { FIXP_DBL tmp = (*spectrum++) << shiftBits; FIXP_DBL lineNrg = fMultDiv2(tmp, tmp); chaosMeasureSfb = fMultAddDiv2(chaosMeasureSfb, lineNrg, *chaosMeasure++); } /* calc tonalityPerSfb */ if (chaosMeasureSfb != FL2FXCONST_DBL(0.0)) { /* add ld(convtone)/64 and 2/64 bec.fMultDiv2 */ chaosMeasureSfbLD64 = CalcLdData((chaosMeasureSfb)) - sfbEnergyLD64[i]; chaosMeasureSfbLD64 += FL2FXCONST_DBL(3.0f / 64) - ((FIXP_DBL)(shiftBits) << (DFRACT_BITS - 6)); if (chaosMeasureSfbLD64 > FL2FXCONST_DBL(-0.0519051)) /* > ld(0.05)+ld(2) */ { if (chaosMeasureSfbLD64 <= FL2FXCONST_DBL(0.0)) sfbTonality[i] = FX_DBL2FX_SGL(fMultDiv2(chaosMeasureSfbLD64, normlog) << 7); else sfbTonality[i] = FL2FXCONST_SGL(0.0); } else sfbTonality[i] = (FIXP_SGL)MAXVAL_SGL; } else sfbTonality[i] = (FIXP_SGL)MAXVAL_SGL; } } fdk-aac-2.0.2/libAACenc/src/tonality.h000066400000000000000000000123061404225310700173450ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Lohwasser Description: Calculate tonality index *******************************************************************************/ #ifndef TONALITY_H #define TONALITY_H #include "common_fix.h" #include "chaosmeasure.h" void FDKaacEnc_CalculateFullTonality(FIXP_DBL *RESTRICT spectrum, INT *RESTRICT sfbMaxScaleSpec, FIXP_DBL *RESTRICT sfbEnergyLD64, FIXP_SGL *RESTRICT sfbTonality, INT sfbCnt, const INT *sfbOffset, INT usePns); #endif /* TONALITY_H */ fdk-aac-2.0.2/libAACenc/src/transform.cpp000066400000000000000000000244241404225310700200540ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): Tobias Chalupka Description: FDKaacLdEnc_MdctTransform480: The module FDKaacLdEnc_MdctTransform will perform the MDCT. The MDCT supports the sine window and the zero padded window. The algorithm of the MDCT can be divided in Windowing, PreModulation, Fft and PostModulation. *******************************************************************************/ #include "transform.h" #include "dct.h" #include "psy_const.h" #include "aacEnc_rom.h" #include "FDK_tools_rom.h" #if defined(__arm__) #endif INT FDKaacEnc_Transform_Real(const INT_PCM *pTimeData, FIXP_DBL *RESTRICT mdctData, const INT blockType, const INT windowShape, INT *prevWindowShape, H_MDCT mdctPers, const INT frameLength, INT *pMdctData_e, INT filterType) { const INT_PCM *RESTRICT timeData; UINT numSpec; UINT numMdctLines; UINT offset; int fr; /* fr: right window slope length */ SHORT mdctData_e[8]; timeData = pTimeData; if (blockType == SHORT_WINDOW) { numSpec = 8; numMdctLines = frameLength >> 3; } else { numSpec = 1; numMdctLines = frameLength; } offset = (windowShape == LOL_WINDOW) ? ((frameLength * 3) >> 2) : 0; switch (blockType) { case LONG_WINDOW: case STOP_WINDOW: fr = frameLength - offset; break; case START_WINDOW: /* or StopStartSequence */ case SHORT_WINDOW: fr = frameLength >> 3; break; default: FDK_ASSERT(0); return -1; } mdct_block(mdctPers, timeData, frameLength, mdctData, numSpec, numMdctLines, FDKgetWindowSlope(fr, windowShape), fr, mdctData_e); if (blockType == SHORT_WINDOW) { if (!(mdctData_e[0] == mdctData_e[1] && mdctData_e[1] == mdctData_e[2] && mdctData_e[2] == mdctData_e[3] && mdctData_e[3] == mdctData_e[4] && mdctData_e[4] == mdctData_e[5] && mdctData_e[5] == mdctData_e[6] && mdctData_e[6] == mdctData_e[7])) { return -1; } } *prevWindowShape = windowShape; *pMdctData_e = mdctData_e[0]; return 0; } INT FDKaacEnc_Transform_Real_Eld(const INT_PCM *pTimeData, FIXP_DBL *RESTRICT mdctData, const INT blockType, const INT windowShape, INT *prevWindowShape, const INT frameLength, INT *mdctData_e, INT filterType, FIXP_DBL *RESTRICT overlapAddBuffer) { const INT_PCM *RESTRICT timeData; INT i; /* tl: transform length fl: left window slope length nl: left window slope offset fr: right window slope length nr: right window slope offset */ const FIXP_WTB *pWindowELD = NULL; int N = frameLength; int L = frameLength; timeData = pTimeData; if (blockType != LONG_WINDOW) { return -1; } /* * MDCT scale: * + 1: fMultDiv2() in windowing. * + 1: Because of factor 1/2 in Princen-Bradley compliant windowed TDAC. */ *mdctData_e = 1 + 1; switch (frameLength) { case 512: pWindowELD = ELDAnalysis512; break; case 480: pWindowELD = ELDAnalysis480; break; case 256: pWindowELD = ELDAnalysis256; *mdctData_e += 1; break; case 240: pWindowELD = ELDAnalysis240; *mdctData_e += 1; break; case 128: pWindowELD = ELDAnalysis128; *mdctData_e += 2; break; case 120: pWindowELD = ELDAnalysis120; *mdctData_e += 2; break; default: FDK_ASSERT(0); return -1; } for (i = 0; i < N / 4; i++) { FIXP_DBL z0, outval; z0 = (fMult((FIXP_PCM)timeData[L + N * 3 / 4 - 1 - i], pWindowELD[N / 2 - 1 - i]) << (WTS0 - 1)) + (fMult((FIXP_PCM)timeData[L + N * 3 / 4 + i], pWindowELD[N / 2 + i]) << (WTS0 - 1)); outval = (fMultDiv2((FIXP_PCM)timeData[L + N * 3 / 4 - 1 - i], pWindowELD[N + N / 2 - 1 - i]) >> (-WTS1)); outval += (fMultDiv2((FIXP_PCM)timeData[L + N * 3 / 4 + i], pWindowELD[N + N / 2 + i]) >> (-WTS1)); outval += (fMultDiv2(overlapAddBuffer[N / 2 + i], pWindowELD[2 * N + i]) >> (-WTS2 - 1)); overlapAddBuffer[N / 2 + i] = overlapAddBuffer[i]; overlapAddBuffer[i] = z0; mdctData[i] = overlapAddBuffer[N / 2 + i] + (fMultDiv2(overlapAddBuffer[N + N / 2 - 1 - i], pWindowELD[2 * N + N / 2 + i]) >> (-WTS2 - 1)); mdctData[N - 1 - i] = outval; overlapAddBuffer[N + N / 2 - 1 - i] = outval; } for (i = N / 4; i < N / 2; i++) { FIXP_DBL z0, outval; z0 = fMult((FIXP_PCM)timeData[L + N * 3 / 4 - 1 - i], pWindowELD[N / 2 - 1 - i]) << (WTS0 - 1); outval = (fMultDiv2((FIXP_PCM)timeData[L + N * 3 / 4 - 1 - i], pWindowELD[N + N / 2 - 1 - i]) >> (-WTS1)); outval += (fMultDiv2(overlapAddBuffer[N / 2 + i], pWindowELD[2 * N + i]) >> (-WTS2 - 1)); overlapAddBuffer[N / 2 + i] = overlapAddBuffer[i] + (fMult((FIXP_PCM)timeData[L - N / 4 + i], pWindowELD[N / 2 + i]) << (WTS0 - 1)); overlapAddBuffer[i] = z0; mdctData[i] = overlapAddBuffer[N / 2 + i] + (fMultDiv2(overlapAddBuffer[N + N / 2 - 1 - i], pWindowELD[2 * N + N / 2 + i]) >> (-WTS2 - 1)); mdctData[N - 1 - i] = outval; overlapAddBuffer[N + N / 2 - 1 - i] = outval; } dct_IV(mdctData, frameLength, mdctData_e); *prevWindowShape = windowShape; return 0; } fdk-aac-2.0.2/libAACenc/src/transform.h000066400000000000000000000167551404225310700175310ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** AAC encoder library ****************************** Author(s): M. Werner Description: MDCT Transform *******************************************************************************/ #ifndef TRANSFORM_H #define TRANSFORM_H #include "mdct.h" #include "common_fix.h" #define WTS0 1 #define WTS1 0 #define WTS2 -2 /** * \brief: Performe MDCT transform of time domain data. * \param timeData pointer to time domain input signal. * \param mdctData pointer to store frequency domain output data. * \param blockType index indicating the type of block. Either * LONG_WINDOW, START_WINDOW, SHORT_WINDOW or STOP_WINDOW. * \param windowShape index indicating the window slope type to be used. * Values allowed are either SINE_WINDOW or KBD_WINDOW. * \param previndowShape index indicating the window slope type used * in the last frame. * Values allowed are either SINE_WINDOW or KBD_WINDOW. * \param frameLength length of the block. Either 1024 or 960. * \param mdctData_e pointer to an INT where the exponent of the frequency * domain output data is stored into. * \param filterType xxx * \return 0 in case of success, non-zero in case of error (inconsistent * parameters). */ INT FDKaacEnc_Transform_Real(const INT_PCM* pTimeData, FIXP_DBL* RESTRICT mdctData, const INT blockType, const INT windowShape, INT* prevWindowShape, H_MDCT mdctPers, const INT frameLength, INT* pMdctData_e, INT filterType); /** * \brief: Performe ELD filterbnank transform of time domain data. * \param timeData pointer to time domain input signal. * \param mdctData pointer to store frequency domain output data. * \param blockType index indicating the type of block. Either * LONG_WINDOW, START_WINDOW, SHORT_WINDOW or STOP_WINDOW. * \param windowShape index indicating the window slope type to be used. * Values allowed are either SINE_WINDOW or KBD_WINDOW. * \param previndowShape index indicating the window slope type used * in the last frame. * Values allowed are either SINE_WINDOW or KBD_WINDOW. * \param frameLength length of the block. Either 1024 or 960. * \param mdctData_e pointer to an INT where the exponent of the frequency * domain output data is stored into. * \param filterType xxx * \param overlapAddBuffer overlap add buffer for overlap of ELD filterbank * \return 0 in case of success, non-zero in case of error (inconsistent * parameters). */ INT FDKaacEnc_Transform_Real_Eld(const INT_PCM* pTimeData, FIXP_DBL* RESTRICT mdctData, const INT blockType, const INT windowShape, INT* prevWindowShape, const INT frameLength, INT* mdctData_e, INT filterType, FIXP_DBL* RESTRICT overlapAddBuffer); #endif /* #!defined (TRANSFORM_H) */ fdk-aac-2.0.2/libArithCoding/000077500000000000000000000000001404225310700156615ustar00rootroot00000000000000fdk-aac-2.0.2/libArithCoding/include/000077500000000000000000000000001404225310700173045ustar00rootroot00000000000000fdk-aac-2.0.2/libArithCoding/include/ac_arith_coder.h000066400000000000000000000140421404225310700224040ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /************************** Arithmetic coder library *************************** Author(s): Oliver Weiss Description: Interface for Spectral Noiseless Coding Scheme based on an Arithmetic Coder in Conjunction with an Adaptive Context *******************************************************************************/ #ifndef AC_ARITH_CODER_H #define AC_ARITH_CODER_H #include "common_fix.h" #include "FDK_bitstream.h" #include "FDK_audio.h" typedef enum { ARITH_CODER_OK = 0, ARITH_CODER_ERROR = 5 } ARITH_CODING_ERROR; typedef struct { SHORT m_numberLinesPrev; UCHAR c_prev[(1024 / 2) + 4]; /* 2-tuple context of previous frame, 4 bit */ } CArcoData; /* prototypes */ CArcoData *CArco_Create(void); void CArco_Destroy(CArcoData *pArcoData); /** * \brief decode a spectral data element by using an adaptive context dependent * arithmetic coding scheme * \param hBs bit stream handle * \param spectrum pointer to quantized data output. * \param lg number of quantized spectral coefficients (output by the arithmetic * decoder). * \param lg_max max number of quantized spectral coefficients. * \param arith_reset_flag flag which indicates if the spectral noiseless * context must be reset * \return void */ ARITH_CODING_ERROR CArco_DecodeArithData(CArcoData *pArcoData, HANDLE_FDK_BITSTREAM hBs, FIXP_DBL *RESTRICT spectrum, int lg, int lg_max, int arith_reset_flag); #endif /* AC_ARITH_CODER_H */ fdk-aac-2.0.2/libArithCoding/src/000077500000000000000000000000001404225310700164505ustar00rootroot00000000000000fdk-aac-2.0.2/libArithCoding/src/ac_arith_coder.cpp000066400000000000000000000776061404225310700221220ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /************************** Arithmetic coder library *************************** Author(s): Youliy Ninov, Oliver Weiss Description: Definition of Spectral Noiseless Coding Scheme based on an Arithmetic Coder in Conjunction with an Adaptive Context *******************************************************************************/ #include "ac_arith_coder.h" #define cbitsnew 16 #define stat_bitsnew 14 #define ari_q4new (((long)1 << cbitsnew) - 1) /* 0xFFFF */ #define ari_q1new (ari_q4new / 4 + 1) /* 0x4000 */ #define ari_q2new (2 * ari_q1new) /* 0x8000 */ #define ari_q3new (3 * ari_q1new) /* 0xC000 */ #define VAL_ESC 16 /* Arithmetic coder library info */ #define AC_LIB_VL0 2 #define AC_LIB_VL1 0 #define AC_LIB_VL2 0 #define AC_LIB_TITLE "Arithmetic Coder Lib" #ifdef __ANDROID__ #define AC_LIB_BUILD_DATE "" #define AC_LIB_BUILD_TIME "" #else #define AC_LIB_BUILD_DATE __DATE__ #define AC_LIB_BUILD_TIME __TIME__ #endif const SHORT ari_lsb2[3][4] = { {12571, 10569, 3696, 0}, {12661, 5700, 3751, 0}, {10827, 6884, 2929, 0}}; H_ALLOC_MEM(ArcoData, CArcoData) /*! The structure ArcoData contains 2-tuple context of previous frame.
Dimension: 1 */ C_ALLOC_MEM(ArcoData, CArcoData, 1) /* This define triggers the use of the pre-known return values of function get_pk_v2() for the cases, where parameter s is in range 0x00000000..0x0000000F. Note: These 16 bytes have been moved into the first 4 entries of ari_merged_hash_ps that are no more referenced. */ static const ULONG ari_merged_hash_ps[742] = { 0x00001044UL, 0x00003D0AUL, 0x00005350UL, 0x000074D6UL, 0x0000A49FUL, 0x0000F96EUL, 0x00111000UL, 0x01111E83UL, 0x01113146UL, 0x01114036UL, 0x01116863UL, 0x011194E9UL, 0x0111F7EEUL, 0x0112269BUL, 0x01124775UL, 0x01126DA1UL, 0x0112D912UL, 0x01131AF0UL, 0x011336DDUL, 0x01135CF5UL, 0x01139DF8UL, 0x01141A5BUL, 0x01144773UL, 0x01146CF5UL, 0x0114FDE9UL, 0x01166CF5UL, 0x0116FDE4UL, 0x01174CF3UL, 0x011FFDCFUL, 0x01211CC2UL, 0x01213B2DUL, 0x01214036UL, 0x01216863UL, 0x012194D2UL, 0x0122197FUL, 0x01223AADUL, 0x01224036UL, 0x01226878UL, 0x0122A929UL, 0x0122F4ABUL, 0x01232B2DUL, 0x012347B6UL, 0x01237DF8UL, 0x0123B929UL, 0x012417DDUL, 0x01245D76UL, 0x01249DF8UL, 0x0124F912UL, 0x01255D75UL, 0x0125FDE9UL, 0x01265D75UL, 0x012B8DF7UL, 0x01311E2AUL, 0x01313B5EUL, 0x0131687BUL, 0x01321A6DUL, 0x013237BCUL, 0x01326863UL, 0x0132F4EEUL, 0x01332B5EUL, 0x01335DA1UL, 0x01338E24UL, 0x01341A5EUL, 0x01343DB6UL, 0x01348DF8UL, 0x01351935UL, 0x01355DB7UL, 0x0135FE12UL, 0x01376DF7UL, 0x013FFE29UL, 0x01400024UL, 0x01423821UL, 0x014318F6UL, 0x01433821UL, 0x0143F8E5UL, 0x01443DA1UL, 0x01486E38UL, 0x014FF929UL, 0x01543EE3UL, 0x015FF912UL, 0x016F298CUL, 0x018A5A69UL, 0x021007F1UL, 0x02112C2CUL, 0x02114B48UL, 0x02117353UL, 0x0211F4AEUL, 0x02122FEAUL, 0x02124B48UL, 0x02127850UL, 0x0212F72EUL, 0x02133A9EUL, 0x02134036UL, 0x02138864UL, 0x021414ADUL, 0x0214379EUL, 0x02145DB6UL, 0x0214FE1FUL, 0x02166DB7UL, 0x02200DC4UL, 0x02212FEAUL, 0x022147A0UL, 0x02218369UL, 0x0221F7EEUL, 0x02222AADUL, 0x02224788UL, 0x02226863UL, 0x02229929UL, 0x0222F4ABUL, 0x02232A9EUL, 0x02234F08UL, 0x02237864UL, 0x0223A929UL, 0x022417DEUL, 0x02244F36UL, 0x02248863UL, 0x02251A74UL, 0x02256DA1UL, 0x0225FE12UL, 0x02263DB6UL, 0x02276DF7UL, 0x022FFE29UL, 0x0231186DUL, 0x023137BCUL, 0x02314020UL, 0x02319ED6UL, 0x0232196DUL, 0x023237BCUL, 0x02325809UL, 0x02329429UL, 0x023317EDUL, 0x02333F08UL, 0x02335809UL, 0x023378E4UL, 0x02341A7CUL, 0x02344221UL, 0x0234A8D3UL, 0x023514BCUL, 0x02354221UL, 0x0235F8DFUL, 0x02364861UL, 0x023FFE29UL, 0x02400024UL, 0x0241583BUL, 0x024214C8UL, 0x02424809UL, 0x02427EE6UL, 0x02431708UL, 0x02434809UL, 0x02436ED0UL, 0x02441A76UL, 0x02443821UL, 0x024458E3UL, 0x0244F91FUL, 0x02454863UL, 0x0246190AUL, 0x02464863UL, 0x024FF929UL, 0x02525ED0UL, 0x025314E1UL, 0x025348FBUL, 0x025419A1UL, 0x025458D0UL, 0x0254F4E5UL, 0x02552861UL, 0x025FF912UL, 0x02665993UL, 0x027F5A69UL, 0x029F1481UL, 0x02CF28A4UL, 0x03100AF0UL, 0x031120AAUL, 0x031147A0UL, 0x03118356UL, 0x031217ECUL, 0x03123B5EUL, 0x03124008UL, 0x03127350UL, 0x031314AAUL, 0x0313201EUL, 0x03134F08UL, 0x03136863UL, 0x03141A5EUL, 0x03143F3CUL, 0x03200847UL, 0x03212AADUL, 0x03214F20UL, 0x03218ED6UL, 0x032218ADUL, 0x032237BCUL, 0x03225809UL, 0x03229416UL, 0x032317EDUL, 0x03234F20UL, 0x03237350UL, 0x0323FA6BUL, 0x03243F08UL, 0x03246863UL, 0x0324F925UL, 0x03254221UL, 0x0325F8DFUL, 0x03264821UL, 0x032FFE29UL, 0x03311E47UL, 0x03313F08UL, 0x0331580DUL, 0x033214DEUL, 0x03323F08UL, 0x03324020UL, 0x03326350UL, 0x033294E9UL, 0x033317DEUL, 0x03333F08UL, 0x0333627BUL, 0x0333A9A9UL, 0x033417FCUL, 0x03343220UL, 0x0334627BUL, 0x0334A9A9UL, 0x0335148AUL, 0x03353220UL, 0x033588E4UL, 0x03361A4AUL, 0x03363821UL, 0x0336F8D2UL, 0x03376863UL, 0x03411939UL, 0x0341583BUL, 0x034214C8UL, 0x03424809UL, 0x03426ED0UL, 0x03431588UL, 0x03434809UL, 0x03436ED0UL, 0x03441A48UL, 0x0344480DUL, 0x03446ED0UL, 0x03451A4AUL, 0x03453809UL, 0x03455EFBUL, 0x034614CAUL, 0x03463849UL, 0x034F8924UL, 0x03500A69UL, 0x035252D0UL, 0x035314E0UL, 0x0353324DUL, 0x03535ED0UL, 0x035414E0UL, 0x0354324DUL, 0x03545ED0UL, 0x0354F4E8UL, 0x0355384DUL, 0x03555ED0UL, 0x0355F4DFUL, 0x03564350UL, 0x035969A6UL, 0x035FFA52UL, 0x036649A6UL, 0x036FFA52UL, 0x037F4F66UL, 0x039D7492UL, 0x03BF6892UL, 0x03DF8A1FUL, 0x04100B84UL, 0x04112107UL, 0x0411520DUL, 0x041214EAUL, 0x04124F20UL, 0x04131EDEUL, 0x04133F08UL, 0x04135809UL, 0x0413F42BUL, 0x04142F08UL, 0x04200847UL, 0x042121FCUL, 0x04214209UL, 0x04221407UL, 0x0422203CUL, 0x04224209UL, 0x04226350UL, 0x04231A7CUL, 0x04234209UL, 0x0423637BUL, 0x04241A7CUL, 0x04243220UL, 0x0424627BUL, 0x042514C8UL, 0x04254809UL, 0x042FF8E9UL, 0x04311E47UL, 0x04313220UL, 0x0431527BUL, 0x043214FCUL, 0x04323220UL, 0x04326250UL, 0x043315BCUL, 0x04333220UL, 0x0433527BUL, 0x04338413UL, 0x04341488UL, 0x04344809UL, 0x04346ED0UL, 0x04351F48UL, 0x0435527BUL, 0x0435F9A5UL, 0x04363809UL, 0x04375EFBUL, 0x043FF929UL, 0x04412E79UL, 0x0441427BUL, 0x044219B9UL, 0x04423809UL, 0x0442537BUL, 0x044314C8UL, 0x04432020UL, 0x0443527BUL, 0x044414CAUL, 0x04443809UL, 0x0444537BUL, 0x04448993UL, 0x0445148AUL, 0x04453809UL, 0x04455ED0UL, 0x0445F4E5UL, 0x0446384DUL, 0x045009A6UL, 0x045272D3UL, 0x045314A0UL, 0x0453324DUL, 0x04535ED0UL, 0x045415A0UL, 0x0454324DUL, 0x04545ED0UL, 0x04551F60UL, 0x0455324DUL, 0x04562989UL, 0x04564350UL, 0x045FF4D2UL, 0x04665993UL, 0x047FFF62UL, 0x048FF725UL, 0x049F44BDUL, 0x04BFB7E5UL, 0x04EF8A25UL, 0x04FFFB98UL, 0x051131F9UL, 0x051212C7UL, 0x05134209UL, 0x05200247UL, 0x05211007UL, 0x05213E60UL, 0x052212C7UL, 0x05224209UL, 0x052319BCUL, 0x05233220UL, 0x0523527BUL, 0x052414C8UL, 0x05243820UL, 0x053112F9UL, 0x05313E49UL, 0x05321439UL, 0x05323E49UL, 0x0532537BUL, 0x053314C8UL, 0x0533480DUL, 0x05337413UL, 0x05341488UL, 0x0534527BUL, 0x0534F4EBUL, 0x05353809UL, 0x05356ED0UL, 0x0535F4E5UL, 0x0536427BUL, 0x054119B9UL, 0x054212F9UL, 0x05423249UL, 0x05426ED3UL, 0x05431739UL, 0x05433249UL, 0x05435ED0UL, 0x0543F4EBUL, 0x05443809UL, 0x05445ED0UL, 0x0544F4E8UL, 0x0545324DUL, 0x054FF992UL, 0x055362D3UL, 0x0553F5ABUL, 0x05544350UL, 0x055514CAUL, 0x0555427BUL, 0x0555F4E5UL, 0x0556327BUL, 0x055FF4D2UL, 0x05665993UL, 0x05774F53UL, 0x059FF728UL, 0x05CC37FDUL, 0x05EFBA28UL, 0x05FFFB98UL, 0x061131F9UL, 0x06121407UL, 0x06133E60UL, 0x061A72E4UL, 0x06211E47UL, 0x06214E4BUL, 0x062214C7UL, 0x06223E60UL, 0x062312F9UL, 0x06233E60UL, 0x063112F9UL, 0x06313E4CUL, 0x063219B9UL, 0x06323E49UL, 0x06331439UL, 0x06333809UL, 0x06336EE6UL, 0x0633F5ABUL, 0x06343809UL, 0x0634F42BUL, 0x0635427BUL, 0x063FF992UL, 0x064342FBUL, 0x0643F4EBUL, 0x0644427BUL, 0x064524C9UL, 0x06655993UL, 0x0666170AUL, 0x066652E6UL, 0x067A6F56UL, 0x0698473DUL, 0x06CF67D2UL, 0x06EF3A26UL, 0x06FFFAD8UL, 0x071131CCUL, 0x07211307UL, 0x07222E79UL, 0x072292DCUL, 0x07234E4BUL, 0x073112F9UL, 0x07322339UL, 0x073632CBUL, 0x073FF992UL, 0x074432CBUL, 0x075549A6UL, 0x0776FF68UL, 0x07774350UL, 0x0788473DUL, 0x07CF4516UL, 0x07EF3A26UL, 0x07FFFAD8UL, 0x08222E79UL, 0x083112F9UL, 0x0834330BUL, 0x0845338BUL, 0x08756F5CUL, 0x0887F725UL, 0x08884366UL, 0x08AF649CUL, 0x08F00898UL, 0x08FFFAD8UL, 0x091111C7UL, 0x0932330BUL, 0x0945338BUL, 0x09774F7DUL, 0x0998C725UL, 0x09996416UL, 0x09EF87E5UL, 0x09FFFAD8UL, 0x0A34330BUL, 0x0A45338BUL, 0x0A77467DUL, 0x0AA9F52BUL, 0x0AAA6416UL, 0x0ABD67DFUL, 0x0AFFFA18UL, 0x0B33330BUL, 0x0B4443A6UL, 0x0B76467DUL, 0x0BB9751FUL, 0x0BBB59BDUL, 0x0BEF5892UL, 0x0BFFFAD8UL, 0x0C221339UL, 0x0C53338EUL, 0x0C76367DUL, 0x0CCAF52EUL, 0x0CCC6996UL, 0x0CFFFA18UL, 0x0D44438EUL, 0x0D64264EUL, 0x0DDCF52EUL, 0x0DDD5996UL, 0x0DFFFA18UL, 0x0E43338EUL, 0x0E68465CUL, 0x0EEE651CUL, 0x0EFFFA18UL, 0x0F33238EUL, 0x0F553659UL, 0x0F8F451CUL, 0x0FAFF8AEUL, 0x0FF00A2EUL, 0x0FFF1ACCUL, 0x0FFF33BDUL, 0x0FFF7522UL, 0x0FFFFAD8UL, 0x10002C72UL, 0x1111103EUL, 0x11121E83UL, 0x11131E9AUL, 0x1121115AUL, 0x11221170UL, 0x112316F0UL, 0x1124175DUL, 0x11311CC2UL, 0x11321182UL, 0x11331D42UL, 0x11411D48UL, 0x11421836UL, 0x11431876UL, 0x11441DF5UL, 0x1152287BUL, 0x12111903UL, 0x1212115AUL, 0x121316F0UL, 0x12211B30UL, 0x12221B30UL, 0x12231B02UL, 0x12311184UL, 0x12321D04UL, 0x12331784UL, 0x12411D39UL, 0x12412020UL, 0x12422220UL, 0x12511D89UL, 0x1252227BUL, 0x1258184AUL, 0x12832992UL, 0x1311171AUL, 0x13121B30UL, 0x1312202CUL, 0x131320AAUL, 0x132120AAUL, 0x132220ADUL, 0x13232FEDUL, 0x13312107UL, 0x13322134UL, 0x13332134UL, 0x13411D39UL, 0x13431E74UL, 0x13441834UL, 0x134812B4UL, 0x1352230BUL, 0x13611E4BUL, 0x136522E4UL, 0x141113C2UL, 0x141211C4UL, 0x143121F9UL, 0x143221F9UL, 0x143321CAUL, 0x14351D34UL, 0x14431E47UL, 0x14441E74UL, 0x144612B4UL, 0x1452230EUL, 0x14551E74UL, 0x1471130EUL, 0x151113C2UL, 0x152121F9UL, 0x153121F9UL, 0x153221F9UL, 0x15331007UL, 0x15522E4EUL, 0x15551E74UL, 0x1571130EUL, 0x161113C7UL, 0x162121F9UL, 0x163121F9UL, 0x16611E79UL, 0x16661334UL, 0x171113C7UL, 0x172121F9UL, 0x17451E47UL, 0x1771130CUL, 0x181113C7UL, 0x18211E47UL, 0x18511E4CUL, 0x1882130CUL, 0x191113C7UL, 0x19331E79UL, 0x1A111307UL, 0x1A311E79UL, 0x1F52230EUL, 0x200003C1UL, 0x20001027UL, 0x20004467UL, 0x200079E7UL, 0x2000E5EFUL, 0x21100BC0UL, 0x211129C0UL, 0x21114011UL, 0x211189E7UL, 0x2111F5EFUL, 0x21124011UL, 0x21127455UL, 0x211325C0UL, 0x21134011UL, 0x21137455UL, 0x211425C0UL, 0x21212440UL, 0x21213001UL, 0x2121F9EFUL, 0x21222540UL, 0x21226455UL, 0x2122F5EFUL, 0x21233051UL, 0x2123F56FUL, 0x21244451UL, 0x21312551UL, 0x21323451UL, 0x21332551UL, 0x21844555UL, 0x221125C0UL, 0x22113011UL, 0x2211F9EFUL, 0x22123051UL, 0x2212F9EFUL, 0x221329D1UL, 0x22212541UL, 0x22213011UL, 0x2221F9EFUL, 0x22223451UL, 0x2222F9EFUL, 0x22232551UL, 0x2223F56FUL, 0x22312551UL, 0x223229D1UL, 0x2232F56FUL, 0x22332551UL, 0x2233F56FUL, 0x22875555UL, 0x22DAB5D7UL, 0x23112BD1UL, 0x23115467UL, 0x231225D1UL, 0x232129D1UL, 0x232229D1UL, 0x2322F9EFUL, 0x23233451UL, 0x2323F9EFUL, 0x23312551UL, 0x233229D1UL, 0x2332F9EFUL, 0x2333F56FUL, 0x237FF557UL, 0x238569D5UL, 0x23D955D7UL, 0x24100BE7UL, 0x248789E7UL, 0x24E315D7UL, 0x24FFFBEFUL, 0x259869E7UL, 0x25DFF5EFUL, 0x25FFFBEFUL, 0x268789E7UL, 0x26DFA5D7UL, 0x26FFFBEFUL, 0x279649E7UL, 0x27E425D7UL, 0x27FFFBEFUL, 0x288879E7UL, 0x28EFF5EFUL, 0x28FFFBEFUL, 0x298439E7UL, 0x29F115EFUL, 0x29FFFBEFUL, 0x2A7659E7UL, 0x2AEF75D7UL, 0x2AFFFBEFUL, 0x2B7C89E7UL, 0x2BEF95D7UL, 0x2BFFFBEFUL, 0x2C6659E7UL, 0x2CD555D7UL, 0x2CFFFBEFUL, 0x2D6329E7UL, 0x2DDD55E7UL, 0x2DFFFBEBUL, 0x2E8479D7UL, 0x2EEE35E7UL, 0x2EFFFBEFUL, 0x2F5459E7UL, 0x2FCF85D7UL, 0x2FFEFBEBUL, 0x2FFFA5EFUL, 0x2FFFEBEFUL, 0x30001AE7UL, 0x30002001UL, 0x311129C0UL, 0x31221015UL, 0x31232000UL, 0x31332451UL, 0x32112540UL, 0x32131027UL, 0x32212440UL, 0x33452455UL, 0x4000F9D7UL, 0x4122F9D7UL, 0x43F65555UL, 0x43FFF5D7UL, 0x44F55567UL, 0x44FFF5D7UL, 0x45F00557UL, 0x45FFF5D7UL, 0x46F659D7UL, 0x471005E7UL, 0x47F449E7UL, 0x481005E7UL, 0x48EFA9D5UL, 0x48FFF5EFUL, 0x49F449E7UL, 0x49FFF5EFUL, 0x4AEA79E7UL, 0x4AFFF5EFUL, 0x4BE9C9D5UL, 0x4BFFF5EFUL, 0x4CE549E7UL, 0x4CFFF5EFUL, 0x4DE359E7UL, 0x4DFFF5D7UL, 0x4EE469E7UL, 0x4EFFF5D7UL, 0x4FEF39E7UL, 0x4FFFF5EFUL, 0x6000F9E7UL, 0x69FFF557UL, 0x6FFFF9D7UL, 0x811009D7UL, 0x8EFFF555UL, 0xFFFFF9E7UL}; static const SHORT ari_pk[64][17] = { {708, 706, 579, 569, 568, 567, 479, 469, 297, 138, 97, 91, 72, 52, 38, 34, 0}, {7619, 6917, 6519, 6412, 5514, 5003, 4683, 4563, 3907, 3297, 3125, 3060, 2904, 2718, 2631, 2590, 0}, {7263, 4888, 4810, 4803, 1889, 415, 335, 327, 195, 72, 52, 49, 36, 20, 15, 14, 0}, {3626, 2197, 2188, 2187, 582, 57, 47, 46, 30, 12, 9, 8, 6, 4, 3, 2, 0}, {7806, 5541, 5451, 5441, 2720, 834, 691, 674, 487, 243, 179, 167, 139, 98, 77, 70, 0}, {6684, 4101, 4058, 4055, 1748, 426, 368, 364, 322, 257, 235, 232, 228, 222, 217, 215, 0}, {9162, 5964, 5831, 5819, 3269, 866, 658, 638, 535, 348, 258, 244, 234, 214, 195, 186, 0}, {10638, 8491, 8365, 8351, 4418, 2067, 1859, 1834, 1190, 601, 495, 478, 356, 217, 174, 164, 0}, {13389, 10514, 10032, 9961, 7166, 3488, 2655, 2524, 2015, 1140, 760, 672, 585, 426, 325, 283, 0}, {14861, 12788, 12115, 11952, 9987, 6657, 5323, 4984, 4324, 3001, 2205, 1943, 1764, 1394, 1115, 978, 0}, {12876, 10004, 9661, 9610, 7107, 3435, 2711, 2595, 2257, 1508, 1059, 952, 893, 753, 609, 538, 0}, {15125, 13591, 13049, 12874, 11192, 8543, 7406, 7023, 6291, 4922, 4104, 3769, 3465, 2890, 2486, 2275, 0}, {14574, 13106, 12731, 12638, 10453, 7947, 7233, 7037, 6031, 4618, 4081, 3906, 3465, 2802, 2476, 2349, 0}, {15070, 13179, 12517, 12351, 10742, 7657, 6200, 5825, 5264, 3998, 3014, 2662, 2510, 2153, 1799, 1564, 0}, {15542, 14466, 14007, 13844, 12489, 10409, 9481, 9132, 8305, 6940, 6193, 5867, 5458, 4743, 4291, 4047, 0}, {15165, 14384, 14084, 13934, 12911, 11485, 10844, 10513, 10002, 8993, 8380, 8051, 7711, 7036, 6514, 6233, 0}, {15642, 14279, 13625, 13393, 12348, 9971, 8405, 7858, 7335, 6119, 4918, 4376, 4185, 3719, 3231, 2860, 0}, {13408, 13407, 11471, 11218, 11217, 11216, 9473, 9216, 6480, 3689, 2857, 2690, 2256, 1732, 1405, 1302, 0}, {16098, 15584, 15191, 14931, 14514, 13578, 12703, 12103, 11830, 11172, 10475, 9867, 9695, 9281, 8825, 8389, 0}, {15844, 14873, 14277, 13996, 13230, 11535, 10205, 9543, 9107, 8086, 7085, 6419, 6214, 5713, 5195, 4731, 0}, {16131, 15720, 15443, 15276, 14848, 13971, 13314, 12910, 12591, 11874, 11225, 10788, 10573, 10077, 9585, 9209, 0}, {16331, 16330, 12283, 11435, 11434, 11433, 8725, 8049, 6065, 4138, 3187, 2842, 2529, 2171, 1907, 1745, 0}, {16011, 15292, 14782, 14528, 14008, 12767, 11556, 10921, 10591, 9759, 8813, 8043, 7855, 7383, 6863, 6282, 0}, {16380, 16379, 15159, 14610, 14609, 14608, 12859, 12111, 11046, 9536, 8348, 7713, 7216, 6533, 5964, 5546, 0}, {16367, 16333, 16294, 16253, 16222, 16143, 16048, 15947, 15915, 15832, 15731, 15619, 15589, 15512, 15416, 15310, 0}, {15967, 15319, 14937, 14753, 14010, 12638, 11787, 11360, 10805, 9706, 8934, 8515, 8166, 7456, 6911, 6575, 0}, {4906, 3005, 2985, 2984, 875, 102, 83, 81, 47, 17, 12, 11, 8, 5, 4, 3, 0}, {7217, 4346, 4269, 4264, 1924, 428, 340, 332, 280, 203, 179, 175, 171, 164, 159, 157, 0}, {16010, 15415, 15032, 14805, 14228, 13043, 12168, 11634, 11265, 10419, 9645, 9110, 8892, 8378, 7850, 7437, 0}, {8573, 5218, 5046, 5032, 2787, 771, 555, 533, 443, 286, 218, 205, 197, 181, 168, 162, 0}, {11474, 8095, 7822, 7796, 4632, 1443, 1046, 1004, 748, 351, 218, 194, 167, 121, 93, 83, 0}, {16152, 15764, 15463, 15264, 14925, 14189, 13536, 13070, 12846, 12314, 11763, 11277, 11131, 10777, 10383, 10011, 0}, {14187, 11654, 11043, 10919, 8498, 4885, 3778, 3552, 2947, 1835, 1283, 1134, 998, 749, 585, 514, 0}, {14162, 11527, 10759, 10557, 8601, 5417, 4105, 3753, 3286, 2353, 1708, 1473, 1370, 1148, 959, 840, 0}, {16205, 15902, 15669, 15498, 15213, 14601, 14068, 13674, 13463, 12970, 12471, 12061, 11916, 11564, 11183, 10841, 0}, {15043, 12972, 12092, 11792, 10265, 7446, 5934, 5379, 4883, 3825, 3036, 2647, 2507, 2185, 1901, 1699, 0}, {15320, 13694, 12782, 12352, 11191, 8936, 7433, 6671, 6255, 5366, 4622, 4158, 4020, 3712, 3420, 3198, 0}, {16255, 16020, 15768, 15600, 15416, 14963, 14440, 14006, 13875, 13534, 13137, 12697, 12602, 12364, 12084, 11781, 0}, {15627, 14503, 13906, 13622, 12557, 10527, 9269, 8661, 8117, 6933, 5994, 5474, 5222, 4664, 4166, 3841, 0}, {16366, 16365, 14547, 14160, 14159, 14158, 11969, 11473, 8735, 6147, 4911, 4530, 3865, 3180, 2710, 2473, 0}, {16257, 16038, 15871, 15754, 15536, 15071, 14673, 14390, 14230, 13842, 13452, 13136, 13021, 12745, 12434, 12154, 0}, {15855, 14971, 14338, 13939, 13239, 11782, 10585, 9805, 9444, 8623, 7846, 7254, 7079, 6673, 6262, 5923, 0}, {9492, 6318, 6197, 6189, 3004, 652, 489, 477, 333, 143, 96, 90, 78, 60, 50, 47, 0}, {16313, 16191, 16063, 15968, 15851, 15590, 15303, 15082, 14968, 14704, 14427, 14177, 14095, 13899, 13674, 13457, 0}, {8485, 5473, 5389, 5383, 2411, 494, 386, 377, 278, 150, 117, 112, 103, 89, 81, 78, 0}, {10497, 7154, 6959, 6943, 3788, 1004, 734, 709, 517, 238, 152, 138, 120, 90, 72, 66, 0}, {16317, 16226, 16127, 16040, 15955, 15762, 15547, 15345, 15277, 15111, 14922, 14723, 14671, 14546, 14396, 14239, 0}, {16382, 16381, 15858, 15540, 15539, 15538, 14704, 14168, 13768, 13092, 12452, 11925, 11683, 11268, 10841, 10460, 0}, {5974, 3798, 3758, 3755, 1275, 205, 166, 162, 95, 35, 26, 24, 18, 11, 8, 7, 0}, {3532, 2258, 2246, 2244, 731, 135, 118, 115, 87, 45, 36, 34, 29, 21, 17, 16, 0}, {7466, 4882, 4821, 4811, 2476, 886, 788, 771, 688, 531, 469, 457, 437, 400, 369, 361, 0}, {9580, 5772, 5291, 5216, 3444, 1496, 1025, 928, 806, 578, 433, 384, 366, 331, 296, 273, 0}, {10692, 7730, 7543, 7521, 4679, 1746, 1391, 1346, 1128, 692, 495, 458, 424, 353, 291, 268, 0}, {11040, 7132, 6549, 6452, 4377, 1875, 1253, 1130, 958, 631, 431, 370, 346, 296, 253, 227, 0}, {12687, 9332, 8701, 8585, 6266, 3093, 2182, 2004, 1683, 1072, 712, 608, 559, 458, 373, 323, 0}, {13429, 9853, 8860, 8584, 6806, 4039, 2862, 2478, 2239, 1764, 1409, 1224, 1178, 1077, 979, 903, 0}, {14685, 12163, 11061, 10668, 9101, 6345, 4871, 4263, 3908, 3200, 2668, 2368, 2285, 2106, 1942, 1819, 0}, {13295, 11302, 10999, 10945, 7947, 5036, 4490, 4385, 3391, 2185, 1836, 1757, 1424, 998, 833, 785, 0}, {4992, 2993, 2972, 2970, 1269, 575, 552, 549, 530, 505, 497, 495, 493, 489, 486, 485, 0}, {15419, 13862, 13104, 12819, 11429, 8753, 7220, 6651, 6020, 4667, 3663, 3220, 2995, 2511, 2107, 1871, 0}, {12468, 9263, 8912, 8873, 5758, 2193, 1625, 1556, 1187, 589, 371, 330, 283, 200, 149, 131, 0}, {15870, 15076, 14615, 14369, 13586, 12034, 10990, 10423, 9953, 8908, 8031, 7488, 7233, 6648, 6101, 5712, 0}, {1693, 978, 976, 975, 194, 18, 16, 15, 11, 7, 6, 5, 4, 3, 2, 1, 0}, {7992, 5218, 5147, 5143, 2152, 366, 282, 276, 173, 59, 38, 35, 27, 16, 11, 10, 0}}; typedef struct { int low; int high; int vobf; } Tastat; static inline INT mul_sbc_14bits(INT r, INT c) { return (((INT)r) * ((INT)c)) >> stat_bitsnew; } static inline INT ari_decode_14bits(HANDLE_FDK_BITSTREAM hBs, Tastat *s, const SHORT *RESTRICT c_freq, int cfl) { INT symbol; INT low, high, range, value; INT c; const SHORT *p; low = s->low; high = s->high; value = s->vobf; range = high - low + 1; c = (((int)(value - low + 1)) << stat_bitsnew) - ((int)1); p = (const SHORT *)(c_freq - 1); if (cfl == (VAL_ESC + 1)) { /* In 50% of all cases, the first entry is the right one, so we check it * prior to all others */ if ((p[1] * range) > c) { p += 1; if ((p[8] * range) > c) { p += 8; } if ((p[4] * range) > c) { p += 4; } if ((p[2] * range) > c) { p += 2; } if ((p[1] * range) > c) { p += 1; } } } else if (cfl == 4) { if ((p[2] * range) > c) { p += 2; } if ((p[1] * range) > c) { p += 1; } } else if (cfl == 2) { if ((p[1] * range) > c) { p += 1; } } else if (cfl == 27) { const SHORT *p_24 = p + 24; if ((p[16] * range) > c) { p += 16; } if ((p[8] * range) > c) { p += 8; } if (p != p_24) { if ((p[4] * range) > c) { p += 4; } } if ((p[2] * range) > c) { p += 2; } if (p != &p_24[2]) { if ((p[1] * range) > c) { p += 1; } } } symbol = (INT)(p - (const SHORT *)(c_freq - 1)); if (symbol) { high = low + mul_sbc_14bits(range, c_freq[symbol - 1]) - 1; } low += mul_sbc_14bits(range, c_freq[symbol]); USHORT us_high = (USHORT)high; USHORT us_low = (USHORT)low; while (1) { if (us_high & 0x8000) { if (!(us_low & 0x8000)) { if (us_low & 0x4000 && !(us_high & 0x4000)) { us_low -= 0x4000; us_high -= 0x4000; value -= 0x4000; } else break; } } us_low = us_low << 1; us_high = (us_high << 1) | 1; value = (value << 1) | FDKreadBit(hBs); } s->low = (int)us_low; s->high = (int)us_high; s->vobf = value & 0xFFFF; return symbol; } static inline void copyTableAmrwbArith2(UCHAR tab[], int sizeIn, int sizeOut) { int i; int j; int k = 2; tab += 2; if (sizeIn < sizeOut) { tab[sizeOut + 0] = tab[sizeIn + 0]; tab[sizeOut + 1] = tab[sizeIn + 1]; if (sizeIn < (sizeOut >> 2)) { k = 8; } else if (sizeIn == (sizeOut >> 2)) { k = 4; } i = sizeOut - 1; j = sizeIn - 1; for (; i >= 0; j--) { UCHAR tq_data0 = tab[j]; for (int l = (k >> 1); l > 0; l--) { tab[i--] = tq_data0; tab[i--] = tq_data0; } } } else { if (sizeOut < (sizeIn >> 2)) { k = 8; } else if (sizeOut == (sizeIn >> 2)) { k = 4; } for (i = 0, j = 0; i < sizeOut; j += k) { UCHAR tq_data0 = tab[j]; tab[i++] = tq_data0; } tab[sizeOut + 0] = tab[sizeIn + 0]; tab[sizeOut + 1] = tab[sizeIn + 1]; } } static inline ULONG get_pk_v2(ULONG s) { const ULONG *p = ari_merged_hash_ps; ULONG s12 = (fMax((UINT)s, (UINT)1) << 12) - 1; if (s12 > p[485]) { p += 486; /* 742 - 256 = 486 */ } else { if (s12 > p[255]) p += 256; } if (s12 > p[127]) { p += 128; } if (s12 > p[63]) { p += 64; } if (s12 > p[31]) { p += 32; } if (s12 > p[15]) { p += 16; } if (s12 > p[7]) { p += 8; } if (s12 > p[3]) { p += 4; } if (s12 > p[1]) { p += 2; } ULONG j = p[0]; if (s12 > j) j = p[1]; if (s != (j >> 12)) j >>= 6; return (j & 0x3F); } static ARITH_CODING_ERROR decode2(HANDLE_FDK_BITSTREAM bbuf, UCHAR *RESTRICT c_prev, FIXP_DBL *RESTRICT pSpectralCoefficient, INT n, INT nt) { Tastat as; int i, l, r; INT lev, esc_nb, pki; USHORT state_inc; UINT s; ARITH_CODING_ERROR ErrorStatus = ARITH_CODER_OK; int c_3 = 0; /* context of current frame 3 time steps ago */ int c_2 = 0; /* context of current frame 2 time steps ago */ int c_1 = 0; /* context of current frame 1 time steps ago */ int c_0 = 1; /* context of current frame to be calculated */ /* ari_start_decoding_14bits */ as.low = 0; as.high = ari_q4new; as.vobf = FDKreadBits(bbuf, cbitsnew); /* arith_map_context */ state_inc = c_prev[0] << 12; for (i = 0; i < n; i++) { /* arith_get_context */ s = state_inc >> 8; s = s + (c_prev[i + 1] << 8); s = (s << 4) + c_1; state_inc = s; if (i > 3) { /* Cumulative amplitude below 2 */ if ((c_1 + c_2 + c_3) < 5) { s += 0x10000; } } /* MSBs decoding */ for (lev = esc_nb = 0;;) { pki = get_pk_v2(s + (esc_nb << (VAL_ESC + 1))); r = ari_decode_14bits(bbuf, &as, ari_pk[pki], VAL_ESC + 1); if (r < VAL_ESC) { break; } lev++; if (lev > 23) return ARITH_CODER_ERROR; if (esc_nb < 7) { esc_nb++; } } /* Stop symbol */ if (r == 0) { if (esc_nb > 0) { break; /* Stop symbol */ } c_0 = 1; } else /* if (r==0) */ { INT b = r >> 2; INT a = r & 0x3; /* LSBs decoding */ for (l = 0; l < lev; l++) { { int pidx = (a == 0) ? 1 : ((b == 0) ? 0 : 2); r = ari_decode_14bits(bbuf, &as, ari_lsb2[pidx], 4); } a = (a << 1) | (r & 1); b = (b << 1) | (r >> 1); } pSpectralCoefficient[2 * i] = (FIXP_DBL)a; pSpectralCoefficient[2 * i + 1] = (FIXP_DBL)b; c_0 = a + b + 1; if (c_0 > 0xF) { c_0 = 0xF; } } /* endif (r==0) */ /* arith_update_context */ c_3 = c_2; c_2 = c_1; c_1 = c_0; c_prev[i] = (UCHAR)c_0; } /* for (i=0; i> (bits - 1))) { pSpectralCoefficient[2 * i] = -pSpectralCoefficient[2 * i]; } if (pSpectralCoefficient[2 * i + 1] != (FIXP_DBL)0 && !(r & 1)) { pSpectralCoefficient[2 * i + 1] = -pSpectralCoefficient[2 * i + 1]; } } } FDKmemset(&c_prev[i], 1, sizeof(c_prev[0]) * (nt - i)); return ErrorStatus; } CArcoData *CArco_Create(void) { return GetArcoData(); } void CArco_Destroy(CArcoData *pArcoData) { FreeArcoData(&pArcoData); } ARITH_CODING_ERROR CArco_DecodeArithData(CArcoData *pArcoData, HANDLE_FDK_BITSTREAM hBs, FIXP_DBL *RESTRICT mdctSpectrum, int lg, int lg_max, int arith_reset_flag) { ARITH_CODING_ERROR ErrorStatus = ARITH_CODER_OK; /* Check lg and lg_max consistency. */ if (lg_max < lg) { return ARITH_CODER_ERROR; } FDKmemclear(mdctSpectrum, lg_max * sizeof(FIXP_DBL)); /* arith_map_context */ if (arith_reset_flag) { FDKmemclear(pArcoData->c_prev, sizeof(pArcoData->c_prev[0]) * ((lg_max / 2) + 4)); } else { if (lg_max != pArcoData->m_numberLinesPrev) { if (pArcoData->m_numberLinesPrev == 0) { /* Cannot decode without a valid AC context */ return ARITH_CODER_ERROR; } /* short-to-long or long-to-short block transition */ /* Current length differs compared to previous - perform up/downmix of * m_qbuf */ copyTableAmrwbArith2(pArcoData->c_prev, pArcoData->m_numberLinesPrev >> 1, lg_max >> 1); } } pArcoData->m_numberLinesPrev = lg_max; if (lg > 0) { ErrorStatus = decode2(hBs, pArcoData->c_prev + 2, mdctSpectrum, lg >> 1, lg_max >> 1); } else { FDKmemset(&pArcoData->c_prev[2], 1, sizeof(pArcoData->c_prev[2]) * (lg_max >> 1)); } if ((INT)FDKgetValidBits(hBs) < 0) { return ARITH_CODER_ERROR; } return ErrorStatus; } fdk-aac-2.0.2/libDRCdec/000077500000000000000000000000001404225310700145525ustar00rootroot00000000000000fdk-aac-2.0.2/libDRCdec/include/000077500000000000000000000000001404225310700161755ustar00rootroot00000000000000fdk-aac-2.0.2/libDRCdec/include/FDK_drcDecLib.h000066400000000000000000000303761404225310700206560ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /************************* MPEG-D DRC decoder library ************************** Author(s): Bernhard Neugebauer Description: MPEG-D DRC Decoder *******************************************************************************/ #ifndef FDK_DRCDECLIB_H #define FDK_DRCDECLIB_H #include "FDK_bitstream.h" #include "FDK_audio.h" #include "common_fix.h" /* DRC decoder according to ISO/IEC 23003-4 (MPEG-D DRC) */ /* including ISO/IEC 23003-4/AMD1 (Amendment 1) */ #ifdef __cplusplus extern "C" { #endif #define DRC_DEC_LOUDNESS_NOT_PRESENT (LONG)0x7FFFFFFE typedef struct s_drc_decoder* HANDLE_DRC_DECODER; typedef struct s_uni_drc_interface* HANDLE_UNI_DRC_INTERFACE; typedef struct s_selection_process_output* HANDLE_SEL_PROC_OUTPUT; typedef enum { DRC_DEC_SELECTION = 0x1, /* DRC decoder instance for DRC set selection only */ DRC_DEC_GAIN = 0x2, /* DRC decoder instance for applying DRC only */ DRC_DEC_ALL = 0x3 /* DRC decoder with full functionality */ } DRC_DEC_FUNCTIONAL_RANGE; typedef enum { /* get and set userparams */ DRC_DEC_BOOST, DRC_DEC_COMPRESS, /* set only userparams */ DRC_DEC_LOUDNESS_NORMALIZATION_ON, DRC_DEC_TARGET_LOUDNESS, /**< target loudness in dB, with exponent e = 7 */ DRC_DEC_EFFECT_TYPE, DRC_DEC_EFFECT_TYPE_FALLBACK_CODE, DRC_DEC_LOUDNESS_MEASUREMENT_METHOD, DRC_DEC_ALBUM_MODE, /* set only system (not user) parameters */ DRC_DEC_DOWNMIX_ID, DRC_DEC_TARGET_CHANNEL_COUNT_REQUESTED, /**< number of output channels notified to FDK_drcDecLib for choosing an appropriate downmixInstruction */ DRC_DEC_BASE_CHANNEL_COUNT, DRC_DEC_FRAME_SIZE, DRC_DEC_SAMPLE_RATE, /* get only system parameters */ DRC_DEC_IS_MULTIBAND_DRC_1, DRC_DEC_IS_MULTIBAND_DRC_2, DRC_DEC_IS_ACTIVE, /**< MPEG-D DRC payload is present and at least one of Dynamic Range Control (DRC) or Loudness Normalization (LN) is activated */ DRC_DEC_TARGET_CHANNEL_COUNT_SELECTED, /**< number of output channels if appropriate downmixInstruction exists */ DRC_DEC_OUTPUT_LOUDNESS /**< output loudness in dB, with exponent e = 7, or DRC_DEC_LOUDNESS_NOT_PRESENT if no loudness is contained in the bitstream */ } DRC_DEC_USERPARAM; typedef enum { DRC_DEC_OK = 0, DRC_DEC_NOT_OK = -10000, DRC_DEC_OUT_OF_MEMORY, DRC_DEC_NOT_OPENED, DRC_DEC_NOT_READY, DRC_DEC_PARAM_OUT_OF_RANGE, DRC_DEC_INVALID_PARAM, DRC_DEC_UNSUPPORTED_FUNCTION } DRC_DEC_ERROR; typedef enum { DRC_DEC_TEST_TIME_DOMAIN = -100, DRC_DEC_TEST_QMF_DOMAIN, DRC_DEC_TEST_STFT_DOMAIN, DRC_DEC_CODEC_MODE_UNDEFINED = -1, DRC_DEC_MPEG_4_AAC, DRC_DEC_MPEG_D_USAC, DRC_DEC_MPEG_H_3DA } DRC_DEC_CODEC_MODE; /* Apply only DRC sets dedicated to processing location. DRC1: before downmix DRC2: before or after downmix (AMD1: only after downmix) DRC3: after downmix */ typedef enum { DRC_DEC_DRC1, DRC_DEC_DRC1_DRC2, DRC_DEC_DRC2, DRC_DEC_DRC3, DRC_DEC_DRC2_DRC3 } DRC_DEC_LOCATION; DRC_DEC_ERROR FDK_drcDec_Open(HANDLE_DRC_DECODER* phDrcDec, const DRC_DEC_FUNCTIONAL_RANGE functionalRange); DRC_DEC_ERROR FDK_drcDec_SetCodecMode(HANDLE_DRC_DECODER hDrcDec, const DRC_DEC_CODEC_MODE codecMode); DRC_DEC_ERROR FDK_drcDec_Init(HANDLE_DRC_DECODER hDrcDec, const int frameSize, const int sampleRate, const int baseChannelCount); DRC_DEC_ERROR FDK_drcDec_Close(HANDLE_DRC_DECODER* phDrcDec); /* set single user request */ DRC_DEC_ERROR FDK_drcDec_SetParam(HANDLE_DRC_DECODER hDrcDec, const DRC_DEC_USERPARAM requestType, const FIXP_DBL requestValue); LONG FDK_drcDec_GetParam(HANDLE_DRC_DECODER hDrcDec, const DRC_DEC_USERPARAM requestType); DRC_DEC_ERROR FDK_drcDec_SetInterfaceParameters(HANDLE_DRC_DECODER hDrcDec, HANDLE_UNI_DRC_INTERFACE uniDrcInterface); DRC_DEC_ERROR FDK_drcDec_SetSelectionProcessMpeghParameters_simple( HANDLE_DRC_DECODER hDrcDec, const int groupPresetIdRequested, const int numGroupIdsRequested, const int* groupIdsRequested); DRC_DEC_ERROR FDK_drcDec_SetDownmixInstructions(HANDLE_DRC_DECODER hDrcDec, const int numDowmixId, const int* downmixId, const int* targetLayout, const int* targetChannelCount); void FDK_drcDec_SetSelectionProcessOutput( HANDLE_DRC_DECODER hDrcDec, HANDLE_SEL_PROC_OUTPUT hSelProcOutput); HANDLE_SEL_PROC_OUTPUT FDK_drcDec_GetSelectionProcessOutput(HANDLE_DRC_DECODER hDrcDec); LONG /* FIXP_DBL, e = 7 */ FDK_drcDec_GetGroupLoudness(HANDLE_SEL_PROC_OUTPUT hSelProcOutput, const int groupID, int* groupLoudnessAvailable); void FDK_drcDec_SetChannelGains(HANDLE_DRC_DECODER hDrcDec, const int numChannels, const int frameSize, FIXP_DBL* channelGainDb, FIXP_DBL* audioBuffer, const int audioBufferChannelOffset); DRC_DEC_ERROR FDK_drcDec_ReadUniDrcConfig(HANDLE_DRC_DECODER hDrcDec, HANDLE_FDK_BITSTREAM hBitstream); DRC_DEC_ERROR FDK_drcDec_ReadLoudnessInfoSet(HANDLE_DRC_DECODER hDrcDec, HANDLE_FDK_BITSTREAM hBitstream); DRC_DEC_ERROR FDK_drcDec_ReadLoudnessBox(HANDLE_DRC_DECODER hDrcDec, HANDLE_FDK_BITSTREAM hBitstream); DRC_DEC_ERROR FDK_drcDec_ReadDownmixInstructions_Box(HANDLE_DRC_DECODER hDrcDec, HANDLE_FDK_BITSTREAM hBitstream); DRC_DEC_ERROR FDK_drcDec_ReadUniDrcInstructions_Box(HANDLE_DRC_DECODER hDrcDec, HANDLE_FDK_BITSTREAM hBitstream); DRC_DEC_ERROR FDK_drcDec_ReadUniDrcCoefficients_Box(HANDLE_DRC_DECODER hDrcDec, HANDLE_FDK_BITSTREAM hBitstream); DRC_DEC_ERROR FDK_drcDec_ReadUniDrcGain(HANDLE_DRC_DECODER hDrcDec, HANDLE_FDK_BITSTREAM hBitstream); /* either call FDK_drcDec_ReadUniDrcConfig, FDK_drcDec_ReadLoudnessInfoSet and FDK_drcDec_ReadUniDrcGain separately, or call FDK_drcDec_ReadUniDrc */ DRC_DEC_ERROR FDK_drcDec_ReadUniDrc(HANDLE_DRC_DECODER hDrcDec, HANDLE_FDK_BITSTREAM hBitstream); /* calling sequence: FDK_drcDec_Read...() FDK_drcDec_SetChannelGains() FDK_drcDec_Preprocess() FDK_drcDec_Process...() */ DRC_DEC_ERROR FDK_drcDec_Preprocess(HANDLE_DRC_DECODER hDrcDec); DRC_DEC_ERROR FDK_drcDec_ProcessTime(HANDLE_DRC_DECODER hDrcDec, const int delaySamples, const DRC_DEC_LOCATION drcLocation, const int channelOffset, const int drcChannelOffset, const int numChannelsProcessed, FIXP_DBL* realBuffer, const int timeDataChannelOffset); DRC_DEC_ERROR FDK_drcDec_ProcessFreq(HANDLE_DRC_DECODER hDrcDec, const int delaySamples, const DRC_DEC_LOCATION drcLocation, const int channelOffset, const int drcChannelOffset, const int numChannelsProcessed, const int processSingleTimeslot, FIXP_DBL** realBuffer, FIXP_DBL** imagBuffer); DRC_DEC_ERROR FDK_drcDec_ApplyDownmix(HANDLE_DRC_DECODER hDrcDec, int* reverseInChannelMap, int* reverseOutChannelMap, FIXP_DBL* realBuffer, int* pNChannels); /* Get library info for this module. */ DRC_DEC_ERROR FDK_drcDec_GetLibInfo(LIB_INFO* info); #ifdef __cplusplus } #endif #endif fdk-aac-2.0.2/libDRCdec/src/000077500000000000000000000000001404225310700153415ustar00rootroot00000000000000fdk-aac-2.0.2/libDRCdec/src/FDK_drcDecLib.cpp000066400000000000000000000770351404225310700203600ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /************************* MPEG-D DRC decoder library ************************** Author(s): Bernhard Neugebauer Description: MPEG-D DRC Decoder *******************************************************************************/ #include "drcDec_reader.h" #include "drcDec_gainDecoder.h" #include "FDK_drcDecLib.h" #include "drcDec_selectionProcess.h" #include "drcDec_tools.h" /* Decoder library info */ #define DRCDEC_LIB_VL0 2 #define DRCDEC_LIB_VL1 1 #define DRCDEC_LIB_VL2 0 #define DRCDEC_LIB_TITLE "MPEG-D DRC Decoder Lib" #ifdef SUPPRESS_BUILD_DATE_INFO #define DRCDEC_LIB_BUILD_DATE "" #define DRCDEC_LIB_BUILD_TIME "" #else #define DRCDEC_LIB_BUILD_DATE __DATE__ #define DRCDEC_LIB_BUILD_TIME __TIME__ #endif typedef enum { DRC_DEC_NOT_INITIALIZED = 0, DRC_DEC_INITIALIZED, DRC_DEC_NEW_GAIN_PAYLOAD, DRC_DEC_INTERPOLATION_PREPARED } DRC_DEC_STATUS; struct s_drc_decoder { DRC_DEC_CODEC_MODE codecMode; DRC_DEC_FUNCTIONAL_RANGE functionalRange; DRC_DEC_STATUS status; /* handles of submodules */ HANDLE_DRC_GAIN_DECODER hGainDec; HANDLE_DRC_SELECTION_PROCESS hSelectionProc; int selProcInputDiff; /* data structs */ UNI_DRC_CONFIG uniDrcConfig; LOUDNESS_INFO_SET loudnessInfoSet; UNI_DRC_GAIN uniDrcGain; SEL_PROC_OUTPUT selProcOutput; } DRC_DECODER; static int _getGainStatus(HANDLE_UNI_DRC_GAIN hUniDrcGain) { return hUniDrcGain->status; } static int isResetNeeded(HANDLE_DRC_DECODER hDrcDec, const SEL_PROC_OUTPUT oldSelProcOutput) { int i, resetNeeded = 0; if (hDrcDec->selProcOutput.numSelectedDrcSets != oldSelProcOutput.numSelectedDrcSets) { resetNeeded = 1; } else { for (i = 0; i < hDrcDec->selProcOutput.numSelectedDrcSets; i++) { if (hDrcDec->selProcOutput.selectedDrcSetIds[i] != oldSelProcOutput.selectedDrcSetIds[i]) resetNeeded = 1; if (hDrcDec->selProcOutput.selectedDownmixIds[i] != oldSelProcOutput.selectedDownmixIds[i]) resetNeeded = 1; } } if (hDrcDec->selProcOutput.boost != oldSelProcOutput.boost) resetNeeded = 1; if (hDrcDec->selProcOutput.compress != oldSelProcOutput.compress) resetNeeded = 1; /* Note: Changes in downmix matrix are not caught, as they don't affect the * DRC gain decoder */ return resetNeeded; } static void startSelectionProcess(HANDLE_DRC_DECODER hDrcDec) { int uniDrcConfigHasChanged = 0; SEL_PROC_OUTPUT oldSelProcOutput = hDrcDec->selProcOutput; if (!hDrcDec->status) return; if (hDrcDec->functionalRange & DRC_DEC_SELECTION) { uniDrcConfigHasChanged = hDrcDec->uniDrcConfig.diff; if (hDrcDec->uniDrcConfig.diff || hDrcDec->loudnessInfoSet.diff || hDrcDec->selProcInputDiff) { /* in case of an error, signal that selection process was not successful */ hDrcDec->selProcOutput.numSelectedDrcSets = 0; drcDec_SelectionProcess_Process( hDrcDec->hSelectionProc, &(hDrcDec->uniDrcConfig), &(hDrcDec->loudnessInfoSet), &(hDrcDec->selProcOutput)); hDrcDec->selProcInputDiff = 0; hDrcDec->uniDrcConfig.diff = 0; hDrcDec->loudnessInfoSet.diff = 0; } } if (hDrcDec->functionalRange & DRC_DEC_GAIN) { if (isResetNeeded(hDrcDec, oldSelProcOutput) || uniDrcConfigHasChanged) { drcDec_GainDecoder_Config(hDrcDec->hGainDec, &(hDrcDec->uniDrcConfig), hDrcDec->selProcOutput.numSelectedDrcSets, hDrcDec->selProcOutput.selectedDrcSetIds, hDrcDec->selProcOutput.selectedDownmixIds); } } } DRC_DEC_ERROR FDK_drcDec_Open(HANDLE_DRC_DECODER* phDrcDec, const DRC_DEC_FUNCTIONAL_RANGE functionalRange) { DRC_ERROR dErr = DE_OK; DRCDEC_SELECTION_PROCESS_RETURN sErr = DRCDEC_SELECTION_PROCESS_NO_ERROR; HANDLE_DRC_DECODER hDrcDec; *phDrcDec = (HANDLE_DRC_DECODER)FDKcalloc(1, sizeof(DRC_DECODER)); if (!*phDrcDec) return DRC_DEC_OUT_OF_MEMORY; hDrcDec = *phDrcDec; hDrcDec->functionalRange = functionalRange; hDrcDec->status = DRC_DEC_NOT_INITIALIZED; hDrcDec->codecMode = DRC_DEC_CODEC_MODE_UNDEFINED; if (hDrcDec->functionalRange & DRC_DEC_SELECTION) { sErr = drcDec_SelectionProcess_Create(&(hDrcDec->hSelectionProc)); if (sErr) return DRC_DEC_OUT_OF_MEMORY; sErr = drcDec_SelectionProcess_Init(hDrcDec->hSelectionProc); if (sErr) return DRC_DEC_NOT_OK; hDrcDec->selProcInputDiff = 1; } if (hDrcDec->functionalRange & DRC_DEC_GAIN) { dErr = drcDec_GainDecoder_Open(&(hDrcDec->hGainDec)); if (dErr) return DRC_DEC_OUT_OF_MEMORY; } return DRC_DEC_OK; } DRC_DEC_ERROR FDK_drcDec_SetCodecMode(HANDLE_DRC_DECODER hDrcDec, const DRC_DEC_CODEC_MODE codecMode) { DRC_ERROR dErr = DE_OK; DRCDEC_SELECTION_PROCESS_RETURN sErr = DRCDEC_SELECTION_PROCESS_NO_ERROR; if (hDrcDec == NULL) return DRC_DEC_NOT_OPENED; if (hDrcDec->codecMode == DRC_DEC_CODEC_MODE_UNDEFINED) { /* Set codec mode, if it is set for the first time */ hDrcDec->codecMode = codecMode; if (hDrcDec->functionalRange & DRC_DEC_SELECTION) { sErr = drcDec_SelectionProcess_SetCodecMode( hDrcDec->hSelectionProc, (SEL_PROC_CODEC_MODE)codecMode); if (sErr) return DRC_DEC_NOT_OK; hDrcDec->selProcInputDiff = 1; } if (hDrcDec->functionalRange & DRC_DEC_GAIN) { DELAY_MODE delayMode; int timeDomainSupported; SUBBAND_DOMAIN_MODE subbandDomainSupported; switch (hDrcDec->codecMode) { case DRC_DEC_MPEG_4_AAC: case DRC_DEC_MPEG_D_USAC: case DRC_DEC_MPEG_H_3DA: default: delayMode = DM_REGULAR_DELAY; } switch (hDrcDec->codecMode) { case DRC_DEC_MPEG_4_AAC: case DRC_DEC_MPEG_D_USAC: timeDomainSupported = 1; subbandDomainSupported = SDM_OFF; break; case DRC_DEC_MPEG_H_3DA: timeDomainSupported = 1; subbandDomainSupported = SDM_STFT256; break; case DRC_DEC_TEST_TIME_DOMAIN: timeDomainSupported = 1; subbandDomainSupported = SDM_OFF; break; case DRC_DEC_TEST_QMF_DOMAIN: timeDomainSupported = 0; subbandDomainSupported = SDM_QMF64; break; case DRC_DEC_TEST_STFT_DOMAIN: timeDomainSupported = 0; subbandDomainSupported = SDM_STFT256; break; default: timeDomainSupported = 0; subbandDomainSupported = SDM_OFF; } dErr = drcDec_GainDecoder_SetCodecDependentParameters( hDrcDec->hGainDec, delayMode, timeDomainSupported, subbandDomainSupported); if (dErr) return DRC_DEC_NOT_OK; } } /* Don't allow changing codecMode if it has already been set. */ if (hDrcDec->codecMode != codecMode) return DRC_DEC_NOT_OK; return DRC_DEC_OK; } DRC_DEC_ERROR FDK_drcDec_Init(HANDLE_DRC_DECODER hDrcDec, const int frameSize, const int sampleRate, const int baseChannelCount) { DRC_ERROR dErr = DE_OK; DRCDEC_SELECTION_PROCESS_RETURN sErr = DRCDEC_SELECTION_PROCESS_NO_ERROR; if (hDrcDec == NULL || frameSize == 0 || sampleRate == 0 || baseChannelCount == 0) return DRC_DEC_OK; /* return without doing anything */ if (hDrcDec->functionalRange & DRC_DEC_SELECTION) { sErr = drcDec_SelectionProcess_SetParam( hDrcDec->hSelectionProc, SEL_PROC_BASE_CHANNEL_COUNT, (FIXP_DBL)baseChannelCount, &(hDrcDec->selProcInputDiff)); if (sErr) return DRC_DEC_NOT_OK; sErr = drcDec_SelectionProcess_SetParam( hDrcDec->hSelectionProc, SEL_PROC_SAMPLE_RATE, (FIXP_DBL)sampleRate, &(hDrcDec->selProcInputDiff)); if (sErr) return DRC_DEC_NOT_OK; } if (hDrcDec->functionalRange & DRC_DEC_GAIN) { dErr = drcDec_GainDecoder_SetParam(hDrcDec->hGainDec, GAIN_DEC_FRAME_SIZE, frameSize); if (dErr) return DRC_DEC_NOT_OK; dErr = drcDec_GainDecoder_SetParam(hDrcDec->hGainDec, GAIN_DEC_SAMPLE_RATE, sampleRate); if (dErr) return DRC_DEC_NOT_OK; dErr = drcDec_GainDecoder_Init(hDrcDec->hGainDec); if (dErr) return DRC_DEC_NOT_OK; } hDrcDec->status = DRC_DEC_INITIALIZED; startSelectionProcess(hDrcDec); return DRC_DEC_OK; } DRC_DEC_ERROR FDK_drcDec_Close(HANDLE_DRC_DECODER* phDrcDec) { HANDLE_DRC_DECODER hDrcDec; if (phDrcDec == NULL) { return DRC_DEC_OK; } hDrcDec = *phDrcDec; if (hDrcDec == NULL) return DRC_DEC_NOT_OPENED; if (hDrcDec->functionalRange & DRC_DEC_GAIN) { drcDec_GainDecoder_Close(&(hDrcDec->hGainDec)); } if (hDrcDec->functionalRange & DRC_DEC_SELECTION) { drcDec_SelectionProcess_Delete(&(hDrcDec->hSelectionProc)); } FDKfree(*phDrcDec); *phDrcDec = NULL; return DRC_DEC_OK; } DRC_DEC_ERROR FDK_drcDec_SetParam(HANDLE_DRC_DECODER hDrcDec, const DRC_DEC_USERPARAM requestType, const FIXP_DBL requestValue) { DRC_ERROR dErr = DE_OK; DRCDEC_SELECTION_PROCESS_RETURN sErr = DRCDEC_SELECTION_PROCESS_NO_ERROR; int invalidParameter = 0; if (hDrcDec == NULL) return DRC_DEC_NOT_OPENED; if (hDrcDec->functionalRange & DRC_DEC_GAIN) { switch (requestType) { case DRC_DEC_SAMPLE_RATE: dErr = drcDec_GainDecoder_SetParam( hDrcDec->hGainDec, GAIN_DEC_SAMPLE_RATE, (int)requestValue); if (dErr) return DRC_DEC_PARAM_OUT_OF_RANGE; break; case DRC_DEC_FRAME_SIZE: dErr = drcDec_GainDecoder_SetParam( hDrcDec->hGainDec, GAIN_DEC_FRAME_SIZE, (int)requestValue); if (dErr) return DRC_DEC_PARAM_OUT_OF_RANGE; break; default: invalidParameter |= DRC_DEC_GAIN; } } if (hDrcDec->functionalRange & DRC_DEC_SELECTION) { switch (requestType) { case DRC_DEC_BOOST: sErr = drcDec_SelectionProcess_SetParam(hDrcDec->hSelectionProc, SEL_PROC_BOOST, requestValue, &(hDrcDec->selProcInputDiff)); if (sErr) return DRC_DEC_PARAM_OUT_OF_RANGE; break; case DRC_DEC_COMPRESS: sErr = drcDec_SelectionProcess_SetParam(hDrcDec->hSelectionProc, SEL_PROC_COMPRESS, requestValue, &(hDrcDec->selProcInputDiff)); if (sErr) return DRC_DEC_PARAM_OUT_OF_RANGE; break; case DRC_DEC_LOUDNESS_NORMALIZATION_ON: sErr = drcDec_SelectionProcess_SetParam( hDrcDec->hSelectionProc, SEL_PROC_LOUDNESS_NORMALIZATION_ON, requestValue, &(hDrcDec->selProcInputDiff)); if (sErr) return DRC_DEC_PARAM_OUT_OF_RANGE; break; case DRC_DEC_TARGET_LOUDNESS: sErr = drcDec_SelectionProcess_SetParam( hDrcDec->hSelectionProc, SEL_PROC_TARGET_LOUDNESS, requestValue, &(hDrcDec->selProcInputDiff)); if (sErr) return DRC_DEC_PARAM_OUT_OF_RANGE; break; case DRC_DEC_EFFECT_TYPE: sErr = drcDec_SelectionProcess_SetParam( hDrcDec->hSelectionProc, SEL_PROC_EFFECT_TYPE, requestValue, &(hDrcDec->selProcInputDiff)); if (sErr) return DRC_DEC_PARAM_OUT_OF_RANGE; break; case DRC_DEC_DOWNMIX_ID: sErr = drcDec_SelectionProcess_SetParam( hDrcDec->hSelectionProc, SEL_PROC_DOWNMIX_ID, requestValue, &(hDrcDec->selProcInputDiff)); if (sErr) return DRC_DEC_PARAM_OUT_OF_RANGE; break; case DRC_DEC_TARGET_CHANNEL_COUNT_REQUESTED: sErr = drcDec_SelectionProcess_SetParam( hDrcDec->hSelectionProc, SEL_PROC_TARGET_CHANNEL_COUNT, requestValue, &(hDrcDec->selProcInputDiff)); if (sErr) return DRC_DEC_PARAM_OUT_OF_RANGE; break; case DRC_DEC_BASE_CHANNEL_COUNT: sErr = drcDec_SelectionProcess_SetParam( hDrcDec->hSelectionProc, SEL_PROC_BASE_CHANNEL_COUNT, requestValue, &(hDrcDec->selProcInputDiff)); if (sErr) return DRC_DEC_NOT_OK; break; case DRC_DEC_LOUDNESS_MEASUREMENT_METHOD: sErr = drcDec_SelectionProcess_SetParam( hDrcDec->hSelectionProc, SEL_PROC_LOUDNESS_MEASUREMENT_METHOD, requestValue, &(hDrcDec->selProcInputDiff)); if (sErr) return DRC_DEC_PARAM_OUT_OF_RANGE; break; case DRC_DEC_ALBUM_MODE: sErr = drcDec_SelectionProcess_SetParam( hDrcDec->hSelectionProc, SEL_PROC_ALBUM_MODE, requestValue, &(hDrcDec->selProcInputDiff)); if (sErr) return DRC_DEC_PARAM_OUT_OF_RANGE; break; default: invalidParameter |= DRC_DEC_SELECTION; } } if (invalidParameter == hDrcDec->functionalRange) return DRC_DEC_INVALID_PARAM; /* All parameters need a new start of the selection process */ startSelectionProcess(hDrcDec); return DRC_DEC_OK; } LONG FDK_drcDec_GetParam(HANDLE_DRC_DECODER hDrcDec, const DRC_DEC_USERPARAM requestType) { if (hDrcDec == NULL) return DRC_DEC_NOT_OPENED; switch (requestType) { case DRC_DEC_BOOST: return (LONG)hDrcDec->selProcOutput.boost; case DRC_DEC_COMPRESS: return (LONG)hDrcDec->selProcOutput.compress; case DRC_DEC_IS_MULTIBAND_DRC_1: return (LONG)bitstreamContainsMultibandDrc(&hDrcDec->uniDrcConfig, 0); case DRC_DEC_IS_MULTIBAND_DRC_2: return (LONG)bitstreamContainsMultibandDrc(&hDrcDec->uniDrcConfig, 0x7F); case DRC_DEC_IS_ACTIVE: { /* MPEG-D DRC is considered active (and overrides MPEG-4 DRC), if * uniDrc payload is present (loudnessInfoSet and/or uniDrcConfig) * at least one of DRC and Loudness Control is switched on */ int drcOn = drcDec_SelectionProcess_GetParam( hDrcDec->hSelectionProc, SEL_PROC_DYNAMIC_RANGE_CONTROL_ON); int lnOn = drcDec_SelectionProcess_GetParam( hDrcDec->hSelectionProc, SEL_PROC_LOUDNESS_NORMALIZATION_ON); int uniDrcPayloadPresent = (hDrcDec->loudnessInfoSet.loudnessInfoCount > 0); uniDrcPayloadPresent |= (hDrcDec->loudnessInfoSet.loudnessInfoAlbumCount > 0); uniDrcPayloadPresent |= (hDrcDec->uniDrcConfig.drcInstructionsUniDrcCount > 0); uniDrcPayloadPresent |= (hDrcDec->uniDrcConfig.downmixInstructionsCount > 0); return (LONG)(uniDrcPayloadPresent && (drcOn || lnOn)); } case DRC_DEC_TARGET_CHANNEL_COUNT_SELECTED: return (LONG)hDrcDec->selProcOutput.targetChannelCount; case DRC_DEC_OUTPUT_LOUDNESS: return (LONG)hDrcDec->selProcOutput.outputLoudness; default: return 0; } } DRC_DEC_ERROR FDK_drcDec_SetInterfaceParameters(HANDLE_DRC_DECODER hDrcDec, HANDLE_UNI_DRC_INTERFACE hUniDrcInterface) { return DRC_DEC_UNSUPPORTED_FUNCTION; } DRC_DEC_ERROR FDK_drcDec_SetSelectionProcessMpeghParameters_simple( HANDLE_DRC_DECODER hDrcDec, const int groupPresetIdRequested, const int numGroupIdsRequested, const int* groupIdsRequested) { return DRC_DEC_UNSUPPORTED_FUNCTION; } DRC_DEC_ERROR FDK_drcDec_SetDownmixInstructions(HANDLE_DRC_DECODER hDrcDec, const int numDownmixId, const int* downmixId, const int* targetLayout, const int* targetChannelCount) { return DRC_DEC_UNSUPPORTED_FUNCTION; } void FDK_drcDec_SetSelectionProcessOutput( HANDLE_DRC_DECODER hDrcDec, HANDLE_SEL_PROC_OUTPUT hSelProcOutput) {} HANDLE_SEL_PROC_OUTPUT FDK_drcDec_GetSelectionProcessOutput(HANDLE_DRC_DECODER hDrcDec) { if (hDrcDec == NULL) return NULL; return &(hDrcDec->selProcOutput); } LONG /* FIXP_DBL, e = 7 */ FDK_drcDec_GetGroupLoudness(HANDLE_SEL_PROC_OUTPUT hSelProcOutput, const int groupID, int* groupLoudnessAvailable) { return (LONG)0; } void FDK_drcDec_SetChannelGains(HANDLE_DRC_DECODER hDrcDec, const int numChannels, const int frameSize, FIXP_DBL* channelGainDb, FIXP_DBL* audioBuffer, const int audioBufferChannelOffset) { int err; if (hDrcDec == NULL) return; err = drcDec_GainDecoder_SetLoudnessNormalizationGainDb( hDrcDec->hGainDec, hDrcDec->selProcOutput.loudnessNormalizationGainDb); if (err) return; drcDec_GainDecoder_SetChannelGains(hDrcDec->hGainDec, numChannels, frameSize, channelGainDb, audioBufferChannelOffset, audioBuffer); } DRC_DEC_ERROR FDK_drcDec_ReadUniDrcConfig(HANDLE_DRC_DECODER hDrcDec, HANDLE_FDK_BITSTREAM hBitstream) { DRC_ERROR dErr = DE_OK; if (hDrcDec == NULL) return DRC_DEC_NOT_OPENED; if (hDrcDec->codecMode == DRC_DEC_MPEG_D_USAC) { dErr = drcDec_readUniDrcConfig(hBitstream, &(hDrcDec->uniDrcConfig)); } else return DRC_DEC_NOT_OK; if (dErr) { /* clear config, if parsing error occured */ FDKmemclear(&hDrcDec->uniDrcConfig, sizeof(hDrcDec->uniDrcConfig)); hDrcDec->uniDrcConfig.diff = 1; } startSelectionProcess(hDrcDec); return DRC_DEC_OK; } DRC_DEC_ERROR FDK_drcDec_ReadDownmixInstructions_Box(HANDLE_DRC_DECODER hDrcDec, HANDLE_FDK_BITSTREAM hBitstream) { DRC_ERROR dErr = DE_OK; if (hDrcDec == NULL) return DRC_DEC_NOT_OPENED; return DRC_DEC_NOT_OK; if (dErr) { /* clear config, if parsing error occurred */ FDKmemclear(&hDrcDec->uniDrcConfig.downmixInstructions, sizeof(hDrcDec->uniDrcConfig.downmixInstructions)); hDrcDec->uniDrcConfig.downmixInstructionsCount = 0; hDrcDec->uniDrcConfig.downmixInstructionsCountV0 = 0; hDrcDec->uniDrcConfig.downmixInstructionsCountV1 = 0; hDrcDec->uniDrcConfig.diff = 1; } startSelectionProcess(hDrcDec); return DRC_DEC_OK; } DRC_DEC_ERROR FDK_drcDec_ReadUniDrcInstructions_Box(HANDLE_DRC_DECODER hDrcDec, HANDLE_FDK_BITSTREAM hBitstream) { DRC_ERROR dErr = DE_OK; if (hDrcDec == NULL) return DRC_DEC_NOT_OPENED; return DRC_DEC_NOT_OK; if (dErr) { /* clear config, if parsing error occurred */ FDKmemclear(&hDrcDec->uniDrcConfig.drcInstructionsUniDrc, sizeof(hDrcDec->uniDrcConfig.drcInstructionsUniDrc)); hDrcDec->uniDrcConfig.drcInstructionsUniDrcCount = 0; hDrcDec->uniDrcConfig.drcInstructionsUniDrcCountV0 = 0; hDrcDec->uniDrcConfig.drcInstructionsUniDrcCountV1 = 0; hDrcDec->uniDrcConfig.diff = 1; } startSelectionProcess(hDrcDec); return DRC_DEC_OK; } DRC_DEC_ERROR FDK_drcDec_ReadUniDrcCoefficients_Box(HANDLE_DRC_DECODER hDrcDec, HANDLE_FDK_BITSTREAM hBitstream) { DRC_ERROR dErr = DE_OK; if (hDrcDec == NULL) return DRC_DEC_NOT_OPENED; return DRC_DEC_NOT_OK; if (dErr) { /* clear config, if parsing error occurred */ FDKmemclear(&hDrcDec->uniDrcConfig.drcCoefficientsUniDrc, sizeof(hDrcDec->uniDrcConfig.drcCoefficientsUniDrc)); hDrcDec->uniDrcConfig.drcCoefficientsUniDrcCount = 0; hDrcDec->uniDrcConfig.drcCoefficientsUniDrcCountV0 = 0; hDrcDec->uniDrcConfig.drcCoefficientsUniDrcCountV1 = 0; hDrcDec->uniDrcConfig.diff = 1; } startSelectionProcess(hDrcDec); return DRC_DEC_OK; } DRC_DEC_ERROR FDK_drcDec_ReadLoudnessInfoSet(HANDLE_DRC_DECODER hDrcDec, HANDLE_FDK_BITSTREAM hBitstream) { DRC_ERROR dErr = DE_OK; if (hDrcDec == NULL) return DRC_DEC_NOT_OPENED; if (hDrcDec->codecMode == DRC_DEC_MPEG_D_USAC) { dErr = drcDec_readLoudnessInfoSet(hBitstream, &(hDrcDec->loudnessInfoSet)); } else return DRC_DEC_NOT_OK; if (dErr) { /* clear config, if parsing error occurred */ FDKmemclear(&hDrcDec->loudnessInfoSet, sizeof(hDrcDec->loudnessInfoSet)); hDrcDec->loudnessInfoSet.diff = 1; } startSelectionProcess(hDrcDec); return DRC_DEC_OK; } DRC_DEC_ERROR FDK_drcDec_ReadLoudnessBox(HANDLE_DRC_DECODER hDrcDec, HANDLE_FDK_BITSTREAM hBitstream) { DRC_ERROR dErr = DE_OK; if (hDrcDec == NULL) return DRC_DEC_NOT_OPENED; return DRC_DEC_NOT_OK; if (dErr) { /* clear config, if parsing error occurred */ FDKmemclear(&hDrcDec->loudnessInfoSet, sizeof(hDrcDec->loudnessInfoSet)); hDrcDec->loudnessInfoSet.diff = 1; } startSelectionProcess(hDrcDec); return DRC_DEC_OK; } DRC_DEC_ERROR FDK_drcDec_ReadUniDrcGain(HANDLE_DRC_DECODER hDrcDec, HANDLE_FDK_BITSTREAM hBitstream) { DRC_ERROR dErr = DE_OK; if (hDrcDec == NULL) return DRC_DEC_NOT_OPENED; if (!hDrcDec->status) { return DRC_DEC_OK; } dErr = drcDec_readUniDrcGain( hBitstream, &(hDrcDec->uniDrcConfig), drcDec_GainDecoder_GetFrameSize(hDrcDec->hGainDec), drcDec_GainDecoder_GetDeltaTminDefault(hDrcDec->hGainDec), &(hDrcDec->uniDrcGain)); if (dErr) return DRC_DEC_NOT_OK; if (_getGainStatus(&(hDrcDec->uniDrcGain))) { hDrcDec->status = DRC_DEC_NEW_GAIN_PAYLOAD; } return DRC_DEC_OK; } DRC_DEC_ERROR FDK_drcDec_ReadUniDrc(HANDLE_DRC_DECODER hDrcDec, HANDLE_FDK_BITSTREAM hBitstream) { DRC_ERROR dErr = DE_OK; if (hDrcDec == NULL) return DRC_DEC_NOT_OPENED; if (!hDrcDec->status) return DRC_DEC_NOT_READY; dErr = drcDec_readUniDrc( hBitstream, &(hDrcDec->uniDrcConfig), &(hDrcDec->loudnessInfoSet), drcDec_GainDecoder_GetFrameSize(hDrcDec->hGainDec), drcDec_GainDecoder_GetDeltaTminDefault(hDrcDec->hGainDec), &(hDrcDec->uniDrcGain)); startSelectionProcess(hDrcDec); if (dErr) return DRC_DEC_NOT_OK; if (_getGainStatus(&(hDrcDec->uniDrcGain))) { hDrcDec->status = DRC_DEC_NEW_GAIN_PAYLOAD; } return DRC_DEC_OK; } DRC_DEC_ERROR FDK_drcDec_Preprocess(HANDLE_DRC_DECODER hDrcDec) { DRC_ERROR dErr = DE_OK; if (hDrcDec == NULL) return DRC_DEC_NOT_OPENED; if (!hDrcDec->status) return DRC_DEC_NOT_READY; if (!(hDrcDec->functionalRange & DRC_DEC_GAIN)) return DRC_DEC_NOT_OK; if (hDrcDec->status != DRC_DEC_NEW_GAIN_PAYLOAD) { /* no new gain payload was read, e.g. during concalment or flushing. Generate DRC gains based on the stored DRC gains of last frames */ drcDec_GainDecoder_Conceal(hDrcDec->hGainDec, &(hDrcDec->uniDrcConfig), &(hDrcDec->uniDrcGain)); } dErr = drcDec_GainDecoder_Preprocess( hDrcDec->hGainDec, &(hDrcDec->uniDrcGain), hDrcDec->selProcOutput.loudnessNormalizationGainDb, hDrcDec->selProcOutput.boost, hDrcDec->selProcOutput.compress); if (dErr) return DRC_DEC_NOT_OK; hDrcDec->status = DRC_DEC_INTERPOLATION_PREPARED; return DRC_DEC_OK; } DRC_DEC_ERROR FDK_drcDec_ProcessTime(HANDLE_DRC_DECODER hDrcDec, const int delaySamples, const DRC_DEC_LOCATION drcLocation, const int channelOffset, const int drcChannelOffset, const int numChannelsProcessed, FIXP_DBL* realBuffer, const int timeDataChannelOffset) { DRC_ERROR dErr = DE_OK; if (hDrcDec == NULL) return DRC_DEC_NOT_OPENED; if (!(hDrcDec->functionalRange & DRC_DEC_GAIN)) return DRC_DEC_NOT_OK; if (hDrcDec->status != DRC_DEC_INTERPOLATION_PREPARED) return DRC_DEC_NOT_READY; dErr = drcDec_GainDecoder_ProcessTimeDomain( hDrcDec->hGainDec, delaySamples, (GAIN_DEC_LOCATION)drcLocation, channelOffset, drcChannelOffset, numChannelsProcessed, timeDataChannelOffset, realBuffer); if (dErr) return DRC_DEC_NOT_OK; return DRC_DEC_OK; } DRC_DEC_ERROR FDK_drcDec_ProcessFreq(HANDLE_DRC_DECODER hDrcDec, const int delaySamples, const DRC_DEC_LOCATION drcLocation, const int channelOffset, const int drcChannelOffset, const int numChannelsProcessed, const int processSingleTimeslot, FIXP_DBL** realBuffer, FIXP_DBL** imagBuffer) { DRC_ERROR dErr = DE_OK; if (hDrcDec == NULL) return DRC_DEC_NOT_OPENED; if (!(hDrcDec->functionalRange & DRC_DEC_GAIN)) return DRC_DEC_NOT_OK; if (hDrcDec->status != DRC_DEC_INTERPOLATION_PREPARED) return DRC_DEC_NOT_READY; dErr = drcDec_GainDecoder_ProcessSubbandDomain( hDrcDec->hGainDec, delaySamples, (GAIN_DEC_LOCATION)drcLocation, channelOffset, drcChannelOffset, numChannelsProcessed, processSingleTimeslot, realBuffer, imagBuffer); if (dErr) return DRC_DEC_NOT_OK; return DRC_DEC_OK; } DRC_DEC_ERROR FDK_drcDec_ApplyDownmix(HANDLE_DRC_DECODER hDrcDec, int* reverseInChannelMap, int* reverseOutChannelMap, FIXP_DBL* realBuffer, int* pNChannels) { SEL_PROC_OUTPUT* pSelProcOutput = &(hDrcDec->selProcOutput); int baseChCnt = pSelProcOutput->baseChannelCount; int targetChCnt = pSelProcOutput->targetChannelCount; int frameSize, n, ic, oc; FIXP_DBL tmp_out[8]; FIXP_DBL* audioChannels[8]; if (hDrcDec == NULL) return DRC_DEC_NOT_OPENED; if (!(hDrcDec->functionalRange & DRC_DEC_GAIN)) return DRC_DEC_NOT_OK; /* only downmix is performed here, no upmix. Downmix is only performed if downmix coefficients are provided. All other cases of downmix and upmix are treated by pcmDmx library. */ if (pSelProcOutput->downmixMatrixPresent == 0) return DRC_DEC_OK; /* no downmix */ if (targetChCnt >= baseChCnt) return DRC_DEC_OK; /* downmix only */ /* sanity checks */ if (realBuffer == NULL) return DRC_DEC_NOT_OK; if (reverseInChannelMap == NULL) return DRC_DEC_NOT_OK; if (reverseOutChannelMap == NULL) return DRC_DEC_NOT_OK; if (baseChCnt > 8) return DRC_DEC_NOT_OK; if (baseChCnt != *pNChannels) return DRC_DEC_NOT_OK; if (targetChCnt > 8) return DRC_DEC_NOT_OK; frameSize = drcDec_GainDecoder_GetFrameSize(hDrcDec->hGainDec); for (ic = 0; ic < baseChCnt; ic++) { audioChannels[ic] = &(realBuffer[ic * frameSize]); } /* in-place downmix */ for (n = 0; n < frameSize; n++) { for (oc = 0; oc < targetChCnt; oc++) { tmp_out[oc] = (FIXP_DBL)0; for (ic = 0; ic < baseChCnt; ic++) { tmp_out[oc] += fMultDiv2(audioChannels[ic][n], pSelProcOutput->downmixMatrix[reverseInChannelMap[ic]] [reverseOutChannelMap[oc]]) << 3; } } for (oc = 0; oc < targetChCnt; oc++) { if (oc >= baseChCnt) break; audioChannels[oc][n] = tmp_out[oc]; } } for (oc = targetChCnt; oc < baseChCnt; oc++) { FDKmemset(audioChannels[oc], 0, frameSize * sizeof(FIXP_DBL)); } *pNChannels = targetChCnt; return DRC_DEC_OK; } /* Get library info for this module. */ DRC_DEC_ERROR FDK_drcDec_GetLibInfo(LIB_INFO* info) { int i; if (info == NULL) { return DRC_DEC_INVALID_PARAM; } /* Search for next free tab */ for (i = 0; i < FDK_MODULE_LAST; i++) { if (info[i].module_id == FDK_NONE) break; } if (i == FDK_MODULE_LAST) { return DRC_DEC_NOT_OK; } /* Add the library info */ info[i].module_id = FDK_UNIDRCDEC; info[i].version = LIB_VERSION(DRCDEC_LIB_VL0, DRCDEC_LIB_VL1, DRCDEC_LIB_VL2); LIB_VERSION_STRING(info + i); info[i].build_date = DRCDEC_LIB_BUILD_DATE; info[i].build_time = DRCDEC_LIB_BUILD_TIME; info[i].title = DRCDEC_LIB_TITLE; return DRC_DEC_OK; } fdk-aac-2.0.2/libDRCdec/src/drcDec_gainDecoder.cpp000066400000000000000000000403571404225310700215260ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /************************* MPEG-D DRC decoder library ************************** Author(s): Description: *******************************************************************************/ #include "drcDec_types.h" #include "drcDec_gainDecoder.h" #include "drcGainDec_preprocess.h" #include "drcGainDec_init.h" #include "drcGainDec_process.h" #include "drcDec_tools.h" /*******************************************/ /* static functions */ /*******************************************/ static int _fitsLocation(DRC_INSTRUCTIONS_UNI_DRC* pInst, const GAIN_DEC_LOCATION drcLocation) { int downmixId = pInst->drcApplyToDownmix ? pInst->downmixId[0] : 0; switch (drcLocation) { case GAIN_DEC_DRC1: return (downmixId == 0); case GAIN_DEC_DRC1_DRC2: return ((downmixId == 0) || (downmixId == DOWNMIX_ID_ANY_DOWNMIX)); case GAIN_DEC_DRC2: return (downmixId == DOWNMIX_ID_ANY_DOWNMIX); case GAIN_DEC_DRC3: return ((downmixId != 0) && (downmixId != DOWNMIX_ID_ANY_DOWNMIX)); case GAIN_DEC_DRC2_DRC3: return (downmixId != 0); } return 0; } static void _setChannelGains(HANDLE_DRC_GAIN_DECODER hGainDec, const int numChannelGains, const FIXP_DBL* channelGainDb) { int i, channelGain_e; FIXP_DBL channelGain; FDK_ASSERT(numChannelGains <= 8); for (i = 0; i < numChannelGains; i++) { if (channelGainDb[i] == (FIXP_DBL)MINVAL_DBL) { hGainDec->channelGain[i] = (FIXP_DBL)0; } else { /* add loudness normalisation gain (dB) to channel gain (dB) */ FIXP_DBL tmp_channelGainDb = (channelGainDb[i] >> 1) + (hGainDec->loudnessNormalisationGainDb >> 2); tmp_channelGainDb = SATURATE_LEFT_SHIFT(tmp_channelGainDb, 1, DFRACT_BITS); channelGain = dB2lin(tmp_channelGainDb, 8, &channelGain_e); hGainDec->channelGain[i] = scaleValue(channelGain, channelGain_e - 8); } } } /*******************************************/ /* public functions */ /*******************************************/ DRC_ERROR drcDec_GainDecoder_Open(HANDLE_DRC_GAIN_DECODER* phGainDec) { DRC_GAIN_DECODER* hGainDec = NULL; hGainDec = (DRC_GAIN_DECODER*)FDKcalloc(1, sizeof(DRC_GAIN_DECODER)); if (hGainDec == NULL) return DE_MEMORY_ERROR; hGainDec->multiBandActiveDrcIndex = -1; hGainDec->channelGainActiveDrcIndex = -1; *phGainDec = hGainDec; return DE_OK; } DRC_ERROR drcDec_GainDecoder_Init(HANDLE_DRC_GAIN_DECODER hGainDec) { DRC_ERROR err = DE_OK; err = initGainDec(hGainDec); if (err) return err; initDrcGainBuffers(hGainDec->frameSize, &hGainDec->drcGainBuffers); return err; } DRC_ERROR drcDec_GainDecoder_SetParam(HANDLE_DRC_GAIN_DECODER hGainDec, const GAIN_DEC_PARAM paramType, const int paramValue) { switch (paramType) { case GAIN_DEC_FRAME_SIZE: if (paramValue < 0) return DE_PARAM_OUT_OF_RANGE; hGainDec->frameSize = paramValue; break; case GAIN_DEC_SAMPLE_RATE: if (paramValue < 0) return DE_PARAM_OUT_OF_RANGE; hGainDec->deltaTminDefault = getDeltaTmin(paramValue); break; default: return DE_PARAM_INVALID; } return DE_OK; } DRC_ERROR drcDec_GainDecoder_SetCodecDependentParameters( HANDLE_DRC_GAIN_DECODER hGainDec, const DELAY_MODE delayMode, const int timeDomainSupported, const SUBBAND_DOMAIN_MODE subbandDomainSupported) { if ((delayMode != DM_REGULAR_DELAY) && (delayMode != DM_LOW_DELAY)) { return DE_NOT_OK; } hGainDec->delayMode = delayMode; hGainDec->timeDomainSupported = timeDomainSupported; hGainDec->subbandDomainSupported = subbandDomainSupported; return DE_OK; } DRC_ERROR drcDec_GainDecoder_Config(HANDLE_DRC_GAIN_DECODER hGainDec, HANDLE_UNI_DRC_CONFIG hUniDrcConfig, const UCHAR numSelectedDrcSets, const SCHAR* selectedDrcSetIds, const UCHAR* selectedDownmixIds) { DRC_ERROR err = DE_OK; int a; hGainDec->nActiveDrcs = 0; hGainDec->multiBandActiveDrcIndex = -1; hGainDec->channelGainActiveDrcIndex = -1; for (a = 0; a < numSelectedDrcSets; a++) { err = initActiveDrc(hGainDec, hUniDrcConfig, selectedDrcSetIds[a], selectedDownmixIds[a]); if (err) return err; } err = initActiveDrcOffset(hGainDec); if (err) return err; return err; } DRC_ERROR drcDec_GainDecoder_Close(HANDLE_DRC_GAIN_DECODER* phGainDec) { if (*phGainDec != NULL) { FDKfree(*phGainDec); *phGainDec = NULL; } return DE_OK; } DRC_ERROR drcDec_GainDecoder_Preprocess(HANDLE_DRC_GAIN_DECODER hGainDec, HANDLE_UNI_DRC_GAIN hUniDrcGain, const FIXP_DBL loudnessNormalizationGainDb, const FIXP_SGL boost, const FIXP_SGL compress) { DRC_ERROR err = DE_OK; int a, c; /* lnbPointer is the index on the most recent node buffer */ hGainDec->drcGainBuffers.lnbPointer++; if (hGainDec->drcGainBuffers.lnbPointer >= NUM_LNB_FRAMES) hGainDec->drcGainBuffers.lnbPointer = 0; for (a = 0; a < hGainDec->nActiveDrcs; a++) { /* prepare gain interpolation of sequences used by copying and modifying * nodes in node buffers */ err = prepareDrcGain(hGainDec, hUniDrcGain, compress, boost, loudnessNormalizationGainDb, a); if (err) return err; } for (a = 0; a < MAX_ACTIVE_DRCS; a++) { for (c = 0; c < 8; c++) { hGainDec->activeDrc[a] .lnbIndexForChannel[c][hGainDec->drcGainBuffers.lnbPointer] = -1; /* "no DRC processing" */ } hGainDec->activeDrc[a].subbandGainsReady = 0; } for (c = 0; c < 8; c++) { hGainDec->drcGainBuffers .channelGain[c][hGainDec->drcGainBuffers.lnbPointer] = FL2FXCONST_DBL(1.0f / (float)(1 << 8)); } return err; } /* create gain sequence out of gain sequences of last frame for concealment and * flushing */ DRC_ERROR drcDec_GainDecoder_Conceal(HANDLE_DRC_GAIN_DECODER hGainDec, HANDLE_UNI_DRC_CONFIG hUniDrcConfig, HANDLE_UNI_DRC_GAIN hUniDrcGain) { int seq, gainSequenceCount; DRC_COEFFICIENTS_UNI_DRC* pCoef = selectDrcCoefficients(hUniDrcConfig, LOCATION_SELECTED); if (pCoef && pCoef->gainSequenceCount) { gainSequenceCount = fMin(pCoef->gainSequenceCount, (UCHAR)12); } else { gainSequenceCount = 1; } for (seq = 0; seq < gainSequenceCount; seq++) { int lastNodeIndex = 0; FIXP_SGL lastGainDb = (FIXP_SGL)0; lastNodeIndex = hUniDrcGain->nNodes[seq] - 1; if ((lastNodeIndex >= 0) && (lastNodeIndex < 16)) { lastGainDb = hUniDrcGain->gainNode[seq][lastNodeIndex].gainDb; } hUniDrcGain->nNodes[seq] = 1; if (lastGainDb > (FIXP_SGL)0) { hUniDrcGain->gainNode[seq][0].gainDb = FX_DBL2FX_SGL(fMult(FL2FXCONST_SGL(0.9f), lastGainDb)); } else { hUniDrcGain->gainNode[seq][0].gainDb = FX_DBL2FX_SGL(fMult(FL2FXCONST_SGL(0.98f), lastGainDb)); } hUniDrcGain->gainNode[seq][0].time = hGainDec->frameSize - 1; } return DE_OK; } void drcDec_GainDecoder_SetChannelGains(HANDLE_DRC_GAIN_DECODER hGainDec, const int numChannels, const int frameSize, const FIXP_DBL* channelGainDb, const int audioBufferChannelOffset, FIXP_DBL* audioBuffer) { int c, i; if (hGainDec->channelGainActiveDrcIndex >= 0) { /* channel gains will be applied in drcDec_GainDecoder_ProcessTimeDomain or * drcDec_GainDecoder_ProcessSubbandDomain, respectively. */ _setChannelGains(hGainDec, numChannels, channelGainDb); if (!hGainDec->status) { /* overwrite previous channel gains at startup */ DRC_GAIN_BUFFERS* pDrcGainBuffers = &hGainDec->drcGainBuffers; for (c = 0; c < numChannels; c++) { for (i = 0; i < NUM_LNB_FRAMES; i++) { pDrcGainBuffers->channelGain[c][i] = hGainDec->channelGain[c]; } } hGainDec->status = 1; } } else { /* smooth and apply channel gains */ FIXP_DBL prevChannelGain[8]; for (c = 0; c < numChannels; c++) { prevChannelGain[c] = hGainDec->channelGain[c]; } _setChannelGains(hGainDec, numChannels, channelGainDb); if (!hGainDec->status) { /* overwrite previous channel gains at startup */ for (c = 0; c < numChannels; c++) prevChannelGain[c] = hGainDec->channelGain[c]; hGainDec->status = 1; } for (c = 0; c < numChannels; c++) { INT n_min = fMin(fMin(CntLeadingZeros(prevChannelGain[c]), CntLeadingZeros(hGainDec->channelGain[c])) - 1, 9); FIXP_DBL gain = prevChannelGain[c] << n_min; FIXP_DBL stepsize = ((hGainDec->channelGain[c] << n_min) - gain); if (stepsize != (FIXP_DBL)0) { if (frameSize == 1024) stepsize = stepsize >> 10; else stepsize = (LONG)stepsize / frameSize; } n_min = 9 - n_min; #ifdef FUNCTION_drcDec_GainDecoder_SetChannelGains_func1 drcDec_GainDecoder_SetChannelGains_func1(audioBuffer, gain, stepsize, n_min, frameSize); #else for (i = 0; i < frameSize; i++) { audioBuffer[i] = fMultDiv2(audioBuffer[i], gain) << n_min; gain += stepsize; } #endif audioBuffer += audioBufferChannelOffset; } } } DRC_ERROR drcDec_GainDecoder_ProcessTimeDomain( HANDLE_DRC_GAIN_DECODER hGainDec, const int delaySamples, const GAIN_DEC_LOCATION drcLocation, const int channelOffset, const int drcChannelOffset, const int numChannelsProcessed, const int timeDataChannelOffset, FIXP_DBL* audioIOBuffer) { DRC_ERROR err = DE_OK; int a; if (!hGainDec->timeDomainSupported) { return DE_NOT_OK; } for (a = 0; a < hGainDec->nActiveDrcs; a++) { if (!_fitsLocation(hGainDec->activeDrc[a].pInst, drcLocation)) continue; /* Apply DRC */ err = processDrcTime(hGainDec, a, delaySamples, channelOffset, drcChannelOffset, numChannelsProcessed, timeDataChannelOffset, audioIOBuffer); if (err) return err; } return err; } DRC_ERROR drcDec_GainDecoder_ProcessSubbandDomain( HANDLE_DRC_GAIN_DECODER hGainDec, const int delaySamples, const GAIN_DEC_LOCATION drcLocation, const int channelOffset, const int drcChannelOffset, const int numChannelsProcessed, const int processSingleTimeslot, FIXP_DBL* audioIOBufferReal[], FIXP_DBL* audioIOBufferImag[]) { DRC_ERROR err = DE_OK; int a; if (hGainDec->subbandDomainSupported == SDM_OFF) { return DE_NOT_OK; } for (a = 0; a < hGainDec->nActiveDrcs; a++) { if (!_fitsLocation(hGainDec->activeDrc[a].pInst, drcLocation)) continue; /* Apply DRC */ err = processDrcSubband(hGainDec, a, delaySamples, channelOffset, drcChannelOffset, numChannelsProcessed, processSingleTimeslot, audioIOBufferReal, audioIOBufferImag); if (err) return err; } return err; } DRC_ERROR drcDec_GainDecoder_SetLoudnessNormalizationGainDb( HANDLE_DRC_GAIN_DECODER hGainDec, FIXP_DBL loudnessNormalizationGainDb) { hGainDec->loudnessNormalisationGainDb = loudnessNormalizationGainDb; return DE_OK; } int drcDec_GainDecoder_GetFrameSize(HANDLE_DRC_GAIN_DECODER hGainDec) { if (hGainDec == NULL) return -1; return hGainDec->frameSize; } int drcDec_GainDecoder_GetDeltaTminDefault(HANDLE_DRC_GAIN_DECODER hGainDec) { if (hGainDec == NULL) return -1; return hGainDec->deltaTminDefault; } fdk-aac-2.0.2/libDRCdec/src/drcDec_gainDecoder.h000066400000000000000000000241571404225310700211730ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /************************* MPEG-D DRC decoder library ************************** Author(s): Description: *******************************************************************************/ #ifndef DRCDEC_GAINDECODER_H #define DRCDEC_GAINDECODER_H #include "drcDecoder.h" /* Definitions common to gainDecoder submodule */ #define NUM_LNB_FRAMES \ 5 /* previous frame + this frame + one frame for DM_REGULAR_DELAY + (maximum \ delaySamples)/frameSize */ /* QMF64 */ #define SUBBAND_NUM_BANDS_QMF64 64 #define SUBBAND_DOWNSAMPLING_FACTOR_QMF64 64 #define SUBBAND_ANALYSIS_DELAY_QMF64 320 /* QMF71 (according to ISO/IEC 23003-1:2007) */ #define SUBBAND_NUM_BANDS_QMF71 71 #define SUBBAND_DOWNSAMPLING_FACTOR_QMF71 64 #define SUBBAND_ANALYSIS_DELAY_QMF71 320 + 384 /* STFT256 (according to ISO/IEC 23008-3:2015/AMD3) */ #define SUBBAND_NUM_BANDS_STFT256 256 #define SUBBAND_DOWNSAMPLING_FACTOR_STFT256 256 #define SUBBAND_ANALYSIS_DELAY_STFT256 256 typedef enum { GAIN_DEC_DRC1, GAIN_DEC_DRC1_DRC2, GAIN_DEC_DRC2, GAIN_DEC_DRC3, GAIN_DEC_DRC2_DRC3 } GAIN_DEC_LOCATION; typedef enum { GAIN_DEC_FRAME_SIZE, GAIN_DEC_SAMPLE_RATE } GAIN_DEC_PARAM; typedef struct { FIXP_DBL gainLin; /* e = 7 */ SHORT time; } NODE_LIN; typedef struct { GAIN_INTERPOLATION_TYPE gainInterpolationType; int nNodes[NUM_LNB_FRAMES]; /* number of nodes, saturated to 16 */ NODE_LIN linearNode[NUM_LNB_FRAMES][16]; } LINEAR_NODE_BUFFER; typedef struct { int lnbPointer; LINEAR_NODE_BUFFER linearNodeBuffer[12]; LINEAR_NODE_BUFFER dummyLnb; FIXP_DBL channelGain[8][NUM_LNB_FRAMES]; /* e = 8 */ } DRC_GAIN_BUFFERS; typedef struct { int activeDrcOffset; DRC_INSTRUCTIONS_UNI_DRC* pInst; DRC_COEFFICIENTS_UNI_DRC* pCoef; DUCKING_MODIFICATION duckingModificationForChannelGroup[8]; SCHAR channelGroupForChannel[8]; UCHAR bandCountForChannelGroup[8]; UCHAR gainElementForGroup[8]; UCHAR channelGroupIsParametricDrc[8]; UCHAR gainElementCount; /* number of different DRC gains inluding all DRC bands */ int lnbIndexForChannel[8][NUM_LNB_FRAMES]; int subbandGainsReady; } ACTIVE_DRC; typedef struct { int deltaTminDefault; INT frameSize; FIXP_DBL loudnessNormalisationGainDb; DELAY_MODE delayMode; int nActiveDrcs; ACTIVE_DRC activeDrc[MAX_ACTIVE_DRCS]; int multiBandActiveDrcIndex; int channelGainActiveDrcIndex; FIXP_DBL channelGain[8]; /* e = 8 */ DRC_GAIN_BUFFERS drcGainBuffers; FIXP_DBL subbandGains[12][4 * 1024 / 256]; FIXP_DBL dummySubbandGains[4 * 1024 / 256]; int status; int timeDomainSupported; SUBBAND_DOMAIN_MODE subbandDomainSupported; } DRC_GAIN_DECODER, *HANDLE_DRC_GAIN_DECODER; /* init functions */ DRC_ERROR drcDec_GainDecoder_Open(HANDLE_DRC_GAIN_DECODER* phGainDec); DRC_ERROR drcDec_GainDecoder_Init(HANDLE_DRC_GAIN_DECODER hGainDec); DRC_ERROR drcDec_GainDecoder_SetParam(HANDLE_DRC_GAIN_DECODER hGainDec, const GAIN_DEC_PARAM paramType, const int paramValue); DRC_ERROR drcDec_GainDecoder_SetCodecDependentParameters( HANDLE_DRC_GAIN_DECODER hGainDec, const DELAY_MODE delayMode, const int timeDomainSupported, const SUBBAND_DOMAIN_MODE subbandDomainSupported); DRC_ERROR drcDec_GainDecoder_Config(HANDLE_DRC_GAIN_DECODER hGainDec, HANDLE_UNI_DRC_CONFIG hUniDrcConfig, const UCHAR numSelectedDrcSets, const SCHAR* selectedDrcSetIds, const UCHAR* selectedDownmixIds); /* close functions */ DRC_ERROR drcDec_GainDecoder_Close(HANDLE_DRC_GAIN_DECODER* phGainDec); /* process functions */ /* call drcDec_GainDecoder_Preprocess first */ DRC_ERROR drcDec_GainDecoder_Preprocess(HANDLE_DRC_GAIN_DECODER hGainDec, HANDLE_UNI_DRC_GAIN hUniDrcGain, const FIXP_DBL loudnessNormalizationGainDb, const FIXP_SGL boost, const FIXP_SGL compress); /* Then call one of drcDec_GainDecoder_ProcessTimeDomain or * drcDec_GainDecoder_ProcessSubbandDomain */ DRC_ERROR drcDec_GainDecoder_ProcessTimeDomain( HANDLE_DRC_GAIN_DECODER hGainDec, const int delaySamples, const GAIN_DEC_LOCATION drcLocation, const int channelOffset, const int drcChannelOffset, const int numChannelsProcessed, const int timeDataChannelOffset, FIXP_DBL* audioIOBuffer); DRC_ERROR drcDec_GainDecoder_ProcessSubbandDomain( HANDLE_DRC_GAIN_DECODER hGainDec, const int delaySamples, GAIN_DEC_LOCATION drcLocation, const int channelOffset, const int drcChannelOffset, const int numChannelsProcessed, const int processSingleTimeslot, FIXP_DBL* audioIOBufferReal[], FIXP_DBL* audioIOBufferImag[]); DRC_ERROR drcDec_GainDecoder_Conceal(HANDLE_DRC_GAIN_DECODER hGainDec, HANDLE_UNI_DRC_CONFIG hUniDrcConfig, HANDLE_UNI_DRC_GAIN hUniDrcGain); DRC_ERROR drcDec_GainDecoder_SetLoudnessNormalizationGainDb( HANDLE_DRC_GAIN_DECODER hGainDec, FIXP_DBL loudnessNormalizationGainDb); int drcDec_GainDecoder_GetFrameSize(HANDLE_DRC_GAIN_DECODER hGainDec); int drcDec_GainDecoder_GetDeltaTminDefault(HANDLE_DRC_GAIN_DECODER hGainDec); void drcDec_GainDecoder_SetChannelGains(HANDLE_DRC_GAIN_DECODER hGainDec, const int numChannels, const int frameSize, const FIXP_DBL* channelGainDb, const int audioBufferChannelOffset, FIXP_DBL* audioBuffer); #endif fdk-aac-2.0.2/libDRCdec/src/drcDec_reader.cpp000066400000000000000000002124501404225310700205570ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /************************* MPEG-D DRC decoder library ************************** Author(s): Description: *******************************************************************************/ #include "fixpoint_math.h" #include "drcDec_reader.h" #include "drcDec_tools.h" #include "drcDec_rom.h" #include "drcDecoder.h" /* MPEG-D DRC AMD 1 */ #define UNIDRCCONFEXT_PARAM_DRC 0x1 #define UNIDRCCONFEXT_V1 0x2 #define UNIDRCLOUDEXT_EQ 0x1 #define UNIDRCGAINEXT_TERM 0x0 #define UNIDRCLOUDEXT_TERM 0x0 #define UNIDRCCONFEXT_TERM 0x0 static int _getZ(const int nNodesMax) { /* Z is the minimum codeword length that is needed to encode all possible * timeDelta values */ /* Z = ceil(log2(2*nNodesMax)) */ int Z = 1; while ((1 << Z) < (2 * nNodesMax)) { Z++; } return Z; } static int _getTimeDeltaMin(const GAIN_SET* pGset, const int deltaTminDefault) { if (pGset->timeDeltaMinPresent) { return pGset->timeDeltaMin; } else { return deltaTminDefault; } } /* compare and assign */ static inline int _compAssign(UCHAR* dest, const UCHAR src) { int diff = 0; if (*dest != src) diff = 1; *dest = src; return diff; } static inline int _compAssign(ULONG* dest, const ULONG src) { int diff = 0; if (*dest != src) diff = 1; *dest = src; return diff; } typedef const SCHAR (*Huffman)[2]; int _decodeHuffmanCW(Huffman h, /*!< pointer to huffman codebook table */ HANDLE_FDK_BITSTREAM hBs) /*!< Handle to bitbuffer */ { SCHAR index = 0; int value, bit; while (index >= 0) { bit = FDKreadBits(hBs, 1); index = h[index][bit]; } value = index + 64; /* Add offset */ return value; } /**********/ /* uniDrc */ /**********/ DRC_ERROR drcDec_readUniDrc(HANDLE_FDK_BITSTREAM hBs, HANDLE_UNI_DRC_CONFIG hUniDrcConfig, HANDLE_LOUDNESS_INFO_SET hLoudnessInfoSet, const int frameSize, const int deltaTminDefault, HANDLE_UNI_DRC_GAIN hUniDrcGain) { DRC_ERROR err = DE_OK; int loudnessInfoSetPresent, uniDrcConfigPresent; loudnessInfoSetPresent = FDKreadBits(hBs, 1); if (loudnessInfoSetPresent) { uniDrcConfigPresent = FDKreadBits(hBs, 1); if (uniDrcConfigPresent) { err = drcDec_readUniDrcConfig(hBs, hUniDrcConfig); if (err) { /* clear config, if parsing error occured */ FDKmemclear(hUniDrcConfig, sizeof(UNI_DRC_CONFIG)); hUniDrcConfig->diff = 1; } } err = drcDec_readLoudnessInfoSet(hBs, hLoudnessInfoSet); if (err) { /* clear config, if parsing error occured */ FDKmemclear(hLoudnessInfoSet, sizeof(LOUDNESS_INFO_SET)); hLoudnessInfoSet->diff = 1; } } err = drcDec_readUniDrcGain(hBs, hUniDrcConfig, frameSize, deltaTminDefault, hUniDrcGain); return err; } /**************/ /* uniDrcGain */ /**************/ static FIXP_SGL _decodeGainInitial( HANDLE_FDK_BITSTREAM hBs, const GAIN_CODING_PROFILE gainCodingProfile) { int sign, magn; FIXP_SGL gainInitial = (FIXP_SGL)0; switch (gainCodingProfile) { case GCP_REGULAR: sign = FDKreadBits(hBs, 1); magn = FDKreadBits(hBs, 8); gainInitial = (FIXP_SGL)(magn << (FRACT_BITS - 1 - 3 - 7)); /* magn * 0.125; */ if (sign) gainInitial = -gainInitial; break; case GCP_FADING: sign = FDKreadBits(hBs, 1); if (sign == 0) gainInitial = (FIXP_SGL)0; else { magn = FDKreadBits(hBs, 10); gainInitial = -(FIXP_SGL)( (magn + 1) << (FRACT_BITS - 1 - 3 - 7)); /* - (magn + 1) * 0.125; */ } break; case GCP_CLIPPING_DUCKING: sign = FDKreadBits(hBs, 1); if (sign == 0) gainInitial = (FIXP_SGL)0; else { magn = FDKreadBits(hBs, 8); gainInitial = -(FIXP_SGL)( (magn + 1) << (FRACT_BITS - 1 - 3 - 7)); /* - (magn + 1) * 0.125; */ } break; case GCP_CONSTANT: break; } return gainInitial; } static int _decodeNNodes(HANDLE_FDK_BITSTREAM hBs) { int nNodes = 0, endMarker = 0; /* decode number of nodes */ while (endMarker != 1) { nNodes++; if (nNodes >= 128) break; endMarker = FDKreadBits(hBs, 1); } return nNodes; } static void _decodeGains(HANDLE_FDK_BITSTREAM hBs, const GAIN_CODING_PROFILE gainCodingProfile, const int nNodes, GAIN_NODE* pNodes) { int k, deltaGain; Huffman deltaGainCodebook; pNodes[0].gainDb = _decodeGainInitial(hBs, gainCodingProfile); if (gainCodingProfile == GCP_CLIPPING_DUCKING) { deltaGainCodebook = (Huffman)&deltaGain_codingProfile_2_huffman; } else { deltaGainCodebook = (Huffman)&deltaGain_codingProfile_0_1_huffman; } for (k = 1; k < nNodes; k++) { deltaGain = _decodeHuffmanCW(deltaGainCodebook, hBs); if (k >= 16) continue; /* gain_dB_e = 7 */ pNodes[k].gainDb = pNodes[k - 1].gainDb + (FIXP_SGL)(deltaGain << (FRACT_BITS - 1 - 7 - 3)); /* pNodes[k-1].gainDb + 0.125*deltaGain */ } } static void _decodeSlopes(HANDLE_FDK_BITSTREAM hBs, const GAIN_INTERPOLATION_TYPE gainInterpolationType, const int nNodes, GAIN_NODE* pNodes) { int k = 0; if (gainInterpolationType == GIT_SPLINE) { /* decode slope steepness */ for (k = 0; k < nNodes; k++) { _decodeHuffmanCW((Huffman)&slopeSteepness_huffman, hBs); } } } static int _decodeTimeDelta(HANDLE_FDK_BITSTREAM hBs, const int Z) { int prefix, mu; prefix = FDKreadBits(hBs, 2); switch (prefix) { case 0x0: return 1; case 0x1: mu = FDKreadBits(hBs, 2); return mu + 2; case 0x2: mu = FDKreadBits(hBs, 3); return mu + 6; case 0x3: mu = FDKreadBits(hBs, Z); return mu + 14; default: return 0; } } static void _decodeTimes(HANDLE_FDK_BITSTREAM hBs, const int deltaTmin, const int frameSize, const int fullFrame, const int timeOffset, const int Z, const int nNodes, GAIN_NODE* pNodes) { int timeDelta, k; int timeOffs = timeOffset; int frameEndFlag, nodeTimeTmp, nodeResFlag; if (fullFrame == 0) { frameEndFlag = FDKreadBits(hBs, 1); } else { frameEndFlag = 1; } if (frameEndFlag == 1) { /* frameEndFlag == 1 signals that the last node is at the end of the DRC frame */ nodeResFlag = 0; for (k = 0; k < nNodes - 1; k++) { /* decode a delta time value */ timeDelta = _decodeTimeDelta(hBs, Z); if (k >= (16 - 1)) continue; /* frameEndFlag == 1 needs special handling for last node with node * reservoir */ nodeTimeTmp = timeOffs + timeDelta * deltaTmin; if (nodeTimeTmp > frameSize + timeOffset) { if (nodeResFlag == 0) { pNodes[k].time = frameSize + timeOffset; nodeResFlag = 1; } pNodes[k + 1].time = nodeTimeTmp; } else { pNodes[k].time = nodeTimeTmp; } timeOffs = nodeTimeTmp; } if (nodeResFlag == 0) { k = fMin(k, 16 - 1); pNodes[k].time = frameSize + timeOffset; } } else { for (k = 0; k < nNodes; k++) { /* decode a delta time value */ timeDelta = _decodeTimeDelta(hBs, Z); if (k >= 16) continue; pNodes[k].time = timeOffs + timeDelta * deltaTmin; timeOffs = pNodes[k].time; } } } static void _readNodes(HANDLE_FDK_BITSTREAM hBs, GAIN_SET* gainSet, const int frameSize, const int timeDeltaMin, UCHAR* pNNodes, GAIN_NODE* pNodes) { int timeOffset, drcGainCodingMode, nNodes; int Z = _getZ(frameSize / timeDeltaMin); if (gainSet->timeAlignment == 0) { timeOffset = -1; } else { timeOffset = -timeDeltaMin + (timeDeltaMin - 1) / 2; /* timeOffset = - deltaTmin + floor((deltaTmin-1)/2); */ } drcGainCodingMode = FDKreadBits(hBs, 1); if (drcGainCodingMode == 0) { /* "simple" mode: only one node at the end of the frame with slope = 0 */ nNodes = 1; pNodes[0].gainDb = _decodeGainInitial( hBs, (GAIN_CODING_PROFILE)gainSet->gainCodingProfile); pNodes[0].time = frameSize + timeOffset; } else { nNodes = _decodeNNodes(hBs); _decodeSlopes(hBs, (GAIN_INTERPOLATION_TYPE)gainSet->gainInterpolationType, nNodes, pNodes); _decodeTimes(hBs, timeDeltaMin, frameSize, gainSet->fullFrame, timeOffset, Z, nNodes, pNodes); _decodeGains(hBs, (GAIN_CODING_PROFILE)gainSet->gainCodingProfile, nNodes, pNodes); } *pNNodes = (UCHAR)nNodes; } static void _readDrcGainSequence(HANDLE_FDK_BITSTREAM hBs, GAIN_SET* gainSet, const int frameSize, const int timeDeltaMin, UCHAR* pNNodes, GAIN_NODE pNodes[16]) { SHORT timeBufPrevFrame[16], timeBufCurFrame[16]; int nNodesNodeRes, nNodesCur, k, m; if (gainSet->gainCodingProfile == GCP_CONSTANT) { *pNNodes = 1; pNodes[0].time = frameSize - 1; pNodes[0].gainDb = (FIXP_SGL)0; } else { _readNodes(hBs, gainSet, frameSize, timeDeltaMin, pNNodes, pNodes); /* count number of nodes in node reservoir */ nNodesNodeRes = 0; nNodesCur = 0; /* count and buffer nodes from node reservoir */ for (k = 0; k < *pNNodes; k++) { if (k >= 16) continue; if (pNodes[k].time >= frameSize) { /* write node reservoir times into buffer */ timeBufPrevFrame[nNodesNodeRes] = pNodes[k].time; nNodesNodeRes++; } else { /* times from current frame */ timeBufCurFrame[nNodesCur] = pNodes[k].time; nNodesCur++; } } /* compose right time order (bit reservoir first) */ for (k = 0; k < nNodesNodeRes; k++) { /* subtract two time frameSize: one to remove node reservoir offset and * one to get the negative index relative to the current frame */ pNodes[k].time = timeBufPrevFrame[k] - 2 * frameSize; } /* ...and times from current frame */ for (m = 0; m < nNodesCur; m++, k++) { pNodes[k].time = timeBufCurFrame[m]; } } } static DRC_ERROR _readUniDrcGainExtension(HANDLE_FDK_BITSTREAM hBs, UNI_DRC_GAIN_EXTENSION* pExt) { DRC_ERROR err = DE_OK; int k, bitSizeLen, extSizeBits, bitSize; k = 0; pExt->uniDrcGainExtType[k] = FDKreadBits(hBs, 4); while (pExt->uniDrcGainExtType[k] != UNIDRCGAINEXT_TERM) { if (k >= (8 - 1)) return DE_MEMORY_ERROR; bitSizeLen = FDKreadBits(hBs, 3); extSizeBits = bitSizeLen + 4; bitSize = FDKreadBits(hBs, extSizeBits); pExt->extBitSize[k] = bitSize + 1; switch (pExt->uniDrcGainExtType[k]) { /* add future extensions here */ default: FDKpushFor(hBs, pExt->extBitSize[k]); break; } k++; pExt->uniDrcGainExtType[k] = FDKreadBits(hBs, 4); } return err; } DRC_ERROR drcDec_readUniDrcGain(HANDLE_FDK_BITSTREAM hBs, HANDLE_UNI_DRC_CONFIG hUniDrcConfig, const int frameSize, const int deltaTminDefault, HANDLE_UNI_DRC_GAIN hUniDrcGain) { DRC_ERROR err = DE_OK; int seq, gainSequenceCount; DRC_COEFFICIENTS_UNI_DRC* pCoef = selectDrcCoefficients(hUniDrcConfig, LOCATION_SELECTED); if (hUniDrcGain == NULL) return DE_NOT_OK; hUniDrcGain->status = 0; if (pCoef) { gainSequenceCount = fMin(pCoef->gainSequenceCount, (UCHAR)12); } else { gainSequenceCount = 0; } for (seq = 0; seq < gainSequenceCount; seq++) { UCHAR index = pCoef->gainSetIndexForGainSequence[seq]; GAIN_SET* gainSet; int timeDeltaMin; UCHAR tmpNNodes = 0; GAIN_NODE tmpNodes[16]; if ((index >= pCoef->gainSetCount) || (index >= 12)) return DE_NOT_OK; gainSet = &(pCoef->gainSet[index]); timeDeltaMin = _getTimeDeltaMin(gainSet, deltaTminDefault); _readDrcGainSequence(hBs, gainSet, frameSize, timeDeltaMin, &tmpNNodes, tmpNodes); hUniDrcGain->nNodes[seq] = tmpNNodes; FDKmemcpy(hUniDrcGain->gainNode[seq], tmpNodes, fMin(tmpNNodes, (UCHAR)16) * sizeof(GAIN_NODE)); } hUniDrcGain->uniDrcGainExtPresent = FDKreadBits(hBs, 1); if (hUniDrcGain->uniDrcGainExtPresent == 1) { err = _readUniDrcGainExtension(hBs, &(hUniDrcGain->uniDrcGainExtension)); if (err) return err; } if (err == DE_OK && gainSequenceCount > 0) { hUniDrcGain->status = 1; } return err; } /****************/ /* uniDrcConfig */ /****************/ static void _decodeDuckingModification(HANDLE_FDK_BITSTREAM hBs, DUCKING_MODIFICATION* pDMod, int isBox) { int bsDuckingScaling, sigma, mu; if (isBox) FDKpushFor(hBs, 7); /* reserved */ pDMod->duckingScalingPresent = FDKreadBits(hBs, 1); if (pDMod->duckingScalingPresent) { if (isBox) FDKpushFor(hBs, 4); /* reserved */ bsDuckingScaling = FDKreadBits(hBs, 4); sigma = bsDuckingScaling >> 3; mu = bsDuckingScaling & 0x7; if (sigma) { pDMod->duckingScaling = (FIXP_SGL)( (7 - mu) << (FRACT_BITS - 1 - 3 - 2)); /* 1.0 - 0.125 * (1 + mu); */ } else { pDMod->duckingScaling = (FIXP_SGL)( (9 + mu) << (FRACT_BITS - 1 - 3 - 2)); /* 1.0 + 0.125 * (1 + mu); */ } } else { pDMod->duckingScaling = (FIXP_SGL)(1 << (FRACT_BITS - 1 - 2)); /* 1.0 */ } } static void _decodeGainModification(HANDLE_FDK_BITSTREAM hBs, const int version, int bandCount, GAIN_MODIFICATION* pGMod, int isBox) { int sign, bsGainOffset, bsAttenuationScaling, bsAmplificationScaling; if (version > 0) { int b, shapeFilterPresent; if (isBox) { FDKpushFor(hBs, 4); /* reserved */ bandCount = FDKreadBits(hBs, 4); } for (b = 0; b < bandCount; b++) { if (isBox) { FDKpushFor(hBs, 4); /* reserved */ pGMod[b].targetCharacteristicLeftPresent = FDKreadBits(hBs, 1); pGMod[b].targetCharacteristicRightPresent = FDKreadBits(hBs, 1); pGMod[b].gainScalingPresent = FDKreadBits(hBs, 1); pGMod[b].gainOffsetPresent = FDKreadBits(hBs, 1); } if (!isBox) pGMod[b].targetCharacteristicLeftPresent = FDKreadBits(hBs, 1); if (pGMod[b].targetCharacteristicLeftPresent) { if (isBox) FDKpushFor(hBs, 4); /* reserved */ pGMod[b].targetCharacteristicLeftIndex = FDKreadBits(hBs, 4); } if (!isBox) pGMod[b].targetCharacteristicRightPresent = FDKreadBits(hBs, 1); if (pGMod[b].targetCharacteristicRightPresent) { if (isBox) FDKpushFor(hBs, 4); /* reserved */ pGMod[b].targetCharacteristicRightIndex = FDKreadBits(hBs, 4); } if (!isBox) pGMod[b].gainScalingPresent = FDKreadBits(hBs, 1); if (pGMod[b].gainScalingPresent) { bsAttenuationScaling = FDKreadBits(hBs, 4); pGMod[b].attenuationScaling = (FIXP_SGL)( bsAttenuationScaling << (FRACT_BITS - 1 - 3 - 2)); /* bsAttenuationScaling * 0.125; */ bsAmplificationScaling = FDKreadBits(hBs, 4); pGMod[b].amplificationScaling = (FIXP_SGL)( bsAmplificationScaling << (FRACT_BITS - 1 - 3 - 2)); /* bsAmplificationScaling * 0.125; */ } if (!isBox) pGMod[b].gainOffsetPresent = FDKreadBits(hBs, 1); if (pGMod[b].gainOffsetPresent) { if (isBox) FDKpushFor(hBs, 2); /* reserved */ sign = FDKreadBits(hBs, 1); bsGainOffset = FDKreadBits(hBs, 5); pGMod[b].gainOffset = (FIXP_SGL)( (1 + bsGainOffset) << (FRACT_BITS - 1 - 2 - 4)); /* (1+bsGainOffset) * 0.25; */ if (sign) { pGMod[b].gainOffset = -pGMod[b].gainOffset; } } } if (bandCount == 1) { shapeFilterPresent = FDKreadBits(hBs, 1); if (shapeFilterPresent) { if (isBox) FDKpushFor(hBs, 3); /* reserved */ FDKpushFor(hBs, 4); /* pGMod->shapeFilterIndex */ } else { if (isBox) FDKpushFor(hBs, 7); /* reserved */ } } } else { int b, gainScalingPresent, gainOffsetPresent; FIXP_SGL attenuationScaling = FL2FXCONST_SGL(1.0f / (float)(1 << 2)), amplificationScaling = FL2FXCONST_SGL(1.0f / (float)(1 << 2)), gainOffset = (FIXP_SGL)0; if (isBox) FDKpushFor(hBs, 7); /* reserved */ gainScalingPresent = FDKreadBits(hBs, 1); if (gainScalingPresent) { bsAttenuationScaling = FDKreadBits(hBs, 4); attenuationScaling = (FIXP_SGL)( bsAttenuationScaling << (FRACT_BITS - 1 - 3 - 2)); /* bsAttenuationScaling * 0.125; */ bsAmplificationScaling = FDKreadBits(hBs, 4); amplificationScaling = (FIXP_SGL)( bsAmplificationScaling << (FRACT_BITS - 1 - 3 - 2)); /* bsAmplificationScaling * 0.125; */ } if (isBox) FDKpushFor(hBs, 7); /* reserved */ gainOffsetPresent = FDKreadBits(hBs, 1); if (gainOffsetPresent) { if (isBox) FDKpushFor(hBs, 2); /* reserved */ sign = FDKreadBits(hBs, 1); bsGainOffset = FDKreadBits(hBs, 5); gainOffset = (FIXP_SGL)((1 + bsGainOffset) << (FRACT_BITS - 1 - 2 - 4)); /* (1+bsGainOffset) * 0.25; */ if (sign) { gainOffset = -gainOffset; } } for (b = 0; b < 4; b++) { pGMod[b].targetCharacteristicLeftPresent = 0; pGMod[b].targetCharacteristicRightPresent = 0; pGMod[b].gainScalingPresent = gainScalingPresent; pGMod[b].attenuationScaling = attenuationScaling; pGMod[b].amplificationScaling = amplificationScaling; pGMod[b].gainOffsetPresent = gainOffsetPresent; pGMod[b].gainOffset = gainOffset; } } } static void _readDrcCharacteristic(HANDLE_FDK_BITSTREAM hBs, const int version, DRC_CHARACTERISTIC* pDChar, int isBox) { if (version == 0) { if (isBox) FDKpushFor(hBs, 1); /* reserved */ pDChar->cicpIndex = FDKreadBits(hBs, 7); if (pDChar->cicpIndex > 0) { pDChar->present = 1; pDChar->isCICP = 1; } else { pDChar->present = 0; } } else { pDChar->present = FDKreadBits(hBs, 1); if (isBox) pDChar->isCICP = FDKreadBits(hBs, 1); if (pDChar->present) { if (!isBox) pDChar->isCICP = FDKreadBits(hBs, 1); if (pDChar->isCICP) { if (isBox) FDKpushFor(hBs, 1); /* reserved */ pDChar->cicpIndex = FDKreadBits(hBs, 7); } else { pDChar->custom.left = FDKreadBits(hBs, 4); pDChar->custom.right = FDKreadBits(hBs, 4); } } } } static void _readBandBorder(HANDLE_FDK_BITSTREAM hBs, BAND_BORDER* pBBord, int drcBandType, int isBox) { if (drcBandType) { if (isBox) FDKpushFor(hBs, 4); /* reserved */ pBBord->crossoverFreqIndex = FDKreadBits(hBs, 4); } else { if (isBox) FDKpushFor(hBs, 6); /* reserved */ pBBord->startSubBandIndex = FDKreadBits(hBs, 10); } } static DRC_ERROR _readGainSet(HANDLE_FDK_BITSTREAM hBs, const int version, int* gainSequenceIndex, GAIN_SET* pGSet, int isBox) { if (isBox) FDKpushFor(hBs, 2); /* reserved */ pGSet->gainCodingProfile = FDKreadBits(hBs, 2); pGSet->gainInterpolationType = FDKreadBits(hBs, 1); pGSet->fullFrame = FDKreadBits(hBs, 1); pGSet->timeAlignment = FDKreadBits(hBs, 1); pGSet->timeDeltaMinPresent = FDKreadBits(hBs, 1); if (pGSet->timeDeltaMinPresent) { int bsTimeDeltaMin; if (isBox) FDKpushFor(hBs, 5); /* reserved */ bsTimeDeltaMin = FDKreadBits(hBs, 11); pGSet->timeDeltaMin = bsTimeDeltaMin + 1; } if (pGSet->gainCodingProfile != GCP_CONSTANT) { int i; if (isBox) FDKpushFor(hBs, 3); /* reserved */ pGSet->bandCount = FDKreadBits(hBs, 4); if (pGSet->bandCount > 4) return DE_MEMORY_ERROR; if ((pGSet->bandCount > 1) || isBox) { pGSet->drcBandType = FDKreadBits(hBs, 1); } for (i = 0; i < pGSet->bandCount; i++) { if (version == 0) { *gainSequenceIndex = (*gainSequenceIndex) + 1; } else { int indexPresent; indexPresent = (isBox) ? 1 : FDKreadBits(hBs, 1); if (indexPresent) { int bsIndex; bsIndex = FDKreadBits(hBs, 6); *gainSequenceIndex = bsIndex; } else { *gainSequenceIndex = (*gainSequenceIndex) + 1; } } pGSet->gainSequenceIndex[i] = *gainSequenceIndex; _readDrcCharacteristic(hBs, version, &(pGSet->drcCharacteristic[i]), isBox); } for (i = 1; i < pGSet->bandCount; i++) { _readBandBorder(hBs, &(pGSet->bandBorder[i]), pGSet->drcBandType, isBox); } } else { pGSet->bandCount = 1; *gainSequenceIndex = (*gainSequenceIndex) + 1; pGSet->gainSequenceIndex[0] = *gainSequenceIndex; } return DE_OK; } static DRC_ERROR _readCustomDrcCharacteristic(HANDLE_FDK_BITSTREAM hBs, const CHARACTERISTIC_SIDE side, UCHAR* pCharacteristicFormat, CUSTOM_DRC_CHAR* pCChar, int isBox) { if (isBox) FDKpushFor(hBs, 7); /* reserved */ *pCharacteristicFormat = FDKreadBits(hBs, 1); if (*pCharacteristicFormat == CF_SIGMOID) { int bsGain, bsIoRatio, bsExp; if (isBox) FDKpushFor(hBs, 1); /* reserved */ bsGain = FDKreadBits(hBs, 6); if (side == CS_LEFT) { pCChar->sigmoid.gain = (FIXP_SGL)(bsGain << (FRACT_BITS - 1 - 6)); } else { pCChar->sigmoid.gain = (FIXP_SGL)(-bsGain << (FRACT_BITS - 1 - 6)); } bsIoRatio = FDKreadBits(hBs, 4); /* pCChar->sigmoid.ioRatio = 0.05 + 0.15 * bsIoRatio; */ pCChar->sigmoid.ioRatio = FL2FXCONST_SGL(0.05f / (float)(1 << 2)) + (FIXP_SGL)((((3 * bsIoRatio) << (FRACT_BITS - 1)) / 5) >> 4); bsExp = FDKreadBits(hBs, 4); if (bsExp < 15) { pCChar->sigmoid.exp = (FIXP_SGL)((1 + 2 * bsExp) << (FRACT_BITS - 1 - 5)); } else { pCChar->sigmoid.exp = (FIXP_SGL)MAXVAL_SGL; /* represents infinity */ } pCChar->sigmoid.flipSign = FDKreadBits(hBs, 1); } else { /* CF_NODES */ int i, bsCharacteristicNodeCount, bsNodeLevelDelta, bsNodeGain; if (isBox) FDKpushFor(hBs, 6); /* reserved */ bsCharacteristicNodeCount = FDKreadBits(hBs, 2); pCChar->nodes.characteristicNodeCount = bsCharacteristicNodeCount + 1; if (pCChar->nodes.characteristicNodeCount > 4) return DE_MEMORY_ERROR; pCChar->nodes.nodeLevel[0] = DRC_INPUT_LOUDNESS_TARGET_SGL; pCChar->nodes.nodeGain[0] = (FIXP_SGL)0; for (i = 0; i < pCChar->nodes.characteristicNodeCount; i++) { if (isBox) FDKpushFor(hBs, 3); /* reserved */ bsNodeLevelDelta = FDKreadBits(hBs, 5); if (side == CS_LEFT) { pCChar->nodes.nodeLevel[i + 1] = pCChar->nodes.nodeLevel[i] - (FIXP_SGL)((1 + bsNodeLevelDelta) << (FRACT_BITS - 1 - 7)); } else { pCChar->nodes.nodeLevel[i + 1] = pCChar->nodes.nodeLevel[i] + (FIXP_SGL)((1 + bsNodeLevelDelta) << (FRACT_BITS - 1 - 7)); } bsNodeGain = FDKreadBits(hBs, 8); pCChar->nodes.nodeGain[i + 1] = (FIXP_SGL)( (bsNodeGain - 128) << (FRACT_BITS - 1 - 1 - 7)); /* 0.5f * bsNodeGain - 64.0f; */ } } return DE_OK; } static void _skipLoudEqInstructions(HANDLE_FDK_BITSTREAM hBs) { int i; int downmixIdPresent, additionalDownmixIdPresent, additionalDownmixIdCount = 0; int drcSetIdPresent, additionalDrcSetIdPresent, additionalDrcSetIdCount = 0; int eqSetIdPresent, additionalEqSetIdPresent, additionalEqSetIdCount = 0; int loudEqGainSequenceCount, drcCharacteristicFormatIsCICP; FDKpushFor(hBs, 4); /* loudEqSetId */ FDKpushFor(hBs, 4); /* drcLocation */ downmixIdPresent = FDKreadBits(hBs, 1); if (downmixIdPresent) { FDKpushFor(hBs, 7); /* downmixId */ additionalDownmixIdPresent = FDKreadBits(hBs, 1); if (additionalDownmixIdPresent) { additionalDownmixIdCount = FDKreadBits(hBs, 7); for (i = 0; i < additionalDownmixIdCount; i++) { FDKpushFor(hBs, 7); /* additionalDownmixId */ } } } drcSetIdPresent = FDKreadBits(hBs, 1); if (drcSetIdPresent) { FDKpushFor(hBs, 6); /* drcSetId */ additionalDrcSetIdPresent = FDKreadBits(hBs, 1); if (additionalDrcSetIdPresent) { additionalDrcSetIdCount = FDKreadBits(hBs, 6); for (i = 0; i < additionalDrcSetIdCount; i++) { FDKpushFor(hBs, 6); /* additionalDrcSetId; */ } } } eqSetIdPresent = FDKreadBits(hBs, 1); if (eqSetIdPresent) { FDKpushFor(hBs, 6); /* eqSetId */ additionalEqSetIdPresent = FDKreadBits(hBs, 1); if (additionalEqSetIdPresent) { additionalEqSetIdCount = FDKreadBits(hBs, 6); for (i = 0; i < additionalEqSetIdCount; i++) { FDKpushFor(hBs, 6); /* additionalEqSetId; */ } } } FDKpushFor(hBs, 1); /* loudnessAfterDrc */ FDKpushFor(hBs, 1); /* loudnessAfterEq */ loudEqGainSequenceCount = FDKreadBits(hBs, 6); for (i = 0; i < loudEqGainSequenceCount; i++) { FDKpushFor(hBs, 6); /* gainSequenceIndex */ drcCharacteristicFormatIsCICP = FDKreadBits(hBs, 1); if (drcCharacteristicFormatIsCICP) { FDKpushFor(hBs, 7); /* drcCharacteristic */ } else { FDKpushFor(hBs, 4); /* drcCharacteristicLeftIndex */ FDKpushFor(hBs, 4); /* drcCharacteristicRightIndex */ } FDKpushFor(hBs, 6); /* frequencyRangeIndex */ FDKpushFor(hBs, 3); /* bsLoudEqScaling */ FDKpushFor(hBs, 5); /* bsLoudEqOffset */ } } static void _skipEqSubbandGainSpline(HANDLE_FDK_BITSTREAM hBs) { int nEqNodes, k, bits; nEqNodes = FDKreadBits(hBs, 5); nEqNodes += 2; for (k = 0; k < nEqNodes; k++) { bits = FDKreadBits(hBs, 1); if (!bits) { FDKpushFor(hBs, 4); } } FDKpushFor(hBs, 4 * (nEqNodes - 1)); bits = FDKreadBits(hBs, 2); switch (bits) { case 0: FDKpushFor(hBs, 5); break; case 1: case 2: FDKpushFor(hBs, 4); break; case 3: FDKpushFor(hBs, 3); break; } FDKpushFor(hBs, 5 * (nEqNodes - 1)); } static void _skipEqCoefficients(HANDLE_FDK_BITSTREAM hBs) { int j, k; int eqDelayMaxPresent; int uniqueFilterBlockCount, filterElementCount, filterElementGainPresent; int uniqueTdFilterElementCount, eqFilterFormat, bsRealZeroRadiusOneCount, realZeroCount, genericZeroCount, realPoleCount, complexPoleCount, firFilterOrder; int uniqueEqSubbandGainsCount, eqSubbandGainRepresentation, eqSubbandGainCount; int eqSubbandGainFormat; eqDelayMaxPresent = FDKreadBits(hBs, 1); if (eqDelayMaxPresent) { FDKpushFor(hBs, 8); /* bsEqDelayMax */ } uniqueFilterBlockCount = FDKreadBits(hBs, 6); for (j = 0; j < uniqueFilterBlockCount; j++) { filterElementCount = FDKreadBits(hBs, 6); for (k = 0; k < filterElementCount; k++) { FDKpushFor(hBs, 6); /* filterElementIndex */ filterElementGainPresent = FDKreadBits(hBs, 1); if (filterElementGainPresent) { FDKpushFor(hBs, 10); /* bsFilterElementGain */ } } } uniqueTdFilterElementCount = FDKreadBits(hBs, 6); for (j = 0; j < uniqueTdFilterElementCount; j++) { eqFilterFormat = FDKreadBits(hBs, 1); if (eqFilterFormat == 0) { /* pole/zero */ bsRealZeroRadiusOneCount = FDKreadBits(hBs, 3); realZeroCount = FDKreadBits(hBs, 6); genericZeroCount = FDKreadBits(hBs, 6); realPoleCount = FDKreadBits(hBs, 4); complexPoleCount = FDKreadBits(hBs, 4); FDKpushFor(hBs, 2 * bsRealZeroRadiusOneCount * 1); FDKpushFor(hBs, realZeroCount * 8); FDKpushFor(hBs, genericZeroCount * 14); FDKpushFor(hBs, realPoleCount * 8); FDKpushFor(hBs, complexPoleCount * 14); } else { /* FIR coefficients */ firFilterOrder = FDKreadBits(hBs, 7); FDKpushFor(hBs, 1); FDKpushFor(hBs, (firFilterOrder / 2 + 1) * 11); } } uniqueEqSubbandGainsCount = FDKreadBits(hBs, 6); if (uniqueEqSubbandGainsCount > 0) { eqSubbandGainRepresentation = FDKreadBits(hBs, 1); eqSubbandGainFormat = FDKreadBits(hBs, 4); switch (eqSubbandGainFormat) { case GF_QMF32: eqSubbandGainCount = 32; break; case GF_QMFHYBRID39: eqSubbandGainCount = 39; break; case GF_QMF64: eqSubbandGainCount = 64; break; case GF_QMFHYBRID71: eqSubbandGainCount = 71; break; case GF_QMF128: eqSubbandGainCount = 128; break; case GF_QMFHYBRID135: eqSubbandGainCount = 135; break; case GF_UNIFORM: default: eqSubbandGainCount = FDKreadBits(hBs, 8); eqSubbandGainCount++; break; } for (k = 0; k < uniqueEqSubbandGainsCount; k++) { if (eqSubbandGainRepresentation == 1) { _skipEqSubbandGainSpline(hBs); } else { FDKpushFor(hBs, eqSubbandGainCount * 9); } } } } static void _skipTdFilterCascade(HANDLE_FDK_BITSTREAM hBs, const int eqChannelGroupCount) { int i, eqCascadeGainPresent, filterBlockCount, eqPhaseAlignmentPresent; for (i = 0; i < eqChannelGroupCount; i++) { eqCascadeGainPresent = FDKreadBits(hBs, 1); if (eqCascadeGainPresent) { FDKpushFor(hBs, 10); /* bsEqCascadeGain */ } filterBlockCount = FDKreadBits(hBs, 4); FDKpushFor(hBs, filterBlockCount * 7); /* filterBlockIndex */ } eqPhaseAlignmentPresent = FDKreadBits(hBs, 1); { if (eqPhaseAlignmentPresent) { for (i = 0; i < eqChannelGroupCount; i++) { FDKpushFor(hBs, (eqChannelGroupCount - i - 1) * 1); } } } } static DRC_ERROR _skipEqInstructions(HANDLE_FDK_BITSTREAM hBs, HANDLE_UNI_DRC_CONFIG hUniDrcConfig) { DRC_ERROR err = DE_OK; int c, i, k, channelCount; int downmixIdPresent, downmixId, eqApplyToDownmix, additionalDownmixIdPresent, additionalDownmixIdCount = 0; int additionalDrcSetIdPresent, additionalDrcSetIdCount; int dependsOnEqSetPresent, eqChannelGroupCount, tdFilterCascadePresent, subbandGainsPresent, eqTransitionDurationPresent; UCHAR eqChannelGroupForChannel[8]; FDKpushFor(hBs, 6); /* eqSetId */ FDKpushFor(hBs, 4); /* eqSetComplexityLevel */ downmixIdPresent = FDKreadBits(hBs, 1); if (downmixIdPresent) { downmixId = FDKreadBits(hBs, 7); eqApplyToDownmix = FDKreadBits(hBs, 1); additionalDownmixIdPresent = FDKreadBits(hBs, 1); if (additionalDownmixIdPresent) { additionalDownmixIdCount = FDKreadBits(hBs, 7); FDKpushFor(hBs, additionalDownmixIdCount * 7); /* additionalDownmixId */ } } else { downmixId = 0; eqApplyToDownmix = 0; } FDKpushFor(hBs, 6); /* drcSetId */ additionalDrcSetIdPresent = FDKreadBits(hBs, 1); if (additionalDrcSetIdPresent) { additionalDrcSetIdCount = FDKreadBits(hBs, 6); for (i = 0; i < additionalDrcSetIdCount; i++) { FDKpushFor(hBs, 6); /* additionalDrcSetId */ } } FDKpushFor(hBs, 16); /* eqSetPurpose */ dependsOnEqSetPresent = FDKreadBits(hBs, 1); if (dependsOnEqSetPresent) { FDKpushFor(hBs, 6); /* dependsOnEqSet */ } else { FDKpushFor(hBs, 1); /* noIndependentEqUse */ } channelCount = hUniDrcConfig->channelLayout.baseChannelCount; if ((downmixIdPresent == 1) && (eqApplyToDownmix == 1) && (downmixId != 0) && (downmixId != DOWNMIX_ID_ANY_DOWNMIX) && (additionalDownmixIdCount == 0)) { DOWNMIX_INSTRUCTIONS* pDown = selectDownmixInstructions(hUniDrcConfig, downmixId); if (pDown == NULL) return DE_NOT_OK; channelCount = pDown->targetChannelCount; /* targetChannelCountFromDownmixId*/ } else if ((downmixId == DOWNMIX_ID_ANY_DOWNMIX) || (additionalDownmixIdCount > 1)) { channelCount = 1; } eqChannelGroupCount = 0; for (c = 0; c < channelCount; c++) { int newGroup = 1; if (c >= 8) return DE_MEMORY_ERROR; eqChannelGroupForChannel[c] = FDKreadBits(hBs, 7); for (k = 0; k < c; k++) { if (eqChannelGroupForChannel[c] == eqChannelGroupForChannel[k]) { newGroup = 0; } } if (newGroup == 1) { eqChannelGroupCount += 1; } } tdFilterCascadePresent = FDKreadBits(hBs, 1); if (tdFilterCascadePresent) { _skipTdFilterCascade(hBs, eqChannelGroupCount); } subbandGainsPresent = FDKreadBits(hBs, 1); if (subbandGainsPresent) { FDKpushFor(hBs, eqChannelGroupCount * 6); /* subbandGainsIndex */ } eqTransitionDurationPresent = FDKreadBits(hBs, 1); if (eqTransitionDurationPresent) { FDKpushFor(hBs, 5); /* bsEqTransitionDuration */ } return err; } static void _skipDrcCoefficientsBasic(HANDLE_FDK_BITSTREAM hBs) { FDKpushFor(hBs, 4); /* drcLocation */ FDKpushFor(hBs, 7); /* drcCharacteristic */ } static DRC_ERROR _readDrcCoefficientsUniDrc(HANDLE_FDK_BITSTREAM hBs, const int version, DRC_COEFFICIENTS_UNI_DRC* pCoef) { DRC_ERROR err = DE_OK; int i, bsDrcFrameSize; int gainSequenceIndex = -1; pCoef->drcLocation = FDKreadBits(hBs, 4); pCoef->drcFrameSizePresent = FDKreadBits(hBs, 1); if (pCoef->drcFrameSizePresent == 1) { bsDrcFrameSize = FDKreadBits(hBs, 15); pCoef->drcFrameSize = bsDrcFrameSize + 1; } if (version == 0) { int gainSequenceCount = 0, gainSetCount; pCoef->characteristicLeftCount = 0; pCoef->characteristicRightCount = 0; gainSetCount = FDKreadBits(hBs, 6); pCoef->gainSetCount = fMin(gainSetCount, 12); for (i = 0; i < gainSetCount; i++) { GAIN_SET tmpGset; FDKmemclear(&tmpGset, sizeof(GAIN_SET)); err = _readGainSet(hBs, version, &gainSequenceIndex, &tmpGset, 0); if (err) return err; gainSequenceCount += tmpGset.bandCount; if (i >= 12) continue; pCoef->gainSet[i] = tmpGset; } pCoef->gainSequenceCount = gainSequenceCount; } else { /* (version == 1) */ UCHAR drcCharacteristicLeftPresent, drcCharacteristicRightPresent; UCHAR shapeFiltersPresent, shapeFilterCount, tmpPresent; int gainSetCount; drcCharacteristicLeftPresent = FDKreadBits(hBs, 1); if (drcCharacteristicLeftPresent) { pCoef->characteristicLeftCount = FDKreadBits(hBs, 4); if ((pCoef->characteristicLeftCount + 1) > 16) return DE_MEMORY_ERROR; for (i = 0; i < pCoef->characteristicLeftCount; i++) { err = _readCustomDrcCharacteristic( hBs, CS_LEFT, &(pCoef->characteristicLeftFormat[i + 1]), &(pCoef->customCharacteristicLeft[i + 1]), 0); if (err) return err; } } drcCharacteristicRightPresent = FDKreadBits(hBs, 1); if (drcCharacteristicRightPresent) { pCoef->characteristicRightCount = FDKreadBits(hBs, 4); if ((pCoef->characteristicRightCount + 1) > 16) return DE_MEMORY_ERROR; for (i = 0; i < pCoef->characteristicRightCount; i++) { err = _readCustomDrcCharacteristic( hBs, CS_RIGHT, &(pCoef->characteristicRightFormat[i + 1]), &(pCoef->customCharacteristicRight[i + 1]), 0); if (err) return err; } } shapeFiltersPresent = FDKreadBits(hBs, 1); if (shapeFiltersPresent) { shapeFilterCount = FDKreadBits(hBs, 4); for (i = 0; i < shapeFilterCount; i++) { tmpPresent = FDKreadBits(hBs, 1); if (tmpPresent) /* lfCutParams */ FDKpushFor(hBs, 5); tmpPresent = FDKreadBits(hBs, 1); if (tmpPresent) /* lfBoostParams */ FDKpushFor(hBs, 5); tmpPresent = FDKreadBits(hBs, 1); if (tmpPresent) /* hfCutParams */ FDKpushFor(hBs, 5); tmpPresent = FDKreadBits(hBs, 1); if (tmpPresent) /* hfBoostParams */ FDKpushFor(hBs, 5); } } pCoef->gainSequenceCount = FDKreadBits(hBs, 6); gainSetCount = FDKreadBits(hBs, 6); pCoef->gainSetCount = fMin(gainSetCount, 12); for (i = 0; i < gainSetCount; i++) { GAIN_SET tmpGset; FDKmemclear(&tmpGset, sizeof(GAIN_SET)); err = _readGainSet(hBs, version, &gainSequenceIndex, &tmpGset, 0); if (err) return err; if (i >= 12) continue; pCoef->gainSet[i] = tmpGset; } } for (i = 0; i < 12; i++) { pCoef->gainSetIndexForGainSequence[i] = 255; } for (i = 0; i < pCoef->gainSetCount; i++) { int b; for (b = 0; b < pCoef->gainSet[i].bandCount; b++) { if (pCoef->gainSet[i].gainSequenceIndex[b] >= 12) continue; pCoef->gainSetIndexForGainSequence[pCoef->gainSet[i] .gainSequenceIndex[b]] = i; } } return err; } static void _skipDrcInstructionsBasic(HANDLE_FDK_BITSTREAM hBs) { int drcSetEffect; int additionalDownmixIdPresent, additionalDownmixIdCount, limiterPeakTargetPresent; int drcSetTargetLoudnessPresent, drcSetTargetLoudnessValueLowerPresent; FDKpushFor(hBs, 6); /* drcSetId */ FDKpushFor(hBs, 4); /* drcLocation */ FDKpushFor(hBs, 7); /* downmixId */ additionalDownmixIdPresent = FDKreadBits(hBs, 1); if (additionalDownmixIdPresent) { additionalDownmixIdCount = FDKreadBits(hBs, 3); FDKpushFor(hBs, 7 * additionalDownmixIdCount); /* additionalDownmixId */ } drcSetEffect = FDKreadBits(hBs, 16); if (!(drcSetEffect & (EB_DUCK_OTHER | EB_DUCK_SELF))) { limiterPeakTargetPresent = FDKreadBits(hBs, 1); if (limiterPeakTargetPresent) { FDKpushFor(hBs, 8); /* bsLimiterPeakTarget */ } } drcSetTargetLoudnessPresent = FDKreadBits(hBs, 1); if (drcSetTargetLoudnessPresent) { FDKpushFor(hBs, 6); /* bsDrcSetTargetLoudnessValueUpper */ drcSetTargetLoudnessValueLowerPresent = FDKreadBits(hBs, 1); if (drcSetTargetLoudnessValueLowerPresent) { FDKpushFor(hBs, 6); /* bsDrcSetTargetLoudnessValueLower */ } } } static DRC_ERROR _readDrcInstructionsUniDrc(HANDLE_FDK_BITSTREAM hBs, const int version, HANDLE_UNI_DRC_CONFIG hUniDrcConfig, DRC_INSTRUCTIONS_UNI_DRC* pInst) { DRC_ERROR err = DE_OK; int i, g, c; int downmixIdPresent, additionalDownmixIdPresent, additionalDownmixIdCount; int bsLimiterPeakTarget, channelCount; DRC_COEFFICIENTS_UNI_DRC* pCoef = NULL; int repeatParameters, bsRepeatParametersCount; int repeatSequenceIndex, bsRepeatSequenceCount; SCHAR* gainSetIndex = pInst->gainSetIndex; SCHAR channelGroupForChannel[8]; DUCKING_MODIFICATION duckingModificationForChannelGroup[8]; pInst->drcSetId = FDKreadBits(hBs, 6); if (version == 0) { /* Assume all v0 DRC sets to be manageable in terms of complexity */ pInst->drcSetComplexityLevel = 2; } else { pInst->drcSetComplexityLevel = FDKreadBits(hBs, 4); } pInst->drcLocation = FDKreadBits(hBs, 4); if (version == 0) { downmixIdPresent = 1; } else { downmixIdPresent = FDKreadBits(hBs, 1); } if (downmixIdPresent) { pInst->downmixId[0] = FDKreadBits(hBs, 7); if (version == 0) { if (pInst->downmixId[0] == 0) pInst->drcApplyToDownmix = 0; else pInst->drcApplyToDownmix = 1; } else { pInst->drcApplyToDownmix = FDKreadBits(hBs, 1); } additionalDownmixIdPresent = FDKreadBits(hBs, 1); if (additionalDownmixIdPresent) { additionalDownmixIdCount = FDKreadBits(hBs, 3); if ((1 + additionalDownmixIdCount) > 8) return DE_MEMORY_ERROR; for (i = 0; i < additionalDownmixIdCount; i++) { pInst->downmixId[i + 1] = FDKreadBits(hBs, 7); } pInst->downmixIdCount = 1 + additionalDownmixIdCount; } else { pInst->downmixIdCount = 1; } } else { pInst->downmixId[0] = 0; pInst->downmixIdCount = 1; } pInst->drcSetEffect = FDKreadBits(hBs, 16); if ((pInst->drcSetEffect & (EB_DUCK_OTHER | EB_DUCK_SELF)) == 0) { pInst->limiterPeakTargetPresent = FDKreadBits(hBs, 1); if (pInst->limiterPeakTargetPresent) { bsLimiterPeakTarget = FDKreadBits(hBs, 8); pInst->limiterPeakTarget = -(FIXP_SGL)( bsLimiterPeakTarget << (FRACT_BITS - 1 - 3 - 5)); /* - bsLimiterPeakTarget * 0.125; */ } } pInst->drcSetTargetLoudnessPresent = FDKreadBits(hBs, 1); /* set default values */ pInst->drcSetTargetLoudnessValueUpper = 0; pInst->drcSetTargetLoudnessValueLower = -63; if (pInst->drcSetTargetLoudnessPresent == 1) { int bsDrcSetTargetLoudnessValueUpper, bsDrcSetTargetLoudnessValueLower; int drcSetTargetLoudnessValueLowerPresent; bsDrcSetTargetLoudnessValueUpper = FDKreadBits(hBs, 6); pInst->drcSetTargetLoudnessValueUpper = bsDrcSetTargetLoudnessValueUpper - 63; drcSetTargetLoudnessValueLowerPresent = FDKreadBits(hBs, 1); if (drcSetTargetLoudnessValueLowerPresent == 1) { bsDrcSetTargetLoudnessValueLower = FDKreadBits(hBs, 6); pInst->drcSetTargetLoudnessValueLower = bsDrcSetTargetLoudnessValueLower - 63; } } pInst->dependsOnDrcSetPresent = FDKreadBits(hBs, 1); pInst->noIndependentUse = 0; if (pInst->dependsOnDrcSetPresent) { pInst->dependsOnDrcSet = FDKreadBits(hBs, 6); } else { pInst->noIndependentUse = FDKreadBits(hBs, 1); } if (version == 0) { pInst->requiresEq = 0; } else { pInst->requiresEq = FDKreadBits(hBs, 1); } pCoef = selectDrcCoefficients(hUniDrcConfig, pInst->drcLocation); pInst->drcChannelCount = channelCount = hUniDrcConfig->channelLayout.baseChannelCount; if (pInst->drcSetEffect & (EB_DUCK_OTHER | EB_DUCK_SELF)) { DUCKING_MODIFICATION* pDModForChannel = pInst->duckingModificationForChannel; c = 0; while (c < channelCount) { int bsGainSetIndex; bsGainSetIndex = FDKreadBits(hBs, 6); if (c >= 8) return DE_MEMORY_ERROR; gainSetIndex[c] = bsGainSetIndex - 1; _decodeDuckingModification(hBs, &(pDModForChannel[c]), 0); c++; repeatParameters = FDKreadBits(hBs, 1); if (repeatParameters == 1) { bsRepeatParametersCount = FDKreadBits(hBs, 5); bsRepeatParametersCount += 1; for (i = 0; i < bsRepeatParametersCount; i++) { if (c >= 8) return DE_MEMORY_ERROR; gainSetIndex[c] = gainSetIndex[c - 1]; pDModForChannel[c] = pDModForChannel[c - 1]; c++; } } } if (c > channelCount) { return DE_NOT_OK; } err = deriveDrcChannelGroups( pInst->drcSetEffect, pInst->drcChannelCount, gainSetIndex, pDModForChannel, &pInst->nDrcChannelGroups, pInst->gainSetIndexForChannelGroup, channelGroupForChannel, duckingModificationForChannelGroup); if (err) return (err); } else { int deriveChannelCount = 0; if (((version == 0) || (pInst->drcApplyToDownmix != 0)) && (pInst->downmixId[0] != DOWNMIX_ID_BASE_LAYOUT) && (pInst->downmixId[0] != DOWNMIX_ID_ANY_DOWNMIX) && (pInst->downmixIdCount == 1)) { if (hUniDrcConfig->downmixInstructionsCount != 0) { DOWNMIX_INSTRUCTIONS* pDown = selectDownmixInstructions(hUniDrcConfig, pInst->downmixId[0]); if (pDown == NULL) return DE_NOT_OK; pInst->drcChannelCount = channelCount = pDown->targetChannelCount; /* targetChannelCountFromDownmixId*/ } else { deriveChannelCount = 1; channelCount = 1; } } else if (((version == 0) || (pInst->drcApplyToDownmix != 0)) && ((pInst->downmixId[0] == DOWNMIX_ID_ANY_DOWNMIX) || (pInst->downmixIdCount > 1))) { /* Set maximum channel count as upper border. The effective channel count * is set at the process function. */ pInst->drcChannelCount = 8; channelCount = 1; } c = 0; while (c < channelCount) { int bsGainSetIndex; bsGainSetIndex = FDKreadBits(hBs, 6); if (c >= 8) return DE_MEMORY_ERROR; gainSetIndex[c] = bsGainSetIndex - 1; c++; repeatSequenceIndex = FDKreadBits(hBs, 1); if (repeatSequenceIndex == 1) { bsRepeatSequenceCount = FDKreadBits(hBs, 5); bsRepeatSequenceCount += 1; if (deriveChannelCount) { channelCount = 1 + bsRepeatSequenceCount; } for (i = 0; i < bsRepeatSequenceCount; i++) { if (c >= 8) return DE_MEMORY_ERROR; gainSetIndex[c] = bsGainSetIndex - 1; c++; } } } if (c > channelCount) { return DE_NOT_OK; } if (deriveChannelCount) { pInst->drcChannelCount = channelCount; } /* DOWNMIX_ID_ANY_DOWNMIX: channelCount is 1. Distribute gainSetIndex to all * channels. */ if ((pInst->downmixId[0] == DOWNMIX_ID_ANY_DOWNMIX) || (pInst->downmixIdCount > 1)) { for (c = 1; c < pInst->drcChannelCount; c++) { gainSetIndex[c] = gainSetIndex[0]; } } err = deriveDrcChannelGroups(pInst->drcSetEffect, pInst->drcChannelCount, gainSetIndex, NULL, &pInst->nDrcChannelGroups, pInst->gainSetIndexForChannelGroup, channelGroupForChannel, NULL); if (err) return (err); for (g = 0; g < pInst->nDrcChannelGroups; g++) { int set, bandCount; set = pInst->gainSetIndexForChannelGroup[g]; /* get bandCount */ if (pCoef != NULL && set < pCoef->gainSetCount) { bandCount = pCoef->gainSet[set].bandCount; } else { bandCount = 1; } _decodeGainModification(hBs, version, bandCount, pInst->gainModificationForChannelGroup[g], 0); } } return err; } static DRC_ERROR _readChannelLayout(HANDLE_FDK_BITSTREAM hBs, CHANNEL_LAYOUT* pChan) { DRC_ERROR err = DE_OK; pChan->baseChannelCount = FDKreadBits(hBs, 7); if (pChan->baseChannelCount > 8) return DE_NOT_OK; pChan->layoutSignalingPresent = FDKreadBits(hBs, 1); if (pChan->layoutSignalingPresent) { pChan->definedLayout = FDKreadBits(hBs, 8); if (pChan->definedLayout == 0) { int i; for (i = 0; i < pChan->baseChannelCount; i++) { if (i < 8) { pChan->speakerPosition[i] = FDKreadBits(hBs, 7); } else { FDKpushFor(hBs, 7); } } } } return err; } static DRC_ERROR _readDownmixInstructions(HANDLE_FDK_BITSTREAM hBs, const int version, CHANNEL_LAYOUT* pChan, DOWNMIX_INSTRUCTIONS* pDown) { DRC_ERROR err = DE_OK; pDown->downmixId = FDKreadBits(hBs, 7); pDown->targetChannelCount = FDKreadBits(hBs, 7); pDown->targetLayout = FDKreadBits(hBs, 8); pDown->downmixCoefficientsPresent = FDKreadBits(hBs, 1); if (pDown->downmixCoefficientsPresent) { int nDownmixCoeffs = pDown->targetChannelCount * pChan->baseChannelCount; int i; if (nDownmixCoeffs > 8 * 8) return DE_NOT_OK; if (version == 0) { pDown->bsDownmixOffset = 0; for (i = 0; i < nDownmixCoeffs; i++) { /* LFE downmix coefficients are not supported. */ pDown->downmixCoefficient[i] = downmixCoeff[FDKreadBits(hBs, 4)]; } } else { pDown->bsDownmixOffset = FDKreadBits(hBs, 4); for (i = 0; i < nDownmixCoeffs; i++) { pDown->downmixCoefficient[i] = downmixCoeffV1[FDKreadBits(hBs, 5)]; } } } return err; } static DRC_ERROR _readDrcExtensionV1(HANDLE_FDK_BITSTREAM hBs, HANDLE_UNI_DRC_CONFIG hUniDrcConfig) { DRC_ERROR err = DE_OK; int downmixInstructionsV1Present; int drcCoeffsAndInstructionsUniDrcV1Present; int loudEqInstructionsPresent, loudEqInstructionsCount; int eqPresent, eqInstructionsCount; int i, offset; int diff = hUniDrcConfig->diff; downmixInstructionsV1Present = FDKreadBits(hBs, 1); if (downmixInstructionsV1Present == 1) { diff |= _compAssign(&hUniDrcConfig->downmixInstructionsCountV1, FDKreadBits(hBs, 7)); offset = hUniDrcConfig->downmixInstructionsCountV0; hUniDrcConfig->downmixInstructionsCount = fMin( (UCHAR)(offset + hUniDrcConfig->downmixInstructionsCountV1), (UCHAR)6); for (i = 0; i < hUniDrcConfig->downmixInstructionsCountV1; i++) { DOWNMIX_INSTRUCTIONS tmpDown; FDKmemclear(&tmpDown, sizeof(DOWNMIX_INSTRUCTIONS)); err = _readDownmixInstructions(hBs, 1, &hUniDrcConfig->channelLayout, &tmpDown); if (err) return err; if ((offset + i) >= 6) continue; if (!diff) diff |= (FDKmemcmp(&tmpDown, &(hUniDrcConfig->downmixInstructions[offset + i]), sizeof(DOWNMIX_INSTRUCTIONS)) != 0); hUniDrcConfig->downmixInstructions[offset + i] = tmpDown; } } else { diff |= _compAssign(&hUniDrcConfig->downmixInstructionsCountV1, 0); } drcCoeffsAndInstructionsUniDrcV1Present = FDKreadBits(hBs, 1); if (drcCoeffsAndInstructionsUniDrcV1Present == 1) { diff |= _compAssign(&hUniDrcConfig->drcCoefficientsUniDrcCountV1, FDKreadBits(hBs, 3)); offset = hUniDrcConfig->drcCoefficientsUniDrcCountV0; hUniDrcConfig->drcCoefficientsUniDrcCount = fMin((UCHAR)(offset + hUniDrcConfig->drcCoefficientsUniDrcCountV1), (UCHAR)2); for (i = 0; i < hUniDrcConfig->drcCoefficientsUniDrcCountV1; i++) { DRC_COEFFICIENTS_UNI_DRC tmpCoef; FDKmemclear(&tmpCoef, sizeof(DRC_COEFFICIENTS_UNI_DRC)); err = _readDrcCoefficientsUniDrc(hBs, 1, &tmpCoef); if (err) return err; if ((offset + i) >= 2) continue; if (!diff) diff |= (FDKmemcmp(&tmpCoef, &(hUniDrcConfig->drcCoefficientsUniDrc[offset + i]), sizeof(DRC_COEFFICIENTS_UNI_DRC)) != 0); hUniDrcConfig->drcCoefficientsUniDrc[offset + i] = tmpCoef; } diff |= _compAssign(&hUniDrcConfig->drcInstructionsUniDrcCountV1, FDKreadBits(hBs, 6)); offset = hUniDrcConfig->drcInstructionsUniDrcCount; hUniDrcConfig->drcInstructionsUniDrcCount = fMin((UCHAR)(offset + hUniDrcConfig->drcInstructionsUniDrcCountV1), (UCHAR)12); for (i = 0; i < hUniDrcConfig->drcInstructionsUniDrcCount; i++) { DRC_INSTRUCTIONS_UNI_DRC tmpInst; FDKmemclear(&tmpInst, sizeof(DRC_INSTRUCTIONS_UNI_DRC)); err = _readDrcInstructionsUniDrc(hBs, 1, hUniDrcConfig, &tmpInst); if (err) return err; if ((offset + i) >= 12) continue; if (!diff) diff |= (FDKmemcmp(&tmpInst, &(hUniDrcConfig->drcInstructionsUniDrc[offset + i]), sizeof(DRC_INSTRUCTIONS_UNI_DRC)) != 0); hUniDrcConfig->drcInstructionsUniDrc[offset + i] = tmpInst; } } else { diff |= _compAssign(&hUniDrcConfig->drcCoefficientsUniDrcCountV1, 0); diff |= _compAssign(&hUniDrcConfig->drcInstructionsUniDrcCountV1, 0); } loudEqInstructionsPresent = FDKreadBits(hBs, 1); if (loudEqInstructionsPresent == 1) { loudEqInstructionsCount = FDKreadBits(hBs, 4); for (i = 0; i < loudEqInstructionsCount; i++) { _skipLoudEqInstructions(hBs); } } eqPresent = FDKreadBits(hBs, 1); if (eqPresent == 1) { _skipEqCoefficients(hBs); eqInstructionsCount = FDKreadBits(hBs, 4); for (i = 0; i < eqInstructionsCount; i++) { _skipEqInstructions(hBs, hUniDrcConfig); } } hUniDrcConfig->diff = diff; return err; } static DRC_ERROR _readUniDrcConfigExtension( HANDLE_FDK_BITSTREAM hBs, HANDLE_UNI_DRC_CONFIG hUniDrcConfig) { DRC_ERROR err = DE_OK; int k, bitSizeLen, extSizeBits, bitSize; INT nBitsRemaining; UNI_DRC_CONFIG_EXTENSION* pExt = &(hUniDrcConfig->uniDrcConfigExt); k = 0; pExt->uniDrcConfigExtType[k] = FDKreadBits(hBs, 4); while (pExt->uniDrcConfigExtType[k] != UNIDRCCONFEXT_TERM) { if (k >= (8 - 1)) return DE_MEMORY_ERROR; bitSizeLen = FDKreadBits(hBs, 4); extSizeBits = bitSizeLen + 4; bitSize = FDKreadBits(hBs, extSizeBits); pExt->extBitSize[k] = bitSize + 1; nBitsRemaining = (INT)FDKgetValidBits(hBs); switch (pExt->uniDrcConfigExtType[k]) { case UNIDRCCONFEXT_V1: err = _readDrcExtensionV1(hBs, hUniDrcConfig); if (err) return err; if (nBitsRemaining != ((INT)pExt->extBitSize[k] + (INT)FDKgetValidBits(hBs))) return DE_NOT_OK; break; case UNIDRCCONFEXT_PARAM_DRC: /* add future extensions here */ default: FDKpushFor(hBs, pExt->extBitSize[k]); break; } k++; pExt->uniDrcConfigExtType[k] = FDKreadBits(hBs, 4); } return err; } DRC_ERROR drcDec_readUniDrcConfig(HANDLE_FDK_BITSTREAM hBs, HANDLE_UNI_DRC_CONFIG hUniDrcConfig) { DRC_ERROR err = DE_OK; int i, diff = 0; int drcDescriptionBasicPresent, drcCoefficientsBasicCount, drcInstructionsBasicCount; CHANNEL_LAYOUT tmpChan; FDKmemclear(&tmpChan, sizeof(CHANNEL_LAYOUT)); if (hUniDrcConfig == NULL) return DE_NOT_OK; diff |= _compAssign(&hUniDrcConfig->sampleRatePresent, FDKreadBits(hBs, 1)); if (hUniDrcConfig->sampleRatePresent == 1) { diff |= _compAssign(&hUniDrcConfig->sampleRate, FDKreadBits(hBs, 18) + 1000); } diff |= _compAssign(&hUniDrcConfig->downmixInstructionsCountV0, FDKreadBits(hBs, 7)); drcDescriptionBasicPresent = FDKreadBits(hBs, 1); if (drcDescriptionBasicPresent == 1) { drcCoefficientsBasicCount = FDKreadBits(hBs, 3); drcInstructionsBasicCount = FDKreadBits(hBs, 4); } else { drcCoefficientsBasicCount = 0; drcInstructionsBasicCount = 0; } diff |= _compAssign(&hUniDrcConfig->drcCoefficientsUniDrcCountV0, FDKreadBits(hBs, 3)); diff |= _compAssign(&hUniDrcConfig->drcInstructionsUniDrcCountV0, FDKreadBits(hBs, 6)); err = _readChannelLayout(hBs, &tmpChan); if (err) return err; if (!diff) diff |= (FDKmemcmp(&tmpChan, &hUniDrcConfig->channelLayout, sizeof(CHANNEL_LAYOUT)) != 0); hUniDrcConfig->channelLayout = tmpChan; hUniDrcConfig->downmixInstructionsCount = fMin(hUniDrcConfig->downmixInstructionsCountV0, (UCHAR)6); for (i = 0; i < hUniDrcConfig->downmixInstructionsCountV0; i++) { DOWNMIX_INSTRUCTIONS tmpDown; FDKmemclear(&tmpDown, sizeof(DOWNMIX_INSTRUCTIONS)); err = _readDownmixInstructions(hBs, 0, &hUniDrcConfig->channelLayout, &tmpDown); if (err) return err; if (i >= 6) continue; if (!diff) diff |= (FDKmemcmp(&tmpDown, &(hUniDrcConfig->downmixInstructions[i]), sizeof(DOWNMIX_INSTRUCTIONS)) != 0); hUniDrcConfig->downmixInstructions[i] = tmpDown; } for (i = 0; i < drcCoefficientsBasicCount; i++) { _skipDrcCoefficientsBasic(hBs); } for (i = 0; i < drcInstructionsBasicCount; i++) { _skipDrcInstructionsBasic(hBs); } hUniDrcConfig->drcCoefficientsUniDrcCount = fMin(hUniDrcConfig->drcCoefficientsUniDrcCountV0, (UCHAR)2); for (i = 0; i < hUniDrcConfig->drcCoefficientsUniDrcCountV0; i++) { DRC_COEFFICIENTS_UNI_DRC tmpCoef; FDKmemclear(&tmpCoef, sizeof(DRC_COEFFICIENTS_UNI_DRC)); err = _readDrcCoefficientsUniDrc(hBs, 0, &tmpCoef); if (err) return err; if (i >= 2) continue; if (!diff) diff |= (FDKmemcmp(&tmpCoef, &(hUniDrcConfig->drcCoefficientsUniDrc[i]), sizeof(DRC_COEFFICIENTS_UNI_DRC)) != 0); hUniDrcConfig->drcCoefficientsUniDrc[i] = tmpCoef; } hUniDrcConfig->drcInstructionsUniDrcCount = fMin(hUniDrcConfig->drcInstructionsUniDrcCountV0, (UCHAR)12); for (i = 0; i < hUniDrcConfig->drcInstructionsUniDrcCountV0; i++) { DRC_INSTRUCTIONS_UNI_DRC tmpInst; FDKmemclear(&tmpInst, sizeof(DRC_INSTRUCTIONS_UNI_DRC)); err = _readDrcInstructionsUniDrc(hBs, 0, hUniDrcConfig, &tmpInst); if (err) return err; if (i >= 12) continue; if (!diff) diff |= (FDKmemcmp(&tmpInst, &(hUniDrcConfig->drcInstructionsUniDrc[i]), sizeof(DRC_INSTRUCTIONS_UNI_DRC)) != 0); hUniDrcConfig->drcInstructionsUniDrc[i] = tmpInst; } diff |= _compAssign(&hUniDrcConfig->uniDrcConfigExtPresent, FDKreadBits(hBs, 1)); hUniDrcConfig->diff = diff; if (hUniDrcConfig->uniDrcConfigExtPresent == 1) { err = _readUniDrcConfigExtension(hBs, hUniDrcConfig); if (err) return err; } return err; } /*******************/ /* loudnessInfoSet */ /*******************/ static DRC_ERROR _decodeMethodValue(HANDLE_FDK_BITSTREAM hBs, const UCHAR methodDefinition, FIXP_DBL* methodValue, INT isBox) { int tmp; FIXP_DBL val; switch (methodDefinition) { case MD_UNKNOWN_OTHER: case MD_PROGRAM_LOUDNESS: case MD_ANCHOR_LOUDNESS: case MD_MAX_OF_LOUDNESS_RANGE: case MD_MOMENTARY_LOUDNESS_MAX: case MD_SHORT_TERM_LOUDNESS_MAX: tmp = FDKreadBits(hBs, 8); val = FL2FXCONST_DBL(-57.75f / (float)(1 << 7)) + (FIXP_DBL)( tmp << (DFRACT_BITS - 1 - 2 - 7)); /* -57.75 + tmp * 0.25; */ break; case MD_LOUDNESS_RANGE: tmp = FDKreadBits(hBs, 8); if (tmp == 0) val = (FIXP_DBL)0; else if (tmp <= 128) val = (FIXP_DBL)(tmp << (DFRACT_BITS - 1 - 2 - 7)); /* tmp * 0.25; */ else if (tmp <= 204) { val = (FIXP_DBL)(tmp << (DFRACT_BITS - 1 - 1 - 7)) - FL2FXCONST_DBL(32.0f / (float)(1 << 7)); /* 0.5 * tmp - 32.0f; */ } else { /* downscale by 1 more bit to prevent overflow at intermediate result */ val = (FIXP_DBL)(tmp << (DFRACT_BITS - 1 - 8)) - FL2FXCONST_DBL(134.0f / (float)(1 << 8)); /* tmp - 134.0; */ val <<= 1; } break; case MD_MIXING_LEVEL: tmp = FDKreadBits(hBs, isBox ? 8 : 5); val = (FIXP_DBL)(tmp << (DFRACT_BITS - 1 - 7)) + FL2FXCONST_DBL(80.0f / (float)(1 << 7)); /* tmp + 80.0; */ break; case MD_ROOM_TYPE: tmp = FDKreadBits(hBs, isBox ? 8 : 2); val = (FIXP_DBL)(tmp << (DFRACT_BITS - 1 - 7)); /* tmp; */ break; case MD_SHORT_TERM_LOUDNESS: tmp = FDKreadBits(hBs, 8); val = FL2FXCONST_DBL(-116.0f / (float)(1 << 7)) + (FIXP_DBL)( tmp << (DFRACT_BITS - 1 - 1 - 7)); /* -116.0 + tmp * 0.5; */ break; default: return DE_NOT_OK; /* invalid methodDefinition value */ } *methodValue = val; return DE_OK; } static DRC_ERROR _readLoudnessMeasurement(HANDLE_FDK_BITSTREAM hBs, LOUDNESS_MEASUREMENT* pMeas) { DRC_ERROR err = DE_OK; pMeas->methodDefinition = FDKreadBits(hBs, 4); err = _decodeMethodValue(hBs, pMeas->methodDefinition, &pMeas->methodValue, 0); if (err) return err; pMeas->measurementSystem = FDKreadBits(hBs, 4); pMeas->reliability = FDKreadBits(hBs, 2); return err; } static DRC_ERROR _readLoudnessInfo(HANDLE_FDK_BITSTREAM hBs, const int version, LOUDNESS_INFO* loudnessInfo) { DRC_ERROR err = DE_OK; int bsSamplePeakLevel, bsTruePeakLevel, i; int measurementCount; loudnessInfo->drcSetId = FDKreadBits(hBs, 6); if (version >= 1) { loudnessInfo->eqSetId = FDKreadBits(hBs, 6); } else { loudnessInfo->eqSetId = 0; } loudnessInfo->downmixId = FDKreadBits(hBs, 7); loudnessInfo->samplePeakLevelPresent = FDKreadBits(hBs, 1); if (loudnessInfo->samplePeakLevelPresent) { bsSamplePeakLevel = FDKreadBits(hBs, 12); if (bsSamplePeakLevel == 0) { loudnessInfo->samplePeakLevelPresent = 0; loudnessInfo->samplePeakLevel = (FIXP_DBL)0; } else { /* 20.0 - bsSamplePeakLevel * 0.03125; */ loudnessInfo->samplePeakLevel = FL2FXCONST_DBL(20.0f / (float)(1 << 7)) - (FIXP_DBL)(bsSamplePeakLevel << (DFRACT_BITS - 1 - 5 - 7)); } } loudnessInfo->truePeakLevelPresent = FDKreadBits(hBs, 1); if (loudnessInfo->truePeakLevelPresent) { bsTruePeakLevel = FDKreadBits(hBs, 12); if (bsTruePeakLevel == 0) { loudnessInfo->truePeakLevelPresent = 0; loudnessInfo->truePeakLevel = (FIXP_DBL)0; } else { loudnessInfo->truePeakLevel = FL2FXCONST_DBL(20.0f / (float)(1 << 7)) - (FIXP_DBL)(bsTruePeakLevel << (DFRACT_BITS - 1 - 5 - 7)); } loudnessInfo->truePeakLevelMeasurementSystem = FDKreadBits(hBs, 4); loudnessInfo->truePeakLevelReliability = FDKreadBits(hBs, 2); } measurementCount = FDKreadBits(hBs, 4); loudnessInfo->measurementCount = fMin(measurementCount, 8); for (i = 0; i < measurementCount; i++) { LOUDNESS_MEASUREMENT tmpMeas; FDKmemclear(&tmpMeas, sizeof(LOUDNESS_MEASUREMENT)); err = _readLoudnessMeasurement(hBs, &tmpMeas); if (err) return err; if (i >= 8) continue; loudnessInfo->loudnessMeasurement[i] = tmpMeas; } return err; } static DRC_ERROR _readLoudnessInfoSetExtEq( HANDLE_FDK_BITSTREAM hBs, HANDLE_LOUDNESS_INFO_SET hLoudnessInfoSet) { DRC_ERROR err = DE_OK; int i, offset; int diff = hLoudnessInfoSet->diff; diff |= _compAssign(&hLoudnessInfoSet->loudnessInfoAlbumCountV1, FDKreadBits(hBs, 6)); diff |= _compAssign(&hLoudnessInfoSet->loudnessInfoCountV1, FDKreadBits(hBs, 6)); offset = hLoudnessInfoSet->loudnessInfoAlbumCountV0; hLoudnessInfoSet->loudnessInfoAlbumCount = fMin( (UCHAR)(offset + hLoudnessInfoSet->loudnessInfoAlbumCountV1), (UCHAR)12); for (i = 0; i < hLoudnessInfoSet->loudnessInfoAlbumCountV1; i++) { LOUDNESS_INFO tmpLoud; FDKmemclear(&tmpLoud, sizeof(LOUDNESS_INFO)); err = _readLoudnessInfo(hBs, 1, &tmpLoud); if (err) return err; if ((offset + i) >= 12) continue; if (!diff) diff |= (FDKmemcmp(&tmpLoud, &(hLoudnessInfoSet->loudnessInfoAlbum[offset + i]), sizeof(LOUDNESS_INFO)) != 0); hLoudnessInfoSet->loudnessInfoAlbum[offset + i] = tmpLoud; } offset = hLoudnessInfoSet->loudnessInfoCountV0; hLoudnessInfoSet->loudnessInfoCount = fMin((UCHAR)(offset + hLoudnessInfoSet->loudnessInfoCountV1), (UCHAR)12); for (i = 0; i < hLoudnessInfoSet->loudnessInfoCountV1; i++) { LOUDNESS_INFO tmpLoud; FDKmemclear(&tmpLoud, sizeof(LOUDNESS_INFO)); err = _readLoudnessInfo(hBs, 1, &tmpLoud); if (err) return err; if ((offset + i) >= 12) continue; if (!diff) diff |= (FDKmemcmp(&tmpLoud, &(hLoudnessInfoSet->loudnessInfo[offset + i]), sizeof(LOUDNESS_INFO)) != 0); hLoudnessInfoSet->loudnessInfo[offset + i] = tmpLoud; } hLoudnessInfoSet->diff = diff; return err; } static DRC_ERROR _readLoudnessInfoSetExtension( HANDLE_FDK_BITSTREAM hBs, HANDLE_LOUDNESS_INFO_SET hLoudnessInfoSet) { DRC_ERROR err = DE_OK; int k, bitSizeLen, extSizeBits, bitSize; INT nBitsRemaining; LOUDNESS_INFO_SET_EXTENSION* pExt = &(hLoudnessInfoSet->loudnessInfoSetExt); k = 0; pExt->loudnessInfoSetExtType[k] = FDKreadBits(hBs, 4); while (pExt->loudnessInfoSetExtType[k] != UNIDRCLOUDEXT_TERM) { if (k >= (8 - 1)) return DE_MEMORY_ERROR; bitSizeLen = FDKreadBits(hBs, 4); extSizeBits = bitSizeLen + 4; bitSize = FDKreadBits(hBs, extSizeBits); pExt->extBitSize[k] = bitSize + 1; nBitsRemaining = (INT)FDKgetValidBits(hBs); switch (pExt->loudnessInfoSetExtType[k]) { case UNIDRCLOUDEXT_EQ: err = _readLoudnessInfoSetExtEq(hBs, hLoudnessInfoSet); if (err) return err; if (nBitsRemaining != ((INT)pExt->extBitSize[k] + (INT)FDKgetValidBits(hBs))) return DE_NOT_OK; break; /* add future extensions here */ default: FDKpushFor(hBs, pExt->extBitSize[k]); break; } k++; pExt->loudnessInfoSetExtType[k] = FDKreadBits(hBs, 4); } return err; } /* Parser for loundessInfoSet() */ DRC_ERROR drcDec_readLoudnessInfoSet(HANDLE_FDK_BITSTREAM hBs, HANDLE_LOUDNESS_INFO_SET hLoudnessInfoSet) { DRC_ERROR err = DE_OK; int i, diff = 0; if (hLoudnessInfoSet == NULL) return DE_NOT_OK; diff |= _compAssign(&hLoudnessInfoSet->loudnessInfoAlbumCountV0, FDKreadBits(hBs, 6)); diff |= _compAssign(&hLoudnessInfoSet->loudnessInfoCountV0, FDKreadBits(hBs, 6)); hLoudnessInfoSet->loudnessInfoAlbumCount = fMin(hLoudnessInfoSet->loudnessInfoAlbumCountV0, (UCHAR)12); for (i = 0; i < hLoudnessInfoSet->loudnessInfoAlbumCountV0; i++) { LOUDNESS_INFO tmpLoud; FDKmemclear(&tmpLoud, sizeof(LOUDNESS_INFO)); err = _readLoudnessInfo(hBs, 0, &tmpLoud); if (err) return err; if (i >= 12) continue; if (!diff) diff |= (FDKmemcmp(&tmpLoud, &(hLoudnessInfoSet->loudnessInfoAlbum[i]), sizeof(LOUDNESS_INFO)) != 0); hLoudnessInfoSet->loudnessInfoAlbum[i] = tmpLoud; } hLoudnessInfoSet->loudnessInfoCount = fMin(hLoudnessInfoSet->loudnessInfoCountV0, (UCHAR)12); for (i = 0; i < hLoudnessInfoSet->loudnessInfoCountV0; i++) { LOUDNESS_INFO tmpLoud; FDKmemclear(&tmpLoud, sizeof(LOUDNESS_INFO)); err = _readLoudnessInfo(hBs, 0, &tmpLoud); if (err) return err; if (i >= 12) continue; if (!diff) diff |= (FDKmemcmp(&tmpLoud, &(hLoudnessInfoSet->loudnessInfo[i]), sizeof(LOUDNESS_INFO)) != 0); hLoudnessInfoSet->loudnessInfo[i] = tmpLoud; } diff |= _compAssign(&hLoudnessInfoSet->loudnessInfoSetExtPresent, FDKreadBits(hBs, 1)); hLoudnessInfoSet->diff = diff; if (hLoudnessInfoSet->loudnessInfoSetExtPresent) { err = _readLoudnessInfoSetExtension(hBs, hLoudnessInfoSet); if (err) return err; } return err; } fdk-aac-2.0.2/libDRCdec/src/drcDec_reader.h000066400000000000000000000131221404225310700202170ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /************************* MPEG-D DRC decoder library ************************** Author(s): Description: *******************************************************************************/ #ifndef DRCDEC_READER_H #define DRCDEC_READER_H #include "drcDecoder.h" #include "drcDec_types.h" #include "FDK_bitstream.h" DRC_ERROR drcDec_readUniDrc(HANDLE_FDK_BITSTREAM hBs, HANDLE_UNI_DRC_CONFIG hUniDrcConfig, HANDLE_LOUDNESS_INFO_SET hLoudnessInfoSet, const int frameSize, const int deltaTminDefault, HANDLE_UNI_DRC_GAIN hUniDrcGain); DRC_ERROR drcDec_readUniDrcGain(HANDLE_FDK_BITSTREAM hBs, HANDLE_UNI_DRC_CONFIG hUniDrcConfig, const int frameSize, const int deltaTminDefault, HANDLE_UNI_DRC_GAIN hUniDrcGain); DRC_ERROR drcDec_readUniDrcConfig(HANDLE_FDK_BITSTREAM hBs, HANDLE_UNI_DRC_CONFIG hUniDrcConfig); DRC_ERROR drcDec_readLoudnessInfoSet(HANDLE_FDK_BITSTREAM hBs, HANDLE_LOUDNESS_INFO_SET hLoudnessInfoSet); #endif fdk-aac-2.0.2/libDRCdec/src/drcDec_rom.cpp000066400000000000000000000361361404225310700201170ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /************************* MPEG-D DRC decoder library ************************** Author(s): Description: *******************************************************************************/ #include "drcDec_types.h" #include "drcDec_rom.h" const SCHAR deltaGain_codingProfile_0_1_huffman[24][2] = { {1, 2}, {3, 4}, {-63, -65}, {5, -66}, {-64, 6}, {-80, 7}, {8, 9}, {-68, 10}, {11, 12}, {-56, -67}, {-61, 13}, {-62, -69}, {14, 15}, {16, -72}, {-71, 17}, {-70, -60}, {18, -59}, {19, 20}, {21, -79}, {-57, -73}, {22, -58}, {-76, 23}, {-75, -74}, {-78, -77}}; const SCHAR deltaGain_codingProfile_2_huffman[48][2] = { {1, 2}, {3, 4}, {5, 6}, {7, 8}, {9, 10}, {11, 12}, {13, -65}, {14, -64}, {15, -66}, {16, -67}, {17, 18}, {19, -68}, {20, -63}, {-69, 21}, {-59, 22}, {-61, -62}, {-60, 23}, {24, -58}, {-70, -57}, {-56, -71}, {25, 26}, {27, -55}, {-72, 28}, {-54, 29}, {-53, 30}, {-73, -52}, {31, -74}, {32, 33}, {-75, 34}, {-76, 35}, {-51, 36}, {-78, 37}, {-77, 38}, {-96, 39}, {-48, 40}, {-50, -79}, {41, 42}, {-80, -81}, {-82, 43}, {44, -49}, {45, -84}, {-83, -89}, {-86, 46}, {-90, -85}, {-91, -93}, {-92, 47}, {-88, -87}, {-95, -94}}; const FIXP_SGL slopeSteepness[] = {FL2FXCONST_SGL(-3.0518f / (float)(1 << 2)), FL2FXCONST_SGL(-1.2207f / (float)(1 << 2)), FL2FXCONST_SGL(-0.4883f / (float)(1 << 2)), FL2FXCONST_SGL(-0.1953f / (float)(1 << 2)), FL2FXCONST_SGL(-0.0781f / (float)(1 << 2)), FL2FXCONST_SGL(-0.0312f / (float)(1 << 2)), FL2FXCONST_SGL(-0.005f / (float)(1 << 2)), FL2FXCONST_SGL(0.0f / (float)(1 << 2)), FL2FXCONST_SGL(0.005f / (float)(1 << 2)), FL2FXCONST_SGL(0.0312f / (float)(1 << 2)), FL2FXCONST_SGL(0.0781f / (float)(1 << 2)), FL2FXCONST_SGL(0.1953f / (float)(1 << 2)), FL2FXCONST_SGL(0.4883f / (float)(1 << 2)), FL2FXCONST_SGL(1.2207f / (float)(1 << 2)), FL2FXCONST_SGL(3.0518f / (float)(1 << 2))}; const SCHAR slopeSteepness_huffman[14][2] = { {1, -57}, {-58, 2}, {3, 4}, {5, 6}, {7, -56}, {8, -60}, {-61, -55}, {9, -59}, {10, -54}, {-64, 11}, {-51, 12}, {-62, -50}, {-63, 13}, {-52, -53}}; const FIXP_DBL downmixCoeff[] = { FL2FXCONST_DBL(1.0000000000 / (float)(1 << 2)), FL2FXCONST_DBL(0.9440608763 / (float)(1 << 2)), FL2FXCONST_DBL(0.8912509381 / (float)(1 << 2)), FL2FXCONST_DBL(0.8413951416 / (float)(1 << 2)), FL2FXCONST_DBL(0.7943282347 / (float)(1 << 2)), FL2FXCONST_DBL(0.7498942093 / (float)(1 << 2)), FL2FXCONST_DBL(0.7079457844 / (float)(1 << 2)), FL2FXCONST_DBL(0.6683439176 / (float)(1 << 2)), FL2FXCONST_DBL(0.6309573445 / (float)(1 << 2)), FL2FXCONST_DBL(0.5956621435 / (float)(1 << 2)), FL2FXCONST_DBL(0.5623413252 / (float)(1 << 2)), FL2FXCONST_DBL(0.5308844442 / (float)(1 << 2)), FL2FXCONST_DBL(0.5011872336 / (float)(1 << 2)), FL2FXCONST_DBL(0.4216965034 / (float)(1 << 2)), FL2FXCONST_DBL(0.3548133892 / (float)(1 << 2)), FL2FXCONST_DBL(0.0000000000 / (float)(1 << 2))}; const FIXP_DBL downmixCoeffV1[] = { FL2FXCONST_DBL(3.1622776602 / (float)(1 << 2)), FL2FXCONST_DBL(1.9952623150 / (float)(1 << 2)), FL2FXCONST_DBL(1.6788040181 / (float)(1 << 2)), FL2FXCONST_DBL(1.4125375446 / (float)(1 << 2)), FL2FXCONST_DBL(1.1885022274 / (float)(1 << 2)), FL2FXCONST_DBL(1.0000000000 / (float)(1 << 2)), FL2FXCONST_DBL(0.9440608763 / (float)(1 << 2)), FL2FXCONST_DBL(0.8912509381 / (float)(1 << 2)), FL2FXCONST_DBL(0.8413951416 / (float)(1 << 2)), FL2FXCONST_DBL(0.7943282347 / (float)(1 << 2)), FL2FXCONST_DBL(0.7498942093 / (float)(1 << 2)), FL2FXCONST_DBL(0.7079457844 / (float)(1 << 2)), FL2FXCONST_DBL(0.6683439176 / (float)(1 << 2)), FL2FXCONST_DBL(0.6309573445 / (float)(1 << 2)), FL2FXCONST_DBL(0.5956621435 / (float)(1 << 2)), FL2FXCONST_DBL(0.5623413252 / (float)(1 << 2)), FL2FXCONST_DBL(0.5308844442 / (float)(1 << 2)), FL2FXCONST_DBL(0.5011872336 / (float)(1 << 2)), FL2FXCONST_DBL(0.4731512590 / (float)(1 << 2)), FL2FXCONST_DBL(0.4466835922 / (float)(1 << 2)), FL2FXCONST_DBL(0.4216965034 / (float)(1 << 2)), FL2FXCONST_DBL(0.3981071706 / (float)(1 << 2)), FL2FXCONST_DBL(0.3548133892 / (float)(1 << 2)), FL2FXCONST_DBL(0.3162277660 / (float)(1 << 2)), FL2FXCONST_DBL(0.2818382931 / (float)(1 << 2)), FL2FXCONST_DBL(0.2511886432 / (float)(1 << 2)), FL2FXCONST_DBL(0.1778279410 / (float)(1 << 2)), FL2FXCONST_DBL(0.1000000000 / (float)(1 << 2)), FL2FXCONST_DBL(0.0562341325 / (float)(1 << 2)), FL2FXCONST_DBL(0.0316227766 / (float)(1 << 2)), FL2FXCONST_DBL(0.0100000000 / (float)(1 << 2)), FL2FXCONST_DBL(0.0000000000 / (float)(1 << 2))}; const CUSTOM_DRC_CHAR_SIGMOID cicpDrcCharSigmoidLeft[] = { {FL2FXCONST_SGL(32.0f / (float)(1 << 6)), FL2FXCONST_SGL(0.0f / (float)(1 << 2)), FL2FXCONST_SGL(9.0f / (float)(1 << 5)), 0}, /* 1 */ {FL2FXCONST_SGL(32.0f / (float)(1 << 6)), FL2FXCONST_SGL(0.2f / (float)(1 << 2)), FL2FXCONST_SGL(9.0f / (float)(1 << 5)), 0}, /* 2 */ {FL2FXCONST_SGL(32.0f / (float)(1 << 6)), FL2FXCONST_SGL(0.4f / (float)(1 << 2)), FL2FXCONST_SGL(9.0f / (float)(1 << 5)), 0}, /* 3 */ {FL2FXCONST_SGL(32.0f / (float)(1 << 6)), FL2FXCONST_SGL(0.6f / (float)(1 << 2)), FL2FXCONST_SGL(9.0f / (float)(1 << 5)), 0}, /* 4 */ {FL2FXCONST_SGL(32.0f / (float)(1 << 6)), FL2FXCONST_SGL(0.8f / (float)(1 << 2)), FL2FXCONST_SGL(6.0f / (float)(1 << 5)), 0}, /* 5 */ {FL2FXCONST_SGL(32.0f / (float)(1 << 6)), FL2FXCONST_SGL(1.0f / (float)(1 << 2)), FL2FXCONST_SGL(5.0f / (float)(1 << 5)), 0}, /* 6 */ }; const CUSTOM_DRC_CHAR_SIGMOID cicpDrcCharSigmoidRight[] = { {FL2FXCONST_SGL(-32.0f / (float)(1 << 6)), FL2FXCONST_SGL(0.0f / (float)(1 << 2)), FL2FXCONST_SGL(12.0f / (float)(1 << 5)), 0}, /* 1 */ {FL2FXCONST_SGL(-32.0f / (float)(1 << 6)), FL2FXCONST_SGL(0.2f / (float)(1 << 2)), FL2FXCONST_SGL(12.0f / (float)(1 << 5)), 0}, /* 2 */ {FL2FXCONST_SGL(-32.0f / (float)(1 << 6)), FL2FXCONST_SGL(0.4f / (float)(1 << 2)), FL2FXCONST_SGL(12.0f / (float)(1 << 5)), 0}, /* 3 */ {FL2FXCONST_SGL(-32.0f / (float)(1 << 6)), FL2FXCONST_SGL(0.6f / (float)(1 << 2)), FL2FXCONST_SGL(10.0f / (float)(1 << 5)), 0}, /* 4 */ {FL2FXCONST_SGL(-32.0f / (float)(1 << 6)), FL2FXCONST_SGL(0.8f / (float)(1 << 2)), FL2FXCONST_SGL(8.0f / (float)(1 << 5)), 0}, /* 5 */ {FL2FXCONST_SGL(-32.0f / (float)(1 << 6)), FL2FXCONST_SGL(1.0f / (float)(1 << 2)), FL2FXCONST_SGL(6.0f / (float)(1 << 5)), 0}, /* 6 */ }; const CUSTOM_DRC_CHAR_NODES cicpDrcCharNodesLeft[] = { {2, {FL2FXCONST_SGL(-31.0f / (float)(1 << 7)), FL2FXCONST_SGL(-41.0f / (float)(1 << 7)), FL2FXCONST_SGL(-53.0f / (float)(1 << 7))}, {FL2FXCONST_SGL(0.0f / (float)(1 << 7)), FL2FXCONST_SGL(0.0f / (float)(1 << 7)), FL2FXCONST_SGL(6.0f / (float)(1 << 7))}}, /* 7 */ {1, {FL2FXCONST_SGL(-31.0f / (float)(1 << 7)), FL2FXCONST_SGL(-43.0f / (float)(1 << 7))}, {FL2FXCONST_SGL(0.0f / (float)(1 << 7)), FL2FXCONST_SGL(6.0f / (float)(1 << 7))}}, /* 8 */ {2, {FL2FXCONST_SGL(-31.0f / (float)(1 << 7)), FL2FXCONST_SGL(-41.0f / (float)(1 << 7)), FL2FXCONST_SGL(-65.0f / (float)(1 << 7))}, {FL2FXCONST_SGL(0.0f / (float)(1 << 7)), FL2FXCONST_SGL(0.0f / (float)(1 << 7)), FL2FXCONST_SGL(12.0f / (float)(1 << 7))}}, /* 9 */ {1, {FL2FXCONST_SGL(-31.0f / (float)(1 << 7)), FL2FXCONST_SGL(-55.0f / (float)(1 << 7))}, {FL2FXCONST_SGL(0.0f / (float)(1 << 7)), FL2FXCONST_SGL(12.0f / (float)(1 << 7))}}, /* 10 */ {1, {FL2FXCONST_SGL(-31.0f / (float)(1 << 7)), FL2FXCONST_SGL(-50.0f / (float)(1 << 7))}, {FL2FXCONST_SGL(0.0f / (float)(1 << 7)), FL2FXCONST_SGL(15.0f / (float)(1 << 7))}} /* 11 */ }; const CUSTOM_DRC_CHAR_NODES cicpDrcCharNodesRight[] = { {4, {FL2FXCONST_SGL(-31.0f / (float)(1 << 7)), FL2FXCONST_SGL(-21.0f / (float)(1 << 7)), FL2FXCONST_SGL(-11.0f / (float)(1 << 7)), FL2FXCONST_SGL(9.0f / (float)(1 << 7)), FL2FXCONST_SGL(19.0f / (float)(1 << 7))}, {FL2FXCONST_SGL(0.0f / (float)(1 << 7)), FL2FXCONST_SGL(0.0f / (float)(1 << 7)), FL2FXCONST_SGL(-5.0f / (float)(1 << 7)), FL2FXCONST_SGL(-24.0f / (float)(1 << 7)), FL2FXCONST_SGL(-34.0f / (float)(1 << 7))}}, /* 7 */ {4, {FL2FXCONST_SGL(-31.0f / (float)(1 << 7)), FL2FXCONST_SGL(-26.0f / (float)(1 << 7)), FL2FXCONST_SGL(-16.0f / (float)(1 << 7)), FL2FXCONST_SGL(4.0f / (float)(1 << 7)), FL2FXCONST_SGL(14.0f / (float)(1 << 7))}, {FL2FXCONST_SGL(0.0f / (float)(1 << 7)), FL2FXCONST_SGL(0.0f / (float)(1 << 7)), FL2FXCONST_SGL(-5.0f / (float)(1 << 7)), FL2FXCONST_SGL(-24.0f / (float)(1 << 7)), FL2FXCONST_SGL(-34.0f / (float)(1 << 7))}}, /* 8 */ {3, {FL2FXCONST_SGL(-31.0f / (float)(1 << 7)), FL2FXCONST_SGL(-21.0f / (float)(1 << 7)), FL2FXCONST_SGL(9.0f / (float)(1 << 7)), FL2FXCONST_SGL(29.0f / (float)(1 << 7))}, {FL2FXCONST_SGL(0.0f / (float)(1 << 7)), FL2FXCONST_SGL(0.0f / (float)(1 << 7)), FL2FXCONST_SGL(-15.0f / (float)(1 << 7)), FL2FXCONST_SGL(-35.0f / (float)(1 << 7))}}, /* 9 */ {4, {FL2FXCONST_SGL(-31.0f / (float)(1 << 7)), FL2FXCONST_SGL(-26.0f / (float)(1 << 7)), FL2FXCONST_SGL(-16.0f / (float)(1 << 7)), FL2FXCONST_SGL(4.0f / (float)(1 << 7)), FL2FXCONST_SGL(14.0f / (float)(1 << 7))}, {FL2FXCONST_SGL(0.0f / (float)(1 << 7)), FL2FXCONST_SGL(0.0f / (float)(1 << 7)), FL2FXCONST_SGL(-5.0f / (float)(1 << 7)), FL2FXCONST_SGL(-24.0f / (float)(1 << 7)), FL2FXCONST_SGL(-34.0f / (float)(1 << 7))}}, /* 10 */ {4, {FL2FXCONST_SGL(-31.0f / (float)(1 << 7)), FL2FXCONST_SGL(-26.0f / (float)(1 << 7)), FL2FXCONST_SGL(-16.0f / (float)(1 << 7)), FL2FXCONST_SGL(4.0f / (float)(1 << 7)), FL2FXCONST_SGL(14.0f / (float)(1 << 7))}, {FL2FXCONST_SGL(0.0f / (float)(1 << 7)), FL2FXCONST_SGL(0.0f / (float)(1 << 7)), FL2FXCONST_SGL(-5.0f / (float)(1 << 7)), FL2FXCONST_SGL(-24.0f / (float)(1 << 7)), FL2FXCONST_SGL(-34.0f / (float)(1 << 7))}} /* 11 */ }; fdk-aac-2.0.2/libDRCdec/src/drcDec_rom.h000066400000000000000000000124251404225310700175570ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /************************* MPEG-D DRC decoder library ************************** Author(s): Description: *******************************************************************************/ #ifndef DRCDEC_ROM_H #define DRCDEC_ROM_H extern const SCHAR deltaGain_codingProfile_0_1_huffman[24][2]; extern const SCHAR deltaGain_codingProfile_2_huffman[48][2]; extern const FIXP_SGL slopeSteepness[]; extern const SCHAR slopeSteepness_huffman[14][2]; extern const FIXP_DBL downmixCoeff[]; extern const FIXP_DBL downmixCoeffV1[]; extern const CUSTOM_DRC_CHAR_SIGMOID cicpDrcCharSigmoidLeft[]; extern const CUSTOM_DRC_CHAR_SIGMOID cicpDrcCharSigmoidRight[]; extern const CUSTOM_DRC_CHAR_NODES cicpDrcCharNodesLeft[]; extern const CUSTOM_DRC_CHAR_NODES cicpDrcCharNodesRight[]; #endif fdk-aac-2.0.2/libDRCdec/src/drcDec_selectionProcess.cpp000066400000000000000000003350561404225310700226510ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /************************* MPEG-D DRC decoder library ************************** Author(s): Andreas Hoelzer Description: DRC Set Selection *******************************************************************************/ #include "drcDec_selectionProcess.h" #include "drcDec_tools.h" typedef enum { DETR_NONE = 0, DETR_NIGHT = 1, DETR_NOISY = 2, DETR_LIMITED = 3, DETR_LOWLEVEL = 4, DETR_DIALOG = 5, DETR_GENERAL_COMPR = 6, DETR_EXPAND = 7, DETR_ARTISTIC = 8, DETR_COUNT } DRC_EFFECT_TYPE_REQUEST; typedef enum { DFRT_EFFECT_TYPE, DFRT_DYNAMIC_RANGE, DFRT_DRC_CHARACTERISTIC } DRC_FEATURE_REQUEST_TYPE; typedef enum { MDR_DEFAULT = 0, MDR_PROGRAM_LOUDNESS = 1, MDR_ANCHOR_LOUDNESS = 2 } METHOD_DEFINITION_REQUEST; typedef enum { MSR_DEFAULT = 0, MSR_BS_1770_4 = 1, MSR_USER = 2, MSR_EXPERT_PANEL = 3, MSR_RESERVED_A = 4, MSR_RESERVED_B = 5, MSR_RESERVED_C = 6, MSR_RESERVED_D = 7, MSR_RESERVED_E = 8 } MEASUREMENT_SYSTEM_REQUEST; typedef enum { LPR_DEFAULT = 0, LPR_OFF = 1, LPR_HIGHPASS = 2 } LOUDNESS_PREPROCESSING_REQUEST; typedef enum { DRMRT_SHORT_TERM_LOUDNESS_TO_AVG = 0, DRMRT_MOMENTARY_LOUDNESS_TO_AVG = 1, DRMRT_TOP_OF_LOUDNESS_RANGE_TO_AVG = 2 } DYN_RANGE_MEASUREMENT_REQUEST_TYPE; typedef enum { TCRT_DOWNMIX_ID = 0, TCRT_TARGET_LAYOUT = 1, TCRT_TARGET_CHANNEL_COUNT = 2 } TARGET_CONFIG_REQUEST_TYPE; typedef shouldBeUnion { struct { UCHAR numRequests; UCHAR numRequestsDesired; DRC_EFFECT_TYPE_REQUEST request[MAX_REQUESTS_DRC_EFFECT_TYPE]; } drcEffectType; struct { DYN_RANGE_MEASUREMENT_REQUEST_TYPE measurementRequestType; UCHAR requestedIsRange; FIXP_DBL requestValue; /* e = 7 */ FIXP_DBL requestValueMin; /* e = 7 */ FIXP_DBL requestValueMax; /* e = 7 */ } dynamicRange; UCHAR drcCharacteristic; } DRC_FEATURE_REQUEST; typedef struct { /* system parameters */ SCHAR baseChannelCount; SCHAR baseLayout; /* not supported */ TARGET_CONFIG_REQUEST_TYPE targetConfigRequestType; UCHAR numDownmixIdRequests; UCHAR downmixIdRequested[MAX_REQUESTS_DOWNMIX_ID]; UCHAR targetLayoutRequested; UCHAR targetChannelCountRequested; LONG audioSampleRate; /* needed for complexity estimation, currently not supported */ /* loudness normalization parameters */ UCHAR loudnessNormalizationOn; FIXP_DBL targetLoudness; /* e = 7 */ UCHAR albumMode; UCHAR peakLimiterPresent; UCHAR loudnessDeviationMax; /* resolution: 1 dB */ METHOD_DEFINITION_REQUEST loudnessMeasurementMethod; MEASUREMENT_SYSTEM_REQUEST loudnessMeasurementSystem; LOUDNESS_PREPROCESSING_REQUEST loudnessMeasurementPreProc; /* not supported */ LONG deviceCutOffFrequency; /* not supported */ FIXP_DBL loudnessNormalizationGainDbMax; /* e = 7 */ FIXP_DBL loudnessNormalizationGainModificationDb; /* e = 7 */ FIXP_DBL outputPeakLevelMax; /* e = 7 */ /* dynamic range control parameters */ UCHAR dynamicRangeControlOn; UCHAR numDrcFeatureRequests; DRC_FEATURE_REQUEST_TYPE drcFeatureRequestType[MAX_REQUESTS_DRC_FEATURE]; DRC_FEATURE_REQUEST drcFeatureRequest[MAX_REQUESTS_DRC_FEATURE]; /* other */ FIXP_SGL boost; /* e = 1 */ FIXP_SGL compress; /* e = 1 */ UCHAR drcCharacteristicTarget; /* not supported */ } SEL_PROC_INPUT, *HANDLE_SEL_PROC_INPUT; /* Table E.1 of ISO/IEC DIS 23003-4: Recommended order of fallback effect type * requests */ static DRC_EFFECT_TYPE_REQUEST fallbackEffectTypeRequests[6][5] = { /* Night */ {DETR_GENERAL_COMPR, DETR_NOISY, DETR_LIMITED, DETR_LOWLEVEL, DETR_DIALOG}, /* Noisy */ {DETR_GENERAL_COMPR, DETR_NIGHT, DETR_LIMITED, DETR_LOWLEVEL, DETR_DIALOG}, /* Limited */ {DETR_GENERAL_COMPR, DETR_NIGHT, DETR_NOISY, DETR_LOWLEVEL, DETR_DIALOG}, /* LowLevel */ {DETR_GENERAL_COMPR, DETR_NOISY, DETR_NIGHT, DETR_LIMITED, DETR_DIALOG}, /* Dialog */ {DETR_GENERAL_COMPR, DETR_NIGHT, DETR_NOISY, DETR_LIMITED, DETR_LOWLEVEL}, /* General */ {DETR_NIGHT, DETR_NOISY, DETR_LIMITED, DETR_LOWLEVEL, DETR_DIALOG}}; /*******************************************/ typedef struct { UCHAR selectionFlag; UCHAR downmixIdRequestIndex; FIXP_DBL outputPeakLevel; /* e = 7 */ FIXP_DBL loudnessNormalizationGainDbAdjusted; /* e = 7 */ FIXP_DBL outputLoudness; /* e = 7 */ DRC_INSTRUCTIONS_UNI_DRC* pInst; } DRCDEC_SELECTION_DATA; typedef struct { UCHAR numData; DRCDEC_SELECTION_DATA data[(12 + 1 + 6)]; } DRCDEC_SELECTION; /*******************************************/ /* helper functions */ /*******************************************/ static int _isError(int x) { if (x < DRCDEC_SELECTION_PROCESS_WARNING) { return 1; } return 0; } /* compare and assign */ static inline int _compAssign(UCHAR* dest, const UCHAR src) { int diff = 0; if (*dest != src) diff = 1; *dest = src; return diff; } static inline int _compAssign(SCHAR* dest, const SCHAR src) { int diff = 0; if (*dest != src) diff = 1; *dest = src; return diff; } static inline int _compAssign(FIXP_DBL* dest, const FIXP_DBL src) { int diff = 0; if (*dest != src) diff = 1; *dest = src; return diff; } static inline int _compAssign(FIXP_SGL* dest, const FIXP_SGL src) { int diff = 0; if (*dest != src) diff = 1; *dest = src; return diff; } static inline int _compAssign(TARGET_CONFIG_REQUEST_TYPE* dest, const int src) { int diff = 0; if (*dest != src) diff = 1; *dest = (TARGET_CONFIG_REQUEST_TYPE)src; return diff; } static inline int _compAssign(METHOD_DEFINITION_REQUEST* dest, const int src) { int diff = 0; if (*dest != src) diff = 1; *dest = (METHOD_DEFINITION_REQUEST)src; return diff; } static inline int _compAssign(DRC_FEATURE_REQUEST_TYPE* dest, const int src) { int diff = 0; if (*dest != src) diff = 1; *dest = (DRC_FEATURE_REQUEST_TYPE)src; return diff; } static inline int _compAssign(DRC_EFFECT_TYPE_REQUEST* dest, const int src) { int diff = 0; if (*dest != src) diff = 1; *dest = (DRC_EFFECT_TYPE_REQUEST)src; return diff; } static DRCDEC_SELECTION_DATA* _drcdec_selection_addNew( DRCDEC_SELECTION* pSelection); static DRCDEC_SELECTION_DATA* _drcdec_selection_add( DRCDEC_SELECTION* pSelection, DRCDEC_SELECTION_DATA* pDataIn); static int _drcdec_selection_clear(DRCDEC_SELECTION* pSelection); static int _drcdec_selection_getNumber(DRCDEC_SELECTION* pSelection); static int _drcdec_selection_setNumber(DRCDEC_SELECTION* pSelection, int num); static DRCDEC_SELECTION_DATA* _drcdec_selection_getAt( DRCDEC_SELECTION* pSelection, int at); static int _swapSelectionAndClear(DRCDEC_SELECTION** ppCandidatesPotential, DRCDEC_SELECTION** ppCandidatesSelected); static int _swapSelection(DRCDEC_SELECTION** ppCandidatesPotential, DRCDEC_SELECTION** ppCandidatesSelected); /*******************************************/ /* declarations of static functions */ /*******************************************/ static DRCDEC_SELECTION_PROCESS_RETURN _initDefaultParams( HANDLE_SEL_PROC_INPUT hSelProcInput); static DRCDEC_SELECTION_PROCESS_RETURN _initCodecModeParams( HANDLE_SEL_PROC_INPUT hSelProcInput, const SEL_PROC_CODEC_MODE codecMode); static DRCDEC_SELECTION_PROCESS_RETURN _drcSetPreSelection( SEL_PROC_INPUT* hSelProcInput, HANDLE_UNI_DRC_CONFIG hUniDrcConfig, HANDLE_LOUDNESS_INFO_SET hLoudnessInfoSet, DRCDEC_SELECTION** ppCandidatesPotential, DRCDEC_SELECTION** ppCandidatesSelected, SEL_PROC_CODEC_MODE codecMode); static DRCDEC_SELECTION_PROCESS_RETURN _drcSetFinalSelection_peakValue0( DRCDEC_SELECTION* pCandidatesPotential, DRCDEC_SELECTION* pCandidatesSelected); static DRCDEC_SELECTION_PROCESS_RETURN _dynamicRangeMeasurement( HANDLE_LOUDNESS_INFO_SET hLoudnessInfoSet, DRC_INSTRUCTIONS_UNI_DRC* pInst, UCHAR downmixIdRequested, DYN_RANGE_MEASUREMENT_REQUEST_TYPE dynamicRangeMeasurementType, int albumMode, int* peakToAveragePresent, FIXP_DBL* peakToAverage); static DRCDEC_SELECTION_PROCESS_RETURN _channelLayoutToDownmixIdMapping( HANDLE_SEL_PROC_INPUT hSelProcInput, HANDLE_UNI_DRC_CONFIG hUniDrcConfig); static DRCDEC_SELECTION_PROCESS_RETURN _generateVirtualDrcSets( HANDLE_SEL_PROC_INPUT hSelProcInput, HANDLE_UNI_DRC_CONFIG hUniDrcConfig, SEL_PROC_CODEC_MODE codecMode); static DRCDEC_SELECTION_PROCESS_RETURN _drcSetRequestSelection( SEL_PROC_INPUT* hSelProcInput, HANDLE_UNI_DRC_CONFIG hUniDrcConfig, HANDLE_LOUDNESS_INFO_SET hLoudnessInfoSet, DRCDEC_SELECTION** ppCandidatesPotential, DRCDEC_SELECTION** ppCandidatesSelected); static DRCDEC_SELECTION_PROCESS_RETURN _drcSetFinalSelection( HANDLE_SEL_PROC_INPUT hSelProcInput, HANDLE_UNI_DRC_CONFIG hUniDrcConfig, DRCDEC_SELECTION** ppCandidatesPotential, DRCDEC_SELECTION** ppCandidatesSelected, SEL_PROC_CODEC_MODE codecMode); static DRCDEC_SELECTION_PROCESS_RETURN _generateOutputInfo( HANDLE_SEL_PROC_INPUT hSelProcInput, HANDLE_SEL_PROC_OUTPUT hSelProcOutput, HANDLE_UNI_DRC_CONFIG hUniDrcConfig, HANDLE_LOUDNESS_INFO_SET hLoudnessInfoSet, DRCDEC_SELECTION_DATA* pSelectionData, SEL_PROC_CODEC_MODE codecMode); static DRCDEC_SELECTION_PROCESS_RETURN _selectDownmixMatrix( HANDLE_SEL_PROC_OUTPUT hSelProcOutput, HANDLE_UNI_DRC_CONFIG hUniDrcConfig); static DRCDEC_SELECTION_PROCESS_RETURN _getLoudness( HANDLE_LOUDNESS_INFO_SET hLoudnessInfoSet, int albumMode, METHOD_DEFINITION_REQUEST measurementMethodRequested, MEASUREMENT_SYSTEM_REQUEST measurementSystemRequested, FIXP_DBL targetLoudness, int drcSetId, int downmixIdRequested, FIXP_DBL* pLoudnessNormalizationGain, FIXP_DBL* pLoudness); static DRCDEC_SELECTION_PROCESS_RETURN _getMixingLevel( HANDLE_LOUDNESS_INFO_SET hLoudnessInfoSet, int downmixIdRequested, int drcSetIdRequested, int albumMode, FIXP_DBL* pMixingLevel); static DRCDEC_SELECTION_PROCESS_RETURN _getSignalPeakLevel( HANDLE_SEL_PROC_INPUT hSelProcInput, HANDLE_UNI_DRC_CONFIG hUniDrcConfig, HANDLE_LOUDNESS_INFO_SET hLoudnessInfoSet, DRC_INSTRUCTIONS_UNI_DRC* pInst, int downmixIdRequested, int* explicitPeakInformationPresent, FIXP_DBL* signalPeakLevelOut, /* e = 7 */ SEL_PROC_CODEC_MODE codecMode); static DRCDEC_SELECTION_PROCESS_RETURN _extractLoudnessPeakToAverageValue( LOUDNESS_INFO* loudnessInfo, DYN_RANGE_MEASUREMENT_REQUEST_TYPE dynamicRangeMeasurementType, int* pLoudnessPeakToAverageValuePresent, FIXP_DBL* pLoudnessPeakToAverageValue); static DRCDEC_SELECTION_PROCESS_RETURN _selectAlbumLoudness( HANDLE_LOUDNESS_INFO_SET hLoudnessInfoSet, DRCDEC_SELECTION* pCandidatesPotential, DRCDEC_SELECTION* pCandidatesSelected); static int _findMethodDefinition(LOUDNESS_INFO* pLoudnessInfo, int methodDefinition, int startIndex); /*******************************************/ /* public functions */ /*******************************************/ struct s_drcdec_selection_process { SEL_PROC_CODEC_MODE codecMode; SEL_PROC_INPUT selProcInput; DRCDEC_SELECTION selectionData[2]; /* 2 instances, one before and one after selection */ }; DRCDEC_SELECTION_PROCESS_RETURN drcDec_SelectionProcess_Create(HANDLE_DRC_SELECTION_PROCESS* phInstance) { HANDLE_DRC_SELECTION_PROCESS hInstance; hInstance = (HANDLE_DRC_SELECTION_PROCESS)FDKcalloc( 1, sizeof(struct s_drcdec_selection_process)); if (!hInstance) return DRCDEC_SELECTION_PROCESS_OUTOFMEMORY; hInstance->codecMode = SEL_PROC_CODEC_MODE_UNDEFINED; *phInstance = hInstance; return DRCDEC_SELECTION_PROCESS_NO_ERROR; } DRCDEC_SELECTION_PROCESS_RETURN drcDec_SelectionProcess_Init(HANDLE_DRC_SELECTION_PROCESS hInstance) { if (!hInstance) return DRCDEC_SELECTION_PROCESS_NOT_OK; _initDefaultParams(&hInstance->selProcInput); return DRCDEC_SELECTION_PROCESS_NO_ERROR; } DRCDEC_SELECTION_PROCESS_RETURN drcDec_SelectionProcess_SetCodecMode(HANDLE_DRC_SELECTION_PROCESS hInstance, const SEL_PROC_CODEC_MODE codecMode) { DRCDEC_SELECTION_PROCESS_RETURN retVal = DRCDEC_SELECTION_PROCESS_NO_ERROR; if (!hInstance) return DRCDEC_SELECTION_PROCESS_NOT_OK; switch (codecMode) { case SEL_PROC_MPEG_4_AAC: case SEL_PROC_MPEG_D_USAC: case SEL_PROC_TEST_TIME_DOMAIN: case SEL_PROC_TEST_QMF_DOMAIN: case SEL_PROC_TEST_STFT_DOMAIN: hInstance->codecMode = codecMode; break; case SEL_PROC_CODEC_MODE_UNDEFINED: default: return DRCDEC_SELECTION_PROCESS_NOT_OK; } retVal = _initCodecModeParams(&(hInstance->selProcInput), hInstance->codecMode = codecMode); return retVal; } DRCDEC_SELECTION_PROCESS_RETURN drcDec_SelectionProcess_SetParam(HANDLE_DRC_SELECTION_PROCESS hInstance, const SEL_PROC_USER_PARAM requestType, FIXP_DBL requestValue, int* pDiff) { INT requestValueInt = (INT)requestValue; int i, diff = 0; SEL_PROC_INPUT* pSelProcInput = &(hInstance->selProcInput); switch (requestType) { case SEL_PROC_LOUDNESS_NORMALIZATION_ON: if ((requestValueInt != 0) && (requestValueInt != 1)) return DRCDEC_SELECTION_PROCESS_PARAM_OUT_OF_RANGE; diff |= _compAssign(&pSelProcInput->loudnessNormalizationOn, requestValueInt); break; case SEL_PROC_TARGET_LOUDNESS: /* Lower boundary: drcSetTargetLoudnessValueLower default value. Upper boundary: drcSetTargetLoudnessValueUpper default value */ if ((requestValue < FL2FXCONST_DBL(-63.0f / (float)(1 << 7))) || (requestValue > (FIXP_DBL)0)) return DRCDEC_SELECTION_PROCESS_PARAM_OUT_OF_RANGE; if (requestValue > FL2FXCONST_DBL(-10.0f / (float)(1 << 7))) /* recommended maximum value */ requestValue = FL2FXCONST_DBL(-10.0f / (float)(1 << 7)); diff |= _compAssign(&pSelProcInput->targetLoudness, requestValue); break; case SEL_PROC_EFFECT_TYPE: if ((requestValueInt < -1) || (requestValueInt >= DETR_COUNT)) return DRCDEC_SELECTION_PROCESS_PARAM_OUT_OF_RANGE; /* Caution. This overrides all drcFeatureRequests requested so far! */ if (requestValueInt == -1) { diff |= _compAssign(&pSelProcInput->dynamicRangeControlOn, 0); } else if (requestValueInt == DETR_NONE) { diff |= _compAssign(&pSelProcInput->dynamicRangeControlOn, 1); diff |= _compAssign(&pSelProcInput->numDrcFeatureRequests, 0); } else { diff |= _compAssign(&pSelProcInput->dynamicRangeControlOn, 1); diff |= _compAssign(&pSelProcInput->numDrcFeatureRequests, 1); diff |= _compAssign(&pSelProcInput->drcFeatureRequestType[0], DFRT_EFFECT_TYPE); diff |= _compAssign(&pSelProcInput->drcFeatureRequest[0] .drcEffectType.numRequestsDesired, 1); diff |= _compAssign( &pSelProcInput->drcFeatureRequest[0].drcEffectType.request[0], requestValueInt); if ((requestValueInt > DETR_NONE) && (requestValueInt <= DETR_GENERAL_COMPR)) { /* use fallback effect type requests */ for (i = 0; i < 5; i++) { diff |= _compAssign(&pSelProcInput->drcFeatureRequest[0] .drcEffectType.request[i + 1], fallbackEffectTypeRequests[requestValueInt - 1][i]); } diff |= _compAssign( &pSelProcInput->drcFeatureRequest[0].drcEffectType.numRequests, 6); } else { diff |= _compAssign( &pSelProcInput->drcFeatureRequest[0].drcEffectType.numRequests, 1); } } break; case SEL_PROC_LOUDNESS_MEASUREMENT_METHOD: if ((requestValueInt < 0) || (requestValueInt > 2)) return DRCDEC_SELECTION_PROCESS_PARAM_OUT_OF_RANGE; diff |= _compAssign(&pSelProcInput->loudnessMeasurementMethod, requestValueInt); break; case SEL_PROC_ALBUM_MODE: if ((requestValueInt < 0) || (requestValueInt > 1)) return DRCDEC_SELECTION_PROCESS_PARAM_OUT_OF_RANGE; diff |= _compAssign(&pSelProcInput->albumMode, requestValueInt); break; case SEL_PROC_DOWNMIX_ID: diff |= _compAssign(&pSelProcInput->targetConfigRequestType, TCRT_DOWNMIX_ID); if (requestValueInt < 0) { /* negative requests signal no downmixId */ diff |= _compAssign(&pSelProcInput->numDownmixIdRequests, 0); } else { diff |= _compAssign(&pSelProcInput->numDownmixIdRequests, 1); diff |= _compAssign(&pSelProcInput->downmixIdRequested[0], requestValueInt); } break; case SEL_PROC_TARGET_LAYOUT: /* Request target layout according to ChannelConfiguration in ISO/IEC * 23001-8 (CICP) */ if ((requestValueInt < 1) || (requestValueInt > 63)) return DRCDEC_SELECTION_PROCESS_PARAM_OUT_OF_RANGE; diff |= _compAssign(&pSelProcInput->targetConfigRequestType, TCRT_TARGET_LAYOUT); diff |= _compAssign(&pSelProcInput->targetLayoutRequested, requestValueInt); break; case SEL_PROC_TARGET_CHANNEL_COUNT: if ((requestValueInt < 1) || (requestValueInt > 8)) return DRCDEC_SELECTION_PROCESS_PARAM_OUT_OF_RANGE; diff |= _compAssign(&pSelProcInput->targetConfigRequestType, TCRT_TARGET_CHANNEL_COUNT); diff |= _compAssign(&pSelProcInput->targetChannelCountRequested, requestValueInt); break; case SEL_PROC_BASE_CHANNEL_COUNT: if (requestValueInt < 0) return DRCDEC_SELECTION_PROCESS_PARAM_OUT_OF_RANGE; diff |= _compAssign(&pSelProcInput->baseChannelCount, requestValueInt); break; case SEL_PROC_SAMPLE_RATE: if (requestValueInt < 0) return DRCDEC_SELECTION_PROCESS_PARAM_OUT_OF_RANGE; diff |= _compAssign(&pSelProcInput->audioSampleRate, requestValueInt); break; case SEL_PROC_BOOST: if ((requestValue < (FIXP_DBL)0) || (requestValue > FL2FXCONST_DBL(1.0f / (float)(1 << 1)))) return DRCDEC_SELECTION_PROCESS_PARAM_OUT_OF_RANGE; diff |= _compAssign( &pSelProcInput->boost, FX_DBL2FX_SGL( requestValue + (FIXP_DBL)(1 << 15))); /* convert to FIXP_SGL with rounding */ break; case SEL_PROC_COMPRESS: if ((requestValue < (FIXP_DBL)0) || (requestValue > FL2FXCONST_DBL(1.0f / (float)(1 << 1)))) return DRCDEC_SELECTION_PROCESS_PARAM_OUT_OF_RANGE; diff |= _compAssign( &pSelProcInput->compress, FX_DBL2FX_SGL( requestValue + (FIXP_DBL)(1 << 15))); /* convert to FIXP_SGL with rounding */ break; default: return DRCDEC_SELECTION_PROCESS_INVALID_PARAM; } if (pDiff != NULL) { *pDiff |= diff; } return DRCDEC_SELECTION_PROCESS_NO_ERROR; } FIXP_DBL drcDec_SelectionProcess_GetParam(HANDLE_DRC_SELECTION_PROCESS hInstance, const SEL_PROC_USER_PARAM requestType) { SEL_PROC_INPUT* pSelProcInput = &(hInstance->selProcInput); switch (requestType) { case SEL_PROC_LOUDNESS_NORMALIZATION_ON: return (FIXP_DBL)pSelProcInput->loudnessNormalizationOn; case SEL_PROC_DYNAMIC_RANGE_CONTROL_ON: return (FIXP_DBL)pSelProcInput->dynamicRangeControlOn; default: return (FIXP_DBL)0; } } DRCDEC_SELECTION_PROCESS_RETURN drcDec_SelectionProcess_Delete(HANDLE_DRC_SELECTION_PROCESS* phInstance) { if (phInstance == NULL || *phInstance == NULL) return DRCDEC_SELECTION_PROCESS_INVALID_HANDLE; FDKfree(*phInstance); *phInstance = NULL; return DRCDEC_SELECTION_PROCESS_NO_ERROR; } DRCDEC_SELECTION_PROCESS_RETURN drcDec_SelectionProcess_Process(HANDLE_DRC_SELECTION_PROCESS hInstance, HANDLE_UNI_DRC_CONFIG hUniDrcConfig, HANDLE_LOUDNESS_INFO_SET hLoudnessInfoSet, HANDLE_SEL_PROC_OUTPUT hSelProcOutput) { DRCDEC_SELECTION_PROCESS_RETURN retVal = DRCDEC_SELECTION_PROCESS_NO_ERROR; DRCDEC_SELECTION* pCandidatesSelected; DRCDEC_SELECTION* pCandidatesPotential; if (hInstance == NULL) return DRCDEC_SELECTION_PROCESS_INVALID_HANDLE; pCandidatesSelected = &(hInstance->selectionData[0]); pCandidatesPotential = &(hInstance->selectionData[1]); _drcdec_selection_setNumber(pCandidatesSelected, 0); _drcdec_selection_setNumber(pCandidatesPotential, 0); retVal = _generateVirtualDrcSets(&(hInstance->selProcInput), hUniDrcConfig, hInstance->codecMode); if (retVal) return (retVal); if (hInstance->selProcInput.baseChannelCount != hUniDrcConfig->channelLayout.baseChannelCount) { hInstance->selProcInput.baseChannelCount = hUniDrcConfig->channelLayout.baseChannelCount; } if ((hInstance->selProcInput.targetConfigRequestType != 0) || (hInstance->selProcInput.targetConfigRequestType == 0 && hInstance->selProcInput.numDownmixIdRequests == 0)) { retVal = _channelLayoutToDownmixIdMapping(&(hInstance->selProcInput), hUniDrcConfig); if (_isError(retVal)) return (retVal); } retVal = _drcSetPreSelection(&(hInstance->selProcInput), hUniDrcConfig, hLoudnessInfoSet, &pCandidatesPotential, &pCandidatesSelected, hInstance->codecMode); if (retVal) return (retVal); if (hInstance->selProcInput.albumMode) { _swapSelectionAndClear(&pCandidatesPotential, &pCandidatesSelected); retVal = _selectAlbumLoudness(hLoudnessInfoSet, pCandidatesPotential, pCandidatesSelected); if (retVal) return (retVal); if (_drcdec_selection_getNumber(pCandidatesSelected) == 0) { _swapSelection(&pCandidatesPotential, &pCandidatesSelected); } } _swapSelectionAndClear(&pCandidatesPotential, &pCandidatesSelected); retVal = _drcSetRequestSelection(&(hInstance->selProcInput), hUniDrcConfig, hLoudnessInfoSet, &pCandidatesPotential, &pCandidatesSelected); if (retVal) return (retVal); retVal = _drcSetFinalSelection(&(hInstance->selProcInput), hUniDrcConfig, &pCandidatesPotential, &pCandidatesSelected, hInstance->codecMode); if (retVal) return (retVal); retVal = _generateOutputInfo( &(hInstance->selProcInput), hSelProcOutput, hUniDrcConfig, hLoudnessInfoSet, &(pCandidatesSelected->data[0]), hInstance->codecMode); if (_isError(retVal)) return (retVal); retVal = _selectDownmixMatrix(hSelProcOutput, hUniDrcConfig); if (retVal) return (retVal); return DRCDEC_SELECTION_PROCESS_NO_ERROR; } /*******************************************/ /* static functions */ /*******************************************/ static DRCDEC_SELECTION_PROCESS_RETURN _initDefaultParams( HANDLE_SEL_PROC_INPUT hSelProcInput) { DRCDEC_SELECTION_PROCESS_RETURN retVal = DRCDEC_SELECTION_PROCESS_NO_ERROR; if (hSelProcInput == NULL) return DRCDEC_SELECTION_PROCESS_INVALID_HANDLE; /* system parameters */ hSelProcInput->baseChannelCount = -1; hSelProcInput->baseLayout = -1; hSelProcInput->targetConfigRequestType = TCRT_DOWNMIX_ID; hSelProcInput->numDownmixIdRequests = 0; /* loudness normalization parameters */ hSelProcInput->albumMode = 0; hSelProcInput->peakLimiterPresent = 0; hSelProcInput->loudnessNormalizationOn = 1; hSelProcInput->targetLoudness = FL2FXCONST_DBL(-24.0f / (float)(1 << 7)); hSelProcInput->loudnessDeviationMax = DEFAULT_LOUDNESS_DEVIATION_MAX; hSelProcInput->loudnessMeasurementMethod = MDR_ANCHOR_LOUDNESS; hSelProcInput->loudnessMeasurementSystem = MSR_EXPERT_PANEL; hSelProcInput->loudnessMeasurementPreProc = LPR_DEFAULT; hSelProcInput->deviceCutOffFrequency = 500; hSelProcInput->loudnessNormalizationGainDbMax = (FIXP_DBL)MAXVAL_DBL; /* infinity as default */ hSelProcInput->loudnessNormalizationGainModificationDb = (FIXP_DBL)0; hSelProcInput->outputPeakLevelMax = (FIXP_DBL)0; if (hSelProcInput->peakLimiterPresent == 1) { hSelProcInput->outputPeakLevelMax = FL2FXCONST_DBL(6.0f / (float)(1 << 7)); } /* dynamic range control parameters */ hSelProcInput->dynamicRangeControlOn = 1; hSelProcInput->numDrcFeatureRequests = 0; /* other parameters */ hSelProcInput->boost = FL2FXCONST_SGL(1.f / (float)(1 << 1)); hSelProcInput->compress = FL2FXCONST_SGL(1.f / (float)(1 << 1)); hSelProcInput->drcCharacteristicTarget = 0; return retVal; } static DRCDEC_SELECTION_PROCESS_RETURN _initCodecModeParams( HANDLE_SEL_PROC_INPUT hSelProcInput, const SEL_PROC_CODEC_MODE codecMode) { DRCDEC_SELECTION_PROCESS_RETURN retVal = DRCDEC_SELECTION_PROCESS_NO_ERROR; if (hSelProcInput == NULL) return DRCDEC_SELECTION_PROCESS_INVALID_HANDLE; switch (codecMode) { case SEL_PROC_MPEG_H_3DA: hSelProcInput->loudnessDeviationMax = 0; hSelProcInput->peakLimiterPresent = 1; /* peak limiter is mandatory */ /* The peak limiter also has to catch overshoots due to user interactivity, downmixing etc. Therefore the maximum output peak level is reduced to 0 dB. */ hSelProcInput->outputPeakLevelMax = (FIXP_DBL)0; break; case SEL_PROC_MPEG_4_AAC: case SEL_PROC_MPEG_D_USAC: hSelProcInput->loudnessDeviationMax = DEFAULT_LOUDNESS_DEVIATION_MAX; hSelProcInput->peakLimiterPresent = 1; /* A peak limiter is present at the end of the decoder, therefore we can * allow for a maximum output peak level greater than full scale */ hSelProcInput->outputPeakLevelMax = FL2FXCONST_DBL(6.0f / (float)(1 << 7)); break; case SEL_PROC_TEST_TIME_DOMAIN: case SEL_PROC_TEST_QMF_DOMAIN: case SEL_PROC_TEST_STFT_DOMAIN: /* for testing, adapt to default settings in reference software */ hSelProcInput->loudnessNormalizationOn = 0; hSelProcInput->dynamicRangeControlOn = 0; break; case SEL_PROC_CODEC_MODE_UNDEFINED: default: hSelProcInput->loudnessDeviationMax = DEFAULT_LOUDNESS_DEVIATION_MAX; hSelProcInput->peakLimiterPresent = 0; } return retVal; } static DRCDEC_SELECTION_PROCESS_RETURN _channelLayoutToDownmixIdMapping( HANDLE_SEL_PROC_INPUT hSelProcInput, HANDLE_UNI_DRC_CONFIG hUniDrcConfig) { DRCDEC_SELECTION_PROCESS_RETURN retVal = DRCDEC_SELECTION_PROCESS_NO_ERROR; DOWNMIX_INSTRUCTIONS* pDown = NULL; int i; hSelProcInput->numDownmixIdRequests = 0; switch (hSelProcInput->targetConfigRequestType) { case TCRT_DOWNMIX_ID: if (hSelProcInput->numDownmixIdRequests == 0) { hSelProcInput->downmixIdRequested[0] = 0; hSelProcInput->numDownmixIdRequests = 1; } break; case TCRT_TARGET_LAYOUT: if (hSelProcInput->targetLayoutRequested == hSelProcInput->baseLayout) { hSelProcInput->downmixIdRequested[0] = 0; hSelProcInput->numDownmixIdRequests = 1; } if (hSelProcInput->numDownmixIdRequests == 0) { for (i = 0; i < hUniDrcConfig->downmixInstructionsCount; i++) { pDown = &(hUniDrcConfig->downmixInstructions[i]); if (hSelProcInput->targetLayoutRequested == pDown->targetLayout) { hSelProcInput ->downmixIdRequested[hSelProcInput->numDownmixIdRequests] = pDown->downmixId; hSelProcInput->numDownmixIdRequests++; } } } if (hSelProcInput->baseLayout == -1) { retVal = DRCDEC_SELECTION_PROCESS_WARNING; } if (hSelProcInput->numDownmixIdRequests == 0) { hSelProcInput->downmixIdRequested[0] = 0; hSelProcInput->numDownmixIdRequests = 1; retVal = DRCDEC_SELECTION_PROCESS_WARNING; } break; case TCRT_TARGET_CHANNEL_COUNT: if (hSelProcInput->targetChannelCountRequested == hSelProcInput->baseChannelCount) { hSelProcInput->downmixIdRequested[0] = 0; hSelProcInput->numDownmixIdRequests = 1; } if (hSelProcInput->numDownmixIdRequests == 0) { for (i = 0; i < hUniDrcConfig->downmixInstructionsCount; i++) { pDown = &(hUniDrcConfig->downmixInstructions[i]); if (hSelProcInput->targetChannelCountRequested == pDown->targetChannelCount) { hSelProcInput ->downmixIdRequested[hSelProcInput->numDownmixIdRequests] = pDown->downmixId; hSelProcInput->numDownmixIdRequests++; } } } if (hSelProcInput->baseChannelCount == -1) { retVal = DRCDEC_SELECTION_PROCESS_WARNING; } if (hSelProcInput->numDownmixIdRequests == 0) { retVal = DRCDEC_SELECTION_PROCESS_WARNING; hSelProcInput->downmixIdRequested[0] = 0; hSelProcInput->numDownmixIdRequests = 1; } break; default: return DRCDEC_SELECTION_PROCESS_NOT_OK; } return retVal; } /*******************************************/ /* Note: Numbering of DRC pre-selection steps according to MPEG-D Part-4 DRC * Amd1 */ /* #1: DownmixId of DRC set matches the requested downmixId. #2: Output channel layout of DRC set matches the requested layout. #3: Channel count of DRC set matches the requested channel count. */ static DRCDEC_SELECTION_PROCESS_RETURN _preSelectionRequirement123( int nRequestedDownmixId, DRC_INSTRUCTIONS_UNI_DRC* pDrcInstructionUniDrc, int* pMatchFound) { int i; *pMatchFound = 0; for (i = 0; i < pDrcInstructionUniDrc->downmixIdCount; i++) { if ((pDrcInstructionUniDrc->downmixId[i] == nRequestedDownmixId) || (pDrcInstructionUniDrc->downmixId[i] == DOWNMIX_ID_ANY_DOWNMIX) || ((pDrcInstructionUniDrc->downmixId[i] == DOWNMIX_ID_BASE_LAYOUT) && (pDrcInstructionUniDrc->drcSetId > 0))) { *pMatchFound = 1; break; } } return DRCDEC_SELECTION_PROCESS_NO_ERROR; } /* #4: The DRC set is not a "Fade-" or "Ducking-" only DRC set. */ static DRCDEC_SELECTION_PROCESS_RETURN _preSelectionRequirement4( DRC_INSTRUCTIONS_UNI_DRC* pDrcInstruction, int nDynamicRangeControlOn, int* pMatchFound) { *pMatchFound = 0; if (nDynamicRangeControlOn == 1) { if ((pDrcInstruction->drcSetEffect != EB_FADE) && (pDrcInstruction->drcSetEffect != EB_DUCK_OTHER) && (pDrcInstruction->drcSetEffect != EB_DUCK_SELF) && (pDrcInstruction->drcSetEffect != 0 || pDrcInstruction->drcSetId < 0)) { *pMatchFound = 1; } } else { *pMatchFound = 1; } return DRCDEC_SELECTION_PROCESS_NO_ERROR; } /* #5: The number of DRC bands is supported. Moreover, gainSetIndex and * gainSequenceIndex are within the allowed range. */ static DRCDEC_SELECTION_PROCESS_RETURN _preSelectionRequirement5( DRC_INSTRUCTIONS_UNI_DRC* pDrcInstructionUniDrc, DRC_COEFFICIENTS_UNI_DRC* pCoef, int* pMatchFound) { int b, i; *pMatchFound = 1; if (pDrcInstructionUniDrc->drcSetId < 0) /* virtual DRC sets are okay */ { return DRCDEC_SELECTION_PROCESS_NO_ERROR; } if (pCoef == NULL) /* check for parametricDRC */ { *pMatchFound = 0; /* parametricDRC not supported */ return DRCDEC_SELECTION_PROCESS_NO_ERROR; } if (pCoef->drcLocation != pDrcInstructionUniDrc ->drcLocation) /* drcLocation must be LOCATION_SELECTED */ { *pMatchFound = 0; return DRCDEC_SELECTION_PROCESS_NO_ERROR; } for (i = 0; i < pDrcInstructionUniDrc->nDrcChannelGroups; i++) { int indexDrcCoeff = pDrcInstructionUniDrc->gainSetIndexForChannelGroup[i]; int bandCount = 0; if (indexDrcCoeff >= 12) { *pMatchFound = 0; return DRCDEC_SELECTION_PROCESS_NO_ERROR; } if (indexDrcCoeff > pCoef->gainSetCount - 1) /* check for parametricDRC */ { continue; } GAIN_SET* gainSet = &(pCoef->gainSet[indexDrcCoeff]); bandCount = gainSet->bandCount; if (bandCount > 4) { *pMatchFound = 0; } for (b = 0; b < bandCount; b++) { if ((gainSet->gainSequenceIndex[b] >= 12) || (gainSet->gainSequenceIndex[b] >= pCoef->gainSequenceCount)) { *pMatchFound = 0; return DRCDEC_SELECTION_PROCESS_NO_ERROR; } } } return DRCDEC_SELECTION_PROCESS_NO_ERROR; } /* #6: Independent use of DRC set is permitted.*/ static DRCDEC_SELECTION_PROCESS_RETURN _preSelectionRequirement6( DRC_INSTRUCTIONS_UNI_DRC* pDrcInstructionUniDrc, int* pMatchFound) { *pMatchFound = 0; if (((pDrcInstructionUniDrc->dependsOnDrcSetPresent == 0) && (pDrcInstructionUniDrc->noIndependentUse == 0)) || (pDrcInstructionUniDrc->dependsOnDrcSetPresent == 1)) { *pMatchFound = 1; } return DRCDEC_SELECTION_PROCESS_NO_ERROR; } /* #7: DRC sets that require EQ are only permitted if EQ is supported. */ static DRCDEC_SELECTION_PROCESS_RETURN _preSelectionRequirement7( DRC_INSTRUCTIONS_UNI_DRC* pDrcInstructionUniDrc, int* pMatchFound) { *pMatchFound = 1; if (pDrcInstructionUniDrc->requiresEq) { /* EQ is not supported */ *pMatchFound = 0; } return DRCDEC_SELECTION_PROCESS_NO_ERROR; } static void _setSelectionDataInfo( DRCDEC_SELECTION_DATA* pData, FIXP_DBL loudness, /* e = 7 */ FIXP_DBL loudnessNormalizationGainDb, /* e = 7 */ FIXP_DBL loudnessNormalizationGainDbMax, /* e = 7 */ FIXP_DBL loudnessDeviationMax, /* e = 7 */ FIXP_DBL signalPeakLevel, /* e = 7 */ FIXP_DBL outputPeakLevelMax, /* e = 7 */ int applyAdjustment) { FIXP_DBL adjustment = 0; /* e = 8 */ /* use e = 8 for all function parameters to prevent overflow */ loudness >>= 1; loudnessNormalizationGainDb >>= 1; loudnessNormalizationGainDbMax >>= 1; loudnessDeviationMax >>= 1; signalPeakLevel >>= 1; outputPeakLevelMax >>= 1; if (applyAdjustment) { adjustment = fMax((FIXP_DBL)0, signalPeakLevel + loudnessNormalizationGainDb - outputPeakLevelMax); adjustment = fMin(adjustment, fMax((FIXP_DBL)0, loudnessDeviationMax)); } pData->loudnessNormalizationGainDbAdjusted = fMin( loudnessNormalizationGainDb - adjustment, loudnessNormalizationGainDbMax); pData->outputLoudness = loudness + pData->loudnessNormalizationGainDbAdjusted; pData->outputPeakLevel = signalPeakLevel + pData->loudnessNormalizationGainDbAdjusted; /* shift back to e = 7 using saturation */ pData->loudnessNormalizationGainDbAdjusted = SATURATE_LEFT_SHIFT( pData->loudnessNormalizationGainDbAdjusted, 1, DFRACT_BITS); pData->outputLoudness = SATURATE_LEFT_SHIFT(pData->outputLoudness, 1, DFRACT_BITS); pData->outputPeakLevel = SATURATE_LEFT_SHIFT(pData->outputPeakLevel, 1, DFRACT_BITS); } static int _targetLoudnessInRange( DRC_INSTRUCTIONS_UNI_DRC* pDrcInstructionUniDrc, FIXP_DBL targetLoudness) { int retVal = 0; FIXP_DBL drcSetTargetLoudnessValueUpper = ((FIXP_DBL)pDrcInstructionUniDrc->drcSetTargetLoudnessValueUpper) << (DFRACT_BITS - 1 - 7); FIXP_DBL drcSetTargetLoudnessValueLower = ((FIXP_DBL)pDrcInstructionUniDrc->drcSetTargetLoudnessValueLower) << (DFRACT_BITS - 1 - 7); if (pDrcInstructionUniDrc->drcSetTargetLoudnessPresent && drcSetTargetLoudnessValueUpper >= targetLoudness && drcSetTargetLoudnessValueLower < targetLoudness) { retVal = 1; } return retVal; } static int _drcSetIsUsable(HANDLE_UNI_DRC_CONFIG hUniDrcConfig, DRC_INSTRUCTIONS_UNI_DRC* pInst) { int usable = 0; DRC_COEFFICIENTS_UNI_DRC* pCoef = selectDrcCoefficients(hUniDrcConfig, LOCATION_SELECTED); /* check if ID is unique */ if (selectDrcInstructions(hUniDrcConfig, pInst->drcSetId) != pInst) return 0; /* sanity check on drcInstructions */ _preSelectionRequirement5(pInst, pCoef, &usable); return usable; } /* #8: The range of the target loudness specified for a DRC set has to include * the requested decoder target loudness. */ static DRCDEC_SELECTION_PROCESS_RETURN _preSelectionRequirement8( SEL_PROC_INPUT* hSelProcInput, int downmixIdIndex, HANDLE_UNI_DRC_CONFIG hUniDrcConfig, HANDLE_LOUDNESS_INFO_SET hLoudnessInfoSet, DRC_INSTRUCTIONS_UNI_DRC* pDrcInstructionUniDrc, DRCDEC_SELECTION* pCandidatesPotential, DRCDEC_SELECTION* pCandidatesSelected, SEL_PROC_CODEC_MODE codecMode) { DRCDEC_SELECTION_PROCESS_RETURN retVal = DRCDEC_SELECTION_PROCESS_NO_ERROR; int explicitPeakInformationPresent; FIXP_DBL signalPeakLevel; int addToCandidate = 0; FIXP_DBL loudnessNormalizationGainDb; FIXP_DBL loudness; FIXP_DBL loudnessDeviationMax = ((FIXP_DBL)hSelProcInput->loudnessDeviationMax) << (DFRACT_BITS - 1 - 7); { retVal = _getLoudness(hLoudnessInfoSet, hSelProcInput->albumMode, hSelProcInput->loudnessMeasurementMethod, hSelProcInput->loudnessMeasurementSystem, hSelProcInput->targetLoudness, pDrcInstructionUniDrc->drcSetId, hSelProcInput->downmixIdRequested[downmixIdIndex], &loudnessNormalizationGainDb, &loudness); if (retVal) return (retVal); } if (!hSelProcInput->loudnessNormalizationOn) { loudnessNormalizationGainDb = (FIXP_DBL)0; } retVal = _getSignalPeakLevel( hSelProcInput, hUniDrcConfig, hLoudnessInfoSet, pDrcInstructionUniDrc, hSelProcInput->downmixIdRequested[downmixIdIndex], &explicitPeakInformationPresent, &signalPeakLevel, codecMode ); if (retVal) return (retVal); if (hSelProcInput->dynamicRangeControlOn) { if (explicitPeakInformationPresent == 0) { if (pDrcInstructionUniDrc->drcSetTargetLoudnessPresent && ((hSelProcInput->loudnessNormalizationOn && _targetLoudnessInRange(pDrcInstructionUniDrc, hSelProcInput->targetLoudness)) || !hSelProcInput->loudnessNormalizationOn)) { DRCDEC_SELECTION_DATA* pData = _drcdec_selection_addNew(pCandidatesSelected); if (pData == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; _setSelectionDataInfo(pData, loudness, loudnessNormalizationGainDb, hSelProcInput->loudnessNormalizationGainDbMax, loudnessDeviationMax, signalPeakLevel, hSelProcInput->outputPeakLevelMax, 0); pData->downmixIdRequestIndex = downmixIdIndex; pData->pInst = pDrcInstructionUniDrc; pData->selectionFlag = 1; /* signal pre-selection step dealing with drcSetTargetLoudness */ if (hSelProcInput->loudnessNormalizationOn) { pData->outputPeakLevel = hSelProcInput->targetLoudness - (((FIXP_DBL)pData->pInst->drcSetTargetLoudnessValueUpper) << (DFRACT_BITS - 1 - 7)); } else { pData->outputPeakLevel = (FIXP_DBL)0; } } else { if ((!hSelProcInput->loudnessNormalizationOn) || (!pDrcInstructionUniDrc->drcSetTargetLoudnessPresent) || (hSelProcInput->loudnessNormalizationOn && _targetLoudnessInRange(pDrcInstructionUniDrc, hSelProcInput->targetLoudness))) { addToCandidate = 1; } } } else { addToCandidate = 1; } if (addToCandidate) { DRCDEC_SELECTION_DATA* pData = _drcdec_selection_addNew(pCandidatesPotential); if (pData == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; _setSelectionDataInfo(pData, loudness, loudnessNormalizationGainDb, hSelProcInput->loudnessNormalizationGainDbMax, loudnessDeviationMax, signalPeakLevel, hSelProcInput->outputPeakLevelMax, 0); pData->downmixIdRequestIndex = downmixIdIndex; pData->pInst = pDrcInstructionUniDrc; pData->selectionFlag = 0; } } else { if (pDrcInstructionUniDrc->drcSetId < 0) { DRCDEC_SELECTION_DATA* pData = _drcdec_selection_addNew(pCandidatesSelected); if (pData == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; _setSelectionDataInfo(pData, loudness, loudnessNormalizationGainDb, hSelProcInput->loudnessNormalizationGainDbMax, loudnessDeviationMax, signalPeakLevel, hSelProcInput->outputPeakLevelMax, 1); pData->downmixIdRequestIndex = downmixIdIndex; pData->pInst = pDrcInstructionUniDrc; pData->selectionFlag = 0; } } return DRCDEC_SELECTION_PROCESS_NO_ERROR; } /* #9: Clipping is minimized. */ static DRCDEC_SELECTION_PROCESS_RETURN _preSelectionRequirement9( SEL_PROC_INPUT* hSelProcInput, DRCDEC_SELECTION* pCandidatesPotential, DRCDEC_SELECTION* pCandidatesSelected) { int i; for (i = 0; i < _drcdec_selection_getNumber(pCandidatesPotential); i++) { DRCDEC_SELECTION_DATA* pCandidate = _drcdec_selection_getAt(pCandidatesPotential, i); if (pCandidate == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; if (pCandidate->outputPeakLevel <= hSelProcInput->outputPeakLevelMax) { if (_drcdec_selection_add(pCandidatesSelected, pCandidate) == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; } } return DRCDEC_SELECTION_PROCESS_NO_ERROR; } static DRCDEC_SELECTION_PROCESS_RETURN _drcSetPreSelectionSingleInstruction( SEL_PROC_INPUT* hSelProcInput, int downmixIdIndex, HANDLE_UNI_DRC_CONFIG hUniDrcConfig, HANDLE_LOUDNESS_INFO_SET hLoudnessInfoSet, DRC_INSTRUCTIONS_UNI_DRC* pDrcInstructionUniDrc, DRCDEC_SELECTION* pCandidatesPotential, DRCDEC_SELECTION* pCandidatesSelected, SEL_PROC_CODEC_MODE codecMode) { DRCDEC_SELECTION_PROCESS_RETURN retVal = DRCDEC_SELECTION_PROCESS_NO_ERROR; int matchFound = 0; DRC_COEFFICIENTS_UNI_DRC* pCoef = selectDrcCoefficients(hUniDrcConfig, LOCATION_SELECTED); retVal = _preSelectionRequirement123( hSelProcInput->downmixIdRequested[downmixIdIndex], pDrcInstructionUniDrc, &matchFound); if (!retVal && matchFound) retVal = _preSelectionRequirement4(pDrcInstructionUniDrc, hSelProcInput->dynamicRangeControlOn, &matchFound); if (!retVal && matchFound) retVal = _preSelectionRequirement5(pDrcInstructionUniDrc, pCoef, &matchFound); if (!retVal && matchFound) retVal = _preSelectionRequirement6(pDrcInstructionUniDrc, &matchFound); if (!retVal && matchFound) retVal = _preSelectionRequirement7(pDrcInstructionUniDrc, &matchFound); if (!retVal && matchFound) retVal = _preSelectionRequirement8( hSelProcInput, downmixIdIndex, hUniDrcConfig, hLoudnessInfoSet, pDrcInstructionUniDrc, pCandidatesPotential, pCandidatesSelected, codecMode); return retVal; } static DRCDEC_SELECTION_PROCESS_RETURN _drcSetSelectionAddCandidates( SEL_PROC_INPUT* hSelProcInput, DRCDEC_SELECTION* pCandidatesPotential, DRCDEC_SELECTION* pCandidatesSelected) { DRCDEC_SELECTION_PROCESS_RETURN retVal = DRCDEC_SELECTION_PROCESS_NO_ERROR; int nHitCount = 0; int i; DRCDEC_SELECTION_DATA* pCandidate = NULL; DRC_INSTRUCTIONS_UNI_DRC* pDrcInstructionUniDrc = NULL; for (i = 0; i < _drcdec_selection_getNumber(pCandidatesPotential); i++) { pCandidate = _drcdec_selection_getAt(pCandidatesPotential, i); if (pCandidate == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; pDrcInstructionUniDrc = pCandidate->pInst; if (_targetLoudnessInRange(pDrcInstructionUniDrc, hSelProcInput->targetLoudness)) { nHitCount++; } } if (nHitCount != 0) { for (i = 0; i < _drcdec_selection_getNumber(pCandidatesPotential); i++) { pCandidate = _drcdec_selection_getAt(pCandidatesPotential, i); if (pCandidate == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; pDrcInstructionUniDrc = pCandidate->pInst; if (_targetLoudnessInRange(pDrcInstructionUniDrc, hSelProcInput->targetLoudness)) { if (_drcdec_selection_add(pCandidatesSelected, pCandidate) == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; } } } else { FIXP_DBL lowestPeakLevel = MAXVAL_DBL; /* e = 7 */ FIXP_DBL peakLevel = 0; /* e = 7 */ for (i = 0; i < _drcdec_selection_getNumber(pCandidatesPotential); i++) { pCandidate = _drcdec_selection_getAt(pCandidatesPotential, i); if (pCandidate == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; peakLevel = pCandidate->outputPeakLevel; if (peakLevel < lowestPeakLevel) { lowestPeakLevel = peakLevel; } } /* add all with lowest peak level or max 1dB above */ for (i = 0; i < _drcdec_selection_getNumber(pCandidatesPotential); i++) { FIXP_DBL loudnessDeviationMax = ((FIXP_DBL)hSelProcInput->loudnessDeviationMax) << (DFRACT_BITS - 1 - 7); /* e = 7 */ pCandidate = _drcdec_selection_getAt(pCandidatesPotential, i); if (pCandidate == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; peakLevel = pCandidate->outputPeakLevel; if (peakLevel == lowestPeakLevel || peakLevel <= lowestPeakLevel + FL2FXCONST_DBL(1.0f / (float)(1 << 7))) { FIXP_DBL adjustment = fMax((FIXP_DBL)0, peakLevel - hSelProcInput->outputPeakLevelMax); adjustment = fMin(adjustment, fMax((FIXP_DBL)0, loudnessDeviationMax)); pCandidate->loudnessNormalizationGainDbAdjusted -= adjustment; pCandidate->outputPeakLevel -= adjustment; pCandidate->outputLoudness -= adjustment; if (_drcdec_selection_add(pCandidatesSelected, pCandidate) == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; } } } return retVal; } static DRCDEC_SELECTION_PROCESS_RETURN _dependentDrcInstruction( HANDLE_UNI_DRC_CONFIG hUniDrcConfig, DRC_INSTRUCTIONS_UNI_DRC* pInst, DRC_INSTRUCTIONS_UNI_DRC** ppDrcInstructionsDependent) { int i; DRC_INSTRUCTIONS_UNI_DRC* pDependentDrc = NULL; for (i = 0; i < hUniDrcConfig->drcInstructionsUniDrcCount; i++) { pDependentDrc = (DRC_INSTRUCTIONS_UNI_DRC*)&(hUniDrcConfig->drcInstructionsUniDrc[i]); if (pDependentDrc->drcSetId == pInst->dependsOnDrcSet) { break; } } if (i == hUniDrcConfig->drcInstructionsUniDrcCount) { return DRCDEC_SELECTION_PROCESS_NOT_OK; } if (pDependentDrc->dependsOnDrcSetPresent == 1) { return DRCDEC_SELECTION_PROCESS_NOT_OK; } *ppDrcInstructionsDependent = pDependentDrc; return DRCDEC_SELECTION_PROCESS_NO_ERROR; } static DRCDEC_SELECTION_PROCESS_RETURN _selectDrcSetEffectNone( HANDLE_UNI_DRC_CONFIG hUniDrcConfig, DRCDEC_SELECTION* pCandidatesPotential, DRCDEC_SELECTION* pCandidatesSelected) { int i; for (i = 0; i < _drcdec_selection_getNumber(pCandidatesPotential); i++) { DRCDEC_SELECTION_DATA* pCandidate = _drcdec_selection_getAt(pCandidatesPotential, i); if (pCandidate == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; if ((pCandidate->pInst->drcSetEffect & 0xff) == 0) { if (_drcdec_selection_add(pCandidatesSelected, pCandidate) == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; } } return DRCDEC_SELECTION_PROCESS_NO_ERROR; } static DRCDEC_SELECTION_PROCESS_RETURN _selectSingleEffectType( HANDLE_UNI_DRC_CONFIG hUniDrcConfig, DRC_EFFECT_TYPE_REQUEST effectType, DRCDEC_SELECTION* pCandidatesPotential, DRCDEC_SELECTION* pCandidatesSelected) { int i; DRCDEC_SELECTION_PROCESS_RETURN retVal = DRCDEC_SELECTION_PROCESS_NO_ERROR; DRC_INSTRUCTIONS_UNI_DRC* pInst; DRC_INSTRUCTIONS_UNI_DRC* pDrcInstructionsDependent; if (effectType == DETR_NONE) { retVal = _selectDrcSetEffectNone(hUniDrcConfig, pCandidatesPotential, pCandidatesSelected); if (retVal) return (retVal); } else { int effectBitPosition = 1 << (effectType - 1); for (i = 0; i < _drcdec_selection_getNumber(pCandidatesPotential); i++) { DRCDEC_SELECTION_DATA* pCandidate = _drcdec_selection_getAt(pCandidatesPotential, i); if (pCandidate == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; pInst = pCandidate->pInst; if (!pInst->dependsOnDrcSetPresent) { if ((pInst->drcSetEffect & effectBitPosition)) { if (_drcdec_selection_add(pCandidatesSelected, pCandidate) == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; } } else { retVal = _dependentDrcInstruction(hUniDrcConfig, pInst, &pDrcInstructionsDependent); if (retVal) return (retVal); if (((pInst->drcSetEffect & effectBitPosition)) || ((pDrcInstructionsDependent->drcSetEffect & effectBitPosition))) { if (_drcdec_selection_add(pCandidatesSelected, pCandidate) == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; } } } } return retVal; } static DRCDEC_SELECTION_PROCESS_RETURN _selectEffectTypeFeature( HANDLE_UNI_DRC_CONFIG hUniDrcConfig, DRC_FEATURE_REQUEST drcFeatureRequest, DRCDEC_SELECTION** ppCandidatesPotential, DRCDEC_SELECTION** ppCandidatesSelected) { DRCDEC_SELECTION_PROCESS_RETURN retVal = DRCDEC_SELECTION_PROCESS_NO_ERROR; int i; int desiredEffectTypeFound = 0; for (i = 0; i < drcFeatureRequest.drcEffectType.numRequestsDesired; i++) { retVal = _selectSingleEffectType( hUniDrcConfig, drcFeatureRequest.drcEffectType.request[i], *ppCandidatesPotential, *ppCandidatesSelected); if (retVal) return (retVal); if (_drcdec_selection_getNumber(*ppCandidatesSelected)) { desiredEffectTypeFound = 1; _swapSelectionAndClear(ppCandidatesPotential, ppCandidatesSelected); } } if (!desiredEffectTypeFound) { for (i = drcFeatureRequest.drcEffectType.numRequestsDesired; i < drcFeatureRequest.drcEffectType.numRequests; i++) { retVal = _selectSingleEffectType( hUniDrcConfig, drcFeatureRequest.drcEffectType.request[i], *ppCandidatesPotential, *ppCandidatesSelected); if (retVal) return (retVal); if (_drcdec_selection_getNumber(*ppCandidatesSelected)) { _swapSelectionAndClear(ppCandidatesPotential, ppCandidatesSelected); break; } } } _swapSelection(ppCandidatesPotential, ppCandidatesSelected); return retVal; } static DRCDEC_SELECTION_PROCESS_RETURN _selectDynamicRange( HANDLE_UNI_DRC_CONFIG hUniDrcConfig, HANDLE_LOUDNESS_INFO_SET hLoudnessInfoSet, DRC_FEATURE_REQUEST drcFeatureRequest, UCHAR* pDownmixIdRequested, int albumMode, DRCDEC_SELECTION* pCandidatesPotential, DRCDEC_SELECTION* ppCandidatesSelected) { DRCDEC_SELECTION_PROCESS_RETURN retVal = DRCDEC_SELECTION_PROCESS_NO_ERROR; int i; int peakToAveragePresent; FIXP_DBL peakToAverage; FIXP_DBL minVal = MAXVAL_DBL; FIXP_DBL val = 0; int numSelectedCandidates = _drcdec_selection_getNumber(ppCandidatesSelected); for (i = 0; i < _drcdec_selection_getNumber(pCandidatesPotential); i++) { DRCDEC_SELECTION_DATA* pCandidate = _drcdec_selection_getAt(pCandidatesPotential, i); if (pCandidate == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; retVal = _dynamicRangeMeasurement( hLoudnessInfoSet, pCandidate->pInst, pDownmixIdRequested[pCandidate->downmixIdRequestIndex], drcFeatureRequest.dynamicRange.measurementRequestType, albumMode, &peakToAveragePresent, &peakToAverage); if (retVal) return (retVal); if (peakToAveragePresent) { if (!drcFeatureRequest.dynamicRange.requestedIsRange) { val = fAbs(drcFeatureRequest.dynamicRange.requestValue - peakToAverage); if (minVal > val) { minVal = val; _drcdec_selection_setNumber(ppCandidatesSelected, numSelectedCandidates); } if (_drcdec_selection_add(ppCandidatesSelected, pCandidate) == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; } else { if ((peakToAverage >= drcFeatureRequest.dynamicRange.requestValueMin) && (peakToAverage <= drcFeatureRequest.dynamicRange.requestValueMax)) { if (_drcdec_selection_add(ppCandidatesSelected, pCandidate) == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; } } } } return retVal; } static DRCDEC_SELECTION_PROCESS_RETURN _selectSingleDrcCharacteristic( HANDLE_UNI_DRC_CONFIG hUniDrcConfig, int requestedDrcCharacteristic, DRCDEC_SELECTION** ppCandidatesPotential, DRCDEC_SELECTION** ppCandidatesSelected) { int i, j, b; int hit = 0; DRC_INSTRUCTIONS_UNI_DRC* pInst = NULL; DRC_COEFFICIENTS_UNI_DRC* pCoef = NULL; GAIN_SET* pGainSet = NULL; if (requestedDrcCharacteristic < 1) { return DRCDEC_SELECTION_PROCESS_NOT_OK; } pCoef = selectDrcCoefficients(hUniDrcConfig, LOCATION_SELECTED); if (pCoef == NULL) /* check for parametricDRC */ { return DRCDEC_SELECTION_PROCESS_NO_ERROR; } for (i = 0; i < _drcdec_selection_getNumber(*ppCandidatesPotential); i++) { DRCDEC_SELECTION_DATA* pCandidate = _drcdec_selection_getAt(*ppCandidatesPotential, i); if (pCandidate == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; pInst = pCandidate->pInst; hit = 0; for (j = 0; j < pInst->nDrcChannelGroups; j++) { int bandCount = 0; int indexDrcCoeff = pInst->gainSetIndexForChannelGroup[j]; if (indexDrcCoeff > pCoef->gainSetCount - 1) /* check for parametricDRC */ { return DRCDEC_SELECTION_PROCESS_NO_ERROR; } pGainSet = &(pCoef->gainSet[indexDrcCoeff]); bandCount = pGainSet->bandCount; for (b = 0; b < bandCount; b++) { if ((pGainSet->drcCharacteristic[b].isCICP) && (pGainSet->drcCharacteristic[b].cicpIndex == requestedDrcCharacteristic)) { hit = 1; break; } } if (hit) break; } if (hit) { if (_drcdec_selection_add(*ppCandidatesSelected, pCandidate) == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; } } if (_drcdec_selection_getNumber(*ppCandidatesSelected)) { _swapSelectionAndClear(ppCandidatesPotential, ppCandidatesSelected); } return DRCDEC_SELECTION_PROCESS_NO_ERROR; } static DRCDEC_SELECTION_PROCESS_RETURN _selectDrcCharacteristic( HANDLE_UNI_DRC_CONFIG hUniDrcConfig, int drcCharacteristicRequested, DRCDEC_SELECTION** ppCandidatesPotential, DRCDEC_SELECTION** ppCandidatesSelected) { DRCDEC_SELECTION_PROCESS_RETURN retVal = DRCDEC_SELECTION_PROCESS_NO_ERROR; const int secondTry[12] = {0, 2, 3, 4, 5, 6, 5, 9, 10, 7, 8, 10}; retVal = _selectSingleDrcCharacteristic( hUniDrcConfig, drcCharacteristicRequested, ppCandidatesPotential, ppCandidatesSelected); if (retVal) return (retVal); if ((drcCharacteristicRequested <= 11) && (_drcdec_selection_getNumber(*ppCandidatesSelected) == 0)) { retVal = _selectSingleDrcCharacteristic( hUniDrcConfig, secondTry[drcCharacteristicRequested], ppCandidatesPotential, ppCandidatesSelected); if (retVal) return (retVal); } if (_drcdec_selection_getNumber(*ppCandidatesSelected) == 0) { if ((drcCharacteristicRequested >= 2) && (drcCharacteristicRequested <= 5)) { retVal = _selectSingleDrcCharacteristic( hUniDrcConfig, drcCharacteristicRequested - 1, ppCandidatesPotential, ppCandidatesSelected); if (retVal) return (retVal); } else if (drcCharacteristicRequested == 11) { retVal = _selectSingleDrcCharacteristic( hUniDrcConfig, 9, ppCandidatesPotential, ppCandidatesSelected); if (retVal) return (retVal); } } _swapSelection(ppCandidatesPotential, ppCandidatesSelected); return retVal; } static DRCDEC_SELECTION_PROCESS_RETURN _drcSetFinalSelection_peakValue0( DRCDEC_SELECTION* pCandidatesPotential, DRCDEC_SELECTION* pCandidatesSelected) { int i; for (i = 0; i < _drcdec_selection_getNumber(pCandidatesPotential); i++) { DRCDEC_SELECTION_DATA* pCandidate = _drcdec_selection_getAt(pCandidatesPotential, i); if (pCandidate == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; if (pCandidate->outputPeakLevel <= FIXP_DBL(0)) { if (_drcdec_selection_add(pCandidatesSelected, pCandidate) == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; } } return DRCDEC_SELECTION_PROCESS_NO_ERROR; } static DRCDEC_SELECTION_PROCESS_RETURN _drcSetFinalSelection_downmixId( HANDLE_SEL_PROC_INPUT hSelProcInput, DRCDEC_SELECTION** ppCandidatesPotential, DRCDEC_SELECTION** ppCandidatesSelected) { int i, j; DRCDEC_SELECTION_DATA* pCandidate = NULL; DRC_INSTRUCTIONS_UNI_DRC* pInst = NULL; for (i = 0; i < _drcdec_selection_getNumber(*ppCandidatesPotential); i++) { pCandidate = _drcdec_selection_getAt(*ppCandidatesPotential, i); if (pCandidate == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; pInst = pCandidate->pInst; for (j = 0; j < pInst->downmixIdCount; j++) { if (DOWNMIX_ID_BASE_LAYOUT != pInst->downmixId[j] && DOWNMIX_ID_ANY_DOWNMIX != pInst->downmixId[j] && hSelProcInput ->downmixIdRequested[pCandidate->downmixIdRequestIndex] == pInst->downmixId[j]) { if (_drcdec_selection_add(*ppCandidatesSelected, pCandidate) == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; } } } if (_drcdec_selection_getNumber(*ppCandidatesSelected) == 0) { _swapSelection(ppCandidatesPotential, ppCandidatesSelected); } return DRCDEC_SELECTION_PROCESS_NO_ERROR; } static int _crossSum(int value) { int sum = 0; while (value != 0) { if ((value & 1) == 1) { sum++; } value >>= 1; } return sum; } static DRCDEC_SELECTION_PROCESS_RETURN _drcSetFinalSelection_effectTypes( DRCDEC_SELECTION* pCandidatesPotential, DRCDEC_SELECTION* pCandidatesSelected) { int i; int minNumEffects = 1000; int numEffects = 0; int effects = 0; DRCDEC_SELECTION_DATA* pCandidate = NULL; DRC_INSTRUCTIONS_UNI_DRC* pInst = NULL; for (i = 0; i < _drcdec_selection_getNumber(pCandidatesPotential); i++) { pCandidate = _drcdec_selection_getAt(pCandidatesPotential, i); if (pCandidate == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; pInst = pCandidate->pInst; effects = pInst->drcSetEffect; effects &= 0xffff ^ (EB_GENERAL_COMPR); numEffects = _crossSum(effects); if (numEffects < minNumEffects) { minNumEffects = numEffects; } } /* add all with minimum number of effects */ for (i = 0; i < _drcdec_selection_getNumber(pCandidatesPotential); i++) { pCandidate = _drcdec_selection_getAt(pCandidatesPotential, i); if (pCandidate == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; pInst = pCandidate->pInst; effects = pInst->drcSetEffect; effects &= 0xffff ^ (EB_GENERAL_COMPR); numEffects = _crossSum(effects); if (numEffects == minNumEffects) { if (_drcdec_selection_add(pCandidatesSelected, pCandidate) == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; } } return DRCDEC_SELECTION_PROCESS_NO_ERROR; } static DRCDEC_SELECTION_PROCESS_RETURN _selectSmallestTargetLoudnessValueUpper( DRCDEC_SELECTION* pCandidatesPotential, DRCDEC_SELECTION* pCandidatesSelected) { int i; SCHAR minVal = 0x7F; SCHAR val = 0; DRCDEC_SELECTION_DATA* pCandidate = NULL; for (i = 0; i < _drcdec_selection_getNumber(pCandidatesPotential); i++) { pCandidate = _drcdec_selection_getAt(pCandidatesPotential, i); if (pCandidate == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; val = pCandidate->pInst->drcSetTargetLoudnessValueUpper; if (val < minVal) { minVal = val; } } /* add all with same smallest drcSetTargetLoudnessValueUpper */ for (i = 0; i < _drcdec_selection_getNumber(pCandidatesPotential); i++) { pCandidate = _drcdec_selection_getAt(pCandidatesPotential, i); if (pCandidate == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; val = pCandidate->pInst->drcSetTargetLoudnessValueUpper; if (val == minVal) { if (_drcdec_selection_add(pCandidatesSelected, pCandidate) == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; } } return DRCDEC_SELECTION_PROCESS_NO_ERROR; } static DRCDEC_SELECTION_PROCESS_RETURN _drcSetFinalSelection_targetLoudness( FIXP_DBL targetLoudness, DRCDEC_SELECTION* pCandidatesPotential, DRCDEC_SELECTION* pCandidatesSelected) { DRCDEC_SELECTION_PROCESS_RETURN retVal = DRCDEC_SELECTION_PROCESS_NO_ERROR; int i; DRCDEC_SELECTION_DATA* pCandidate = NULL; for (i = 0; i < _drcdec_selection_getNumber(pCandidatesPotential); i++) { pCandidate = _drcdec_selection_getAt(pCandidatesPotential, i); if (pCandidate == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; if (pCandidate->selectionFlag == 0) { if (_drcdec_selection_add(pCandidatesSelected, pCandidate) == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; } } if (_drcdec_selection_getNumber(pCandidatesSelected) == 0) { retVal = _selectSmallestTargetLoudnessValueUpper(pCandidatesPotential, pCandidatesSelected); if (retVal) return (retVal); } if (_drcdec_selection_getNumber(pCandidatesSelected) > 1) { DRC_INSTRUCTIONS_UNI_DRC* pDrcInstructionUniDrc = NULL; _swapSelectionAndClear(&pCandidatesPotential, &pCandidatesSelected); for (i = 0; i < _drcdec_selection_getNumber(pCandidatesPotential); i++) { pCandidate = _drcdec_selection_getAt(pCandidatesPotential, i); if (pCandidate == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; pDrcInstructionUniDrc = pCandidate->pInst; if (_targetLoudnessInRange(pDrcInstructionUniDrc, targetLoudness)) { if (_drcdec_selection_add(pCandidatesSelected, pCandidate) == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; } } if (_drcdec_selection_getNumber(pCandidatesSelected) > 1) { _swapSelectionAndClear(&pCandidatesPotential, &pCandidatesSelected); retVal = _selectSmallestTargetLoudnessValueUpper(pCandidatesPotential, pCandidatesSelected); if (retVal) return (retVal); } } return retVal; } static DRCDEC_SELECTION_PROCESS_RETURN _drcSetFinalSelection_peakValueLargest( DRCDEC_SELECTION* pCandidatesPotential, DRCDEC_SELECTION* pCandidatesSelected) { int i; FIXP_DBL largestPeakLevel = MINVAL_DBL; FIXP_DBL peakLevel = 0; DRCDEC_SELECTION_DATA* pCandidate = NULL; for (i = 0; i < _drcdec_selection_getNumber(pCandidatesPotential); i++) { pCandidate = _drcdec_selection_getAt(pCandidatesPotential, i); if (pCandidate == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; peakLevel = pCandidate->outputPeakLevel; if (peakLevel > largestPeakLevel) { largestPeakLevel = peakLevel; } } /* add all with same largest peak level */ for (i = 0; i < _drcdec_selection_getNumber(pCandidatesPotential); i++) { pCandidate = _drcdec_selection_getAt(pCandidatesPotential, i); if (pCandidate == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; peakLevel = pCandidate->outputPeakLevel; if (peakLevel == largestPeakLevel) { if (_drcdec_selection_add(pCandidatesSelected, pCandidate) == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; } } return DRCDEC_SELECTION_PROCESS_NO_ERROR; } static DRCDEC_SELECTION_PROCESS_RETURN _drcSetFinalSelection_drcSetId( DRCDEC_SELECTION* pCandidatesPotential, DRCDEC_SELECTION* pCandidatesSelected) { int i; int largestId = -1000; int id = 0; DRCDEC_SELECTION_DATA* pCandidate = NULL; DRCDEC_SELECTION_DATA* pCandidateSelected = NULL; for (i = 0; i < _drcdec_selection_getNumber(pCandidatesPotential); i++) { pCandidate = _drcdec_selection_getAt(pCandidatesPotential, i); if (pCandidate == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; id = pCandidate->pInst->drcSetId; if (id > largestId) { largestId = id; pCandidateSelected = pCandidate; } } if (pCandidateSelected != NULL) { if (_drcdec_selection_add(pCandidatesSelected, pCandidateSelected) == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; } else { return DRCDEC_SELECTION_PROCESS_NOT_OK; } return DRCDEC_SELECTION_PROCESS_NO_ERROR; } static DRCDEC_SELECTION_PROCESS_RETURN _drcSetFinalSelection( HANDLE_SEL_PROC_INPUT hSelProcInput, HANDLE_UNI_DRC_CONFIG hUniDrcConfig, DRCDEC_SELECTION** ppCandidatesPotential, DRCDEC_SELECTION** ppCandidatesSelected, SEL_PROC_CODEC_MODE codecMode) { DRCDEC_SELECTION_PROCESS_RETURN retVal = DRCDEC_SELECTION_PROCESS_NO_ERROR; if (_drcdec_selection_getNumber(*ppCandidatesPotential) == 0) { return DRCDEC_SELECTION_PROCESS_NOT_OK; } else if (_drcdec_selection_getNumber(*ppCandidatesPotential) == 1) { _swapSelection(ppCandidatesPotential, ppCandidatesSelected); /* finished */ } else /* > 1 */ { retVal = _drcSetFinalSelection_peakValue0(*ppCandidatesPotential, *ppCandidatesSelected); if (retVal) return (retVal); if (_drcdec_selection_getNumber(*ppCandidatesSelected) > 1) { _swapSelectionAndClear(ppCandidatesPotential, ppCandidatesSelected); retVal = _drcSetFinalSelection_downmixId( hSelProcInput, ppCandidatesPotential, ppCandidatesSelected); if (retVal) return (retVal); } if (_drcdec_selection_getNumber(*ppCandidatesSelected) > 1) { _swapSelectionAndClear(ppCandidatesPotential, ppCandidatesSelected); retVal = _drcSetFinalSelection_effectTypes(*ppCandidatesPotential, *ppCandidatesSelected); if (retVal) return (retVal); } if (_drcdec_selection_getNumber(*ppCandidatesSelected) > 1) { _swapSelectionAndClear(ppCandidatesPotential, ppCandidatesSelected); retVal = _drcSetFinalSelection_targetLoudness( hSelProcInput->targetLoudness, *ppCandidatesPotential, *ppCandidatesSelected); if (retVal) return (retVal); } if (_drcdec_selection_getNumber(*ppCandidatesSelected) > 1) { _swapSelectionAndClear(ppCandidatesPotential, ppCandidatesSelected); retVal = _drcSetFinalSelection_peakValueLargest(*ppCandidatesPotential, *ppCandidatesSelected); if (retVal) return (retVal); } if (_drcdec_selection_getNumber(*ppCandidatesSelected) > 1) { _swapSelectionAndClear(ppCandidatesPotential, ppCandidatesSelected); retVal = _drcSetFinalSelection_drcSetId(*ppCandidatesPotential, *ppCandidatesSelected); if (retVal) return (retVal); } } if (_drcdec_selection_getNumber(*ppCandidatesSelected) == 0) { return DRCDEC_SELECTION_PROCESS_NOT_OK; } return retVal; } static DRCDEC_SELECTION_PROCESS_RETURN _generateVirtualDrcSets( HANDLE_SEL_PROC_INPUT hSelProcInput, HANDLE_UNI_DRC_CONFIG hUniDrcConfig, SEL_PROC_CODEC_MODE codecMode) { int i; int nMixes = hUniDrcConfig->downmixInstructionsCount + 1; int index = hUniDrcConfig->drcInstructionsUniDrcCount; int indexVirtual = -1; DRC_INSTRUCTIONS_UNI_DRC* pDrcInstruction = &(hUniDrcConfig->drcInstructionsUniDrc[index]); if (codecMode == SEL_PROC_MPEG_H_3DA) { nMixes = 1; } if ((index + nMixes) > (12 + 1 + 6)) { return DRCDEC_SELECTION_PROCESS_NOT_OK; } FDKmemset(pDrcInstruction, 0, sizeof(DRC_INSTRUCTIONS_UNI_DRC)); pDrcInstruction->drcSetId = indexVirtual; index++; indexVirtual--; pDrcInstruction->downmixIdCount = 1; if ((codecMode == SEL_PROC_MPEG_H_3DA) && (hSelProcInput->numDownmixIdRequests)) { pDrcInstruction->downmixId[0] = hSelProcInput->downmixIdRequested[0]; } else { pDrcInstruction->downmixId[0] = DOWNMIX_ID_BASE_LAYOUT; } for (i = 1; i < nMixes; i++) { pDrcInstruction = &(hUniDrcConfig->drcInstructionsUniDrc[index]); FDKmemset(pDrcInstruction, 0, sizeof(DRC_INSTRUCTIONS_UNI_DRC)); pDrcInstruction->drcSetId = indexVirtual; pDrcInstruction->downmixId[0] = hUniDrcConfig->downmixInstructions[i - 1].downmixId; pDrcInstruction->downmixIdCount = 1; index++; indexVirtual--; } hUniDrcConfig->drcInstructionsCountInclVirtual = hUniDrcConfig->drcInstructionsUniDrcCount + nMixes; return DRCDEC_SELECTION_PROCESS_NO_ERROR; } static DRCDEC_SELECTION_PROCESS_RETURN _generateOutputInfo( HANDLE_SEL_PROC_INPUT hSelProcInput, HANDLE_SEL_PROC_OUTPUT hSelProcOutput, HANDLE_UNI_DRC_CONFIG hUniDrcConfig, HANDLE_LOUDNESS_INFO_SET hLoudnessInfoSet, DRCDEC_SELECTION_DATA* pSelectionData, SEL_PROC_CODEC_MODE codecMode) { DRCDEC_SELECTION_PROCESS_RETURN retVal = DRCDEC_SELECTION_PROCESS_NO_ERROR; int i, j; int hasDependend = 0; int hasFading = 0; int hasDucking = 0; int selectedDrcSetIds; int selectedDownmixIds; FIXP_DBL mixingLevel = 0; int albumMode = hSelProcInput->albumMode; UCHAR* pDownmixIdRequested = hSelProcInput->downmixIdRequested; FIXP_SGL boost = hSelProcInput->boost; FIXP_SGL compress = hSelProcInput->compress; hSelProcOutput->numSelectedDrcSets = 1; hSelProcOutput->selectedDrcSetIds[0] = pSelectionData->pInst->drcSetId; hSelProcOutput->selectedDownmixIds[0] = pSelectionData->pInst->drcApplyToDownmix == 1 ? pSelectionData->pInst->downmixId[0] : 0; hSelProcOutput->loudnessNormalizationGainDb = pSelectionData->loudnessNormalizationGainDbAdjusted + hSelProcInput->loudnessNormalizationGainModificationDb; hSelProcOutput->outputPeakLevelDb = pSelectionData->outputPeakLevel; hSelProcOutput->outputLoudness = pSelectionData->outputLoudness; hSelProcOutput->boost = boost; hSelProcOutput->compress = compress; hSelProcOutput->baseChannelCount = hUniDrcConfig->channelLayout.baseChannelCount; hSelProcOutput->targetChannelCount = hUniDrcConfig->channelLayout.baseChannelCount; hSelProcOutput->activeDownmixId = pDownmixIdRequested[pSelectionData->downmixIdRequestIndex]; _getMixingLevel(hLoudnessInfoSet, *pDownmixIdRequested, hSelProcOutput->selectedDrcSetIds[0], albumMode, &mixingLevel); hSelProcOutput->mixingLevel = mixingLevel; /*dependent*/ if (pSelectionData->pInst->dependsOnDrcSetPresent) { int dependsOnDrcSetID = pSelectionData->pInst->dependsOnDrcSet; for (i = 0; i < hUniDrcConfig->drcInstructionsCountInclVirtual; i++) { DRC_INSTRUCTIONS_UNI_DRC* pInst = &(hUniDrcConfig->drcInstructionsUniDrc[i]); if (!_drcSetIsUsable(hUniDrcConfig, pInst)) continue; if (pInst->drcSetId == dependsOnDrcSetID) { hSelProcOutput->selectedDrcSetIds[hSelProcOutput->numSelectedDrcSets] = hUniDrcConfig->drcInstructionsUniDrc[i].drcSetId; hSelProcOutput->selectedDownmixIds[hSelProcOutput->numSelectedDrcSets] = hUniDrcConfig->drcInstructionsUniDrc[i].drcApplyToDownmix == 1 ? hUniDrcConfig->drcInstructionsUniDrc[i].downmixId[0] : 0; hSelProcOutput->numSelectedDrcSets++; hasDependend = 1; break; } } } /* fading */ if (hSelProcInput->albumMode == 0) { for (i = 0; i < hUniDrcConfig->drcInstructionsUniDrcCount; i++) { DRC_INSTRUCTIONS_UNI_DRC* pInst = &(hUniDrcConfig->drcInstructionsUniDrc[i]); if (!_drcSetIsUsable(hUniDrcConfig, pInst)) continue; if (pInst->drcSetEffect & EB_FADE) { if (pInst->downmixId[0] == DOWNMIX_ID_ANY_DOWNMIX) { hSelProcOutput->numSelectedDrcSets = hasDependend + 1; hSelProcOutput ->selectedDrcSetIds[hSelProcOutput->numSelectedDrcSets] = hUniDrcConfig->drcInstructionsUniDrc[i].drcSetId; hSelProcOutput ->selectedDownmixIds[hSelProcOutput->numSelectedDrcSets] = hUniDrcConfig->drcInstructionsUniDrc[i].drcApplyToDownmix == 1 ? hUniDrcConfig->drcInstructionsUniDrc[i].downmixId[0] : 0; hSelProcOutput->numSelectedDrcSets++; hasFading = 1; } else { retVal = DRCDEC_SELECTION_PROCESS_NOT_OK; if (retVal) return DRCDEC_SELECTION_PROCESS_NOT_OK; } } } } /* ducking */ for (i = 0; i < hUniDrcConfig->drcInstructionsUniDrcCount; i++) { DRC_INSTRUCTIONS_UNI_DRC* pInst = &(hUniDrcConfig->drcInstructionsUniDrc[i]); if (!_drcSetIsUsable(hUniDrcConfig, pInst)) continue; if (pInst->drcSetEffect & (EB_DUCK_OTHER | EB_DUCK_SELF)) { for (j = 0; j < pInst->downmixIdCount; j++) { if (pInst->downmixId[j] == hSelProcOutput->activeDownmixId) { hSelProcOutput->numSelectedDrcSets = hasDependend + 1; /* ducking overrides fading */ hSelProcOutput ->selectedDrcSetIds[hSelProcOutput->numSelectedDrcSets] = hUniDrcConfig->drcInstructionsUniDrc[i].drcSetId; /* force ducking DRC set to be processed on base layout */ hSelProcOutput ->selectedDownmixIds[hSelProcOutput->numSelectedDrcSets] = 0; hSelProcOutput->numSelectedDrcSets++; hasDucking = 1; } } } } /* repeat for DOWNMIX_ID_BASE_LAYOUT if no ducking found*/ if (!hasDucking) { for (i = 0; i < hUniDrcConfig->drcInstructionsUniDrcCount; i++) { DRC_INSTRUCTIONS_UNI_DRC* pInst = &(hUniDrcConfig->drcInstructionsUniDrc[i]); if (!_drcSetIsUsable(hUniDrcConfig, pInst)) continue; if (pInst->drcSetEffect & (EB_DUCK_OTHER | EB_DUCK_SELF)) { for (j = 0; j < pInst->downmixIdCount; j++) { if (pInst->downmixId[j] == DOWNMIX_ID_BASE_LAYOUT) { hSelProcOutput->numSelectedDrcSets = hasDependend + hasFading + 1; hSelProcOutput ->selectedDrcSetIds[hSelProcOutput->numSelectedDrcSets] = hUniDrcConfig->drcInstructionsUniDrc[i].drcSetId; /* force ducking DRC set to be processed on base layout */ hSelProcOutput ->selectedDownmixIds[hSelProcOutput->numSelectedDrcSets] = 0; hSelProcOutput->numSelectedDrcSets++; } } } } } if (hSelProcOutput->numSelectedDrcSets > 3) { /* maximum permitted number of applied DRC sets is 3, see section 6.3.5 of * ISO/IEC 23003-4 */ hSelProcOutput->numSelectedDrcSets = 0; return DRCDEC_SELECTION_PROCESS_NOT_OK; } /* sorting: Ducking/Fading -> Dependent -> Selected */ if (hSelProcOutput->numSelectedDrcSets == 3) { selectedDrcSetIds = hSelProcOutput->selectedDrcSetIds[0]; selectedDownmixIds = hSelProcOutput->selectedDownmixIds[0]; hSelProcOutput->selectedDrcSetIds[0] = hSelProcOutput->selectedDrcSetIds[2]; hSelProcOutput->selectedDownmixIds[0] = hSelProcOutput->selectedDownmixIds[2]; hSelProcOutput->selectedDrcSetIds[2] = selectedDrcSetIds; hSelProcOutput->selectedDownmixIds[2] = selectedDownmixIds; } else if (hSelProcOutput->numSelectedDrcSets == 2) { selectedDrcSetIds = hSelProcOutput->selectedDrcSetIds[0]; selectedDownmixIds = hSelProcOutput->selectedDownmixIds[0]; hSelProcOutput->selectedDrcSetIds[0] = hSelProcOutput->selectedDrcSetIds[1]; hSelProcOutput->selectedDownmixIds[0] = hSelProcOutput->selectedDownmixIds[1]; hSelProcOutput->selectedDrcSetIds[1] = selectedDrcSetIds; hSelProcOutput->selectedDownmixIds[1] = selectedDownmixIds; } return retVal; } static DRCDEC_SELECTION_PROCESS_RETURN _selectDownmixMatrix( HANDLE_SEL_PROC_OUTPUT hSelProcOutput, HANDLE_UNI_DRC_CONFIG hUniDrcConfig) { int i; hSelProcOutput->baseChannelCount = hUniDrcConfig->channelLayout.baseChannelCount; hSelProcOutput->targetChannelCount = hUniDrcConfig->channelLayout.baseChannelCount; hSelProcOutput->targetLayout = -1; hSelProcOutput->downmixMatrixPresent = 0; if (hSelProcOutput->activeDownmixId != 0) { for (i = 0; i < hUniDrcConfig->downmixInstructionsCount; i++) { DOWNMIX_INSTRUCTIONS* pDown = &(hUniDrcConfig->downmixInstructions[i]); if (pDown->targetChannelCount > 8) { continue; } if (hSelProcOutput->activeDownmixId == pDown->downmixId) { hSelProcOutput->targetChannelCount = pDown->targetChannelCount; hSelProcOutput->targetLayout = pDown->targetLayout; if (pDown->downmixCoefficientsPresent) { int j, k; FIXP_DBL downmixOffset = getDownmixOffset( pDown, hSelProcOutput->baseChannelCount); /* e = 1 */ for (j = 0; j < hSelProcOutput->baseChannelCount; j++) { for (k = 0; k < hSelProcOutput->targetChannelCount; k++) { hSelProcOutput->downmixMatrix[j][k] = fMultDiv2( downmixOffset, pDown->downmixCoefficient[j + k * hSelProcOutput ->baseChannelCount]) << 2; } } hSelProcOutput->downmixMatrixPresent = 1; } break; } } } return DRCDEC_SELECTION_PROCESS_NO_ERROR; } static DRCDEC_SELECTION_PROCESS_RETURN _drcSetPreSelection( SEL_PROC_INPUT* hSelProcInput, HANDLE_UNI_DRC_CONFIG hUniDrcConfig, HANDLE_LOUDNESS_INFO_SET hLoudnessInfoSet, DRCDEC_SELECTION** ppCandidatesPotential, DRCDEC_SELECTION** ppCandidatesSelected, SEL_PROC_CODEC_MODE codecMode) { DRCDEC_SELECTION_PROCESS_RETURN retVal = DRCDEC_SELECTION_PROCESS_NO_ERROR; int i, j; for (i = 0; i < hSelProcInput->numDownmixIdRequests; i++) { for (j = 0; j < hUniDrcConfig->drcInstructionsCountInclVirtual; j++) { DRC_INSTRUCTIONS_UNI_DRC* pDrcInstruction = &(hUniDrcConfig->drcInstructionsUniDrc[j]); /* check if ID is unique */ if (selectDrcInstructions(hUniDrcConfig, pDrcInstruction->drcSetId) != pDrcInstruction) continue; retVal = _drcSetPreSelectionSingleInstruction( hSelProcInput, i, hUniDrcConfig, hLoudnessInfoSet, pDrcInstruction, *ppCandidatesPotential, *ppCandidatesSelected, codecMode); if (retVal) return DRCDEC_SELECTION_PROCESS_NOT_OK; } } retVal = _preSelectionRequirement9(hSelProcInput, *ppCandidatesPotential, *ppCandidatesSelected); if (retVal) return DRCDEC_SELECTION_PROCESS_NOT_OK; if (_drcdec_selection_getNumber(*ppCandidatesSelected) == 0) { retVal = _drcSetSelectionAddCandidates( hSelProcInput, *ppCandidatesPotential, *ppCandidatesSelected); if (retVal) return DRCDEC_SELECTION_PROCESS_NOT_OK; } return retVal; } static DRCDEC_SELECTION_PROCESS_RETURN _drcSetRequestSelection( SEL_PROC_INPUT* hSelProcInput, HANDLE_UNI_DRC_CONFIG hUniDrcConfig, HANDLE_LOUDNESS_INFO_SET hLoudnessInfoSet, DRCDEC_SELECTION** ppCandidatesPotential, DRCDEC_SELECTION** ppCandidatesSelected) { DRCDEC_SELECTION_PROCESS_RETURN retVal; int i; if (_drcdec_selection_getNumber(*ppCandidatesPotential) == 0) { retVal = DRCDEC_SELECTION_PROCESS_NOT_OK; if (retVal) return DRCDEC_SELECTION_PROCESS_NOT_OK; } if (hSelProcInput->dynamicRangeControlOn) { if (hSelProcInput->numDrcFeatureRequests == 0) { retVal = _selectDrcSetEffectNone(hUniDrcConfig, *ppCandidatesPotential, *ppCandidatesSelected); if (retVal) return (retVal); if (_drcdec_selection_getNumber(*ppCandidatesSelected) == 0) { DRC_FEATURE_REQUEST fallbackRequest; fallbackRequest.drcEffectType.numRequests = 5; fallbackRequest.drcEffectType.numRequestsDesired = 5; fallbackRequest.drcEffectType.request[0] = DETR_GENERAL_COMPR; fallbackRequest.drcEffectType.request[1] = DETR_NIGHT; fallbackRequest.drcEffectType.request[2] = DETR_NOISY; fallbackRequest.drcEffectType.request[3] = DETR_LIMITED; fallbackRequest.drcEffectType.request[4] = DETR_LOWLEVEL; retVal = _selectEffectTypeFeature(hUniDrcConfig, fallbackRequest, ppCandidatesPotential, ppCandidatesSelected); if (retVal) return DRCDEC_SELECTION_PROCESS_NOT_OK; } _swapSelectionAndClear(ppCandidatesPotential, ppCandidatesSelected); } else { for (i = 0; i < hSelProcInput->numDrcFeatureRequests; i++) { if (hSelProcInput->drcFeatureRequestType[i] == DFRT_EFFECT_TYPE) { retVal = _selectEffectTypeFeature( hUniDrcConfig, hSelProcInput->drcFeatureRequest[i], ppCandidatesPotential, ppCandidatesSelected); _swapSelectionAndClear(ppCandidatesPotential, ppCandidatesSelected); if (retVal) return DRCDEC_SELECTION_PROCESS_NOT_OK; } else if (hSelProcInput->drcFeatureRequestType[i] == DFRT_DYNAMIC_RANGE) { retVal = _selectDynamicRange( hUniDrcConfig, hLoudnessInfoSet, hSelProcInput->drcFeatureRequest[i], hSelProcInput->downmixIdRequested, hSelProcInput->albumMode, *ppCandidatesPotential, *ppCandidatesSelected); if (_drcdec_selection_getNumber(*ppCandidatesSelected) > 0) { _swapSelectionAndClear(ppCandidatesPotential, ppCandidatesSelected); } if (retVal) return DRCDEC_SELECTION_PROCESS_NOT_OK; } else if (hSelProcInput->drcFeatureRequestType[i] == DFRT_DRC_CHARACTERISTIC) { retVal = _selectDrcCharacteristic( hUniDrcConfig, hSelProcInput->drcFeatureRequest[i].drcCharacteristic, ppCandidatesPotential, ppCandidatesSelected); if (_drcdec_selection_getNumber(*ppCandidatesSelected) > 0) { _swapSelectionAndClear(ppCandidatesPotential, ppCandidatesSelected); } if (retVal) return DRCDEC_SELECTION_PROCESS_NOT_OK; } } } } return DRCDEC_SELECTION_PROCESS_NO_ERROR; } /*******************************************/ static DRCDEC_SELECTION_PROCESS_RETURN _dynamicRangeMeasurement( HANDLE_LOUDNESS_INFO_SET hLoudnessInfoSet, DRC_INSTRUCTIONS_UNI_DRC* pInst, UCHAR downmixIdRequested, DYN_RANGE_MEASUREMENT_REQUEST_TYPE dynamicRangeMeasurementType, int albumMode, int* pPeakToAveragePresent, FIXP_DBL* pPeakToAverage) { int i; DRCDEC_SELECTION_PROCESS_RETURN retVal = DRCDEC_SELECTION_PROCESS_NO_ERROR; int drcSetId = fMax(0, pInst->drcSetId); *pPeakToAveragePresent = 0; if (albumMode) { for (i = 0; i < hLoudnessInfoSet->loudnessInfoAlbumCount; i++) { LOUDNESS_INFO* pLoudnessInfo = &(hLoudnessInfoSet->loudnessInfoAlbum[i]); if (drcSetId == pLoudnessInfo->drcSetId) { if (downmixIdRequested == pLoudnessInfo->downmixId) { retVal = _extractLoudnessPeakToAverageValue( pLoudnessInfo, dynamicRangeMeasurementType, pPeakToAveragePresent, pPeakToAverage); if (retVal) return (retVal); } } } } if (*pPeakToAveragePresent == 0) { for (i = 0; i < hLoudnessInfoSet->loudnessInfoCount; i++) { LOUDNESS_INFO* pLoudnessInfo = &(hLoudnessInfoSet->loudnessInfo[i]); if (drcSetId == pLoudnessInfo->drcSetId) { if (downmixIdRequested == pLoudnessInfo->downmixId) { retVal = _extractLoudnessPeakToAverageValue( pLoudnessInfo, dynamicRangeMeasurementType, pPeakToAveragePresent, pPeakToAverage); if (retVal) return (retVal); } } } } return retVal; } /*******************************************/ static DRCDEC_SELECTION_DATA* _drcdec_selection_addNew( DRCDEC_SELECTION* pSelection) { if (pSelection->numData < (12 + 1 + 6)) { DRCDEC_SELECTION_DATA* pData = &(pSelection->data[pSelection->numData]); FDKmemset(pData, 0, sizeof(DRCDEC_SELECTION_DATA)); pSelection->numData++; return pData; } else { return NULL; } } static DRCDEC_SELECTION_DATA* _drcdec_selection_add( DRCDEC_SELECTION* pSelection, DRCDEC_SELECTION_DATA* pDataIn) { if (pSelection->numData < (12 + 1 + 6)) { DRCDEC_SELECTION_DATA* pData = &(pSelection->data[pSelection->numData]); FDKmemcpy(pData, pDataIn, sizeof(DRCDEC_SELECTION_DATA)); pSelection->numData++; return pData; } else { return NULL; } } static int _drcdec_selection_clear(DRCDEC_SELECTION* pSelection) { return pSelection->numData = 0; } static int _drcdec_selection_getNumber(DRCDEC_SELECTION* pSelection) { return pSelection->numData; } static int _drcdec_selection_setNumber(DRCDEC_SELECTION* pSelection, int num) { if (num >= 0 && num < pSelection->numData) { return pSelection->numData = num; } else { return pSelection->numData; } } static DRCDEC_SELECTION_DATA* _drcdec_selection_getAt( DRCDEC_SELECTION* pSelection, int at) { if (at >= 0 && at < (12 + 1 + 6)) { return &(pSelection->data[at]); } else { return NULL; } } static int _swapSelectionAndClear(DRCDEC_SELECTION** ppCandidatesPotential, DRCDEC_SELECTION** ppCandidatesSelected) { DRCDEC_SELECTION* pTmp = *ppCandidatesPotential; *ppCandidatesPotential = *ppCandidatesSelected; *ppCandidatesSelected = pTmp; _drcdec_selection_clear(*ppCandidatesSelected); return 0; } static int _swapSelection(DRCDEC_SELECTION** ppCandidatesPotential, DRCDEC_SELECTION** ppCandidatesSelected) { DRCDEC_SELECTION* pTmp = *ppCandidatesPotential; *ppCandidatesPotential = *ppCandidatesSelected; *ppCandidatesSelected = pTmp; return 0; } /*******************************************/ static LOUDNESS_INFO* _getLoudnessInfoStructure( HANDLE_LOUDNESS_INFO_SET hLoudnessInfoSet, int drcSetId, int downmixId, int albumMode) { int i, j; int count; LOUDNESS_INFO* pLoudnessInfo = NULL; if (albumMode) { count = hLoudnessInfoSet->loudnessInfoAlbumCount; pLoudnessInfo = hLoudnessInfoSet->loudnessInfoAlbum; } else { count = hLoudnessInfoSet->loudnessInfoCount; pLoudnessInfo = hLoudnessInfoSet->loudnessInfo; } for (i = 0; i < count; i++) { if ((pLoudnessInfo[i].drcSetId == drcSetId) && (pLoudnessInfo[i].downmixId == downmixId)) { for (j = 0; j < pLoudnessInfo[i].measurementCount; j++) { if ((pLoudnessInfo[i].loudnessMeasurement[j].methodDefinition == 1) || (pLoudnessInfo[i].loudnessMeasurement[j].methodDefinition == 2)) { return &pLoudnessInfo[i]; } } } } return NULL; } static LOUDNESS_INFO* _getApplicableLoudnessInfoStructure( HANDLE_LOUDNESS_INFO_SET hLoudnessInfoSet, int drcSetId, int downmixIdRequested, int albumMode) { LOUDNESS_INFO* pLoudnessInfo = NULL; /* default value */ pLoudnessInfo = _getLoudnessInfoStructure(hLoudnessInfoSet, drcSetId, downmixIdRequested, albumMode); /* fallback values */ if (pLoudnessInfo == NULL) { pLoudnessInfo = _getLoudnessInfoStructure(hLoudnessInfoSet, drcSetId, 0x7F, albumMode); } if (pLoudnessInfo == NULL) { pLoudnessInfo = _getLoudnessInfoStructure(hLoudnessInfoSet, 0x3F, downmixIdRequested, albumMode); } if (pLoudnessInfo == NULL) { pLoudnessInfo = _getLoudnessInfoStructure(hLoudnessInfoSet, 0, downmixIdRequested, albumMode); } if (pLoudnessInfo == NULL) { pLoudnessInfo = _getLoudnessInfoStructure(hLoudnessInfoSet, 0x3F, 0x7F, albumMode); } if (pLoudnessInfo == NULL) { pLoudnessInfo = _getLoudnessInfoStructure(hLoudnessInfoSet, 0, 0x7F, albumMode); } if (pLoudnessInfo == NULL) { pLoudnessInfo = _getLoudnessInfoStructure(hLoudnessInfoSet, drcSetId, 0, albumMode); } if (pLoudnessInfo == NULL) { pLoudnessInfo = _getLoudnessInfoStructure(hLoudnessInfoSet, 0x3F, 0, albumMode); } if (pLoudnessInfo == NULL) { pLoudnessInfo = _getLoudnessInfoStructure(hLoudnessInfoSet, 0, 0, albumMode); } return pLoudnessInfo; } /*******************************************/ typedef struct { FIXP_DBL value; int order; } VALUE_ORDER; void _initValueOrder(VALUE_ORDER* pValue) { pValue->value = (FIXP_DBL)0; pValue->order = -1; } enum { MS_BONUS0 = 0, MS_BONUS1770, MS_BONUSUSER, MS_BONUSEXPERT, MS_RESA, MS_RESB, MS_RESC, MS_RESD, MS_RESE, MS_PROGRAMLOUDNESS, MS_PEAKLOUDNESS }; static DRCDEC_SELECTION_PROCESS_RETURN _getMethodValue( VALUE_ORDER* pValueOrder, FIXP_DBL value, int measurementSystem, int measurementSystemRequested) { const int rows = 11; const int columns = 12; const int pOrdering[rows][columns] = { {0, 0, 8, 0, 1, 3, 0, 5, 6, 7, 4, 2}, /* default = bonus1770 */ {0, 0, 8, 0, 1, 3, 0, 5, 6, 7, 4, 2}, /* bonus1770 */ {0, 0, 1, 0, 8, 5, 0, 2, 3, 4, 6, 7}, /* bonusUser */ {0, 0, 3, 0, 1, 8, 0, 4, 5, 6, 7, 2}, /* bonusExpert */ {0, 0, 5, 0, 1, 3, 0, 8, 6, 7, 4, 2}, /* ResA */ {0, 0, 5, 0, 1, 3, 0, 6, 8, 7, 4, 2}, /* ResB */ {0, 0, 5, 0, 1, 3, 0, 6, 7, 8, 4, 2}, /* ResC */ {0, 0, 3, 0, 1, 7, 0, 4, 5, 6, 8, 2}, /* ResD */ {0, 0, 1, 0, 7, 5, 0, 2, 3, 4, 6, 8}, /* ResE */ {0, 0, 1, 0, 0, 0, 0, 2, 3, 4, 0, 0}, /* ProgramLoudness */ {0, 7, 0, 0, 0, 0, 6, 5, 4, 3, 2, 1} /* PeakLoudness */ }; if (measurementSystemRequested < 0 || measurementSystemRequested >= rows || measurementSystem < 0 || measurementSystem >= columns) { return DRCDEC_SELECTION_PROCESS_NOT_OK; } if (pOrdering[measurementSystemRequested][measurementSystem] > pValueOrder->order) { pValueOrder->order = pOrdering[measurementSystemRequested][measurementSystem]; pValueOrder->value = value; } return DRCDEC_SELECTION_PROCESS_NO_ERROR; } /*******************************************/ static DRCDEC_SELECTION_PROCESS_RETURN _getLoudness( HANDLE_LOUDNESS_INFO_SET hLoudnessInfoSet, int albumMode, METHOD_DEFINITION_REQUEST measurementMethodRequested, MEASUREMENT_SYSTEM_REQUEST measurementSystemRequested, FIXP_DBL targetLoudness, /* e = 7 */ int drcSetId, int downmixIdRequested, FIXP_DBL* pLoudnessNormalizationGain, /* e = 7 */ FIXP_DBL* pLoudness) /* e = 7 */ { int index; LOUDNESS_INFO* pLoudnessInfo = NULL; VALUE_ORDER valueOrder; /* map MDR_DEFAULT to MDR_PROGRAM_LOUDNESS */ METHOD_DEFINITION_REQUEST requestedMethodDefinition = measurementMethodRequested < MDR_ANCHOR_LOUDNESS ? MDR_PROGRAM_LOUDNESS : MDR_ANCHOR_LOUDNESS; if (measurementMethodRequested > MDR_ANCHOR_LOUDNESS) { return DRCDEC_SELECTION_PROCESS_NOT_OK; } _initValueOrder(&valueOrder); *pLoudness = UNDEFINED_LOUDNESS_VALUE; *pLoudnessNormalizationGain = (FIXP_DBL)0; if (drcSetId < 0) { drcSetId = 0; } pLoudnessInfo = _getApplicableLoudnessInfoStructure( hLoudnessInfoSet, drcSetId, downmixIdRequested, albumMode); if (albumMode && (pLoudnessInfo == NULL)) { pLoudnessInfo = _getApplicableLoudnessInfoStructure( hLoudnessInfoSet, drcSetId, downmixIdRequested, 0); } if (pLoudnessInfo == NULL) { return DRCDEC_SELECTION_PROCESS_NO_ERROR; } index = -1; do { index = _findMethodDefinition(pLoudnessInfo, requestedMethodDefinition, index + 1); if (index >= 0) { _getMethodValue( &valueOrder, pLoudnessInfo->loudnessMeasurement[index].methodValue, pLoudnessInfo->loudnessMeasurement[index].measurementSystem, measurementSystemRequested); } } while (index >= 0); /* repeat with other method definition */ if (valueOrder.order == -1) { index = -1; do { index = _findMethodDefinition( pLoudnessInfo, requestedMethodDefinition == MDR_PROGRAM_LOUDNESS ? MDR_ANCHOR_LOUDNESS : MDR_PROGRAM_LOUDNESS, index + 1); if (index >= 0) { _getMethodValue( &valueOrder, pLoudnessInfo->loudnessMeasurement[index].methodValue, pLoudnessInfo->loudnessMeasurement[index].measurementSystem, measurementSystemRequested); } } while (index >= 0); } if (valueOrder.order == -1) { return DRCDEC_SELECTION_PROCESS_NOT_OK; } else { *pLoudnessNormalizationGain = targetLoudness - valueOrder.value; *pLoudness = valueOrder.value; } return DRCDEC_SELECTION_PROCESS_NO_ERROR; } /*******************************************/ static int _truePeakLevelIsPresent(HANDLE_LOUDNESS_INFO_SET hLoudnessInfoSet, int drcSetId, int downmixId, int albumMode) { int i; int count; LOUDNESS_INFO* pLoudnessInfo = NULL; if (albumMode) { count = hLoudnessInfoSet->loudnessInfoAlbumCount; pLoudnessInfo = hLoudnessInfoSet->loudnessInfoAlbum; } else { count = hLoudnessInfoSet->loudnessInfoCount; pLoudnessInfo = hLoudnessInfoSet->loudnessInfo; } for (i = 0; i < count; i++) { if ((pLoudnessInfo[i].drcSetId == drcSetId) && (pLoudnessInfo[i].downmixId == downmixId)) { if (pLoudnessInfo[i].truePeakLevelPresent) return 1; } } return 0; } static DRCDEC_SELECTION_PROCESS_RETURN _getTruePeakLevel( HANDLE_LOUDNESS_INFO_SET hLoudnessInfoSet, int drcSetId, int downmixId, int albumMode, FIXP_DBL* pTruePeakLevel) { int i; int count; LOUDNESS_INFO* pLoudnessInfo = NULL; if (albumMode) { count = hLoudnessInfoSet->loudnessInfoAlbumCount; pLoudnessInfo = hLoudnessInfoSet->loudnessInfoAlbum; } else { count = hLoudnessInfoSet->loudnessInfoCount; pLoudnessInfo = hLoudnessInfoSet->loudnessInfo; } for (i = 0; i < count; i++) { if ((pLoudnessInfo[i].drcSetId == drcSetId) && (pLoudnessInfo[i].downmixId == downmixId)) { if (pLoudnessInfo[i].truePeakLevelPresent) { *pTruePeakLevel = pLoudnessInfo[i].truePeakLevel; return DRCDEC_SELECTION_PROCESS_NO_ERROR; } } } return DRCDEC_SELECTION_PROCESS_NOT_OK; } static int _samplePeakLevelIsPresent(HANDLE_LOUDNESS_INFO_SET hLoudnessInfoSet, int drcSetId, int downmixId, int albumMode) { int i; int count; LOUDNESS_INFO* pLoudnessInfo = NULL; if (albumMode) { count = hLoudnessInfoSet->loudnessInfoAlbumCount; pLoudnessInfo = hLoudnessInfoSet->loudnessInfoAlbum; } else { count = hLoudnessInfoSet->loudnessInfoCount; pLoudnessInfo = hLoudnessInfoSet->loudnessInfo; } for (i = 0; i < count; i++) { if ((pLoudnessInfo[i].drcSetId == drcSetId) && (pLoudnessInfo[i].downmixId == downmixId)) { if (pLoudnessInfo[i].samplePeakLevelPresent) return 1; } } return 0; } static DRCDEC_SELECTION_PROCESS_RETURN _getSamplePeakLevel( HANDLE_LOUDNESS_INFO_SET hLoudnessInfoSet, int drcSetId, int downmixId, int albumMode, FIXP_DBL* pSamplePeakLevel /* e = 7 */ ) { int i; int count; LOUDNESS_INFO* pLoudnessInfo = NULL; if (albumMode) { count = hLoudnessInfoSet->loudnessInfoAlbumCount; pLoudnessInfo = hLoudnessInfoSet->loudnessInfoAlbum; } else { count = hLoudnessInfoSet->loudnessInfoCount; pLoudnessInfo = hLoudnessInfoSet->loudnessInfo; } for (i = 0; i < count; i++) { if ((pLoudnessInfo[i].drcSetId == drcSetId) && (pLoudnessInfo[i].downmixId == downmixId)) { if (pLoudnessInfo[i].samplePeakLevelPresent) { *pSamplePeakLevel = pLoudnessInfo[i].samplePeakLevel; return DRCDEC_SELECTION_PROCESS_NO_ERROR; } } } return DRCDEC_SELECTION_PROCESS_NOT_OK; } static int _limiterPeakTargetIsPresent( DRC_INSTRUCTIONS_UNI_DRC* pDrcInstruction, int drcSetId, int downmixId) { int i; if (pDrcInstruction->limiterPeakTargetPresent) { if ((pDrcInstruction->downmixId[0] == downmixId) || (pDrcInstruction->downmixId[0] == 0x7F)) { return 1; } for (i = 0; i < pDrcInstruction->downmixIdCount; i++) { if (pDrcInstruction->downmixId[i] == downmixId) { return 1; } } } return 0; } static DRCDEC_SELECTION_PROCESS_RETURN _getLimiterPeakTarget( DRC_INSTRUCTIONS_UNI_DRC* pDrcInstruction, int drcSetId, int downmixId, FIXP_DBL* pLimiterPeakTarget) { int i; if (pDrcInstruction->limiterPeakTargetPresent) { if ((pDrcInstruction->downmixId[0] == downmixId) || (pDrcInstruction->downmixId[0] == 0x7F)) { *pLimiterPeakTarget = ((FX_SGL2FX_DBL(pDrcInstruction->limiterPeakTarget) >> 2)); return DRCDEC_SELECTION_PROCESS_NO_ERROR; } for (i = 0; i < pDrcInstruction->downmixIdCount; i++) { if (pDrcInstruction->downmixId[i] == downmixId) { *pLimiterPeakTarget = ((FX_SGL2FX_DBL(pDrcInstruction->limiterPeakTarget) >> 2)); return DRCDEC_SELECTION_PROCESS_NO_ERROR; } } } return DRCDEC_SELECTION_PROCESS_NOT_OK; } static int _downmixCoefficientsArePresent(HANDLE_UNI_DRC_CONFIG hUniDrcConfig, int downmixId, int* pIndex) { int i; *pIndex = -1; for (i = 0; i < hUniDrcConfig->downmixInstructionsCount; i++) { if (hUniDrcConfig->downmixInstructions[i].downmixId == downmixId) { if (hUniDrcConfig->downmixInstructions[i].downmixCoefficientsPresent) { if (hUniDrcConfig->downmixInstructions[i].targetChannelCount > 8) return 0; *pIndex = i; return 1; } } } return 0; } static DRCDEC_SELECTION_PROCESS_RETURN _getSignalPeakLevel( HANDLE_SEL_PROC_INPUT hSelProcInput, HANDLE_UNI_DRC_CONFIG hUniDrcConfig, HANDLE_LOUDNESS_INFO_SET hLoudnessInfoSet, DRC_INSTRUCTIONS_UNI_DRC* pInst, int downmixIdRequested, int* explicitPeakInformationPresent, FIXP_DBL* signalPeakLevelOut, /* e = 7 */ SEL_PROC_CODEC_MODE codecMode ) { DRCDEC_SELECTION_PROCESS_RETURN retVal = DRCDEC_SELECTION_PROCESS_NO_ERROR; int albumMode = hSelProcInput->albumMode; FIXP_DBL signalPeakLevelTmp = (FIXP_DBL)0; FIXP_DBL signalPeakLevel = FIXP_DBL(0); int dmxId = downmixIdRequested; int drcSetId = pInst->drcSetId; if (drcSetId < 0) { drcSetId = 0; } *explicitPeakInformationPresent = 1; if (_truePeakLevelIsPresent(hLoudnessInfoSet, drcSetId, dmxId, albumMode)) { retVal = _getTruePeakLevel(hLoudnessInfoSet, drcSetId, dmxId, albumMode, &signalPeakLevel); if (retVal) return (retVal); } else if (_samplePeakLevelIsPresent(hLoudnessInfoSet, drcSetId, dmxId, albumMode)) { retVal = _getSamplePeakLevel(hLoudnessInfoSet, drcSetId, dmxId, albumMode, &signalPeakLevel); if (retVal) return (retVal); } else if (_truePeakLevelIsPresent(hLoudnessInfoSet, 0x3F, dmxId, albumMode)) { retVal = _getTruePeakLevel(hLoudnessInfoSet, 0x3F, dmxId, albumMode, &signalPeakLevel); if (retVal) return (retVal); } else if (_samplePeakLevelIsPresent(hLoudnessInfoSet, 0x3F, dmxId, albumMode)) { retVal = _getSamplePeakLevel(hLoudnessInfoSet, 0x3F, dmxId, albumMode, &signalPeakLevel); if (retVal) return (retVal); } else if (_limiterPeakTargetIsPresent(pInst, drcSetId, dmxId)) { retVal = _getLimiterPeakTarget(pInst, drcSetId, dmxId, &signalPeakLevel); if (retVal) return (retVal); } else if (dmxId != 0) { int downmixInstructionIndex = 0; FIXP_DBL downmixPeakLevelDB = 0; *explicitPeakInformationPresent = 0; signalPeakLevelTmp = FIXP_DBL(0); if (_downmixCoefficientsArePresent(hUniDrcConfig, dmxId, &downmixInstructionIndex)) { FIXP_DBL dB_m; int dB_e; FIXP_DBL coeff; FIXP_DBL sum, maxSum; /* e = 7, so it is possible to sum up up to 32 downmix coefficients (with e = 2) */ int i, j; DOWNMIX_INSTRUCTIONS* pDown = &(hUniDrcConfig->downmixInstructions[downmixInstructionIndex]); FIXP_DBL downmixOffset = getDownmixOffset( pDown, hUniDrcConfig->channelLayout.baseChannelCount); /* e = 1 */ maxSum = (FIXP_DBL)0; for (i = 0; i < pDown->targetChannelCount; i++) { sum = (FIXP_DBL)0; for (j = 0; j < hUniDrcConfig->channelLayout.baseChannelCount; j++) { coeff = pDown->downmixCoefficient[j + i * hUniDrcConfig->channelLayout .baseChannelCount]; sum += coeff >> 5; } if (maxSum < sum) maxSum = sum; } maxSum = fMultDiv2(maxSum, downmixOffset) << 2; if (maxSum == FL2FXCONST_DBL(1.0f / (float)(1 << 7))) { downmixPeakLevelDB = (FIXP_DBL)0; } else { dB_m = lin2dB(maxSum, 7, &dB_e); /* e_maxSum = 7 */ downmixPeakLevelDB = scaleValue(dB_m, dB_e - 7); /* e_downmixPeakLevelDB = 7 */ } } if (_truePeakLevelIsPresent(hLoudnessInfoSet, drcSetId, 0, albumMode)) { retVal = _getTruePeakLevel(hLoudnessInfoSet, drcSetId, 0, albumMode, &signalPeakLevelTmp); if (retVal) return (retVal); } else if (_samplePeakLevelIsPresent(hLoudnessInfoSet, drcSetId, 0, albumMode)) { retVal = _getSamplePeakLevel(hLoudnessInfoSet, drcSetId, 0, albumMode, &signalPeakLevelTmp); if (retVal) return (retVal); } else if (_truePeakLevelIsPresent(hLoudnessInfoSet, 0x3F, 0, albumMode)) { retVal = _getTruePeakLevel(hLoudnessInfoSet, 0x3F, 0, albumMode, &signalPeakLevelTmp); if (retVal) return (retVal); } else if (_samplePeakLevelIsPresent(hLoudnessInfoSet, 0x3F, 0, albumMode)) { retVal = _getSamplePeakLevel(hLoudnessInfoSet, 0x3F, 0, albumMode, &signalPeakLevelTmp); if (retVal) return (retVal); } else if (_limiterPeakTargetIsPresent(pInst, drcSetId, 0)) { retVal = _getLimiterPeakTarget(pInst, drcSetId, 0, &signalPeakLevelTmp); if (retVal) return (retVal); } signalPeakLevel = signalPeakLevelTmp + downmixPeakLevelDB; } else { signalPeakLevel = FIXP_DBL(0); /* worst case estimate */ *explicitPeakInformationPresent = FIXP_DBL(0); } *signalPeakLevelOut = signalPeakLevel; return retVal; } static DRCDEC_SELECTION_PROCESS_RETURN _extractLoudnessPeakToAverageValue( LOUDNESS_INFO* loudnessInfo, DYN_RANGE_MEASUREMENT_REQUEST_TYPE dynamicRangeMeasurementType, int* pLoudnessPeakToAverageValuePresent, FIXP_DBL* pLoudnessPeakToAverageValue) { int i; VALUE_ORDER valueOrderLoudness; VALUE_ORDER valueOrderPeakLoudness; _initValueOrder(&valueOrderLoudness); _initValueOrder(&valueOrderPeakLoudness); LOUDNESS_MEASUREMENT* pLoudnessMeasure = NULL; *pLoudnessPeakToAverageValuePresent = 0; for (i = 0; i < loudnessInfo->measurementCount; i++) { pLoudnessMeasure = &(loudnessInfo->loudnessMeasurement[i]); if (pLoudnessMeasure->methodDefinition == MD_PROGRAM_LOUDNESS) { _getMethodValue(&valueOrderLoudness, pLoudnessMeasure->methodValue, pLoudnessMeasure->measurementSystem, MS_PROGRAMLOUDNESS); } if ((dynamicRangeMeasurementType == DRMRT_SHORT_TERM_LOUDNESS_TO_AVG) && (pLoudnessMeasure->methodDefinition == MD_SHORT_TERM_LOUDNESS_MAX)) { _getMethodValue(&valueOrderPeakLoudness, pLoudnessMeasure->methodValue, pLoudnessMeasure->measurementSystem, MS_PEAKLOUDNESS); } if ((dynamicRangeMeasurementType == DRMRT_MOMENTARY_LOUDNESS_TO_AVG) && (pLoudnessMeasure->methodDefinition == MD_MOMENTARY_LOUDNESS_MAX)) { _getMethodValue(&valueOrderPeakLoudness, pLoudnessMeasure->methodValue, pLoudnessMeasure->measurementSystem, MS_PEAKLOUDNESS); } if ((dynamicRangeMeasurementType == DRMRT_TOP_OF_LOUDNESS_RANGE_TO_AVG) && (pLoudnessMeasure->methodDefinition == MD_MAX_OF_LOUDNESS_RANGE)) { _getMethodValue(&valueOrderPeakLoudness, pLoudnessMeasure->methodValue, pLoudnessMeasure->measurementSystem, MS_PEAKLOUDNESS); } } if ((valueOrderLoudness.order > -1) && (valueOrderPeakLoudness.order > -1)) { *pLoudnessPeakToAverageValue = valueOrderPeakLoudness.value - valueOrderLoudness.value; *pLoudnessPeakToAverageValuePresent = 1; } return DRCDEC_SELECTION_PROCESS_NO_ERROR; } /*******************************************/ static DRCDEC_SELECTION_PROCESS_RETURN _selectAlbumLoudness( HANDLE_LOUDNESS_INFO_SET hLoudnessInfoSet, DRCDEC_SELECTION* pCandidatesPotential, DRCDEC_SELECTION* pCandidatesSelected) { int i, j; for (i = 0; i < _drcdec_selection_getNumber(pCandidatesPotential); i++) { DRCDEC_SELECTION_DATA* pCandidate = _drcdec_selection_getAt(pCandidatesPotential, i); if (pCandidate == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; for (j = 0; j < hLoudnessInfoSet->loudnessInfoAlbumCount; j++) { if (pCandidate->pInst->drcSetId == hLoudnessInfoSet->loudnessInfoAlbum[j].drcSetId) { if (_drcdec_selection_add(pCandidatesSelected, pCandidate) == NULL) return DRCDEC_SELECTION_PROCESS_NOT_OK; } } } return DRCDEC_SELECTION_PROCESS_NO_ERROR; } /*******************************************/ static int _findMethodDefinition(LOUDNESS_INFO* pLoudnessInfo, int methodDefinition, int startIndex) { int i; int index = -1; for (i = startIndex; i < pLoudnessInfo->measurementCount; i++) { if (pLoudnessInfo->loudnessMeasurement[i].methodDefinition == methodDefinition) { index = i; break; } } return index; } /*******************************************/ static DRCDEC_SELECTION_PROCESS_RETURN _getMixingLevel( HANDLE_LOUDNESS_INFO_SET hLoudnessInfoSet, int downmixIdRequested, int drcSetIdRequested, int albumMode, FIXP_DBL* pMixingLevel) { const FIXP_DBL mixingLevelDefault = FL2FXCONST_DBL(85.0f / (float)(1 << 7)); int i; int count; LOUDNESS_INFO* pLoudnessInfo = NULL; *pMixingLevel = mixingLevelDefault; if (drcSetIdRequested < 0) { drcSetIdRequested = 0; } if (albumMode) { count = hLoudnessInfoSet->loudnessInfoAlbumCount; pLoudnessInfo = hLoudnessInfoSet->loudnessInfoAlbum; } else { count = hLoudnessInfoSet->loudnessInfoCount; pLoudnessInfo = hLoudnessInfoSet->loudnessInfo; } for (i = 0; i < count; i++) { if ((drcSetIdRequested == pLoudnessInfo[i].drcSetId) && ((downmixIdRequested == pLoudnessInfo[i].downmixId) || (DOWNMIX_ID_ANY_DOWNMIX == pLoudnessInfo[i].downmixId))) { int index = _findMethodDefinition(&pLoudnessInfo[i], MD_MIXING_LEVEL, 0); if (index >= 0) { *pMixingLevel = pLoudnessInfo[i].loudnessMeasurement[index].methodValue; break; } } } return DRCDEC_SELECTION_PROCESS_NO_ERROR; } /*******************************************/ fdk-aac-2.0.2/libDRCdec/src/drcDec_selectionProcess.h000066400000000000000000000207011404225310700223020ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /************************* MPEG-D DRC decoder library ************************** Author(s): Andreas Hoelzer Description: DRC Set Selection *******************************************************************************/ #ifndef DRCDEC_SELECTIONPROCESS_H #define DRCDEC_SELECTIONPROCESS_H #include "drcDec_types.h" #include "drcDecoder.h" /* DRC set selection according to section 6.2 of ISO/IEC 23003-4 (MPEG-D DRC) */ /* including ISO/IEC 23003-4/AMD1 (Amendment 1) */ typedef struct s_drcdec_selection_process* HANDLE_DRC_SELECTION_PROCESS; #define UNDEFINED_LOUDNESS_VALUE (FIXP_DBL)(MAXVAL_DBL - 1) typedef enum { DRCDEC_SELECTION_PROCESS_NO_ERROR = 0, DRCDEC_SELECTION_PROCESS_WARNING = -1000, DRCDEC_SELECTION_PROCESS_NOT_OK = -2000, DRCDEC_SELECTION_PROCESS_OUTOFMEMORY, DRCDEC_SELECTION_PROCESS_INVALID_HANDLE, DRCDEC_SELECTION_PROCESS_NOT_SUPPORTED, DRCDEC_SELECTION_PROCESS_INVALID_PARAM, DRCDEC_SELECTION_PROCESS_PARAM_OUT_OF_RANGE } DRCDEC_SELECTION_PROCESS_RETURN; typedef enum { SEL_PROC_TEST_TIME_DOMAIN = -100, SEL_PROC_TEST_QMF_DOMAIN, SEL_PROC_TEST_STFT_DOMAIN, SEL_PROC_CODEC_MODE_UNDEFINED = -1, SEL_PROC_MPEG_4_AAC, SEL_PROC_MPEG_D_USAC, SEL_PROC_MPEG_H_3DA } SEL_PROC_CODEC_MODE; typedef enum { /* set and get user param */ SEL_PROC_LOUDNESS_NORMALIZATION_ON, /* get only user param */ SEL_PROC_DYNAMIC_RANGE_CONTROL_ON, /* set only user params */ SEL_PROC_TARGET_LOUDNESS, SEL_PROC_EFFECT_TYPE, SEL_PROC_EFFECT_TYPE_FALLBACK_CODE, SEL_PROC_LOUDNESS_MEASUREMENT_METHOD, SEL_PROC_ALBUM_MODE, SEL_PROC_DOWNMIX_ID, SEL_PROC_TARGET_LAYOUT, SEL_PROC_TARGET_CHANNEL_COUNT, SEL_PROC_BASE_CHANNEL_COUNT, SEL_PROC_SAMPLE_RATE, SEL_PROC_BOOST, SEL_PROC_COMPRESS } SEL_PROC_USER_PARAM; typedef struct s_selection_process_output { FIXP_DBL outputPeakLevelDb; /* e = 7 */ FIXP_DBL loudnessNormalizationGainDb; /* e = 7 */ FIXP_DBL outputLoudness; /* e = 7 */ UCHAR numSelectedDrcSets; SCHAR selectedDrcSetIds[MAX_ACTIVE_DRCS]; UCHAR selectedDownmixIds[MAX_ACTIVE_DRCS]; UCHAR activeDownmixId; UCHAR baseChannelCount; UCHAR targetChannelCount; SCHAR targetLayout; UCHAR downmixMatrixPresent; FIXP_DBL downmixMatrix[8][8]; /* e = 2 */ FIXP_SGL boost; /* e = 1 */ FIXP_SGL compress; /* e = 1 */ FIXP_DBL mixingLevel; /* e = 7 */ } SEL_PROC_OUTPUT, *HANDLE_SEL_PROC_OUTPUT; DRCDEC_SELECTION_PROCESS_RETURN drcDec_SelectionProcess_Create(HANDLE_DRC_SELECTION_PROCESS* phInstance); DRCDEC_SELECTION_PROCESS_RETURN drcDec_SelectionProcess_Delete(HANDLE_DRC_SELECTION_PROCESS* phInstance); DRCDEC_SELECTION_PROCESS_RETURN drcDec_SelectionProcess_Init(HANDLE_DRC_SELECTION_PROCESS hInstance); DRCDEC_SELECTION_PROCESS_RETURN drcDec_SelectionProcess_SetCodecMode(HANDLE_DRC_SELECTION_PROCESS hInstance, const SEL_PROC_CODEC_MODE codecMode); DRCDEC_SELECTION_PROCESS_RETURN drcDec_SelectionProcess_SetParam(HANDLE_DRC_SELECTION_PROCESS hInstance, const SEL_PROC_USER_PARAM requestType, FIXP_DBL requestValue, int* pDiff); FIXP_DBL drcDec_SelectionProcess_GetParam(HANDLE_DRC_SELECTION_PROCESS hInstance, const SEL_PROC_USER_PARAM requestType); DRCDEC_SELECTION_PROCESS_RETURN drcDec_SelectionProcess_SetMpeghParams( HANDLE_DRC_SELECTION_PROCESS hInstance, const int numGroupIdsRequested, const int* groupIdRequested, const int numGroupPresetIdsRequested, const int* groupPresetIdRequested, const int* numMembersGroupPresetIdsRequested, const int groupPresetIdRequestedPreference, int* pDiff); DRCDEC_SELECTION_PROCESS_RETURN drcDec_SelectionProcess_Process(HANDLE_DRC_SELECTION_PROCESS hInstance, HANDLE_UNI_DRC_CONFIG hUniDrcConfig, HANDLE_LOUDNESS_INFO_SET hLoudnessInfoSet, HANDLE_SEL_PROC_OUTPUT hSelProcOutput); #endif fdk-aac-2.0.2/libDRCdec/src/drcDec_tools.cpp000066400000000000000000000310241404225310700204510ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /************************* MPEG-D DRC decoder library ************************** Author(s): Description: *******************************************************************************/ #include "drcDec_types.h" #include "drcDec_tools.h" #include "fixpoint_math.h" #include "drcDecoder.h" int getDeltaTmin(const int sampleRate) { /* half_ms = round (0.0005 * sampleRate); */ int half_ms = (sampleRate + 1000) / 2000; int deltaTmin = 1; if (sampleRate < 1000) { return DE_NOT_OK; } while (deltaTmin <= half_ms) { deltaTmin = deltaTmin << 1; } return deltaTmin; } DRC_COEFFICIENTS_UNI_DRC* selectDrcCoefficients( HANDLE_UNI_DRC_CONFIG hUniDrcConfig, const int location) { int n; int c = -1; for (n = 0; n < hUniDrcConfig->drcCoefficientsUniDrcCount; n++) { if (hUniDrcConfig->drcCoefficientsUniDrc[n].drcLocation == location) { c = n; } } if (c >= 0) { return &(hUniDrcConfig->drcCoefficientsUniDrc[c]); } return NULL; /* possible during bitstream parsing */ } DRC_INSTRUCTIONS_UNI_DRC* selectDrcInstructions( HANDLE_UNI_DRC_CONFIG hUniDrcConfig, const int drcSetId) { int i; for (i = 0; i < hUniDrcConfig->drcInstructionsCountInclVirtual; i++) { if (hUniDrcConfig->drcInstructionsUniDrc[i].drcSetId == drcSetId) { return &(hUniDrcConfig->drcInstructionsUniDrc[i]); } } return NULL; } DOWNMIX_INSTRUCTIONS* selectDownmixInstructions( HANDLE_UNI_DRC_CONFIG hUniDrcConfig, const int downmixId) { int i; for (i = 0; i < hUniDrcConfig->downmixInstructionsCount; i++) { if (hUniDrcConfig->downmixInstructions[i].downmixId == downmixId) { return &(hUniDrcConfig->downmixInstructions[i]); } } return NULL; } DRC_ERROR deriveDrcChannelGroups( const int drcSetEffect, /* in */ const int channelCount, /* in */ const SCHAR* gainSetIndex, /* in */ const DUCKING_MODIFICATION* duckingModificationForChannel, /* in */ UCHAR* nDrcChannelGroups, /* out */ SCHAR* uniqueIndex, /* out (gainSetIndexForChannelGroup) */ SCHAR* groupForChannel, /* out */ DUCKING_MODIFICATION* duckingModificationForChannelGroup) /* out */ { int duckingSequence = -1; int c, n, g, match, idx; FIXP_SGL factor; FIXP_SGL uniqueScaling[8]; for (g = 0; g < 8; g++) { uniqueIndex[g] = -10; uniqueScaling[g] = FIXP_SGL(-1.0f); } g = 0; if (drcSetEffect & EB_DUCK_OTHER) { for (c = 0; c < channelCount; c++) { match = 0; if (c >= 8) return DE_MEMORY_ERROR; idx = gainSetIndex[c]; factor = duckingModificationForChannel[c].duckingScaling; if (idx < 0) { for (n = 0; n < g; n++) { if (uniqueScaling[n] == factor) { match = 1; groupForChannel[c] = n; break; } } if (match == 0) { if (g >= 8) return DE_MEMORY_ERROR; uniqueIndex[g] = idx; uniqueScaling[g] = factor; groupForChannel[c] = g; g++; } } else { if ((duckingSequence > 0) && (duckingSequence != idx)) { return DE_NOT_OK; } duckingSequence = idx; groupForChannel[c] = -1; } } if (duckingSequence == -1) { return DE_NOT_OK; } } else if (drcSetEffect & EB_DUCK_SELF) { for (c = 0; c < channelCount; c++) { match = 0; if (c >= 8) return DE_MEMORY_ERROR; idx = gainSetIndex[c]; factor = duckingModificationForChannel[c].duckingScaling; if (idx >= 0) { for (n = 0; n < g; n++) { if ((uniqueIndex[n] == idx) && (uniqueScaling[n] == factor)) { match = 1; groupForChannel[c] = n; break; } } if (match == 0) { if (g >= 8) return DE_MEMORY_ERROR; uniqueIndex[g] = idx; uniqueScaling[g] = factor; groupForChannel[c] = g; g++; } } else { groupForChannel[c] = -1; } } } else { /* no ducking */ for (c = 0; c < channelCount; c++) { if (c >= 8) return DE_MEMORY_ERROR; idx = gainSetIndex[c]; match = 0; if (idx >= 0) { for (n = 0; n < g; n++) { if (uniqueIndex[n] == idx) { match = 1; groupForChannel[c] = n; break; } } if (match == 0) { if (g >= 8) return DE_MEMORY_ERROR; uniqueIndex[g] = idx; groupForChannel[c] = g; g++; } } else { groupForChannel[c] = -1; } } } *nDrcChannelGroups = g; if (drcSetEffect & (EB_DUCK_OTHER | EB_DUCK_SELF)) { for (g = 0; g < *nDrcChannelGroups; g++) { if (drcSetEffect & EB_DUCK_OTHER) { uniqueIndex[g] = duckingSequence; } duckingModificationForChannelGroup[g].duckingScaling = uniqueScaling[g]; if (uniqueScaling[g] != FL2FXCONST_SGL(1.0f / (float)(1 << 2))) { duckingModificationForChannelGroup[g].duckingScalingPresent = 1; } else { duckingModificationForChannelGroup[g].duckingScalingPresent = 0; } } } return DE_OK; } FIXP_DBL dB2lin(const FIXP_DBL dB_m, const int dB_e, int* pLin_e) { /* get linear value from dB. return lin_val = 10^(dB_val/20) = 2^(log2(10)/20*dB_val) with dB_val = dB_m *2^dB_e and lin_val = lin_m * 2^lin_e */ FIXP_DBL lin_m = f2Pow(fMult(dB_m, FL2FXCONST_DBL(0.1660964f * (float)(1 << 2))), dB_e - 2, pLin_e); return lin_m; } FIXP_DBL lin2dB(const FIXP_DBL lin_m, const int lin_e, int* pDb_e) { /* get dB value from linear value. return dB_val = 20*log10(lin_val) with dB_val = dB_m *2^dB_e and lin_val = lin_m * 2^lin_e */ FIXP_DBL dB_m; if (lin_m == (FIXP_DBL)0) { /* return very small value representing -inf */ dB_m = (FIXP_DBL)MINVAL_DBL; *pDb_e = DFRACT_BITS - 1; } else { /* 20*log10(lin_val) = 20/log2(10)*log2(lin_val) */ dB_m = fMultDiv2(FL2FXCONST_DBL(6.02059991f / (float)(1 << 3)), fLog2(lin_m, lin_e, pDb_e)); *pDb_e += 3 + 1; } return dB_m; } FIXP_DBL approxDb2lin(const FIXP_DBL dB_m, const int dB_e, int* pLin_e) { /* get linear value from approximate dB. return lin_val = 2^(dB_val/6) with dB_val = dB_m *2^dB_e and lin_val = lin_m * 2^lin_e */ FIXP_DBL lin_m = f2Pow(fMult(dB_m, FL2FXCONST_DBL(0.1666667f * (float)(1 << 2))), dB_e - 2, pLin_e); return lin_m; } int bitstreamContainsMultibandDrc(HANDLE_UNI_DRC_CONFIG hUniDrcConfig, const int downmixId) { int i, g, d, seq; DRC_INSTRUCTIONS_UNI_DRC* pInst; DRC_COEFFICIENTS_UNI_DRC* pCoef = NULL; int isMultiband = 0; pCoef = selectDrcCoefficients(hUniDrcConfig, LOCATION_SELECTED); if (pCoef == NULL) return 0; for (i = 0; i < hUniDrcConfig->drcInstructionsUniDrcCount; i++) { pInst = &(hUniDrcConfig->drcInstructionsUniDrc[i]); for (d = 0; d < pInst->downmixIdCount; d++) { if (downmixId == pInst->downmixId[d]) { for (g = 0; g < pInst->nDrcChannelGroups; g++) { seq = pInst->gainSetIndexForChannelGroup[g]; if (pCoef->gainSet[seq].bandCount > 1) { isMultiband = 1; } } } } } return isMultiband; } FIXP_DBL getDownmixOffset(DOWNMIX_INSTRUCTIONS* pDown, int baseChannelCount) { FIXP_DBL downmixOffset = FL2FXCONST_DBL(1.0f / (1 << 1)); /* e = 1 */ if ((pDown->bsDownmixOffset == 1) || (pDown->bsDownmixOffset == 2)) { int e_a, e_downmixOffset; FIXP_DBL a, q; if (baseChannelCount <= pDown->targetChannelCount) return downmixOffset; q = fDivNorm((FIXP_DBL)pDown->targetChannelCount, (FIXP_DBL)baseChannelCount); /* e = 0 */ a = lin2dB(q, 0, &e_a); if (pDown->bsDownmixOffset == 2) { e_a += 1; /* a *= 2 */ } /* a = 0.5 * round (a) */ a = fixp_round(a, e_a) >> 1; downmixOffset = dB2lin(a, e_a, &e_downmixOffset); downmixOffset = scaleValue(downmixOffset, e_downmixOffset - 1); } return downmixOffset; } fdk-aac-2.0.2/libDRCdec/src/drcDec_tools.h000066400000000000000000000142261404225310700201230ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /************************* MPEG-D DRC decoder library ************************** Author(s): Description: *******************************************************************************/ #ifndef DRCDEC_TOOLS_H #define DRCDEC_TOOLS_H #include "drcDec_types.h" #include "drcDec_selectionProcess.h" int getDeltaTmin(const int sampleRate); DRC_COEFFICIENTS_UNI_DRC* selectDrcCoefficients( HANDLE_UNI_DRC_CONFIG hUniDrcConfig, const int location); DRC_INSTRUCTIONS_UNI_DRC* selectDrcInstructions( HANDLE_UNI_DRC_CONFIG hUniDrcConfig, const int drcSetId); DOWNMIX_INSTRUCTIONS* selectDownmixInstructions( HANDLE_UNI_DRC_CONFIG hUniDrcConfig, const int downmixId); DRC_ERROR deriveDrcChannelGroups( const int drcSetEffect, /* in */ const int channelCount, /* in */ const SCHAR* gainSetIndex, /* in */ const DUCKING_MODIFICATION* duckingModificationForChannel, /* in */ UCHAR* nDrcChannelGroups, /* out */ SCHAR* uniqueIndex, /* out (gainSetIndexForChannelGroup) */ SCHAR* groupForChannel, /* out */ DUCKING_MODIFICATION* duckingModificationForChannelGroup); /* out */ FIXP_DBL dB2lin(const FIXP_DBL dB_m, const int dB_e, int* pLin_e); FIXP_DBL lin2dB(const FIXP_DBL lin_m, const int lin_e, int* pDb_e); FIXP_DBL approxDb2lin(const FIXP_DBL dB_m, const int dB_e, int* pLin_e); int bitstreamContainsMultibandDrc(HANDLE_UNI_DRC_CONFIG hUniDrcConfig, const int downmixId); FIXP_DBL getDownmixOffset(DOWNMIX_INSTRUCTIONS* pDown, int baseChannelCount); #endif fdk-aac-2.0.2/libDRCdec/src/drcDec_types.h000066400000000000000000000312751404225310700201320ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /************************* MPEG-D DRC decoder library ************************** Author(s): Description: *******************************************************************************/ #ifndef DRCDEC_TYPES_H #define DRCDEC_TYPES_H #include "common_fix.h" /* Data structures corresponding to static and dynamic DRC/Loudness payload as defined in section 7 of MPEG-D DRC standard, ISO/IEC 23003-4 */ /**************/ /* uniDrcGain */ /**************/ typedef struct { FIXP_SGL gainDb; /* e = 7 */ SHORT time; } GAIN_NODE; /* uniDrcGainExtension() (Table 56) */ typedef struct { UCHAR uniDrcGainExtType[8]; ULONG extBitSize[8 - 1]; } UNI_DRC_GAIN_EXTENSION; /* uniDrcGain() (Table 55) */ typedef struct { UCHAR nNodes[12]; /* unsaturated value, i.e. as provided in bitstream */ GAIN_NODE gainNode[12][16]; UCHAR uniDrcGainExtPresent; UNI_DRC_GAIN_EXTENSION uniDrcGainExtension; /* derived data */ UCHAR status; } UNI_DRC_GAIN, *HANDLE_UNI_DRC_GAIN; /****************/ /* uniDrcConfig */ /****************/ typedef enum { EB_NIGHT = 0x0001, EB_NOISY = 0x0002, EB_LIMITED = 0x0004, EB_LOWLEVEL = 0x0008, EB_DIALOG = 0x0010, EB_GENERAL_COMPR = 0x0020, EB_EXPAND = 0x0040, EB_ARTISTIC = 0x0080, EB_CLIPPING = 0x0100, EB_FADE = 0x0200, EB_DUCK_OTHER = 0x0400, EB_DUCK_SELF = 0x0800 } EFFECT_BIT; typedef enum { GCP_REGULAR = 0, GCP_FADING = 1, GCP_CLIPPING_DUCKING = 2, GCP_CONSTANT = 3 } GAIN_CODING_PROFILE; typedef enum { GIT_SPLINE = 0, GIT_LINEAR = 1 } GAIN_INTERPOLATION_TYPE; typedef enum { CS_LEFT = 0, CS_RIGHT = 1 } CHARACTERISTIC_SIDE; typedef enum { CF_SIGMOID = 0, CF_NODES = 1 } CHARACTERISTIC_FORMAT; typedef enum { GF_QMF32 = 0x1, GF_QMFHYBRID39 = 0x2, GF_QMF64 = 0x3, GF_QMFHYBRID71 = 0x4, GF_QMF128 = 0x5, GF_QMFHYBRID135 = 0x6, GF_UNIFORM = 0x7 } EQ_SUBBAND_GAIN_FORMAT; typedef struct { UCHAR duckingScalingPresent; FIXP_SGL duckingScaling; /* e = 2 */ } DUCKING_MODIFICATION; typedef struct { UCHAR targetCharacteristicLeftPresent; UCHAR targetCharacteristicLeftIndex; UCHAR targetCharacteristicRightPresent; UCHAR targetCharacteristicRightIndex; UCHAR gainScalingPresent; FIXP_SGL attenuationScaling; /* e = 2 */ FIXP_SGL amplificationScaling; /* e = 2 */ UCHAR gainOffsetPresent; FIXP_SGL gainOffset; /* e = 4 */ } GAIN_MODIFICATION; typedef union { UCHAR crossoverFreqIndex; USHORT startSubBandIndex; } BAND_BORDER; typedef struct { UCHAR left; UCHAR right; } CUSTOM_INDEX; typedef struct { UCHAR present; UCHAR isCICP; union { UCHAR cicpIndex; CUSTOM_INDEX custom; }; } DRC_CHARACTERISTIC; typedef struct { UCHAR gainCodingProfile; UCHAR gainInterpolationType; UCHAR fullFrame; UCHAR timeAlignment; UCHAR timeDeltaMinPresent; USHORT timeDeltaMin; UCHAR bandCount; UCHAR drcBandType; UCHAR gainSequenceIndex[4]; DRC_CHARACTERISTIC drcCharacteristic[4]; BAND_BORDER bandBorder[4]; } GAIN_SET; typedef struct { FIXP_SGL gain; /* e = 6 */ FIXP_SGL ioRatio; /* e = 2 */ FIXP_SGL exp; /* e = 5 */ UCHAR flipSign; } CUSTOM_DRC_CHAR_SIGMOID; typedef struct { UCHAR characteristicNodeCount; FIXP_SGL nodeLevel[4 + 1]; /* e = 7 */ FIXP_SGL nodeGain[4 + 1]; /* e = 7 */ } CUSTOM_DRC_CHAR_NODES; typedef shouldBeUnion { CUSTOM_DRC_CHAR_SIGMOID sigmoid; CUSTOM_DRC_CHAR_NODES nodes; } CUSTOM_DRC_CHAR; /* drcCoefficientsUniDrc() (Table 67) */ typedef struct { UCHAR drcLocation; UCHAR drcFrameSizePresent; USHORT drcFrameSize; UCHAR characteristicLeftCount; UCHAR characteristicLeftFormat[16]; CUSTOM_DRC_CHAR customCharacteristicLeft[16]; UCHAR characteristicRightCount; UCHAR characteristicRightFormat[16]; CUSTOM_DRC_CHAR customCharacteristicRight[16]; UCHAR gainSequenceCount; /* unsaturated value, i.e. as provided in bitstream */ UCHAR gainSetCount; /* saturated to 12 */ GAIN_SET gainSet[12]; /* derived data */ UCHAR gainSetIndexForGainSequence[12]; } DRC_COEFFICIENTS_UNI_DRC; /* drcInstructionsUniDrc() (Table 72) */ typedef struct { SCHAR drcSetId; UCHAR drcSetComplexityLevel; UCHAR drcLocation; UCHAR drcApplyToDownmix; UCHAR downmixIdCount; UCHAR downmixId[8]; USHORT drcSetEffect; UCHAR limiterPeakTargetPresent; FIXP_SGL limiterPeakTarget; /* e = 5 */ UCHAR drcSetTargetLoudnessPresent; SCHAR drcSetTargetLoudnessValueUpper; SCHAR drcSetTargetLoudnessValueLower; UCHAR dependsOnDrcSetPresent; union { SCHAR dependsOnDrcSet; UCHAR noIndependentUse; }; UCHAR requiresEq; shouldBeUnion { GAIN_MODIFICATION gainModificationForChannelGroup[8][4]; DUCKING_MODIFICATION duckingModificationForChannel[8]; }; SCHAR gainSetIndex[8]; /* derived data */ UCHAR drcChannelCount; UCHAR nDrcChannelGroups; SCHAR gainSetIndexForChannelGroup[8]; } DRC_INSTRUCTIONS_UNI_DRC; /* channelLayout() (Table 62) */ typedef struct { UCHAR baseChannelCount; UCHAR layoutSignalingPresent; UCHAR definedLayout; UCHAR speakerPosition[8]; } CHANNEL_LAYOUT; /* downmixInstructions() (Table 63) */ typedef struct { UCHAR downmixId; UCHAR targetChannelCount; UCHAR targetLayout; UCHAR downmixCoefficientsPresent; UCHAR bsDownmixOffset; FIXP_DBL downmixCoefficient[8 * 8]; /* e = 2 */ } DOWNMIX_INSTRUCTIONS; typedef struct { UCHAR uniDrcConfigExtType[8]; ULONG extBitSize[8 - 1]; } UNI_DRC_CONFIG_EXTENSION; /* uniDrcConfig() (Table 57) */ typedef struct { UCHAR sampleRatePresent; ULONG sampleRate; UCHAR downmixInstructionsCountV0; UCHAR downmixInstructionsCountV1; UCHAR downmixInstructionsCount; /* saturated to 6 */ UCHAR drcCoefficientsUniDrcCountV0; UCHAR drcCoefficientsUniDrcCountV1; UCHAR drcCoefficientsUniDrcCount; /* saturated to 2 */ UCHAR drcInstructionsUniDrcCountV0; UCHAR drcInstructionsUniDrcCountV1; UCHAR drcInstructionsUniDrcCount; /* saturated to (12 + 1 + 6) */ CHANNEL_LAYOUT channelLayout; DOWNMIX_INSTRUCTIONS downmixInstructions[6]; DRC_COEFFICIENTS_UNI_DRC drcCoefficientsUniDrc[2]; DRC_INSTRUCTIONS_UNI_DRC drcInstructionsUniDrc[(12 + 1 + 6)]; UCHAR uniDrcConfigExtPresent; UNI_DRC_CONFIG_EXTENSION uniDrcConfigExt; /* derived data */ UCHAR drcInstructionsCountInclVirtual; UCHAR diff; } UNI_DRC_CONFIG, *HANDLE_UNI_DRC_CONFIG; /*******************/ /* loudnessInfoSet */ /*******************/ typedef enum { MD_UNKNOWN_OTHER = 0, MD_PROGRAM_LOUDNESS = 1, MD_ANCHOR_LOUDNESS = 2, MD_MAX_OF_LOUDNESS_RANGE = 3, MD_MOMENTARY_LOUDNESS_MAX = 4, MD_SHORT_TERM_LOUDNESS_MAX = 5, MD_LOUDNESS_RANGE = 6, MD_MIXING_LEVEL = 7, MD_ROOM_TYPE = 8, MD_SHORT_TERM_LOUDNESS = 9 } METHOD_DEFINITION; typedef enum { MS_UNKNOWN_OTHER = 0, MS_EBU_R_128 = 1, MS_BS_1770_4 = 2, MS_BS_1770_4_PRE_PROCESSING = 3, MS_USER = 4, MS_EXPERT_PANEL = 5, MS_BS_1771_1 = 6, MS_RESERVED_A = 7, MS_RESERVED_B = 8, MS_RESERVED_C = 9, MS_RESERVED_D = 10, MS_RESERVED_E = 11 } MEASUREMENT_SYSTEM; typedef enum { R_UKNOWN = 0, R_UNVERIFIED = 1, R_CEILING = 2, R_ACCURATE = 3 } RELIABILITY; typedef struct { UCHAR methodDefinition; FIXP_DBL methodValue; /* e = 7 for all methodDefinitions */ UCHAR measurementSystem; UCHAR reliability; } LOUDNESS_MEASUREMENT; /* loudnessInfo() (Table 59) */ typedef struct { SCHAR drcSetId; UCHAR eqSetId; UCHAR downmixId; UCHAR samplePeakLevelPresent; FIXP_DBL samplePeakLevel; /* e = 7 */ UCHAR truePeakLevelPresent; FIXP_DBL truePeakLevel; /* e = 7 */ UCHAR truePeakLevelMeasurementSystem; UCHAR truePeakLevelReliability; UCHAR measurementCount; /* saturated to 8 */ LOUDNESS_MEASUREMENT loudnessMeasurement[8]; } LOUDNESS_INFO; /* loudnessInfoSetExtension() (Table 61) */ typedef struct { UCHAR loudnessInfoSetExtType[8]; ULONG extBitSize[8 - 1]; } LOUDNESS_INFO_SET_EXTENSION; /* loudnessInfoSet() (Table 58) */ typedef struct { UCHAR loudnessInfoAlbumCountV0; UCHAR loudnessInfoAlbumCountV1; UCHAR loudnessInfoAlbumCount; /* saturated to 12 */ UCHAR loudnessInfoCountV0; UCHAR loudnessInfoCountV1; UCHAR loudnessInfoCount; /* saturated to 12 */ LOUDNESS_INFO loudnessInfoAlbum[12]; LOUDNESS_INFO loudnessInfo[12]; UCHAR loudnessInfoSetExtPresent; LOUDNESS_INFO_SET_EXTENSION loudnessInfoSetExt; /* derived data */ UCHAR diff; } LOUDNESS_INFO_SET, *HANDLE_LOUDNESS_INFO_SET; #endif fdk-aac-2.0.2/libDRCdec/src/drcDecoder.h000066400000000000000000000133721404225310700175560ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /************************* MPEG-D DRC decoder library ************************** Author(s): Description: *******************************************************************************/ #ifndef DRCDECODER_H #define DRCDECODER_H /* drcDecoder.h: definitions used in all submodules */ #define MAX_ACTIVE_DRCS 3 typedef enum { DM_REGULAR_DELAY = 0, DM_LOW_DELAY = 1 } DELAY_MODE; typedef enum { DE_OK = 0, DE_NOT_OK = -100, DE_PARAM_OUT_OF_RANGE, DE_PARAM_INVALID, DE_MEMORY_ERROR } DRC_ERROR; typedef enum { SDM_OFF, SDM_QMF64, SDM_QMF71, SDM_STFT256 } SUBBAND_DOMAIN_MODE; #define DOWNMIX_ID_BASE_LAYOUT 0x0 #define DOWNMIX_ID_ANY_DOWNMIX 0x7F #define DRC_SET_ID_NO_DRC 0x0 #define DRC_SET_ID_ANY_DRC 0x3F #define LOCATION_MP4_INSTREAM_UNIDRC 0x1 #define LOCATION_MP4_DYN_RANGE_INFO 0x2 #define LOCATION_MP4_COMPRESSION_VALUE 0x3 #define LOCATION_SELECTED \ LOCATION_MP4_INSTREAM_UNIDRC /* set to location selected by system */ #define MAX_REQUESTS_DOWNMIX_ID 15 #define MAX_REQUESTS_DRC_FEATURE 7 #define MAX_REQUESTS_DRC_EFFECT_TYPE 15 #define DEFAULT_LOUDNESS_DEVIATION_MAX 63 #define DRC_INPUT_LOUDNESS_TARGET FL2FXCONST_DBL(-31.0f / (float)(1 << 7)) #define DRC_INPUT_LOUDNESS_TARGET_SGL FL2FXCONST_SGL(-31.0f / (float)(1 << 7)) #endif fdk-aac-2.0.2/libDRCdec/src/drcGainDec_init.cpp000066400000000000000000000276131404225310700210640ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /************************* MPEG-D DRC decoder library ************************** Author(s): Description: *******************************************************************************/ #include "drcDec_types.h" #include "drcDec_tools.h" #include "drcDec_gainDecoder.h" #include "drcGainDec_init.h" static DRC_ERROR _generateDrcInstructionsDerivedData( HANDLE_DRC_GAIN_DECODER hGainDec, HANDLE_UNI_DRC_CONFIG hUniDrcConfig, DRC_INSTRUCTIONS_UNI_DRC* pInst, DRC_COEFFICIENTS_UNI_DRC* pCoef, ACTIVE_DRC* pActiveDrc) { DRC_ERROR err = DE_OK; int g; int gainElementCount = 0; UCHAR nDrcChannelGroups = 0; SCHAR gainSetIndexForChannelGroup[8]; err = deriveDrcChannelGroups( pInst->drcSetEffect, pInst->drcChannelCount, pInst->gainSetIndex, pInst->drcSetEffect & (EB_DUCK_OTHER | EB_DUCK_SELF) ? pInst->duckingModificationForChannel : NULL, &nDrcChannelGroups, gainSetIndexForChannelGroup, pActiveDrc->channelGroupForChannel, pInst->drcSetEffect & (EB_DUCK_OTHER | EB_DUCK_SELF) ? pActiveDrc->duckingModificationForChannelGroup : NULL); if (err) return (err); /* sanity check */ if (nDrcChannelGroups != pInst->nDrcChannelGroups) return DE_NOT_OK; for (g = 0; g < pInst->nDrcChannelGroups; g++) { if (gainSetIndexForChannelGroup[g] != pInst->gainSetIndexForChannelGroup[g]) return DE_NOT_OK; } for (g = 0; g < pInst->nDrcChannelGroups; g++) { int seq = pInst->gainSetIndexForChannelGroup[g]; if (seq != -1 && (hUniDrcConfig->drcCoefficientsUniDrcCount == 0 || seq >= pCoef->gainSetCount)) { pActiveDrc->channelGroupIsParametricDrc[g] = 1; } else { pActiveDrc->channelGroupIsParametricDrc[g] = 0; if (seq >= pCoef->gainSetCount) { return DE_NOT_OK; } } } /* gainElementCount */ if (pInst->drcSetEffect & (EB_DUCK_OTHER | EB_DUCK_SELF)) { for (g = 0; g < pInst->nDrcChannelGroups; g++) { pActiveDrc->bandCountForChannelGroup[g] = 1; } pActiveDrc->gainElementCount = pInst->nDrcChannelGroups; /* one gain element per channel group */ } else { for (g = 0; g < pInst->nDrcChannelGroups; g++) { if (pActiveDrc->channelGroupIsParametricDrc[g]) { gainElementCount++; pActiveDrc->bandCountForChannelGroup[g] = 1; } else { int seq, bandCount; seq = pInst->gainSetIndexForChannelGroup[g]; bandCount = pCoef->gainSet[seq].bandCount; pActiveDrc->bandCountForChannelGroup[g] = bandCount; gainElementCount += bandCount; } } pActiveDrc->gainElementCount = gainElementCount; } /* prepare gainElementForGroup (cumulated sum of bandCountForChannelGroup) */ pActiveDrc->gainElementForGroup[0] = 0; for (g = 1; g < pInst->nDrcChannelGroups; g++) { pActiveDrc->gainElementForGroup[g] = pActiveDrc->gainElementForGroup[g - 1] + pActiveDrc->bandCountForChannelGroup[g - 1]; /* index of first gain sequence in channel group */ } return DE_OK; } DRC_ERROR initGainDec(HANDLE_DRC_GAIN_DECODER hGainDec) { int i, j, k; /* sanity check */ if (hGainDec->deltaTminDefault > hGainDec->frameSize) return DE_NOT_OK; for (i = 0; i < MAX_ACTIVE_DRCS; i++) { for (j = 0; j < 8; j++) { /* use startup node at the beginning */ hGainDec->activeDrc[i].lnbIndexForChannel[j][0] = 0; for (k = 1; k < NUM_LNB_FRAMES; k++) { hGainDec->activeDrc[i].lnbIndexForChannel[j][k] = -1; } } } for (j = 0; j < 8; j++) { hGainDec->channelGain[j] = FL2FXCONST_DBL(1.0f / (float)(1 << 8)); } for (i = 0; i < 4 * 1024 / 256; i++) { hGainDec->dummySubbandGains[i] = FL2FXCONST_DBL(1.0f / (float)(1 << 7)); } hGainDec->status = 0; /* startup */ return DE_OK; } void initDrcGainBuffers(const int frameSize, DRC_GAIN_BUFFERS* drcGainBuffers) { int i, c, j; /* prepare 12 instances of node buffers */ for (i = 0; i < 12; i++) { for (j = 0; j < NUM_LNB_FRAMES; j++) { drcGainBuffers->linearNodeBuffer[i].nNodes[j] = 1; drcGainBuffers->linearNodeBuffer[i].linearNode[j][0].gainLin = FL2FXCONST_DBL(1.0f / (float)(1 << 7)); if (j == 0) { drcGainBuffers->linearNodeBuffer[i].linearNode[j][0].time = 0; /* initialize last node with startup node */ } else { drcGainBuffers->linearNodeBuffer[i].linearNode[j][0].time = frameSize - 1; } } } /* prepare dummyLnb, a linearNodeBuffer containing a constant gain of 0 dB, * for the "no DRC processing" case */ drcGainBuffers->dummyLnb.gainInterpolationType = GIT_LINEAR; for (i = 0; i < NUM_LNB_FRAMES; i++) { drcGainBuffers->dummyLnb.nNodes[i] = 1; drcGainBuffers->dummyLnb.linearNode[i][0].gainLin = FL2FXCONST_DBL(1.0f / (float)(1 << 7)); drcGainBuffers->dummyLnb.linearNode[i][0].time = frameSize - 1; } /* prepare channelGain delay line */ for (c = 0; c < 8; c++) { for (i = 0; i < NUM_LNB_FRAMES; i++) { drcGainBuffers->channelGain[c][i] = FL2FXCONST_DBL(1.0f / (float)(1 << 8)); } } drcGainBuffers->lnbPointer = 0; } DRC_ERROR initActiveDrc(HANDLE_DRC_GAIN_DECODER hGainDec, HANDLE_UNI_DRC_CONFIG hUniDrcConfig, const int drcSetIdSelected, const int downmixIdSelected) { int g, isMultiband = 0; DRC_ERROR err = DE_OK; DRC_INSTRUCTIONS_UNI_DRC* pInst = NULL; DRC_COEFFICIENTS_UNI_DRC* pCoef = NULL; pInst = selectDrcInstructions(hUniDrcConfig, drcSetIdSelected); if (pInst == NULL) { return DE_NOT_OK; } if (pInst->drcSetId >= 0) { pCoef = selectDrcCoefficients(hUniDrcConfig, pInst->drcLocation); if (pCoef == NULL) { return DE_NOT_OK; } if (pCoef->drcFrameSizePresent) { if (pCoef->drcFrameSize != hGainDec->frameSize) { return DE_NOT_OK; } } err = _generateDrcInstructionsDerivedData( hGainDec, hUniDrcConfig, pInst, pCoef, &(hGainDec->activeDrc[hGainDec->nActiveDrcs])); if (err) return err; } hGainDec->activeDrc[hGainDec->nActiveDrcs].pInst = pInst; hGainDec->activeDrc[hGainDec->nActiveDrcs].pCoef = pCoef; for (g = 0; g < pInst->nDrcChannelGroups; g++) { if (hGainDec->activeDrc[hGainDec->nActiveDrcs].bandCountForChannelGroup[g] > 1) { if (hGainDec->multiBandActiveDrcIndex != -1) { return DE_NOT_OK; } isMultiband = 1; } } if (isMultiband) { /* Keep activeDrc index of multiband DRC set */ hGainDec->multiBandActiveDrcIndex = hGainDec->nActiveDrcs; } if ((hGainDec->channelGainActiveDrcIndex == -1) && (downmixIdSelected == DOWNMIX_ID_BASE_LAYOUT) && (hUniDrcConfig->drcInstructionsUniDrcCount > 0)) { /* use this activeDrc to apply channelGains */ hGainDec->channelGainActiveDrcIndex = hGainDec->nActiveDrcs; } hGainDec->nActiveDrcs++; if (hGainDec->nActiveDrcs > MAX_ACTIVE_DRCS) return DE_NOT_OK; return DE_OK; } DRC_ERROR initActiveDrcOffset(HANDLE_DRC_GAIN_DECODER hGainDec) { int a, accGainElementCount; accGainElementCount = 0; for (a = 0; a < hGainDec->nActiveDrcs; a++) { hGainDec->activeDrc[a].activeDrcOffset = accGainElementCount; accGainElementCount += hGainDec->activeDrc[a].gainElementCount; if (accGainElementCount > 12) { hGainDec->nActiveDrcs = a; return DE_NOT_OK; } } return DE_OK; } fdk-aac-2.0.2/libDRCdec/src/drcGainDec_init.h000066400000000000000000000122041404225310700205170ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /************************* MPEG-D DRC decoder library ************************** Author(s): Description: *******************************************************************************/ #ifndef DRCGAINDEC_INIT_H #define DRCGAINDEC_INIT_H DRC_ERROR initGainDec(HANDLE_DRC_GAIN_DECODER hGainDec); void initDrcGainBuffers(const int frameSize, DRC_GAIN_BUFFERS* drcGainBuffers); DRC_ERROR initActiveDrc(HANDLE_DRC_GAIN_DECODER hGainDec, HANDLE_UNI_DRC_CONFIG hUniDrcConfig, const int drcSetIdSelected, const int downmixIdSelected); DRC_ERROR initActiveDrcOffset(HANDLE_DRC_GAIN_DECODER hGainDec); #endif fdk-aac-2.0.2/libDRCdec/src/drcGainDec_preprocess.cpp000066400000000000000000000665461404225310700223160ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /************************* MPEG-D DRC decoder library ************************** Author(s): Description: *******************************************************************************/ #include "drcDec_types.h" #include "drcDec_gainDecoder.h" #include "drcGainDec_preprocess.h" #include "drcDec_tools.h" #include "FDK_matrixCalloc.h" #include "drcDec_rom.h" #define SLOPE_FACTOR_DB_TO_LINEAR \ FL2FXCONST_DBL(0.1151f * (float)(1 << 3)) /* ln(10) / 20 */ typedef struct { int drcSetEffect; DUCKING_MODIFICATION* pDMod; GAIN_MODIFICATION* pGMod; int drcCharacteristicPresent; CHARACTERISTIC_FORMAT characteristicFormatSource[2]; const CUSTOM_DRC_CHAR* pCCharSource[2]; CHARACTERISTIC_FORMAT characteristicFormatTarget[2]; const CUSTOM_DRC_CHAR* pCCharTarget[2]; int slopeIsNegative; int limiterPeakTargetPresent; FIXP_SGL limiterPeakTarget; FIXP_DBL loudnessNormalizationGainDb; FIXP_SGL compress; FIXP_SGL boost; } NODE_MODIFICATION; static DRC_ERROR _getCicpCharacteristic( const int cicpCharacteristic, CHARACTERISTIC_FORMAT pCharacteristicFormat[2], const CUSTOM_DRC_CHAR* pCCharSource[2]) { if ((cicpCharacteristic < 1) || (cicpCharacteristic > 11)) { return DE_NOT_OK; } if (cicpCharacteristic < 7) { /* sigmoid characteristic */ pCharacteristicFormat[CS_LEFT] = CF_SIGMOID; pCCharSource[CS_LEFT] = (const CUSTOM_DRC_CHAR*)(&cicpDrcCharSigmoidLeft[cicpCharacteristic - 1]); pCharacteristicFormat[CS_RIGHT] = CF_SIGMOID; pCCharSource[CS_RIGHT] = (const CUSTOM_DRC_CHAR*)(&cicpDrcCharSigmoidRight[cicpCharacteristic - 1]); } else { /* nodes characteristic */ pCharacteristicFormat[CS_LEFT] = CF_NODES; pCCharSource[CS_LEFT] = (const CUSTOM_DRC_CHAR*)(&cicpDrcCharNodesLeft[cicpCharacteristic - 7]); pCharacteristicFormat[CS_RIGHT] = CF_NODES; pCCharSource[CS_RIGHT] = (const CUSTOM_DRC_CHAR*)(&cicpDrcCharNodesRight[cicpCharacteristic - 7]); } return DE_OK; } static int _getSign(FIXP_SGL in) { if (in > (FIXP_DBL)0) return 1; if (in < (FIXP_DBL)0) return -1; return 0; } static DRC_ERROR _getSlopeSign(const CHARACTERISTIC_FORMAT drcCharFormat, const CUSTOM_DRC_CHAR* pCChar, int* pSlopeSign) { if (drcCharFormat == CF_SIGMOID) { *pSlopeSign = (pCChar->sigmoid.flipSign ? 1 : -1); } else { int k, slopeSign = 0, tmp_slopeSign; for (k = 0; k < pCChar->nodes.characteristicNodeCount; k++) { if (pCChar->nodes.nodeLevel[k + 1] > pCChar->nodes.nodeLevel[k]) { tmp_slopeSign = _getSign(pCChar->nodes.nodeGain[k + 1] - pCChar->nodes.nodeGain[k]); } else { tmp_slopeSign = -_getSign(pCChar->nodes.nodeGain[k + 1] - pCChar->nodes.nodeGain[k]); } if ((slopeSign || tmp_slopeSign) && (slopeSign == -tmp_slopeSign)) return DE_NOT_OK; /* DRC characteristic is not invertible */ else slopeSign = tmp_slopeSign; } *pSlopeSign = slopeSign; } return DE_OK; } static DRC_ERROR _isSlopeNegative(const CHARACTERISTIC_FORMAT drcCharFormat[2], const CUSTOM_DRC_CHAR* pCChar[2], int* pSlopeIsNegative) { DRC_ERROR err = DE_OK; int slopeSign[2] = {0, 0}; err = _getSlopeSign(drcCharFormat[CS_LEFT], pCChar[CS_LEFT], &slopeSign[CS_LEFT]); if (err) return err; err = _getSlopeSign(drcCharFormat[CS_RIGHT], pCChar[CS_RIGHT], &slopeSign[CS_RIGHT]); if (err) return err; if ((slopeSign[CS_LEFT] || slopeSign[CS_RIGHT]) && (slopeSign[CS_LEFT] == -slopeSign[CS_RIGHT])) return DE_NOT_OK; /* DRC characteristic is not invertible */ *pSlopeIsNegative = (slopeSign[CS_LEFT] < 0); return DE_OK; } static DRC_ERROR _prepareDrcCharacteristic(const DRC_CHARACTERISTIC* pDChar, DRC_COEFFICIENTS_UNI_DRC* pCoef, const int b, NODE_MODIFICATION* pNodeMod) { DRC_ERROR err = DE_OK; pNodeMod->drcCharacteristicPresent = pDChar->present; if (pNodeMod->drcCharacteristicPresent) { if (pDChar->isCICP == 1) { err = _getCicpCharacteristic(pDChar->cicpIndex, pNodeMod->characteristicFormatSource, pNodeMod->pCCharSource); if (err) return err; } else { pNodeMod->characteristicFormatSource[CS_LEFT] = (CHARACTERISTIC_FORMAT) pCoef->characteristicLeftFormat[pDChar->custom.left]; pNodeMod->pCCharSource[CS_LEFT] = &(pCoef->customCharacteristicLeft[pDChar->custom.left]); pNodeMod->characteristicFormatSource[CS_RIGHT] = (CHARACTERISTIC_FORMAT) pCoef->characteristicRightFormat[pDChar->custom.right]; pNodeMod->pCCharSource[CS_RIGHT] = &(pCoef->customCharacteristicRight[pDChar->custom.right]); } err = _isSlopeNegative(pNodeMod->characteristicFormatSource, pNodeMod->pCCharSource, &pNodeMod->slopeIsNegative); if (err) return err; if (pNodeMod->pGMod != NULL) { if (pNodeMod->pGMod[b].targetCharacteristicLeftPresent) { pNodeMod->characteristicFormatTarget[CS_LEFT] = (CHARACTERISTIC_FORMAT)pCoef->characteristicLeftFormat [pNodeMod->pGMod[b].targetCharacteristicLeftIndex]; pNodeMod->pCCharTarget[CS_LEFT] = &(pCoef->customCharacteristicLeft [pNodeMod->pGMod[b].targetCharacteristicLeftIndex]); } if (pNodeMod->pGMod[b].targetCharacteristicRightPresent) { pNodeMod->characteristicFormatTarget[CS_RIGHT] = (CHARACTERISTIC_FORMAT)pCoef->characteristicRightFormat [pNodeMod->pGMod[b].targetCharacteristicRightIndex]; pNodeMod->pCCharTarget[CS_RIGHT] = &(pCoef->customCharacteristicRight [pNodeMod->pGMod[b].targetCharacteristicRightIndex]); } } } return DE_OK; } static DRC_ERROR _compressorIO_sigmoid_common( const FIXP_DBL tmp, /* e = 7 */ const FIXP_DBL gainDbLimit, /* e = 6 */ const FIXP_DBL exp, /* e = 5 */ const int inverse, FIXP_DBL* out) /* e = 7 */ { FIXP_DBL x, tmp1, tmp2, invExp, denom; int e_x, e_tmp1, e_tmp2, e_invExp, e_denom, e_out; if (exp < FL2FXCONST_DBL(1.0f / (float)(1 << 5))) { return DE_NOT_OK; } /* x = tmp / gainDbLimit; */ x = fDivNormSigned(tmp, gainDbLimit, &e_x); e_x += 7 - 6; if (x < (FIXP_DBL)0) { return DE_NOT_OK; } /* out = tmp / pow(1.0f +/- pow(x, exp), 1.0f/exp); */ tmp1 = fPow(x, e_x, exp, 5, &e_tmp1); if (inverse) tmp1 = -tmp1; tmp2 = fAddNorm(FL2FXCONST_DBL(1.0f / (float)(1 << 1)), 1, tmp1, e_tmp1, &e_tmp2); invExp = fDivNorm(FL2FXCONST_DBL(1.0f / (float)(1 << 1)), exp, &e_invExp); e_invExp += 1 - 5; if (tmp2 < (FIXP_DBL)0) { return DE_NOT_OK; } denom = fPow(tmp2, e_tmp2, invExp, e_invExp, &e_denom); *out = fDivNormSigned(tmp, denom, &e_out); e_out += 7 - e_denom; *out = scaleValueSaturate(*out, e_out - 7); return DE_OK; } static DRC_ERROR _compressorIO_sigmoid(const CUSTOM_DRC_CHAR_SIGMOID* pCChar, const FIXP_DBL inLevelDb, /* e = 7 */ FIXP_DBL* outGainDb) /* e = 7 */ { FIXP_DBL tmp; FIXP_SGL exp = pCChar->exp; DRC_ERROR err = DE_OK; tmp = fMultDiv2((DRC_INPUT_LOUDNESS_TARGET >> 1) - (inLevelDb >> 1), pCChar->ioRatio); tmp = SATURATE_LEFT_SHIFT(tmp, 2 + 1 + 1, DFRACT_BITS); if (exp < (FIXP_SGL)MAXVAL_SGL) { /* x = tmp / gainDbLimit; */ /* *outGainDb = tmp / pow(1.0f + pow(x, exp), 1.0f/exp); */ err = _compressorIO_sigmoid_common(tmp, FX_SGL2FX_DBL(pCChar->gain), FX_SGL2FX_DBL(exp), 0, outGainDb); if (err) return err; } else { *outGainDb = tmp; /* scaling of outGainDb (7) is equal to scaling of tmp (7) */ } if (pCChar->flipSign == 1) { *outGainDb = -*outGainDb; } return err; } static DRC_ERROR _compressorIO_sigmoid_inverse( const CUSTOM_DRC_CHAR_SIGMOID* pCChar, const FIXP_SGL gainDb, FIXP_DBL* inLev) { DRC_ERROR err = DE_OK; FIXP_SGL ioRatio = pCChar->ioRatio; FIXP_SGL exp = pCChar->exp; FIXP_DBL tmp = FX_SGL2FX_DBL(gainDb), tmp_out; int e_out; if (pCChar->flipSign == 1) { tmp = -tmp; } if (exp < (FIXP_SGL)MAXVAL_SGL) { /* x = tmp / gainDbLimit; */ /* tmp = tmp / pow(1.0f - pow(x, exp), 1.0f / exp); */ err = _compressorIO_sigmoid_common(tmp, FX_SGL2FX_DBL(pCChar->gain), FX_SGL2FX_DBL(exp), 1, &tmp); if (err) return err; } if (ioRatio == (FIXP_SGL)0) { return DE_NOT_OK; } tmp_out = fDivNormSigned(tmp, FX_SGL2FX_DBL(ioRatio), &e_out); e_out += 7 - 2; tmp_out = fAddNorm(DRC_INPUT_LOUDNESS_TARGET, 7, -tmp_out, e_out, &e_out); *inLev = scaleValueSaturate(tmp_out, e_out - 7); return err; } static DRC_ERROR _compressorIO_nodes(const CUSTOM_DRC_CHAR_NODES* pCChar, const FIXP_DBL inLevelDb, /* e = 7 */ FIXP_DBL* outGainDb) /* e = 7 */ { int n; FIXP_DBL w; const FIXP_SGL* nodeLevel = pCChar->nodeLevel; const FIXP_SGL* nodeGain = pCChar->nodeGain; if (inLevelDb < DRC_INPUT_LOUDNESS_TARGET) { for (n = 0; n < pCChar->characteristicNodeCount; n++) { if ((inLevelDb <= FX_SGL2FX_DBL(nodeLevel[n])) && (inLevelDb > FX_SGL2FX_DBL(nodeLevel[n + 1]))) { w = fDivNorm(inLevelDb - FX_SGL2FX_DBL(nodeLevel[n + 1]), FX_SGL2FX_DBL(nodeLevel[n] - nodeLevel[n + 1])); *outGainDb = fMult(w, nodeGain[n]) + fMult((FIXP_DBL)MAXVAL_DBL - w, nodeGain[n + 1]); /* *outGainDb = (w * nodeGain[n] + (1.0-w) * nodeGain[n+1]); */ return DE_OK; } } } else { for (n = 0; n < pCChar->characteristicNodeCount; n++) { if ((inLevelDb >= FX_SGL2FX_DBL(nodeLevel[n])) && (inLevelDb < FX_SGL2FX_DBL(nodeLevel[n + 1]))) { w = fDivNorm(FX_SGL2FX_DBL(nodeLevel[n + 1]) - inLevelDb, FX_SGL2FX_DBL(nodeLevel[n + 1] - nodeLevel[n])); *outGainDb = fMult(w, nodeGain[n]) + fMult((FIXP_DBL)MAXVAL_DBL - w, nodeGain[n + 1]); /* *outGainDb = (w * nodeGain[n] + (1.0-w) * nodeGain[n+1]); */ return DE_OK; } } } *outGainDb = FX_SGL2FX_DBL(nodeGain[pCChar->characteristicNodeCount]); return DE_OK; } static DRC_ERROR _compressorIO_nodes_inverse( const CUSTOM_DRC_CHAR_NODES* pCChar, const FIXP_SGL gainDb, /* e = 7 */ FIXP_DBL* inLev) /* e = 7 */ { int n; int k; FIXP_DBL w; int gainIsNegative = 0; const FIXP_SGL* nodeLevel = pCChar->nodeLevel; const FIXP_SGL* nodeGain = pCChar->nodeGain; int nodeCount = pCChar->characteristicNodeCount; for (k = 0; k < nodeCount; k++) { if (pCChar->nodeGain[k + 1] < (FIXP_SGL)0) { gainIsNegative = 1; } } if (gainIsNegative == 1) { if (gainDb <= nodeGain[nodeCount]) { *inLev = FX_SGL2FX_DBL(nodeLevel[nodeCount]); } else { if (gainDb >= (FIXP_SGL)0) { *inLev = DRC_INPUT_LOUDNESS_TARGET; } else { for (n = 0; n < nodeCount; n++) { if ((gainDb <= nodeGain[n]) && (gainDb > nodeGain[n + 1])) { FIXP_SGL gainDelta = nodeGain[n] - nodeGain[n + 1]; if (gainDelta == (FIXP_SGL)0) { *inLev = FX_SGL2FX_DBL(nodeLevel[n]); return DE_OK; } w = fDivNorm(gainDb - nodeGain[n + 1], gainDelta); *inLev = fMult(w, nodeLevel[n]) + fMult((FIXP_DBL)MAXVAL_DBL - w, nodeLevel[n + 1]); /* *inLev = (w * nodeLevel[n] + (1.0-w) * nodeLevel[n+1]); */ return DE_OK; } } *inLev = FX_SGL2FX_DBL(nodeLevel[nodeCount]); } } } else { if (gainDb >= nodeGain[nodeCount]) { *inLev = FX_SGL2FX_DBL(nodeLevel[nodeCount]); } else { if (gainDb <= (FIXP_SGL)0) { *inLev = DRC_INPUT_LOUDNESS_TARGET; } else { for (n = 0; n < nodeCount; n++) { if ((gainDb >= nodeGain[n]) && (gainDb < nodeGain[n + 1])) { FIXP_SGL gainDelta = nodeGain[n + 1] - nodeGain[n]; if (gainDelta == (FIXP_SGL)0) { *inLev = FX_SGL2FX_DBL(nodeLevel[n]); return DE_OK; } w = fDivNorm(nodeGain[n + 1] - gainDb, gainDelta); *inLev = fMult(w, nodeLevel[n]) + fMult((FIXP_DBL)MAXVAL_DBL - w, nodeLevel[n + 1]); /* *inLev = (w * nodeLevel[n] + (1.0-w) * nodeLevel[n+1]); */ return DE_OK; } } *inLev = FX_SGL2FX_DBL(nodeLevel[nodeCount]); } } } return DE_OK; } static DRC_ERROR _mapGain(const CHARACTERISTIC_FORMAT pCCharFormatSource, const CUSTOM_DRC_CHAR* pCCharSource, const CHARACTERISTIC_FORMAT pCCharFormatTarget, const CUSTOM_DRC_CHAR* pCCharTarget, const FIXP_SGL gainInDb, /* e = 7 */ FIXP_DBL* gainOutDb) /* e = 7 */ { FIXP_DBL inLevel = (FIXP_DBL)0; DRC_ERROR err = DE_OK; switch (pCCharFormatSource) { case CF_SIGMOID: err = _compressorIO_sigmoid_inverse( (const CUSTOM_DRC_CHAR_SIGMOID*)pCCharSource, gainInDb, &inLevel); if (err) return err; break; case CF_NODES: err = _compressorIO_nodes_inverse( (const CUSTOM_DRC_CHAR_NODES*)pCCharSource, gainInDb, &inLevel); if (err) return err; break; default: return DE_NOT_OK; } switch (pCCharFormatTarget) { case CF_SIGMOID: err = _compressorIO_sigmoid((const CUSTOM_DRC_CHAR_SIGMOID*)pCCharTarget, inLevel, gainOutDb); if (err) return err; break; case CF_NODES: err = _compressorIO_nodes((const CUSTOM_DRC_CHAR_NODES*)pCCharTarget, inLevel, gainOutDb); if (err) return err; break; default: break; } return DE_OK; } static DRC_ERROR _toLinear( const NODE_MODIFICATION* nodeMod, const int drcBand, const FIXP_SGL gainDb, /* in: gain value in dB, e = 7 */ const FIXP_SGL slopeDb, /* in: slope value in dB/deltaTmin, e = 2 */ FIXP_DBL* gainLin, /* out: linear gain value, e = 7 */ FIXP_DBL* slopeLin) /* out: linear slope value, e = 7 */ { FIXP_DBL gainRatio_m = FL2FXCONST_DBL(1.0f / (float)(1 << 1)); GAIN_MODIFICATION* pGMod = NULL; DUCKING_MODIFICATION* pDMod = nodeMod->pDMod; FIXP_DBL tmp_dbl, gainDb_modified, gainDb_offset, gainDb_out, gainLin_m, slopeLin_m; int gainLin_e, gainRatio_e = 1, gainDb_out_e; if (nodeMod->pGMod != NULL) { pGMod = &(nodeMod->pGMod[drcBand]); } if (((nodeMod->drcSetEffect & (EB_DUCK_OTHER | EB_DUCK_SELF)) == 0) && (nodeMod->drcSetEffect != EB_FADE) && (nodeMod->drcSetEffect != EB_CLIPPING)) { DRC_ERROR err = DE_OK; FIXP_DBL gainDbMapped; if ((pGMod != NULL) && (nodeMod->drcCharacteristicPresent)) { if (((gainDb > (FIXP_SGL)0) && nodeMod->slopeIsNegative) || ((gainDb < (FIXP_SGL)0) && !nodeMod->slopeIsNegative)) { /* left side */ if (pGMod->targetCharacteristicLeftPresent == 1) { err = _mapGain(nodeMod->characteristicFormatSource[CS_LEFT], nodeMod->pCCharSource[CS_LEFT], nodeMod->characteristicFormatTarget[CS_LEFT], nodeMod->pCCharTarget[CS_LEFT], gainDb, &gainDbMapped); if (err) return err; gainRatio_m = fDivNormSigned( gainDbMapped, FX_SGL2FX_DBL(gainDb), &gainRatio_e); /* target characteristic in payload */ } } else { /* if (((gainDb < (FIXP_SGL)0) && nodeMod->slopeIsNegative) || ((gainDb > (FIXP_SGL)0) && !nodeMod->slopeIsNegative)) */ /* right side */ if (pGMod->targetCharacteristicRightPresent == 1) { err = _mapGain(nodeMod->characteristicFormatSource[CS_RIGHT], nodeMod->pCCharSource[CS_RIGHT], nodeMod->characteristicFormatTarget[CS_RIGHT], nodeMod->pCCharTarget[CS_RIGHT], gainDb, &gainDbMapped); if (err) return err; gainRatio_m = fDivNormSigned( gainDbMapped, FX_SGL2FX_DBL(gainDb), &gainRatio_e); /* target characteristic in payload */ } } } if (gainDb < (FIXP_SGL)0) { gainRatio_m = fMultDiv2(gainRatio_m, nodeMod->compress); } else { gainRatio_m = fMultDiv2(gainRatio_m, nodeMod->boost); } gainRatio_e += 2; } if ((pGMod != NULL) && (pGMod->gainScalingPresent == 1)) { if (gainDb < (FIXP_SGL)0) { gainRatio_m = fMultDiv2(gainRatio_m, pGMod->attenuationScaling); } else { gainRatio_m = fMultDiv2(gainRatio_m, pGMod->amplificationScaling); } gainRatio_e += 3; } if ((pDMod != NULL) && (nodeMod->drcSetEffect & (EB_DUCK_OTHER | EB_DUCK_SELF)) && (pDMod->duckingScalingPresent == 1)) { gainRatio_m = fMultDiv2(gainRatio_m, pDMod->duckingScaling); gainRatio_e += 3; } gainDb_modified = fMultDiv2(gainDb, gainRatio_m); /* resulting e: 7 + gainRatio_e + 1*/ gainDb_offset = (FIXP_DBL)0; if ((pGMod != NULL) && (pGMod->gainOffsetPresent == 1)) { /* *gainLin *= (float)pow(2.0, (double)(pGMod->gainOffset/6.0f)); */ gainDb_offset += FX_SGL2FX_DBL(pGMod->gainOffset) >> 4; /* resulting e: 8 */ } if ((nodeMod->limiterPeakTargetPresent == 1) && (nodeMod->drcSetEffect == EB_CLIPPING)) { /* The only drcSetEffect is "clipping prevention" */ /* loudnessNormalizationGainModificationDb is included in * loudnessNormalizationGainDb */ /* *gainLin *= (float)pow(2.0, max(0.0, -nodeModification->limiterPeakTarget * - nodeModification->loudnessNormalizationGainDb)/6.0); */ gainDb_offset += fMax( (FIXP_DBL)0, (FX_SGL2FX_DBL(-nodeMod->limiterPeakTarget) >> 3) - (nodeMod->loudnessNormalizationGainDb >> 1)); /* resulting e: 8 */ } if (gainDb_offset != (FIXP_DBL)0) { gainDb_out = fAddNorm(gainDb_modified, 7 + gainRatio_e + 1, gainDb_offset, 8, &gainDb_out_e); } else { gainDb_out = gainDb_modified; gainDb_out_e = 7 + gainRatio_e + 1; } /* *gainLin = (float)pow(2.0, (double)(gainDb_modified[1] / 6.0f)); */ gainLin_m = approxDb2lin(gainDb_out, gainDb_out_e, &gainLin_e); *gainLin = scaleValueSaturate(gainLin_m, gainLin_e - 7); /* *slopeLin = SLOPE_FACTOR_DB_TO_LINEAR * gainRatio * *gainLin * slopeDb; */ if (slopeDb == (FIXP_SGL)0) { *slopeLin = (FIXP_DBL)0; } else { tmp_dbl = fMult(slopeDb, SLOPE_FACTOR_DB_TO_LINEAR); /* resulting e: 2 - 3 = -1 */ tmp_dbl = fMult(tmp_dbl, gainRatio_m); /* resulting e: -1 + gainRatio_e */ if (gainDb_offset != (FIXP_DBL)0) { /* recalculate gainLin from gainDb that wasn't modified by gainOffset and limiterPeakTarget */ gainLin_m = approxDb2lin(gainDb_modified, 7 + gainRatio_e, &gainLin_e); } slopeLin_m = fMult(tmp_dbl, gainLin_m); *slopeLin = scaleValueSaturate(slopeLin_m, -1 + gainRatio_e + gainLin_e - 7); } if ((nodeMod->limiterPeakTargetPresent == 1) && (nodeMod->drcSetEffect == EB_CLIPPING)) { if (*gainLin >= FL2FXCONST_DBL(1.0f / (float)(1 << 7))) { *gainLin = FL2FXCONST_DBL(1.0f / (float)(1 << 7)); *slopeLin = (FIXP_DBL)0; } } return DE_OK; } /* prepare buffers containing linear nodes for each gain sequence */ DRC_ERROR prepareDrcGain(HANDLE_DRC_GAIN_DECODER hGainDec, HANDLE_UNI_DRC_GAIN hUniDrcGain, const FIXP_SGL compress, const FIXP_SGL boost, const FIXP_DBL loudnessNormalizationGainDb, const int activeDrcIndex) { int b, g, gainElementIndex; DRC_GAIN_BUFFERS* drcGainBuffers = &(hGainDec->drcGainBuffers); NODE_MODIFICATION nodeMod; FDKmemclear(&nodeMod, sizeof(NODE_MODIFICATION)); ACTIVE_DRC* pActiveDrc = &(hGainDec->activeDrc[activeDrcIndex]); DRC_INSTRUCTIONS_UNI_DRC* pInst = pActiveDrc->pInst; if (pInst == NULL) return DE_NOT_OK; nodeMod.drcSetEffect = pInst->drcSetEffect; nodeMod.compress = compress; nodeMod.boost = boost; nodeMod.loudnessNormalizationGainDb = loudnessNormalizationGainDb; nodeMod.limiterPeakTargetPresent = pInst->limiterPeakTargetPresent; nodeMod.limiterPeakTarget = pInst->limiterPeakTarget; gainElementIndex = 0; for (g = 0; g < pInst->nDrcChannelGroups; g++) { int gainSetIndex = 0; int nDrcBands = 0; DRC_COEFFICIENTS_UNI_DRC* pCoef = pActiveDrc->pCoef; if (pCoef == NULL) return DE_NOT_OK; if (!pActiveDrc->channelGroupIsParametricDrc[g]) { gainSetIndex = pInst->gainSetIndexForChannelGroup[g]; if (nodeMod.drcSetEffect & (EB_DUCK_OTHER | EB_DUCK_SELF)) { nodeMod.pDMod = &(pActiveDrc->duckingModificationForChannelGroup[g]); nodeMod.pGMod = NULL; } else { nodeMod.pGMod = pInst->gainModificationForChannelGroup[g]; nodeMod.pDMod = NULL; } nDrcBands = pActiveDrc->bandCountForChannelGroup[g]; for (b = 0; b < nDrcBands; b++) { DRC_ERROR err = DE_OK; GAIN_SET* pGainSet = &(pCoef->gainSet[gainSetIndex]); int seq = pGainSet->gainSequenceIndex[b]; DRC_CHARACTERISTIC* pDChar = &(pGainSet->drcCharacteristic[b]); /* linearNodeBuffer contains a copy of the gain sequences (consisting of nodes) that are relevant for decoding. It also contains gain sequences of previous frames. */ LINEAR_NODE_BUFFER* pLnb = &(drcGainBuffers->linearNodeBuffer[pActiveDrc->activeDrcOffset + gainElementIndex]); int i, lnbp; lnbp = drcGainBuffers->lnbPointer; pLnb->gainInterpolationType = (GAIN_INTERPOLATION_TYPE)pGainSet->gainInterpolationType; err = _prepareDrcCharacteristic(pDChar, pCoef, b, &nodeMod); if (err) return err; /* copy a node buffer and convert from dB to linear */ pLnb->nNodes[lnbp] = fMin((int)hUniDrcGain->nNodes[seq], 16); for (i = 0; i < pLnb->nNodes[lnbp]; i++) { FIXP_DBL gainLin, slopeLin; err = _toLinear(&nodeMod, b, hUniDrcGain->gainNode[seq][i].gainDb, (FIXP_SGL)0, &gainLin, &slopeLin); if (err) return err; pLnb->linearNode[lnbp][i].gainLin = gainLin; pLnb->linearNode[lnbp][i].time = hUniDrcGain->gainNode[seq][i].time; } gainElementIndex++; } } else { /* parametric DRC not supported */ gainElementIndex++; } } return DE_OK; } fdk-aac-2.0.2/libDRCdec/src/drcGainDec_preprocess.h000066400000000000000000000120141404225310700217400ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /************************* MPEG-D DRC decoder library ************************** Author(s): Description: *******************************************************************************/ #ifndef DRCGAINDEC_PREPROCESS_H #define DRCGAINDEC_PREPROCESS_H DRC_ERROR prepareDrcGain(HANDLE_DRC_GAIN_DECODER hGainDec, HANDLE_UNI_DRC_GAIN hUniDrcGain, const FIXP_SGL compress, const FIXP_SGL boost, const FIXP_DBL loudnessNormalizationGainDb, const int activeDrcIndex); #endif fdk-aac-2.0.2/libDRCdec/src/drcGainDec_process.cpp000066400000000000000000000513151404225310700215730ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /************************* MPEG-D DRC decoder library ************************** Author(s): Description: *******************************************************************************/ #include "drcDec_types.h" #include "drcDec_gainDecoder.h" #include "drcGainDec_process.h" #define E_TGAINSTEP 12 static DRC_ERROR _prepareLnbIndex(ACTIVE_DRC* pActiveDrc, const int channelOffset, const int drcChannelOffset, const int numChannelsProcessed, const int lnbPointer) { int g, c; DRC_INSTRUCTIONS_UNI_DRC* pInst = pActiveDrc->pInst; /* channelOffset: start index of physical channels numChannelsProcessed: number of processed channels, physical channels and DRC channels channelOffset + drcChannelOffset: start index of DRC channels, i.e. the channel order referenced in pInst.sequenceIndex */ /* sanity checks */ if ((channelOffset + numChannelsProcessed) > 8) return DE_NOT_OK; if ((channelOffset + drcChannelOffset + numChannelsProcessed) > 8) return DE_NOT_OK; if ((channelOffset + drcChannelOffset) < 0) return DE_NOT_OK; /* prepare lnbIndexForChannel, a map of indices from each channel to its * corresponding linearNodeBuffer instance */ for (c = channelOffset; c < channelOffset + numChannelsProcessed; c++) { if (pInst->drcSetId > 0) { int drcChannel = c + drcChannelOffset; /* fallback for configuration with more physical channels than DRC channels: reuse DRC gain of first channel. This is necessary for HE-AAC mono with stereo output */ if (drcChannel >= pInst->drcChannelCount) drcChannel = 0; g = pActiveDrc->channelGroupForChannel[drcChannel]; if ((g >= 0) && !pActiveDrc->channelGroupIsParametricDrc[g]) { pActiveDrc->lnbIndexForChannel[c][lnbPointer] = pActiveDrc->activeDrcOffset + pActiveDrc->gainElementForGroup[g]; } } } return DE_OK; } static DRC_ERROR _interpolateDrcGain( const GAIN_INTERPOLATION_TYPE gainInterpolationType, const SHORT timePrev, /* time0 */ const SHORT tGainStep, /* time1 - time0 */ const SHORT start, const SHORT stop, const SHORT stepsize, const FIXP_DBL gainLeft, const FIXP_DBL gainRight, const FIXP_DBL slopeLeft, const FIXP_DBL slopeRight, FIXP_DBL* buffer) { int n, n_buf; int start_modulo, start_offset; if (tGainStep < 0) { return DE_NOT_OK; } if (tGainStep == 0) { return DE_OK; } /* get start index offset and buffer index for downsampled interpolation */ /* start_modulo = (start+timePrev)%stepsize; */ /* stepsize is a power of 2 */ start_modulo = (start + timePrev) & (stepsize - 1); start_offset = (start_modulo ? stepsize - start_modulo : 0); /* n_buf = (start + timePrev + start_offset)/stepsize; */ n_buf = (start + timePrev + start_offset) >> (15 - fixnormz_S(stepsize)); { /* gainInterpolationType == GIT_LINEAR */ LONG a; /* runs = ceil((stop - start - start_offset)/stepsize). This works for * stepsize = 2^N only. */ INT runs = (INT)(stop - start - start_offset + stepsize - 1) >> (30 - CountLeadingBits(stepsize)); INT n_min = fMin( fMin(CntLeadingZeros(gainRight), CntLeadingZeros(gainLeft)) - 1, 8); a = (LONG)((gainRight << n_min) - (gainLeft << n_min)) / tGainStep; LONG a_step = a * stepsize; n = start + start_offset; a = a * n + (LONG)(gainLeft << n_min); buffer += n_buf; #if defined(FUNCTION_interpolateDrcGain_func1) interpolateDrcGain_func1(buffer, a, a_step, n_min, runs); #else a -= a_step; n_min = 8 - n_min; for (int i = 0; i < runs; i++) { a += a_step; buffer[i] = fMultDiv2(buffer[i], (FIXP_DBL)a) << n_min; } #endif /* defined(FUNCTION_interpolateDrcGain_func1) */ } return DE_OK; } static DRC_ERROR _processNodeSegments( const int frameSize, const GAIN_INTERPOLATION_TYPE gainInterpolationType, const int nNodes, const NODE_LIN* pNodeLin, const int offset, const SHORT stepsize, const NODE_LIN nodePrevious, /* the last node of the previous frame */ const FIXP_DBL channelGain, FIXP_DBL* buffer) { DRC_ERROR err = DE_OK; SHORT timePrev, duration, start, stop, time; int n; FIXP_DBL gainLin = FL2FXCONST_DBL(1.0f / (float)(1 << 7)), gainLinPrev; FIXP_DBL slopeLin = (FIXP_DBL)0, slopeLinPrev = (FIXP_DBL)0; timePrev = nodePrevious.time + offset; gainLinPrev = nodePrevious.gainLin; for (n = 0; n < nNodes; n++) { time = pNodeLin[n].time + offset; duration = time - timePrev; gainLin = pNodeLin[n].gainLin; if (channelGain != FL2FXCONST_DBL(1.0f / (float)(1 << 8))) gainLin = SATURATE_LEFT_SHIFT(fMultDiv2(gainLin, channelGain), 9, DFRACT_BITS); if ((timePrev >= (frameSize - 1)) || (time < 0)) { /* This segment (between previous and current node) lies outside of this audio frame */ timePrev = time; gainLinPrev = gainLin; slopeLinPrev = slopeLin; continue; } /* start and stop are the boundaries of the region of this segment that lie within this audio frame. Their values are relative to the beginning of this segment. stop is the first sample that isn't processed any more. */ start = fMax(-timePrev, 1); stop = fMin(time, (SHORT)(frameSize - 1)) - timePrev + 1; err = _interpolateDrcGain(gainInterpolationType, timePrev, duration, start, stop, stepsize, gainLinPrev, gainLin, slopeLinPrev, slopeLin, buffer); if (err) return err; timePrev = time; gainLinPrev = gainLin; } return err; } /* process DRC on time-domain signal */ DRC_ERROR processDrcTime(HANDLE_DRC_GAIN_DECODER hGainDec, const int activeDrcIndex, const int delaySamples, const int channelOffset, const int drcChannelOffset, const int numChannelsProcessed, const int timeDataChannelOffset, FIXP_DBL* deinterleavedAudio) { DRC_ERROR err = DE_OK; int c, b, i; ACTIVE_DRC* pActiveDrc = &(hGainDec->activeDrc[activeDrcIndex]); DRC_GAIN_BUFFERS* pDrcGainBuffers = &(hGainDec->drcGainBuffers); int lnbPointer = pDrcGainBuffers->lnbPointer, lnbIx; LINEAR_NODE_BUFFER* pLinearNodeBuffer = pDrcGainBuffers->linearNodeBuffer; LINEAR_NODE_BUFFER* pDummyLnb = &(pDrcGainBuffers->dummyLnb); int offset = 0; if (hGainDec->delayMode == DM_REGULAR_DELAY) { offset = hGainDec->frameSize; } if ((delaySamples + offset) > (NUM_LNB_FRAMES - 2) * hGainDec->frameSize) /* if delaySamples is too big, NUM_LNB_FRAMES should be increased */ return DE_NOT_OK; err = _prepareLnbIndex(pActiveDrc, channelOffset, drcChannelOffset, numChannelsProcessed, lnbPointer); if (err) return err; deinterleavedAudio += channelOffset * timeDataChannelOffset; /* apply channelOffset */ /* signal processing loop */ for (c = channelOffset; c < channelOffset + numChannelsProcessed; c++) { if (activeDrcIndex == hGainDec->channelGainActiveDrcIndex) pDrcGainBuffers->channelGain[c][lnbPointer] = hGainDec->channelGain[c]; b = 0; { LINEAR_NODE_BUFFER *pLnb, *pLnbPrevious; NODE_LIN nodePrevious; int lnbPointerDiff; FIXP_DBL channelGain; /* get pointer to oldest linearNodes */ lnbIx = lnbPointer + 1 - NUM_LNB_FRAMES; while (lnbIx < 0) lnbIx += NUM_LNB_FRAMES; if (activeDrcIndex == hGainDec->channelGainActiveDrcIndex) channelGain = pDrcGainBuffers->channelGain[c][lnbIx]; else channelGain = FL2FXCONST_DBL(1.0f / (float)(1 << 8)); /* Loop over all node buffers in linearNodeBuffer. All nodes which are not relevant for the current frame are sorted out inside _processNodeSegments. */ for (i = 0; i < NUM_LNB_FRAMES - 1; i++) { /* Prepare previous node */ if (pActiveDrc->lnbIndexForChannel[c][lnbIx] >= 0) pLnbPrevious = &( pLinearNodeBuffer[pActiveDrc->lnbIndexForChannel[c][lnbIx] + b]); else pLnbPrevious = pDummyLnb; nodePrevious = pLnbPrevious->linearNode[lnbIx][pLnbPrevious->nNodes[lnbIx] - 1]; nodePrevious.time -= hGainDec->frameSize; if (channelGain != FL2FXCONST_DBL(1.0f / (float)(1 << 8))) nodePrevious.gainLin = SATURATE_LEFT_SHIFT( fMultDiv2(nodePrevious.gainLin, pDrcGainBuffers->channelGain[c][lnbIx]), 9, DFRACT_BITS); /* Prepare current linearNodeBuffer instance */ lnbIx++; if (lnbIx >= NUM_LNB_FRAMES) lnbIx = 0; /* if lnbIndexForChannel changes over time, use the old indices for * smooth transitions */ if (pActiveDrc->lnbIndexForChannel[c][lnbIx] >= 0) pLnb = &( pLinearNodeBuffer[pActiveDrc->lnbIndexForChannel[c][lnbIx] + b]); else /* lnbIndexForChannel = -1 means "no DRC processing", due to drcInstructionsIndex < 0, drcSetId < 0 or channel group < 0 */ pLnb = pDummyLnb; if (activeDrcIndex == hGainDec->channelGainActiveDrcIndex) channelGain = pDrcGainBuffers->channelGain[c][lnbIx]; /* number of frames of offset with respect to lnbPointer */ lnbPointerDiff = i - (NUM_LNB_FRAMES - 2); err = _processNodeSegments( hGainDec->frameSize, pLnb->gainInterpolationType, pLnb->nNodes[lnbIx], pLnb->linearNode[lnbIx], lnbPointerDiff * hGainDec->frameSize + delaySamples + offset, 1, nodePrevious, channelGain, deinterleavedAudio); if (err) return err; } deinterleavedAudio += timeDataChannelOffset; /* proceed to next channel */ } } return DE_OK; } /* process DRC on subband-domain signal */ DRC_ERROR processDrcSubband(HANDLE_DRC_GAIN_DECODER hGainDec, const int activeDrcIndex, const int delaySamples, const int channelOffset, const int drcChannelOffset, const int numChannelsProcessed, const int processSingleTimeslot, FIXP_DBL* deinterleavedAudioReal[], FIXP_DBL* deinterleavedAudioImag[]) { DRC_ERROR err = DE_OK; int b, c, g, m, m_start, m_stop, s, i; FIXP_DBL gainSb; DRC_INSTRUCTIONS_UNI_DRC* pInst = hGainDec->activeDrc[activeDrcIndex].pInst; DRC_GAIN_BUFFERS* pDrcGainBuffers = &(hGainDec->drcGainBuffers); ACTIVE_DRC* pActiveDrc = &(hGainDec->activeDrc[activeDrcIndex]); int activeDrcOffset = pActiveDrc->activeDrcOffset; int lnbPointer = pDrcGainBuffers->lnbPointer, lnbIx; LINEAR_NODE_BUFFER* pLinearNodeBuffer = pDrcGainBuffers->linearNodeBuffer; FIXP_DBL(*subbandGains)[4 * 1024 / 256] = hGainDec->subbandGains; FIXP_DBL* dummySubbandGains = hGainDec->dummySubbandGains; SUBBAND_DOMAIN_MODE subbandDomainMode = hGainDec->subbandDomainSupported; int signalIndex = 0; int frameSizeSb = 0; int nDecoderSubbands; SHORT L = 0; /* L: downsampling factor */ int offset = 0; FIXP_DBL *audioReal = NULL, *audioImag = NULL; if (hGainDec->delayMode == DM_REGULAR_DELAY) { offset = hGainDec->frameSize; } if ((delaySamples + offset) > (NUM_LNB_FRAMES - 2) * hGainDec->frameSize) /* if delaySamples is too big, NUM_LNB_FRAMES should be increased */ return DE_NOT_OK; switch (subbandDomainMode) { #if ((1024 / 256) >= (4096 / SUBBAND_DOWNSAMPLING_FACTOR_QMF64)) case SDM_QMF64: nDecoderSubbands = SUBBAND_NUM_BANDS_QMF64; L = SUBBAND_DOWNSAMPLING_FACTOR_QMF64; /* analysisDelay = SUBBAND_ANALYSIS_DELAY_QMF64; */ break; case SDM_QMF71: nDecoderSubbands = SUBBAND_NUM_BANDS_QMF71; L = SUBBAND_DOWNSAMPLING_FACTOR_QMF71; /* analysisDelay = SUBBAND_ANALYSIS_DELAY_QMF71; */ break; #else case SDM_QMF64: case SDM_QMF71: /* QMF domain processing is not supported. */ return DE_NOT_OK; #endif case SDM_STFT256: nDecoderSubbands = SUBBAND_NUM_BANDS_STFT256; L = SUBBAND_DOWNSAMPLING_FACTOR_STFT256; /* analysisDelay = SUBBAND_ANALYSIS_DELAY_STFT256; */ break; default: return DE_NOT_OK; } /* frameSizeSb = hGainDec->frameSize/L; */ /* L is a power of 2 */ frameSizeSb = hGainDec->frameSize >> (15 - fixnormz_S(L)); /* timeslots per frame */ if ((processSingleTimeslot < 0) || (processSingleTimeslot >= frameSizeSb)) { m_start = 0; m_stop = frameSizeSb; } else { m_start = processSingleTimeslot; m_stop = m_start + 1; } err = _prepareLnbIndex(pActiveDrc, channelOffset, drcChannelOffset, numChannelsProcessed, lnbPointer); if (err) return err; if (!pActiveDrc->subbandGainsReady) /* only for the first time per frame that processDrcSubband is called */ { /* write subbandGains */ for (g = 0; g < pInst->nDrcChannelGroups; g++) { b = 0; { LINEAR_NODE_BUFFER* pLnb = &(pLinearNodeBuffer[activeDrcOffset + pActiveDrc->gainElementForGroup[g] + b]); NODE_LIN nodePrevious; int lnbPointerDiff; for (m = 0; m < frameSizeSb; m++) { subbandGains[activeDrcOffset + g][b * frameSizeSb + m] = FL2FXCONST_DBL(1.0f / (float)(1 << 7)); } lnbIx = lnbPointer - (NUM_LNB_FRAMES - 1); while (lnbIx < 0) lnbIx += NUM_LNB_FRAMES; /* Loop over all node buffers in linearNodeBuffer. All nodes which are not relevant for the current frame are sorted out inside _processNodeSegments. */ for (i = 0; i < NUM_LNB_FRAMES - 1; i++) { /* Prepare previous node */ nodePrevious = pLnb->linearNode[lnbIx][pLnb->nNodes[lnbIx] - 1]; nodePrevious.time -= hGainDec->frameSize; lnbIx++; if (lnbIx >= NUM_LNB_FRAMES) lnbIx = 0; /* number of frames of offset with respect to lnbPointer */ lnbPointerDiff = i - (NUM_LNB_FRAMES - 2); err = _processNodeSegments( hGainDec->frameSize, pLnb->gainInterpolationType, pLnb->nNodes[lnbIx], pLnb->linearNode[lnbIx], lnbPointerDiff * hGainDec->frameSize + delaySamples + offset - (L - 1) / 2, L, nodePrevious, FL2FXCONST_DBL(1.0f / (float)(1 << 8)), &(subbandGains[activeDrcOffset + g][b * frameSizeSb])); if (err) return err; } } } pActiveDrc->subbandGainsReady = 1; } for (c = channelOffset; c < channelOffset + numChannelsProcessed; c++) { FIXP_DBL* thisSubbandGainsBuffer; if (pInst->drcSetId > 0) g = pActiveDrc->channelGroupForChannel[c + drcChannelOffset]; else g = -1; audioReal = deinterleavedAudioReal[signalIndex]; if (subbandDomainMode != SDM_STFT256) { audioImag = deinterleavedAudioImag[signalIndex]; } if ((g >= 0) && !pActiveDrc->channelGroupIsParametricDrc[g]) { thisSubbandGainsBuffer = subbandGains[activeDrcOffset + g]; } else { thisSubbandGainsBuffer = dummySubbandGains; } for (m = m_start; m < m_stop; m++) { INT n_min = 8; { /* single-band DRC */ gainSb = thisSubbandGainsBuffer[m]; if (activeDrcIndex == hGainDec->channelGainActiveDrcIndex) gainSb = SATURATE_LEFT_SHIFT( fMultDiv2(gainSb, hGainDec->channelGain[c]), 9, DFRACT_BITS); /* normalize gainSb for keeping signal precision */ n_min = fMin(CntLeadingZeros(gainSb) - 1, n_min); gainSb <<= n_min; n_min = 8 - n_min; if (subbandDomainMode == SDM_STFT256) { /* For STFT filterbank, real and imaginary parts are interleaved. */ for (s = 0; s < nDecoderSubbands; s++) { *audioReal = fMultDiv2(*audioReal, gainSb) << n_min; audioReal++; *audioReal = fMultDiv2(*audioReal, gainSb) << n_min; audioReal++; } } else { for (s = 0; s < nDecoderSubbands; s++) { *audioReal = fMultDiv2(*audioReal, gainSb) << n_min; audioReal++; *audioImag = fMultDiv2(*audioImag, gainSb) << n_min; audioImag++; } } } } signalIndex++; } return DE_OK; } fdk-aac-2.0.2/libDRCdec/src/drcGainDec_process.h000066400000000000000000000127001404225310700212330ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /************************* MPEG-D DRC decoder library ************************** Author(s): Description: *******************************************************************************/ #ifndef DRCGAINDEC_PROCESS_H #define DRCGAINDEC_PROCESS_H DRC_ERROR processDrcTime(HANDLE_DRC_GAIN_DECODER hGainDec, const int activeDrcIndex, const int delaySamples, const int channelOffset, const int drcChannelOffset, const int numChannelsProcessed, const int timeDataChannelOffset, FIXP_DBL* deinterleavedAudio); DRC_ERROR processDrcSubband(HANDLE_DRC_GAIN_DECODER hGainDec, const int activeDrcIndex, const int delaySamples, const int channelOffset, const int drcChannelOffset, const int numChannelsProcessed, const int processSingleTimeslot, FIXP_DBL* deinterleavedAudioReal[], FIXP_DBL* deinterleavedAudioImag[]); #endif fdk-aac-2.0.2/libFDK/000077500000000000000000000000001404225310700140725ustar00rootroot00000000000000fdk-aac-2.0.2/libFDK/include/000077500000000000000000000000001404225310700155155ustar00rootroot00000000000000fdk-aac-2.0.2/libFDK/include/FDK_archdef.h000066400000000000000000000223531404225310700177530ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Description: *******************************************************************************/ #ifndef FDK_ARCHDEF_H #define FDK_ARCHDEF_H /* Unify some few toolchain specific defines to avoid having large "or" macro * contraptions all over the source code. */ /* Use single macro (the GCC built in macro) for architecture identification * independent of the particular toolchain */ #if defined(__i386__) || defined(__i486__) || defined(__i586__) || \ defined(__i686__) || (defined(_MSC_VER) && defined(_M_IX86)) || \ (defined(_MSC_VER) && defined(_M_X64)) || defined(__x86_64__) #define __x86__ #endif #if defined(_M_ARM) && !defined(__arm__) || defined(__aarch64__) || defined(_M_ARM64) #define __arm__ #endif #if defined(_ARCH_PPC) && !defined(__powerpc__) #define __powerpc__ 1 #endif #if (__TARGET_ARCH_ARM == 5) || defined(__TARGET_FEATURE_DSPMUL) || \ (_M_ARM == 5) || defined(__ARM_ARCH_5TEJ__) || defined(__ARM_ARCH_7EM__) /* Define __ARM_ARCH_5TE__ if armv5te features are supported */ #define __ARM_ARCH_5TE__ #endif #if (__TARGET_ARCH_ARM == 6) || defined(__ARM_ARCH_6J__) || \ defined(__ARM_ARCH_6ZK__) /* Define __ARM_ARCH_6__ if the armv6 intructions are being supported. */ #define __ARM_ARCH_5TE__ #define __ARM_ARCH_6__ #endif #if defined(__TARGET_ARCH_7_R) || defined(__ARM_ARCH_7R__) /* Define __ARM_ARCH_7_A__ if the armv7 intructions are being supported. */ #define __ARM_ARCH_5TE__ #define __ARM_ARCH_6__ #define __ARM_ARCH_7_R__ #endif #if defined(__TARGET_ARCH_7_A) || defined(__ARM_ARCH_7A__) || \ ((__ARM_ARCH == 8) && (__ARM_32BIT_STATE == 1)) /* Define __ARM_ARCH_7_A__ if the armv7 intructions are being supported. */ #define __ARM_ARCH_5TE__ #define __ARM_ARCH_6__ #define __ARM_ARCH_7_A__ #endif #if defined(__TARGET_ARCH_7_M) || defined(__ARM_ARCH_7_M__) /* Define __ARM_ARCH_7M__ if the ARMv7-M instructions are being supported, e.g. * Cortex-M3. */ #define __ARM_ARCH_7M__ #endif #if defined(__TARGET_ARCH_7E_M) || defined(__ARM_ARCH_7E_M__) /* Define __ARM_ARCH_7EM__ if the ARMv7-ME instructions are being supported, * e.g. Cortex-M4. */ #define __ARM_ARCH_7EM__ #endif #if defined(__aarch64__) || defined(_M_ARM64) #define __ARM_ARCH_8__ #endif #ifdef _M_ARM #include "armintr.h" #endif /* Define preferred Multiplication type */ #if defined(__mips__) #define ARCH_PREFER_MULT_16x16 #undef SINETABLE_16BIT #undef POW2COEFF_16BIT #undef LDCOEFF_16BIT #undef WINDOWTABLE_16BIT #elif defined(__arm__) && defined(__ARM_ARCH_8__) #define ARCH_PREFER_MULT_32x16 #define SINETABLE_16BIT #define POW2COEFF_16BIT #define LDCOEFF_16BIT #define WINDOWTABLE_16BIT #elif defined(__arm__) && defined(__ARM_ARCH_5TE__) #define ARCH_PREFER_MULT_32x16 #define SINETABLE_16BIT #define POW2COEFF_16BIT #define LDCOEFF_16BIT #define WINDOWTABLE_16BIT #elif defined(__arm__) && defined(__ARM_ARCH_7M__) #define ARCH_PREFER_MULT_32x16 #define SINETABLE_16BIT #define POW2COEFF_16BIT #define LDCOEFF_16BIT #define WINDOWTABLE_16BIT #elif defined(__arm__) && defined(__ARM_ARCH_7EM__) #define ARCH_PREFER_MULT_32x32 #define ARCH_PREFER_MULT_32x16 #define SINETABLE_16BIT #define POW2COEFF_16BIT #define LDCOEFF_16BIT #define WINDOWTABLE_16BIT #elif defined(__arm__) && !defined(__ARM_ARCH_5TE__) #define ARCH_PREFER_MULT_16x16 #undef SINETABLE_16BIT #undef WINDOWTABLE_16BIT #undef POW2COEFF_16BIT #undef LDCOEFF_16BIT #elif defined(__x86__) #define ARCH_PREFER_MULT_32x16 #define SINETABLE_16BIT #define WINDOWTABLE_16BIT #define POW2COEFF_16BIT #define LDCOEFF_16BIT #elif defined(__powerpc__) #define ARCH_PREFER_MULT_32x32 #define ARCH_PREFER_MULT_32x16 #define SINETABLE_16BIT #define POW2COEFF_16BIT #define LDCOEFF_16BIT #define WINDOWTABLE_16BIT #else #warning >>>> Please set architecture characterization defines for your platform (FDK_HIGH_PERFORMANCE)! <<<< #endif /* Architecture switches */ #ifdef SINETABLE_16BIT #define FIXP_STB FIXP_SGL /* STB sinus Tab used in transformation */ #define FIXP_STP FIXP_SPK #define STC(a) (FX_DBL2FXCONST_SGL(a)) #else #define FIXP_STB FIXP_DBL #define FIXP_STP FIXP_DPK #define STC(a) ((FIXP_DBL)(LONG)(a)) #endif /* defined(SINETABLE_16BIT) */ #define STCP(cos, sin) \ { \ { STC(cos), STC(sin) } \ } #ifdef WINDOWTABLE_16BIT #define FIXP_WTB FIXP_SGL /* single FIXP_SGL values */ #define FX_DBL2FX_WTB(x) FX_DBL2FX_SGL(x) #define FIXP_WTP FIXP_SPK /* packed FIXP_SGL values */ #define WTC(a) FX_DBL2FXCONST_SGL(a) #else /* SINETABLE_16BIT */ #define FIXP_WTB FIXP_DBL #define FX_DBL2FX_WTB(x) (x) #define FIXP_WTP FIXP_DPK #define WTC(a) (FIXP_DBL)(a) #endif /* SINETABLE_16BIT */ #define WTCP(a, b) \ { \ { WTC(a), WTC(b) } \ } #endif /* FDK_ARCHDEF_H */ fdk-aac-2.0.2/libFDK/include/FDK_bitbuffer.h000066400000000000000000000153321404225310700203260ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): M. Lohwasser Description: common bitbuffer read/write routines *******************************************************************************/ #ifndef FDK_BITBUFFER_H #define FDK_BITBUFFER_H #include "FDK_archdef.h" #include "machine_type.h" /* leave 3 bits headroom so MAX_BUFSIZE can be represented in bits as well. */ #define MAX_BUFSIZE_BYTES (0x10000000) typedef struct { UINT ValidBits; UINT ReadOffset; UINT WriteOffset; UINT BitNdx; UCHAR *Buffer; UINT bufSize; UINT bufBits; } FDK_BITBUF; typedef FDK_BITBUF *HANDLE_FDK_BITBUF; #ifdef __cplusplus extern "C" { #endif extern const UINT BitMask[32 + 1]; /** The BitBuffer Functions are called straight from FDK_bitstream Interface. For Functions functional survey look there. */ void FDK_CreateBitBuffer(HANDLE_FDK_BITBUF *hBitBuffer, UCHAR *pBuffer, UINT bufSize); void FDK_InitBitBuffer(HANDLE_FDK_BITBUF hBitBuffer, UCHAR *pBuffer, UINT bufSize, UINT validBits); void FDK_ResetBitBuffer(HANDLE_FDK_BITBUF hBitBuffer); void FDK_DeleteBitBuffer(HANDLE_FDK_BITBUF hBitBuffer); INT FDK_get(HANDLE_FDK_BITBUF hBitBuffer, const UINT numberOfBits); INT FDK_get32(HANDLE_FDK_BITBUF hBitBuf); void FDK_put(HANDLE_FDK_BITBUF hBitBuffer, UINT value, const UINT numberOfBits); INT FDK_getBwd(HANDLE_FDK_BITBUF hBitBuffer, const UINT numberOfBits); void FDK_putBwd(HANDLE_FDK_BITBUF hBitBuffer, UINT value, const UINT numberOfBits); void FDK_pushBack(HANDLE_FDK_BITBUF hBitBuffer, const UINT numberOfBits, UCHAR config); void FDK_pushForward(HANDLE_FDK_BITBUF hBitBuffer, const UINT numberOfBits, UCHAR config); UINT FDK_getValidBits(HANDLE_FDK_BITBUF hBitBuffer); INT FDK_getFreeBits(HANDLE_FDK_BITBUF hBitBuffer); void FDK_Feed(HANDLE_FDK_BITBUF hBitBuffer, const UCHAR inputBuffer[], const UINT bufferSize, UINT *bytesValid); void FDK_Copy(HANDLE_FDK_BITBUF hBitBufDst, HANDLE_FDK_BITBUF hBitBufSrc, UINT *bytesValid); void FDK_Fetch(HANDLE_FDK_BITBUF hBitBuffer, UCHAR outBuf[], UINT *writeBytes); #ifdef __cplusplus } #endif #endif fdk-aac-2.0.2/libFDK/include/FDK_bitstream.h000066400000000000000000000546411404225310700203560ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): M. Lohwasser Description: bitstream interface to bitbuffer routines *******************************************************************************/ #ifndef FDK_BITSTREAM_H #define FDK_BITSTREAM_H #include "FDK_bitbuffer.h" #include "machine_type.h" #include "genericStds.h" #define CACHE_BITS 32 #define BUFSIZE_DUMMY_VALUE MAX_BUFSIZE_BYTES typedef enum { BS_READER, BS_WRITER } FDK_BS_CFG; typedef struct { UINT CacheWord; UINT BitsInCache; FDK_BITBUF hBitBuf; UINT ConfigCache; } FDK_BITSTREAM; typedef FDK_BITSTREAM *HANDLE_FDK_BITSTREAM; /** * \brief CreateBitStream Function. * * Create and initialize bitstream with extern allocated buffer. * * \param pBuffer Pointer to BitBuffer array. * \param bufSize Length of BitBuffer array. (awaits size 2^n and <= * MAX_BUFSIZE_BYTES) * \param config Initialize BitStream as Reader or Writer. */ FDK_INLINE HANDLE_FDK_BITSTREAM FDKcreateBitStream(UCHAR *pBuffer, UINT bufSize, FDK_BS_CFG config = BS_READER) { HANDLE_FDK_BITSTREAM hBitStream = (HANDLE_FDK_BITSTREAM)FDKcalloc(1, sizeof(FDK_BITSTREAM)); if (hBitStream == NULL) return NULL; FDK_InitBitBuffer(&hBitStream->hBitBuf, pBuffer, bufSize, 0); /* init cache */ hBitStream->CacheWord = hBitStream->BitsInCache = 0; hBitStream->ConfigCache = config; return hBitStream; } /** * \brief Initialize BistreamBuffer. BitBuffer can point to filled BitBuffer * array . * * \param hBitStream HANDLE_FDK_BITSTREAM handle * \param pBuffer Pointer to BitBuffer array. * \param bufSize Length of BitBuffer array in bytes. (awaits size 2^n and <= * MAX_BUFSIZE_BYTES) * \param validBits Number of valid BitBuffer filled Bits. * \param config Initialize BitStream as Reader or Writer. * \return void */ FDK_INLINE void FDKinitBitStream(HANDLE_FDK_BITSTREAM hBitStream, UCHAR *pBuffer, UINT bufSize, UINT validBits, FDK_BS_CFG config = BS_READER) { FDK_InitBitBuffer(&hBitStream->hBitBuf, pBuffer, bufSize, validBits); /* init cache */ hBitStream->CacheWord = hBitStream->BitsInCache = 0; hBitStream->ConfigCache = config; } /** * \brief ResetBitbuffer Function. Reset states in BitBuffer and Cache. * * \param hBitStream HANDLE_FDK_BITSTREAM handle * \param config Initialize BitStream as Reader or Writer. * \return void */ FDK_INLINE void FDKresetBitbuffer(HANDLE_FDK_BITSTREAM hBitStream, FDK_BS_CFG config = BS_READER) { FDK_ResetBitBuffer(&hBitStream->hBitBuf); /* init cache */ hBitStream->CacheWord = hBitStream->BitsInCache = 0; hBitStream->ConfigCache = config; } /** DeleteBitStream. Deletes the in Create Bitstream allocated BitStream and BitBuffer. */ FDK_INLINE void FDKdeleteBitStream(HANDLE_FDK_BITSTREAM hBitStream) { FDK_DeleteBitBuffer(&hBitStream->hBitBuf); FDKfree(hBitStream); } /** * \brief ReadBits Function (forward). This function returns a number of * sequential bits from the input bitstream. * * \param hBitStream HANDLE_FDK_BITSTREAM handle * \param numberOfBits The number of bits to be retrieved. ( (0),1 <= * numberOfBits <= 32) * \return the requested bits, right aligned * \return */ FDK_INLINE UINT FDKreadBits(HANDLE_FDK_BITSTREAM hBitStream, const UINT numberOfBits) { UINT bits = 0; INT missingBits = (INT)numberOfBits - (INT)hBitStream->BitsInCache; FDK_ASSERT(numberOfBits <= 32); if (missingBits > 0) { if (missingBits != 32) bits = hBitStream->CacheWord << missingBits; hBitStream->CacheWord = FDK_get32(&hBitStream->hBitBuf); hBitStream->BitsInCache += CACHE_BITS; } hBitStream->BitsInCache -= numberOfBits; return (bits | (hBitStream->CacheWord >> hBitStream->BitsInCache)) & BitMask[numberOfBits]; } FDK_INLINE UINT FDKreadBit(HANDLE_FDK_BITSTREAM hBitStream) { if (!hBitStream->BitsInCache) { hBitStream->CacheWord = FDK_get32(&hBitStream->hBitBuf); hBitStream->BitsInCache = CACHE_BITS - 1; return hBitStream->CacheWord >> 31; } hBitStream->BitsInCache--; return (hBitStream->CacheWord >> hBitStream->BitsInCache) & 1; } /** * \brief Read2Bits Function (forward). This function reads 2 sequential * bits from the input bitstream. It is the optimized version of FDKreadBits() for reading 2 bits. * * \param hBitStream HANDLE_FDK_BITSTREAM handle * \return the requested bits, right aligned * \return */ FDK_INLINE UINT FDKread2Bits(HANDLE_FDK_BITSTREAM hBitStream) { /* ** Version corresponds to optimized FDKreadBits implementation ** calling FDK_get32, that keeps read pointer aligned. */ UINT bits = 0; INT missingBits = 2 - (INT)hBitStream->BitsInCache; if (missingBits > 0) { bits = hBitStream->CacheWord << missingBits; hBitStream->CacheWord = FDK_get32(&hBitStream->hBitBuf); hBitStream->BitsInCache += CACHE_BITS; } hBitStream->BitsInCache -= 2; return (bits | (hBitStream->CacheWord >> hBitStream->BitsInCache)) & 0x3; } /** * \brief ReadBits Function (backward). This function returns a number of * sequential bits from the input bitstream. * * \param hBitStream HANDLE_FDK_BITSTREAM handle * \param numberOfBits The number of bits to be retrieved. * \return the requested bits, right aligned */ FDK_INLINE UINT FDKreadBitsBwd(HANDLE_FDK_BITSTREAM hBitStream, const UINT numberOfBits) { const UINT validMask = BitMask[numberOfBits]; if (hBitStream->BitsInCache <= numberOfBits) { const INT freeBits = (CACHE_BITS - 1) - hBitStream->BitsInCache; hBitStream->CacheWord = (hBitStream->CacheWord << freeBits) | FDK_getBwd(&hBitStream->hBitBuf, freeBits); hBitStream->BitsInCache += freeBits; } hBitStream->BitsInCache -= numberOfBits; return (hBitStream->CacheWord >> hBitStream->BitsInCache) & validMask; } /** * \brief read an integer value using a varying number of bits from the * bitstream * * q.v. ISO/IEC FDIS 23003-3 Table 16 * * \param hBitStream HANDLE_FDK_BITSTREAM handle * \param nBits1 number of bits to read for a small integer value or escape * value * \param nBits2 number of bits to read for a medium sized integer value or * escape value * \param nBits3 number of bits to read for a large integer value * \return integer value read from bitstream */ FDK_INLINE UINT escapedValue(HANDLE_FDK_BITSTREAM hBitStream, int nBits1, int nBits2, int nBits3) { UINT value = FDKreadBits(hBitStream, nBits1); if (value == (UINT)(1 << nBits1) - 1) { UINT valueAdd = FDKreadBits(hBitStream, nBits2); value += valueAdd; if (valueAdd == (UINT)(1 << nBits2) - 1) { value += FDKreadBits(hBitStream, nBits3); } } return value; } /** * \brief return a number of bits from the bitBuffer. * You have to know what you do! Cache has to be synchronized before * using this function. * * \param hBitStream HANDLE_FDK_BITSTREAM handle * \param numBits The number of bits to be retrieved. * \return the requested bits, right aligned */ FDK_INLINE UINT FDKgetBits(HANDLE_FDK_BITSTREAM hBitStream, UINT numBits) { return FDK_get(&hBitStream->hBitBuf, numBits); } /** * \brief WriteBits Function. This function writes numberOfBits of value into * bitstream. * * \param hBitStream HANDLE_FDK_BITSTREAM handle * \param value The data to be written * \param numberOfBits The number of bits to be written * \return Number of bits written */ FDK_INLINE UCHAR FDKwriteBits(HANDLE_FDK_BITSTREAM hBitStream, UINT value, const UINT numberOfBits) { const UINT validMask = BitMask[numberOfBits]; if (hBitStream == NULL) { return numberOfBits; } if ((hBitStream->BitsInCache + numberOfBits) < CACHE_BITS) { hBitStream->BitsInCache += numberOfBits; hBitStream->CacheWord = (hBitStream->CacheWord << numberOfBits) | (value & validMask); } else { /* Put always 32 bits into memory */ /* - fill cache's LSBits with MSBits of value */ /* - store 32 bits in memory using subroutine */ /* - fill remaining bits into cache's LSBits */ /* - upper bits in cache are don't care */ /* Compute number of bits to be filled into cache */ int missing_bits = CACHE_BITS - hBitStream->BitsInCache; int remaining_bits = numberOfBits - missing_bits; value = value & validMask; /* Avoid shift left by 32 positions */ UINT CacheWord = (missing_bits == 32) ? 0 : (hBitStream->CacheWord << missing_bits); CacheWord |= (value >> (remaining_bits)); FDK_put(&hBitStream->hBitBuf, CacheWord, 32); hBitStream->CacheWord = value; hBitStream->BitsInCache = remaining_bits; } return numberOfBits; } /** * \brief WriteBits Function (backward). This function writes numberOfBits of * value into bitstream. * * \param hBitStream HANDLE_FDK_BITSTREAM handle * \param value Variable holds data to be written. * \param numberOfBits The number of bits to be written. * \return number of bits written */ FDK_INLINE UCHAR FDKwriteBitsBwd(HANDLE_FDK_BITSTREAM hBitStream, UINT value, const UINT numberOfBits) { const UINT validMask = BitMask[numberOfBits]; if ((hBitStream->BitsInCache + numberOfBits) <= CACHE_BITS) { hBitStream->BitsInCache += numberOfBits; hBitStream->CacheWord = (hBitStream->CacheWord << numberOfBits) | (value & validMask); } else { FDK_putBwd(&hBitStream->hBitBuf, hBitStream->CacheWord, hBitStream->BitsInCache); hBitStream->BitsInCache = numberOfBits; hBitStream->CacheWord = (value & validMask); } return numberOfBits; } /** * \brief write an integer value using a varying number of bits from the * bitstream * * q.v. ISO/IEC FDIS 23003-3 Table 16 * * \param hBitStream HANDLE_FDK_BITSTREAM handle * \param value the data to be written * \param nBits1 number of bits to write for a small integer value or escape * value * \param nBits2 number of bits to write for a medium sized integer value or * escape value * \param nBits3 number of bits to write for a large integer value * \return number of bits written */ FDK_INLINE UCHAR FDKwriteEscapedValue(HANDLE_FDK_BITSTREAM hBitStream, UINT value, UINT nBits1, UINT nBits2, UINT nBits3) { UCHAR nbits = 0; UINT tmp = (1 << nBits1) - 1; if (value < tmp) { nbits += FDKwriteBits(hBitStream, value, nBits1); } else { nbits += FDKwriteBits(hBitStream, tmp, nBits1); value -= tmp; tmp = (1 << nBits2) - 1; if (value < tmp) { nbits += FDKwriteBits(hBitStream, value, nBits2); } else { nbits += FDKwriteBits(hBitStream, tmp, nBits2); value -= tmp; nbits += FDKwriteBits(hBitStream, value, nBits3); } } return nbits; } /** * \brief SyncCache Function. Clear cache after read forward. * * \param hBitStream HANDLE_FDK_BITSTREAM handle * \return void */ FDK_INLINE void FDKsyncCache(HANDLE_FDK_BITSTREAM hBitStream) { if (hBitStream->ConfigCache == BS_READER) FDK_pushBack(&hBitStream->hBitBuf, hBitStream->BitsInCache, hBitStream->ConfigCache); else if (hBitStream->BitsInCache) /* BS_WRITER */ FDK_put(&hBitStream->hBitBuf, hBitStream->CacheWord, hBitStream->BitsInCache); hBitStream->BitsInCache = 0; hBitStream->CacheWord = 0; } /** * \brief SyncCache Function. Clear cache after read backwards. * * \param hBitStream HANDLE_FDK_BITSTREAM handle * \return void */ FDK_INLINE void FDKsyncCacheBwd(HANDLE_FDK_BITSTREAM hBitStream) { if (hBitStream->ConfigCache == BS_READER) { FDK_pushForward(&hBitStream->hBitBuf, hBitStream->BitsInCache, hBitStream->ConfigCache); } else { /* BS_WRITER */ FDK_putBwd(&hBitStream->hBitBuf, hBitStream->CacheWord, hBitStream->BitsInCache); } hBitStream->BitsInCache = 0; hBitStream->CacheWord = 0; } /** * \brief Byte Alignment Function with anchor * This function performs the byte_alignment() syntactic function on the * input stream, i.e. some bits will be discarded so that the next bits to be * read/written would be aligned on a byte boundary with respect to the * given alignment anchor. * * \param hBitStream HANDLE_FDK_BITSTREAM handle * \param alignmentAnchor bit position to be considered as origin for byte * alignment * \return void */ FDK_INLINE void FDKbyteAlign(HANDLE_FDK_BITSTREAM hBitStream, UINT alignmentAnchor) { FDKsyncCache(hBitStream); if (hBitStream->ConfigCache == BS_READER) { FDK_pushForward( &hBitStream->hBitBuf, (UINT)((INT)8 - (((INT)alignmentAnchor - (INT)FDK_getValidBits(&hBitStream->hBitBuf)) & 0x07)) & 0x07, hBitStream->ConfigCache); } else { FDK_put(&hBitStream->hBitBuf, 0, (8 - ((FDK_getValidBits(&hBitStream->hBitBuf) - alignmentAnchor) & 0x07)) & 0x07); } } /** * \brief Push Back(Cache) / For / BiDirectional Function. * PushBackCache function ungets a number of bits erroneously * read/written by the last Get() call. NB: The number of bits to be stuffed * back into the stream may never exceed the number of bits returned by * the immediately preceding Get() call. * * PushBack function ungets a number of bits (combines cache and bitbuffer * indices) PushFor function gets a number of bits (combines cache and * bitbuffer indices) PushBiDirectional gets/ungets number of bits as * defined in PusBack/For function NB: The sign of bits is not known, so * the function checks direction and calls appropriate function. (positive * sign pushFor, negative sign pushBack ) * * \param hBitStream HANDLE_FDK_BITSTREAM handle * \param numberOfBits The number of bits to be pushed back/for. * \return void */ FDK_INLINE void FDKpushBackCache(HANDLE_FDK_BITSTREAM hBitStream, const UINT numberOfBits) { FDK_ASSERT((hBitStream->BitsInCache + numberOfBits) <= CACHE_BITS); hBitStream->BitsInCache += numberOfBits; } FDK_INLINE void FDKpushBack(HANDLE_FDK_BITSTREAM hBitStream, const UINT numberOfBits) { if ((hBitStream->BitsInCache + numberOfBits) < CACHE_BITS && (hBitStream->ConfigCache == BS_READER)) { hBitStream->BitsInCache += numberOfBits; FDKsyncCache(hBitStream); /* sync cache to avoid invalid cache */ } else { FDKsyncCache(hBitStream); FDK_pushBack(&hBitStream->hBitBuf, numberOfBits, hBitStream->ConfigCache); } } FDK_INLINE void FDKpushFor(HANDLE_FDK_BITSTREAM hBitStream, const UINT numberOfBits) { if ((hBitStream->BitsInCache > numberOfBits) && (hBitStream->ConfigCache == BS_READER)) { hBitStream->BitsInCache -= numberOfBits; } else { FDKsyncCache(hBitStream); FDK_pushForward(&hBitStream->hBitBuf, numberOfBits, hBitStream->ConfigCache); } } FDK_INLINE void FDKpushBiDirectional(HANDLE_FDK_BITSTREAM hBitStream, const INT numberOfBits) { if (numberOfBits >= 0) FDKpushFor(hBitStream, numberOfBits); else FDKpushBack(hBitStream, -numberOfBits); } /** * \brief GetValidBits Function. Clear cache and return valid Bits from * Bitbuffer. * \param hBitStream HANDLE_FDK_BITSTREAM handle * \return amount of valid bits that still can be read or were already written. * */ FDK_INLINE UINT FDKgetValidBits(HANDLE_FDK_BITSTREAM hBitStream) { FDKsyncCache(hBitStream); return FDK_getValidBits(&hBitStream->hBitBuf); } /** * \brief return amount of unused Bits from Bitbuffer. * \param hBitStream HANDLE_FDK_BITSTREAM handle * \return amount of free bits that still can be written into the bitstream */ FDK_INLINE INT FDKgetFreeBits(HANDLE_FDK_BITSTREAM hBitStream) { return FDK_getFreeBits(&hBitStream->hBitBuf); } /** * \brief Fill the BitBuffer with a number of input bytes from external source. * The bytesValid variable returns the number of ramaining valid bytes in * extern inputBuffer. * * \param hBitStream HANDLE_FDK_BITSTREAM handle * \param inputBuffer Pointer to input buffer with bitstream data. * \param bufferSize Total size of inputBuffer array. * \param bytesValid Input: number of valid bytes in inputBuffer. Output: bytes * still left unread in inputBuffer. * \return void */ FDK_INLINE void FDKfeedBuffer(HANDLE_FDK_BITSTREAM hBitStream, const UCHAR inputBuffer[], const UINT bufferSize, UINT *bytesValid) { FDKsyncCache(hBitStream); FDK_Feed(&hBitStream->hBitBuf, inputBuffer, bufferSize, bytesValid); } /** * \brief fill destination BitBuffer with a number of bytes from source * BitBuffer. The bytesValid variable returns the number of ramaining valid * bytes in source BitBuffer. * * \param hBSDst HANDLE_FDK_BITSTREAM handle to write data into * \param hBSSrc HANDLE_FDK_BITSTREAM handle to read data from * \param bytesValid Input: number of valid bytes in inputBuffer. Output: * bytes still left unread in inputBuffer. * \return void */ FDK_INLINE void FDKcopyBuffer(HANDLE_FDK_BITSTREAM hBSDst, HANDLE_FDK_BITSTREAM hBSSrc, UINT *bytesValid) { FDKsyncCache(hBSSrc); FDK_Copy(&hBSDst->hBitBuf, &hBSSrc->hBitBuf, bytesValid); } /** * \brief fill the outputBuffer with all valid bytes hold in BitBuffer. The * WriteBytes variable returns the number of written Bytes. * * \param hBitStream HANDLE_FDK_BITSTREAM handle * \param outputBuffer Pointer to output buffer. * \param writeBytes Number of bytes write to output buffer. * \return void */ FDK_INLINE void FDKfetchBuffer(HANDLE_FDK_BITSTREAM hBitStream, UCHAR *outputBuffer, UINT *writeBytes) { FDKsyncCache(hBitStream); FDK_Fetch(&hBitStream->hBitBuf, outputBuffer, writeBytes); } #endif fdk-aac-2.0.2/libFDK/include/FDK_core.h000066400000000000000000000120701404225310700173020ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Manuel Jander Description: FDK tools versioning support *******************************************************************************/ #ifndef FDK_CORE_H #define FDK_CORE_H #include "FDK_audio.h" #ifdef __cplusplus extern "C" { #endif /** @brief Get FDK_tools library information. * @return Return 0 on success and a negative errorcode on failure (see * errorcodes.h). */ int FDK_toolsGetLibInfo(LIB_INFO *info); #ifdef __cplusplus } #endif #endif fdk-aac-2.0.2/libFDK/include/FDK_crc.h000066400000000000000000000211121404225310700171160ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Description: CRC calculation *******************************************************************************/ #ifndef FDK_CRC_H #define FDK_CRC_H #include "FDK_bitstream.h" #define MAX_CRC_REGS \ 3 /*!< Maximal number of overlapping crc region in ADTS channel pair element \ is two. Select three independent regions preventively. */ /** * This structure describes single crc region used for crc calculation. */ typedef struct { UCHAR isActive; INT maxBits; INT bitBufCntBits; INT validBits; } CCrcRegData; /** * CRC info structure. */ typedef struct { CCrcRegData crcRegData[MAX_CRC_REGS]; /*!< Multiple crc region description. */ const USHORT* pCrcLookup; /*!< Pointer to lookup table filled in FDK_crcInit(). */ USHORT crcPoly; /*!< CRC generator polynom. */ USHORT crcMask; /*!< CRC mask. */ USHORT startValue; /*!< CRC start value. */ UCHAR crcLen; /*!< CRC length. */ UINT regStart; /*!< Start region marker for synchronization. */ UINT regStop; /*!< Stop region marker for synchronization. */ USHORT crcValue; /*!< Crc value to be calculated. */ } FDK_CRCINFO; /** * CRC info handle. */ typedef FDK_CRCINFO* HANDLE_FDK_CRCINFO; /** * \brief Initialize CRC structure. * * The function initializes existing crc info structure with denoted * configuration. * * \param hCrcInfo Pointer to an outlying allocated crc info * structure. * \param crcPoly Configure crc polynom. * \param crcStartValue Configure crc start value. * \param crcLen Configure crc length. * * \return none */ void FDKcrcInit(HANDLE_FDK_CRCINFO hCrcInfo, const UINT crcPoly, const UINT crcStartValue, const UINT crcLen); /** * \brief Reset CRC info structure. * * This function clears all intern states of the crc structure. * * \param hCrcInfo Pointer to crc info stucture. * * \return none */ void FDKcrcReset(HANDLE_FDK_CRCINFO hCrcInfo); /** * \brief Start CRC region with maximum number of bits. * * This function marks position in bitstream to be used as start point for crc * calculation. Bitstream range for crc calculation can be limited or kept * dynamic depending on mBits parameter. The crc region has to be terminated * with FDKcrcEndReg() in each case. * * \param hCrcInfo Pointer to crc info stucture. * \param hBs Pointer to current bit buffer structure. * \param mBits Number of bits in crc region to be calculated. * - mBits > 0: Zero padding will be used for CRC * calculation, if there are less than mBits bits available. * - mBits < 0: No zero padding is done. * - mBits = 0: The number of bits used in crc * calculation is dynamically, depending on bitstream position between * FDKcrcStartReg() and FDKcrcEndReg() * call. * * \return ID for the created region, -1 in case of an error */ INT FDKcrcStartReg(HANDLE_FDK_CRCINFO hCrcInfo, const HANDLE_FDK_BITSTREAM hBs, const INT mBits); /** * \brief Ends CRC region. * * This function terminates crc region specified with FDKcrcStartReg(). The * number of bits in crc region depends on mBits parameter of FDKcrcStartReg(). * This function calculates and updates crc in info structure. * * \param hCrcInfo Pointer to crc info stucture. * \param hBs Pointer to current bit buffer structure. * \param reg Crc region ID created in FDKcrcStartReg(). * * \return 0 on success */ INT FDKcrcEndReg(HANDLE_FDK_CRCINFO hCrcInfo, const HANDLE_FDK_BITSTREAM hBs, const INT reg); /** * \brief This function returns crc value from info struct. * * \param hCrcInfo Pointer to crc info stucture. * * \return CRC value masked with crc length. */ USHORT FDKcrcGetCRC(const HANDLE_FDK_CRCINFO hCrcInfo); #endif /* FDK_CRC_H */ fdk-aac-2.0.2/libFDK/include/FDK_decorrelate.h000066400000000000000000000266371404225310700206610ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Markus Lohwasser Description: FDK Tools Decorrelator *******************************************************************************/ #ifndef FDK_DECORRELATE_H #define FDK_DECORRELATE_H #include "common_fix.h" #define FIXP_MPS FIXP_DBL #ifndef ARCH_PREFER_MULT_32x32 #define FIXP_DECORR FIXP_SGL #define FX_DECORR2FX_DBL FX_SGL2FX_DBL #define FX_DECORR2FX_SGL #define FX_DBL2FX_DECORR FX_DBL2FX_SGL #define FX_SGL2FX_DECORR #define DECORR(a) (FX_DBL2FXCONST_SGL(a)) #define FL2FXCONST_DECORR FL2FXCONST_SGL #else #define FIXP_DECORR FIXP_DBL #define FX_DECORR2FX_DBL #define FX_DECORR2FX_SGL FX_DBL2FX_SGL #define FX_DBL2FX_DECORR #define FX_SGL2FX_DECORR FX_SGL2FX_DBL #define DECORR(a) FIXP_DBL(a) #define FL2FXCONST_DECORR FL2FXCONST_DBL #endif /*--------------- enums -------------------------------*/ /** * Decorrelator types. */ typedef enum { DECORR_MPS, /**< Decorrelator type used by MPS LP/HQ */ DECORR_PS, /**< Decorrelator type used by HEAACv2 and MPS LP */ DECORR_USAC, /**< Decorrelator type used by USAC */ DECORR_LD /**< Decorrelator type used by MPS Low Delay */ } FDK_DECORR_TYPE; /** * Ducker types. */ typedef enum { DUCKER_AUTOMATIC, /**< FDKdecorrelateInit() chooses correct ducker type depending on provided parameters. */ DUCKER_MPS, /**< Force ducker type to MPS. */ DUCKER_PS /**< Force ducker type to PS. */ } FDK_DUCKER_TYPE; /** * Reverb band types. */ typedef enum { NOT_EXIST, /**< Mark reverb band as non-existing (number of bands = 0). */ DELAY, /**< Reverb bands just contains delay elements and no allpass filters. */ COMMON_REAL, /**< Real filter coeffs, common filter coeffs within one reverb band */ COMMON_CPLX, /**< Complex filter coeffs, common filter coeffs within one reverb band */ INDEP_CPLX, /**< Complex filter coeffs, independent filter coeffs for each hybrid band */ INDEP_CPLX_PS /**< PS optimized implementation of general INDEP_CPLX type */ } REVBAND_FILT_TYPE; typedef struct DECORR_DEC *HANDLE_DECORR_DEC; typedef struct DUCKER_INSTANCE { int hybridBands; int parameterBands; int partiallyComplex; FDK_DUCKER_TYPE duckerType; const UCHAR *qs_next; const UCHAR *mapProcBands2HybBands; const UCHAR *mapHybBands2ProcBands; /* interleaved SmoothDirectNrg[] and SmoothReverbNrg[], non-interleaved SmoothDirectNrg[] in case of parametric stereo */ FIXP_MPS SmoothDirRevNrg[2 * (28)]; /* parametric stereo */ FIXP_MPS peakDecay[(28)]; FIXP_MPS peakDiff[(28)]; FIXP_DBL maxValDirectData; FIXP_DBL maxValReverbData; SCHAR scaleDirectNrg; SCHAR scaleReverbNrg; SCHAR scaleSmoothDirRevNrg; SCHAR headroomSmoothDirRevNrg; } DUCKER_INSTANCE; typedef struct DECORR_FILTER_INSTANCE { FIXP_MPS *stateCplx; FIXP_DBL *DelayBufferCplx; const FIXP_DECORR *numeratorReal; const FIXP_STP *coeffsPacked; const FIXP_DECORR *denominatorReal; } DECORR_FILTER_INSTANCE; typedef struct DECORR_DEC { INT L_stateBufferCplx; FIXP_DBL *stateBufferCplx; INT L_delayBufferCplx; FIXP_DBL *delayBufferCplx; const REVBAND_FILT_TYPE *REV_filtType; const UCHAR *REV_bandOffset; const UCHAR *REV_delay; const SCHAR *REV_filterOrder; INT reverbBandDelayBufferIndex[(4)]; UCHAR stateBufferOffset[(3)]; DECORR_FILTER_INSTANCE Filter[(71)]; DUCKER_INSTANCE ducker; int numbins; int partiallyComplex; } DECORR_DEC; /** * \brief Create one instance of Decorrelator. * * \param hDecorrDec A pointer to a decorrelator instance which was * allocated externally. * \param bufferCplx Externally allocated buffer (allocate (2*( ( 825 ) * + ( 373 ) )) FIXP_DBL values). * \param bufLen Length of bufferCplx. Must be >= (2*( ( 825 ) + ( * 373 ) )). * * \return 0 on success. */ INT FDKdecorrelateOpen(HANDLE_DECORR_DEC hDecorrDec, FIXP_DBL *bufferCplx, const INT bufLen); /** * \brief Initialize and configure Decorrelator instance. * * \param hDecorrDec A Decorrelator handle. * \param nrHybBands Number of (hybrid) bands. * \param decorrType Decorrelator type to use. * \param duckerType Ducker type to use (in general use * DUCKER_AUTOMATIC). * \param decorrConfig Depending on decorrType values of 0,1,2 are * allowed. * \param seed Seed of decorrelator instance. Allowed maximum * valued depends on decorrType. * \param partiallyComplex Low power or high quality processing 0: HQ, 1: LQ * (only allowed for DECORR_MPS | DECORR_PS). * \param useFractDelay Indicate usage of fractional delay 0: off, 1: on * (currently not supported). * \param isLegacyPS Indicate if DECORR_PS is used for HEAACv2 (for all * other cases: isLegacyPS = 0). The purpose of this parameter is to select the * correct number of param bands for the ducker. * \param initStatesFlag Indicates whether the states buffer has to be * cleared. * * \return 0 on success. */ INT FDKdecorrelateInit(HANDLE_DECORR_DEC hDecorrDec, const INT nrHybBands, const FDK_DECORR_TYPE decorrType, const FDK_DUCKER_TYPE duckerType, const INT decorrConfig, const INT seed, const INT partiallyComplex, const INT useFractDelay, const INT isLegacyPS, const INT initStatesFlag); /** * \brief Apply Decorrelator on input data. * * Function applies decorrelator and ducker inplace on hybrid input data. * Modified hybrid data will be returned inplace. * * \param hDecorrDec A decorrelator handle. * \param dataRealIn In (hybrid) data. * \param dataImagIn In (hybrid) data. * \param dataRealOut Out (hybrid) data (can be same as dataRealIn for * in-place calculation). * \param dataImagOut Out (hybrid) data (can be same as dataImagIn for * in-place calculation). * \param startHybBand Hybrid band to start with decorrelation. * * \return 0 on success. */ INT FDKdecorrelateApply(HANDLE_DECORR_DEC hDecorrDec, FIXP_DBL *dataRealIn, FIXP_DBL *dataImagIn, FIXP_DBL *dataRealOut, FIXP_DBL *dataImagOut, const INT startHybBand); /** * \brief Destroy a Decorrelator instance. * * Deallocate whole memory of decorraltor and inside ducker. * * \param hDecorrDec Pointer to a decoderrolator handle. Null initialized on * return. * * \return 0 on success. */ INT FDKdecorrelateClose(HANDLE_DECORR_DEC hDecorrDec); /** * \brief Get max value address of direct signal. * * Get max value address of direct signal needed for ducker energy calculation. * * \param hDecorrDec Pointer to a decoderrolator handle. * * \return address of max value */ FIXP_DBL *getAddrDirectSignalMaxVal(HANDLE_DECORR_DEC hDecorrDec); #endif /* FDK_DECORRELATE_H */ fdk-aac-2.0.2/libFDK/include/FDK_hybrid.h000066400000000000000000000242251404225310700176400ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Markus Lohwasser Description: FDK Tools Hybrid Filterbank *******************************************************************************/ #ifndef FDK_HYBRID_H #define FDK_HYBRID_H #include "common_fix.h" /*--------------- enums -------------------------------*/ /** * Hybrid Filterband modes. */ typedef enum { THREE_TO_TEN, THREE_TO_TWELVE, THREE_TO_SIXTEEN } FDK_HYBRID_MODE; /*--------------- structure definitions ---------------*/ typedef const struct FDK_HYBRID_SETUP *HANDLE_FDK_HYBRID_SETUP; typedef struct { FIXP_DBL *bufferLFReal[3]; /*!< LF real filter states. */ FIXP_DBL *bufferLFImag[3]; /*!< LF imag filter states. */ FIXP_DBL *bufferHFReal[13]; /*!< HF real delay lines. */ FIXP_DBL *bufferHFImag[13]; /*!< HF imag delay lines. */ INT bufferLFpos; /*!< Position to write incoming data into ringbuffer. */ INT bufferHFpos; /*!< Delay line positioning. */ INT nrBands; /*!< Number of QMF bands. */ INT cplxBands; /*!< Number of complex QMF bands.*/ UCHAR hfMode; /*!< Flag signalizes treatment of HF bands. */ FIXP_DBL *pLFmemory; /*!< Pointer to LF states buffer. */ FIXP_DBL *pHFmemory; /*!< Pointer to HF states buffer. */ UINT LFmemorySize; /*!< Size of LF states buffer. */ UINT HFmemorySize; /*!< Size of HF states buffer. */ HANDLE_FDK_HYBRID_SETUP pSetup; /*!< Pointer to filter setup. */ } FDK_ANA_HYB_FILTER; typedef struct { INT nrBands; /*!< Number of QMF bands. */ INT cplxBands; /*!< Number of complex QMF bands.*/ HANDLE_FDK_HYBRID_SETUP pSetup; /*!< Pointer to filter setup. */ } FDK_SYN_HYB_FILTER; typedef FDK_ANA_HYB_FILTER *HANDLE_FDK_ANA_HYB_FILTER; typedef FDK_SYN_HYB_FILTER *HANDLE_FDK_SYN_HYB_FILTER; /** * \brief Create one instance of Hybrid Analyis Filterbank. * * \param hAnalysisHybFilter Pointer to an outlying allocated Hybrid Analysis * Filterbank structure. * \param pLFmemory Pointer to outlying buffer used LF filtering. * \param LFmemorySize Size of pLFmemory in bytes. * \param pHFmemory Pointer to outlying buffer used HF delay line. * \param HFmemorySize Size of pLFmemory in bytes. * * \return 0 on success. */ INT FDKhybridAnalysisOpen(HANDLE_FDK_ANA_HYB_FILTER hAnalysisHybFilter, FIXP_DBL *const pLFmemory, const UINT LFmemorySize, FIXP_DBL *const pHFmemory, const UINT HFmemorySize); /** * \brief Initialize and configure Hybrid Analysis Filterbank instance. * * \param hAnalysisHybFilter A Hybrid Analysis Filterbank handle. * \param mode Select hybrid filter configuration. * \param qmfBands Number of qmf bands to be processed. * \param cplxBands Number of complex qmf bands to be processed. * \param initStatesFlag Indicates whether the states buffer has to be * cleared. * * \return 0 on success. */ INT FDKhybridAnalysisInit(HANDLE_FDK_ANA_HYB_FILTER hAnalysisHybFilter, const FDK_HYBRID_MODE mode, const INT qmfBands, const INT cplxBands, const INT initStatesFlag); /** * \brief Adjust Hybrid Analysis Filterbank states. * * \param hAnalysisHybFilter A Hybrid Analysis Filterbank handle. * \param scalingValue Scaling value to be applied on filter states. * * \return 0 on success. */ INT FDKhybridAnalysisScaleStates(HANDLE_FDK_ANA_HYB_FILTER hAnalysisHybFilter, const INT scalingValue); /** * \brief Apply Hybrid Analysis Filterbank on Qmf input data. * * \param hAnalysisHybFilter A Hybrid Analysis Filterbank handle. * \param pQmfReal Qmf input data. * \param pQmfImag Qmf input data. * \param pHybridReal Hybrid output data. * \param pHybridImag Hybrid output data. * * \return 0 on success. */ INT FDKhybridAnalysisApply(HANDLE_FDK_ANA_HYB_FILTER hAnalysisHybFilter, const FIXP_DBL *const pQmfReal, const FIXP_DBL *const pQmfImag, FIXP_DBL *const pHybridReal, FIXP_DBL *const pHybridImag); /** * \brief Close a Hybrid Analysis Filterbank instance. * * \param hAnalysisHybFilter Pointer to a Hybrid Analysis Filterbank instance. * * \return 0 on success. */ INT FDKhybridAnalysisClose(HANDLE_FDK_ANA_HYB_FILTER hAnalysisHybFilter); /** * \brief Initialize and configure Hybrdid Synthesis Filterbank instance. * * \param hSynthesisHybFilter A Hybrid Synthesis Filterbank handle. * \param mode Select hybrid filter configuration. * \param qmfBands Number of qmf bands to be processed. * \param cplxBands Number of complex qmf bands to be processed. * * \return 0 on success. */ INT FDKhybridSynthesisInit(HANDLE_FDK_SYN_HYB_FILTER hSynthesisHybFilter, const FDK_HYBRID_MODE mode, const INT qmfBands, const INT cplxBands); /** * \brief Apply Hybrid Analysis Filterbank on Hybrid data. * * \param hSynthesisHybFilter A Hybrid Analysis Filterbandk handle. * \param pHybridReal Hybrid input data. * \param pHybridImag Hybrid input data. * \param pQmfReal Qmf output data. * \param pQmfImag Qmf output data. * */ void FDKhybridSynthesisApply(HANDLE_FDK_SYN_HYB_FILTER hSynthesisHybFilter, const FIXP_DBL *const pHybridReal, const FIXP_DBL *const pHybridImag, FIXP_DBL *const pQmfReal, FIXP_DBL *const pQmfImag); #endif /* FDK_HYBRID_H */ fdk-aac-2.0.2/libFDK/include/FDK_lpc.h000066400000000000000000000233331404225310700171340ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Manuel Jander Description: LPC related functions *******************************************************************************/ #ifndef FDK_LPC_H #define FDK_LPC_H #include "common_fix.h" #define LPC_MAX_ORDER 24 /* * Experimental solution for lattice filter substitution. * LPC_SYNTHESIS_IIR macro must be activated in aacdec_tns.cpp. * When LPC_SYNTHESIS_IIR enabled, there will be a substitution of the default * lpc synthesis lattice filter by an IIR synthesis filter (with a conversionof * the filter coefs). LPC_TNS related macros are intended to implement the data * types used by the CLpc_Synthesis variant which is used for this solution. * */ /* #define LPC_TNS_LOWER_PRECISION */ typedef FIXP_DBL FIXP_LPC_TNS; #define FX_DBL2FX_LPC_TNS(x) (x) #define FX_DBL2FXCONST_LPC_TNS(x) (x) #define FX_LPC_TNS2FX_DBL(x) (x) #define FL2FXCONST_LPC_TNS(val) FL2FXCONST_DBL(val) #define MAXVAL_LPC_TNS MAXVAL_DBL typedef FIXP_SGL FIXP_LPC; #define FX_DBL2FX_LPC(x) FX_DBL2FX_SGL((FIXP_DBL)(x)) #define FX_DBL2FXCONST_LPC(x) FX_DBL2FXCONST_SGL(x) #define FX_LPC2FX_DBL(x) FX_SGL2FX_DBL(x) #define FL2FXCONST_LPC(val) FL2FXCONST_SGL(val) #define MAXVAL_LPC MAXVAL_SGL /** * \brief Obtain residual signal through LPC analysis. * \param signal pointer to buffer holding signal to be analysed. Residual is * returned there (in place) * \param signal_size the size of the input data in pData * \param lpcCoeff_m the LPC filter coefficient mantissas * \param lpcCoeff_e the LPC filter coefficient exponent * \param order the LPC filter order (size of coeff) * \param filtState Pointer to state buffer of size order * \param filtStateIndex pointer to state index storage */ void CLpc_Analysis(FIXP_DBL signal[], const int signal_size, const FIXP_LPC lpcCoeff_m[], const int lpcCoeff_e, const int order, FIXP_DBL *filtState, int *filtStateIndex); /** * \brief Synthesize signal fom residual through LPC synthesis, using LP * coefficients. * \param signal pointer to buffer holding the residual signal. The synthesis is * returned there (in place) * \param signal_size the size of the input data in pData * \param inc buffer traversal increment for signal * \param coeff the LPC filter coefficients * \param coeff_e exponent of coeff * \param order the LPC filter order (size of coeff) * \param state state buffer of size LPC_MAX_ORDER * \param pStateIndex pointer to state index storage */ void CLpc_Synthesis(FIXP_DBL *signal, const int signal_size, const int signal_e, const int inc, const FIXP_LPC_TNS *lpcCoeff_m, const int lpcCoeff_e, const int order, FIXP_DBL *state, int *pStateIndex); void CLpc_Synthesis(FIXP_DBL *signal, const int signal_size, const int signal_e, const int inc, const FIXP_LPC coeff[], const int coeff_e, const int order, FIXP_DBL *filtState, int *pStateIndex); /** * \brief Synthesize signal fom residual through LPC synthesis, using ParCor * coefficients. The algorithm assumes a filter gain of max 1.0. If the filter * gain is higher, this must be accounted into the values of signal_e * and/or signal_e_out to avoid overflows. * \param signal pointer to buffer holding the residual signal. The synthesis is * returned there (in place) * \param signal_size the size of the input data in pData * \param inc buffer traversal increment for signal * \param coeff the LPC filter coefficients * \param coeff_e exponent of coeff * \param order the LPC filter order (size of coeff) * \param state state buffer of size LPC_MAX_ORDER */ void CLpc_SynthesisLattice(FIXP_DBL *signal, const int signal_size, const int signal_e, const int signal_e_out, const int inc, const FIXP_SGL *coeff, const int order, FIXP_DBL *state); void CLpc_SynthesisLattice(FIXP_DBL *RESTRICT signal, const int signal_size, const int signal_e, const int signal_e_out, const int inc, const FIXP_DBL *RESTRICT coeff, const int order, FIXP_DBL *RESTRICT state); /** * \brief */ INT CLpc_ParcorToLpc(const FIXP_LPC_TNS reflCoeff[], FIXP_LPC_TNS LpcCoeff[], INT numOfCoeff, FIXP_DBL workBuffer[]); INT CLpc_ParcorToLpc(const FIXP_LPC reflCoeff[], FIXP_LPC LpcCoeff[], const int numOfCoeff, FIXP_DBL workBuffer[]); /** * \brief Calculate ParCor (Partial autoCorrelation, reflection) coefficients * from autocorrelation coefficients using the Schur algorithm (instead of * Levinson Durbin). * \param acorr order+1 autocorrelation coefficients * \param reflCoeff output reflection /ParCor coefficients. The first * coefficient which is always 1.0 is ommitted. * \param order number of acorr / reflCoeff coefficients. * \param pPredictionGain_m prediction gain mantissa * \param pPredictionGain_e prediction gain exponent */ void CLpc_AutoToParcor(FIXP_DBL acorr[], const int acorr_e, FIXP_LPC reflCoeff[], const int order, FIXP_DBL *pPredictionGain_m, INT *pPredictionGain_e); #endif /* FDK_LPC_H */ fdk-aac-2.0.2/libFDK/include/FDK_matrixCalloc.h000066400000000000000000000240611404225310700207770ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Description: matrix memory allocation *******************************************************************************/ #ifndef FDK_MATRIXCALLOC_H #define FDK_MATRIXCALLOC_H #include "machine_type.h" #include "genericStds.h" /* It is recommended to use FDK_ALLOCATE_MEMORY_1D instead of fdkCallocMatrix1D */ void* fdkCallocMatrix1D(UINT dim1, UINT size); void* fdkCallocMatrix1D_aligned(UINT dim1, UINT size); /* It is recommended to use FDK_ALLOCATE_MEMORY_1D_INT instead of * fdkCallocMatrix1D_int */ void* fdkCallocMatrix1D_int(UINT dim1, UINT size, MEMORY_SECTION s); void* fdkCallocMatrix1D_int_aligned(UINT dim1, UINT size, MEMORY_SECTION s); /* It is recommended to use FDK_FREE_MEMORY_1D instead of fdkFreeMatrix1D */ void fdkFreeMatrix1D(void* p); void fdkFreeMatrix1D_aligned(void* p); /* It is recommended to use FDK_ALLOCATE_MEMORY_2D instead of fdkCallocMatrix2D */ void** fdkCallocMatrix2D(UINT dim1, UINT dim2, UINT size); void** fdkCallocMatrix2D_aligned(UINT dim1, UINT dim2, UINT size); /* It is recommended to use FDK_ALLOCATE_MEMORY_2D_INT instead of * fdkCallocMatrix2D_int */ void** fdkCallocMatrix2D_int(UINT dim1, UINT dim2, UINT size, MEMORY_SECTION s); /* It is recommended to use FDK_ALLOCATE_MEMORY_2D_INT_ALIGNED instead of * fdkCallocMatrix2D_int_aligned */ void** fdkCallocMatrix2D_int_aligned(UINT dim1, UINT dim2, UINT size, MEMORY_SECTION s); /* It is recommended to use FDK_FREE_MEMORY_2D instead of fdkFreeMatrix2D */ void fdkFreeMatrix2D(void** p); /* It is recommended to use FDK_FREE_MEMORY_2D_ALIGNED instead of * fdkFreeMatrix2D_aligned */ void fdkFreeMatrix2D_aligned(void** p); /* It is recommended to use FDK_ALLOCATE_MEMORY_3D instead of fdkCallocMatrix3D */ void*** fdkCallocMatrix3D(UINT dim1, UINT dim2, UINT dim3, UINT size); /* It is recommended to use FDK_ALLOCATE_MEMORY_3D_INT instead of * fdkCallocMatrix3D_int */ void*** fdkCallocMatrix3D_int(UINT dim1, UINT dim2, UINT dim3, UINT size, MEMORY_SECTION s); /* It is recommended to use FDK_FREE_MEMORY_3D instead of fdkFreeMatrix3D */ void fdkFreeMatrix3D(void*** p); #define FDK_ALLOCATE_MEMORY_1D(a, dim1, type) \ if (((a) = (type*)fdkCallocMatrix1D((dim1), sizeof(type))) == NULL) { \ goto bail; \ } #define FDK_ALLOCATE_MEMORY_1D_ALIGNED(a, dim1, type) \ if (((a) = (type*)fdkCallocMatrix1D_aligned((dim1), sizeof(type))) == \ NULL) { \ goto bail; \ } #define FDK_ALLOCATE_MEMORY_1D_P(a, dim1, type, ptype) \ if (((a) = (ptype)fdkCallocMatrix1D((dim1), sizeof(type))) == NULL) { \ goto bail; \ } #define FDK_ALLOCATE_MEMORY_1D_INT(a, dim1, type, s) \ if (((a) = (type*)fdkCallocMatrix1D_int((dim1), sizeof(type), (s))) == \ NULL) { \ goto bail; \ } #define FDK_FREE_MEMORY_1D(a) \ do { \ fdkFreeMatrix1D((void*)(a)); \ (a) = NULL; \ } while (0) #define FDK_FREE_MEMORY_1D_ALIGNED(a) \ do { \ fdkFreeMatrix1D_aligned((void*)(a)); \ (a) = NULL; \ } while (0) #define FDK_ALLOCATE_MEMORY_2D(a, dim1, dim2, type) \ if (((a) = (type**)fdkCallocMatrix2D((dim1), (dim2), sizeof(type))) == \ NULL) { \ goto bail; \ } #define FDK_ALLOCATE_MEMORY_2D_INT(a, dim1, dim2, type, s) \ if (((a) = (type**)fdkCallocMatrix2D_int((dim1), (dim2), sizeof(type), \ (s))) == NULL) { \ goto bail; \ } #define FDK_ALLOCATE_MEMORY_2D_INT_ALIGNED(a, dim1, dim2, type, s) \ if (((a) = (type**)fdkCallocMatrix2D_int_aligned( \ (dim1), (dim2), sizeof(type), (s))) == NULL) { \ goto bail; \ } #define FDK_FREE_MEMORY_2D(a) \ do { \ fdkFreeMatrix2D((void**)(a)); \ (a) = NULL; \ } while (0) #define FDK_FREE_MEMORY_2D_ALIGNED(a) \ do { \ fdkFreeMatrix2D_aligned((void**)(a)); \ (a) = NULL; \ } while (0) #define FDK_ALLOCATE_MEMORY_3D(a, dim1, dim2, dim3, type) \ if (((a) = (type***)fdkCallocMatrix3D((dim1), (dim2), (dim3), \ sizeof(type))) == NULL) { \ goto bail; \ } #define FDK_ALLOCATE_MEMORY_3D_INT(a, dim1, dim2, dim3, type, s) \ if (((a) = (type***)fdkCallocMatrix3D_int((dim1), (dim2), (dim3), \ sizeof(type), (s))) == NULL) { \ goto bail; \ } #define FDK_FREE_MEMORY_3D(a) \ do { \ fdkFreeMatrix3D((void***)(a)); \ (a) = NULL; \ } while (0) #endif /* FDK_MATRIXCALLOC_H */ fdk-aac-2.0.2/libFDK/include/FDK_qmf_domain.h000066400000000000000000000407521404225310700204740ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Matthias Hildenbrand Description: Module to efficiently handle QMF data for multiple channels and to share the data between e.g. SBR and MPS *******************************************************************************/ #ifndef FDK_QMF_DOMAIN_H #define FDK_QMF_DOMAIN_H #include "qmf.h" typedef enum { QMF_DOMAIN_OK = 0x0, /*!< No error occurred. */ QMF_DOMAIN_OUT_OF_MEMORY = 0x1, /*!< QMF-Configuration demands for more memory than allocated on heap. */ QMF_DOMAIN_INIT_ERROR = 0x2, /*!< An error during filterbank-setup occurred. */ QMF_DOMAIN_RESAMPLER_INIT_ERROR = 0x3 /*!< An error during QMF-resampler-setup occurred. */ } QMF_DOMAIN_ERROR; #define CMPLX_MOD (2) #define QMF_MAX_WB_SECTIONS (5) /* maximum number of workbuffer sections */ #define QMF_WB_SECTION_SIZE (1024 * 2) H_ALLOC_MEM_OVERLAY(QmfWorkBufferCore1, FIXP_DBL) H_ALLOC_MEM_OVERLAY(QmfWorkBufferCore3, FIXP_DBL) H_ALLOC_MEM_OVERLAY(QmfWorkBufferCore4, FIXP_DBL) H_ALLOC_MEM_OVERLAY(QmfWorkBufferCore6, FIXP_DBL) H_ALLOC_MEM_OVERLAY(QmfWorkBufferCore7, FIXP_DBL) #define QMF_DOMAIN_MAX_ANALYSIS_QMF_BANDS (64) #define QMF_DOMAIN_MAX_SYNTHESIS_QMF_BANDS (QMF_MAX_SYNTHESIS_BANDS) #define QMF_DOMAIN_MAX_QMF_PROC_BANDS (64) #define QMF_DOMAIN_MAX_TIMESLOTS (64) #define QMF_DOMAIN_MAX_OV_TIMESLOTS (12) #define QMF_DOMAIN_ANALYSIS_QMF_BANDS_16 (16) #define QMF_DOMAIN_ANALYSIS_QMF_BANDS_24 (24) #define QMF_DOMAIN_ANALYSIS_QMF_BANDS_32 (32) #define QMF_DOMAIN_TIMESLOTS_16 (16) #define QMF_DOMAIN_TIMESLOTS_32 (32) #define QMF_DOMAIN_OV_TIMESLOTS_16 (3) #define QMF_DOMAIN_OV_TIMESLOTS_32 (6) H_ALLOC_MEM(AnaQmfStates, FIXP_DBL) H_ALLOC_MEM(SynQmfStates, FIXP_QSS) H_ALLOC_MEM(QmfSlotsReal, FIXP_DBL *) H_ALLOC_MEM(QmfSlotsImag, FIXP_DBL *) H_ALLOC_MEM(QmfOverlapBuffer, FIXP_DBL) H_ALLOC_MEM(AnaQmfStates16, FIXP_DBL) H_ALLOC_MEM(AnaQmfStates24, FIXP_DBL) H_ALLOC_MEM(AnaQmfStates32, FIXP_DBL) H_ALLOC_MEM(QmfSlotsReal16, FIXP_DBL *) H_ALLOC_MEM(QmfSlotsReal32, FIXP_DBL *) H_ALLOC_MEM(QmfSlotsImag16, FIXP_DBL *) H_ALLOC_MEM(QmfSlotsImag32, FIXP_DBL *) H_ALLOC_MEM(QmfOverlapBuffer16, FIXP_DBL) H_ALLOC_MEM(QmfOverlapBuffer32, FIXP_DBL) /** * Structure to hold the configuration data which is global whithin a QMF domain * instance. */ typedef struct { UCHAR qmfDomainExplicitConfig; /*!< Flag to signal that QMF domain is set explicitly instead of SBR and MPS init routines. */ UCHAR nInputChannels; /*!< Number of QMF input channels. */ UCHAR nInputChannels_requested; /*!< Corresponding requested not yet active configuration parameter. */ UCHAR nOutputChannels; /*!< Number of QMF output channels. */ UCHAR nOutputChannels_requested; /*!< Corresponding requested not yet active configuration parameter. */ UCHAR parkChannel; /*!< signal to automatically allocate additional memory to park a channel if only one processing channel is available. */ UCHAR parkChannel_requested; FIXP_DBL * pWorkBuffer[QMF_MAX_WB_SECTIONS]; /*!< Pointerarray to volatile memory. */ UINT flags; /*!< Flags to be set on all QMF analysis/synthesis filter instances. */ UINT flags_requested; /*!< Corresponding requested not yet active configuration parameter. */ UCHAR nBandsAnalysis; /*!< Number of QMF analysis bands for all input channels. */ UCHAR nBandsAnalysis_requested; /*!< Corresponding requested not yet active configuration parameter. */ USHORT nBandsSynthesis; /*!< Number of QMF synthesis bands for all output channels. */ USHORT nBandsSynthesis_requested; /*!< Corresponding requested not yet active configuration parameter. */ UCHAR nQmfTimeSlots; /*!< Number of QMF time slots (stored in work buffer memory). */ UCHAR nQmfTimeSlots_requested; /*!< Corresponding requested not yet active configuration parameter. */ UCHAR nQmfOvTimeSlots; /*!< Number of QMF overlap/delay time slots (stored in persistent memory). */ UCHAR nQmfOvTimeSlots_requested; /*!< Corresponding requested not yet active configuration parameter. */ UCHAR nQmfProcBands; /*!< Number of QMF bands which are processed by the decoder. Typically this is equal to nBandsSynthesis but it may differ if the QMF based resampler is being used. */ UCHAR nQmfProcBands_requested; /*!< Corresponding requested not yet active configuration parameter. */ UCHAR nQmfProcChannels; /*!< Number of complete QMF channels which need to coexist in memory at the same time. For most cases this is 1 which means the work buffer can be shared between audio channels. */ UCHAR nQmfProcChannels_requested; /*!< Corresponding requested not yet active configuration parameter. */ } FDK_QMF_DOMAIN_GC; typedef FDK_QMF_DOMAIN_GC *HANDLE_FDK_QMF_DOMAIN_GC; /** * Structure representing one QMF input channel. This includes the QMF analysis * and the QMF domain data representation needed by the codec. Work buffer data * may be shared between channels if the codec processes all QMF channels in a * consecutive order. */ typedef struct { HANDLE_FDK_QMF_DOMAIN_GC pGlobalConf; /*!< Pointer to global configuration structure. */ QMF_FILTER_BANK fb; /*!< QMF (analysis) filter bank structure. */ QMF_SCALE_FACTOR scaling; /*!< Structure with scaling information. */ UCHAR workBuf_nTimeSlots; /*!< Work buffer dimension for this channel is (workBuf_nTimeSlots * workBuf_nBands * CMPLX_MOD). */ UCHAR workBuf_nBands; /*!< Work buffer dimension for this channel is (workBuf_nTimeSlots * workBuf_nBands * CMPLX_MOD). */ USHORT workBufferOffset; /*!< Offset within work buffer. */ USHORT workBufferSectSize; /*!< Size of work buffer section. */ FIXP_DBL * pAnaQmfStates; /*!< Pointer to QMF analysis states (persistent memory). */ FIXP_DBL *pOverlapBuffer; /*!< Pointer to QMF overlap/delay memory (persistent memory). */ FIXP_DBL **pWorkBuffer; /*!< Pointer array to available work buffers. */ FIXP_DBL * *hQmfSlotsReal; /*!< Handle for QMF real data time slot pointer array. */ FIXP_DBL **hQmfSlotsImag; /*!< Handle for QMF imaginary data time slot pointer array. */ } FDK_QMF_DOMAIN_IN; typedef FDK_QMF_DOMAIN_IN *HANDLE_FDK_QMF_DOMAIN_IN; /** * Structure representing one QMF output channel. */ typedef struct { QMF_FILTER_BANK fb; /*!< QMF (synthesis) filter bank structure. */ FIXP_QSS *pSynQmfStates; /*!< Pointer to QMF synthesis states (persistent memory). */ } FDK_QMF_DOMAIN_OUT; typedef FDK_QMF_DOMAIN_OUT *HANDLE_FDK_QMF_DOMAIN_OUT; /** * Structure representing the QMF domain for multiple channels. */ typedef struct { FDK_QMF_DOMAIN_GC globalConf; /*!< Global configuration structure. */ FDK_QMF_DOMAIN_IN QmfDomainIn[((8) + (1))]; /*!< Array of QMF domain input structures */ FDK_QMF_DOMAIN_OUT QmfDomainOut[((8) + (1))]; /*!< Array of QMF domain output structures */ } FDK_QMF_DOMAIN; typedef FDK_QMF_DOMAIN *HANDLE_FDK_QMF_DOMAIN; /** * \brief Check whether analysis- and synthesis-filterbank-states have been * initialized. * * \param qd Pointer to QMF domain structure. * * \return 1 if initialized, 0 else */ int FDK_QmfDomain_IsInitialized(const HANDLE_FDK_QMF_DOMAIN qd); /** * \brief Initialize QMF analysis and synthesis filter banks and set up QMF data * representation. * * \param qd Pointer to QMF domain structure. * \param extra_flags Initialize filter banks with extra flags which were not * set in the global config flags field. * * \return 0 on success. */ int FDK_QmfDomain_InitFilterBank(HANDLE_FDK_QMF_DOMAIN qd, UINT extra_flags); /** * \brief When QMF processing of one channel is finished copy the overlap/delay * part into the persistent memory to be used in the next frame. * * \param qd_ch Pointer to a QMF domain input channel. * \param offset * * \return void */ void FDK_QmfDomain_SaveOverlap(HANDLE_FDK_QMF_DOMAIN_IN qd_ch, int offset); /** * \brief Get one slot of QMF data and adapt the scaling. * * \param qd_ch Pointer to a QMF domain input channel. * \param ts Time slot number to be obtained. * \param start_band Start index of QMF bands to be obtained. * \param stop_band Stop index of QMF band to be obtained. * \param pQmfOutReal Output buffer (real QMF data). * \param pQmfOutImag Output buffer (imag QMF data). * \param exp_out Target exponent (scaling) of data. * * \return void */ void FDK_QmfDomain_GetSlot(const HANDLE_FDK_QMF_DOMAIN_IN qd_ch, const int ts, const int start_band, const int stop_band, FIXP_DBL *pQmfOutReal, FIXP_DBL *pQmfOutImag, const int exp_out); /** * \brief Direct access to the work buffer associated with a certain channel (no * time slot pointer array is used). * * \param qd_ch Pointer to a QMF domain input channel. * \param ts Time slot number to be obtained. * \param ppQmfReal Returns the pointer to the requested part of the work buffer * (real time slot). * \param ppQmfImag Returns the pointer to the requested part of the work buffer * (imag time slot). * * \return void */ void FDK_QmfDomain_GetWorkBuffer(const HANDLE_FDK_QMF_DOMAIN_IN qd_ch, const int ts, FIXP_DBL **ppQmfReal, FIXP_DBL **ppQmfImag); /** * \brief For the case that the work buffer associated to this channel is not * identical to the processing channel work buffer copy the data into the * processing channel. * * \param qd_ch Pointer to a QMF domain input channel. * \return void */ void FDK_QmfDomain_WorkBuffer2ProcChannel(const HANDLE_FDK_QMF_DOMAIN_IN qd_ch); /** * \brief For the case of stereoCfgIndex3 with HBE the HBE buffer is copied into * the processing channel work buffer and the processing channel work buffer is * copied into the HBE buffer. * * \param qd_ch Pointer to a QMF domain input channel. * \param ppQmfReal Pointer to a HBE QMF data buffer (real). * \param ppQmfImag Pointer to a HBE QMF data buffer (imag). * * \return void */ void FDK_QmfDomain_QmfData2HBE(HANDLE_FDK_QMF_DOMAIN_IN qd_ch, FIXP_DBL **ppQmfReal, FIXP_DBL **ppQmfImag); /** * \brief Set all fields for requested parametervalues in global config struct * FDK_QMF_DOMAIN_GC to 0. * * \param hgc Pointer to a QMF domain global config struct. */ void FDK_QmfDomain_ClearRequested(HANDLE_FDK_QMF_DOMAIN_GC hgc); /** * \brief Check for parameter-change requests in global config and * (re-)configure QMF domain accordingly. * * \param hqd Pointer to QMF domain * * \return errorcode */ QMF_DOMAIN_ERROR FDK_QmfDomain_Configure(HANDLE_FDK_QMF_DOMAIN hqd); /** * \brief Free QMF workbuffer, QMF persistent memory and configuration * variables. * * \param hqd Pointer to QMF domain */ void FDK_QmfDomain_FreeMem(HANDLE_FDK_QMF_DOMAIN hqd); /** * \brief Clear QMF overlap buffers and QMF filter bank states. * * \param hqd Pointer to QMF domain */ QMF_DOMAIN_ERROR FDK_QmfDomain_ClearPersistentMemory(HANDLE_FDK_QMF_DOMAIN hqd); /** * \brief Free QMF workbuffer and QMF persistent memory. * * \param hqd Pointer to QMF domain * * \param dmx_lp_mode downmix low power mode flag */ void FDK_QmfDomain_Close(HANDLE_FDK_QMF_DOMAIN hqd); #endif /* FDK_QMF_DOMAIN_H */ fdk-aac-2.0.2/libFDK/include/FDK_tools_rom.h000066400000000000000000000352011404225310700203700ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Oliver Moser Description: ROM tables used by FDK tools *******************************************************************************/ #ifndef FDK_TOOLS_ROM_H #define FDK_TOOLS_ROM_H #include "common_fix.h" #include "FDK_audio.h" /* sinetables */ /* None radix2 rotation vectors */ extern RAM_ALIGN const FIXP_STB RotVectorReal60[60]; extern RAM_ALIGN const FIXP_STB RotVectorImag60[60]; extern RAM_ALIGN const FIXP_STB RotVectorReal192[192]; extern RAM_ALIGN const FIXP_STB RotVectorImag192[192]; extern RAM_ALIGN const FIXP_STB RotVectorReal240[210]; extern RAM_ALIGN const FIXP_STB RotVectorImag240[210]; extern RAM_ALIGN const FIXP_STB RotVectorReal480[480]; extern RAM_ALIGN const FIXP_STB RotVectorImag480[480]; extern RAM_ALIGN const FIXP_STB RotVectorReal6[6]; extern RAM_ALIGN const FIXP_STB RotVectorImag6[6]; extern RAM_ALIGN const FIXP_STB RotVectorReal12[12]; extern RAM_ALIGN const FIXP_STB RotVectorImag12[12]; extern RAM_ALIGN const FIXP_STB RotVectorReal24[24]; extern RAM_ALIGN const FIXP_STB RotVectorImag24[24]; extern RAM_ALIGN const FIXP_STB RotVectorReal48[48]; extern RAM_ALIGN const FIXP_STB RotVectorImag48[48]; extern RAM_ALIGN const FIXP_STB RotVectorReal80[80]; extern RAM_ALIGN const FIXP_STB RotVectorImag80[80]; extern RAM_ALIGN const FIXP_STB RotVectorReal96[96]; extern RAM_ALIGN const FIXP_STB RotVectorImag96[96]; extern RAM_ALIGN const FIXP_STB RotVectorReal384[384]; extern RAM_ALIGN const FIXP_STB RotVectorImag384[384]; extern RAM_ALIGN const FIXP_STB RotVectorReal20[20]; extern RAM_ALIGN const FIXP_STB RotVectorImag20[20]; extern RAM_ALIGN const FIXP_STB RotVectorReal120[120]; extern RAM_ALIGN const FIXP_STB RotVectorImag120[120]; /* Regular sine tables */ extern RAM_ALIGN const FIXP_STP SineTable1024[]; extern RAM_ALIGN const FIXP_STP SineTable512[]; extern RAM_ALIGN const FIXP_STP SineTable480[]; extern RAM_ALIGN const FIXP_STP SineTable384[]; extern RAM_ALIGN const FIXP_STP SineTable80[]; #ifdef INCLUDE_SineTable10 extern RAM_ALIGN const FIXP_STP SineTable10[]; #endif /* AAC-LC windows */ extern RAM_ALIGN const FIXP_WTP SineWindow1024[]; extern RAM_ALIGN const FIXP_WTP KBDWindow1024[]; extern RAM_ALIGN const FIXP_WTP SineWindow128[]; extern RAM_ALIGN const FIXP_WTP KBDWindow128[]; extern RAM_ALIGN const FIXP_WTP SineWindow960[]; extern RAM_ALIGN const FIXP_WTP KBDWindow960[]; extern RAM_ALIGN const FIXP_WTP SineWindow120[]; extern RAM_ALIGN const FIXP_WTP KBDWindow120[]; /* AAC-LD windows */ extern RAM_ALIGN const FIXP_WTP SineWindow512[]; #define LowOverlapWindow512 SineWindow128 extern RAM_ALIGN const FIXP_WTP SineWindow480[]; #define LowOverlapWindow480 SineWindow120 /* USAC TCX Window */ extern RAM_ALIGN const FIXP_WTP SineWindow256[256]; extern RAM_ALIGN const FIXP_WTP SineWindow192[]; /* USAC 8/3 windows */ extern RAM_ALIGN const FIXP_WTP SineWindow768[]; extern RAM_ALIGN const FIXP_WTP KBDWindow768[]; extern RAM_ALIGN const FIXP_WTP SineWindow96[]; extern RAM_ALIGN const FIXP_WTP KBDWindow96[]; /* DCT and others */ extern RAM_ALIGN const FIXP_WTP SineWindow64[]; extern RAM_ALIGN const FIXP_WTP SineWindow48[]; extern RAM_ALIGN const FIXP_WTP SineWindow32[]; extern RAM_ALIGN const FIXP_WTP SineWindow24[]; extern RAM_ALIGN const FIXP_WTP SineWindow16[]; extern RAM_ALIGN const FIXP_WTP SineWindow8[]; /** * \brief Helper table for window slope mapping. You should prefer the usage of * the function FDKgetWindowSlope(), this table is only made public for some * optimized access inside dct.cpp. */ extern const FIXP_WTP *const windowSlopes[2][4][9]; /** * \brief Window slope access helper. Obtain a window of given length and shape. * \param length Length of the window slope. * \param shape Shape index of the window slope. 0: sine window, 1: * Kaiser-Bessel. Any other value is applied a mask of 1 to, mapping it to * either 0 or 1. * \param Pointer to window slope or NULL if the requested window slope is not * available. */ const FIXP_WTP *FDKgetWindowSlope(int length, int shape); extern const FIXP_WTP sin_twiddle_L64[]; /* * Filter coefficient type definition */ #if defined(ARCH_PREFER_MULT_16x16) || defined(ARCH_PREFER_MULT_32x16) #define QMF_COEFF_16BIT #endif #define QMF_FILTER_PROTOTYPE_SIZE 640 #define QMF_NO_POLY 5 #ifdef QMF_COEFF_16BIT #define FIXP_PFT FIXP_SGL #define FIXP_QTW FIXP_SGL #define FX_DBL2FX_QTW(x) FX_DBL2FX_SGL(x) #else #define FIXP_PFT FIXP_DBL #define FIXP_QTW FIXP_DBL #define FX_DBL2FX_QTW(x) (x) #endif #define QMF640_PFT_TABLE_SIZE (640 / 2 + QMF_NO_POLY) /* Resampling twiddles for QMF */ /* Not resampling twiddles */ extern RAM_ALIGN const FIXP_QTW qmf_phaseshift_cos32[32]; extern RAM_ALIGN const FIXP_QTW qmf_phaseshift_sin32[32]; /* Adapted analysis post-twiddles for down-sampled HQ SBR */ extern RAM_ALIGN const FIXP_QTW qmf_phaseshift_cos_downsamp32[32]; extern RAM_ALIGN const FIXP_QTW qmf_phaseshift_sin_downsamp32[32]; extern RAM_ALIGN const FIXP_QTW qmf_phaseshift_cos64[64]; extern RAM_ALIGN const FIXP_QTW qmf_phaseshift_sin64[64]; extern RAM_ALIGN const FIXP_PFT qmf_pfilt640[QMF640_PFT_TABLE_SIZE + QMF_NO_POLY]; extern RAM_ALIGN const FIXP_PFT qmf_pfilt640_vector[640]; extern RAM_ALIGN const FIXP_QTW qmf_phaseshift_cos40[40]; extern RAM_ALIGN const FIXP_QTW qmf_phaseshift_sin40[40]; extern RAM_ALIGN const FIXP_PFT qmf_pfilt400[]; extern RAM_ALIGN const FIXP_PFT qmf_pfilt200[]; extern RAM_ALIGN const FIXP_PFT qmf_pfilt120[]; extern RAM_ALIGN const FIXP_QTW qmf_phaseshift_cos24[24]; extern RAM_ALIGN const FIXP_QTW qmf_phaseshift_sin24[24]; extern RAM_ALIGN const FIXP_PFT qmf_pfilt240[]; extern RAM_ALIGN const FIXP_QTW qmf_phaseshift_cos16[16]; extern RAM_ALIGN const FIXP_QTW qmf_phaseshift_sin16[16]; #define QMF640_CLDFB_PFT_TABLE_SIZE (640) #define QMF320_CLDFB_PFT_TABLE_SIZE (320) #define QMF_CLDFB_PFT_SCALE 1 extern const FIXP_QTW qmf_phaseshift_cos32_cldfb_ana[32]; extern const FIXP_QTW qmf_phaseshift_cos32_cldfb_syn[32]; extern const FIXP_QTW qmf_phaseshift_sin32_cldfb[32]; extern const FIXP_QTW qmf_phaseshift_cos16_cldfb_ana[16]; extern const FIXP_QTW qmf_phaseshift_cos16_cldfb_syn[16]; extern const FIXP_QTW qmf_phaseshift_sin16_cldfb[16]; extern const FIXP_QTW qmf_phaseshift_cos8_cldfb_ana[8]; extern const FIXP_QTW qmf_phaseshift_cos8_cldfb_syn[8]; extern const FIXP_QTW qmf_phaseshift_sin8_cldfb[8]; extern const FIXP_QTW qmf_phaseshift_cos64_cldfb[64]; extern const FIXP_QTW qmf_phaseshift_sin64_cldfb[64]; extern RAM_ALIGN const FIXP_PFT qmf_cldfb_640[QMF640_CLDFB_PFT_TABLE_SIZE]; extern RAM_ALIGN const FIXP_PFT qmf_cldfb_320[QMF320_CLDFB_PFT_TABLE_SIZE]; #define QMF160_CLDFB_PFT_TABLE_SIZE (160) extern RAM_ALIGN const FIXP_PFT qmf_cldfb_160[QMF160_CLDFB_PFT_TABLE_SIZE]; #define QMF80_CLDFB_PFT_TABLE_SIZE (80) extern RAM_ALIGN const FIXP_PFT qmf_cldfb_80[QMF80_CLDFB_PFT_TABLE_SIZE]; #define QMF320_MPSLDFB_PFT_TABLE_SIZE (320) #define QMF640_MPSLDFB_PFT_TABLE_SIZE (640) #define QMF_MPSLDFB_PFT_SCALE 1 extern const FIXP_PFT qmf_mpsldfb_320[QMF320_MPSLDFB_PFT_TABLE_SIZE]; extern RAM_ALIGN const FIXP_PFT qmf_mpsldfb_640[QMF640_MPSLDFB_PFT_TABLE_SIZE]; /** * Audio bitstream element specific syntax flags: */ #define AC_EL_GA_CCE 0x00000001 /*!< GA AAC coupling channel element (CCE) */ /* * Raw Data Block list items. */ typedef enum { element_instance_tag, common_window, /* -> decision for link_sequence */ global_gain, ics_info, /* ics_reserved_bit, window_sequence, window_shape, max_sfb, scale_factor_grouping, predictor_data_present, ltp_data_present, ltp_data */ max_sfb, ms, /* ms_mask_present, ms_used */ /*predictor_data_present,*/ /* part of ics_info */ ltp_data_present, ltp_data, section_data, scale_factor_data, pulse, /* pulse_data_present, pulse_data */ tns_data_present, tns_data, gain_control_data_present, gain_control_data, esc1_hcr, esc2_rvlc, spectral_data, scale_factor_data_usac, core_mode, /* -> decision for link_sequence */ common_tw, lpd_channel_stream, tw_data, noise, ac_spectral_data, fac_data, tns_active, /* introduced in MPEG-D usac CD */ tns_data_present_usac, common_max_sfb, coupled_elements, /* only for CCE parsing */ gain_element_lists, /* only for CCE parsing */ /* Non data list items */ adtscrc_start_reg1, adtscrc_start_reg2, adtscrc_end_reg1, adtscrc_end_reg2, drmcrc_start_reg, drmcrc_end_reg, next_channel, next_channel_loop, link_sequence, end_of_sequence } rbd_id_t; struct element_list { const rbd_id_t *id; const struct element_list *next[2]; }; typedef struct element_list element_list_t; /** * \brief get elementary stream pieces list for given parameters. * \param aot audio object type * \param epConfig the epConfig value from the current Audio Specific Config * \param nChannels amount of channels contained in the current element. * \param layer the layer of the current element. * \param elFlags element specific flags. * \return element_list_t parser guidance structure. */ const element_list_t *getBitstreamElementList(AUDIO_OBJECT_TYPE aot, SCHAR epConfig, UCHAR nChannels, UCHAR layer, UINT elFlags); typedef enum { /* n.a. */ FDK_FORMAT_1_0 = 1, /* mono */ FDK_FORMAT_2_0 = 2, /* stereo */ FDK_FORMAT_3_0_FC = 3, /* 3/0.0 */ FDK_FORMAT_3_1_0 = 4, /* 3/1.0 */ FDK_FORMAT_5_0 = 5, /* 3/2.0 */ FDK_FORMAT_5_1 = 6, /* 5.1 */ FDK_FORMAT_7_1_ALT = 7, /* 5/2.1 ALT */ /* 8 n.a.*/ FDK_FORMAT_3_0_RC = 9, /* 2/1.0 */ FDK_FORMAT_2_2_0 = 10, /* 2/2.0 */ FDK_FORMAT_6_1 = 11, /* 3/3.1 */ FDK_FORMAT_7_1 = 12, /* 3/4.1 */ FDK_FORMAT_22_2 = 13, /* 22.2 */ FDK_FORMAT_5_2_1 = 14, /* 5/2.1*/ FDK_FORMAT_5_5_2 = 15, /* 5/5.2 */ FDK_FORMAT_9_1 = 16, /* 5/4.1 */ FDK_FORMAT_6_5_1 = 17, /* 6/5.1 */ FDK_FORMAT_6_7_1 = 18, /* 6/7.1 */ FDK_FORMAT_5_6_1 = 19, /* 5/6.1 */ FDK_FORMAT_7_6_1 = 20, /* 7/6.1 */ FDK_FORMAT_IN_LISTOFCHANNELS = 21, FDK_FORMAT_OUT_LISTOFCHANNELS = 22, /* 20 formats + In & Out list of channels */ FDK_NFORMATS = 23, FDK_FORMAT_FAIL = -1 } FDK_converter_formatid_t; extern const INT format_nchan[FDK_NFORMATS + 9 - 2]; #endif fdk-aac-2.0.2/libFDK/include/FDK_trigFcts.h000066400000000000000000000217111404225310700201410ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Haricharan Lakshman, Manuel Jander Description: Trigonometric functions fixed point fractional implementation. *******************************************************************************/ #if !defined(FDK_TRIGFCTS_H) #define FDK_TRIGFCTS_H #include "common_fix.h" #include "FDK_tools_rom.h" /* Fixed point precision definitions */ #define Q(format) ((FIXP_DBL)(((LONG)1) << (format))) #ifndef M_PI #define M_PI (3.14159265358979323846f) #endif /*! * Inverse tangent function. */ /* --- fixp_atan() ---- */ #define Q_ATANINP (25) // Input in q25, Output in q30 #define Q_ATANOUT (30) #define ATI_SF ((DFRACT_BITS - 1) - Q_ATANINP) /* 6 */ #define ATI_SCALE ((float)(1 << ATI_SF)) #define ATO_SF ((DFRACT_BITS - 1) - Q_ATANOUT) /* 1 ] -pi/2 .. pi/2 [ */ #define ATO_SCALE ((float)(1 << ATO_SF)) /* --- fixp_atan2() --- */ #define Q_ATAN2OUT (29) #define AT2O_SF ((DFRACT_BITS - 1) - Q_ATAN2OUT) /* 2 ] -pi .. pi ] */ #define AT2O_SCALE ((float)(1 << AT2O_SF)) // -------------------- FIXP_DBL fixp_atan(FIXP_DBL x); FIXP_DBL fixp_atan2(FIXP_DBL y, FIXP_DBL x); FIXP_DBL fixp_cos(FIXP_DBL x, int scale); FIXP_DBL fixp_sin(FIXP_DBL x, int scale); #define FIXP_COS_SIN #include "FDK_tools_rom.h" #define SINETAB SineTable512 #define LD 9 #ifndef FUNCTION_inline_fixp_cos_sin #define FUNCTION_inline_fixp_cos_sin /* * Calculates coarse lookup index and sign for sine. * Returns delta x residual. */ static inline FIXP_DBL fixp_sin_cos_residual_inline(FIXP_DBL x, int scale, FIXP_DBL *sine, FIXP_DBL *cosine) { FIXP_DBL residual; int s; int shift = (31 - scale - LD - 1); int ssign = 1; int csign = 1; residual = fMult(x, FL2FXCONST_DBL(1.0 / M_PI)); s = ((LONG)residual) >> shift; residual &= ((1 << shift) - 1); residual = fMult(residual, FL2FXCONST_DBL(M_PI / 4.0)) << 2; residual <<= scale; /* Sine sign symmetry */ if (s & ((1 << LD) << 1)) { ssign = -ssign; } /* Cosine sign symmetry */ if ((s + (1 << LD)) & ((1 << LD) << 1)) { csign = -csign; } s = fAbs(s); s &= (((1 << LD) << 1) - 1); /* Modulo PI */ if (s > (1 << LD)) { s = ((1 << LD) << 1) - s; } { LONG sl, cl; /* Because of packed table */ if (s > (1 << (LD - 1))) { FIXP_STP tmp; /* Cosine/Sine simetry for angles greater than PI/4 */ s = (1 << LD) - s; tmp = SINETAB[s]; sl = (LONG)tmp.v.re; cl = (LONG)tmp.v.im; } else { FIXP_STP tmp; tmp = SINETAB[s]; sl = (LONG)tmp.v.im; cl = (LONG)tmp.v.re; } #ifdef SINETABLE_16BIT *sine = (FIXP_DBL)((sl * ssign) << (DFRACT_BITS - FRACT_BITS)); *cosine = (FIXP_DBL)((cl * csign) << (DFRACT_BITS - FRACT_BITS)); #else /* scale down by 1 for overflow prevention. This is undone at the calling * function. */ *sine = (FIXP_DBL)(sl * ssign) >> 1; *cosine = (FIXP_DBL)(cl * csign) >> 1; #endif } return residual; } /** * \brief Calculate cosine and sine value each of 2 angles different angle * values. * \param x1 first angle value * \param x2 second angle value * \param scale exponent of x1 and x2 * \param out pointer to 4 FIXP_DBL locations, were the values cos(x1), sin(x1), * cos(x2), sin(x2) will be stored into. */ static inline void inline_fixp_cos_sin(FIXP_DBL x1, FIXP_DBL x2, const int scale, FIXP_DBL *out) { FIXP_DBL residual, error0, error1, sine, cosine; residual = fixp_sin_cos_residual_inline(x1, scale, &sine, &cosine); error0 = fMultDiv2(sine, residual); error1 = fMultDiv2(cosine, residual); #ifdef SINETABLE_16BIT *out++ = cosine - (error0 << 1); *out++ = sine + (error1 << 1); #else /* Undo downscaling by 1 which was done at fixp_sin_cos_residual_inline */ *out++ = SATURATE_LEFT_SHIFT(cosine - (error0 << 1), 1, DFRACT_BITS); *out++ = SATURATE_LEFT_SHIFT(sine + (error1 << 1), 1, DFRACT_BITS); #endif residual = fixp_sin_cos_residual_inline(x2, scale, &sine, &cosine); error0 = fMultDiv2(sine, residual); error1 = fMultDiv2(cosine, residual); #ifdef SINETABLE_16BIT *out++ = cosine - (error0 << 1); *out++ = sine + (error1 << 1); #else *out++ = SATURATE_LEFT_SHIFT(cosine - (error0 << 1), 1, DFRACT_BITS); *out++ = SATURATE_LEFT_SHIFT(sine + (error1 << 1), 1, DFRACT_BITS); #endif } #endif #endif /* !defined(FDK_TRIGFCTS_H) */ fdk-aac-2.0.2/libFDK/include/abs.h000066400000000000000000000131451404225310700164370ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): M. Lohwasser Description: fixed point abs definitions *******************************************************************************/ #if !defined(ABS_H) #define ABS_H #if defined(__mips__) #include "mips/abs_mips.h" #elif defined(__x86__) #include "x86/abs_x86.h" #endif /* all cores */ /************************************************************************* ************************************************************************* Software fallbacks for missing functions ************************************************************************** **************************************************************************/ #if !defined(FUNCTION_fixabs_D) inline FIXP_DBL fixabs_D(FIXP_DBL x) { return ((x) > (FIXP_DBL)(0)) ? (x) : -(x); } #endif #if !defined(FUNCTION_fixabs_I) inline INT fixabs_I(INT x) { return ((x) > (INT)(0)) ? (x) : -(x); } #endif #if !defined(FUNCTION_fixabs_S) inline FIXP_SGL fixabs_S(FIXP_SGL x) { return ((x) > (FIXP_SGL)(0)) ? (x) : -(x); } #endif #endif /* ABS_H */ fdk-aac-2.0.2/libFDK/include/arm/000077500000000000000000000000001404225310700162745ustar00rootroot00000000000000fdk-aac-2.0.2/libFDK/include/arm/clz_arm.h000066400000000000000000000137751404225310700201110ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Description: fixed point intrinsics *******************************************************************************/ #if !defined(CLZ_ARM_H) #define CLZ_ARM_H #if defined(__arm__) #if defined(__GNUC__) /* ARM gcc*/ #if defined(__ARM_ARCH_5TE__) || defined(__ARM_ARCH_8__) #define FUNCTION_fixnormz_D #define FUNCTION_fixnorm_D #define FUNCTION_fixnormz_S #define FUNCTION_fixnorm_S #ifdef FUNCTION_fixnormz_D inline INT fixnormz_D(LONG value) { INT result; #if defined(__ARM_ARCH_8__) asm("clz %w0, %w1 " : "=r"(result) : "r"(value)); #else asm("clz %0, %1 " : "=r"(result) : "r"(value)); #endif return result; } #endif /* #ifdef FUNCTION_fixnormz_D */ #ifdef FUNCTION_fixnorm_D inline INT fixnorm_D(LONG value) { if (!value) return 0; if (value < 0) value = ~value; return fixnormz_D(value) - 1; } #endif /* #ifdef FUNCTION_fixnorm_D */ #ifdef FUNCTION_fixnormz_S inline INT fixnormz_S(SHORT value) { INT result; result = (LONG)(value << 16); if (result == 0) result = 16; else result = fixnormz_D(result); return result; } #endif /* #ifdef FUNCTION_fixnormz_S */ #ifdef FUNCTION_fixnorm_S inline INT fixnorm_S(SHORT value) { LONG lvalue = (LONG)(value << 16); if (!lvalue) return 0; if (lvalue < 0) lvalue = ~lvalue; return fixnormz_D(lvalue) - 1; } #endif /* #ifdef FUNCTION_fixnorm_S */ #endif #endif /* arm toolchain */ #endif /* __arm__ */ #endif /* !defined(CLZ_ARM_H) */ fdk-aac-2.0.2/libFDK/include/arm/cplx_mul_arm.h000066400000000000000000000200561404225310700211320ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Description: fixed point intrinsics *******************************************************************************/ #if !defined(CPLX_MUL_ARM_H) #define CPLX_MUL_ARM_H #if defined(__arm__) && defined(__GNUC__) #if defined(__ARM_ARCH_5TE__) || defined(__ARM_ARCH_6__) || \ defined(__ARM_ARCH_8__) #define FUNCTION_cplxMultDiv2_32x16 #define FUNCTION_cplxMultDiv2_32x16X2 #endif #define FUNCTION_cplxMultDiv2_32x32X2 #ifdef FUNCTION_cplxMultDiv2_32x32X2 inline void cplxMultDiv2(FIXP_DBL *c_Re, FIXP_DBL *c_Im, const FIXP_DBL a_Re, const FIXP_DBL a_Im, const FIXP_DBL b_Re, const FIXP_DBL b_Im) { LONG tmp1, tmp2; #ifdef __ARM_ARCH_8__ asm("smull %x0, %w2, %w4; \n" /* tmp1 = a_Re * b_Re */ "smull %x1, %w2, %w5; \n" /* tmp2 = a_Re * b_Im */ "smsubl %x0, %w3, %w5, %x0; \n" /* tmp1 -= a_Im * b_Im */ "smaddl %x1, %w3, %w4, %x1; \n" /* tmp2 += a_Im * b_Re */ "asr %x0, %x0, #32 \n" "asr %x1, %x1, #32 \n" : "=&r"(tmp1), "=&r"(tmp2) : "r"(a_Re), "r"(a_Im), "r"(b_Re), "r"(b_Im)); #elif defined(__ARM_ARCH_6__) asm("smmul %0, %2, %4;\n" /* tmp1 = a_Re * b_Re */ "smmls %0, %3, %5, %0;\n" /* tmp1 -= a_Im * b_Im */ "smmul %1, %2, %5;\n" /* tmp2 = a_Re * b_Im */ "smmla %1, %3, %4, %1;\n" /* tmp2 += a_Im * b_Re */ : "=&r"(tmp1), "=&r"(tmp2) : "r"(a_Re), "r"(a_Im), "r"(b_Re), "r"(b_Im)); #else LONG discard; asm("smull %2, %0, %7, %6;\n" /* tmp1 = -a_Im * b_Im */ "smlal %2, %0, %3, %5;\n" /* tmp1 += a_Re * b_Re */ "smull %2, %1, %3, %6;\n" /* tmp2 = a_Re * b_Im */ "smlal %2, %1, %4, %5;\n" /* tmp2 += a_Im * b_Re */ : "=&r"(tmp1), "=&r"(tmp2), "=&r"(discard) : "r"(a_Re), "r"(a_Im), "r"(b_Re), "r"(b_Im), "r"(-a_Im)); #endif *c_Re = tmp1; *c_Im = tmp2; } #endif /* FUNCTION_cplxMultDiv2_32x32X2 */ #if defined(FUNCTION_cplxMultDiv2_32x16) inline void cplxMultDiv2(FIXP_DBL *c_Re, FIXP_DBL *c_Im, const FIXP_DBL a_Re, const FIXP_DBL a_Im, FIXP_SPK wpk) { #ifdef __ARM_ARCH_8__ FIXP_DBL b_Im = FX_SGL2FX_DBL(wpk.v.im); FIXP_DBL b_Re = FX_SGL2FX_DBL(wpk.v.re); cplxMultDiv2(c_Re, c_Im, a_Re, a_Im, b_Re, b_Im); #else LONG tmp1, tmp2; const LONG w = wpk.w; asm("smulwt %0, %3, %4;\n" "rsb %1,%0,#0;\n" "smlawb %0, %2, %4, %1;\n" "smulwt %1, %2, %4;\n" "smlawb %1, %3, %4, %1;\n" : "=&r"(tmp1), "=&r"(tmp2) : "r"(a_Re), "r"(a_Im), "r"(w)); *c_Re = tmp1; *c_Im = tmp2; #endif } #endif /* FUNCTION_cplxMultDiv2_32x16 */ #ifdef FUNCTION_cplxMultDiv2_32x16X2 inline void cplxMultDiv2(FIXP_DBL *c_Re, FIXP_DBL *c_Im, const FIXP_DBL a_Re, const FIXP_DBL a_Im, const FIXP_SGL b_Re, const FIXP_SGL b_Im) { #ifdef __ARM_ARCH_8__ FIXP_DBL b_re = FX_SGL2FX_DBL(b_Re); FIXP_DBL b_im = FX_SGL2FX_DBL(b_Im); cplxMultDiv2(c_Re, c_Im, a_Re, a_Im, b_re, b_im); #else LONG tmp1, tmp2; asm("smulwb %0, %3, %5;\n" /* %7 = -a_Im * b_Im */ "rsb %1,%0,#0;\n" "smlawb %0, %2, %4, %1;\n" /* tmp1 = a_Re * b_Re - a_Im * b_Im */ "smulwb %1, %2, %5;\n" /* %7 = a_Re * b_Im */ "smlawb %1, %3, %4, %1;\n" /* tmp2 = a_Im * b_Re + a_Re * b_Im */ : "=&r"(tmp1), "=&r"(tmp2) : "r"(a_Re), "r"(a_Im), "r"(b_Re), "r"(b_Im)); *c_Re = tmp1; *c_Im = tmp2; #endif } #endif /* FUNCTION_cplxMultDiv2_32x16X2 */ #endif #endif /* !defined(CPLX_MUL_ARM_H) */ fdk-aac-2.0.2/libFDK/include/arm/fixmadd_arm.h000066400000000000000000000206131404225310700207220ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Description: fixed point intrinsics *******************************************************************************/ #if !defined(FIXMADD_ARM_H) #define FIXMADD_ARM_H #if defined(__arm__) /* ############################################################################# */ #if defined(__GNUC__) && defined(__arm__) /* ############################################################################# */ /* ARM GNU GCC */ #ifdef __ARM_ARCH_8__ #define FUNCTION_fixmadddiv2_DD #ifdef FUNCTION_fixmadddiv2_DD inline FIXP_DBL fixmadddiv2_DD(FIXP_DBL x, const FIXP_DBL a, const FIXP_DBL b) { INT64 result; asm("smull %x0, %w1, %w2; \n" "asr %x0, %x0, #32; \n" "add %w0, %w3, %w0; \n" : "=&r"(result) : "r"(a), "r"(b), "r"(x)); return (INT)result; } #endif /* #ifdef FUNCTION_fixmadddiv2_DD */ #define FUNCTION_fixmsubdiv2_DD #ifdef FUNCTION_fixmsubdiv2_DD inline FIXP_DBL fixmsubdiv2_DD(FIXP_DBL x, const FIXP_DBL a, const FIXP_DBL b) { INT64 result; asm("smull %x0, %w1, %w2; \n" "asr %x0, %x0, #32; \n" "sub %w0, %w3, %w0; \n" : "=&r"(result) : "r"(a), "r"(b), "r"(x)); return (INT)result; } #endif /* #ifdef FUNCTION_fixmsubdiv2_DD */ #elif defined(__ARM_ARCH_6__) #define FUNCTION_fixmadddiv2_DD #ifdef FUNCTION_fixmadddiv2_DD inline FIXP_DBL fixmadddiv2_DD(FIXP_DBL x, const FIXP_DBL a, const FIXP_DBL b) { INT result; asm("smmla %0, %1, %2, %3;\n" : "=r"(result) : "r"(a), "r"(b), "r"(x)); return result; } #endif /* #ifdef FUNCTION_fixmadddiv2_DD */ #define FUNCTION_fixmsubdiv2_DD #ifdef FUNCTION_fixmsubdiv2_DD inline FIXP_DBL fixmsubdiv2_DD(FIXP_DBL x, const FIXP_DBL a, const FIXP_DBL b) { INT result; asm("smmls %0, %1, %2, %3;\n" : "=r"(result) : "r"(a), "r"(b), "r"(x)); return result; } #endif /* #ifdef FUNCTION_fixmsubdiv2_DD */ #else #define FUNCTION_fixmadddiv2_DD #ifdef FUNCTION_fixmadddiv2_DD inline FIXP_DBL fixmadddiv2_DD(FIXP_DBL x, const FIXP_DBL a, const FIXP_DBL b) { INT discard = 0; INT result = x; asm("smlal %0, %1, %2, %3;\n" : "+r"(discard), "+r"(result) : "r"(a), "r"(b)); return result; } #endif /* #ifdef FUNCTION_fixmadddiv2_DD */ #endif #if defined(__ARM_ARCH_5TE__) || defined(__ARM_ARCH_6__) #define FUNCTION_fixmadddiv2_DS #ifdef FUNCTION_fixmadddiv2_DS inline FIXP_DBL fixmadddiv2_DS(FIXP_DBL x, const FIXP_DBL a, const FIXP_SGL b) { INT result; asm("smlawb %0, %1, %2, %3 " : "=r"(result) : "r"(a), "r"(b), "r"(x)); return result; } #endif /* #ifdef FUNCTION_fixmadddiv2_DS */ #endif /* defined(__ARM_ARCH_5TE__) || defined(__ARM_ARCH_6__) */ #define FUNCTION_fixmadddiv2BitExact_DD #ifdef FUNCTION_fixmadddiv2BitExact_DD #define fixmadddiv2BitExact_DD(a, b, c) fixmadddiv2_DD(a, b, c) #endif /* #ifdef FUNCTION_fixmadddiv2BitExact_DD */ #define FUNCTION_fixmsubdiv2BitExact_DD #ifdef FUNCTION_fixmsubdiv2BitExact_DD inline FIXP_DBL fixmsubdiv2BitExact_DD(FIXP_DBL x, const FIXP_DBL a, const FIXP_DBL b) { return x - fixmuldiv2BitExact_DD(a, b); } #endif /* #ifdef FUNCTION_fixmsubdiv2BitExact_DD */ #define FUNCTION_fixmadddiv2BitExact_DS #ifdef FUNCTION_fixmadddiv2BitExact_DS #define fixmadddiv2BitExact_DS(a, b, c) fixmadddiv2_DS(a, b, c) #endif /* #ifdef FUNCTION_fixmadddiv2BitExact_DS */ #define FUNCTION_fixmsubdiv2BitExact_DS #ifdef FUNCTION_fixmsubdiv2BitExact_DS inline FIXP_DBL fixmsubdiv2BitExact_DS(FIXP_DBL x, const FIXP_DBL a, const FIXP_SGL b) { return x - fixmuldiv2BitExact_DS(a, b); } #endif /* #ifdef FUNCTION_fixmsubdiv2BitExact_DS */ /* ############################################################################# */ #endif /* toolchain */ /* ############################################################################# */ #endif /* __arm__ */ #endif /* !defined(FIXMADD_ARM_H) */ fdk-aac-2.0.2/libFDK/include/arm/fixmul_arm.h000066400000000000000000000164361404225310700206220ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Description: fixed point intrinsics *******************************************************************************/ #if !defined(FIXMUL_ARM_H) #define FIXMUL_ARM_H #if defined(__arm__) #if defined(__GNUC__) && defined(__arm__) /* ARM with GNU compiler */ #define FUNCTION_fixmuldiv2_DD #define FUNCTION_fixmuldiv2BitExact_DD #ifdef FUNCTION_fixmuldiv2BitExact_DD #define fixmuldiv2BitExact_DD(a, b) fixmuldiv2_DD(a, b) #endif /* #ifdef FUNCTION_fixmuldiv2BitExact_DD */ #define FUNCTION_fixmulBitExact_DD #ifdef FUNCTION_fixmulBitExact_DD #define fixmulBitExact_DD(a, b) (fixmuldiv2BitExact_DD(a, b) << 1) #endif /* #ifdef FUNCTION_fixmulBitExact_DD */ #define FUNCTION_fixmuldiv2BitExact_DS #ifdef FUNCTION_fixmuldiv2BitExact_DS #define fixmuldiv2BitExact_DS(a, b) fixmuldiv2_DS(a, b) #endif /* #ifdef FUNCTION_fixmuldiv2BitExact_DS */ #define FUNCTION_fixmulBitExact_DS #ifdef FUNCTION_fixmulBitExact_DS #define fixmulBitExact_DS(a, b) fixmul_DS(a, b) #endif /* #ifdef FUNCTION_fixmulBitExact_DS */ #ifdef FUNCTION_fixmuldiv2_DD inline INT fixmuldiv2_DD(const INT a, const INT b) { INT result; #if defined(__ARM_ARCH_8__) INT64 result64; __asm__( "smull %x0, %w1, %w2;\n" "asr %x0, %x0, #32; " : "=r"(result64) : "r"(a), "r"(b)); result = (INT)result64; #elif defined(__ARM_ARCH_6__) || defined(__TARGET_ARCH_7E_M) __asm__("smmul %0, %1, %2" : "=r"(result) : "r"(a), "r"(b)); #else INT discard; __asm__("smull %0, %1, %2, %3" : "=&r"(discard), "=r"(result) : "r"(a), "r"(b)); #endif return result; } #endif /* #ifdef FUNCTION_fixmuldiv2_DD */ #if defined(__ARM_ARCH_8__) #define FUNCTION_fixmuldiv2_SD #ifdef FUNCTION_fixmuldiv2_SD inline INT fixmuldiv2_SD(const SHORT a, const INT b) { return fixmuldiv2_DD((INT)(a << 16), b); } #endif /* #ifdef FUNCTION_fixmuldiv2_SD */ #elif defined(__ARM_ARCH_5TE__) || defined(__ARM_ARCH_6__) #define FUNCTION_fixmuldiv2_SD #ifdef FUNCTION_fixmuldiv2_SD inline INT fixmuldiv2_SD(const SHORT a, const INT b) { INT result; __asm__("smulwb %0, %1, %2" : "=r"(result) : "r"(b), "r"(a)); return result; } #endif /* #ifdef FUNCTION_fixmuldiv2_SD */ #endif #define FUNCTION_fixmul_DD #ifdef FUNCTION_fixmul_DD #if defined(__ARM_ARCH_8__) inline INT fixmul_DD(const INT a, const INT b) { INT64 result64; __asm__( "smull %x0, %w1, %w2;\n" "asr %x0, %x0, #31; " : "=r"(result64) : "r"(a), "r"(b)); return (INT)result64; } #else inline INT fixmul_DD(const INT a, const INT b) { return (fixmuldiv2_DD(a, b) << 1); } #endif /* __ARM_ARCH_8__ */ #endif /* #ifdef FUNCTION_fixmul_DD */ #endif /* defined(__GNUC__) && defined(__arm__) */ #endif /* __arm__ */ #endif /* !defined(FIXMUL_ARM_H) */ fdk-aac-2.0.2/libFDK/include/arm/scale_arm.h000066400000000000000000000147541404225310700204060ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Description: ARM scaling operations *******************************************************************************/ #if !defined(SCALE_ARM_H) #define SCALE_ARM_H #if defined(__GNUC__) /* GCC Compiler */ #if defined(__ARM_ARCH_6__) inline static INT shiftRightSat(INT src, int scale) { INT result; asm("ssat %0,%2,%0;\n" : "=&r"(result) : "r"(src >> scale), "M"(SAMPLE_BITS)); return result; } #define SATURATE_INT_PCM_RIGHT_SHIFT(src, scale) shiftRightSat(src, scale) inline static INT shiftLeftSat(INT src, int scale) { INT result; asm("ssat %0,%2,%0;\n" : "=&r"(result) : "r"(src << scale), "M"(SAMPLE_BITS)); return result; } #define SATURATE_INT_PCM_LEFT_SHIFT(src, scale) shiftLeftSat(src, scale) #endif /* __ARM_ARCH_6__ */ #endif /* compiler selection */ #define FUNCTION_scaleValueInPlace #ifdef FUNCTION_scaleValueInPlace inline void scaleValueInPlace(FIXP_DBL *value, /*!< Value */ INT scalefactor /*!< Scalefactor */ ) { INT newscale; if ((newscale = scalefactor) >= 0) *value <<= newscale; else *value >>= -newscale; } #endif /* #ifdef FUNCTION_scaleValueInPlace */ #define SATURATE_RIGHT_SHIFT(src, scale, dBits) \ ((((LONG)(src) ^ ((LONG)(src) >> (DFRACT_BITS - 1))) >> (scale)) > \ (LONG)(((1U) << ((dBits)-1)) - 1)) \ ? ((LONG)(src) >> (DFRACT_BITS - 1)) ^ (LONG)(((1U) << ((dBits)-1)) - 1) \ : ((LONG)(src) >> (scale)) #define SATURATE_LEFT_SHIFT(src, scale, dBits) \ (((LONG)(src) ^ ((LONG)(src) >> (DFRACT_BITS - 1))) > \ ((LONG)(((1U) << ((dBits)-1)) - 1) >> (scale))) \ ? ((LONG)(src) >> (DFRACT_BITS - 1)) ^ (LONG)(((1U) << ((dBits)-1)) - 1) \ : ((LONG)(src) << (scale)) #endif /* !defined(SCALE_ARM_H) */ fdk-aac-2.0.2/libFDK/include/arm/scramble_arm.h000066400000000000000000000163261404225310700211040ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Description: bitreversal of input data *******************************************************************************/ #if !defined(SCRAMBLE_ARM_H) #define SCRAMBLE_ARM_H #if defined(FUNCTION_scramble) #if defined(__GNUC__) #define FUNCTION_scramble #if defined(__ARM_ARCH_5TE__) #define USE_LDRD_STRD /* LDRD requires 8 byte data alignment. */ #endif inline void scramble(FIXP_DBL x[], INT n) { FDK_ASSERT(!(((INT)x) & (ALIGNMENT_DEFAULT - 1))); asm("mov r2, #1;\n" /* r2(m) = 1; */ "sub r3, %1, #1;\n" /* r3 = n-1; */ "mov r4, #0;\n" /* r4(j) = 0; */ "scramble_m_loop%=:\n" /* { */ "mov r5, %1;\n" /* r5(k) = 1; */ "scramble_k_loop%=:\n" /* { */ "mov r5, r5, lsr #1;\n" /* k >>= 1; */ "eor r4, r4, r5;\n" /* j ^=k; */ "ands r10, r4, r5;\n" /* r10 = r4 & r5; */ "beq scramble_k_loop%=;\n" /* } while (r10 == 0); */ "cmp r4, r2;\n" /* if (r4 < r2) break; */ "bcc scramble_m_loop_end%=;\n" #ifdef USE_LDRD_STRD "mov r5, r2, lsl #3;\n" /* m(r5) = r2*4*2 */ "ldrd r10, [%0, r5];\n" /* r10 = x[r5], x7 = x[r5+1] */ "mov r6, r4, lsl #3;\n" /* j(r6) = r4*4*2 */ "ldrd r8, [%0, r6];\n" /* r8 = x[r6], r9 = x[r6+1]; */ "strd r10, [%0, r6];\n" /* x[r6,r6+1] = r10,r11; */ "strd r8, [%0, r5];\n" /* x[r5,r5+1] = r8,r9; */ #else "mov r5, r2, lsl #3;\n" /* m(r5) = r2*4*2 */ "ldr r10, [%0, r5];\n" "mov r6, r4, lsl #3;\n" /* j(r6) = r4*4*2 */ "ldr r11, [%0, r6];\n" "str r10, [%0, r6];\n" "str r11, [%0, r5];\n" "add r5, r5, #4;" "ldr r10, [%0, r5];\n" "add r6, r6, #4;" "ldr r11, [%0, r6];\n" "str r10, [%0, r6];\n" "str r11, [%0, r5];\n" #endif "scramble_m_loop_end%=:\n" "add r2, r2, #1;\n" /* r2++; */ "cmp r2, r3;\n" "bcc scramble_m_loop%=;\n" /* } while (r2(m) < r3(n-1)); */ : : "r"(x), "r"(n) #ifdef USE_LDRD_STRD : "r2", "r3", "r4", "r5", "r10", "r11", "r8", "r9", "r6"); #else : "r2", "r3", "r4", "r5", "r10", "r11", "r6"); #endif } #else /* Force C implementation if no assembler version available. */ #undef FUNCTION_scramble #endif /* Toolchain selection. */ #endif /* defined(FUNCTION_scramble) */ #endif /* !defined(SCRAMBLE_ARM_H) */ fdk-aac-2.0.2/libFDK/include/autocorr2nd.h000066400000000000000000000131241404225310700201310ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): M. Lohwasser Description: fixed point abs definitions *******************************************************************************/ #ifndef AUTOCORR2ND_H #define AUTOCORR2ND_H #include "common_fix.h" typedef struct { FIXP_DBL r00r; FIXP_DBL r11r; FIXP_DBL r22r; FIXP_DBL r01r; FIXP_DBL r02r; FIXP_DBL r12r; FIXP_DBL r01i; FIXP_DBL r02i; FIXP_DBL r12i; FIXP_DBL det; int det_scale; } ACORR_COEFS; #define LPC_ORDER 2 INT autoCorr2nd_real( ACORR_COEFS *ac, /*!< Pointer to autocorrelation coeffs */ const FIXP_DBL *reBuffer, /*!< Pointer to to real part of spectrum */ const int len /*!< Number of qmf slots */ ); INT autoCorr2nd_cplx( ACORR_COEFS *ac, /*!< Pointer to autocorrelation coeffs */ const FIXP_DBL *reBuffer, /*!< Pointer to to real part of spectrum */ const FIXP_DBL *imBuffer, /*!< Pointer to imag part of spectrum */ const int len /*!< Number of qmf slots */ ); #endif /* AUTOCORR2ND_H */ fdk-aac-2.0.2/libFDK/include/clz.h000066400000000000000000000156051404225310700164650ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Marc Gayer Description: fixed point intrinsics *******************************************************************************/ #if !defined(CLZ_H) #define CLZ_H #include "FDK_archdef.h" #include "machine_type.h" #if defined(__arm__) #include "arm/clz_arm.h" #elif defined(__mips__) #include "mips/clz_mips.h" #elif defined(__x86__) #include "x86/clz_x86.h" #elif defined(__powerpc__) #include "ppc/clz_ppc.h" #endif /* all cores */ /************************************************************************* ************************************************************************* Software fallbacks for missing functions. ************************************************************************** **************************************************************************/ #if !defined(FUNCTION_fixnormz_S) #ifdef FUNCTION_fixnormz_D inline INT fixnormz_S(SHORT a) { if (a < 0) { return 0; } return fixnormz_D((INT)(a)) - 16; } #else inline INT fixnormz_S(SHORT a) { int leadingBits = 0; a = ~a; while (a & 0x8000) { leadingBits++; a <<= 1; } return (leadingBits); } #endif #endif #if !defined(FUNCTION_fixnormz_D) inline INT fixnormz_D(LONG a) { INT leadingBits = 0; a = ~a; while (a & 0x80000000) { leadingBits++; a <<= 1; } return (leadingBits); } #endif /***************************************************************************** functionname: fixnorm_D description: Count leading ones or zeros of operand val for dfract/LONG INT values. Return this value minus 1. Return 0 if operand==0. *****************************************************************************/ #if !defined(FUNCTION_fixnorm_S) #ifdef FUNCTION_fixnorm_D inline INT fixnorm_S(FIXP_SGL val) { if (val == (FIXP_SGL)0) { return 0; } return fixnorm_D((INT)(val)) - 16; } #else inline INT fixnorm_S(FIXP_SGL val) { INT leadingBits = 0; if (val != (FIXP_SGL)0) { if (val < (FIXP_SGL)0) { val = ~val; } leadingBits = fixnormz_S(val) - 1; } return (leadingBits); } #endif #endif #if !defined(FUNCTION_fixnorm_D) inline INT fixnorm_D(FIXP_DBL val) { INT leadingBits = 0; if (val != (FIXP_DBL)0) { if (val < (FIXP_DBL)0) { val = ~val; } leadingBits = fixnormz_D(val) - 1; } return (leadingBits); } #endif #endif /* CLZ_H */ fdk-aac-2.0.2/libFDK/include/common_fix.h000066400000000000000000000430401404225310700200250ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): M. Lohwasser, M. Gayer Description: Flexible fixpoint library configuration *******************************************************************************/ #ifndef COMMON_FIX_H #define COMMON_FIX_H #include "FDK_archdef.h" #include "machine_type.h" /* ***** Start of former fix.h ****** */ /* Define bit sizes of integer fixpoint fractional data types */ #define FRACT_BITS 16 /* single precision */ #define DFRACT_BITS 32 /* double precision */ #define ACCU_BITS 40 /* double precision plus overflow */ /* Fixpoint equivalent type fot PCM audio time domain data. */ #if defined(SAMPLE_BITS) #if (SAMPLE_BITS == DFRACT_BITS) #define FIXP_PCM FIXP_DBL #define MAXVAL_FIXP_PCM MAXVAL_DBL #define MINVAL_FIXP_PCM MINVAL_DBL #define FX_PCM2FX_DBL(x) ((FIXP_DBL)(x)) #define FX_DBL2FX_PCM(x) ((INT_PCM)(x)) #elif (SAMPLE_BITS == FRACT_BITS) #define FIXP_PCM FIXP_SGL #define MAXVAL_FIXP_PCM MAXVAL_SGL #define MINVAL_FIXP_PCM MINVAL_SGL #define FX_PCM2FX_DBL(x) FX_SGL2FX_DBL((FIXP_SGL)(x)) #define FX_DBL2FX_PCM(x) FX_DBL2FX_SGL(x) #else #error SAMPLE_BITS different from FRACT_BITS or DFRACT_BITS not implemented! #endif #endif /* ****** End of former fix.h ****** */ #define SGL_MASK ((1UL << FRACT_BITS) - 1) /* 16bit: (2^16)-1 = 0xFFFF */ #define MAX_SHIFT_SGL \ (FRACT_BITS - 1) /* maximum possible shift for FIXP_SGL values */ #define MAX_SHIFT_DBL \ (DFRACT_BITS - 1) /* maximum possible shift for FIXP_DBL values */ /* Scale factor from/to float/fixpoint values. DO NOT USE THESE VALUES AS * SATURATION LIMITS !! */ #define FRACT_FIX_SCALE ((INT64(1) << (FRACT_BITS - 1))) #define DFRACT_FIX_SCALE ((INT64(1) << (DFRACT_BITS - 1))) /* Max and Min values for saturation purposes. DO NOT USE THESE VALUES AS SCALE * VALUES !! */ #define MAXVAL_SGL \ ((signed)0x00007FFF) /* this has to be synchronized to FRACT_BITS */ #define MINVAL_SGL \ ((signed)0xFFFF8000) /* this has to be synchronized to FRACT_BITS */ #define MAXVAL_DBL \ ((signed)0x7FFFFFFF) /* this has to be synchronized to DFRACT_BITS */ #define MINVAL_DBL \ ((signed)0x80000000) /* this has to be synchronized to DFRACT_BITS */ #define FX_DBL2FXCONST_SGL(val) \ ((((((val) >> (DFRACT_BITS - FRACT_BITS - 1)) + 1) > \ (((LONG)1 << FRACT_BITS) - 1)) && \ ((LONG)(val) > 0)) \ ? (FIXP_SGL)(SHORT)(((LONG)1 << (FRACT_BITS - 1)) - 1) \ : (FIXP_SGL)(SHORT)((((val) >> (DFRACT_BITS - FRACT_BITS - 1)) + 1) >> \ 1)) #define shouldBeUnion union /* unions are possible */ typedef SHORT FIXP_SGL; typedef LONG FIXP_DBL; /* macros for compile-time conversion of constant float values to fixedpoint */ #define FL2FXCONST_SPC FL2FXCONST_DBL #define MINVAL_DBL_CONST MINVAL_DBL #define MINVAL_SGL_CONST MINVAL_SGL #define FL2FXCONST_SGL(val) \ (FIXP_SGL)( \ ((val) >= 0) \ ? ((((double)(val) * (FRACT_FIX_SCALE) + 0.5) >= \ (double)(MAXVAL_SGL)) \ ? (SHORT)(MAXVAL_SGL) \ : (SHORT)((double)(val) * (double)(FRACT_FIX_SCALE) + 0.5)) \ : ((((double)(val) * (FRACT_FIX_SCALE)-0.5) <= \ (double)(MINVAL_SGL_CONST)) \ ? (SHORT)(MINVAL_SGL_CONST) \ : (SHORT)((double)(val) * (double)(FRACT_FIX_SCALE)-0.5))) #define FL2FXCONST_DBL(val) \ (FIXP_DBL)( \ ((val) >= 0) \ ? ((((double)(val) * (DFRACT_FIX_SCALE) + 0.5) >= \ (double)(MAXVAL_DBL)) \ ? (LONG)(MAXVAL_DBL) \ : (LONG)((double)(val) * (double)(DFRACT_FIX_SCALE) + 0.5)) \ : ((((double)(val) * (DFRACT_FIX_SCALE)-0.5) <= \ (double)(MINVAL_DBL_CONST)) \ ? (LONG)(MINVAL_DBL_CONST) \ : (LONG)((double)(val) * (double)(DFRACT_FIX_SCALE)-0.5))) /* macros for runtime conversion of float values to integer fixedpoint. NO * OVERFLOW CHECK!!! */ #define FL2FX_SPC FL2FX_DBL #define FL2FX_SGL(val) \ ((val) > 0.0f ? (SHORT)((val) * (float)(FRACT_FIX_SCALE) + 0.5f) \ : (SHORT)((val) * (float)(FRACT_FIX_SCALE)-0.5f)) #define FL2FX_DBL(val) \ ((val) > 0.0f ? (LONG)((val) * (float)(DFRACT_FIX_SCALE) + 0.5f) \ : (LONG)((val) * (float)(DFRACT_FIX_SCALE)-0.5f)) /* macros for runtime conversion of fixedpoint values to other fixedpoint. NO * ROUNDING!!! */ #define FX_ACC2FX_SGL(val) ((FIXP_SGL)((val) >> (ACCU_BITS - FRACT_BITS))) #define FX_ACC2FX_DBL(val) ((FIXP_DBL)((val) >> (ACCU_BITS - DFRACT_BITS))) #define FX_SGL2FX_ACC(val) ((FIXP_ACC)((LONG)(val) << (ACCU_BITS - FRACT_BITS))) #define FX_SGL2FX_DBL(val) \ ((FIXP_DBL)((LONG)(val) << (DFRACT_BITS - FRACT_BITS))) #define FX_DBL2FX_SGL(val) ((FIXP_SGL)((val) >> (DFRACT_BITS - FRACT_BITS))) /* ############################################################# */ /* macros for runtime conversion of integer fixedpoint values to float. */ /* #define FX_DBL2FL(val) ((float)(pow(2.,-31.)*(float)val)) */ /* version #1 */ #define FX_DBL2FL(val) \ ((float)((double)(val) / (double)DFRACT_FIX_SCALE)) /* version #2 - \ identical to class \ dfract cast from \ dfract to float */ #define FX_DBL2DOUBLE(val) (((double)(val) / (double)DFRACT_FIX_SCALE)) /* ############################################################# */ #include "fixmul.h" FDK_INLINE LONG fMult(SHORT a, SHORT b) { return fixmul_SS(a, b); } FDK_INLINE LONG fMult(SHORT a, LONG b) { return fixmul_SD(a, b); } FDK_INLINE LONG fMult(LONG a, SHORT b) { return fixmul_DS(a, b); } FDK_INLINE LONG fMult(LONG a, LONG b) { return fixmul_DD(a, b); } FDK_INLINE LONG fPow2(LONG a) { return fixpow2_D(a); } FDK_INLINE LONG fPow2(SHORT a) { return fixpow2_S(a); } FDK_INLINE LONG fMultDiv2(SHORT a, SHORT b) { return fixmuldiv2_SS(a, b); } FDK_INLINE LONG fMultDiv2(SHORT a, LONG b) { return fixmuldiv2_SD(a, b); } FDK_INLINE LONG fMultDiv2(LONG a, SHORT b) { return fixmuldiv2_DS(a, b); } FDK_INLINE LONG fMultDiv2(LONG a, LONG b) { return fixmuldiv2_DD(a, b); } FDK_INLINE LONG fPow2Div2(LONG a) { return fixpow2div2_D(a); } FDK_INLINE LONG fPow2Div2(SHORT a) { return fixpow2div2_S(a); } FDK_INLINE LONG fMultDiv2BitExact(LONG a, LONG b) { return fixmuldiv2BitExact_DD(a, b); } FDK_INLINE LONG fMultDiv2BitExact(SHORT a, LONG b) { return fixmuldiv2BitExact_SD(a, b); } FDK_INLINE LONG fMultDiv2BitExact(LONG a, SHORT b) { return fixmuldiv2BitExact_DS(a, b); } FDK_INLINE LONG fMultBitExact(LONG a, LONG b) { return fixmulBitExact_DD(a, b); } FDK_INLINE LONG fMultBitExact(SHORT a, LONG b) { return fixmulBitExact_SD(a, b); } FDK_INLINE LONG fMultBitExact(LONG a, SHORT b) { return fixmulBitExact_DS(a, b); } /* ******************************************************************************** */ #include "abs.h" FDK_INLINE FIXP_DBL fAbs(FIXP_DBL x) { return fixabs_D(x); } FDK_INLINE FIXP_SGL fAbs(FIXP_SGL x) { return fixabs_S(x); } #if !defined(__LP64__) FDK_INLINE INT fAbs(INT x) { return fixabs_I(x); } #endif /* ******************************************************************************** */ #include "clz.h" FDK_INLINE INT fNormz(INT64 x) { INT clz = fixnormz_D((INT)(x >> 32)); if (clz == 32) clz += fixnormz_D((INT)x); return clz; } FDK_INLINE INT fNormz(FIXP_DBL x) { return fixnormz_D(x); } FDK_INLINE INT fNormz(FIXP_SGL x) { return fixnormz_S(x); } FDK_INLINE INT fNorm(FIXP_DBL x) { return fixnorm_D(x); } FDK_INLINE INT fNorm(FIXP_SGL x) { return fixnorm_S(x); } /* ******************************************************************************** */ /* ******************************************************************************** */ /* ******************************************************************************** */ #include "clz.h" #define fixp_abs(x) fAbs(x) #define fixMin(a, b) fMin(a, b) #define fixMax(a, b) fMax(a, b) #define CntLeadingZeros(x) fixnormz_D(x) #define CountLeadingBits(x) fixnorm_D(x) #include "fixmadd.h" /* y = (x+0.5*a*b) */ FDK_INLINE FIXP_DBL fMultAddDiv2(FIXP_DBL x, FIXP_DBL a, FIXP_DBL b) { return fixmadddiv2_DD(x, a, b); } FDK_INLINE FIXP_DBL fMultAddDiv2(FIXP_DBL x, FIXP_SGL a, FIXP_DBL b) { return fixmadddiv2_SD(x, a, b); } FDK_INLINE FIXP_DBL fMultAddDiv2(FIXP_DBL x, FIXP_DBL a, FIXP_SGL b) { return fixmadddiv2_DS(x, a, b); } FDK_INLINE FIXP_DBL fMultAddDiv2(FIXP_DBL x, FIXP_SGL a, FIXP_SGL b) { return fixmadddiv2_SS(x, a, b); } FDK_INLINE FIXP_DBL fPow2AddDiv2(FIXP_DBL x, FIXP_DBL a) { return fixpadddiv2_D(x, a); } FDK_INLINE FIXP_DBL fPow2AddDiv2(FIXP_DBL x, FIXP_SGL a) { return fixpadddiv2_S(x, a); } /* y = 2*(x+0.5*a*b) = (2x+a*b) */ FDK_INLINE FIXP_DBL fMultAdd(FIXP_DBL x, FIXP_DBL a, FIXP_DBL b) { return fixmadd_DD(x, a, b); } inline FIXP_DBL fMultAdd(FIXP_DBL x, FIXP_SGL a, FIXP_DBL b) { return fixmadd_SD(x, a, b); } inline FIXP_DBL fMultAdd(FIXP_DBL x, FIXP_DBL a, FIXP_SGL b) { return fixmadd_DS(x, a, b); } inline FIXP_DBL fMultAdd(FIXP_DBL x, FIXP_SGL a, FIXP_SGL b) { return fixmadd_SS(x, a, b); } inline FIXP_DBL fPow2Add(FIXP_DBL x, FIXP_DBL a) { return fixpadd_D(x, a); } inline FIXP_DBL fPow2Add(FIXP_DBL x, FIXP_SGL a) { return fixpadd_S(x, a); } /* y = (x-0.5*a*b) */ inline FIXP_DBL fMultSubDiv2(FIXP_DBL x, FIXP_DBL a, FIXP_DBL b) { return fixmsubdiv2_DD(x, a, b); } inline FIXP_DBL fMultSubDiv2(FIXP_DBL x, FIXP_SGL a, FIXP_DBL b) { return fixmsubdiv2_SD(x, a, b); } inline FIXP_DBL fMultSubDiv2(FIXP_DBL x, FIXP_DBL a, FIXP_SGL b) { return fixmsubdiv2_DS(x, a, b); } inline FIXP_DBL fMultSubDiv2(FIXP_DBL x, FIXP_SGL a, FIXP_SGL b) { return fixmsubdiv2_SS(x, a, b); } /* y = 2*(x-0.5*a*b) = (2*x-a*b) */ FDK_INLINE FIXP_DBL fMultSub(FIXP_DBL x, FIXP_DBL a, FIXP_DBL b) { return fixmsub_DD(x, a, b); } inline FIXP_DBL fMultSub(FIXP_DBL x, FIXP_SGL a, FIXP_DBL b) { return fixmsub_SD(x, a, b); } inline FIXP_DBL fMultSub(FIXP_DBL x, FIXP_DBL a, FIXP_SGL b) { return fixmsub_DS(x, a, b); } inline FIXP_DBL fMultSub(FIXP_DBL x, FIXP_SGL a, FIXP_SGL b) { return fixmsub_SS(x, a, b); } FDK_INLINE FIXP_DBL fMultAddDiv2BitExact(FIXP_DBL x, FIXP_DBL a, FIXP_DBL b) { return fixmadddiv2BitExact_DD(x, a, b); } FDK_INLINE FIXP_DBL fMultAddDiv2BitExact(FIXP_DBL x, FIXP_SGL a, FIXP_DBL b) { return fixmadddiv2BitExact_SD(x, a, b); } FDK_INLINE FIXP_DBL fMultAddDiv2BitExact(FIXP_DBL x, FIXP_DBL a, FIXP_SGL b) { return fixmadddiv2BitExact_DS(x, a, b); } FDK_INLINE FIXP_DBL fMultSubDiv2BitExact(FIXP_DBL x, FIXP_DBL a, FIXP_DBL b) { return fixmsubdiv2BitExact_DD(x, a, b); } FDK_INLINE FIXP_DBL fMultSubDiv2BitExact(FIXP_DBL x, FIXP_SGL a, FIXP_DBL b) { return fixmsubdiv2BitExact_SD(x, a, b); } FDK_INLINE FIXP_DBL fMultSubDiv2BitExact(FIXP_DBL x, FIXP_DBL a, FIXP_SGL b) { return fixmsubdiv2BitExact_DS(x, a, b); } #include "fixminmax.h" FDK_INLINE FIXP_DBL fMin(FIXP_DBL a, FIXP_DBL b) { return fixmin_D(a, b); } FDK_INLINE FIXP_DBL fMax(FIXP_DBL a, FIXP_DBL b) { return fixmax_D(a, b); } FDK_INLINE FIXP_SGL fMin(FIXP_SGL a, FIXP_SGL b) { return fixmin_S(a, b); } FDK_INLINE FIXP_SGL fMax(FIXP_SGL a, FIXP_SGL b) { return fixmax_S(a, b); } #if !defined(__LP64__) FDK_INLINE INT fMax(INT a, INT b) { return fixmax_I(a, b); } FDK_INLINE INT fMin(INT a, INT b) { return fixmin_I(a, b); } #endif inline UINT fMax(UINT a, UINT b) { return fixmax_UI(a, b); } inline UINT fMin(UINT a, UINT b) { return fixmin_UI(a, b); } inline UCHAR fMax(UCHAR a, UCHAR b) { return (UCHAR)fixmax_UI((UINT)a, (UINT)b); } inline UCHAR fMin(UCHAR a, UCHAR b) { return (UCHAR)fixmin_UI((UINT)a, (UINT)b); } /* Complex data types */ typedef shouldBeUnion { /* vector representation for arithmetic */ struct { FIXP_SGL re; FIXP_SGL im; } v; /* word representation for memory move */ LONG w; } FIXP_SPK; typedef shouldBeUnion { /* vector representation for arithmetic */ struct { FIXP_DBL re; FIXP_DBL im; } v; /* word representation for memory move */ INT64 w; } FIXP_DPK; #include "fixmul.h" #include "fixmadd.h" #include "cplx_mul.h" #include "fixpoint_math.h" #endif fdk-aac-2.0.2/libFDK/include/cplx_mul.h000066400000000000000000000243501404225310700175150ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Description: fixed point intrinsics *******************************************************************************/ #if !defined(CPLX_MUL_H) #define CPLX_MUL_H #include "common_fix.h" #if defined(__arm__) || defined(_M_ARM) #include "arm/cplx_mul_arm.h" #elif defined(__GNUC__) && defined(__mips__) && __mips_isa_rev < 6 #include "mips/cplx_mul_mips.h" #endif /* #if defined all cores: bfin, arm, etc. */ /* ############################################################################# */ /* Fallback generic implementations */ #if !defined(FUNCTION_cplxMultDiv2_32x16X2) #define FUNCTION_cplxMultDiv2_32x16X2 inline void cplxMultDiv2(FIXP_DBL *c_Re, FIXP_DBL *c_Im, const FIXP_DBL a_Re, const FIXP_DBL a_Im, const FIXP_SGL b_Re, const FIXP_SGL b_Im) { *c_Re = fMultDiv2(a_Re, b_Re) - fMultDiv2(a_Im, b_Im); *c_Im = fMultDiv2(a_Re, b_Im) + fMultDiv2(a_Im, b_Re); } #endif #if !defined(FUNCTION_cplxMultDiv2_16x16X2) #define FUNCTION_cplxMultDiv2_16x16X2 inline void cplxMultDiv2(FIXP_DBL *c_Re, FIXP_DBL *c_Im, const FIXP_SGL a_Re, const FIXP_SGL a_Im, const FIXP_SGL b_Re, const FIXP_SGL b_Im) { *c_Re = fMultDiv2(a_Re, b_Re) - fMultDiv2(a_Im, b_Im); *c_Im = fMultDiv2(a_Re, b_Im) + fMultDiv2(a_Im, b_Re); } inline void cplxMultDiv2(FIXP_SGL *c_Re, FIXP_SGL *c_Im, const FIXP_SGL a_Re, const FIXP_SGL a_Im, const FIXP_SGL b_Re, const FIXP_SGL b_Im) { *c_Re = FX_DBL2FX_SGL(fMultDiv2(a_Re, b_Re) - fMultDiv2(a_Im, b_Im)); *c_Im = FX_DBL2FX_SGL(fMultDiv2(a_Re, b_Im) + fMultDiv2(a_Im, b_Re)); } #endif #if !defined(FUNCTION_cplxMultDiv2_32x16) #define FUNCTION_cplxMultDiv2_32x16 inline void cplxMultDiv2(FIXP_DBL *c_Re, FIXP_DBL *c_Im, const FIXP_DBL a_Re, const FIXP_DBL a_Im, const FIXP_SPK w) { cplxMultDiv2(c_Re, c_Im, a_Re, a_Im, w.v.re, w.v.im); } #endif #if !defined(FUNCTION_cplxMultDiv2_16x16) #define FUNCTION_cplxMultDiv2_16x16 inline void cplxMultDiv2(FIXP_DBL *c_Re, FIXP_DBL *c_Im, const FIXP_SGL a_Re, const FIXP_SGL a_Im, const FIXP_SPK w) { cplxMultDiv2(c_Re, c_Im, a_Re, a_Im, w.v.re, w.v.im); } inline void cplxMultDiv2(FIXP_SGL *c_Re, FIXP_SGL *c_Im, const FIXP_SGL a_Re, const FIXP_SGL a_Im, const FIXP_SPK w) { cplxMultDiv2(c_Re, c_Im, a_Re, a_Im, w.v.re, w.v.im); } #endif #if !defined(FUNCTION_cplxMultSubDiv2_32x16X2) #define FUNCTION_cplxMultSubDiv2_32x16X2 inline void cplxMultSubDiv2(FIXP_DBL *c_Re, FIXP_DBL *c_Im, const FIXP_DBL a_Re, const FIXP_DBL a_Im, const FIXP_SGL b_Re, const FIXP_SGL b_Im) { *c_Re -= fMultDiv2(a_Re, b_Re) - fMultDiv2(a_Im, b_Im); *c_Im -= fMultDiv2(a_Re, b_Im) + fMultDiv2(a_Im, b_Re); } #endif #if !defined(FUNCTION_cplxMultDiv2_32x32X2) #define FUNCTION_cplxMultDiv2_32x32X2 inline void cplxMultDiv2(FIXP_DBL *c_Re, FIXP_DBL *c_Im, const FIXP_DBL a_Re, const FIXP_DBL a_Im, const FIXP_DBL b_Re, const FIXP_DBL b_Im) { *c_Re = fMultDiv2(a_Re, b_Re) - fMultDiv2(a_Im, b_Im); *c_Im = fMultDiv2(a_Re, b_Im) + fMultDiv2(a_Im, b_Re); } #endif #if !defined(FUNCTION_cplxMultDiv2_32x32) #define FUNCTION_cplxMultDiv2_32x32 inline void cplxMultDiv2(FIXP_DBL *c_Re, FIXP_DBL *c_Im, const FIXP_DBL a_Re, const FIXP_DBL a_Im, const FIXP_DPK w) { cplxMultDiv2(c_Re, c_Im, a_Re, a_Im, w.v.re, w.v.im); } #endif #if !defined(FUNCTION_cplxMultSubDiv2_32x32X2) #define FUNCTION_cplxMultSubDiv2_32x32X2 inline void cplxMultSubDiv2(FIXP_DBL *c_Re, FIXP_DBL *c_Im, const FIXP_DBL a_Re, const FIXP_DBL a_Im, const FIXP_DBL b_Re, const FIXP_DBL b_Im) { *c_Re -= fMultDiv2(a_Re, b_Re) - fMultDiv2(a_Im, b_Im); *c_Im -= fMultDiv2(a_Re, b_Im) + fMultDiv2(a_Im, b_Re); } #endif /* ############################################################################# */ #if !defined(FUNCTION_cplxMult_32x16X2) #define FUNCTION_cplxMult_32x16X2 inline void cplxMult(FIXP_DBL *c_Re, FIXP_DBL *c_Im, const FIXP_DBL a_Re, const FIXP_DBL a_Im, const FIXP_SGL b_Re, const FIXP_SGL b_Im) { *c_Re = fMult(a_Re, b_Re) - fMult(a_Im, b_Im); *c_Im = fMult(a_Re, b_Im) + fMult(a_Im, b_Re); } inline void cplxMult(FIXP_SGL *c_Re, FIXP_SGL *c_Im, const FIXP_SGL a_Re, const FIXP_SGL a_Im, const FIXP_SGL b_Re, const FIXP_SGL b_Im) { *c_Re = FX_DBL2FX_SGL(fMult(a_Re, b_Re) - fMult(a_Im, b_Im)); *c_Im = FX_DBL2FX_SGL(fMult(a_Re, b_Im) + fMult(a_Im, b_Re)); } #endif #if !defined(FUNCTION_cplxMult_32x16) #define FUNCTION_cplxMult_32x16 inline void cplxMult(FIXP_DBL *c_Re, FIXP_DBL *c_Im, const FIXP_DBL a_Re, const FIXP_DBL a_Im, const FIXP_SPK w) { cplxMult(c_Re, c_Im, a_Re, a_Im, w.v.re, w.v.im); } #endif #if !defined(FUNCTION_cplxMult_32x32X2) #define FUNCTION_cplxMult_32x32X2 inline void cplxMult(FIXP_DBL *c_Re, FIXP_DBL *c_Im, const FIXP_DBL a_Re, const FIXP_DBL a_Im, const FIXP_DBL b_Re, const FIXP_DBL b_Im) { *c_Re = fMult(a_Re, b_Re) - fMult(a_Im, b_Im); *c_Im = fMult(a_Re, b_Im) + fMult(a_Im, b_Re); } #endif #if !defined(FUNCTION_cplxMult_32x32) #define FUNCTION_cplxMult_32x32 inline void cplxMult(FIXP_DBL *c_Re, FIXP_DBL *c_Im, const FIXP_DBL a_Re, const FIXP_DBL a_Im, const FIXP_DPK w) { cplxMult(c_Re, c_Im, a_Re, a_Im, w.v.re, w.v.im); } #endif /* ############################################################################# */ #endif /* CPLX_MUL_H */ fdk-aac-2.0.2/libFDK/include/dct.h000066400000000000000000000172671404225310700164550ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Description: Library functions to calculate standard DCTs. This will most likely be replaced by hand-optimized functions for the specific target processor. *******************************************************************************/ #ifndef DCT_H #define DCT_H #include "common_fix.h" void dct_getTables(const FIXP_WTP **ptwiddle, const FIXP_STP **sin_twiddle, int *sin_step, int length); /** * \brief Calculate DCT type II of given length. The DCT IV is * calculated by a complex FFT, with some pre and post twiddeling. * A factor of sqrt(2/(N-1)) is NOT applied. * \param pDat pointer to input/output data (in place processing). * \param size size of pDat. * \param pDat_e pointer to an integer containing the exponent of the data * referenced by pDat. The exponent is updated accordingly. */ void dct_II(FIXP_DBL *pDat, FIXP_DBL *tmp, int size, int *pDat_e); /** * \brief Calculate DCT type III of given length. The DCT IV is * calculated by a complex FFT, with some pre and post twiddeling. * Note that the factor 0.5 for the sum term x[0] is 1.0 instead of 0.5. * A factor of sqrt(2/N) is NOT applied. * \param pDat pointer to input/output data (in place processing). * \param size size of pDat. * \param pDat_e pointer to an integer containing the exponent of the data * referenced by pDat. The exponent is updated accordingly. */ void dct_III(FIXP_DBL *pDat, FIXP_DBL *tmp, int size, int *pDat_e); /** * \brief Calculate DST type III of given length. The DST III is * calculated by a DCT III of mirrored input and sign-flipping of odd * output coefficients. * Note that the factor 0.5 for the sum term x[N-1] is 1.0 instead of * 0.5. A factor of sqrt(2/N) is NOT applied. * \param pDat pointer to input/output data (in place processing). * \param size size of pDat. * \param pDat_e pointer to an integer containing the exponent of the data * referenced by pDat. The exponent is updated accordingly. */ void dst_III(FIXP_DBL *pDat, FIXP_DBL *tmp, int size, int *pDat_e); /** * \brief Calculate DCT type IV of given length. The DCT IV is * calculated by a complex FFT, with some pre and post twiddeling. * A factor of sqrt(2/N) is NOT applied. * \param pDat pointer to input/output data (in place processing). * \param size size of pDat. * \param pDat_e pointer to an integer containing the exponent of the data * referenced by pDat. The exponent is updated accordingly. */ void dct_IV(FIXP_DBL *pDat, int size, int *pDat_e); /** * \brief Calculate DST type IV of given length. The DST IV is * calculated by a complex FFT, with some pre and post twiddeling. * A factor of sqrt(2/N) is NOT applied. * \param pDat pointer to input/output data (in place processing). * \param size size of pDat. * \param pDat_e pointer to an integer containing the exponent of the data * referenced by pDat. The exponent is updated accordingly. */ void dst_IV(FIXP_DBL *pDat, int size, int *pDat_e); #endif fdk-aac-2.0.2/libFDK/include/fft.h000066400000000000000000000213261404225310700164510ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Josef Hoepfl, DSP Solutions Description: Fix point FFT *******************************************************************************/ #ifndef FFT_H #define FFT_H #include "common_fix.h" /** * \brief Perform an inplace complex valued FFT of length 2^n * * \param length Length of the FFT to be calculated. * \param pInput Input/Output data buffer. The input data must have at least 1 * bit scale headroom. The values are interleaved, real/imag pairs. * \param scalefactor Pointer to an INT, which contains the current scale of the * input data, which is updated according to the FFT scale. */ void fft(int length, FIXP_DBL *pInput, INT *scalefactor); /** * \brief Perform an inplace complex valued IFFT of length 2^n * * \param length Length of the FFT to be calculated. * \param pInput Input/Output data buffer. The input data must have at least 1 * bit scale headroom. The values are interleaved, real/imag pairs. * \param scalefactor Pointer to an INT, which contains the current scale of the * input data, which is updated according to the IFFT scale. */ void ifft(int length, FIXP_DBL *pInput, INT *scalefactor); /* * Frequently used and fixed short length FFTs. */ #ifndef FUNCTION_fft_4 /** * \brief Perform an inplace complex valued FFT of length 4 * * \param pInput Input/Output data buffer. The input data must have at least 1 * bit scale headroom. The values are interleaved, real/imag pairs. */ LNK_SECTION_CODE_L1 static FDK_FORCEINLINE void fft_4(FIXP_DBL *x) { FIXP_DBL a00, a10, a20, a30, tmp0, tmp1; a00 = (x[0] + x[4]) >> 1; /* Re A + Re B */ a10 = (x[2] + x[6]) >> 1; /* Re C + Re D */ a20 = (x[1] + x[5]) >> 1; /* Im A + Im B */ a30 = (x[3] + x[7]) >> 1; /* Im C + Im D */ x[0] = a00 + a10; /* Re A' = Re A + Re B + Re C + Re D */ x[1] = a20 + a30; /* Im A' = Im A + Im B + Im C + Im D */ tmp0 = a00 - x[4]; /* Re A - Re B */ tmp1 = a20 - x[5]; /* Im A - Im B */ x[4] = a00 - a10; /* Re C' = Re A + Re B - Re C - Re D */ x[5] = a20 - a30; /* Im C' = Im A + Im B - Im C - Im D */ a10 = a10 - x[6]; /* Re C - Re D */ a30 = a30 - x[7]; /* Im C - Im D */ x[2] = tmp0 + a30; /* Re B' = Re A - Re B + Im C - Im D */ x[6] = tmp0 - a30; /* Re D' = Re A - Re B - Im C + Im D */ x[3] = tmp1 - a10; /* Im B' = Im A - Im B - Re C + Re D */ x[7] = tmp1 + a10; /* Im D' = Im A - Im B + Re C - Re D */ } #endif /* FUNCTION_fft_4 */ #ifndef FUNCTION_fft_8 LNK_SECTION_CODE_L1 static FDK_FORCEINLINE void fft_8(FIXP_DBL *x) { FIXP_SPK w_PiFOURTH = {{FIXP_SGL(0x5A82), FIXP_SGL(0x5A82)}}; FIXP_DBL a00, a10, a20, a30; FIXP_DBL y[16]; a00 = (x[0] + x[8]) >> 1; a10 = x[4] + x[12]; a20 = (x[1] + x[9]) >> 1; a30 = x[5] + x[13]; y[0] = a00 + (a10 >> 1); y[4] = a00 - (a10 >> 1); y[1] = a20 + (a30 >> 1); y[5] = a20 - (a30 >> 1); a00 = a00 - x[8]; a10 = (a10 >> 1) - x[12]; a20 = a20 - x[9]; a30 = (a30 >> 1) - x[13]; y[2] = a00 + a30; y[6] = a00 - a30; y[3] = a20 - a10; y[7] = a20 + a10; a00 = (x[2] + x[10]) >> 1; a10 = x[6] + x[14]; a20 = (x[3] + x[11]) >> 1; a30 = x[7] + x[15]; y[8] = a00 + (a10 >> 1); y[12] = a00 - (a10 >> 1); y[9] = a20 + (a30 >> 1); y[13] = a20 - (a30 >> 1); a00 = a00 - x[10]; a10 = (a10 >> 1) - x[14]; a20 = a20 - x[11]; a30 = (a30 >> 1) - x[15]; y[10] = a00 + a30; y[14] = a00 - a30; y[11] = a20 - a10; y[15] = a20 + a10; FIXP_DBL vr, vi, ur, ui; ur = y[0] >> 1; ui = y[1] >> 1; vr = y[8]; vi = y[9]; x[0] = ur + (vr >> 1); x[1] = ui + (vi >> 1); x[8] = ur - (vr >> 1); x[9] = ui - (vi >> 1); ur = y[4] >> 1; ui = y[5] >> 1; vi = y[12]; vr = y[13]; x[4] = ur + (vr >> 1); x[5] = ui - (vi >> 1); x[12] = ur - (vr >> 1); x[13] = ui + (vi >> 1); ur = y[10]; ui = y[11]; cplxMultDiv2(&vi, &vr, ui, ur, w_PiFOURTH); ur = y[2]; ui = y[3]; x[2] = (ur >> 1) + vr; x[3] = (ui >> 1) + vi; x[10] = (ur >> 1) - vr; x[11] = (ui >> 1) - vi; ur = y[14]; ui = y[15]; cplxMultDiv2(&vr, &vi, ui, ur, w_PiFOURTH); ur = y[6]; ui = y[7]; x[6] = (ur >> 1) + vr; x[7] = (ui >> 1) - vi; x[14] = (ur >> 1) - vr; x[15] = (ui >> 1) + vi; } #endif /* FUNCTION_fft_8 */ #endif fdk-aac-2.0.2/libFDK/include/fft_rad2.h000066400000000000000000000124611404225310700173610ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): M. Lohwasser, M. Gayer Description: *******************************************************************************/ #ifndef FFT_RAD2_H #define FFT_RAD2_H #include "common_fix.h" /** * \brief Performe an inplace complex valued FFT of 2^n length * * \param x Input/Output data buffer. The input data must have at least 1 bit * scale headroom. The values are interleaved, real/imag pairs. * \param ldn log2 of FFT length * \param trigdata Pointer to a sinetable of a length of at least (2^ldn)/2 sine * values. * \param trigDataSize length of the sinetable "trigdata". */ void dit_fft(FIXP_DBL *x, const INT ldn, const FIXP_STP *trigdata, const INT trigDataSize); #endif /* FFT_RAD2_H */ fdk-aac-2.0.2/libFDK/include/fixmadd.h000066400000000000000000000261361404225310700173120ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): M. Lohwasser, M. Gayer Description: fixed point intrinsics *******************************************************************************/ #if !defined(FIXMADD_H) #define FIXMADD_H #include "FDK_archdef.h" #include "machine_type.h" #include "fixmul.h" #if defined(__arm__) #include "arm/fixmadd_arm.h" #endif /* all cores */ /************************************************************************* ************************************************************************* Software fallbacks for missing functions. ************************************************************************** **************************************************************************/ /* Divide by two versions. */ #if !defined(FUNCTION_fixmadddiv2_DD) inline FIXP_DBL fixmadddiv2_DD(FIXP_DBL x, const FIXP_DBL a, const FIXP_DBL b) { return (x + fMultDiv2(a, b)); } #endif #if !defined(FUNCTION_fixmadddiv2_SD) inline FIXP_DBL fixmadddiv2_SD(FIXP_DBL x, const FIXP_SGL a, const FIXP_DBL b) { #ifdef FUNCTION_fixmadddiv2_DS return fixmadddiv2_DS(x, b, a); #else return fixmadddiv2_DD(x, FX_SGL2FX_DBL(a), b); #endif } #endif #if !defined(FUNCTION_fixmadddiv2_DS) inline FIXP_DBL fixmadddiv2_DS(FIXP_DBL x, const FIXP_DBL a, const FIXP_SGL b) { #ifdef FUNCTION_fixmadddiv2_SD return fixmadddiv2_SD(x, b, a); #else return fixmadddiv2_DD(x, a, FX_SGL2FX_DBL(b)); #endif } #endif #if !defined(FUNCTION_fixmadddiv2_SS) inline FIXP_DBL fixmadddiv2_SS(FIXP_DBL x, const FIXP_SGL a, const FIXP_SGL b) { return x + fMultDiv2(a, b); } #endif #if !defined(FUNCTION_fixmsubdiv2_DD) inline FIXP_DBL fixmsubdiv2_DD(FIXP_DBL x, const FIXP_DBL a, const FIXP_DBL b) { return (x - fMultDiv2(a, b)); } #endif #if !defined(FUNCTION_fixmsubdiv2_SD) inline FIXP_DBL fixmsubdiv2_SD(FIXP_DBL x, const FIXP_SGL a, const FIXP_DBL b) { #ifdef FUNCTION_fixmsubdiv2_DS return fixmsubdiv2_DS(x, b, a); #else return fixmsubdiv2_DD(x, FX_SGL2FX_DBL(a), b); #endif } #endif #if !defined(FUNCTION_fixmsubdiv2_DS) inline FIXP_DBL fixmsubdiv2_DS(FIXP_DBL x, const FIXP_DBL a, const FIXP_SGL b) { #ifdef FUNCTION_fixmsubdiv2_SD return fixmsubdiv2_SD(x, b, a); #else return fixmsubdiv2_DD(x, a, FX_SGL2FX_DBL(b)); #endif } #endif #if !defined(FUNCTION_fixmsubdiv2_SS) inline FIXP_DBL fixmsubdiv2_SS(FIXP_DBL x, const FIXP_SGL a, const FIXP_SGL b) { return x - fMultDiv2(a, b); } #endif #if !defined(FUNCTION_fixmadddiv2BitExact_DD) #define FUNCTION_fixmadddiv2BitExact_DD inline FIXP_DBL fixmadddiv2BitExact_DD(FIXP_DBL x, const FIXP_DBL a, const FIXP_DBL b) { return x + fMultDiv2BitExact(a, b); } #endif #if !defined(FUNCTION_fixmadddiv2BitExact_SD) #define FUNCTION_fixmadddiv2BitExact_SD inline FIXP_DBL fixmadddiv2BitExact_SD(FIXP_DBL x, const FIXP_SGL a, const FIXP_DBL b) { #ifdef FUNCTION_fixmadddiv2BitExact_DS return fixmadddiv2BitExact_DS(x, b, a); #else return x + fMultDiv2BitExact(a, b); #endif } #endif #if !defined(FUNCTION_fixmadddiv2BitExact_DS) #define FUNCTION_fixmadddiv2BitExact_DS inline FIXP_DBL fixmadddiv2BitExact_DS(FIXP_DBL x, const FIXP_DBL a, const FIXP_SGL b) { #ifdef FUNCTION_fixmadddiv2BitExact_SD return fixmadddiv2BitExact_SD(x, b, a); #else return x + fMultDiv2BitExact(a, b); #endif } #endif #if !defined(FUNCTION_fixmsubdiv2BitExact_DD) #define FUNCTION_fixmsubdiv2BitExact_DD inline FIXP_DBL fixmsubdiv2BitExact_DD(FIXP_DBL x, const FIXP_DBL a, const FIXP_DBL b) { return x - fMultDiv2BitExact(a, b); } #endif #if !defined(FUNCTION_fixmsubdiv2BitExact_SD) #define FUNCTION_fixmsubdiv2BitExact_SD inline FIXP_DBL fixmsubdiv2BitExact_SD(FIXP_DBL x, const FIXP_SGL a, const FIXP_DBL b) { #ifdef FUNCTION_fixmsubdiv2BitExact_DS return fixmsubdiv2BitExact_DS(x, b, a); #else return x - fMultDiv2BitExact(a, b); #endif } #endif #if !defined(FUNCTION_fixmsubdiv2BitExact_DS) #define FUNCTION_fixmsubdiv2BitExact_DS inline FIXP_DBL fixmsubdiv2BitExact_DS(FIXP_DBL x, const FIXP_DBL a, const FIXP_SGL b) { #ifdef FUNCTION_fixmsubdiv2BitExact_SD return fixmsubdiv2BitExact_SD(x, b, a); #else return x - fMultDiv2BitExact(a, b); #endif } #endif /* Normal versions */ #if !defined(FUNCTION_fixmadd_DD) inline FIXP_DBL fixmadd_DD(FIXP_DBL x, const FIXP_DBL a, const FIXP_DBL b) { return fixmadddiv2_DD(x, a, b) << 1; } #endif #if !defined(FUNCTION_fixmadd_SD) inline FIXP_DBL fixmadd_SD(FIXP_DBL x, const FIXP_SGL a, const FIXP_DBL b) { #ifdef FUNCTION_fixmadd_DS return fixmadd_DS(x, b, a); #else return fixmadd_DD(x, FX_SGL2FX_DBL(a), b); #endif } #endif #if !defined(FUNCTION_fixmadd_DS) inline FIXP_DBL fixmadd_DS(FIXP_DBL x, const FIXP_DBL a, const FIXP_SGL b) { #ifdef FUNCTION_fixmadd_SD return fixmadd_SD(x, b, a); #else return fixmadd_DD(x, a, FX_SGL2FX_DBL(b)); #endif } #endif #if !defined(FUNCTION_fixmadd_SS) inline FIXP_DBL fixmadd_SS(FIXP_DBL x, const FIXP_SGL a, const FIXP_SGL b) { return (x + fMultDiv2(a, b)) << 1; } #endif #if !defined(FUNCTION_fixmsub_DD) inline FIXP_DBL fixmsub_DD(FIXP_DBL x, const FIXP_DBL a, const FIXP_DBL b) { return fixmsubdiv2_DD(x, a, b) << 1; } #endif #if !defined(FUNCTION_fixmsub_SD) inline FIXP_DBL fixmsub_SD(FIXP_DBL x, const FIXP_SGL a, const FIXP_DBL b) { #ifdef FUNCTION_fixmsub_DS return fixmsub_DS(x, b, a); #else return fixmsub_DD(x, FX_SGL2FX_DBL(a), b); #endif } #endif #if !defined(FUNCTION_fixmsub_DS) inline FIXP_DBL fixmsub_DS(FIXP_DBL x, const FIXP_DBL a, const FIXP_SGL b) { #ifdef FUNCTION_fixmsub_SD return fixmsub_SD(x, b, a); #else return fixmsub_DD(x, a, FX_SGL2FX_DBL(b)); #endif } #endif #if !defined(FUNCTION_fixmsub_SS) inline FIXP_DBL fixmsub_SS(FIXP_DBL x, const FIXP_SGL a, const FIXP_SGL b) { return (x - fMultDiv2(a, b)) << 1; } #endif #if !defined(FUNCTION_fixpow2adddiv2_D) #ifdef FUNCTION_fixmadddiv2_DD #define fixpadddiv2_D(x, a) fixmadddiv2_DD(x, a, a) #else inline INT fixpadddiv2_D(FIXP_DBL x, const FIXP_DBL a) { return (x + fPow2Div2(a)); } #endif #endif #if !defined(FUNCTION_fixpow2add_D) inline INT fixpadd_D(FIXP_DBL x, const FIXP_DBL a) { return (x + fPow2(a)); } #endif #if !defined(FUNCTION_fixpow2adddiv2_S) #ifdef FUNCTION_fixmadddiv2_SS #define fixpadddiv2_S(x, a) fixmadddiv2_SS(x, a, a) #else inline INT fixpadddiv2_S(FIXP_DBL x, const FIXP_SGL a) { return (x + fPow2Div2(a)); } #endif #endif #if !defined(FUNCTION_fixpow2add_S) inline INT fixpadd_S(FIXP_DBL x, const FIXP_SGL a) { return (x + fPow2(a)); } #endif #endif /* FIXMADD_H */ fdk-aac-2.0.2/libFDK/include/fixminmax.h000066400000000000000000000126001404225310700176650ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): M. Lohwasser, M. Gayer Description: min/max inline functions and defines *******************************************************************************/ #ifndef FIXMINMAX_H #define FIXMINMAX_H #include "FDK_archdef.h" #include "machine_type.h" /* Inline Function to determine the smaller/bigger value of two values with same * type. */ template inline T fixmin(T a, T b) { return (a < b ? a : b); } template inline T fixmax(T a, T b) { return (a > b ? a : b); } #define fixmax_D(a, b) fixmax(a, b) #define fixmin_D(a, b) fixmin(a, b) #define fixmax_S(a, b) fixmax(a, b) #define fixmin_S(a, b) fixmin(a, b) #define fixmax_I(a, b) fixmax(a, b) #define fixmin_I(a, b) fixmin(a, b) #define fixmax_UI(a, b) fixmax(a, b) #define fixmin_UI(a, b) fixmin(a, b) #endif fdk-aac-2.0.2/libFDK/include/fixmul.h000066400000000000000000000221261404225310700171750ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Stefan Gewinner Description: fixed point multiplication *******************************************************************************/ #if !defined(FIXMUL_H) #define FIXMUL_H #include "FDK_archdef.h" #include "machine_type.h" #if defined(__arm__) #include "arm/fixmul_arm.h" #elif defined(__mips__) #include "mips/fixmul_mips.h" #elif defined(__x86__) #include "x86/fixmul_x86.h" #elif defined(__powerpc__) #include "ppc/fixmul_ppc.h" #endif /* all cores */ /************************************************************************* ************************************************************************* Software fallbacks for missing functions ************************************************************************** **************************************************************************/ #if !defined(FUNCTION_fixmuldiv2_DD) #define FUNCTION_fixmuldiv2_DD inline LONG fixmuldiv2_DD(const LONG a, const LONG b) { return (LONG)((((INT64)a) * b) >> 32); } #endif #if !defined(FUNCTION_fixmuldiv2BitExact_DD) #define FUNCTION_fixmuldiv2BitExact_DD inline LONG fixmuldiv2BitExact_DD(const LONG a, const LONG b) { return (LONG)((((INT64)a) * b) >> 32); } #endif #if !defined(FUNCTION_fixmul_DD) #define FUNCTION_fixmul_DD inline LONG fixmul_DD(const LONG a, const LONG b) { return fixmuldiv2_DD(a, b) << 1; } #endif #if !defined(FUNCTION_fixmulBitExact_DD) #define FUNCTION_fixmulBitExact_DD inline LONG fixmulBitExact_DD(const LONG a, const LONG b) { return ((LONG)((((INT64)a) * b) >> 32)) << 1; } #endif #if !defined(FUNCTION_fixmuldiv2_SS) #define FUNCTION_fixmuldiv2_SS inline LONG fixmuldiv2_SS(const SHORT a, const SHORT b) { return ((LONG)a * b); } #endif #if !defined(FUNCTION_fixmul_SS) #define FUNCTION_fixmul_SS inline LONG fixmul_SS(const SHORT a, const SHORT b) { return (a * b) << 1; } #endif #if !defined(FUNCTION_fixmuldiv2_SD) #define FUNCTION_fixmuldiv2_SD inline LONG fixmuldiv2_SD(const SHORT a, const LONG b) #ifdef FUNCTION_fixmuldiv2_DS { return fixmuldiv2_DS(b, a); } #else { return fixmuldiv2_DD(FX_SGL2FX_DBL(a), b); } #endif #endif #if !defined(FUNCTION_fixmuldiv2_DS) #define FUNCTION_fixmuldiv2_DS inline LONG fixmuldiv2_DS(const LONG a, const SHORT b) #ifdef FUNCTION_fixmuldiv2_SD { return fixmuldiv2_SD(b, a); } #else { return fixmuldiv2_DD(a, FX_SGL2FX_DBL(b)); } #endif #endif #if !defined(FUNCTION_fixmuldiv2BitExact_SD) #define FUNCTION_fixmuldiv2BitExact_SD inline LONG fixmuldiv2BitExact_SD(const SHORT a, const LONG b) #ifdef FUNCTION_fixmuldiv2BitExact_DS { return fixmuldiv2BitExact_DS(b, a); } #else { return (LONG)((((INT64)a) * b) >> 16); } #endif #endif #if !defined(FUNCTION_fixmuldiv2BitExact_DS) #define FUNCTION_fixmuldiv2BitExact_DS inline LONG fixmuldiv2BitExact_DS(const LONG a, const SHORT b) #ifdef FUNCTION_fixmuldiv2BitExact_SD { return fixmuldiv2BitExact_SD(b, a); } #else { return (LONG)((((INT64)a) * b) >> 16); } #endif #endif #if !defined(FUNCTION_fixmul_SD) #define FUNCTION_fixmul_SD inline LONG fixmul_SD(const SHORT a, const LONG b) { #ifdef FUNCTION_fixmul_DS return fixmul_DS(b, a); #else return fixmuldiv2_SD(a, b) << 1; #endif } #endif #if !defined(FUNCTION_fixmul_DS) #define FUNCTION_fixmul_DS inline LONG fixmul_DS(const LONG a, const SHORT b) { #ifdef FUNCTION_fixmul_SD return fixmul_SD(b, a); #else return fixmuldiv2_DS(a, b) << 1; #endif } #endif #if !defined(FUNCTION_fixmulBitExact_SD) #define FUNCTION_fixmulBitExact_SD inline LONG fixmulBitExact_SD(const SHORT a, const LONG b) #ifdef FUNCTION_fixmulBitExact_DS { return fixmulBitExact_DS(b, a); } #else { return (LONG)(((((INT64)a) * b) >> 16) << 1); } #endif #endif #if !defined(FUNCTION_fixmulBitExact_DS) #define FUNCTION_fixmulBitExact_DS inline LONG fixmulBitExact_DS(const LONG a, const SHORT b) #ifdef FUNCTION_fixmulBitExact_SD { return fixmulBitExact_SD(b, a); } #else { return (LONG)(((((INT64)a) * b) >> 16) << 1); } #endif #endif #if !defined(FUNCTION_fixpow2div2_D) #define FUNCTION_fixpow2div2_D inline LONG fixpow2div2_D(const LONG a) { return fixmuldiv2_DD(a, a); } #endif #if !defined(FUNCTION_fixpow2_D) #define FUNCTION_fixpow2_D inline LONG fixpow2_D(const LONG a) { return fixpow2div2_D(a) << 1; } #endif #if !defined(FUNCTION_fixpow2div2_S) #define FUNCTION_fixpow2div2_S inline LONG fixpow2div2_S(const SHORT a) { return fixmuldiv2_SS(a, a); } #endif #if !defined(FUNCTION_fixpow2_S) #define FUNCTION_fixpow2_S inline LONG fixpow2_S(const SHORT a) { LONG result = fixpow2div2_S(a) << 1; return result ^ (result >> 31); } #endif #endif /* FIXMUL_H */ fdk-aac-2.0.2/libFDK/include/fixpoint_math.h000066400000000000000000000706721404225310700205530ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): M. Gayer Description: Fixed point specific mathematical functions *******************************************************************************/ #ifndef FIXPOINT_MATH_H #define FIXPOINT_MATH_H #include "common_fix.h" #include "scale.h" /* * Data definitions */ #define LD_DATA_SCALING (64.0f) #define LD_DATA_SHIFT 6 /* pow(2, LD_DATA_SHIFT) = LD_DATA_SCALING */ #define MAX_LD_PRECISION 10 #define LD_PRECISION 10 /* Taylor series coefficients for ln(1-x), centered at 0 (MacLaurin polynomial). */ #ifndef LDCOEFF_16BIT LNK_SECTION_CONSTDATA_L1 static const FIXP_DBL ldCoeff[MAX_LD_PRECISION] = { FL2FXCONST_DBL(-1.0), FL2FXCONST_DBL(-1.0 / 2.0), FL2FXCONST_DBL(-1.0 / 3.0), FL2FXCONST_DBL(-1.0 / 4.0), FL2FXCONST_DBL(-1.0 / 5.0), FL2FXCONST_DBL(-1.0 / 6.0), FL2FXCONST_DBL(-1.0 / 7.0), FL2FXCONST_DBL(-1.0 / 8.0), FL2FXCONST_DBL(-1.0 / 9.0), FL2FXCONST_DBL(-1.0 / 10.0)}; #else /* LDCOEFF_16BIT */ LNK_SECTION_CONSTDATA_L1 static const FIXP_SGL ldCoeff[MAX_LD_PRECISION] = { FL2FXCONST_SGL(-1.0), FL2FXCONST_SGL(-1.0 / 2.0), FL2FXCONST_SGL(-1.0 / 3.0), FL2FXCONST_SGL(-1.0 / 4.0), FL2FXCONST_SGL(-1.0 / 5.0), FL2FXCONST_SGL(-1.0 / 6.0), FL2FXCONST_SGL(-1.0 / 7.0), FL2FXCONST_SGL(-1.0 / 8.0), FL2FXCONST_SGL(-1.0 / 9.0), FL2FXCONST_SGL(-1.0 / 10.0)}; #endif /* LDCOEFF_16BIT */ /***************************************************************************** functionname: invSqrtNorm2 description: delivers 1/sqrt(op) normalized to .5...1 and the shift value of the OUTPUT *****************************************************************************/ #define SQRT_BITS 7 #define SQRT_VALUES (128 + 2) #define SQRT_BITS_MASK 0x7f #define SQRT_FRACT_BITS_MASK 0x007FFFFF extern const FIXP_DBL invSqrtTab[SQRT_VALUES]; /* * Hardware specific implementations */ #if defined(__x86__) #include "x86/fixpoint_math_x86.h" #endif /* target architecture selector */ /* * Fallback implementations */ #if !defined(FUNCTION_fIsLessThan) /** * \brief Compares two fixpoint values incl. scaling. * \param a_m mantissa of the first input value. * \param a_e exponent of the first input value. * \param b_m mantissa of the second input value. * \param b_e exponent of the second input value. * \return non-zero if (a_m*2^a_e) < (b_m*2^b_e), 0 otherwise */ FDK_INLINE INT fIsLessThan(FIXP_DBL a_m, INT a_e, FIXP_DBL b_m, INT b_e) { INT n; n = fixnorm_D(a_m); a_m <<= n; a_e -= n; n = fixnorm_D(b_m); b_m <<= n; b_e -= n; if (a_m == (FIXP_DBL)0) a_e = b_e; if (b_m == (FIXP_DBL)0) b_e = a_e; if (a_e > b_e) { return ((b_m >> fMin(a_e - b_e, DFRACT_BITS - 1)) > a_m); } else { return ((a_m >> fMin(b_e - a_e, DFRACT_BITS - 1)) < b_m); } } FDK_INLINE INT fIsLessThan(FIXP_SGL a_m, INT a_e, FIXP_SGL b_m, INT b_e) { INT n; n = fixnorm_S(a_m); a_m <<= n; a_e -= n; n = fixnorm_S(b_m); b_m <<= n; b_e -= n; if (a_m == (FIXP_SGL)0) a_e = b_e; if (b_m == (FIXP_SGL)0) b_e = a_e; if (a_e > b_e) { return ((b_m >> fMin(a_e - b_e, FRACT_BITS - 1)) > a_m); } else { return ((a_m >> fMin(b_e - a_e, FRACT_BITS - 1)) < b_m); } } #endif /** * \brief deprecated. Use fLog2() instead. */ #define CalcLdData(op) fLog2(op, 0) void LdDataVector(FIXP_DBL *srcVector, FIXP_DBL *destVector, INT number); extern const UINT exp2_tab_long[32]; extern const UINT exp2w_tab_long[32]; extern const UINT exp2x_tab_long[32]; LNK_SECTION_CODE_L1 FDK_INLINE FIXP_DBL CalcInvLdData(const FIXP_DBL x) { int set_zero = (x < FL2FXCONST_DBL(-31.0 / 64.0)) ? 0 : 1; int set_max = (x >= FL2FXCONST_DBL(31.0 / 64.0)) | (x == FL2FXCONST_DBL(0.0)); FIXP_SGL frac = (FIXP_SGL)((LONG)x & 0x3FF); UINT index3 = (UINT)(LONG)(x >> 10) & 0x1F; UINT index2 = (UINT)(LONG)(x >> 15) & 0x1F; UINT index1 = (UINT)(LONG)(x >> 20) & 0x1F; int exp = fMin(31, ((x > FL2FXCONST_DBL(0.0f)) ? (31 - (int)(x >> 25)) : (int)(-(x >> 25)))); UINT lookup1 = exp2_tab_long[index1] * set_zero; UINT lookup2 = exp2w_tab_long[index2]; UINT lookup3 = exp2x_tab_long[index3]; UINT lookup3f = lookup3 + (UINT)(LONG)fMultDiv2((FIXP_DBL)(0x0016302F), (FIXP_SGL)frac); UINT lookup12 = (UINT)(LONG)fMult((FIXP_DBL)lookup1, (FIXP_DBL)lookup2); UINT lookup = (UINT)(LONG)fMult((FIXP_DBL)lookup12, (FIXP_DBL)lookup3f); FIXP_DBL retVal = (lookup << 3) >> exp; if (set_max) { retVal = (FIXP_DBL)MAXVAL_DBL; } return retVal; } void InitLdInt(); FIXP_DBL CalcLdInt(INT i); extern const USHORT sqrt_tab[49]; inline FIXP_DBL sqrtFixp_lookup(FIXP_DBL x) { UINT y = (INT)x; UCHAR is_zero = (y == 0); INT zeros = fixnormz_D(y) & 0x1e; y <<= zeros; UINT idx = (y >> 26) - 16; USHORT frac = (y >> 10) & 0xffff; USHORT nfrac = 0xffff ^ frac; UINT t = (UINT)nfrac * sqrt_tab[idx] + (UINT)frac * sqrt_tab[idx + 1]; t = t >> (zeros >> 1); return (is_zero ? 0 : t); } inline FIXP_DBL sqrtFixp_lookup(FIXP_DBL x, INT *x_e) { UINT y = (INT)x; INT e; if (x == (FIXP_DBL)0) { return x; } /* Normalize */ e = fixnormz_D(y); y <<= e; e = *x_e - e + 2; /* Correct odd exponent. */ if (e & 1) { y >>= 1; e++; } /* Get square root */ UINT idx = (y >> 26) - 16; USHORT frac = (y >> 10) & 0xffff; USHORT nfrac = 0xffff ^ frac; UINT t = (UINT)nfrac * sqrt_tab[idx] + (UINT)frac * sqrt_tab[idx + 1]; /* Write back exponent */ *x_e = e >> 1; return (FIXP_DBL)(LONG)(t >> 1); } void InitInvSqrtTab(); #ifndef FUNCTION_invSqrtNorm2 /** * \brief calculate 1.0/sqrt(op) * \param op_m mantissa of input value. * \param result_e pointer to return the exponent of the result * \return mantissa of the result */ /***************************************************************************** delivers 1/sqrt(op) normalized to .5...1 and the shift value of the OUTPUT, i.e. the denormalized result is 1/sqrt(op) = invSqrtNorm(op) * 2^(shift) uses Newton-iteration for approximation Q(n+1) = Q(n) + Q(n) * (0.5 - 2 * V * Q(n)^2) with Q = 0.5* V ^-0.5; 0.5 <= V < 1.0 *****************************************************************************/ static FDK_FORCEINLINE FIXP_DBL invSqrtNorm2(FIXP_DBL op, INT *shift) { FIXP_DBL val = op; FIXP_DBL reg1, reg2; if (val == FL2FXCONST_DBL(0.0)) { *shift = 16; return ((LONG)MAXVAL_DBL); /* maximum positive value */ } #define INVSQRTNORM2_LINEAR_INTERPOLATE #define INVSQRTNORM2_LINEAR_INTERPOLATE_HQ /* normalize input, calculate shift value */ FDK_ASSERT(val > FL2FXCONST_DBL(0.0)); *shift = fNormz(val) - 1; /* CountLeadingBits() is not necessary here since test value is always > 0 */ val <<= *shift; /* normalized input V */ *shift += 2; /* bias for exponent */ #if defined(INVSQRTNORM2_LINEAR_INTERPOLATE) INT index = (INT)(val >> (DFRACT_BITS - 1 - (SQRT_BITS + 1))) & SQRT_BITS_MASK; FIXP_DBL Fract = (FIXP_DBL)(((INT)val & SQRT_FRACT_BITS_MASK) << (SQRT_BITS + 1)); FIXP_DBL diff = invSqrtTab[index + 1] - invSqrtTab[index]; reg1 = invSqrtTab[index] + (fMultDiv2(diff, Fract) << 1); #if defined(INVSQRTNORM2_LINEAR_INTERPOLATE_HQ) /* reg1 = t[i] + (t[i+1]-t[i])*fract ... already computed ... + (1-fract)fract*(t[i+2]-t[i+1])/2 */ if (Fract != (FIXP_DBL)0) { /* fract = fract * (1 - fract) */ Fract = fMultDiv2(Fract, (FIXP_DBL)((ULONG)0x80000000 - (ULONG)Fract)) << 1; diff = diff - (invSqrtTab[index + 2] - invSqrtTab[index + 1]); reg1 = fMultAddDiv2(reg1, Fract, diff); } #endif /* INVSQRTNORM2_LINEAR_INTERPOLATE_HQ */ #else #error \ "Either define INVSQRTNORM2_NEWTON_ITERATE or INVSQRTNORM2_LINEAR_INTERPOLATE" #endif /* calculate the output exponent = input exp/2 */ if (*shift & 0x00000001) { /* odd shift values ? */ /* Note: Do not use rounded value 0x5A82799A to avoid overflow with * shift-by-2 */ reg2 = (FIXP_DBL)0x5A827999; /* FL2FXCONST_DBL(0.707106781186547524400844362104849f);*/ /* 1/sqrt(2); */ reg1 = fMultDiv2(reg1, reg2) << 2; } *shift = *shift >> 1; return (reg1); } #endif /* FUNCTION_invSqrtNorm2 */ #ifndef FUNCTION_sqrtFixp static FDK_FORCEINLINE FIXP_DBL sqrtFixp(FIXP_DBL op) { INT tmp_exp = 0; FIXP_DBL tmp_inv = invSqrtNorm2(op, &tmp_exp); FDK_ASSERT(tmp_exp > 0); return ((FIXP_DBL)(fMultDiv2((op << (tmp_exp - 1)), tmp_inv) << 2)); } #endif /* FUNCTION_sqrtFixp */ #ifndef FUNCTION_invFixp /** * \brief calculate 1.0/op * \param op mantissa of the input value. * \return mantissa of the result with implicit exponent of 31 * \exceptions are provided for op=0,1 setting max. positive value */ static inline FIXP_DBL invFixp(FIXP_DBL op) { if ((op == (FIXP_DBL)0x00000000) || (op == (FIXP_DBL)0x00000001)) { return ((LONG)MAXVAL_DBL); } INT tmp_exp; FIXP_DBL tmp_inv = invSqrtNorm2(op, &tmp_exp); FDK_ASSERT((31 - (2 * tmp_exp + 1)) >= 0); int shift = 31 - (2 * tmp_exp + 1); tmp_inv = fPow2Div2(tmp_inv); if (shift) { tmp_inv = ((tmp_inv >> (shift - 1)) + (FIXP_DBL)1) >> 1; } return tmp_inv; } /** * \brief calculate 1.0/(op_m * 2^op_e) * \param op_m mantissa of the input value. * \param op_e pointer into were the exponent of the input value is stored, and * the result will be stored into. * \return mantissa of the result */ static inline FIXP_DBL invFixp(FIXP_DBL op_m, int *op_e) { if ((op_m == (FIXP_DBL)0x00000000) || (op_m == (FIXP_DBL)0x00000001)) { *op_e = 31 - *op_e; return ((LONG)MAXVAL_DBL); } INT tmp_exp; FIXP_DBL tmp_inv = invSqrtNorm2(op_m, &tmp_exp); *op_e = (tmp_exp << 1) - *op_e + 1; return fPow2Div2(tmp_inv); } #endif /* FUNCTION_invFixp */ #ifndef FUNCTION_schur_div /** * \brief Divide two FIXP_DBL values with given precision. * \param num dividend * \param denum divisor * \param count amount of significant bits of the result (starting to the MSB) * \return num/divisor */ FIXP_DBL schur_div(FIXP_DBL num, FIXP_DBL denum, INT count); #endif /* FUNCTION_schur_div */ FIXP_DBL mul_dbl_sgl_rnd(const FIXP_DBL op1, const FIXP_SGL op2); #ifndef FUNCTION_fMultNorm /** * \brief multiply two values with normalization, thus max precision. * Author: Robert Weidner * * \param f1 first factor * \param f2 second factor * \param result_e pointer to an INT where the exponent of the result is stored * into * \return mantissa of the product f1*f2 */ FIXP_DBL fMultNorm(FIXP_DBL f1, FIXP_DBL f2, INT *result_e); /** * \brief Multiply 2 values using maximum precision. The exponent of the result * is 0. * \param f1_m mantissa of factor 1 * \param f2_m mantissa of factor 2 * \return mantissa of the result with exponent equal to 0 */ inline FIXP_DBL fMultNorm(FIXP_DBL f1, FIXP_DBL f2) { FIXP_DBL m; INT e; m = fMultNorm(f1, f2, &e); m = scaleValueSaturate(m, e); return m; } /** * \brief Multiply 2 values with exponent and use given exponent for the * mantissa of the result. * \param f1_m mantissa of factor 1 * \param f1_e exponent of factor 1 * \param f2_m mantissa of factor 2 * \param f2_e exponent of factor 2 * \param result_e exponent for the returned mantissa of the result * \return mantissa of the result with exponent equal to result_e */ inline FIXP_DBL fMultNorm(FIXP_DBL f1_m, INT f1_e, FIXP_DBL f2_m, INT f2_e, INT result_e) { FIXP_DBL m; INT e; m = fMultNorm(f1_m, f2_m, &e); m = scaleValueSaturate(m, e + f1_e + f2_e - result_e); return m; } #endif /* FUNCTION_fMultNorm */ #ifndef FUNCTION_fMultI /** * \brief Multiplies a fractional value and a integer value and performs * rounding to nearest * \param a fractional value * \param b integer value * \return integer value */ inline INT fMultI(FIXP_DBL a, INT b) { FIXP_DBL m, mi; INT m_e; m = fMultNorm(a, (FIXP_DBL)b, &m_e); if (m_e < (INT)0) { if (m_e > (INT)-DFRACT_BITS) { m = m >> ((-m_e) - 1); mi = (m + (FIXP_DBL)1) >> 1; } else { mi = (FIXP_DBL)0; } } else { mi = scaleValueSaturate(m, m_e); } return ((INT)mi); } #endif /* FUNCTION_fMultI */ #ifndef FUNCTION_fMultIfloor /** * \brief Multiplies a fractional value and a integer value and performs floor * rounding * \param a fractional value * \param b integer value * \return integer value */ inline INT fMultIfloor(FIXP_DBL a, INT b) { FIXP_DBL m, mi; INT m_e; m = fMultNorm(a, (FIXP_DBL)b, &m_e); if (m_e < (INT)0) { if (m_e > (INT)-DFRACT_BITS) { mi = m >> (-m_e); } else { mi = (FIXP_DBL)0; if (m < (FIXP_DBL)0) { mi = (FIXP_DBL)-1; } } } else { mi = scaleValueSaturate(m, m_e); } return ((INT)mi); } #endif /* FUNCTION_fMultIfloor */ #ifndef FUNCTION_fMultIceil /** * \brief Multiplies a fractional value and a integer value and performs ceil * rounding * \param a fractional value * \param b integer value * \return integer value */ inline INT fMultIceil(FIXP_DBL a, INT b) { FIXP_DBL m, mi; INT m_e; m = fMultNorm(a, (FIXP_DBL)b, &m_e); if (m_e < (INT)0) { if (m_e > (INT) - (DFRACT_BITS - 1)) { mi = (m >> (-m_e)); if ((LONG)m & ((1 << (-m_e)) - 1)) { mi = mi + (FIXP_DBL)1; } } else { if (m > (FIXP_DBL)0) { mi = (FIXP_DBL)1; } else { if ((m_e == -(DFRACT_BITS - 1)) && (m == (FIXP_DBL)MINVAL_DBL)) { mi = (FIXP_DBL)-1; } else { mi = (FIXP_DBL)0; } } } } else { mi = scaleValueSaturate(m, m_e); } return ((INT)mi); } #endif /* FUNCTION_fMultIceil */ #ifndef FUNCTION_fDivNorm /** * \brief Divide 2 FIXP_DBL values with normalization of input values. * \param num numerator * \param denum denominator * \param result_e pointer to an INT where the exponent of the result is stored * into * \return num/denum with exponent = *result_e */ FIXP_DBL fDivNorm(FIXP_DBL num, FIXP_DBL denom, INT *result_e); /** * \brief Divide 2 positive FIXP_DBL values with normalization of input values. * \param num numerator * \param denum denominator * \return num/denum with exponent = 0 */ FIXP_DBL fDivNorm(FIXP_DBL num, FIXP_DBL denom); /** * \brief Divide 2 signed FIXP_DBL values with normalization of input values. * \param num numerator * \param denum denominator * \param result_e pointer to an INT where the exponent of the result is stored * into * \return num/denum with exponent = *result_e */ FIXP_DBL fDivNormSigned(FIXP_DBL L_num, FIXP_DBL L_denum, INT *result_e); /** * \brief Divide 2 signed FIXP_DBL values with normalization of input values. * \param num numerator * \param denum denominator * \return num/denum with exponent = 0 */ FIXP_DBL fDivNormSigned(FIXP_DBL num, FIXP_DBL denom); #endif /* FUNCTION_fDivNorm */ /** * \brief Adjust mantissa to exponent -1 * \param a_m mantissa of value to be adjusted * \param pA_e pointer to the exponen of a_m * \return adjusted mantissa */ inline FIXP_DBL fAdjust(FIXP_DBL a_m, INT *pA_e) { INT shift; shift = fNorm(a_m) - 1; *pA_e -= shift; return scaleValue(a_m, shift); } #ifndef FUNCTION_fAddNorm /** * \brief Add two values with normalization * \param a_m mantissa of first summand * \param a_e exponent of first summand * \param a_m mantissa of second summand * \param a_e exponent of second summand * \param pResult_e pointer to where the exponent of the result will be stored * to. * \return mantissa of result */ inline FIXP_DBL fAddNorm(FIXP_DBL a_m, INT a_e, FIXP_DBL b_m, INT b_e, INT *pResult_e) { INT result_e; FIXP_DBL result_m; /* If one of the summands is zero, return the other. This is necessary for the summation of a very small number to zero */ if (a_m == (FIXP_DBL)0) { *pResult_e = b_e; return b_m; } if (b_m == (FIXP_DBL)0) { *pResult_e = a_e; return a_m; } a_m = fAdjust(a_m, &a_e); b_m = fAdjust(b_m, &b_e); if (a_e > b_e) { result_m = a_m + (b_m >> fMin(a_e - b_e, DFRACT_BITS - 1)); result_e = a_e; } else { result_m = (a_m >> fMin(b_e - a_e, DFRACT_BITS - 1)) + b_m; result_e = b_e; } *pResult_e = result_e; return result_m; } inline FIXP_DBL fAddNorm(FIXP_DBL a_m, INT a_e, FIXP_DBL b_m, INT b_e, INT result_e) { FIXP_DBL result_m; a_m = scaleValue(a_m, a_e - result_e); b_m = scaleValue(b_m, b_e - result_e); result_m = a_m + b_m; return result_m; } #endif /* FUNCTION_fAddNorm */ /** * \brief Divide 2 FIXP_DBL values with normalization of input values. * \param num numerator * \param denum denomintator * \return num/denum with exponent = 0 */ FIXP_DBL fDivNormHighPrec(FIXP_DBL L_num, FIXP_DBL L_denum, INT *result_e); #ifndef FUNCTION_fPow /** * \brief return 2 ^ (exp_m * 2^exp_e) * \param exp_m mantissa of the exponent to 2.0f * \param exp_e exponent of the exponent to 2.0f * \param result_e pointer to a INT where the exponent of the result will be * stored into * \return mantissa of the result */ FIXP_DBL f2Pow(const FIXP_DBL exp_m, const INT exp_e, INT *result_e); /** * \brief return 2 ^ (exp_m * 2^exp_e). This version returns only the mantissa * with implicit exponent of zero. * \param exp_m mantissa of the exponent to 2.0f * \param exp_e exponent of the exponent to 2.0f * \return mantissa of the result */ FIXP_DBL f2Pow(const FIXP_DBL exp_m, const INT exp_e); /** * \brief return x ^ (exp_m * 2^exp_e), where log2(x) = baseLd_m * 2^(baseLd_e). * This saves the need to compute log2() of constant values (when x is a * constant). * \param baseLd_m mantissa of log2() of x. * \param baseLd_e exponent of log2() of x. * \param exp_m mantissa of the exponent to 2.0f * \param exp_e exponent of the exponent to 2.0f * \param result_e pointer to a INT where the exponent of the result will be * stored into * \return mantissa of the result */ FIXP_DBL fLdPow(FIXP_DBL baseLd_m, INT baseLd_e, FIXP_DBL exp_m, INT exp_e, INT *result_e); /** * \brief return x ^ (exp_m * 2^exp_e), where log2(x) = baseLd_m * 2^(baseLd_e). * This saves the need to compute log2() of constant values (when x is a * constant). This version does not return an exponent, which is * implicitly 0. * \param baseLd_m mantissa of log2() of x. * \param baseLd_e exponent of log2() of x. * \param exp_m mantissa of the exponent to 2.0f * \param exp_e exponent of the exponent to 2.0f * \return mantissa of the result */ FIXP_DBL fLdPow(FIXP_DBL baseLd_m, INT baseLd_e, FIXP_DBL exp_m, INT exp_e); /** * \brief return (base_m * 2^base_e) ^ (exp * 2^exp_e). Use fLdPow() instead * whenever possible. * \param base_m mantissa of the base. * \param base_e exponent of the base. * \param exp_m mantissa of power to be calculated of the base. * \param exp_e exponent of power to be calculated of the base. * \param result_e pointer to a INT where the exponent of the result will be * stored into. * \return mantissa of the result. */ FIXP_DBL fPow(FIXP_DBL base_m, INT base_e, FIXP_DBL exp_m, INT exp_e, INT *result_e); /** * \brief return (base_m * 2^base_e) ^ N * \param base_m mantissa of the base. Must not be negative. * \param base_e exponent of the base * \param N power to be calculated of the base * \param result_e pointer to a INT where the exponent of the result will be * stored into * \return mantissa of the result */ FIXP_DBL fPowInt(FIXP_DBL base_m, INT base_e, INT N, INT *result_e); #endif /* #ifndef FUNCTION_fPow */ #ifndef FUNCTION_fLog2 /** * \brief Calculate log(argument)/log(2) (logarithm with base 2). deprecated. * Use fLog2() instead. * \param arg mantissa of the argument * \param arg_e exponent of the argument * \param result_e pointer to an INT to store the exponent of the result * \return the mantissa of the result. * \param */ FIXP_DBL CalcLog2(FIXP_DBL arg, INT arg_e, INT *result_e); /** * \brief calculate logarithm of base 2 of x_m * 2^(x_e) * \param x_m mantissa of the input value. * \param x_e exponent of the input value. * \param pointer to an INT where the exponent of the result is returned into. * \return mantissa of the result. */ FDK_INLINE FIXP_DBL fLog2(FIXP_DBL x_m, INT x_e, INT *result_e) { FIXP_DBL result_m; /* Short cut for zero and negative numbers. */ if (x_m <= FL2FXCONST_DBL(0.0f)) { *result_e = DFRACT_BITS - 1; return FL2FXCONST_DBL(-1.0f); } /* Calculate log2() */ { FIXP_DBL x2_m; /* Move input value x_m * 2^x_e toward 1.0, where the taylor approximation of the function log(1-x) centered at 0 is most accurate. */ { INT b_norm; b_norm = fNormz(x_m) - 1; x2_m = x_m << b_norm; x_e = x_e - b_norm; } /* map x from log(x) domain to log(1-x) domain. */ x2_m = -(x2_m + FL2FXCONST_DBL(-1.0)); /* Taylor polynomial approximation of ln(1-x) */ { FIXP_DBL px2_m; result_m = FL2FXCONST_DBL(0.0); px2_m = x2_m; for (int i = 0; i < LD_PRECISION; i++) { result_m = fMultAddDiv2(result_m, ldCoeff[i], px2_m); px2_m = fMult(px2_m, x2_m); } } /* Multiply result with 1/ln(2) = 1.0 + 0.442695040888 (get log2(x) from * ln(x) result). */ result_m = fMultAddDiv2(result_m, result_m, FL2FXCONST_DBL(2.0 * 0.4426950408889634073599246810019)); /* Add exponent part. log2(x_m * 2^x_e) = log2(x_m) + x_e */ if (x_e != 0) { int enorm; enorm = DFRACT_BITS - fNorm((FIXP_DBL)x_e); /* The -1 in the right shift of result_m compensates the fMultDiv2() above * in the taylor polynomial evaluation loop.*/ result_m = (result_m >> (enorm - 1)) + ((FIXP_DBL)x_e << (DFRACT_BITS - 1 - enorm)); *result_e = enorm; } else { /* 1 compensates the fMultDiv2() above in the taylor polynomial evaluation * loop.*/ *result_e = 1; } } return result_m; } /** * \brief calculate logarithm of base 2 of x_m * 2^(x_e) * \param x_m mantissa of the input value. * \param x_e exponent of the input value. * \return mantissa of the result with implicit exponent of LD_DATA_SHIFT. */ FDK_INLINE FIXP_DBL fLog2(FIXP_DBL x_m, INT x_e) { if (x_m <= FL2FXCONST_DBL(0.0f)) { x_m = FL2FXCONST_DBL(-1.0f); } else { INT result_e; x_m = fLog2(x_m, x_e, &result_e); x_m = scaleValue(x_m, result_e - LD_DATA_SHIFT); } return x_m; } #endif /* FUNCTION_fLog2 */ #ifndef FUNCTION_fAddSaturate /** * \brief Add with saturation of the result. * \param a first summand * \param b second summand * \return saturated sum of a and b. */ inline FIXP_SGL fAddSaturate(const FIXP_SGL a, const FIXP_SGL b) { LONG sum; sum = (LONG)(SHORT)a + (LONG)(SHORT)b; sum = fMax(fMin((INT)sum, (INT)MAXVAL_SGL), (INT)MINVAL_SGL); return (FIXP_SGL)(SHORT)sum; } /** * \brief Add with saturation of the result. * \param a first summand * \param b second summand * \return saturated sum of a and b. */ inline FIXP_DBL fAddSaturate(const FIXP_DBL a, const FIXP_DBL b) { LONG sum; sum = (LONG)(a >> 1) + (LONG)(b >> 1); sum = fMax(fMin((INT)sum, (INT)(MAXVAL_DBL >> 1)), (INT)(MINVAL_DBL >> 1)); return (FIXP_DBL)(LONG)(sum << 1); } #endif /* FUNCTION_fAddSaturate */ INT fixp_floorToInt(FIXP_DBL f_inp, INT sf); FIXP_DBL fixp_floor(FIXP_DBL f_inp, INT sf); INT fixp_ceilToInt(FIXP_DBL f_inp, INT sf); FIXP_DBL fixp_ceil(FIXP_DBL f_inp, INT sf); INT fixp_truncateToInt(FIXP_DBL f_inp, INT sf); FIXP_DBL fixp_truncate(FIXP_DBL f_inp, INT sf); INT fixp_roundToInt(FIXP_DBL f_inp, INT sf); FIXP_DBL fixp_round(FIXP_DBL f_inp, INT sf); /***************************************************************************** array for 1/n, n=1..80 ****************************************************************************/ extern const FIXP_DBL invCount[80]; LNK_SECTION_INITCODE inline void InitInvInt(void) {} /** * \brief Calculate the value of 1/i where i is a integer value. It supports * input values from 1 upto (80-1). * \param intValue Integer input value. * \param FIXP_DBL representation of 1/intValue */ inline FIXP_DBL GetInvInt(int intValue) { return invCount[fMin(fMax(intValue, 0), 80 - 1)]; } #endif /* FIXPOINT_MATH_H */ fdk-aac-2.0.2/libFDK/include/huff_nodes.h000066400000000000000000000163771404225310700200240ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Omer Osman Description: MPEG-D SAC/USAC/SAOC Huffman Part0 Tables *******************************************************************************/ #ifndef HUFF_NODES_H #define HUFF_NODES_H #include "genericStds.h" typedef struct { SHORT nodeTab[39][2]; } HUFF_RES_NODES; /* 1D Nodes */ typedef struct { SHORT nodeTab[30][2]; } HUFF_CLD_NOD_1D; typedef struct { SHORT nodeTab[7][2]; } HUFF_ICC_NOD_1D; typedef struct { SHORT nodeTab[50][2]; } HUFF_CPC_NOD_1D; typedef struct { SHORT nodeTab[15][2]; } HUFF_OLD_NOD_1D; typedef struct { SHORT nodeTab[63][2]; } HUFF_NRG_NOD_1D; /* 2D Nodes */ typedef struct { SHORT lav3[15][2]; SHORT lav5[35][2]; SHORT lav7[63][2]; SHORT lav9[99][2]; } HUFF_CLD_NOD_2D; typedef struct { SHORT lav1[3][2]; SHORT lav3[15][2]; SHORT lav5[35][2]; SHORT lav7[63][2]; } HUFF_ICC_NOD_2D; typedef struct { SHORT lav3[15][2]; SHORT lav6[48][2]; SHORT lav9[99][2]; SHORT lav12[168][2]; } HUFF_OLD_NOD_2D; typedef struct { SHORT lav3[15][2]; SHORT lav5[35][2]; SHORT lav7[63][2]; SHORT lav9[99][2]; } HUFF_NRG_NOD_2D_df; typedef struct { SHORT lav3[15][2]; SHORT lav6[48][2]; SHORT lav9[99][2]; SHORT lav12[168][2]; } HUFF_NRG_NOD_2D_dt; typedef struct { HUFF_NRG_NOD_2D_df df[2]; HUFF_NRG_NOD_2D_dt dt[2]; HUFF_NRG_NOD_2D_df dp[2]; } HUFF_NRG_NOD_2D; /* Complete bs Parameter Nodes */ typedef struct { const HUFF_CLD_NOD_1D *h1D[3]; const HUFF_CLD_NOD_2D *h2D[3][2]; } HUFF_CLD_NODES; typedef struct { const HUFF_ICC_NOD_1D *h1D[3]; const HUFF_ICC_NOD_2D *h2D[3][2]; } HUFF_ICC_NODES; typedef struct { const HUFF_OLD_NOD_1D *h1D[3]; const HUFF_OLD_NOD_2D *h2D[3][2]; } HUFF_OLD_NODES; typedef struct { const HUFF_NRG_NOD_1D *h1D[3]; const HUFF_NRG_NOD_2D *h2D; } HUFF_NRG_NODES; /* parameter instance */ typedef struct { SHORT cld[30][2]; SHORT icc[7][2]; SHORT ipd[7][2]; SHORT old[15][2]; SHORT nrg[63][2]; } HUFF_PT0_NODES; typedef struct { SHORT nodeTab[3][2]; } HUFF_LAV_NODES; /* USAC specific */ typedef struct { SHORT nodeTab[7][2]; } HUFF_IPD_NOD_1D; typedef struct { SHORT lav1[3][2]; SHORT lav3[15][2]; SHORT lav5[35][2]; SHORT lav7[63][2]; } HUFF_IPD_NOD_2D; typedef struct { HUFF_IPD_NOD_1D h1D[3]; HUFF_IPD_NOD_2D h2D[3][2]; } HUFF_IPD_NODES; /* non-lossy coding decoder */ extern const HUFF_PT0_NODES FDK_huffPart0Nodes; extern const HUFF_LAV_NODES FDK_huffLavIdxNodes; extern const HUFF_ICC_NODES FDK_huffICCNodes; extern const HUFF_CLD_NODES FDK_huffCLDNodes; extern const HUFF_RES_NODES FDK_huffReshapeNodes; extern const HUFF_OLD_NODES huffOLDNodes; extern const HUFF_IPD_NODES FDK_huffIPDNodes; #endif /* HUFF_NODES_H */ fdk-aac-2.0.2/libFDK/include/mdct.h000066400000000000000000000242531404225310700166230ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Manuel Jander, Josef Hoepfl, Youliy Ninov, Daniel Hagel Description: MDCT/MDST routines *******************************************************************************/ #ifndef MDCT_H #define MDCT_H #include "common_fix.h" #define MDCT_OUT_HEADROOM 2 /* Output additional headroom */ #define PCM_OUT_BITS DFRACT_BITS #define PCM_OUT_HEADROOM 8 /* Must have the same values as DMXH_HEADROOM */ #define MDCT_OUTPUT_SCALE (-MDCT_OUT_HEADROOM + (DFRACT_BITS - PCM_OUT_BITS)) /* Refer to "Output word length" in ISO/IEC 14496-3:2008(E) 23.2.3.6 */ #define MDCT_OUTPUT_GAIN 16 #define IMDCT_SCALE(x, s) \ SATURATE_RIGHT_SHIFT((x), ((s) + MDCT_OUTPUT_SCALE), PCM_OUT_BITS) #define IMDCT_SCALE_DBL(x) (FIXP_DBL)(x) #define IMDCT_SCALE_DBL_LSH1(x) SATURATE_LEFT_SHIFT_ALT((x), 1, DFRACT_BITS) #define MLT_FLAG_CURR_ALIAS_SYMMETRY 1 typedef enum { BLOCK_LONG = 0, /* normal long block */ BLOCK_START, /* long start block */ BLOCK_SHORT, /* 8 short blocks sequence */ BLOCK_STOP /* long stop block*/ } BLOCK_TYPE; typedef enum { SHAPE_SINE = 0, SHAPE_KBD, SHAPE_LOL } WINDOW_SHAPE; /** * \brief MDCT persistent data */ typedef struct { union { FIXP_DBL *freq; FIXP_DBL *time; } overlap; /**< Pointer to overlap memory */ const FIXP_WTP *prev_wrs; /**< pointer to previous right window slope */ int prev_tl; /**< previous transform length */ int prev_nr; /**< previous right window offset */ int prev_fr; /**< previous right window slope length */ int ov_offset; /**< overlap time data fill level */ int ov_size; /**< Overlap buffer size in words */ int prevAliasSymmetry; int prevPrevAliasSymmetry; FIXP_DBL *pFacZir; FIXP_DBL *pAsymOvlp; /**< pointer to asymmetric overlap (used for stereo LPD transition) */ } mdct_t; typedef mdct_t *H_MDCT; /** * \brief Initialize as valid MDCT handle * * \param hMdct handle of an allocated MDCT handle. * \param overlap pointer to FIXP_DBL overlap buffer. * \param overlapBufferSize size in FIXP_DBLs of the given overlap buffer. */ void mdct_init(H_MDCT hMdct, FIXP_DBL *overlap, INT overlapBufferSize); /** * \brief perform MDCT transform (time domain to frequency domain) with given * parameters. * * \param hMdct handle of an allocated MDCT handle. * \param pTimeData pointer to input time domain signal * \param noInSamples number of input samples * \param mdctData pointer to where the resulting MDCT spectrum will be stored * into. * \param nSpec number of spectra * \param pMdctData_e pointer to the input data exponent. Updated accordingly on * return for output data. * \return number of input samples processed. */ INT mdct_block(H_MDCT hMdct, const INT_PCM *pTimeData, const INT noInSamples, FIXP_DBL *RESTRICT mdctData, const INT nSpec, const INT tl, const FIXP_WTP *pRightWindowPart, const INT fr, SHORT *pMdctData_e); /** * \brief add/multiply 2/N transform gain and MPEG4 part 3 defined output gain * (see definition of MDCT_OUTPUT_GAIN) to given mantissa factor and exponent. * \param pGain pointer to the mantissa of a gain factor to be applied to IMDCT * data. * \param pExponent pointer to the exponent of a gain factor to be applied to * IMDCT data. * \param tl length of the IMDCT where the gain *pGain * (2 ^ *pExponent) will * be applied to. */ void imdct_gain(FIXP_DBL *pGain, int *pExponent, int tl); /** * \brief drain buffered output samples into given buffer. Changes the MDCT * state. */ INT imdct_drain(H_MDCT hMdct, FIXP_DBL *pTimeData, INT nrSamplesRoom); /** * \brief Copy overlap time domain data to given buffer. Does not change the * MDCT state. * \return number of actually copied samples (ov + nr). */ INT imdct_copy_ov_and_nr(H_MDCT hMdct, FIXP_DBL *pTimeData, INT nrSamples); /** * \brief Adapt MDCT parameters for non-matching window slopes. * \param hMdct handle of an allocated MDCT handle. * \param pfl pointer to left overlap window side length. * \param pnl pointer to length of the left n part of the window. * \param tl transform length. * \param wls pointer to the left side overlap window coefficients. * \param noOutSamples desired number of output samples. */ void imdct_adapt_parameters(H_MDCT hMdct, int *pfl, int *pnl, int tl, const FIXP_WTP *wls, int noOutSamples); /** * \brief perform several inverse MLT transforms (frequency domain to time * domain) with given parameters. * * \param hMdct handle of an allocated MDCT handle. * \param output pointer to where the output time domain signal will be stored * into. * \param spectrum pointer to the input MDCT spectra. * \param scalefactors exponents of the input spectrum. * \param nSpec number of MDCT spectrums. * \param noOutSamples desired number of output samples. * \param tl transform length. * \param wls pointer to the left side overlap window coefficients. * \param fl left overlap window side length. * \param wrs pointer to the right side overlap window coefficients of all * individual IMDCTs. * \param fr right overlap window side length of all individual IMDCTs. * \param gain factor to apply to output samples (if != 0). * \param flags flags controlling the type of transform * \return number of output samples returned. */ INT imlt_block(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *spectrum, const SHORT scalefactor[], const INT nSpec, const INT noOutSamples, const INT tl, const FIXP_WTP *wls, INT fl, const FIXP_WTP *wrs, const INT fr, FIXP_DBL gain, int flags); #endif /* MDCT_H */ fdk-aac-2.0.2/libFDK/include/mips/000077500000000000000000000000001404225310700164655ustar00rootroot00000000000000fdk-aac-2.0.2/libFDK/include/mips/abs_mips.h000066400000000000000000000124621404225310700204400ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Description: fixed point intrinsics *******************************************************************************/ #if !defined(ABS_MIPS_H) #define ABS_MIPS_H #if defined(__mips__) #if defined(__GNUC__) && defined(__mips__) #if defined(__mips_dsp) && !defined(__mips16) #define FUNCTION_fixabs_D #define FUNCTION_fixabs_I #define FUNCTION_fixabs_S inline FIXP_DBL fixabs_D(FIXP_DBL x) { return __builtin_mips_absq_s_w(x); } inline FIXP_SGL fixabs_S(FIXP_SGL x) { return ((x) > (FIXP_SGL)(0)) ? (x) : -(x); } inline INT fixabs_I(INT x) { return __builtin_mips_absq_s_w(x); } #endif /* __mips_dsp */ #endif /* defined(__GNUC__) && defined(__mips__) */ #endif /*__mips__ */ #endif /* !defined(ABS_MIPS_H) */ fdk-aac-2.0.2/libFDK/include/mips/clz_mips.h000066400000000000000000000124721404225310700204640ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Description: fixed point intrinsics *******************************************************************************/ #if !defined(CLZ_MIPS_H) #define CLZ_MIPS_H #if defined(__mips__) #if defined(__mips__) && (__GNUC__ == 2) && (mips >= 32) #define FUNCTION_fixnormz_D inline INT fixnormz_D(LONG value) { INT result; __asm__("clz %0,%1" : "=d"(result) : "d"(value)); return result; } #elif defined(__mips__) && (__GNUC__ == 3) && (__mips >= 32) #define FUNCTION_fixnormz_D INT inline fixnormz_D(LONG value) { INT result; __asm__("clz %[result], %[value]" : [result] "=r"(result) : [value] "r"(value)); return result; } #endif #endif /* __mips__ */ #endif /* !defined(CLZ_MIPS_H) */ fdk-aac-2.0.2/libFDK/include/mips/cplx_mul_mips.h000066400000000000000000000135541404225310700215210ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Description: fixed point intrinsics *******************************************************************************/ #if !defined(CPLX_MUL_MIPS_H) #define CPLX_MUL_MIPS_H #if defined(__GNUC__) && defined(__mips__) //#define FUNCTION_cplxMultDiv2_32x16 //#define FUNCTION_cplxMultDiv2_32x16X2 #define FUNCTION_cplxMultDiv2_32x32X2 //#define FUNCTION_cplxMult_32x16 //#define FUNCTION_cplxMult_32x16X2 #define FUNCTION_cplxMult_32x32X2 #if defined(FUNCTION_cplxMultDiv2_32x32X2) inline void cplxMultDiv2(FIXP_DBL *c_Re, FIXP_DBL *c_Im, FIXP_DBL a_Re, FIXP_DBL a_Im, FIXP_DBL b_Re, FIXP_DBL b_Im) { *c_Re = (((long long)a_Re * (long long)b_Re) - ((long long)a_Im * (long long)b_Im))>>32; *c_Im = (((long long)a_Re * (long long)b_Im) + ((long long)a_Im * (long long)b_Re))>>32; } #endif #if defined(FUNCTION_cplxMult_32x32X2) inline void cplxMult(FIXP_DBL *c_Re, FIXP_DBL *c_Im, FIXP_DBL a_Re, FIXP_DBL a_Im, FIXP_DBL b_Re, FIXP_DBL b_Im) { *c_Re = ((((long long)a_Re * (long long)b_Re) - ((long long)a_Im * (long long)b_Im))>>32)<<1; *c_Im = ((((long long)a_Re * (long long)b_Im) + ((long long)a_Im * (long long)b_Re))>>32)<<1; } #endif #endif /* defined(__GNUC__) && defined(__mips__) */ #endif /* !defined(CPLX_MUL_MIPS_H) */ fdk-aac-2.0.2/libFDK/include/mips/fixmul_mips.h000066400000000000000000000124301404225310700211720ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Description: fixed point intrinsics *******************************************************************************/ #if !defined(FIXMUL_MIPS_H) #define FIXMUL_MIPS_H #if defined(__mips__) #if (__GNUC__) && defined(__mips__) /* MIPS GCC based compiler */ #define FUNCTION_fixmuldiv2_DD #define FUNCTION_fixmuldiv2BitExact_DD #define fixmuldiv2BitExact_DD(a, b) fixmuldiv2_DD(a, b) inline INT fixmuldiv2_DD(const INT a, const INT b) { INT result; result = ((long long)a * b) >> 32; return result; } #endif /* (__GNUC__) && defined(__mips__) */ #endif /* __mips__ */ #define FUNCTION_fixmulBitExact_DD #define fixmulBitExact_DD fixmul_DD #endif /* !defined(FIXMUL_MIPS_H) */ fdk-aac-2.0.2/libFDK/include/mips/scale_mips.h000066400000000000000000000123551404225310700207630ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Description: *******************************************************************************/ #ifndef SCALE_MIPS_H #define SCALE_MIPS_H #if defined(__mips_dsp) && !defined(__mips16) /*! * * \brief Scale input value by 2^{scale} and saturate output to 2^{dBits-1} * \return scaled and saturated value * * This macro scales src value right or left and applies saturation to * (2^dBits)-1 maxima output. */ #define SATURATE_RIGHT_SHIFT(src, scale, dBits) \ (__builtin_mips_shll_s_w((src) >> (scale), (DFRACT_BITS - (dBits))) >> \ (DFRACT_BITS - (dBits))) #endif /*__mips_dsp */ #endif /* SCALE_MIPS_H */ fdk-aac-2.0.2/libFDK/include/mips/scramble_mips.h000066400000000000000000000123621404225310700214620ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Description: *******************************************************************************/ #ifndef SCRAMBLE_MIPS_H #define SCRAMBLE_MIPS_H #define FUNCTION_scramble #if defined(FUNCTION_scramble) inline void scramble(FIXP_DBL *x, INT n) { INT m, j; int ldn = 1; do { ldn++; } while ((1 << ldn) < n); for (m = 1, j = 0; m < n - 1; m++) { j = __builtin_mips_bitrev(m) >> (16 - ldn); if (j > m) { FIXP_DBL tmp; tmp = x[2 * m]; x[2 * m] = x[2 * j]; x[2 * j] = tmp; tmp = x[2 * m + 1]; x[2 * m + 1] = x[2 * j + 1]; x[2 * j + 1] = tmp; } } } #endif #endif /* SCRAMBLE_MIPS_H */ fdk-aac-2.0.2/libFDK/include/nlc_dec.h000066400000000000000000000141551404225310700172630ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Omer Osman Description: SAC/SAOC Dec Noiseless Coding *******************************************************************************/ #ifndef NLC_DEC_H #define NLC_DEC_H #include "FDK_bitstream.h" #include "huff_nodes.h" #include "common_fix.h" typedef enum { SAC_DECODER, SAOC_DECODER, USAC_DECODER } DECODER_TYPE; typedef enum { t_CLD, t_ICC, t_IPD, t_OLD, t_IOC, t_NRG, t_DCLD, t_DMG, t_PDG } DATA_TYPE; typedef enum { BACKWARDS = 0x0, FORWARDS = 0x1 } DIRECTION; typedef enum { DIFF_FREQ = 0x0, DIFF_TIME = 0x1 } DIFF_TYPE; typedef enum { HUFF_1D = 0x0, HUFF_2D = 0x1 } CODING_SCHEME; typedef enum { FREQ_PAIR = 0x0, TIME_PAIR = 0x1 } PAIRING; #ifndef HUFFDEC_PARAMS #define HUFFDEC_PARMS #define PAIR_SHIFT 4 #define PAIR_MASK 0xf #define MAX_ENTRIES 168 #define HANDLE_HUFF_NODE const SHORT(*)[MAX_ENTRIES][2] #endif /* HUFFDECPARAMS */ #define HUFFDEC_OK 0 #define HUFFDEC_NOTOK (-1) typedef int ERROR_t; ERROR_t EcDataPairDec(DECODER_TYPE DECODER, HANDLE_FDK_BITSTREAM strm, SCHAR *aaOutData1, SCHAR *aaOutData2, SCHAR *aHistory, DATA_TYPE data_type, int startBand, int dataBands, int pair_flag, int coarse_flag, int allowDiffTimeBack_flag); /* needed for GES- & STP-tool */ ERROR_t huff_dec_reshape(HANDLE_FDK_BITSTREAM strm, int *out_data, int num_val); extern ERROR_t sym_restoreIPD(HANDLE_FDK_BITSTREAM strm, int lav, SCHAR data[2]); #endif fdk-aac-2.0.2/libFDK/include/ppc/000077500000000000000000000000001404225310700162775ustar00rootroot00000000000000fdk-aac-2.0.2/libFDK/include/ppc/clz_ppc.h000066400000000000000000000120521404225310700201020ustar00rootroot00000000000000 /* ----------------------------------------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Frderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------------------------------------- */ /*************************** Fraunhofer IIS FDK Tools ********************** Author(s): Description: fixed point intrinsics ******************************************************************************/ #if defined(__powerpc__) && (defined(__GNUC__) || defined(__xlC__)) #define FUNCTION_fixnormz_D inline INT fixnormz_D(LONG value) { INT result; __asm__ ("cntlzw %0, %1" : "=r" (result) : "r" (value)); return result; } #endif /* __powerpc__ && (__GNUC__ || __xlC__) */ fdk-aac-2.0.2/libFDK/include/ppc/fixmul_ppc.h000066400000000000000000000126501404225310700206220ustar00rootroot00000000000000 /* ----------------------------------------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Frderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------------------------------------- */ /*************************** Fraunhofer IIS FDK Tools ********************** Author(s): Description: fixed point intrinsics ******************************************************************************/ #if defined(__powerpc__) && (defined(__GNUC__) || defined(__xlC__)) #define FUNCTION_fixmuldiv2_DD #define FUNCTION_fixmuldiv2BitExact_DD #define fixmuldiv2BitExact_DD(a,b) fixmuldiv2_DD(a,b) #define FUNCTION_fixmulBitExact_DD #define fixmulBitExact_DD(a,b) fixmul_DD(a,b) #define FUNCTION_fixmuldiv2BitExact_DS #define fixmuldiv2BitExact_DS(a,b) fixmuldiv2_DS(a,b) #define FUNCTION_fixmulBitExact_DS #define fixmulBitExact_DS(a,b) fixmul_DS(a,b) inline INT fixmuldiv2_DD (const INT a, const INT b) { INT result; __asm__ ("mulhw %0, %1, %2" : "=r" (result) : "r" (a), "r" (b)); return result; } #endif /* __powerpc__ && (__GNUC__ || __xlC__) */ fdk-aac-2.0.2/libFDK/include/qmf.h000066400000000000000000000344221404225310700164560ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Description: *******************************************************************************/ /*! \file qmf.h \brief Complex qmf analysis/synthesis \author Markus Werner */ #ifndef QMF_H #define QMF_H #include "common_fix.h" #include "FDK_tools_rom.h" #include "dct.h" #define FIXP_QAS FIXP_PCM #define QAS_BITS SAMPLE_BITS #define INT_PCM_QMFIN INT_PCM #define FIXP_QSS FIXP_DBL #define QSS_BITS DFRACT_BITS /* Flags for QMF intialization */ /* Low Power mode flag */ #define QMF_FLAG_LP 1 /* Filter is not symmetric. This flag is set internally in the QMF * initialization as required. */ /* DO NOT PASS THIS FLAG TO qmfInitAnalysisFilterBank or * qmfInitSynthesisFilterBank */ #define QMF_FLAG_NONSYMMETRIC 2 /* Complex Low Delay Filter Bank (or std symmetric filter bank) */ #define QMF_FLAG_CLDFB 4 /* Flag indicating that the states should be kept. */ #define QMF_FLAG_KEEP_STATES 8 /* Complex Low Delay Filter Bank used in MPEG Surround Encoder */ #define QMF_FLAG_MPSLDFB 16 /* Complex Low Delay Filter Bank used in MPEG Surround Encoder allows a * optimized calculation of the modulation in qmfForwardModulationHQ() */ #define QMF_FLAG_MPSLDFB_OPTIMIZE_MODULATION 32 /* Flag to indicate HE-AAC down-sampled SBR mode (decoder) -> adapt analysis * post twiddling */ #define QMF_FLAG_DOWNSAMPLED 64 #define QMF_MAX_SYNTHESIS_BANDS (64) /*! * \brief Algorithmic scaling in sbrForwardModulation() * * The scaling in sbrForwardModulation() is caused by: * * \li 1 R_SHIFT in sbrForwardModulation() * \li 5/6 R_SHIFT in dct3() if using 32/64 Bands * \li 1 omitted gain of 2.0 in qmfForwardModulation() */ #define ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK 7 /*! * \brief Algorithmic scaling in cplxSynthesisQmfFiltering() * * The scaling in cplxSynthesisQmfFiltering() is caused by: * * \li 5/6 R_SHIFT in dct2() if using 32/64 Bands * \li 1 omitted gain of 2.0 in qmfInverseModulation() * \li -6 division by 64 in synthesis filterbank * \li x bits external influence */ #define ALGORITHMIC_SCALING_IN_SYNTHESIS_FILTERBANK 1 typedef struct { int lb_scale; /*!< Scale of low band area */ int ov_lb_scale; /*!< Scale of adjusted overlap low band area */ int hb_scale; /*!< Scale of high band area */ int ov_hb_scale; /*!< Scale of adjusted overlap high band area */ } QMF_SCALE_FACTOR; struct QMF_FILTER_BANK { const FIXP_PFT *p_filter; /*!< Pointer to filter coefficients */ void *FilterStates; /*!< Pointer to buffer of filter states FIXP_PCM in analyse and FIXP_DBL in synthesis filter */ int FilterSize; /*!< Size of prototype filter. */ const FIXP_QTW *t_cos; /*!< Modulation tables. */ const FIXP_QTW *t_sin; int filterScale; /*!< filter scale */ int no_channels; /*!< Total number of channels (subbands) */ int no_col; /*!< Number of time slots */ int lsb; /*!< Top of low subbands */ int usb; /*!< Top of high subbands */ int synScalefactor; /*!< Scale factor of synthesis qmf (syn only) */ int outScalefactor; /*!< Scale factor of output data (syn only) */ FIXP_DBL outGain_m; /*!< Mantissa of gain output data (syn only) (init with 0x80000000 to ignore) */ int outGain_e; /*!< Exponent of gain output data (syn only) */ UINT flags; /*!< flags */ UCHAR p_stride; /*!< Stride Factor of polyphase filters */ }; typedef struct QMF_FILTER_BANK *HANDLE_QMF_FILTER_BANK; int qmfInitAnalysisFilterBank( HANDLE_QMF_FILTER_BANK h_Qmf, /*!< QMF Handle */ FIXP_QAS *pFilterStates, /*!< Pointer to filter state buffer */ int noCols, /*!< Number of time slots */ int lsb, /*!< Number of lower bands */ int usb, /*!< Number of upper bands */ int no_channels, /*!< Number of critically sampled bands */ int flags); /*!< Flags */ #if SAMPLE_BITS == 16 int qmfInitAnalysisFilterBank( HANDLE_QMF_FILTER_BANK h_Qmf, /*!< QMF Handle */ FIXP_DBL *pFilterStates, /*!< Pointer to filter state buffer */ int noCols, /*!< Number of time slots */ int lsb, /*!< Number of lower bands */ int usb, /*!< Number of upper bands */ int no_channels, /*!< Number of critically sampled bands */ int flags); /*!< Flags */ #endif void qmfAnalysisFiltering( HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank */ FIXP_DBL **qmfReal, /*!< Pointer to real subband slots */ FIXP_DBL **qmfImag, /*!< Pointer to imag subband slots */ QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data */ const INT_PCM *timeIn, /*!< Time signal */ const int timeIn_e, /*!< Exponent of audio data */ const int stride, /*!< Stride factor of audio data */ FIXP_DBL *pWorkBuffer /*!< pointer to temporal working buffer */ ); #if SAMPLE_BITS == 16 void qmfAnalysisFiltering( HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank */ FIXP_DBL **qmfReal, /*!< Pointer to real subband slots */ FIXP_DBL **qmfImag, /*!< Pointer to imag subband slots */ QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data */ const LONG *timeIn, /*!< Time signal */ const int timeIn_e, /*!< Exponent of audio data */ const int stride, /*!< Stride factor of audio data */ FIXP_DBL *pWorkBuffer /*!< pointer to temporary working buffer */ ); #endif void qmfAnalysisFilteringSlot( HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Synthesis Bank */ FIXP_DBL *qmfReal, /*!< Low and High band, real */ FIXP_DBL *qmfImag, /*!< Low and High band, imag */ const INT_PCM *timeIn, /*!< Pointer to input */ const int stride, /*!< stride factor of input */ FIXP_DBL *pWorkBuffer /*!< pointer to temporal working buffer */ ); #if SAMPLE_BITS == 16 void qmfAnalysisFilteringSlot( HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Synthesis Bank */ FIXP_DBL *qmfReal, /*!< Low and High band, real */ FIXP_DBL *qmfImag, /*!< Low and High band, imag */ const LONG *timeIn, /*!< Pointer to input */ const int stride, /*!< stride factor of input */ FIXP_DBL *pWorkBuffer /*!< pointer to temporary working buffer */ ); #endif int qmfInitSynthesisFilterBank( HANDLE_QMF_FILTER_BANK h_Qmf, /*!< QMF Handle */ FIXP_QSS *pFilterStates, /*!< Pointer to filter state buffer */ int noCols, /*!< Number of time slots */ int lsb, /*!< Number of lower bands */ int usb, /*!< Number of upper bands */ int no_channels, /*!< Number of critically sampled bands */ int flags); /*!< Flags */ void qmfSynthesisFiltering( HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */ FIXP_DBL **QmfBufferReal, /*!< Pointer to real subband slots */ FIXP_DBL **QmfBufferImag, /*!< Pointer to imag subband slots */ const QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data */ const int ov_len, /*!< Length of band overlap */ INT_PCM *timeOut, /*!< Time signal */ const INT stride, /*!< Stride factor of audio data */ FIXP_DBL *pWorkBuffer /*!< pointer to temporary working buffer. It must be aligned */ ); #if SAMPLE_BITS == 16 void qmfSynthesisFiltering( HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */ FIXP_DBL **QmfBufferReal, /*!< Pointer to real subband slots */ FIXP_DBL **QmfBufferImag, /*!< Pointer to imag subband slots */ const QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data */ const int ov_len, /*!< Length of band overlap */ LONG *timeOut, /*!< Time signal */ const int timeOut_e, /*!< Target exponent for timeOut */ FIXP_DBL *pWorkBuffer /*!< pointer to temporary working buffer */ ); #endif void qmfSynthesisFilteringSlot(HANDLE_QMF_FILTER_BANK synQmf, const FIXP_DBL *realSlot, const FIXP_DBL *imagSlot, const int scaleFactorLowBand, const int scaleFactorHighBand, INT_PCM *timeOut, const int timeOut_e, FIXP_DBL *pWorkBuffer); #if SAMPLE_BITS == 16 void qmfSynthesisFilteringSlot(HANDLE_QMF_FILTER_BANK synQmf, const FIXP_DBL *realSlot, const FIXP_DBL *imagSlot, const int scaleFactorLowBand, const int scaleFactorHighBand, LONG *timeOut, const int timeOut_e, FIXP_DBL *pWorkBuffer); #endif void qmfChangeOutScalefactor( HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */ int outScalefactor /*!< New scaling factor for output data */ ); int qmfGetOutScalefactor( HANDLE_QMF_FILTER_BANK synQmf /*!< Handle of Qmf Synthesis Bank */ ); void qmfChangeOutGain( HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */ FIXP_DBL outputGain, /*!< New gain for output data (mantissa) */ int outputGainScale /*!< New gain for output data (exponent) */ ); #endif /*ifndef QMF_H */ fdk-aac-2.0.2/libFDK/include/qmf_pcm.h000066400000000000000000000565071404225310700173250ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Markus Lohwasser, Josef Hoepfl, Manuel Jander Description: QMF filterbank *******************************************************************************/ #ifndef QMF_PCM_H #define QMF_PCM_H /* All Synthesis functions dependent on datatype INT_PCM_QMFOUT Should only be included by qmf.cpp, but not compiled separately, please exclude compilation from project, if done otherwise. Is optional included twice to duplicate all functions with two different pre-definitions, as: #define INT_PCM_QMFOUT LONG and ... #define INT_PCM_QMFOUT SHORT needed to run QMF synthesis in both 16bit and 32bit sample output format. */ #define QSSCALE (0) #define FX_DBL2FX_QSS(x) (x) #define FX_QSS2FX_DBL(x) (x) /*! \brief Perform Synthesis Prototype Filtering on a single slot of input data. The filter takes 2 * qmf->no_channels of input data and generates qmf->no_channels time domain output samples. */ /* static */ #ifndef FUNCTION_qmfSynPrototypeFirSlot void qmfSynPrototypeFirSlot( #else void qmfSynPrototypeFirSlot_fallback( #endif HANDLE_QMF_FILTER_BANK qmf, FIXP_DBL *RESTRICT realSlot, /*!< Input: Pointer to real Slot */ FIXP_DBL *RESTRICT imagSlot, /*!< Input: Pointer to imag Slot */ INT_PCM_QMFOUT *RESTRICT timeOut, /*!< Time domain data */ int stride) { FIXP_QSS *FilterStates = (FIXP_QSS *)qmf->FilterStates; int no_channels = qmf->no_channels; const FIXP_PFT *p_Filter = qmf->p_filter; int p_stride = qmf->p_stride; int j; FIXP_QSS *RESTRICT sta = FilterStates; const FIXP_PFT *RESTRICT p_flt, *RESTRICT p_fltm; int scale = (DFRACT_BITS - SAMPLE_BITS_QMFOUT) - 1 - qmf->outScalefactor - qmf->outGain_e; p_flt = p_Filter + p_stride * QMF_NO_POLY; /* 5th of 330 */ p_fltm = p_Filter + (qmf->FilterSize / 2) - p_stride * QMF_NO_POLY; /* 5 + (320 - 2*5) = 315th of 330 */ FIXP_SGL gain = FX_DBL2FX_SGL(qmf->outGain_m); FIXP_DBL rnd_val = 0; if (scale > 0) { if (scale < (DFRACT_BITS - 1)) rnd_val = FIXP_DBL(1 << (scale - 1)); else scale = (DFRACT_BITS - 1); } else { scale = fMax(scale, -(DFRACT_BITS - 1)); } for (j = no_channels - 1; j >= 0; j--) { FIXP_DBL imag = imagSlot[j]; /* no_channels-1 .. 0 */ FIXP_DBL real = realSlot[j]; /* no_channels-1 .. 0 */ { INT_PCM_QMFOUT tmp; FIXP_DBL Are = fMultAddDiv2(FX_QSS2FX_DBL(sta[0]), p_fltm[0], real); /* This PCM formatting performs: - multiplication with 16-bit gain, if not -1.0f - rounding, if shift right is applied - apply shift left (or right) with saturation to 32 (or 16) bits - store output with --stride in 32 (or 16) bit format */ if (gain != (FIXP_SGL)(-32768)) /* -1.0f */ { Are = fMult(Are, gain); } if (scale >= 0) { FDK_ASSERT( Are <= (Are + rnd_val)); /* Round-addition must not overflow, might be equal for rnd_val=0 */ tmp = (INT_PCM_QMFOUT)( SATURATE_RIGHT_SHIFT(Are + rnd_val, scale, SAMPLE_BITS_QMFOUT)); } else { tmp = (INT_PCM_QMFOUT)( SATURATE_LEFT_SHIFT(Are, -scale, SAMPLE_BITS_QMFOUT)); } { timeOut[(j)*stride] = tmp; } } sta[0] = FX_DBL2FX_QSS(fMultAddDiv2(FX_QSS2FX_DBL(sta[1]), p_flt[4], imag)); sta[1] = FX_DBL2FX_QSS(fMultAddDiv2(FX_QSS2FX_DBL(sta[2]), p_fltm[1], real)); sta[2] = FX_DBL2FX_QSS(fMultAddDiv2(FX_QSS2FX_DBL(sta[3]), p_flt[3], imag)); sta[3] = FX_DBL2FX_QSS(fMultAddDiv2(FX_QSS2FX_DBL(sta[4]), p_fltm[2], real)); sta[4] = FX_DBL2FX_QSS(fMultAddDiv2(FX_QSS2FX_DBL(sta[5]), p_flt[2], imag)); sta[5] = FX_DBL2FX_QSS(fMultAddDiv2(FX_QSS2FX_DBL(sta[6]), p_fltm[3], real)); sta[6] = FX_DBL2FX_QSS(fMultAddDiv2(FX_QSS2FX_DBL(sta[7]), p_flt[1], imag)); sta[7] = FX_DBL2FX_QSS(fMultAddDiv2(FX_QSS2FX_DBL(sta[8]), p_fltm[4], real)); sta[8] = FX_DBL2FX_QSS(fMultDiv2(p_flt[0], imag)); p_flt += (p_stride * QMF_NO_POLY); p_fltm -= (p_stride * QMF_NO_POLY); sta += 9; // = (2*QMF_NO_POLY-1); } } #ifndef FUNCTION_qmfSynPrototypeFirSlot_NonSymmetric /*! \brief Perform Synthesis Prototype Filtering on a single slot of input data. The filter takes 2 * qmf->no_channels of input data and generates qmf->no_channels time domain output samples. */ static void qmfSynPrototypeFirSlot_NonSymmetric( HANDLE_QMF_FILTER_BANK qmf, FIXP_DBL *RESTRICT realSlot, /*!< Input: Pointer to real Slot */ FIXP_DBL *RESTRICT imagSlot, /*!< Input: Pointer to imag Slot */ INT_PCM_QMFOUT *RESTRICT timeOut, /*!< Time domain data */ int stride) { FIXP_QSS *FilterStates = (FIXP_QSS *)qmf->FilterStates; int no_channels = qmf->no_channels; const FIXP_PFT *p_Filter = qmf->p_filter; int p_stride = qmf->p_stride; int j; FIXP_QSS *RESTRICT sta = FilterStates; const FIXP_PFT *RESTRICT p_flt, *RESTRICT p_fltm; int scale = (DFRACT_BITS - SAMPLE_BITS_QMFOUT) - 1 - qmf->outScalefactor - qmf->outGain_e; p_flt = p_Filter; /*!< Pointer to first half of filter coefficients */ p_fltm = &p_flt[qmf->FilterSize / 2]; /* at index 320, overall 640 coefficients */ FIXP_SGL gain = FX_DBL2FX_SGL(qmf->outGain_m); FIXP_DBL rnd_val = (FIXP_DBL)0; if (scale > 0) { if (scale < (DFRACT_BITS - 1)) rnd_val = FIXP_DBL(1 << (scale - 1)); else scale = (DFRACT_BITS - 1); } else { scale = fMax(scale, -(DFRACT_BITS - 1)); } for (j = no_channels - 1; j >= 0; j--) { FIXP_DBL imag = imagSlot[j]; /* no_channels-1 .. 0 */ FIXP_DBL real = realSlot[j]; /* no_channels-1 .. 0 */ { INT_PCM_QMFOUT tmp; FIXP_DBL Are = sta[0] + FX_DBL2FX_QSS(fMultDiv2(p_fltm[4], real)); /* This PCM formatting performs: - multiplication with 16-bit gain, if not -1.0f - rounding, if shift right is applied - apply shift left (or right) with saturation to 32 (or 16) bits - store output with --stride in 32 (or 16) bit format */ if (gain != (FIXP_SGL)(-32768)) /* -1.0f */ { Are = fMult(Are, gain); } if (scale > 0) { FDK_ASSERT(Are < (Are + rnd_val)); /* Round-addition must not overflow */ tmp = (INT_PCM_QMFOUT)( SATURATE_RIGHT_SHIFT(Are + rnd_val, scale, SAMPLE_BITS_QMFOUT)); } else { tmp = (INT_PCM_QMFOUT)( SATURATE_LEFT_SHIFT(Are, -scale, SAMPLE_BITS_QMFOUT)); } timeOut[j * stride] = tmp; } sta[0] = sta[1] + FX_DBL2FX_QSS(fMultDiv2(p_flt[4], imag)); sta[1] = sta[2] + FX_DBL2FX_QSS(fMultDiv2(p_fltm[3], real)); sta[2] = sta[3] + FX_DBL2FX_QSS(fMultDiv2(p_flt[3], imag)); sta[3] = sta[4] + FX_DBL2FX_QSS(fMultDiv2(p_fltm[2], real)); sta[4] = sta[5] + FX_DBL2FX_QSS(fMultDiv2(p_flt[2], imag)); sta[5] = sta[6] + FX_DBL2FX_QSS(fMultDiv2(p_fltm[1], real)); sta[6] = sta[7] + FX_DBL2FX_QSS(fMultDiv2(p_flt[1], imag)); sta[7] = sta[8] + FX_DBL2FX_QSS(fMultDiv2(p_fltm[0], real)); sta[8] = FX_DBL2FX_QSS(fMultDiv2(p_flt[0], imag)); p_flt += (p_stride * QMF_NO_POLY); p_fltm += (p_stride * QMF_NO_POLY); sta += 9; // = (2*QMF_NO_POLY-1); } } #endif /* FUNCTION_qmfSynPrototypeFirSlot_NonSymmetric */ void qmfSynthesisFilteringSlot(HANDLE_QMF_FILTER_BANK synQmf, const FIXP_DBL *realSlot, const FIXP_DBL *imagSlot, const int scaleFactorLowBand, const int scaleFactorHighBand, INT_PCM_QMFOUT *timeOut, const int stride, FIXP_DBL *pWorkBuffer) { if (!(synQmf->flags & QMF_FLAG_LP)) qmfInverseModulationHQ(synQmf, realSlot, imagSlot, scaleFactorLowBand, scaleFactorHighBand, pWorkBuffer); else { if (synQmf->flags & QMF_FLAG_CLDFB) { qmfInverseModulationLP_odd(synQmf, realSlot, scaleFactorLowBand, scaleFactorHighBand, pWorkBuffer); } else { qmfInverseModulationLP_even(synQmf, realSlot, scaleFactorLowBand, scaleFactorHighBand, pWorkBuffer); } } if (synQmf->flags & QMF_FLAG_NONSYMMETRIC) { qmfSynPrototypeFirSlot_NonSymmetric(synQmf, pWorkBuffer, pWorkBuffer + synQmf->no_channels, timeOut, stride); } else { qmfSynPrototypeFirSlot(synQmf, pWorkBuffer, pWorkBuffer + synQmf->no_channels, timeOut, stride); } } /*! * * \brief Perform complex-valued subband synthesis of the * low band and the high band and store the * time domain data in timeOut * * First step: Calculate the proper scaling factor of current * spectral data in qmfReal/qmfImag, old spectral data in the overlap * range and filter states. * * Second step: Perform Frequency-to-Time mapping with inverse * Modulation slot-wise. * * Third step: Perform FIR-filter slot-wise. To save space for filter * states, the MAC operations are executed directly on the filter states * instead of accumulating several products in the accumulator. The * buffer shift at the end of the function should be replaced by a * modulo operation, which is available on some DSPs. * * Last step: Copy the upper part of the spectral data to the overlap buffer. * * The qmf coefficient table is symmetric. The symmetry is exploited by * shrinking the coefficient table to half the size. The addressing mode * takes care of the symmetries. If the #define #QMFTABLE_FULL is set, * coefficient addressing works on the full table size. The code will be * slightly faster and slightly more compact. * * Workbuffer requirement: 2 x sizeof(**QmfBufferReal) * synQmf->no_channels * The workbuffer must be aligned */ void qmfSynthesisFiltering( HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */ FIXP_DBL **QmfBufferReal, /*!< Low and High band, real */ FIXP_DBL **QmfBufferImag, /*!< Low and High band, imag */ const QMF_SCALE_FACTOR *scaleFactor, const INT ov_len, /*!< split Slot of overlap and actual slots */ INT_PCM_QMFOUT *timeOut, /*!< Pointer to output */ const INT stride, /*!< stride factor of output */ FIXP_DBL *pWorkBuffer /*!< pointer to temporal working buffer */ ) { int i; int L = synQmf->no_channels; int scaleFactorHighBand; int scaleFactorLowBand_ov, scaleFactorLowBand_no_ov; FDK_ASSERT(synQmf->no_channels >= synQmf->lsb); FDK_ASSERT(synQmf->no_channels >= synQmf->usb); /* adapt scaling */ scaleFactorHighBand = -ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK - scaleFactor->hb_scale - synQmf->filterScale; scaleFactorLowBand_ov = -ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK - scaleFactor->ov_lb_scale - synQmf->filterScale; scaleFactorLowBand_no_ov = -ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK - scaleFactor->lb_scale - synQmf->filterScale; for (i = 0; i < synQmf->no_col; i++) /* ----- no_col loop ----- */ { const FIXP_DBL *QmfBufferImagSlot = NULL; int scaleFactorLowBand = (i < ov_len) ? scaleFactorLowBand_ov : scaleFactorLowBand_no_ov; if (!(synQmf->flags & QMF_FLAG_LP)) QmfBufferImagSlot = QmfBufferImag[i]; qmfSynthesisFilteringSlot(synQmf, QmfBufferReal[i], QmfBufferImagSlot, scaleFactorLowBand, scaleFactorHighBand, timeOut + (i * L * stride), stride, pWorkBuffer); } /* no_col loop i */ } /*! * * \brief Create QMF filter bank instance * * * \return 0 if successful * */ int qmfInitAnalysisFilterBank( HANDLE_QMF_FILTER_BANK h_Qmf, /*!< Returns handle */ FIXP_QAS *pFilterStates, /*!< Handle to filter states */ int noCols, /*!< Number of timeslots per frame */ int lsb, /*!< lower end of QMF */ int usb, /*!< upper end of QMF */ int no_channels, /*!< Number of channels (bands) */ int flags) /*!< Low Power flag */ { int err = qmfInitFilterBank(h_Qmf, pFilterStates, noCols, lsb, usb, no_channels, flags, 0); if (!(flags & QMF_FLAG_KEEP_STATES) && (h_Qmf->FilterStates != NULL)) { FDKmemclear(h_Qmf->FilterStates, (2 * QMF_NO_POLY - 1) * h_Qmf->no_channels * sizeof(FIXP_QAS)); } FDK_ASSERT(h_Qmf->no_channels >= h_Qmf->lsb); return err; } #ifndef FUNCTION_qmfAnaPrototypeFirSlot /*! \brief Perform Analysis Prototype Filtering on a single slot of input data. */ static void qmfAnaPrototypeFirSlot( FIXP_DBL *analysisBuffer, INT no_channels, /*!< Number channels of analysis filter */ const FIXP_PFT *p_filter, INT p_stride, /*!< Stride of analysis filter */ FIXP_QAS *RESTRICT pFilterStates) { INT k; FIXP_DBL accu; const FIXP_PFT *RESTRICT p_flt = p_filter; FIXP_DBL *RESTRICT pData_0 = analysisBuffer + 2 * no_channels - 1; FIXP_DBL *RESTRICT pData_1 = analysisBuffer; FIXP_QAS *RESTRICT sta_0 = (FIXP_QAS *)pFilterStates; FIXP_QAS *RESTRICT sta_1 = (FIXP_QAS *)pFilterStates + (2 * QMF_NO_POLY * no_channels) - 1; INT pfltStep = QMF_NO_POLY * (p_stride); INT staStep1 = no_channels << 1; INT staStep2 = (no_channels << 3) - 1; /* Rewind one less */ /* FIR filters 127..64 0..63 */ for (k = 0; k < no_channels; k++) { accu = fMultDiv2(p_flt[0], *sta_1); sta_1 -= staStep1; accu += fMultDiv2(p_flt[1], *sta_1); sta_1 -= staStep1; accu += fMultDiv2(p_flt[2], *sta_1); sta_1 -= staStep1; accu += fMultDiv2(p_flt[3], *sta_1); sta_1 -= staStep1; accu += fMultDiv2(p_flt[4], *sta_1); *pData_1++ = (accu << 1); sta_1 += staStep2; p_flt += pfltStep; accu = fMultDiv2(p_flt[0], *sta_0); sta_0 += staStep1; accu += fMultDiv2(p_flt[1], *sta_0); sta_0 += staStep1; accu += fMultDiv2(p_flt[2], *sta_0); sta_0 += staStep1; accu += fMultDiv2(p_flt[3], *sta_0); sta_0 += staStep1; accu += fMultDiv2(p_flt[4], *sta_0); *pData_0-- = (accu << 1); sta_0 -= staStep2; } } #endif /* !defined(FUNCTION_qmfAnaPrototypeFirSlot) */ #ifndef FUNCTION_qmfAnaPrototypeFirSlot_NonSymmetric /*! \brief Perform Analysis Prototype Filtering on a single slot of input data. */ static void qmfAnaPrototypeFirSlot_NonSymmetric( FIXP_DBL *analysisBuffer, int no_channels, /*!< Number channels of analysis filter */ const FIXP_PFT *p_filter, int p_stride, /*!< Stride of analysis filter */ FIXP_QAS *RESTRICT pFilterStates) { const FIXP_PFT *RESTRICT p_flt = p_filter; int p, k; for (k = 0; k < 2 * no_channels; k++) { FIXP_DBL accu = (FIXP_DBL)0; p_flt += QMF_NO_POLY * (p_stride - 1); /* Perform FIR-Filter */ for (p = 0; p < QMF_NO_POLY; p++) { accu += fMultDiv2(*p_flt++, pFilterStates[2 * no_channels * p]); } analysisBuffer[2 * no_channels - 1 - k] = (accu << 1); pFilterStates++; } } #endif /* FUNCTION_qmfAnaPrototypeFirSlot_NonSymmetric */ /* * \brief Perform one QMF slot analysis of the time domain data of timeIn * with specified stride and stores the real part of the subband * samples in rSubband, and the imaginary part in iSubband * * Note: anaQmf->lsb can be greater than anaQmf->no_channels in case * of implicit resampling (USAC with reduced 3/4 core frame length). */ void qmfAnalysisFilteringSlot( HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Synthesis Bank */ FIXP_DBL *qmfReal, /*!< Low and High band, real */ FIXP_DBL *qmfImag, /*!< Low and High band, imag */ const INT_PCM_QMFIN *RESTRICT timeIn, /*!< Pointer to input */ const int stride, /*!< stride factor of input */ FIXP_DBL *pWorkBuffer /*!< pointer to temporal working buffer */ ) { int offset = anaQmf->no_channels * (QMF_NO_POLY * 2 - 1); /* Feed time signal into oldest anaQmf->no_channels states */ { FIXP_QAS *FilterStatesAnaTmp = ((FIXP_QAS *)anaQmf->FilterStates) + offset; /* Feed and scale actual time in slot */ for (int i = anaQmf->no_channels >> 1; i != 0; i--) { /* Place INT_PCM value left aligned in scaledTimeIn */ *FilterStatesAnaTmp++ = (FIXP_QAS)*timeIn; timeIn += stride; *FilterStatesAnaTmp++ = (FIXP_QAS)*timeIn; timeIn += stride; } } if (anaQmf->flags & QMF_FLAG_NONSYMMETRIC) { qmfAnaPrototypeFirSlot_NonSymmetric(pWorkBuffer, anaQmf->no_channels, anaQmf->p_filter, anaQmf->p_stride, (FIXP_QAS *)anaQmf->FilterStates); } else { qmfAnaPrototypeFirSlot(pWorkBuffer, anaQmf->no_channels, anaQmf->p_filter, anaQmf->p_stride, (FIXP_QAS *)anaQmf->FilterStates); } if (anaQmf->flags & QMF_FLAG_LP) { if (anaQmf->flags & QMF_FLAG_CLDFB) qmfForwardModulationLP_odd(anaQmf, pWorkBuffer, qmfReal); else qmfForwardModulationLP_even(anaQmf, pWorkBuffer, qmfReal); } else { qmfForwardModulationHQ(anaQmf, pWorkBuffer, qmfReal, qmfImag); } /* Shift filter states Should be realized with modulo addressing on a DSP instead of a true buffer shift */ FDKmemmove(anaQmf->FilterStates, (FIXP_QAS *)anaQmf->FilterStates + anaQmf->no_channels, offset * sizeof(FIXP_QAS)); } /*! * * \brief Perform complex-valued subband filtering of the time domain * data of timeIn and stores the real part of the subband * samples in rAnalysis, and the imaginary part in iAnalysis * The qmf coefficient table is symmetric. The symmetry is expoited by * shrinking the coefficient table to half the size. The addressing mode * takes care of the symmetries. * * * \sa PolyphaseFiltering */ void qmfAnalysisFiltering( HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank */ FIXP_DBL **qmfReal, /*!< Pointer to real subband slots */ FIXP_DBL **qmfImag, /*!< Pointer to imag subband slots */ QMF_SCALE_FACTOR *scaleFactor, const INT_PCM_QMFIN *timeIn, /*!< Time signal */ const int timeIn_e, const int stride, FIXP_DBL *pWorkBuffer /*!< pointer to temporal working buffer */ ) { int i; int no_channels = anaQmf->no_channels; scaleFactor->lb_scale = -ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK - timeIn_e; scaleFactor->lb_scale -= anaQmf->filterScale; for (i = 0; i < anaQmf->no_col; i++) { FIXP_DBL *qmfImagSlot = NULL; if (!(anaQmf->flags & QMF_FLAG_LP)) { qmfImagSlot = qmfImag[i]; } qmfAnalysisFilteringSlot(anaQmf, qmfReal[i], qmfImagSlot, timeIn, stride, pWorkBuffer); timeIn += no_channels * stride; } /* no_col loop i */ } #endif /* QMF_PCM_H */ fdk-aac-2.0.2/libFDK/include/scale.h000066400000000000000000000265551404225310700167720ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Description: Scaling operations *******************************************************************************/ #ifndef SCALE_H #define SCALE_H #include "common_fix.h" #include "genericStds.h" #include "fixminmax.h" #define SCALE_INLINE #if defined(__arm__) #include "arm/scale_arm.h" #elif defined(__mips__) #include "mips/scale_mips.h" #endif void scaleValues(FIXP_SGL *vector, INT len, INT scalefactor); void scaleValues(FIXP_DBL *vector, INT len, INT scalefactor); void scaleValues(FIXP_DBL *dst, const FIXP_DBL *src, INT len, INT scalefactor); #if (SAMPLE_BITS == 16) void scaleValues(FIXP_PCM *dst, const FIXP_DBL *src, INT len, INT scalefactor); #endif void scaleValues(FIXP_SGL *dst, const FIXP_SGL *src, INT len, INT scalefactor); void scaleCplxValues(FIXP_DBL *r_dst, FIXP_DBL *i_dst, const FIXP_DBL *r_src, const FIXP_DBL *i_src, INT len, INT scalefactor); void scaleValuesWithFactor(FIXP_DBL *vector, FIXP_DBL factor, INT len, INT scalefactor); void scaleValuesSaturate(FIXP_DBL *vector, INT len, INT scalefactor); void scaleValuesSaturate(FIXP_DBL *dst, const FIXP_DBL *src, INT len, INT scalefactor); void scaleValuesSaturate(FIXP_SGL *dst, const FIXP_DBL *src, INT len, INT scalefactor); void scaleValuesSaturate(FIXP_SGL *vector, INT len, INT scalefactor); void scaleValuesSaturate(FIXP_SGL *dst, const FIXP_SGL *src, INT len, INT scalefactor); INT getScalefactorShort(const SHORT *vector, INT len); INT getScalefactorPCM(const INT_PCM *vector, INT len, INT stride); INT getScalefactor(const FIXP_DBL *vector, INT len); INT getScalefactor(const FIXP_SGL *vector, INT len); #ifndef FUNCTION_scaleValue /*! * * \brief Multiply input by \f$ 2^{scalefactor} \f$ * * \return Scaled input * */ #define FUNCTION_scaleValue inline FIXP_DBL scaleValue(const FIXP_DBL value, /*!< Value */ INT scalefactor /*!< Scalefactor */ ) { if (scalefactor > 0) return (value << scalefactor); else return (value >> (-scalefactor)); } inline FIXP_SGL scaleValue(const FIXP_SGL value, /*!< Value */ INT scalefactor /*!< Scalefactor */ ) { if (scalefactor > 0) return (value << scalefactor); else return (value >> (-scalefactor)); } #endif #ifndef FUNCTION_scaleValueSaturate /*! * * \brief Multiply input by \f$ 2^{scalefactor} \f$ * \param value The value to be scaled. * \param the shift amount * \return \f$ value * 2^scalefactor \f$ * */ #define FUNCTION_scaleValueSaturate inline FIXP_DBL scaleValueSaturate(const FIXP_DBL value, INT scalefactor /* in range -31 ... +31 */ ) { int headroom = fixnormz_D( (INT)value ^ (INT)((value >> 31))); /* headroom in range 1...32 */ if (scalefactor >= 0) { /* shift left: saturate in case of headroom less/equal scalefactor */ if (headroom <= scalefactor) { if (value > (FIXP_DBL)0) return (FIXP_DBL)MAXVAL_DBL; /* 0x7FFF.FFFF */ else return (FIXP_DBL)MINVAL_DBL + (FIXP_DBL)1; /* 0x8000.0001 */ } else { return fMax((value << scalefactor), (FIXP_DBL)MINVAL_DBL + (FIXP_DBL)1); } } else { scalefactor = -scalefactor; /* shift right: clear in case of 32-headroom greater/equal -scalefactor */ if ((DFRACT_BITS - headroom) <= scalefactor) { return (FIXP_DBL)0; } else { return fMax((value >> scalefactor), (FIXP_DBL)MINVAL_DBL + (FIXP_DBL)1); } } } #endif #ifndef FUNCTION_scaleValueInPlace /*! * * \brief Multiply input by \f$ 2^{scalefactor} \f$ in place * * \return void * */ #define FUNCTION_scaleValueInPlace inline void scaleValueInPlace(FIXP_DBL *value, /*!< Value */ INT scalefactor /*!< Scalefactor */ ) { INT newscale; /* Note: The assignment inside the if conditional allows combining a load with * the compare to zero (on ARM and maybe others) */ if ((newscale = (scalefactor)) >= 0) { *(value) <<= newscale; } else { *(value) >>= -newscale; } } #endif /*! * * \brief Scale input value by 2^{scale} and saturate output to 2^{dBits-1} * \return scaled and saturated value * * This macro scales src value right or left and applies saturation to * (2^dBits)-1 maxima output. */ #ifndef SATURATE_RIGHT_SHIFT #define SATURATE_RIGHT_SHIFT(src, scale, dBits) \ ((((LONG)(src) >> (scale)) > (LONG)(((1U) << ((dBits)-1)) - 1)) \ ? (LONG)(((1U) << ((dBits)-1)) - 1) \ : (((LONG)(src) >> (scale)) < ~((LONG)(((1U) << ((dBits)-1)) - 1))) \ ? ~((LONG)(((1U) << ((dBits)-1)) - 1)) \ : ((LONG)(src) >> (scale))) #endif #ifndef SATURATE_LEFT_SHIFT #define SATURATE_LEFT_SHIFT(src, scale, dBits) \ (((LONG)(src) > ((LONG)(((1U) << ((dBits)-1)) - 1) >> (scale))) \ ? (LONG)(((1U) << ((dBits)-1)) - 1) \ : ((LONG)(src) < ~((LONG)(((1U) << ((dBits)-1)) - 1) >> (scale))) \ ? ~((LONG)(((1U) << ((dBits)-1)) - 1)) \ : ((LONG)(src) << (scale))) #endif #ifndef SATURATE_SHIFT #define SATURATE_SHIFT(src, scale, dBits) \ (((scale) < 0) ? SATURATE_LEFT_SHIFT((src), -(scale), (dBits)) \ : SATURATE_RIGHT_SHIFT((src), (scale), (dBits))) #endif /* * Alternative shift and saturate left, saturates to -0.99999 instead of -1.0000 * to avoid problems when inverting the sign of the result. */ #ifndef SATURATE_LEFT_SHIFT_ALT #define SATURATE_LEFT_SHIFT_ALT(src, scale, dBits) \ (((LONG)(src) > ((LONG)(((1U) << ((dBits)-1)) - 1) >> (scale))) \ ? (LONG)(((1U) << ((dBits)-1)) - 1) \ : ((LONG)(src) <= ~((LONG)(((1U) << ((dBits)-1)) - 1) >> (scale))) \ ? ~((LONG)(((1U) << ((dBits)-1)) - 2)) \ : ((LONG)(src) << (scale))) #endif #ifndef SATURATE_RIGHT_SHIFT_ALT #define SATURATE_RIGHT_SHIFT_ALT(src, scale, dBits) \ ((((LONG)(src) >> (scale)) > (LONG)(((1U) << ((dBits)-1)) - 1)) \ ? (LONG)(((1U) << ((dBits)-1)) - 1) \ : (((LONG)(src) >> (scale)) < ~((LONG)(((1U) << ((dBits)-1)) - 2))) \ ? ~((LONG)(((1U) << ((dBits)-1)) - 2)) \ : ((LONG)(src) >> (scale))) #endif #ifndef SATURATE_INT_PCM_RIGHT_SHIFT #define SATURATE_INT_PCM_RIGHT_SHIFT(src, scale) \ SATURATE_RIGHT_SHIFT(src, scale, SAMPLE_BITS) #endif #ifndef SATURATE_INT_PCM_LEFT_SHIFT #define SATURATE_INT_PCM_LEFT_SHIFT(src, scale) \ SATURATE_LEFT_SHIFT(src, scale, SAMPLE_BITS) #endif #endif /* #ifndef SCALE_H */ fdk-aac-2.0.2/libFDK/include/scramble.h000066400000000000000000000134351404225310700174640ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Description: *******************************************************************************/ #ifndef SCRAMBLE_H #define SCRAMBLE_H #include "common_fix.h" #if defined(__arm__) #include "arm/scramble_arm.h" #elif defined(__mips__) && defined(__mips_dsp) && !defined(__mips16) #include "mips/scramble_mips.h" #endif /***************************************************************************** functionname: scramble description: bitreversal of input data returns: input: output: *****************************************************************************/ #if !defined(FUNCTION_scramble) /* default scramble functionality */ inline void scramble(FIXP_DBL *x, INT length) { INT m, k, j; FDK_ASSERT(!(((INT)(INT64)x) & (ALIGNMENT_DEFAULT - 1))); C_ALLOC_ALIGNED_CHECK(x); for (m = 1, j = 0; m < length - 1; m++) { { for (k = length >> 1; (!((j ^= k) & k)); k >>= 1) ; } if (j > m) { FIXP_DBL tmp; tmp = x[2 * m]; x[2 * m] = x[2 * j]; x[2 * j] = tmp; tmp = x[2 * m + 1]; x[2 * m + 1] = x[2 * j + 1]; x[2 * j + 1] = tmp; } } } #endif /* !defined(FUNCTION_scramble) */ #endif /* SCRAMBLE_H */ fdk-aac-2.0.2/libFDK/include/x86/000077500000000000000000000000001404225310700161425ustar00rootroot00000000000000fdk-aac-2.0.2/libFDK/include/x86/abs_x86.h000066400000000000000000000122001404225310700175600ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Description: fixed point intrinsics *******************************************************************************/ #if !defined(ABS_X86_H) #define ABS_X86_H #if defined(__x86__) #if defined(__x86_64__) inline INT fixabs_D(INT x) { return ((x) > (INT)(0)) ? (x) : -(x); } inline INT fixabs_S(INT x) { return ((x) > (INT)(0)) ? (x) : -(x); } #define fixabs_I(x) fixabs_D(x) #define FUNCTION_fixabs_S #define FUNCTION_fixabs_D #define FUNCTION_fixabs_I #endif /* __x86_64__ */ #endif /*__x86__ */ #endif /* !defined(ABS_X86_H) */ fdk-aac-2.0.2/libFDK/include/x86/clz_x86.h000066400000000000000000000134761404225310700176230ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Manuel Jander Description: x86 version of count leading zero / bits *******************************************************************************/ #if !defined(CLZ_X86_H) #define CLZ_X86_H #if defined(__GNUC__) && (defined(__x86__) || defined(__x86_64__)) #define FUNCTION_fixnormz_D #define FUNCTION_fixnorm_D inline INT fixnormz_D(LONG value) { INT result; if (value != 0) { result = __builtin_clz(value); } else { result = 32; } return result; } inline INT fixnorm_D(LONG value) { INT result; if (value == 0) { return 0; } if (value < 0) { value = ~value; } result = fixnormz_D(value); return result - 1; } #elif (_MSC_VER > 1200) && (defined(_M_IX86) || defined(_M_X64)) #include #define FUNCTION_fixnormz_D #define FUNCTION_fixnorm_D inline INT fixnormz_D(LONG value) { unsigned long result = 0; unsigned char err; err = _BitScanReverse(&result, value); if (err) { return 31 - result; } else { return 32; } } inline INT fixnorm_D(LONG value) { INT result; if (value == 0) { return 0; } if (value < 0) { value = ~value; } result = fixnormz_D(value); return result - 1; } #endif /* toolchain */ #endif /* !defined(CLZ_X86_H) */ fdk-aac-2.0.2/libFDK/include/x86/fixmul_x86.h000066400000000000000000000147571404225310700203420ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Description: fixed point intrinsics *******************************************************************************/ #if !defined(FIXMUL_X86_H) #define FIXMUL_X86_H #if defined(__x86__) #if defined(_MSC_VER) && defined(_M_IX86) /* Intel x86 */ #define FUNCTION_fixmul_DD #define FUNCTION_fixmuldiv2_DD #define FUNCTION_fixmuldiv2BitExact_DD #define fixmuldiv2BitExact_DD(a, b) fixmuldiv2_DD(a, b) #define FUNCTION_fixmulBitExact_DD #define fixmulBitExact_DD(a, b) fixmul_DD(a, b) #define FUNCTION_fixmuldiv2BitExact_DS #define fixmuldiv2BitExact_DS(a, b) fixmuldiv2_DS(a, b) #define FUNCTION_fixmulBitExact_DS #define fixmulBitExact_DS(a, b) fixmul_DS(a, b) inline INT fixmul_DD(INT a, const INT b) { __asm { mov eax, a imul b shl edx, 1 mov a, edx } return a; } inline INT fixmuldiv2_DD(INT a, const INT b) { __asm { mov eax, a imul b mov a, edx } return a; } /* ############################################################################# */ #elif (defined(__GNUC__) || defined(__gnu_linux__)) && defined(__x86__) #define FUNCTION_fixmul_DD #define FUNCTION_fixmuldiv2_DD #define FUNCTION_fixmuldiv2BitExact_DD #define fixmuldiv2BitExact_DD(a, b) fixmuldiv2_DD(a, b) #define FUNCTION_fixmulBitExact_DD #define fixmulBitExact_DD(a, b) fixmul_DD(a, b) #define FUNCTION_fixmuldiv2BitExact_DS #define fixmuldiv2BitExact_DS(a, b) fixmuldiv2_DS(a, b) #define FUNCTION_fixmulBitExact_DS #define fixmulBitExact_DS(a, b) fixmul_DS(a, b) inline INT fixmul_DD(INT a, const INT b) { INT result; asm("imul %2;\n" "shl $1, %0;\n" : "=d"(result), "+a"(a) : "r"(b)); return result; } inline INT fixmuldiv2_DD(INT a, const INT b) { INT result; asm("imul %2;" : "=d"(result), "+a"(a) : "r"(b)); return result; } #endif /* (defined(__GNUC__)||defined(__gnu_linux__)) && defined(__x86__) */ #endif /* __x86__ */ #endif /* !defined(FIXMUL_X86_H) */ fdk-aac-2.0.2/libFDK/include/x86/fixpoint_math_x86.h000066400000000000000000000172741404225310700217040ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Manuel Jander Description: Fixed point specific mathematical functions for x86 *******************************************************************************/ #if !defined(FIXPOINT_MATH_X86_H) #define FIXPOINT_MATH_X86_H #define FUNCTION_sqrtFixp #include #ifdef FUNCTION_sqrtFixp static inline FIXP_DBL sqrtFixp(const FIXP_DBL op) { FIXP_DBL result; /* result = * (FIXP_DBL)(INT)(sqrt((double)(INT)op)*46340.950011841578559133736114903); */ result = (FIXP_DBL)(INT)(sqrt((float)(INT)op) * 46340.9492f); FDK_ASSERT(result >= (FIXP_DBL)0); return result; } #endif /* FUNCTION_sqrtFixp */ #include #define FUNCTION_invSqrtNorm2 /** * \brief calculate 1.0/sqrt(op) * \param op_m mantissa of input value. * \param result_e pointer to return the exponent of the result * \return mantissa of the result */ #ifdef FUNCTION_invSqrtNorm2 inline FIXP_DBL invSqrtNorm2(FIXP_DBL op_m, INT *result_e) { float result; if (op_m == (FIXP_DBL)0) { *result_e = 16; return ((LONG)0x7fffffff); } result = (float)(1.0 / sqrt(0.5f * (float)(INT)op_m)); result = (float)ldexp(frexpf(result, result_e), DFRACT_BITS - 1); *result_e += 15; FDK_ASSERT(result >= 0); return (FIXP_DBL)(INT)result; } #endif /* FUNCTION_invSqrtNorm2 */ #define FUNCTION_invFixp /** * \brief calculate 1.0/op * \param op mantissa of the input value. * \return mantissa of the result with implizit exponent of 31 */ #ifdef FUNCTION_invFixp inline FIXP_DBL invFixp(FIXP_DBL op) { float result; INT result_e; if ((op == (FIXP_DBL)0) || (op == (FIXP_DBL)1)) { return ((LONG)0x7fffffff); } result = (float)(1.0 / (float)(INT)op); result = frexpf(result, &result_e); result = ldexpf(result, 31 + result_e); return (FIXP_DBL)(INT)result; } /** * \brief calculate 1.0/(op_m * 2^op_e) * \param op_m mantissa of the input value. * \param op_e pointer into were the exponent of the input value is stored, and * the result will be stored into. * \return mantissa of the result */ inline FIXP_DBL invFixp(FIXP_DBL op_m, int *op_e) { float result; INT result_e; if ((op_m == (FIXP_DBL)0x00000000) || (op_m == (FIXP_DBL)0x00000001)) { *op_e = 31 - *op_e; return ((LONG)0x7fffffff); } result = (float)(1.0 / (float)(INT)op_m); result = ldexpf(frexpf(result, &result_e), DFRACT_BITS - 1); *op_e = result_e - *op_e + 31; return (FIXP_DBL)(INT)result; } #endif /* FUNCTION_invFixp */ #define FUNCTION_schur_div /** * \brief Divide two FIXP_DBL values with given precision. * \param num dividend * \param denum divisor * \param count amount of significant bits of the result (starting to the MSB) * \return num/divisor */ #ifdef FUNCTION_schur_div inline FIXP_DBL schur_div(FIXP_DBL num, FIXP_DBL denum, INT count) { (void)count; /* same asserts than for fallback implementation */ FDK_ASSERT(num >= (FIXP_DBL)0); FDK_ASSERT(denum > (FIXP_DBL)0); FDK_ASSERT(num <= denum); return (num == denum) ? (FIXP_DBL)MAXVAL_DBL : (FIXP_DBL)(INT)(((INT64)(INT)num << 31) / (INT)denum); } #endif /* FUNCTION_schur_div */ #endif /* !defined(FIXPOINT_MATH_X86_H) */ fdk-aac-2.0.2/libFDK/src/000077500000000000000000000000001404225310700146615ustar00rootroot00000000000000fdk-aac-2.0.2/libFDK/src/FDK_bitbuffer.cpp000066400000000000000000000433661404225310700200350ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): M. Lohwasser Description: common bitbuffer read/write routines *******************************************************************************/ #include "FDK_bitbuffer.h" #include "genericStds.h" #include "common_fix.h" #include "fixminmax.h" const UINT BitMask[32 + 1] = { 0x0, 0x1, 0x3, 0x7, 0xf, 0x1f, 0x3f, 0x7f, 0xff, 0x1ff, 0x3ff, 0x7ff, 0xfff, 0x1fff, 0x3fff, 0x7fff, 0xffff, 0x1ffff, 0x3ffff, 0x7ffff, 0xfffff, 0x1fffff, 0x3fffff, 0x7fffff, 0xffffff, 0x1ffffff, 0x3ffffff, 0x7ffffff, 0xfffffff, 0x1fffffff, 0x3fffffff, 0x7fffffff, 0xffffffff}; void FDK_CreateBitBuffer(HANDLE_FDK_BITBUF *hBitBuf, UCHAR *pBuffer, UINT bufSize) { FDK_InitBitBuffer(*hBitBuf, pBuffer, bufSize, 0); FDKmemclear((*hBitBuf)->Buffer, bufSize * sizeof(UCHAR)); } void FDK_DeleteBitBuffer(HANDLE_FDK_BITBUF hBitBuf) { ; } void FDK_InitBitBuffer(HANDLE_FDK_BITBUF hBitBuf, UCHAR *pBuffer, UINT bufSize, UINT validBits) { hBitBuf->ValidBits = validBits; hBitBuf->ReadOffset = 0; hBitBuf->WriteOffset = 0; hBitBuf->BitNdx = 0; hBitBuf->Buffer = pBuffer; hBitBuf->bufSize = bufSize; hBitBuf->bufBits = (bufSize << 3); /*assure bufsize (2^n) */ FDK_ASSERT(hBitBuf->ValidBits <= hBitBuf->bufBits); FDK_ASSERT((bufSize > 0) && (bufSize <= MAX_BUFSIZE_BYTES)); { UINT x = 0, n = bufSize; for (x = 0; n > 0; x++, n >>= 1) { } if (bufSize != ((UINT)1 << (x - 1))) { FDK_ASSERT(0); } } } void FDK_ResetBitBuffer(HANDLE_FDK_BITBUF hBitBuf) { hBitBuf->ValidBits = 0; hBitBuf->ReadOffset = 0; hBitBuf->WriteOffset = 0; hBitBuf->BitNdx = 0; } #ifndef FUNCTION_FDK_get INT FDK_get(HANDLE_FDK_BITBUF hBitBuf, const UINT numberOfBits) { UINT byteOffset = hBitBuf->BitNdx >> 3; UINT bitOffset = hBitBuf->BitNdx & 0x07; hBitBuf->BitNdx = (hBitBuf->BitNdx + numberOfBits) & (hBitBuf->bufBits - 1); hBitBuf->ValidBits -= numberOfBits; UINT byteMask = hBitBuf->bufSize - 1; UINT tx = (hBitBuf->Buffer[byteOffset & byteMask] << 24) | (hBitBuf->Buffer[(byteOffset + 1) & byteMask] << 16) | (hBitBuf->Buffer[(byteOffset + 2) & byteMask] << 8) | hBitBuf->Buffer[(byteOffset + 3) & byteMask]; if (bitOffset) { tx <<= bitOffset; tx |= hBitBuf->Buffer[(byteOffset + 4) & byteMask] >> (8 - bitOffset); } return (tx >> (32 - numberOfBits)); } #endif /* #ifndef FUNCTION_FDK_get */ #ifndef FUNCTION_FDK_get32 INT FDK_get32(HANDLE_FDK_BITBUF hBitBuf) { UINT BitNdx = hBitBuf->BitNdx + 32; hBitBuf->BitNdx = BitNdx & (hBitBuf->bufBits - 1); hBitBuf->ValidBits = (UINT)((INT)hBitBuf->ValidBits - (INT)32); UINT byteOffset = (BitNdx - 1) >> 3; if (BitNdx <= hBitBuf->bufBits) { UINT cache = (hBitBuf->Buffer[(byteOffset - 3)] << 24) | (hBitBuf->Buffer[(byteOffset - 2)] << 16) | (hBitBuf->Buffer[(byteOffset - 1)] << 8) | hBitBuf->Buffer[(byteOffset - 0)]; if ((BitNdx = (BitNdx & 7)) != 0) { cache = (cache >> (8 - BitNdx)) | ((UINT)hBitBuf->Buffer[byteOffset - 4] << (24 + BitNdx)); } return (cache); } else { UINT byte_mask = hBitBuf->bufSize - 1; UINT cache = (hBitBuf->Buffer[(byteOffset - 3) & byte_mask] << 24) | (hBitBuf->Buffer[(byteOffset - 2) & byte_mask] << 16) | (hBitBuf->Buffer[(byteOffset - 1) & byte_mask] << 8) | hBitBuf->Buffer[(byteOffset - 0) & byte_mask]; if ((BitNdx = (BitNdx & 7)) != 0) { cache = (cache >> (8 - BitNdx)) | ((UINT)hBitBuf->Buffer[(byteOffset - 4) & byte_mask] << (24 + BitNdx)); } return (cache); } } #endif INT FDK_getBwd(HANDLE_FDK_BITBUF hBitBuf, const UINT numberOfBits) { UINT byteOffset = hBitBuf->BitNdx >> 3; UINT bitOffset = hBitBuf->BitNdx & 0x07; UINT byteMask = hBitBuf->bufSize - 1; int i; hBitBuf->BitNdx = (hBitBuf->BitNdx - numberOfBits) & (hBitBuf->bufBits - 1); hBitBuf->ValidBits += numberOfBits; UINT tx = hBitBuf->Buffer[(byteOffset - 3) & byteMask] << 24 | hBitBuf->Buffer[(byteOffset - 2) & byteMask] << 16 | hBitBuf->Buffer[(byteOffset - 1) & byteMask] << 8 | hBitBuf->Buffer[byteOffset & byteMask]; UINT txa = 0x0; tx >>= (8 - bitOffset); if (bitOffset && numberOfBits > 24) { tx |= hBitBuf->Buffer[(byteOffset - 4) & byteMask] << (24 + bitOffset); } /* in place turn around */ for (i = 0; i < 16; i++) { UINT bitMaskR = 0x00000001 << i; UINT bitMaskL = 0x80000000 >> i; txa |= (tx & bitMaskR) << (31 - (i << 1)); txa |= (tx & bitMaskL) >> (31 - (i << 1)); } return (txa >> (32 - numberOfBits)); } void FDK_put(HANDLE_FDK_BITBUF hBitBuf, UINT value, const UINT numberOfBits) { if (numberOfBits != 0) { UINT byteOffset0 = hBitBuf->BitNdx >> 3; UINT bitOffset = hBitBuf->BitNdx & 0x7; hBitBuf->BitNdx = (hBitBuf->BitNdx + numberOfBits) & (hBitBuf->bufBits - 1); hBitBuf->ValidBits += numberOfBits; UINT byteMask = hBitBuf->bufSize - 1; UINT byteOffset1 = (byteOffset0 + 1) & byteMask; UINT byteOffset2 = (byteOffset0 + 2) & byteMask; UINT byteOffset3 = (byteOffset0 + 3) & byteMask; // Create tmp containing free bits at the left border followed by bits to // write, LSB's are cleared, if available Create mask to apply upon all // buffer bytes UINT tmp = (value << (32 - numberOfBits)) >> bitOffset; UINT mask = ~((BitMask[numberOfBits] << (32 - numberOfBits)) >> bitOffset); // read all 4 bytes from buffer and create a 32-bit cache UINT cache = (((UINT)hBitBuf->Buffer[byteOffset0]) << 24) | (((UINT)hBitBuf->Buffer[byteOffset1]) << 16) | (((UINT)hBitBuf->Buffer[byteOffset2]) << 8) | (((UINT)hBitBuf->Buffer[byteOffset3]) << 0); cache = (cache & mask) | tmp; hBitBuf->Buffer[byteOffset0] = (UCHAR)(cache >> 24); hBitBuf->Buffer[byteOffset1] = (UCHAR)(cache >> 16); hBitBuf->Buffer[byteOffset2] = (UCHAR)(cache >> 8); hBitBuf->Buffer[byteOffset3] = (UCHAR)(cache >> 0); if ((bitOffset + numberOfBits) > 32) { UINT byteOffset4 = (byteOffset0 + 4) & byteMask; // remaining bits: in range 1..7 // replace MSBits of next byte in buffer by LSBits of "value" int bits = (bitOffset + numberOfBits) & 7; cache = (UINT)hBitBuf->Buffer[byteOffset4] & (~(BitMask[bits] << (8 - bits))); cache |= value << (8 - bits); hBitBuf->Buffer[byteOffset4] = (UCHAR)cache; } } } void FDK_putBwd(HANDLE_FDK_BITBUF hBitBuf, UINT value, const UINT numberOfBits) { UINT byteOffset = hBitBuf->BitNdx >> 3; UINT bitOffset = 7 - (hBitBuf->BitNdx & 0x07); UINT byteMask = hBitBuf->bufSize - 1; UINT mask = ~(BitMask[numberOfBits] << bitOffset); UINT tmp = 0x0000; int i; hBitBuf->BitNdx = (hBitBuf->BitNdx - numberOfBits) & (hBitBuf->bufBits - 1); hBitBuf->ValidBits -= numberOfBits; /* in place turn around */ for (i = 0; i < 16; i++) { UINT bitMaskR = 0x00000001 << i; UINT bitMaskL = 0x80000000 >> i; tmp |= (value & bitMaskR) << (31 - (i << 1)); tmp |= (value & bitMaskL) >> (31 - (i << 1)); } value = tmp; tmp = value >> (32 - numberOfBits) << bitOffset; hBitBuf->Buffer[byteOffset & byteMask] = (hBitBuf->Buffer[byteOffset & byteMask] & (mask)) | (UCHAR)(tmp); hBitBuf->Buffer[(byteOffset - 1) & byteMask] = (hBitBuf->Buffer[(byteOffset - 1) & byteMask] & (mask >> 8)) | (UCHAR)(tmp >> 8); hBitBuf->Buffer[(byteOffset - 2) & byteMask] = (hBitBuf->Buffer[(byteOffset - 2) & byteMask] & (mask >> 16)) | (UCHAR)(tmp >> 16); hBitBuf->Buffer[(byteOffset - 3) & byteMask] = (hBitBuf->Buffer[(byteOffset - 3) & byteMask] & (mask >> 24)) | (UCHAR)(tmp >> 24); if ((bitOffset + numberOfBits) > 32) { hBitBuf->Buffer[(byteOffset - 4) & byteMask] = (UCHAR)(value >> (64 - numberOfBits - bitOffset)) | (hBitBuf->Buffer[(byteOffset - 4) & byteMask] & ~(BitMask[bitOffset] >> (32 - numberOfBits))); } } #ifndef FUNCTION_FDK_pushBack void FDK_pushBack(HANDLE_FDK_BITBUF hBitBuf, const UINT numberOfBits, UCHAR config) { hBitBuf->ValidBits = (config == 0) ? (UINT)((INT)hBitBuf->ValidBits + (INT)numberOfBits) : ((UINT)((INT)hBitBuf->ValidBits - (INT)numberOfBits)); hBitBuf->BitNdx = ((UINT)((INT)hBitBuf->BitNdx - (INT)numberOfBits)) & (hBitBuf->bufBits - 1); } #endif void FDK_pushForward(HANDLE_FDK_BITBUF hBitBuf, const UINT numberOfBits, UCHAR config) { hBitBuf->ValidBits = (config == 0) ? ((UINT)((INT)hBitBuf->ValidBits - (INT)numberOfBits)) : (UINT)((INT)hBitBuf->ValidBits + (INT)numberOfBits); hBitBuf->BitNdx = (UINT)((INT)hBitBuf->BitNdx + (INT)numberOfBits) & (hBitBuf->bufBits - 1); } #ifndef FUNCTION_FDK_getValidBits UINT FDK_getValidBits(HANDLE_FDK_BITBUF hBitBuf) { return hBitBuf->ValidBits; } #endif /* #ifndef FUNCTION_FDK_getValidBits */ INT FDK_getFreeBits(HANDLE_FDK_BITBUF hBitBuf) { return (hBitBuf->bufBits - hBitBuf->ValidBits); } void FDK_Feed(HANDLE_FDK_BITBUF hBitBuf, const UCHAR *RESTRICT inputBuffer, const UINT bufferSize, UINT *bytesValid) { inputBuffer = &inputBuffer[bufferSize - *bytesValid]; UINT bTotal = 0; UINT bToRead = fMin(hBitBuf->bufBits, (UINT)fMax(0, ((INT)hBitBuf->bufBits - (INT)hBitBuf->ValidBits))) >> 3; UINT noOfBytes = fMin(bToRead, *bytesValid); //(bToRead < *bytesValid) ? bToRead : *bytesValid ; while (noOfBytes > 0) { /* split read to buffer size */ bToRead = hBitBuf->bufSize - hBitBuf->ReadOffset; bToRead = fMin(bToRead, noOfBytes); //(bToRead < noOfBytes) ? bToRead : noOfBytes ; /* copy 'bToRead' bytes from 'ptr' to inputbuffer */ FDKmemcpy(&hBitBuf->Buffer[hBitBuf->ReadOffset], inputBuffer, bToRead * sizeof(UCHAR)); /* add noOfBits to number of valid bits in buffer */ hBitBuf->ValidBits = (UINT)((INT)hBitBuf->ValidBits + (INT)(bToRead << 3)); bTotal += bToRead; inputBuffer += bToRead; hBitBuf->ReadOffset = (hBitBuf->ReadOffset + bToRead) & (hBitBuf->bufSize - 1); noOfBytes -= bToRead; } *bytesValid -= bTotal; } void CopyAlignedBlock(HANDLE_FDK_BITBUF h_BitBufSrc, UCHAR *RESTRICT dstBuffer, UINT bToRead) { UINT byteOffset = h_BitBufSrc->BitNdx >> 3; const UINT byteMask = h_BitBufSrc->bufSize - 1; UCHAR *RESTRICT pBBB = h_BitBufSrc->Buffer; for (UINT i = 0; i < bToRead; i++) { dstBuffer[i] = pBBB[(byteOffset + i) & byteMask]; } bToRead <<= 3; h_BitBufSrc->BitNdx = (h_BitBufSrc->BitNdx + bToRead) & (h_BitBufSrc->bufBits - 1); h_BitBufSrc->ValidBits -= bToRead; } void FDK_Copy(HANDLE_FDK_BITBUF h_BitBufDst, HANDLE_FDK_BITBUF h_BitBufSrc, UINT *bytesValid) { INT bTotal = 0; /* limit noOfBytes to valid bytes in src buffer and available bytes in dst * buffer */ UINT bToRead = h_BitBufSrc->ValidBits >> 3; UINT noOfBytes = fMin(bToRead, *bytesValid); //(*bytesValid < bToRead) ? *bytesValid : bToRead ; bToRead = FDK_getFreeBits(h_BitBufDst); noOfBytes = fMin(bToRead, noOfBytes); //(bToRead < noOfBytes) ? bToRead : noOfBytes; while (noOfBytes > 0) { /* Split Read to buffer size */ bToRead = h_BitBufDst->bufSize - h_BitBufDst->ReadOffset; bToRead = fMin(noOfBytes, bToRead); //(noOfBytes < bToRead) ? noOfBytes : bToRead ; /* copy 'bToRead' bytes from buffer to buffer */ if (!(h_BitBufSrc->BitNdx & 0x07)) { CopyAlignedBlock(h_BitBufSrc, h_BitBufDst->Buffer + h_BitBufDst->ReadOffset, bToRead); } else { for (UINT i = 0; i < bToRead; i++) { h_BitBufDst->Buffer[h_BitBufDst->ReadOffset + i] = (UCHAR)FDK_get(h_BitBufSrc, 8); } } /* add noOfBits to number of valid bits in buffer */ h_BitBufDst->ValidBits += bToRead << 3; bTotal += bToRead; h_BitBufDst->ReadOffset = (h_BitBufDst->ReadOffset + bToRead) & (h_BitBufDst->bufSize - 1); noOfBytes -= bToRead; } *bytesValid -= bTotal; } void FDK_Fetch(HANDLE_FDK_BITBUF hBitBuf, UCHAR *outBuf, UINT *writeBytes) { UCHAR *RESTRICT outputBuffer = outBuf; UINT bTotal = 0; UINT bToWrite = (hBitBuf->ValidBits) >> 3; UINT noOfBytes = fMin(bToWrite, *writeBytes); //(bToWrite < *writeBytes) ? bToWrite : *writeBytes ; while (noOfBytes > 0) { /* split write to buffer size */ bToWrite = hBitBuf->bufSize - hBitBuf->WriteOffset; bToWrite = fMin( bToWrite, noOfBytes); //(bToWrite < noOfBytes) ? bToWrite : noOfBytes ; /* copy 'bToWrite' bytes from bitbuffer to outputbuffer */ FDKmemcpy(outputBuffer, &hBitBuf->Buffer[hBitBuf->WriteOffset], bToWrite * sizeof(UCHAR)); /* sub noOfBits from number of valid bits in buffer */ hBitBuf->ValidBits -= bToWrite << 3; bTotal += bToWrite; outputBuffer += bToWrite; hBitBuf->WriteOffset = (hBitBuf->WriteOffset + bToWrite) & (hBitBuf->bufSize - 1); noOfBytes -= bToWrite; } *writeBytes = bTotal; } fdk-aac-2.0.2/libFDK/src/FDK_core.cpp000066400000000000000000000133411404225310700170030ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Manuel Jander Description: FDK tools versioning support *******************************************************************************/ #include "FDK_core.h" /* FDK tools library info */ #define FDK_TOOLS_LIB_VL0 3 #define FDK_TOOLS_LIB_VL1 1 #define FDK_TOOLS_LIB_VL2 0 #define FDK_TOOLS_LIB_TITLE "FDK Tools" #ifdef SUPPRESS_BUILD_DATE_INFO #define FDK_TOOLS_LIB_BUILD_DATE "" #define FDK_TOOLS_LIB_BUILD_TIME "" #else #define FDK_TOOLS_LIB_BUILD_DATE __DATE__ #define FDK_TOOLS_LIB_BUILD_TIME __TIME__ #endif int FDK_toolsGetLibInfo(LIB_INFO *info) { UINT v; int i; if (info == NULL) { return -1; } /* search for next free tab */ i = FDKlibInfo_lookup(info, FDK_TOOLS); if (i < 0) return -1; info += i; v = LIB_VERSION(FDK_TOOLS_LIB_VL0, FDK_TOOLS_LIB_VL1, FDK_TOOLS_LIB_VL2); FDKsprintf(info->versionStr, "%d.%d.%d", ((v >> 24) & 0xff), ((v >> 16) & 0xff), ((v >> 8) & 0xff)); info->module_id = FDK_TOOLS; info->version = v; info->build_date = FDK_TOOLS_LIB_BUILD_DATE; info->build_time = FDK_TOOLS_LIB_BUILD_TIME; info->title = FDK_TOOLS_LIB_TITLE; info->flags = 1; return 0; } fdk-aac-2.0.2/libFDK/src/FDK_crc.cpp000066400000000000000000000471421404225310700166300ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Description: CRC calculation *******************************************************************************/ #include "FDK_crc.h" /*---------------- constants -----------------------*/ /** * \brief This table defines precalculated lookup tables for crc polynom x^16 * + x^15 + x^2 + x^0. */ static const USHORT crcLookup_16_15_2_0[256] = { 0x0000, 0x8005, 0x800f, 0x000a, 0x801b, 0x001e, 0x0014, 0x8011, 0x8033, 0x0036, 0x003c, 0x8039, 0x0028, 0x802d, 0x8027, 0x0022, 0x8063, 0x0066, 0x006c, 0x8069, 0x0078, 0x807d, 0x8077, 0x0072, 0x0050, 0x8055, 0x805f, 0x005a, 0x804b, 0x004e, 0x0044, 0x8041, 0x80c3, 0x00c6, 0x00cc, 0x80c9, 0x00d8, 0x80dd, 0x80d7, 0x00d2, 0x00f0, 0x80f5, 0x80ff, 0x00fa, 0x80eb, 0x00ee, 0x00e4, 0x80e1, 0x00a0, 0x80a5, 0x80af, 0x00aa, 0x80bb, 0x00be, 0x00b4, 0x80b1, 0x8093, 0x0096, 0x009c, 0x8099, 0x0088, 0x808d, 0x8087, 0x0082, 0x8183, 0x0186, 0x018c, 0x8189, 0x0198, 0x819d, 0x8197, 0x0192, 0x01b0, 0x81b5, 0x81bf, 0x01ba, 0x81ab, 0x01ae, 0x01a4, 0x81a1, 0x01e0, 0x81e5, 0x81ef, 0x01ea, 0x81fb, 0x01fe, 0x01f4, 0x81f1, 0x81d3, 0x01d6, 0x01dc, 0x81d9, 0x01c8, 0x81cd, 0x81c7, 0x01c2, 0x0140, 0x8145, 0x814f, 0x014a, 0x815b, 0x015e, 0x0154, 0x8151, 0x8173, 0x0176, 0x017c, 0x8179, 0x0168, 0x816d, 0x8167, 0x0162, 0x8123, 0x0126, 0x012c, 0x8129, 0x0138, 0x813d, 0x8137, 0x0132, 0x0110, 0x8115, 0x811f, 0x011a, 0x810b, 0x010e, 0x0104, 0x8101, 0x8303, 0x0306, 0x030c, 0x8309, 0x0318, 0x831d, 0x8317, 0x0312, 0x0330, 0x8335, 0x833f, 0x033a, 0x832b, 0x032e, 0x0324, 0x8321, 0x0360, 0x8365, 0x836f, 0x036a, 0x837b, 0x037e, 0x0374, 0x8371, 0x8353, 0x0356, 0x035c, 0x8359, 0x0348, 0x834d, 0x8347, 0x0342, 0x03c0, 0x83c5, 0x83cf, 0x03ca, 0x83db, 0x03de, 0x03d4, 0x83d1, 0x83f3, 0x03f6, 0x03fc, 0x83f9, 0x03e8, 0x83ed, 0x83e7, 0x03e2, 0x83a3, 0x03a6, 0x03ac, 0x83a9, 0x03b8, 0x83bd, 0x83b7, 0x03b2, 0x0390, 0x8395, 0x839f, 0x039a, 0x838b, 0x038e, 0x0384, 0x8381, 0x0280, 0x8285, 0x828f, 0x028a, 0x829b, 0x029e, 0x0294, 0x8291, 0x82b3, 0x02b6, 0x02bc, 0x82b9, 0x02a8, 0x82ad, 0x82a7, 0x02a2, 0x82e3, 0x02e6, 0x02ec, 0x82e9, 0x02f8, 0x82fd, 0x82f7, 0x02f2, 0x02d0, 0x82d5, 0x82df, 0x02da, 0x82cb, 0x02ce, 0x02c4, 0x82c1, 0x8243, 0x0246, 0x024c, 0x8249, 0x0258, 0x825d, 0x8257, 0x0252, 0x0270, 0x8275, 0x827f, 0x027a, 0x826b, 0x026e, 0x0264, 0x8261, 0x0220, 0x8225, 0x822f, 0x022a, 0x823b, 0x023e, 0x0234, 0x8231, 0x8213, 0x0216, 0x021c, 0x8219, 0x0208, 0x820d, 0x8207, 0x0202}; /** * \brief This table defines precalculated lookup tables for crc polynom x^16 * + x^12 + x^5 + x^0. */ static const USHORT crcLookup_16_12_5_0[256] = { 0x0000, 0x1021, 0x2042, 0x3063, 0x4084, 0x50a5, 0x60c6, 0x70e7, 0x8108, 0x9129, 0xa14a, 0xb16b, 0xc18c, 0xd1ad, 0xe1ce, 0xf1ef, 0x1231, 0x0210, 0x3273, 0x2252, 0x52b5, 0x4294, 0x72f7, 0x62d6, 0x9339, 0x8318, 0xb37b, 0xa35a, 0xd3bd, 0xc39c, 0xf3ff, 0xe3de, 0x2462, 0x3443, 0x0420, 0x1401, 0x64e6, 0x74c7, 0x44a4, 0x5485, 0xa56a, 0xb54b, 0x8528, 0x9509, 0xe5ee, 0xf5cf, 0xc5ac, 0xd58d, 0x3653, 0x2672, 0x1611, 0x0630, 0x76d7, 0x66f6, 0x5695, 0x46b4, 0xb75b, 0xa77a, 0x9719, 0x8738, 0xf7df, 0xe7fe, 0xd79d, 0xc7bc, 0x48c4, 0x58e5, 0x6886, 0x78a7, 0x0840, 0x1861, 0x2802, 0x3823, 0xc9cc, 0xd9ed, 0xe98e, 0xf9af, 0x8948, 0x9969, 0xa90a, 0xb92b, 0x5af5, 0x4ad4, 0x7ab7, 0x6a96, 0x1a71, 0x0a50, 0x3a33, 0x2a12, 0xdbfd, 0xcbdc, 0xfbbf, 0xeb9e, 0x9b79, 0x8b58, 0xbb3b, 0xab1a, 0x6ca6, 0x7c87, 0x4ce4, 0x5cc5, 0x2c22, 0x3c03, 0x0c60, 0x1c41, 0xedae, 0xfd8f, 0xcdec, 0xddcd, 0xad2a, 0xbd0b, 0x8d68, 0x9d49, 0x7e97, 0x6eb6, 0x5ed5, 0x4ef4, 0x3e13, 0x2e32, 0x1e51, 0x0e70, 0xff9f, 0xefbe, 0xdfdd, 0xcffc, 0xbf1b, 0xaf3a, 0x9f59, 0x8f78, 0x9188, 0x81a9, 0xb1ca, 0xa1eb, 0xd10c, 0xc12d, 0xf14e, 0xe16f, 0x1080, 0x00a1, 0x30c2, 0x20e3, 0x5004, 0x4025, 0x7046, 0x6067, 0x83b9, 0x9398, 0xa3fb, 0xb3da, 0xc33d, 0xd31c, 0xe37f, 0xf35e, 0x02b1, 0x1290, 0x22f3, 0x32d2, 0x4235, 0x5214, 0x6277, 0x7256, 0xb5ea, 0xa5cb, 0x95a8, 0x8589, 0xf56e, 0xe54f, 0xd52c, 0xc50d, 0x34e2, 0x24c3, 0x14a0, 0x0481, 0x7466, 0x6447, 0x5424, 0x4405, 0xa7db, 0xb7fa, 0x8799, 0x97b8, 0xe75f, 0xf77e, 0xc71d, 0xd73c, 0x26d3, 0x36f2, 0x0691, 0x16b0, 0x6657, 0x7676, 0x4615, 0x5634, 0xd94c, 0xc96d, 0xf90e, 0xe92f, 0x99c8, 0x89e9, 0xb98a, 0xa9ab, 0x5844, 0x4865, 0x7806, 0x6827, 0x18c0, 0x08e1, 0x3882, 0x28a3, 0xcb7d, 0xdb5c, 0xeb3f, 0xfb1e, 0x8bf9, 0x9bd8, 0xabbb, 0xbb9a, 0x4a75, 0x5a54, 0x6a37, 0x7a16, 0x0af1, 0x1ad0, 0x2ab3, 0x3a92, 0xfd2e, 0xed0f, 0xdd6c, 0xcd4d, 0xbdaa, 0xad8b, 0x9de8, 0x8dc9, 0x7c26, 0x6c07, 0x5c64, 0x4c45, 0x3ca2, 0x2c83, 0x1ce0, 0x0cc1, 0xef1f, 0xff3e, 0xcf5d, 0xdf7c, 0xaf9b, 0xbfba, 0x8fd9, 0x9ff8, 0x6e17, 0x7e36, 0x4e55, 0x5e74, 0x2e93, 0x3eb2, 0x0ed1, 0x1ef0}; /** * \brief This table defines precalculated lookup tables for crc polynom x^16 * + x^15 + x^5 + x^0. */ static const USHORT crcLookup_16_15_5_0[256] = { 0x0000, 0x8021, 0x8063, 0x0042, 0x80e7, 0x00c6, 0x0084, 0x80a5, 0x81ef, 0x01ce, 0x018c, 0x81ad, 0x0108, 0x8129, 0x816b, 0x014a, 0x83ff, 0x03de, 0x039c, 0x83bd, 0x0318, 0x8339, 0x837b, 0x035a, 0x0210, 0x8231, 0x8273, 0x0252, 0x82f7, 0x02d6, 0x0294, 0x82b5, 0x87df, 0x07fe, 0x07bc, 0x879d, 0x0738, 0x8719, 0x875b, 0x077a, 0x0630, 0x8611, 0x8653, 0x0672, 0x86d7, 0x06f6, 0x06b4, 0x8695, 0x0420, 0x8401, 0x8443, 0x0462, 0x84c7, 0x04e6, 0x04a4, 0x8485, 0x85cf, 0x05ee, 0x05ac, 0x858d, 0x0528, 0x8509, 0x854b, 0x056a, 0x8f9f, 0x0fbe, 0x0ffc, 0x8fdd, 0x0f78, 0x8f59, 0x8f1b, 0x0f3a, 0x0e70, 0x8e51, 0x8e13, 0x0e32, 0x8e97, 0x0eb6, 0x0ef4, 0x8ed5, 0x0c60, 0x8c41, 0x8c03, 0x0c22, 0x8c87, 0x0ca6, 0x0ce4, 0x8cc5, 0x8d8f, 0x0dae, 0x0dec, 0x8dcd, 0x0d68, 0x8d49, 0x8d0b, 0x0d2a, 0x0840, 0x8861, 0x8823, 0x0802, 0x88a7, 0x0886, 0x08c4, 0x88e5, 0x89af, 0x098e, 0x09cc, 0x89ed, 0x0948, 0x8969, 0x892b, 0x090a, 0x8bbf, 0x0b9e, 0x0bdc, 0x8bfd, 0x0b58, 0x8b79, 0x8b3b, 0x0b1a, 0x0a50, 0x8a71, 0x8a33, 0x0a12, 0x8ab7, 0x0a96, 0x0ad4, 0x8af5, 0x9f1f, 0x1f3e, 0x1f7c, 0x9f5d, 0x1ff8, 0x9fd9, 0x9f9b, 0x1fba, 0x1ef0, 0x9ed1, 0x9e93, 0x1eb2, 0x9e17, 0x1e36, 0x1e74, 0x9e55, 0x1ce0, 0x9cc1, 0x9c83, 0x1ca2, 0x9c07, 0x1c26, 0x1c64, 0x9c45, 0x9d0f, 0x1d2e, 0x1d6c, 0x9d4d, 0x1de8, 0x9dc9, 0x9d8b, 0x1daa, 0x18c0, 0x98e1, 0x98a3, 0x1882, 0x9827, 0x1806, 0x1844, 0x9865, 0x992f, 0x190e, 0x194c, 0x996d, 0x19c8, 0x99e9, 0x99ab, 0x198a, 0x9b3f, 0x1b1e, 0x1b5c, 0x9b7d, 0x1bd8, 0x9bf9, 0x9bbb, 0x1b9a, 0x1ad0, 0x9af1, 0x9ab3, 0x1a92, 0x9a37, 0x1a16, 0x1a54, 0x9a75, 0x1080, 0x90a1, 0x90e3, 0x10c2, 0x9067, 0x1046, 0x1004, 0x9025, 0x916f, 0x114e, 0x110c, 0x912d, 0x1188, 0x91a9, 0x91eb, 0x11ca, 0x937f, 0x135e, 0x131c, 0x933d, 0x1398, 0x93b9, 0x93fb, 0x13da, 0x1290, 0x92b1, 0x92f3, 0x12d2, 0x9277, 0x1256, 0x1214, 0x9235, 0x975f, 0x177e, 0x173c, 0x971d, 0x17b8, 0x9799, 0x97db, 0x17fa, 0x16b0, 0x9691, 0x96d3, 0x16f2, 0x9657, 0x1676, 0x1634, 0x9615, 0x14a0, 0x9481, 0x94c3, 0x14e2, 0x9447, 0x1466, 0x1424, 0x9405, 0x954f, 0x156e, 0x152c, 0x950d, 0x15a8, 0x9589, 0x95cb, 0x15ea, }; /*--------------- function declarations --------------------*/ static inline INT calcCrc_Bits(USHORT *const pCrc, USHORT crcMask, USHORT crcPoly, HANDLE_FDK_BITSTREAM hBs, INT nBits); static inline INT calcCrc_Bytes(USHORT *const pCrc, const USHORT *pCrcLookup, HANDLE_FDK_BITSTREAM hBs, INT nBytes); static void crcCalc(HANDLE_FDK_CRCINFO hCrcInfo, HANDLE_FDK_BITSTREAM hBs, const INT reg); /*------------- function definitions ----------------*/ void FDKcrcInit(HANDLE_FDK_CRCINFO hCrcInfo, const UINT crcPoly, const UINT crcStartValue, const UINT crcLen) { /* crc polynom example: x^16 + x^15 + x^5 + x^0 (1) 1000 0000 0010 0001 -> 0x8021 x^16 + x^15 + x^2 + x^0 (1) 1000 0000 0000 0101 -> 0x8005 x^16 + x^12 + x^5 + x^0 (1) 0001 0000 0010 0001 -> 0x1021 x^8 + x^4 + x^3 + x^2 + x^0 (1) 0001 1101 -> 0x001d */ hCrcInfo->crcLen = crcLen; hCrcInfo->crcPoly = crcPoly; hCrcInfo->startValue = crcStartValue; hCrcInfo->crcMask = (crcLen) ? (1 << (crcLen - 1)) : 0; FDKcrcReset(hCrcInfo); hCrcInfo->pCrcLookup = 0; /* Preset 0 for "crcLen" != 16 or unknown 16-bit polynoms "crcPoly" */ if (hCrcInfo->crcLen == 16) { switch (crcPoly) { case 0x8021: hCrcInfo->pCrcLookup = crcLookup_16_15_5_0; break; case 0x8005: hCrcInfo->pCrcLookup = crcLookup_16_15_2_0; break; case 0x1021: hCrcInfo->pCrcLookup = crcLookup_16_12_5_0; break; case 0x001d: default: /* no lookup table */ break; } } } void FDKcrcReset(HANDLE_FDK_CRCINFO hCrcInfo) { int i; hCrcInfo->crcValue = hCrcInfo->startValue; for (i = 0; i < MAX_CRC_REGS; i++) { hCrcInfo->crcRegData[i].isActive = 0; } hCrcInfo->regStart = 0; hCrcInfo->regStop = 0; } INT FDKcrcStartReg(HANDLE_FDK_CRCINFO hCrcInfo, const HANDLE_FDK_BITSTREAM hBs, const INT mBits) { int reg = hCrcInfo->regStart; FDK_ASSERT(hCrcInfo->crcRegData[reg].isActive == 0); hCrcInfo->crcRegData[reg].isActive = 1; hCrcInfo->crcRegData[reg].maxBits = mBits; hCrcInfo->crcRegData[reg].validBits = (INT)FDKgetValidBits(hBs); hCrcInfo->crcRegData[reg].bitBufCntBits = 0; hCrcInfo->regStart = (hCrcInfo->regStart + 1) % MAX_CRC_REGS; return (reg); } INT FDKcrcEndReg(HANDLE_FDK_CRCINFO hCrcInfo, const HANDLE_FDK_BITSTREAM hBs, const INT reg) { FDK_ASSERT((reg == (INT)hCrcInfo->regStop) && (hCrcInfo->crcRegData[reg].isActive == 1)); if (hBs->ConfigCache == BS_WRITER) { hCrcInfo->crcRegData[reg].bitBufCntBits = (INT)FDKgetValidBits(hBs) - hCrcInfo->crcRegData[reg].validBits; } else { hCrcInfo->crcRegData[reg].bitBufCntBits = hCrcInfo->crcRegData[reg].validBits - (INT)FDKgetValidBits(hBs); } if (hCrcInfo->crcRegData[reg].maxBits == 0) { hCrcInfo->crcRegData[reg].maxBits = hCrcInfo->crcRegData[reg].bitBufCntBits; } crcCalc(hCrcInfo, hBs, reg); hCrcInfo->crcRegData[reg].isActive = 0; hCrcInfo->regStop = (hCrcInfo->regStop + 1) % MAX_CRC_REGS; return 0; } USHORT FDKcrcGetCRC(const HANDLE_FDK_CRCINFO hCrcInfo) { return (hCrcInfo->crcValue & (((hCrcInfo->crcMask - 1) << 1) + 1)); } /** * \brief Calculate crc bits. * * Calculate crc starting at current bitstream postion over nBits. * * \param pCrc Pointer to an outlying allocated crc info * structure. * \param crcMask CrcMask in use. * \param crcPoly Crc polynom in use. * \param hBs Handle to current bit buffer structure. * \param nBits Number of processing bits. * * \return Number of processed bits. */ static inline INT calcCrc_Bits(USHORT *const pCrc, USHORT crcMask, USHORT crcPoly, HANDLE_FDK_BITSTREAM hBs, INT nBits) { int i; USHORT crc = *pCrc; /* get crc value */ if (hBs != NULL) { for (i = 0; (i < nBits); i++) { USHORT tmp = FDKreadBit(hBs); // process single bit tmp ^= ((crc & crcMask) ? 1 : 0); if (tmp != 0) tmp = crcPoly; crc <<= 1; crc ^= tmp; } } else { for (i = 0; (i < nBits); i++) { USHORT tmp = (crc & crcMask) ? crcPoly : 0; // process single bit crc <<= 1; crc ^= tmp; } } *pCrc = crc; /* update crc value */ return nBits; } /** * \brief Calculate crc bytes. * * Calculate crc starting at current bitstream postion over nBytes. * * \param pCrc Pointer to an outlying allocated crc info * structure. * \param pCrcLookup Pointer to lookup table used for fast crc * calculation. * \param hBs Handle to current bit buffer structure. * \param nBits Number of processing bytes. * * \return Number of processed bits. */ static inline INT calcCrc_Bytes(USHORT *const pCrc, const USHORT *pCrcLookup, HANDLE_FDK_BITSTREAM hBs, INT nBytes) { int i; USHORT crc = *pCrc; /* get crc value */ if (hBs != NULL) { ULONG data; INT bits; for (i = 0; i < (nBytes >> 2); i++) { data = (ULONG)FDKreadBits(hBs, 32); crc = (crc << 8) ^ pCrcLookup[((crc >> 8) ^ ((USHORT)(data >> 24))) & 0xFF]; crc = (crc << 8) ^ pCrcLookup[((crc >> 8) ^ ((USHORT)(data >> 16))) & 0xFF]; crc = (crc << 8) ^ pCrcLookup[((crc >> 8) ^ ((USHORT)(data >> 8))) & 0xFF]; crc = (crc << 8) ^ pCrcLookup[((crc >> 8) ^ ((USHORT)(data >> 0))) & 0xFF]; } bits = (nBytes & 3) << 3; if (bits > 0) { data = (ULONG)FDKreadBits(hBs, bits); for (bits -= 8; bits >= 0; bits -= 8) crc = (crc << 8) ^ pCrcLookup[((crc >> 8) ^ (USHORT)(data >> bits)) & 0xFF]; } } else { for (i = 0; i < nBytes; i++) { crc = (crc << 8) ^ pCrcLookup[(crc >> 8) & 0xFF]; } } *pCrc = crc; /* update crc value */ return (nBytes); } /** * \brief Calculate crc. * * Calculate crc. Lenght depends on mBits parameter in FDKcrcStartReg() * configuration. * * \param hCrcInfo Pointer to an outlying allocated crc info * structure. * \param hBs Pointer to current bit buffer structure. * \param reg Crc region ID. * * \return Number of processed bits. */ static void crcCalc(HANDLE_FDK_CRCINFO hCrcInfo, HANDLE_FDK_BITSTREAM hBs, const INT reg) { USHORT crc = hCrcInfo->crcValue; CCrcRegData *rD = &hCrcInfo->crcRegData[reg]; FDK_BITSTREAM bsReader; if (hBs->ConfigCache == BS_READER) { bsReader = *hBs; FDKpushBiDirectional(&bsReader, -(rD->validBits - (INT)FDKgetValidBits(&bsReader))); } else { FDKinitBitStream(&bsReader, hBs->hBitBuf.Buffer, hBs->hBitBuf.bufSize, hBs->hBitBuf.ValidBits, BS_READER); FDKpushBiDirectional(&bsReader, rD->validBits); } int bits, rBits; rBits = (rD->maxBits >= 0) ? rD->maxBits : -rD->maxBits; /* ramaining bits */ if ((rD->maxBits > 0) && ((rD->bitBufCntBits >> 3 << 3) < rBits)) { bits = rD->bitBufCntBits; } else { bits = rBits; } int words = bits >> 3; /* processing bytes */ int mBits = bits & 0x7; /* modulo bits */ if (hCrcInfo->pCrcLookup) { rBits -= (calcCrc_Bytes(&crc, hCrcInfo->pCrcLookup, &bsReader, words) << 3); } else { rBits -= calcCrc_Bits(&crc, hCrcInfo->crcMask, hCrcInfo->crcPoly, &bsReader, words << 3); } /* remaining valid bits*/ if (mBits != 0) { rBits -= calcCrc_Bits(&crc, hCrcInfo->crcMask, hCrcInfo->crcPoly, &bsReader, mBits); } if (rBits != 0) { /* zero bytes */ if ((hCrcInfo->pCrcLookup) && (rBits > 8)) { rBits -= (calcCrc_Bytes(&crc, hCrcInfo->pCrcLookup, NULL, rBits >> 3) << 3); } /* remaining zero bits */ if (rBits != 0) { calcCrc_Bits(&crc, hCrcInfo->crcMask, hCrcInfo->crcPoly, NULL, rBits); } } hCrcInfo->crcValue = crc; } fdk-aac-2.0.2/libFDK/src/FDK_decorrelate.cpp000066400000000000000000002055131404225310700203500ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Markus Lohwasser Description: FDK Tools Decorrelator *******************************************************************************/ #include "FDK_decorrelate.h" #define PC_NUM_BANDS (8) #define PC_NUM_HYB_BANDS (PC_NUM_BANDS - 3 + 10) #define DUCK_ALPHA (0.8f) #define DUCK_GAMMA (1.5f) #define ABS_THR (1e-9f * 32768 * 32768) #define ABS_THR_FDK ((FIXP_DBL)1) #define DECORR_ZERO_PADDING 0 #define DECORR_FILTER_ORDER_BAND_0_MPS (20) #define DECORR_FILTER_ORDER_BAND_1_MPS (15) #define DECORR_FILTER_ORDER_BAND_2_MPS (6) #define DECORR_FILTER_ORDER_BAND_3_MPS (3) #define DECORR_FILTER_ORDER_BAND_0_USAC (10) #define DECORR_FILTER_ORDER_BAND_1_USAC (8) #define DECORR_FILTER_ORDER_BAND_2_USAC (3) #define DECORR_FILTER_ORDER_BAND_3_USAC (2) #define DECORR_FILTER_ORDER_BAND_0_LD (0) #define DECORR_FILTER_ORDER_BAND_1_LD (DECORR_FILTER_ORDER_BAND_1_MPS) #define DECORR_FILTER_ORDER_BAND_2_LD (DECORR_FILTER_ORDER_BAND_2_MPS) #define DECORR_FILTER_ORDER_BAND_3_LD (DECORR_FILTER_ORDER_BAND_3_MPS) #define MAX_DECORR_SEED_MPS \ (5) /* 4 is worst case for 7272 mode for low power */ /* 5 is worst case for 7271 and 7272 mode for high quality */ #define MAX_DECORR_SEED_USAC (1) #define MAX_DECORR_SEED_LD (4) #define DECORR_FILTER_ORDER_PS (12) #define NUM_DECORR_CONFIGS \ (3) /* different configs defined by bsDecorrConfig bitstream field */ /* REV_bandOffset_... tables map (hybrid) bands to the corresponding reverb bands. Within each reverb band the same processing is applied. Instead of QMF split frequencies the corresponding hybrid band offsets are stored directly */ static const UCHAR REV_bandOffset_MPS_HQ[NUM_DECORR_CONFIGS][(4)] = { {8, 21, 30, 71}, {8, 56, 71, 71}, {0, 21, 71, 71}}; /* REV_bandOffset_USAC[] are equivalent to REV_bandOffset_MPS_HQ */ static const UCHAR REV_bandOffset_PS_HQ[(4)] = {30, 42, 71, 71}; static const UCHAR REV_bandOffset_PS_LP[(4)] = {14, 42, 71, 71}; static const UCHAR REV_bandOffset_LD[NUM_DECORR_CONFIGS][(4)] = { {0, 14, 23, 64}, {0, 49, 64, 64}, {0, 14, 64, 64}}; /* REV_delay_... tables define the number of delay elements within each reverb * band */ /* REV_filterOrder_... tables define the filter order within each reverb band */ static const UCHAR REV_delay_MPS[(4)] = {8, 7, 2, 1}; static const SCHAR REV_filterOrder_MPS[(4)] = { DECORR_FILTER_ORDER_BAND_0_MPS, DECORR_FILTER_ORDER_BAND_1_MPS, DECORR_FILTER_ORDER_BAND_2_MPS, DECORR_FILTER_ORDER_BAND_3_MPS}; static const UCHAR REV_delay_PS_HQ[(4)] = {2, 14, 1, 0}; static const UCHAR REV_delay_PS_LP[(4)] = {8, 14, 1, 0}; static const SCHAR REV_filterOrder_PS[(4)] = {DECORR_FILTER_ORDER_PS, -1, -1, -1}; static const UCHAR REV_delay_USAC[(4)] = {11, 10, 5, 2}; static const SCHAR REV_filterOrder_USAC[(4)] = { DECORR_FILTER_ORDER_BAND_0_USAC, DECORR_FILTER_ORDER_BAND_1_USAC, DECORR_FILTER_ORDER_BAND_2_USAC, DECORR_FILTER_ORDER_BAND_3_USAC}; /* REV_filtType_... tables define the type of processing (filtering with different properties or pure delay) done in each reverb band. This is mapped to specialized routines. */ static const REVBAND_FILT_TYPE REV_filtType_MPS[(4)] = { COMMON_REAL, COMMON_REAL, COMMON_REAL, COMMON_REAL}; static const REVBAND_FILT_TYPE REV_filtType_PS[(4)] = {INDEP_CPLX_PS, DELAY, DELAY, NOT_EXIST}; /* initialization values of ring buffer offsets for the 3 concatenated allpass * filters (PS type decorrelator). */ static const UCHAR stateBufferOffsetInit[(3)] = {0, 6, 14}; static const REVBAND_FILT_TYPE REV_filtType_LD[(4)] = { NOT_EXIST, COMMON_REAL, COMMON_REAL, COMMON_REAL}; /*** mapping of hybrid bands to processing (/parameter?) bands ***/ /* table for PS decorr running in legacy PS decoder. */ static const UCHAR kernels_20_to_71_PS[(71) + 1] = { 0, 0, 1, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 14, 15, 15, 15, 16, 16, 16, 16, 17, 17, 17, 17, 17, 18, 18, 18, 18, 18, 18, 18, 18, 18, 18, 18, 18, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19}; /*** mapping of processing (/parameter?) bands to hybrid bands ***/ /* table for PS decorr running in legacy PS decoder. */ static const UCHAR kernels_20_to_71_offset_PS[(20) + 1] = { 0, 2, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 18, 21, 25, 30, 42, 71}; static const UCHAR kernels_28_to_71[(71) + 1] = { 0, 0, 1, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 17, 18, 18, 19, 19, 20, 20, 21, 21, 21, 22, 22, 22, 23, 23, 23, 23, 24, 24, 24, 24, 24, 25, 25, 25, 25, 25, 25, 26, 26, 26, 26, 26, 26, 26, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27}; static const UCHAR kernels_28_to_71_offset[(28) + 1] = { 0, 2, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 21, 23, 25, 27, 30, 33, 37, 42, 48, 55, 71}; /* LD-MPS defined in SAOC standart (mapping qmf -> param bands)*/ static const UCHAR kernels_23_to_64[(64) + 1] = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 12, 13, 13, 14, 14, 15, 15, 16, 16, 16, 17, 17, 17, 18, 18, 18, 18, 19, 19, 19, 19, 19, 20, 20, 20, 20, 20, 20, 21, 21, 21, 21, 21, 21, 21, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, }; static const UCHAR kernels_23_to_64_offset[(23) + 1] = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 14, 16, 18, 20, 23, 26, 30, 35, 41, 48, 64}; static inline int SpatialDecGetProcessingBand(int hybridBand, const UCHAR *tab) { return tab[hybridBand]; } /* helper inline function */ static inline int SpatialDecGetQmfBand(int paramBand, const UCHAR *tab) { return (int)tab[paramBand]; } #define DUCKER_MAX_NRG_SCALE (24) #define DUCKER_HEADROOM_BITS (2) #define FILTER_SF (2) #ifdef ARCH_PREFER_MULT_32x32 #define FIXP_DUCK_GAIN FIXP_DBL #define FX_DBL2FX_DUCK_GAIN #define FL2FXCONST_DUCK FL2FXCONST_DBL #else #define FIXP_DUCK_GAIN FIXP_SGL #define FX_DBL2FX_DUCK_GAIN FX_DBL2FX_SGL #define FL2FXCONST_DUCK FL2FXCONST_SGL #endif #define PS_DUCK_PEAK_DECAY_FACTOR (0.765928338364649f) #define PS_DUCK_FILTER_COEFF (0.25f) #define DUCK_ALPHA_FDK FL2FXCONST_DUCK(DUCK_ALPHA) #define DUCK_ONE_MINUS_ALPHA_X4_FDK FL2FXCONST_DUCK(4.0f * (1.0f - DUCK_ALPHA)) #define DUCK_GAMMA_FDK FL2FXCONST_DUCK(DUCK_GAMMA / 2) #define PS_DUCK_PEAK_DECAY_FACTOR_FDK FL2FXCONST_DUCK(PS_DUCK_PEAK_DECAY_FACTOR) #define PS_DUCK_FILTER_COEFF_FDK FL2FXCONST_DUCK(PS_DUCK_FILTER_COEFF) RAM_ALIGN const FIXP_STP DecorrPsCoeffsCplx[][4] = { {STCP(0x5d6940eb, 0x5783153e), STCP(0xadcd41a8, 0x0e0373ed), STCP(0xbad41f3e, 0x14fba045), STCP(0xc1eb6694, 0x0883227d)}, {STCP(0x5d6940eb, 0xa87ceac2), STCP(0xadcd41a8, 0xf1fc8c13), STCP(0xbad41f3e, 0xeb045fbb), STCP(0xc1eb6694, 0xf77cdd83)}, {STCP(0xaec24162, 0x62e9d75b), STCP(0xb7169316, 0x28751048), STCP(0xd224c0cc, 0x37e05050), STCP(0xc680864f, 0x18e88cba)}, {STCP(0xaec24162, 0x9d1628a5), STCP(0xb7169316, 0xd78aefb8), STCP(0xd224c0cc, 0xc81fafb0), STCP(0xc680864f, 0xe7177346)}, {STCP(0x98012341, 0x4aa00ed1), STCP(0xc89ca1b2, 0xc1ab6bff), STCP(0xf8ea394e, 0xb8106bf4), STCP(0xcf542d73, 0xd888b99b)}, {STCP(0x43b137b3, 0x6ca2ca40), STCP(0xe0649cc4, 0xb2d69cca), STCP(0x22130c21, 0xc0405382), STCP(0xdbbf8fba, 0xcce3c7cc)}, {STCP(0x28fc4d71, 0x86bd3b87), STCP(0x09ccfeb9, 0xad319baf), STCP(0x46e51f02, 0xf1e5ea55), STCP(0xf30d5e34, 0xc2b0e335)}, {STCP(0xc798f756, 0x72e73c7d), STCP(0x3b6c3c1e, 0xc580dc72), STCP(0x2828a6ba, 0x3c1a14fb), STCP(0x14b733bb, 0xc4dcaae1)}, {STCP(0x46dcadd3, 0x956795c7), STCP(0x52f32fae, 0xf78048cd), STCP(0xd7d75946, 0x3c1a14fb), STCP(0x306017cb, 0xd82c0a75)}, {STCP(0xabe197de, 0x607a675e), STCP(0x460cef6e, 0x2d3b264e), STCP(0xb91ae0fe, 0xf1e5ea55), STCP(0x3e03e5e0, 0xf706590e)}, {STCP(0xb1b4f509, 0x9abcaf5f), STCP(0xfeb0b4be, 0x535fb8ba), STCP(0x1ba96f8e, 0xbd37e6d8), STCP(0x30f6dbbb, 0x271a0743)}, {STCP(0xce75b52a, 0x89f9be61), STCP(0xb26e4dda, 0x101054c5), STCP(0x1a475d2e, 0x3f714b19), STCP(0xf491f154, 0x3a6baf46)}, {STCP(0xee8fdfcb, 0x813181fa), STCP(0xe11e1a00, 0xbb9a6039), STCP(0xc3e582f5, 0xe71ab533), STCP(0xc9eb35e2, 0x0ffd212a)}, {STCP(0x0fd7d92f, 0x80fbf975), STCP(0x38adccbc, 0xd571bbf4), STCP(0x38c3aefc, 0xe87cc794), STCP(0xdafe8c3d, 0xd9b16100)}, {STCP(0x300d9e10, 0x895cc359), STCP(0x32b9843e, 0x2b52adcc), STCP(0xe9ded9f4, 0x356ce0ed), STCP(0x0fdd5ca3, 0xd072932e)}, {STCP(0x4d03b4f8, 0x99c2dec3), STCP(0xe2bc8d94, 0x3744e195), STCP(0xeb40ec55, 0xcde9ed22), STCP(0x2e67e231, 0xf893470b)}, {STCP(0x64c4deb3, 0xb112790f), STCP(0xc7b32682, 0xf099172d), STCP(0x2ebf44cf, 0x135d014a), STCP(0x1a2bacd5, 0x23334254)}, {STCP(0x75b5f9aa, 0xcdb81e14), STCP(0x028d9bb1, 0xc9dc45b9), STCP(0xd497893f, 0x11faeee9), STCP(0xee40ff71, 0x24a91b85)}, {STCP(0x7eb1cd81, 0xedc3feec), STCP(0x31491897, 0xf765f6d8), STCP(0x1098dc89, 0xd7ee574e), STCP(0xda6b816d, 0x011f35cf)}, {STCP(0x7f1cde01, 0x0f0b7727), STCP(0x118ce49d, 0x2a5ecda4), STCP(0x0f36ca28, 0x24badaa3), STCP(0xef2908a4, 0xe1ee3743)}, {STCP(0x76efee25, 0x2f4e8c3a), STCP(0xdde3be2a, 0x17f92215), STCP(0xde9bf36c, 0xf22b4839), STCP(0x1128fc0c, 0xe5c95f5a)}, {STCP(0x66b87d65, 0x4c5ede42), STCP(0xe43f351a, 0xe6bf22dc), STCP(0x1e0d3e85, 0xf38d5a9a), STCP(0x1c0f44a3, 0x02c92fe3)}, {STCP(0x4f8f36b7, 0x6445680f), STCP(0x10867ea2, 0xe3072740), STCP(0xf4ef6cfa, 0x1ab67076), STCP(0x09562a8a, 0x1742bb8b)}, {STCP(0x3304f6ec, 0x7564812a), STCP(0x1be4f1a8, 0x0894d75a), STCP(0xf6517f5b, 0xe8a05d98), STCP(0xf1bb0053, 0x10a78853)}, {STCP(0x1307b2c5, 0x7e93d532), STCP(0xfe098e27, 0x18f02a58), STCP(0x1408d459, 0x084c6e44), STCP(0xedafe5bd, 0xfbc15b2e)}, {STCP(0xf1c111cd, 0x7f346c97), STCP(0xeb5ca6a0, 0x02efee93), STCP(0xef4df9b6, 0x06ea5be4), STCP(0xfc149289, 0xf0d53ce4)}, {STCP(0xd1710001, 0x773b6beb), STCP(0xfa1aeb8c, 0xf06655ff), STCP(0x05884983, 0xf2a4c7c5), STCP(0x094f13df, 0xf79c01bf)}, {STCP(0xb446be0b, 0x6732cfca), STCP(0x0a743752, 0xf9220dfa), STCP(0x04263722, 0x0a046a2c), STCP(0x08ced80b, 0x0347e9c2)}, {STCP(0x9c3b1202, 0x503018a5), STCP(0x05fcf01a, 0x05cd8529), STCP(0xf95263e2, 0xfd3bdb3f), STCP(0x00c68cf9, 0x0637cb7f)}, {STCP(0x8aee2710, 0x33c187ec), STCP(0xfdd253f8, 0x038e09b9), STCP(0x0356ce0f, 0xfe9ded9f), STCP(0xfd6c3054, 0x01c8060a)}}; const FIXP_DECORR DecorrNumeratorReal0_USAC [MAX_DECORR_SEED_USAC][DECORR_FILTER_ORDER_BAND_0_USAC + 1] = { {DECORR(0x05bf4880), DECORR(0x08321c00), DECORR(0xe9315ee0), DECORR(0x07d9dd20), DECORR(0x02224994), DECORR(0x0009d200), DECORR(0xf8a29358), DECORR(0xf4e310d0), DECORR(0xef901fc0), DECORR(0xebda0460), DECORR(0x40000000)}}; const FIXP_DECORR DecorrNumeratorReal1_USAC [MAX_DECORR_SEED_USAC][DECORR_FILTER_ORDER_BAND_1_USAC + 1] = { {DECORR(0xf82f8378), DECORR(0xfef588c2), DECORR(0x02eddbd8), DECORR(0x041c2450), DECORR(0xf7edcd60), DECORR(0x07e29310), DECORR(0xfa4ece48), DECORR(0xed9f8a20), DECORR(0x40000000)}}; /* identical to MPS coeffs for reverb band 3: DecorrNumeratorReal3[0] */ const FIXP_DECORR DecorrNumeratorReal2_USAC[MAX_DECORR_SEED_USAC] [DECORR_FILTER_ORDER_BAND_2_USAC + 1] = { {DECORR(0x0248e8a8), DECORR(0xfde95838), DECORR(0x084823c0), DECORR(0x40000000)}}; const FIXP_DECORR DecorrNumeratorReal3_USAC[MAX_DECORR_SEED_USAC] [DECORR_FILTER_ORDER_BAND_3_USAC + 1] = { {DECORR(0xff2b020c), DECORR(0x02393830), DECORR(0x40000000)}}; /* const FIXP_DECORR DecorrNumeratorReal0_LD[MAX_DECORR_SEED_LD][] does not * exist */ RAM_ALIGN const FIXP_DECORR DecorrNumeratorReal1_LD[MAX_DECORR_SEED_LD] [DECORR_FILTER_ORDER_BAND_1_LD + 1] = { { DECORR(0xf310cb29), DECORR(0x1932d745), DECORR(0x0cc2d917), DECORR(0xddde064e), DECORR(0xf234a626), DECORR(0x198551a6), DECORR(0x17141b6a), DECORR(0xf298803d), DECORR(0xef98be92), DECORR(0x09ea1706), DECORR(0x28fbdff4), DECORR(0x1a869eb9), DECORR(0xdeefe147), DECORR(0xcde2adda), DECORR(0x13ddc619), DECORR(0x40000000), }, { DECORR(0x041d7dbf), DECORR(0x01b7309c), DECORR(0xfb599834), DECORR(0x092fc5ed), DECORR(0xf2fd7c25), DECORR(0xdd51e2eb), DECORR(0xf62fe72b), DECORR(0x0b15d588), DECORR(0xf1f091a7), DECORR(0xed1bbbfe), DECORR(0x03526899), DECORR(0x180cb256), DECORR(0xecf1433d), DECORR(0xf626ab95), DECORR(0x197dd27e), DECORR(0x40000000), }, { DECORR(0x157a786c), DECORR(0x0028c98c), DECORR(0xf5eff57b), DECORR(0x11f7d04f), DECORR(0xf390d28d), DECORR(0x18947081), DECORR(0xe5dc2319), DECORR(0xf4cc0235), DECORR(0x2394d47f), DECORR(0xe069230e), DECORR(0x03a1a773), DECORR(0xfbc9b092), DECORR(0x15a0173b), DECORR(0x0e9ecdf0), DECORR(0xd309b2c7), DECORR(0x40000000), }, { DECORR(0xe0ce703b), DECORR(0xe508b672), DECORR(0xef362398), DECORR(0xffe788ef), DECORR(0x2fda3749), DECORR(0x4671c0c6), DECORR(0x3c003494), DECORR(0x2387707c), DECORR(0xd2107d2e), DECORR(0xb3e47e08), DECORR(0xacd0abca), DECORR(0xc70791df), DECORR(0x0b586e85), DECORR(0x2f11cda7), DECORR(0x3a4a210b), DECORR(0x40000000), }, }; RAM_ALIGN const FIXP_DECORR DecorrNumeratorReal2_LD[MAX_DECORR_SEED_LD] [DECORR_FILTER_ORDER_BAND_2_LD + 1 + DECORR_ZERO_PADDING] = { { DECORR(0xffb4a234), DECORR(0x01ac71a2), DECORR(0xf2bca010), DECORR(0xfe3d7593), DECORR(0x093e9976), DECORR(0xf2c5f3f5), DECORR(0x40000000), }, { DECORR(0xe303afb8), DECORR(0xcd70c2bb), DECORR(0xf1e2ad7e), DECORR(0x0c8ffbe2), DECORR(0x21f80abf), DECORR(0x3d08410c), DECORR(0x40000000), }, { DECORR(0xe26809d5), DECORR(0x0efbcfa4), DECORR(0x210c1a97), DECORR(0xfe60af4e), DECORR(0xeda01a51), DECORR(0x00faf468), DECORR(0x40000000), }, { DECORR(0x1edc5d64), DECORR(0xe5b2e35c), DECORR(0xe94b1c45), DECORR(0x30a6f1e1), DECORR(0xf04e52de), DECORR(0xe30de45a), DECORR(0x40000000), }, }; RAM_ALIGN const FIXP_DECORR DecorrNumeratorReal3_LD[MAX_DECORR_SEED_LD] [DECORR_FILTER_ORDER_BAND_3_LD + 1] = { { DECORR(0x0248e8a7), DECORR(0xfde9583b), DECORR(0x084823bb), DECORR(0x40000000), }, { DECORR(0x1db22d0e), DECORR(0xfc773992), DECORR(0x0e819a74), DECORR(0x40000000), }, { DECORR(0x0fcb923a), DECORR(0x0154b7ff), DECORR(0xe70cb647), DECORR(0x40000000), }, { DECORR(0xe39f559b), DECORR(0xe06dd6ca), DECORR(0x19f71f71), DECORR(0x40000000), }, }; FIXP_DBL *getAddrDirectSignalMaxVal(HANDLE_DECORR_DEC self) { return &(self->ducker.maxValDirectData); } static INT DecorrFilterInit(DECORR_FILTER_INSTANCE *const self, FIXP_MPS *pStateBufferCplx, FIXP_DBL *pDelayBufferCplx, INT *offsetStateBuffer, INT *offsetDelayBuffer, INT const decorr_seed, INT const reverb_band, INT const useFractDelay, INT const noSampleDelay, INT const filterOrder, FDK_DECORR_TYPE const decorrType) { INT errorCode = 0; switch (decorrType) { case DECORR_USAC: if (useFractDelay) { return 1; } else { FDK_ASSERT(decorr_seed == 0); switch (reverb_band) { case 0: self->numeratorReal = DecorrNumeratorReal0_USAC[decorr_seed]; break; case 1: self->numeratorReal = DecorrNumeratorReal1_USAC[decorr_seed]; break; case 2: self->numeratorReal = DecorrNumeratorReal2_USAC[decorr_seed]; break; case 3: self->numeratorReal = DecorrNumeratorReal3_USAC[decorr_seed]; break; } } break; case DECORR_LD: FDK_ASSERT(decorr_seed < MAX_DECORR_SEED_LD); switch (reverb_band) { case 0: self->numeratorReal = NULL; break; case 1: self->numeratorReal = DecorrNumeratorReal1_LD[decorr_seed]; break; case 2: self->numeratorReal = DecorrNumeratorReal2_LD[decorr_seed]; break; case 3: self->numeratorReal = DecorrNumeratorReal3_LD[decorr_seed]; break; } break; default: return 1; } self->stateCplx = pStateBufferCplx + (*offsetStateBuffer); *offsetStateBuffer += 2 * filterOrder; self->DelayBufferCplx = pDelayBufferCplx + (*offsetDelayBuffer); *offsetDelayBuffer += 2 * noSampleDelay; return errorCode; } /******************************************************************************* *******************************************************************************/ static INT DecorrFilterInitPS(DECORR_FILTER_INSTANCE *const self, FIXP_MPS *pStateBufferCplx, FIXP_DBL *pDelayBufferCplx, INT *offsetStateBuffer, INT *offsetDelayBuffer, INT const hybridBand, INT const reverbBand, INT const noSampleDelay) { INT errorCode = 0; if (reverbBand == 0) { self->coeffsPacked = DecorrPsCoeffsCplx[hybridBand]; self->stateCplx = pStateBufferCplx + (*offsetStateBuffer); *offsetStateBuffer += 2 * DECORR_FILTER_ORDER_PS; } self->DelayBufferCplx = pDelayBufferCplx + (*offsetDelayBuffer); *offsetDelayBuffer += 2 * noSampleDelay; return errorCode; } LNK_SECTION_CODE_L1 static INT DecorrFilterApplyPASS(DECORR_FILTER_INSTANCE const filter[], FIXP_DBL *dataRealIn, FIXP_DBL *dataImagIn, FIXP_DBL *dataRealOut, FIXP_DBL *dataImagOut, INT start, INT stop, INT reverbBandNoSampleDelay, INT reverbBandDelayBufferIndex) { INT i; INT offset = 2 * reverbBandNoSampleDelay; FIXP_MPS *pDelayBuffer = &filter[start].DelayBufferCplx[reverbBandDelayBufferIndex]; /* Memory for the delayline has been allocated in a consecutive order, so we can address from filter to filter with a constant length. Be aware that real and imaginary part of the delayline are stored in interleaved order. */ if (dataImagIn == NULL) { for (i = start; i < stop; i++) { FIXP_DBL tmp; tmp = *pDelayBuffer; *pDelayBuffer = dataRealIn[i]; dataRealOut[i] = tmp; pDelayBuffer += offset; } } else { if ((i = stop - start) != 0) { dataRealIn += start; dataImagIn += start; dataRealOut += start; dataImagOut += start; do { FIXP_DBL delay_re, delay_im, real, imag; real = *dataRealIn++; imag = *dataImagIn++; delay_re = pDelayBuffer[0]; delay_im = pDelayBuffer[1]; pDelayBuffer[0] = real; pDelayBuffer[1] = imag; *dataRealOut++ = delay_re; *dataImagOut++ = delay_im; pDelayBuffer += offset; } while (--i != 0); } } return (INT)0; } #ifndef FUNCTION_DecorrFilterApplyREAL LNK_SECTION_CODE_L1 static INT DecorrFilterApplyREAL(DECORR_FILTER_INSTANCE const filter[], FIXP_DBL *dataRealIn, FIXP_DBL *dataImagIn, FIXP_DBL *dataRealOut, FIXP_DBL *dataImagOut, INT start, INT stop, INT reverbFilterOrder, INT reverbBandNoSampleDelay, INT reverbBandDelayBufferIndex) { INT i, j; FIXP_DBL xReal, xImag, yReal, yImag; const FIXP_DECORR *pFilter = filter[start].numeratorReal; INT offsetDelayBuffer = (2 * reverbBandNoSampleDelay) - 1; FIXP_MPS *pDelayBuffer = &filter[start].DelayBufferCplx[reverbBandDelayBufferIndex]; INT offsetStates = 2 * reverbFilterOrder; FIXP_DBL *pStates = filter[start].stateCplx; /* Memory for the delayline has been allocated in a consecutive order, so we can address from filter to filter with a constant length. The same is valid for the states. Be aware that real and imaginary part of the delayline and the states are stored in interleaved order. All filter in a reverb band have the same filter coefficients. Exploit symmetry: numeratorReal[i] = denominatorReal[reverbFilterLength-1-i] Do not accumulate the highest states which are always zero. */ if (reverbFilterOrder == 2) { FIXP_DECORR nFilt0L, nFilt0H; nFilt0L = pFilter[0]; nFilt0H = pFilter[1]; for (i = start; i < stop; i++) { xReal = *pDelayBuffer; *pDelayBuffer = dataRealIn[i]; pDelayBuffer++; xImag = *pDelayBuffer; *pDelayBuffer = dataImagIn[i]; pDelayBuffer += offsetDelayBuffer; yReal = (pStates[0] + fMultDiv2(xReal, nFilt0L)) << FILTER_SF; yImag = (pStates[1] + fMultDiv2(xImag, nFilt0L)) << FILTER_SF; dataRealOut[i] = yReal; dataImagOut[i] = yImag; pStates[0] = pStates[2] + fMultDiv2(xReal, nFilt0H) - fMultDiv2(yReal, nFilt0H); pStates[1] = pStates[3] + fMultDiv2(xImag, nFilt0H) - fMultDiv2(yImag, nFilt0H); pStates[2] = (xReal >> FILTER_SF) - fMultDiv2(yReal, nFilt0L); pStates[3] = (xImag >> FILTER_SF) - fMultDiv2(yImag, nFilt0L); pStates += offsetStates; } } else if (reverbFilterOrder == 3) { FIXP_DECORR nFilt0L, nFilt0H, nFilt1L; nFilt0L = pFilter[0]; nFilt0H = pFilter[1]; nFilt1L = pFilter[2]; for (i = start; i < stop; i++) { xReal = *pDelayBuffer; *pDelayBuffer = dataRealIn[i]; pDelayBuffer++; xImag = *pDelayBuffer; *pDelayBuffer = dataImagIn[i]; pDelayBuffer += offsetDelayBuffer; yReal = (pStates[0] + fMultDiv2(xReal, nFilt0L)) << FILTER_SF; yImag = (pStates[1] + fMultDiv2(xImag, nFilt0L)) << FILTER_SF; dataRealOut[i] = yReal; dataImagOut[i] = yImag; pStates[0] = pStates[2] + fMultDiv2(xReal, nFilt0H) - fMultDiv2(yReal, nFilt1L); pStates[1] = pStates[3] + fMultDiv2(xImag, nFilt0H) - fMultDiv2(yImag, nFilt1L); pStates[2] = pStates[4] + fMultDiv2(xReal, nFilt1L) - fMultDiv2(yReal, nFilt0H); pStates[3] = pStates[5] + fMultDiv2(xImag, nFilt1L) - fMultDiv2(yImag, nFilt0H); pStates[4] = (xReal >> FILTER_SF) - fMultDiv2(yReal, nFilt0L); pStates[5] = (xImag >> FILTER_SF) - fMultDiv2(yImag, nFilt0L); pStates += offsetStates; } } else if (reverbFilterOrder == 6) { FIXP_DECORR nFilt0L, nFilt0H, nFilt1L, nFilt1H, nFilt2L, nFilt2H; nFilt0L = pFilter[0]; nFilt0H = pFilter[1]; nFilt1L = pFilter[2]; nFilt1H = pFilter[3]; nFilt2L = pFilter[4]; nFilt2H = pFilter[5]; for (i = start; i < stop; i++) { xReal = *pDelayBuffer; *pDelayBuffer = dataRealIn[i]; pDelayBuffer++; xImag = *pDelayBuffer; *pDelayBuffer = dataImagIn[i]; pDelayBuffer += offsetDelayBuffer; yReal = (pStates[0] + fMultDiv2(xReal, nFilt0L)) << FILTER_SF; yImag = (pStates[1] + fMultDiv2(xImag, nFilt0L)) << FILTER_SF; dataRealOut[i] = yReal; dataImagOut[i] = yImag; pStates[0] = pStates[2] + fMultDiv2(xReal, nFilt0H) - fMultDiv2(yReal, nFilt2H); pStates[1] = pStates[3] + fMultDiv2(xImag, nFilt0H) - fMultDiv2(yImag, nFilt2H); pStates[2] = pStates[4] + fMultDiv2(xReal, nFilt1L) - fMultDiv2(yReal, nFilt2L); pStates[3] = pStates[5] + fMultDiv2(xImag, nFilt1L) - fMultDiv2(yImag, nFilt2L); pStates[4] = pStates[6] + fMultDiv2(xReal, nFilt1H) - fMultDiv2(yReal, nFilt1H); pStates[5] = pStates[7] + fMultDiv2(xImag, nFilt1H) - fMultDiv2(yImag, nFilt1H); pStates[6] = pStates[8] + fMultDiv2(xReal, nFilt2L) - fMultDiv2(yReal, nFilt1L); pStates[7] = pStates[9] + fMultDiv2(xImag, nFilt2L) - fMultDiv2(yImag, nFilt1L); pStates[8] = pStates[10] + fMultDiv2(xReal, nFilt2H) - fMultDiv2(yReal, nFilt0H); pStates[9] = pStates[11] + fMultDiv2(xImag, nFilt2H) - fMultDiv2(yImag, nFilt0H); pStates[10] = (xReal >> FILTER_SF) - fMultDiv2(yReal, nFilt0L); pStates[11] = (xImag >> FILTER_SF) - fMultDiv2(yImag, nFilt0L); pStates += offsetStates; } } else { FIXP_DECORR nFilt0L, nFilt0H; for (i = start; i < stop; i++) { xReal = *pDelayBuffer; *pDelayBuffer = dataRealIn[i]; pDelayBuffer++; xImag = *pDelayBuffer; *pDelayBuffer = dataImagIn[i]; pDelayBuffer += offsetDelayBuffer; nFilt0L = pFilter[0]; yReal = (pStates[0] + fMultDiv2(xReal, nFilt0L)) << 2; yImag = (pStates[1] + fMultDiv2(xImag, nFilt0L)) << 2; dataRealOut[i] = yReal; dataImagOut[i] = yImag; for (j = 1; j < reverbFilterOrder; j++) { nFilt0L = pFilter[j]; nFilt0H = pFilter[reverbFilterOrder - j]; pStates[2 * j - 2] = pStates[2 * j] + fMultDiv2(xReal, nFilt0L) - fMultDiv2(yReal, nFilt0H); pStates[2 * j - 1] = pStates[2 * j + 1] + fMultDiv2(xImag, nFilt0L) - fMultDiv2(yImag, nFilt0H); } nFilt0L = pFilter[j]; nFilt0H = pFilter[reverbFilterOrder - j]; pStates[2 * j - 2] = fMultDiv2(xReal, nFilt0L) - fMultDiv2(yReal, nFilt0H); pStates[2 * j - 1] = fMultDiv2(xImag, nFilt0L) - fMultDiv2(yImag, nFilt0H); pStates += offsetStates; } } return (INT)0; } #endif /* #ifndef FUNCTION_DecorrFilterApplyREAL */ #ifndef FUNCTION_DecorrFilterApplyCPLX_PS LNK_SECTION_CODE_L1 static INT DecorrFilterApplyCPLX_PS( DECORR_FILTER_INSTANCE const filter[], FIXP_DBL *dataRealIn, FIXP_DBL *dataImagIn, FIXP_DBL *dataRealOut, FIXP_DBL *dataImagOut, INT start, INT stop, INT reverbFilterOrder, INT reverbBandNoSampleDelay, INT reverbBandDelayBufferIndex, UCHAR *stateBufferOffset) { /* r = real, j = imaginary */ FIXP_DBL r_data_a, j_data_a, r_data_b, j_data_b, r_stage_mult, j_stage_mult; FIXP_STP rj_coeff; /* get pointer to current position in input delay buffer of filter with * starting-index */ FIXP_DBL *pDelayBuffer = &filter[start].DelayBufferCplx[reverbBandDelayBufferIndex]; /* increases by 2 every other call of this function */ /* determine the increment for this pointer to get to the correct position in * the delay buffer of the next filter */ INT offsetDelayBuffer = (2 * reverbBandNoSampleDelay) - 1; /* pointer to current position in state buffer */ FIXP_DBL *pStates = filter[start].stateCplx; INT pStatesIncrement = 2 * reverbFilterOrder; /* stateBufferOffset-pointers */ FIXP_DBL *pStateBufferOffset0 = pStates + stateBufferOffset[0]; FIXP_DBL *pStateBufferOffset1 = pStates + stateBufferOffset[1]; FIXP_DBL *pStateBufferOffset2 = pStates + stateBufferOffset[2]; /* traverse all hybrid-bands inbetween start- and stop-index */ for (int i = start; i < stop; i++) { /* 1. input delay (real/imaginary values interleaved) */ /* load delayed real input value */ r_data_a = *pDelayBuffer; /* store incoming real data value to delay buffer and increment pointer */ *pDelayBuffer++ = dataRealIn[i]; /* load delayed imaginary input value */ j_data_a = *pDelayBuffer; /* store incoming imaginary data value to delay buffer */ *pDelayBuffer = dataImagIn[i]; /* increase delay buffer by offset */ pDelayBuffer += offsetDelayBuffer; /* 2. Phi(k)-stage */ /* create pointer to coefficient table (real and imaginary coefficients * interleaved) */ const FIXP_STP *pCoeffs = filter[i].coeffsPacked; /* the first two entries of the coefficient table are the * Phi(k)-multiplicants */ rj_coeff = *pCoeffs++; /* multiply value from input delay buffer by looked-up values */ cplxMultDiv2(&r_data_b, &j_data_b, r_data_a, j_data_a, rj_coeff); /* 3. process all three filter stages */ /* stage 0 */ /* get coefficients from lookup table */ rj_coeff = *pCoeffs++; /* multiply output of last stage by coefficient */ cplxMultDiv2(&r_stage_mult, &j_stage_mult, r_data_b, j_data_b, rj_coeff); r_stage_mult <<= 1; j_stage_mult <<= 1; /* read and add value from state buffer (this is the input for the next * stage) */ r_data_a = r_stage_mult + pStateBufferOffset0[0]; j_data_a = j_stage_mult + pStateBufferOffset0[1]; /* negate r_data_a to perform multiplication with complex conjugate of * rj_coeff */ cplxMultDiv2(&r_stage_mult, &j_stage_mult, -r_data_a, j_data_a, rj_coeff); /* add stage input to shifted result */ r_stage_mult = r_data_b + (r_stage_mult << 1); j_stage_mult = j_data_b - (j_stage_mult << 1); /* store result to state buffer */ pStateBufferOffset0[0] = r_stage_mult; pStateBufferOffset0[1] = j_stage_mult; pStateBufferOffset0 += pStatesIncrement; /* stage 1 */ /* get coefficients from lookup table */ rj_coeff = *pCoeffs++; /* multiply output of last stage by coefficient */ cplxMultDiv2(&r_stage_mult, &j_stage_mult, r_data_a, j_data_a, rj_coeff); r_stage_mult <<= 1; j_stage_mult <<= 1; /* read and add value from state buffer (this is the input for the next * stage) */ r_data_b = r_stage_mult + pStateBufferOffset1[0]; j_data_b = j_stage_mult + pStateBufferOffset1[1]; /* negate r_data_b to perform multiplication with complex conjugate of * rj_coeff */ cplxMultDiv2(&r_stage_mult, &j_stage_mult, -r_data_b, j_data_b, rj_coeff); /* add stage input to shifted result */ r_stage_mult = r_data_a + (r_stage_mult << 1); j_stage_mult = j_data_a - (j_stage_mult << 1); /* store result to state buffer */ pStateBufferOffset1[0] = r_stage_mult; pStateBufferOffset1[1] = j_stage_mult; pStateBufferOffset1 += pStatesIncrement; /* stage 2 */ /* get coefficients from lookup table */ rj_coeff = *pCoeffs++; /* multiply output of last stage by coefficient */ cplxMultDiv2(&r_stage_mult, &j_stage_mult, r_data_b, j_data_b, rj_coeff); r_stage_mult <<= 1; j_stage_mult <<= 1; /* read and add value from state buffer (this is the input for the next * stage) */ r_data_a = r_stage_mult + pStateBufferOffset2[0]; j_data_a = j_stage_mult + pStateBufferOffset2[1]; /* negate r_data_a to perform multiplication with complex conjugate of * rj_coeff */ cplxMultDiv2(&r_stage_mult, &j_stage_mult, -r_data_a, j_data_a, rj_coeff); /* add stage input to shifted result */ r_stage_mult = r_data_b + (r_stage_mult << 1); j_stage_mult = j_data_b - (j_stage_mult << 1); /* store result to state buffer */ pStateBufferOffset2[0] = r_stage_mult; pStateBufferOffset2[1] = j_stage_mult; pStateBufferOffset2 += pStatesIncrement; /* write filter output */ dataRealOut[i] = r_data_a << 1; dataImagOut[i] = j_data_a << 1; } /* end of band/filter loop (outer loop) */ /* update stateBufferOffset with respect to ring buffer boundaries */ if (stateBufferOffset[0] == 4) stateBufferOffset[0] = 0; else stateBufferOffset[0] += 2; if (stateBufferOffset[1] == 12) stateBufferOffset[1] = 6; else stateBufferOffset[1] += 2; if (stateBufferOffset[2] == 22) stateBufferOffset[2] = 14; else stateBufferOffset[2] += 2; return (INT)0; } #endif /* FUNCTION_DecorrFilterApplyCPLX_PS */ /******************************************************************************* *******************************************************************************/ static INT DuckerInit(DUCKER_INSTANCE *const self, int const hybridBands, int partiallyComplex, const FDK_DUCKER_TYPE duckerType, const int nParamBands, int initStatesFlag) { INT errorCode = 0; if (self) { switch (nParamBands) { case (20): FDK_ASSERT(hybridBands == 71); self->mapHybBands2ProcBands = kernels_20_to_71_PS; self->mapProcBands2HybBands = kernels_20_to_71_offset_PS; self->parameterBands = (20); break; case (28): self->mapHybBands2ProcBands = kernels_28_to_71; self->mapProcBands2HybBands = kernels_28_to_71_offset; self->parameterBands = (28); break; case (23): FDK_ASSERT(hybridBands == 64 || hybridBands == 32); self->mapHybBands2ProcBands = kernels_23_to_64; self->mapProcBands2HybBands = kernels_23_to_64_offset; self->parameterBands = (23); break; default: return 1; } self->qs_next = &self->mapProcBands2HybBands[1]; self->maxValDirectData = FL2FXCONST_DBL(-1.0f); self->maxValReverbData = FL2FXCONST_DBL(-1.0f); self->scaleDirectNrg = 2 * DUCKER_MAX_NRG_SCALE; self->scaleReverbNrg = 2 * DUCKER_MAX_NRG_SCALE; self->scaleSmoothDirRevNrg = 2 * DUCKER_MAX_NRG_SCALE; self->headroomSmoothDirRevNrg = 2 * DUCKER_MAX_NRG_SCALE; self->hybridBands = hybridBands; self->partiallyComplex = partiallyComplex; if (initStatesFlag && (duckerType == DUCKER_PS)) { int pb; for (pb = 0; pb < self->parameterBands; pb++) { self->SmoothDirRevNrg[pb] = (FIXP_MPS)0; } } } else errorCode = 1; return errorCode; } /******************************************************************************* *******************************************************************************/ #ifndef FUNCTION_DuckerCalcEnergy static INT DuckerCalcEnergy(DUCKER_INSTANCE *const self, FIXP_DBL const inputReal[(71)], FIXP_DBL const inputImag[(71)], FIXP_DBL energy[(28)], FIXP_DBL inputMaxVal, SCHAR *nrgScale, int mode, /* 1:(ps) 0:(else) */ int startHybBand) { INT err = 0; int qs, maxHybBand; int maxHybridBand = self->hybridBands - 1; maxHybBand = maxHybridBand; FDKmemclear(energy, (28) * sizeof(FIXP_DBL)); if (mode == 1) { int pb; int clz; FIXP_DBL maxVal = FL2FXCONST_DBL(-1.0f); if (maxVal == FL2FXCONST_DBL(-1.0f)) { clz = fMin(getScalefactor(&inputReal[startHybBand], fMax(0, maxHybridBand - startHybBand + 1)), getScalefactor(&inputImag[startHybBand], fMax(0, maxHybBand - startHybBand + 1))); } else { clz = CntLeadingZeros(maxVal) - 1; } clz = fMin(fMax(0, clz - DUCKER_HEADROOM_BITS), DUCKER_MAX_NRG_SCALE); *nrgScale = (SCHAR)clz << 1; /* Initialize pb since it would stay uninitialized for the case startHybBand * > maxHybBand. */ pb = SpatialDecGetProcessingBand(maxHybBand, self->mapHybBands2ProcBands); for (qs = startHybBand; qs <= maxHybBand; qs++) { pb = SpatialDecGetProcessingBand(qs, self->mapHybBands2ProcBands); energy[pb] = SATURATE_LEFT_SHIFT( (energy[pb] >> 1) + (fPow2Div2(inputReal[qs] << clz) >> 1) + (fPow2Div2(inputImag[qs] << clz) >> 1), 1, DFRACT_BITS); } pb++; for (; pb <= SpatialDecGetProcessingBand(maxHybridBand, self->mapHybBands2ProcBands); pb++) { FDK_ASSERT(pb != SpatialDecGetProcessingBand( qs - 1, self->mapHybBands2ProcBands)); int qs_next; FIXP_DBL nrg = 0; qs_next = (int)self->qs_next[pb]; for (; qs < qs_next; qs++) { nrg = fAddSaturate(nrg, fPow2Div2(inputReal[qs] << clz)); } energy[pb] = nrg; } } else { int clz; FIXP_DBL maxVal = FL2FXCONST_DBL(-1.0f); maxVal = inputMaxVal; if (maxVal == FL2FXCONST_DBL(-1.0f)) { clz = fMin(getScalefactor(&inputReal[startHybBand], fMax(0, maxHybridBand - startHybBand + 1)), getScalefactor(&inputImag[startHybBand], fMax(0, maxHybBand - startHybBand + 1))); } else { clz = CntLeadingZeros(maxVal) - 1; } clz = fMin(fMax(0, clz - DUCKER_HEADROOM_BITS), DUCKER_MAX_NRG_SCALE); *nrgScale = (SCHAR)clz << 1; for (qs = startHybBand; qs <= maxHybBand; qs++) { int pb = SpatialDecGetProcessingBand(qs, self->mapHybBands2ProcBands); energy[pb] = SATURATE_LEFT_SHIFT( (energy[pb] >> 1) + (fPow2Div2(inputReal[qs] << clz) >> 1) + (fPow2Div2(inputImag[qs] << clz) >> 1), 1, DFRACT_BITS); } for (; qs <= maxHybridBand; qs++) { int pb = SpatialDecGetProcessingBand(qs, self->mapHybBands2ProcBands); energy[pb] = fAddSaturate(energy[pb], fPow2Div2(inputReal[qs] << clz)); } } { /* Catch overflows which have been observed in erred bitstreams to avoid * assertion failures later. */ int pb; for (pb = 0; pb < (28); pb++) { energy[pb] = (FIXP_DBL)((LONG)energy[pb] & (LONG)MAXVAL_DBL); } } return err; } #endif /* #ifndef FUNCTION_DuckerCalcEnergy */ LNK_SECTION_CODE_L1 static INT DuckerApply(DUCKER_INSTANCE *const self, FIXP_DBL const directNrg[(28)], FIXP_DBL outputReal[(71)], FIXP_DBL outputImag[(71)], int startHybBand) { INT err = 0; int qs = startHybBand; int qs_next = 0; int pb = 0; int startParamBand = 0; int hybBands; int hybridBands = self->hybridBands; C_ALLOC_SCRATCH_START(reverbNrg, FIXP_DBL, (28)); FIXP_DBL *smoothDirRevNrg = &self->SmoothDirRevNrg[0]; FIXP_DUCK_GAIN duckGain = 0; int doScaleNrg = 0; int scaleDirectNrg = 0; int scaleReverbNrg = 0; int scaleSmoothDirRevNrg = 0; FIXP_DBL maxDirRevNrg = FL2FXCONST_DBL(0.0); hybBands = hybridBands; startParamBand = SpatialDecGetProcessingBand(startHybBand, self->mapHybBands2ProcBands); DuckerCalcEnergy(self, outputReal, outputImag, reverbNrg, self->maxValReverbData, &(self->scaleReverbNrg), 0, startHybBand); if ((self->scaleDirectNrg != self->scaleReverbNrg) || (self->scaleDirectNrg != self->scaleSmoothDirRevNrg) || (self->headroomSmoothDirRevNrg == 0)) { int scale; scale = fixMin(self->scaleDirectNrg, self->scaleSmoothDirRevNrg + self->headroomSmoothDirRevNrg - 1); scale = fixMin(scale, self->scaleReverbNrg); scaleDirectNrg = fMax(fMin(self->scaleDirectNrg - scale, (DFRACT_BITS - 1)), -(DFRACT_BITS - 1)); scaleReverbNrg = fMax(fMin(self->scaleReverbNrg - scale, (DFRACT_BITS - 1)), -(DFRACT_BITS - 1)); scaleSmoothDirRevNrg = fMax(fMin(self->scaleSmoothDirRevNrg - scale, (DFRACT_BITS - 1)), -(DFRACT_BITS - 1)); self->scaleSmoothDirRevNrg = (SCHAR)scale; doScaleNrg = 1; } for (pb = startParamBand; pb < self->parameterBands; pb++) { FIXP_DBL tmp1; FIXP_DBL tmp2; INT s; /* smoothDirRevNrg[2*pb ] = fMult(smoothDirRevNrg[2*pb ],DUCK_ALPHA_FDK) + fMultDiv2(directNrg[pb],DUCK_ONE_MINUS_ALPHA_X4_FDK); smoothDirRevNrg[2*pb+1] = fMult(smoothDirRevNrg[2*pb+1],DUCK_ALPHA_FDK) + fMultDiv2(reverbNrg[pb],DUCK_ONE_MINUS_ALPHA_X4_FDK); tmp1 = fMult(smoothDirRevNrg[2*pb],DUCK_GAMMA_FDK); tmp2 = smoothDirRevNrg[2*pb+1] >> 1; */ tmp1 = smoothDirRevNrg[2 * pb + 0]; tmp2 = smoothDirRevNrg[2 * pb + 1]; tmp1 = fMult(tmp1, DUCK_ALPHA_FDK); tmp2 = fMult(tmp2, DUCK_ALPHA_FDK); if (doScaleNrg) { int scaleSmoothDirRevNrg_asExponent = -scaleSmoothDirRevNrg; tmp1 = scaleValue(tmp1, scaleSmoothDirRevNrg_asExponent); tmp2 = scaleValue(tmp2, scaleSmoothDirRevNrg_asExponent); tmp1 = fMultAddDiv2(tmp1, scaleValue(directNrg[pb], -scaleDirectNrg), DUCK_ONE_MINUS_ALPHA_X4_FDK); tmp2 = fMultAddDiv2(tmp2, scaleValue(reverbNrg[pb], -scaleReverbNrg), DUCK_ONE_MINUS_ALPHA_X4_FDK); } else { tmp1 = fMultAddDiv2(tmp1, directNrg[pb], DUCK_ONE_MINUS_ALPHA_X4_FDK); tmp2 = fMultAddDiv2(tmp2, reverbNrg[pb], DUCK_ONE_MINUS_ALPHA_X4_FDK); } smoothDirRevNrg[2 * pb] = tmp1; smoothDirRevNrg[2 * pb + 1] = tmp2; maxDirRevNrg |= fAbs(tmp1); maxDirRevNrg |= fAbs(tmp2); tmp1 = fMult(tmp1, DUCK_GAMMA_FDK); tmp2 = tmp2 >> 1; qs_next = fMin((int)self->qs_next[pb], self->hybridBands); if (tmp2 > tmp1) { /* true for about 20% */ /* gain smaller than 1.0 */ tmp1 = sqrtFixp(tmp1); tmp2 = invSqrtNorm2(tmp2, &s); duckGain = FX_DBL2FX_DUCK_GAIN(fMultDiv2(tmp1, tmp2) << s); } else { /* true for about 80 % */ tmp2 = smoothDirRevNrg[2 * pb] >> 1; tmp1 = fMult(smoothDirRevNrg[2 * pb + 1], DUCK_GAMMA_FDK); if (tmp2 > tmp1) { /* true for about 20% */ if (tmp1 <= (tmp2 >> 2)) { /* limit gain to 2.0 */ if (qs < hybBands) { for (; qs < qs_next; qs++) { outputReal[qs] = outputReal[qs] << 1; outputImag[qs] = outputImag[qs] << 1; } } else { for (; qs < qs_next; qs++) { outputReal[qs] = outputReal[qs] << 1; } } /* skip general gain*output section */ continue; } else { /* gain from 1.0 to 2.0 */ tmp2 = sqrtFixp(tmp2 >> 2); tmp1 = invSqrtNorm2(tmp1, &s); duckGain = FX_DBL2FX_DUCK_GAIN(fMult(tmp1, tmp2) << s); } } else { /* true for about 60% */ /* gain = 1.0; output does not change; update qs index */ qs = qs_next; continue; } } /* general gain*output section */ if (qs < hybBands) { /* true for about 39% */ for (; qs < qs_next; qs++) { /* runs about 2 times */ outputReal[qs] = fMultDiv2(outputReal[qs], duckGain) << 2; outputImag[qs] = fMultDiv2(outputImag[qs], duckGain) << 2; } } else { for (; qs < qs_next; qs++) { outputReal[qs] = fMultDiv2(outputReal[qs], duckGain) << 2; } } } /* pb */ self->headroomSmoothDirRevNrg = (SCHAR)fixMax(0, CntLeadingZeros(maxDirRevNrg) - 1); C_ALLOC_SCRATCH_END(reverbNrg, FIXP_DBL, (28)); return err; } LNK_SECTION_CODE_L1 static INT DuckerApplyPS(DUCKER_INSTANCE *const self, FIXP_DBL const directNrg[(28)], FIXP_DBL outputReal[(71)], FIXP_DBL outputImag[(71)], int startHybBand) { int qs = startHybBand; int pb = 0; int startParamBand = SpatialDecGetProcessingBand(startHybBand, self->mapHybBands2ProcBands); int hybBands; int doScaleNrg = 0; int scaleDirectNrg = 0; int scaleSmoothDirRevNrg = 0; FIXP_DBL maxDirRevNrg = FL2FXCONST_DBL(0.0); if ((self->scaleDirectNrg != self->scaleSmoothDirRevNrg) || (self->headroomSmoothDirRevNrg == 0)) { int scale; scale = fixMin(self->scaleDirectNrg, self->scaleSmoothDirRevNrg + self->headroomSmoothDirRevNrg - 2); scaleDirectNrg = fMax(fMin(self->scaleDirectNrg - scale, (DFRACT_BITS - 1)), -(DFRACT_BITS - 1)); scaleSmoothDirRevNrg = fMax(fMin(self->scaleSmoothDirRevNrg - scale, (DFRACT_BITS - 1)), -(DFRACT_BITS - 1)); self->scaleSmoothDirRevNrg = (SCHAR)scale; doScaleNrg = 1; } hybBands = self->hybridBands; FDK_ASSERT((self->parameterBands == (28)) || (self->parameterBands == (20))); for (pb = startParamBand; pb < self->parameterBands; pb++) { FIXP_DBL directNrg2 = directNrg[pb]; if (doScaleNrg) { directNrg2 = scaleValue(directNrg2, -scaleDirectNrg); self->peakDiff[pb] = scaleValue(self->peakDiff[pb], -scaleSmoothDirRevNrg); self->peakDecay[pb] = scaleValue(self->peakDecay[pb], -scaleSmoothDirRevNrg); self->SmoothDirRevNrg[pb] = scaleValue(self->SmoothDirRevNrg[pb], -scaleSmoothDirRevNrg); } self->peakDecay[pb] = fixMax( directNrg2, fMult(self->peakDecay[pb], PS_DUCK_PEAK_DECAY_FACTOR_FDK)); self->peakDiff[pb] = self->peakDiff[pb] + fMult(PS_DUCK_FILTER_COEFF_FDK, (self->peakDecay[pb] - directNrg2 - self->peakDiff[pb])); self->SmoothDirRevNrg[pb] = fixMax(self->SmoothDirRevNrg[pb] + fMult(PS_DUCK_FILTER_COEFF_FDK, (directNrg2 - self->SmoothDirRevNrg[pb])), FL2FXCONST_DBL(0)); maxDirRevNrg |= fAbs(self->peakDiff[pb]); maxDirRevNrg |= fAbs(self->SmoothDirRevNrg[pb]); if ((self->peakDiff[pb] == FL2FXCONST_DBL(0)) && (self->SmoothDirRevNrg[pb] == FL2FXCONST_DBL(0))) { int qs_next; qs = fMax(qs, SpatialDecGetQmfBand(pb, self->mapProcBands2HybBands)); qs_next = fMin((int)self->qs_next[pb], self->hybridBands); FIXP_DBL *pOutputReal = &outputReal[qs]; FIXP_DBL *pOutputImag = &outputImag[qs]; if (qs < hybBands) { for (; qs < qs_next; qs++) { *pOutputReal++ = FL2FXCONST_DBL(0); *pOutputImag++ = FL2FXCONST_DBL(0); } } else { for (; qs < qs_next; qs++) { *pOutputReal++ = FL2FXCONST_DBL(0); } } } else if (self->peakDiff[pb] != FL2FXCONST_DBL(0)) { FIXP_DBL multiplication = fMult(FL2FXCONST_DUCK(0.75f), self->peakDiff[pb]); if (multiplication > (self->SmoothDirRevNrg[pb] >> 1)) { FIXP_DBL num, denom, duckGain; int scale, qs_next; /* implement x/y as (sqrt(x)*invSqrt(y))^2 */ num = sqrtFixp(self->SmoothDirRevNrg[pb] >> 1); denom = self->peakDiff[pb] + FL2FXCONST_DBL(ABS_THR / (32768.0f * 32768.0f * 128.0f * 1.5f)); denom = invSqrtNorm2(denom, &scale); /* duck output whether duckGain != 1.f */ qs = fMax(qs, SpatialDecGetQmfBand(pb, self->mapProcBands2HybBands)); qs_next = fMin((int)self->qs_next[pb], self->hybridBands); duckGain = fMult(num, denom); duckGain = fPow2Div2(duckGain << scale); duckGain = fMultDiv2(FL2FXCONST_DUCK(2.f / 3.f), duckGain) << 3; FIXP_DBL *pOutputReal = &outputReal[qs]; FIXP_DBL *pOutputImag = &outputImag[qs]; if (qs < hybBands) { for (; qs < qs_next; qs++) { *pOutputReal = fMult(*pOutputReal, duckGain); pOutputReal++; /* don't move in front of "=" above, because then the fract class treats it differently and provides wrong argument to fMult() (seen on win32/msvc8) */ *pOutputImag = fMult(*pOutputImag, duckGain); pOutputImag++; } } else { for (; qs < qs_next; qs++) { *pOutputReal = fMult(*pOutputReal, duckGain); pOutputReal++; } } } } } /* pb */ self->headroomSmoothDirRevNrg = (SCHAR)fixMax(0, CntLeadingZeros(maxDirRevNrg) - 1); return 0; } INT FDKdecorrelateOpen(HANDLE_DECORR_DEC hDecorrDec, FIXP_DBL *bufferCplx, const INT bufLen) { HANDLE_DECORR_DEC self = hDecorrDec; if (bufLen < (2 * ((825) + (373)))) return 1; /* assign all memory to stateBufferCplx. It is reassigned during * FDKdecorrelateInit() */ self->stateBufferCplx = bufferCplx; self->L_stateBufferCplx = 0; self->delayBufferCplx = NULL; self->L_delayBufferCplx = 0; return 0; } static int distributeBuffer(HANDLE_DECORR_DEC self, const int L_stateBuf, const int L_delayBuf) { /* factor 2 because of complex values */ if ((2 * ((825) + (373))) < 2 * (L_stateBuf + L_delayBuf)) { return 1; } self->L_stateBufferCplx = 2 * L_stateBuf; self->delayBufferCplx = self->stateBufferCplx + 2 * L_stateBuf; self->L_delayBufferCplx = 2 * L_delayBuf; return 0; } INT FDKdecorrelateInit(HANDLE_DECORR_DEC hDecorrDec, const INT nrHybBands, const FDK_DECORR_TYPE decorrType, const FDK_DUCKER_TYPE duckerType, const INT decorrConfig, const INT seed, const INT partiallyComplex, const INT useFractDelay, const INT isLegacyPS, const INT initStatesFlag) { INT errorCode = 0; int i, rb, i_start; int nParamBands = 28; INT offsetStateBuffer = 0; INT offsetDelayBuffer = 0; const UCHAR *REV_bandOffset; const SCHAR *REV_filterOrder; hDecorrDec->partiallyComplex = partiallyComplex; hDecorrDec->numbins = nrHybBands; switch (decorrType) { case DECORR_PS: /* ignore decorrConfig, seed */ if (partiallyComplex) { hDecorrDec->REV_bandOffset = REV_bandOffset_PS_LP; hDecorrDec->REV_delay = REV_delay_PS_LP; errorCode = distributeBuffer(hDecorrDec, (168), (533)); } else { hDecorrDec->REV_bandOffset = REV_bandOffset_PS_HQ; hDecorrDec->REV_delay = REV_delay_PS_HQ; errorCode = distributeBuffer(hDecorrDec, (360), (257)); } hDecorrDec->REV_filterOrder = REV_filterOrder_PS; hDecorrDec->REV_filtType = REV_filtType_PS; /* Initialize ring buffer offsets for PS specific filter implementation. */ for (i = 0; i < (3); i++) hDecorrDec->stateBufferOffset[i] = stateBufferOffsetInit[i]; break; case DECORR_USAC: if (partiallyComplex) return 1; if (seed != 0) return 1; hDecorrDec->REV_bandOffset = REV_bandOffset_MPS_HQ[decorrConfig]; /* reverb band layout is inherited from MPS standard */ hDecorrDec->REV_filterOrder = REV_filterOrder_USAC; hDecorrDec->REV_delay = REV_delay_USAC; if (useFractDelay) { return 1; /* not yet supported */ } else { hDecorrDec->REV_filtType = REV_filtType_MPS; /* the filter types are inherited from MPS standard */ } /* bsDecorrConfig == 1 is worst case */ errorCode = distributeBuffer(hDecorrDec, (509), (643)); break; case DECORR_LD: if (partiallyComplex) return 1; if (useFractDelay) return 1; if (decorrConfig > 2) return 1; if (seed > (MAX_DECORR_SEED_LD - 1)) return 1; if (!(nrHybBands == 64 || nrHybBands == 32)) return 1; /* actually just qmf bands and no hybrid bands */ hDecorrDec->REV_bandOffset = REV_bandOffset_LD[decorrConfig]; hDecorrDec->REV_filterOrder = REV_filterOrder_MPS; /* the filter orders are inherited from MPS standard */ hDecorrDec->REV_delay = REV_delay_MPS; /* the delays in each reverb band are inherited from MPS standard */ hDecorrDec->REV_filtType = REV_filtType_LD; errorCode = distributeBuffer(hDecorrDec, (825), (373)); break; default: return 1; } if (errorCode) { return errorCode; } if (initStatesFlag) { FDKmemclear( hDecorrDec->stateBufferCplx, hDecorrDec->L_stateBufferCplx * sizeof(*hDecorrDec->stateBufferCplx)); FDKmemclear( hDecorrDec->delayBufferCplx, hDecorrDec->L_delayBufferCplx * sizeof(*hDecorrDec->delayBufferCplx)); FDKmemclear(hDecorrDec->reverbBandDelayBufferIndex, sizeof(hDecorrDec->reverbBandDelayBufferIndex)); } REV_bandOffset = hDecorrDec->REV_bandOffset; REV_filterOrder = hDecorrDec->REV_filterOrder; i_start = 0; for (rb = 0; rb < (4); rb++) { int i_stop; i_stop = REV_bandOffset[rb]; if (i_stop <= i_start) { continue; } for (i = i_start; i < i_stop; i++) { switch (decorrType) { case DECORR_PS: errorCode = DecorrFilterInitPS( &hDecorrDec->Filter[i], hDecorrDec->stateBufferCplx, hDecorrDec->delayBufferCplx, &offsetStateBuffer, &offsetDelayBuffer, i, rb, hDecorrDec->REV_delay[rb]); break; default: errorCode = DecorrFilterInit( &hDecorrDec->Filter[i], hDecorrDec->stateBufferCplx, hDecorrDec->delayBufferCplx, &offsetStateBuffer, &offsetDelayBuffer, seed, rb, useFractDelay, hDecorrDec->REV_delay[rb], REV_filterOrder[rb], decorrType); break; } } i_start = i_stop; } /* loop over reverbBands */ if (!(offsetStateBuffer <= hDecorrDec->L_stateBufferCplx) || !(offsetDelayBuffer <= hDecorrDec->L_delayBufferCplx)) { return errorCode = 1; } if (duckerType == DUCKER_AUTOMATIC) { /* Choose correct ducker type according to standards: */ switch (decorrType) { case DECORR_PS: hDecorrDec->ducker.duckerType = DUCKER_PS; if (isLegacyPS) { nParamBands = (20); } else { nParamBands = (28); } break; case DECORR_USAC: hDecorrDec->ducker.duckerType = DUCKER_MPS; nParamBands = (28); break; case DECORR_LD: hDecorrDec->ducker.duckerType = DUCKER_MPS; nParamBands = (23); break; default: return 1; } } errorCode = DuckerInit( &hDecorrDec->ducker, hDecorrDec->numbins, hDecorrDec->partiallyComplex, hDecorrDec->ducker.duckerType, nParamBands, initStatesFlag); return errorCode; } INT FDKdecorrelateClose(HANDLE_DECORR_DEC hDecorrDec) { INT err = 0; if (hDecorrDec == NULL) { return 1; } hDecorrDec->stateBufferCplx = NULL; hDecorrDec->L_stateBufferCplx = 0; hDecorrDec->delayBufferCplx = NULL; hDecorrDec->L_delayBufferCplx = 0; return err; } LNK_SECTION_CODE_L1 INT FDKdecorrelateApply(HANDLE_DECORR_DEC hDecorrDec, FIXP_DBL *dataRealIn, FIXP_DBL *dataImagIn, FIXP_DBL *dataRealOut, FIXP_DBL *dataImagOut, const INT startHybBand) { HANDLE_DECORR_DEC self = hDecorrDec; INT err = 0; INT rb, stop, start; if (self != NULL) { int nHybBands = 0; /* copy new samples */ nHybBands = self->numbins; FIXP_DBL directNrg[(28)]; DuckerCalcEnergy( &self->ducker, dataRealIn, dataImagIn, directNrg, self->ducker.maxValDirectData, &(self->ducker.scaleDirectNrg), (self->ducker.duckerType == DUCKER_PS) ? 1 : 0, startHybBand); /* complex-valued hybrid bands */ for (stop = 0, rb = 0; rb < (4); rb++) { start = fMax(stop, startHybBand); stop = fMin(self->REV_bandOffset[rb], (UCHAR)nHybBands); if (start < stop) { switch (hDecorrDec->REV_filtType[rb]) { case DELAY: err = DecorrFilterApplyPASS(&self->Filter[0], dataRealIn, dataImagIn, dataRealOut, dataImagOut, start, stop, self->REV_delay[rb], self->reverbBandDelayBufferIndex[rb]); break; case INDEP_CPLX_PS: err = DecorrFilterApplyCPLX_PS( &self->Filter[0], dataRealIn, dataImagIn, dataRealOut, dataImagOut, start, stop, self->REV_filterOrder[rb], self->REV_delay[rb], self->reverbBandDelayBufferIndex[rb], self->stateBufferOffset); break; case COMMON_REAL: err = DecorrFilterApplyREAL( &self->Filter[0], dataRealIn, dataImagIn, dataRealOut, dataImagOut, start, stop, self->REV_filterOrder[rb], self->REV_delay[rb], self->reverbBandDelayBufferIndex[rb]); break; default: err = 1; break; } if (err != 0) { goto bail; } } /* if start < stop */ } /* loop over reverb bands */ for (rb = 0; rb < (4); rb++) { self->reverbBandDelayBufferIndex[rb] += 2; if (self->reverbBandDelayBufferIndex[rb] >= 2 * self->REV_delay[rb]) self->reverbBandDelayBufferIndex[rb] = 0; } switch (self->ducker.duckerType) { case DUCKER_PS: err = DuckerApplyPS(&self->ducker, directNrg, dataRealOut, dataImagOut, startHybBand); if (err != 0) goto bail; break; default: err = DuckerApply(&self->ducker, directNrg, dataRealOut, dataImagOut, startHybBand); if (err != 0) goto bail; break; } } bail: return err; } fdk-aac-2.0.2/libFDK/src/FDK_hybrid.cpp000066400000000000000000000770211404225310700173410ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Markus Lohwasser Description: FDK Tools Hybrid Filterbank *******************************************************************************/ #include "FDK_hybrid.h" #include "fft.h" /*--------------- defines -----------------------------*/ #define FFT_IDX_R(a) (2 * a) #define FFT_IDX_I(a) (2 * a + 1) #define HYB_COEF8_0 (0.00746082949812f) #define HYB_COEF8_1 (0.02270420949825f) #define HYB_COEF8_2 (0.04546865930473f) #define HYB_COEF8_3 (0.07266113929591f) #define HYB_COEF8_4 (0.09885108575264f) #define HYB_COEF8_5 (0.11793710567217f) #define HYB_COEF8_6 (0.12500000000000f) #define HYB_COEF8_7 (HYB_COEF8_5) #define HYB_COEF8_8 (HYB_COEF8_4) #define HYB_COEF8_9 (HYB_COEF8_3) #define HYB_COEF8_10 (HYB_COEF8_2) #define HYB_COEF8_11 (HYB_COEF8_1) #define HYB_COEF8_12 (HYB_COEF8_0) /*--------------- structure definitions ---------------*/ #if defined(ARCH_PREFER_MULT_32x16) #define FIXP_HTB FIXP_SGL /* SGL data type. */ #define FIXP_HTP FIXP_SPK /* Packed SGL data type. */ #define HTC(a) (FX_DBL2FXCONST_SGL(a)) /* Cast to SGL */ #define FL2FXCONST_HTB FL2FXCONST_SGL #else #define FIXP_HTB FIXP_DBL /* SGL data type. */ #define FIXP_HTP FIXP_DPK /* Packed DBL data type. */ #define HTC(a) ((FIXP_DBL)(LONG)(a)) /* Cast to DBL */ #define FL2FXCONST_HTB FL2FXCONST_DBL #endif #define HTCP(real, imag) \ { \ { HTC(real), HTC(imag) } \ } /* How to arrange the packed values. */ struct FDK_HYBRID_SETUP { UCHAR nrQmfBands; /*!< Number of QMF bands to be converted to hybrid. */ UCHAR nHybBands[3]; /*!< Number of Hybrid bands generated by nrQmfBands. */ UCHAR synHybScale[3]; /*!< Headroom needed in hybrid synthesis filterbank. */ SCHAR kHybrid[3]; /*!< Filter configuration of each QMF band. */ UCHAR protoLen; /*!< Prototype filter length. */ UCHAR filterDelay; /*!< Delay caused by hybrid filter. */ const INT *pReadIdxTable; /*!< Helper table to access input data ringbuffer. */ }; /*--------------- constants ---------------------------*/ static const INT ringbuffIdxTab[2 * 13] = {0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12}; static const FDK_HYBRID_SETUP setup_3_16 = { 3, {8, 4, 4}, {4, 3, 3}, {8, 4, 4}, 13, (13 - 1) / 2, ringbuffIdxTab}; static const FDK_HYBRID_SETUP setup_3_12 = { 3, {8, 2, 2}, {4, 2, 2}, {8, 2, 2}, 13, (13 - 1) / 2, ringbuffIdxTab}; static const FDK_HYBRID_SETUP setup_3_10 = { 3, {6, 2, 2}, {3, 2, 2}, {-8, -2, 2}, 13, (13 - 1) / 2, ringbuffIdxTab}; static const FIXP_HTP HybFilterCoef8[] = { HTCP(0x10000000, 0x00000000), HTCP(0x0df26407, 0xfa391882), HTCP(0xff532109, 0x00acdef7), HTCP(0x08f26d36, 0xf70d92ca), HTCP(0xfee34b5f, 0x02af570f), HTCP(0x038f276e, 0xf7684793), HTCP(0x00000000, 0x05d1eac2), HTCP(0x00000000, 0x05d1eac2), HTCP(0x038f276e, 0x0897b86d), HTCP(0xfee34b5f, 0xfd50a8f1), HTCP(0x08f26d36, 0x08f26d36), HTCP(0xff532109, 0xff532109), HTCP(0x0df26407, 0x05c6e77e)}; static const FIXP_HTB HybFilterCoef2[3] = {FL2FXCONST_HTB(0.01899487526049f), FL2FXCONST_HTB(-0.07293139167538f), FL2FXCONST_HTB(0.30596630545168f)}; static const FIXP_HTB HybFilterCoef4[13] = {FL2FXCONST_HTB(-0.00305151927305f), FL2FXCONST_HTB(-0.00794862316203f), FL2FXCONST_HTB(0.0f), FL2FXCONST_HTB(0.04318924038756f), FL2FXCONST_HTB(0.12542448210445f), FL2FXCONST_HTB(0.21227807049160f), FL2FXCONST_HTB(0.25f), FL2FXCONST_HTB(0.21227807049160f), FL2FXCONST_HTB(0.12542448210445f), FL2FXCONST_HTB(0.04318924038756f), FL2FXCONST_HTB(0.0f), FL2FXCONST_HTB(-0.00794862316203f), FL2FXCONST_HTB(-0.00305151927305f)}; /*--------------- function declarations ---------------*/ static INT kChannelFiltering(const FIXP_DBL *const pQmfReal, const FIXP_DBL *const pQmfImag, const INT *const pReadIdx, FIXP_DBL *const mHybridReal, FIXP_DBL *const mHybridImag, const SCHAR hybridConfig); /*--------------- function definitions ----------------*/ INT FDKhybridAnalysisOpen(HANDLE_FDK_ANA_HYB_FILTER hAnalysisHybFilter, FIXP_DBL *const pLFmemory, const UINT LFmemorySize, FIXP_DBL *const pHFmemory, const UINT HFmemorySize) { INT err = 0; /* Save pointer to extern memory. */ hAnalysisHybFilter->pLFmemory = pLFmemory; hAnalysisHybFilter->LFmemorySize = LFmemorySize; hAnalysisHybFilter->pHFmemory = pHFmemory; hAnalysisHybFilter->HFmemorySize = HFmemorySize; return err; } INT FDKhybridAnalysisInit(HANDLE_FDK_ANA_HYB_FILTER hAnalysisHybFilter, const FDK_HYBRID_MODE mode, const INT qmfBands, const INT cplxBands, const INT initStatesFlag) { int k; INT err = 0; FIXP_DBL *pMem = NULL; HANDLE_FDK_HYBRID_SETUP setup = NULL; switch (mode) { case THREE_TO_TEN: setup = &setup_3_10; break; case THREE_TO_TWELVE: setup = &setup_3_12; break; case THREE_TO_SIXTEEN: setup = &setup_3_16; break; default: err = -1; goto bail; } /* Initialize handle. */ hAnalysisHybFilter->pSetup = setup; if (initStatesFlag) { hAnalysisHybFilter->bufferLFpos = setup->protoLen - 1; hAnalysisHybFilter->bufferHFpos = 0; } hAnalysisHybFilter->nrBands = qmfBands; hAnalysisHybFilter->cplxBands = cplxBands; hAnalysisHybFilter->hfMode = 0; /* Check available memory. */ if (((2 * setup->nrQmfBands * setup->protoLen * sizeof(FIXP_DBL)) > hAnalysisHybFilter->LFmemorySize)) { err = -2; goto bail; } if (hAnalysisHybFilter->HFmemorySize != 0) { if (((setup->filterDelay * ((qmfBands - setup->nrQmfBands) + (cplxBands - setup->nrQmfBands)) * sizeof(FIXP_DBL)) > hAnalysisHybFilter->HFmemorySize)) { err = -3; goto bail; } } /* Distribute LF memory. */ pMem = hAnalysisHybFilter->pLFmemory; for (k = 0; k < setup->nrQmfBands; k++) { hAnalysisHybFilter->bufferLFReal[k] = pMem; pMem += setup->protoLen; hAnalysisHybFilter->bufferLFImag[k] = pMem; pMem += setup->protoLen; } /* Distribute HF memory. */ if (hAnalysisHybFilter->HFmemorySize != 0) { pMem = hAnalysisHybFilter->pHFmemory; for (k = 0; k < setup->filterDelay; k++) { hAnalysisHybFilter->bufferHFReal[k] = pMem; pMem += (qmfBands - setup->nrQmfBands); hAnalysisHybFilter->bufferHFImag[k] = pMem; pMem += (cplxBands - setup->nrQmfBands); } } if (initStatesFlag) { /* Clear LF buffer */ for (k = 0; k < setup->nrQmfBands; k++) { FDKmemclear(hAnalysisHybFilter->bufferLFReal[k], setup->protoLen * sizeof(FIXP_DBL)); FDKmemclear(hAnalysisHybFilter->bufferLFImag[k], setup->protoLen * sizeof(FIXP_DBL)); } if (hAnalysisHybFilter->HFmemorySize != 0) { if (qmfBands > setup->nrQmfBands) { /* Clear HF buffer */ for (k = 0; k < setup->filterDelay; k++) { FDKmemclear(hAnalysisHybFilter->bufferHFReal[k], (qmfBands - setup->nrQmfBands) * sizeof(FIXP_DBL)); FDKmemclear(hAnalysisHybFilter->bufferHFImag[k], (cplxBands - setup->nrQmfBands) * sizeof(FIXP_DBL)); } } } } bail: return err; } INT FDKhybridAnalysisScaleStates(HANDLE_FDK_ANA_HYB_FILTER hAnalysisHybFilter, const INT scalingValue) { INT err = 0; if (hAnalysisHybFilter == NULL) { err = 1; /* invalid handle */ } else { int k; HANDLE_FDK_HYBRID_SETUP setup = hAnalysisHybFilter->pSetup; /* Scale LF buffer */ for (k = 0; k < setup->nrQmfBands; k++) { scaleValues(hAnalysisHybFilter->bufferLFReal[k], setup->protoLen, scalingValue); scaleValues(hAnalysisHybFilter->bufferLFImag[k], setup->protoLen, scalingValue); } if (hAnalysisHybFilter->nrBands > setup->nrQmfBands) { /* Scale HF buffer */ for (k = 0; k < setup->filterDelay; k++) { scaleValues(hAnalysisHybFilter->bufferHFReal[k], (hAnalysisHybFilter->nrBands - setup->nrQmfBands), scalingValue); scaleValues(hAnalysisHybFilter->bufferHFImag[k], (hAnalysisHybFilter->cplxBands - setup->nrQmfBands), scalingValue); } } } return err; } INT FDKhybridAnalysisApply(HANDLE_FDK_ANA_HYB_FILTER hAnalysisHybFilter, const FIXP_DBL *const pQmfReal, const FIXP_DBL *const pQmfImag, FIXP_DBL *const pHybridReal, FIXP_DBL *const pHybridImag) { int k, hybOffset = 0; INT err = 0; const int nrQmfBandsLF = hAnalysisHybFilter->pSetup ->nrQmfBands; /* number of QMF bands to be converted to hybrid */ const int writIndex = hAnalysisHybFilter->bufferLFpos; int readIndex = hAnalysisHybFilter->bufferLFpos; if (++readIndex >= hAnalysisHybFilter->pSetup->protoLen) readIndex = 0; const INT *pBufferLFreadIdx = &hAnalysisHybFilter->pSetup->pReadIdxTable[readIndex]; /* * LF buffer. */ for (k = 0; k < nrQmfBandsLF; k++) { /* New input sample. */ hAnalysisHybFilter->bufferLFReal[k][writIndex] = pQmfReal[k]; hAnalysisHybFilter->bufferLFImag[k][writIndex] = pQmfImag[k]; /* Perform hybrid filtering. */ err |= kChannelFiltering(hAnalysisHybFilter->bufferLFReal[k], hAnalysisHybFilter->bufferLFImag[k], pBufferLFreadIdx, pHybridReal + hybOffset, pHybridImag + hybOffset, hAnalysisHybFilter->pSetup->kHybrid[k]); hybOffset += hAnalysisHybFilter->pSetup->nHybBands[k]; } hAnalysisHybFilter->bufferLFpos = readIndex; /* Index where to write next input sample. */ if (hAnalysisHybFilter->nrBands > nrQmfBandsLF) { /* * HF buffer. */ if (hAnalysisHybFilter->hfMode != 0) { /* HF delay compensation was applied outside. */ FDKmemcpy( pHybridReal + hybOffset, &pQmfReal[nrQmfBandsLF], (hAnalysisHybFilter->nrBands - nrQmfBandsLF) * sizeof(FIXP_DBL)); FDKmemcpy( pHybridImag + hybOffset, &pQmfImag[nrQmfBandsLF], (hAnalysisHybFilter->cplxBands - nrQmfBandsLF) * sizeof(FIXP_DBL)); } else { FDK_ASSERT(hAnalysisHybFilter->HFmemorySize != 0); /* HF delay compensation, filterlength/2. */ FDKmemcpy( pHybridReal + hybOffset, hAnalysisHybFilter->bufferHFReal[hAnalysisHybFilter->bufferHFpos], (hAnalysisHybFilter->nrBands - nrQmfBandsLF) * sizeof(FIXP_DBL)); FDKmemcpy( pHybridImag + hybOffset, hAnalysisHybFilter->bufferHFImag[hAnalysisHybFilter->bufferHFpos], (hAnalysisHybFilter->cplxBands - nrQmfBandsLF) * sizeof(FIXP_DBL)); FDKmemcpy( hAnalysisHybFilter->bufferHFReal[hAnalysisHybFilter->bufferHFpos], &pQmfReal[nrQmfBandsLF], (hAnalysisHybFilter->nrBands - nrQmfBandsLF) * sizeof(FIXP_DBL)); FDKmemcpy( hAnalysisHybFilter->bufferHFImag[hAnalysisHybFilter->bufferHFpos], &pQmfImag[nrQmfBandsLF], (hAnalysisHybFilter->cplxBands - nrQmfBandsLF) * sizeof(FIXP_DBL)); if (++hAnalysisHybFilter->bufferHFpos >= hAnalysisHybFilter->pSetup->filterDelay) hAnalysisHybFilter->bufferHFpos = 0; } } /* process HF part*/ return err; } INT FDKhybridAnalysisClose(HANDLE_FDK_ANA_HYB_FILTER hAnalysisHybFilter) { INT err = 0; if (hAnalysisHybFilter != NULL) { hAnalysisHybFilter->pLFmemory = NULL; hAnalysisHybFilter->pHFmemory = NULL; hAnalysisHybFilter->LFmemorySize = 0; hAnalysisHybFilter->HFmemorySize = 0; } return err; } INT FDKhybridSynthesisInit(HANDLE_FDK_SYN_HYB_FILTER hSynthesisHybFilter, const FDK_HYBRID_MODE mode, const INT qmfBands, const INT cplxBands) { INT err = 0; HANDLE_FDK_HYBRID_SETUP setup = NULL; switch (mode) { case THREE_TO_TEN: setup = &setup_3_10; break; case THREE_TO_TWELVE: setup = &setup_3_12; break; case THREE_TO_SIXTEEN: setup = &setup_3_16; break; default: err = -1; goto bail; } hSynthesisHybFilter->pSetup = setup; hSynthesisHybFilter->nrBands = qmfBands; hSynthesisHybFilter->cplxBands = cplxBands; bail: return err; } void FDKhybridSynthesisApply(HANDLE_FDK_SYN_HYB_FILTER hSynthesisHybFilter, const FIXP_DBL *const pHybridReal, const FIXP_DBL *const pHybridImag, FIXP_DBL *const pQmfReal, FIXP_DBL *const pQmfImag) { int k, n, hybOffset = 0; const INT nrQmfBandsLF = hSynthesisHybFilter->pSetup->nrQmfBands; /* * LF buffer. */ for (k = 0; k < nrQmfBandsLF; k++) { const int nHybBands = hSynthesisHybFilter->pSetup->nHybBands[k]; const int scale = hSynthesisHybFilter->pSetup->synHybScale[k]; FIXP_DBL accu1 = FL2FXCONST_DBL(0.f); FIXP_DBL accu2 = FL2FXCONST_DBL(0.f); /* Perform hybrid filtering. */ for (n = 0; n < nHybBands; n++) { accu1 += pHybridReal[hybOffset + n] >> scale; accu2 += pHybridImag[hybOffset + n] >> scale; } pQmfReal[k] = SATURATE_LEFT_SHIFT(accu1, scale, DFRACT_BITS); pQmfImag[k] = SATURATE_LEFT_SHIFT(accu2, scale, DFRACT_BITS); hybOffset += nHybBands; } if (hSynthesisHybFilter->nrBands > nrQmfBandsLF) { /* * HF buffer. */ FDKmemcpy(&pQmfReal[nrQmfBandsLF], &pHybridReal[hybOffset], (hSynthesisHybFilter->nrBands - nrQmfBandsLF) * sizeof(FIXP_DBL)); FDKmemcpy( &pQmfImag[nrQmfBandsLF], &pHybridImag[hybOffset], (hSynthesisHybFilter->cplxBands - nrQmfBandsLF) * sizeof(FIXP_DBL)); } return; } static void dualChannelFiltering(const FIXP_DBL *const pQmfReal, const FIXP_DBL *const pQmfImag, const INT *const pReadIdx, FIXP_DBL *const mHybridReal, FIXP_DBL *const mHybridImag, const INT invert) { FIXP_DBL r1, r6; FIXP_DBL i1, i6; const FIXP_HTB f0 = HybFilterCoef2[0]; /* corresponds to p1 and p11 */ const FIXP_HTB f1 = HybFilterCoef2[1]; /* corresponds to p3 and p9 */ const FIXP_HTB f2 = HybFilterCoef2[2]; /* corresponds to p5 and p7 */ /* symmetric filter coefficients */ r1 = fMultDiv2(f0, pQmfReal[pReadIdx[1]]) + fMultDiv2(f0, pQmfReal[pReadIdx[11]]); i1 = fMultDiv2(f0, pQmfImag[pReadIdx[1]]) + fMultDiv2(f0, pQmfImag[pReadIdx[11]]); r1 += fMultDiv2(f1, pQmfReal[pReadIdx[3]]) + fMultDiv2(f1, pQmfReal[pReadIdx[9]]); i1 += fMultDiv2(f1, pQmfImag[pReadIdx[3]]) + fMultDiv2(f1, pQmfImag[pReadIdx[9]]); r1 += fMultDiv2(f2, pQmfReal[pReadIdx[5]]) + fMultDiv2(f2, pQmfReal[pReadIdx[7]]); i1 += fMultDiv2(f2, pQmfImag[pReadIdx[5]]) + fMultDiv2(f2, pQmfImag[pReadIdx[7]]); r6 = pQmfReal[pReadIdx[6]] >> 2; i6 = pQmfImag[pReadIdx[6]] >> 2; FDK_ASSERT((invert == 0) || (invert == 1)); mHybridReal[0 + invert] = (r6 + r1) << 1; mHybridImag[0 + invert] = (i6 + i1) << 1; mHybridReal[1 - invert] = (r6 - r1) << 1; mHybridImag[1 - invert] = (i6 - i1) << 1; } static void fourChannelFiltering(const FIXP_DBL *const pQmfReal, const FIXP_DBL *const pQmfImag, const INT *const pReadIdx, FIXP_DBL *const mHybridReal, FIXP_DBL *const mHybridImag, const INT invert) { const FIXP_HTB *p = HybFilterCoef4; FIXP_DBL fft[8]; static const FIXP_DBL cr[13] = { FL2FXCONST_DBL(0.f), FL2FXCONST_DBL(-0.70710678118655f), FL2FXCONST_DBL(-1.f), FL2FXCONST_DBL(-0.70710678118655f), FL2FXCONST_DBL(0.f), FL2FXCONST_DBL(0.70710678118655f), FL2FXCONST_DBL(1.f), FL2FXCONST_DBL(0.70710678118655f), FL2FXCONST_DBL(0.f), FL2FXCONST_DBL(-0.70710678118655f), FL2FXCONST_DBL(-1.f), FL2FXCONST_DBL(-0.70710678118655f), FL2FXCONST_DBL(0.f)}; static const FIXP_DBL ci[13] = { FL2FXCONST_DBL(-1.f), FL2FXCONST_DBL(-0.70710678118655f), FL2FXCONST_DBL(0.f), FL2FXCONST_DBL(0.70710678118655f), FL2FXCONST_DBL(1.f), FL2FXCONST_DBL(0.70710678118655f), FL2FXCONST_DBL(0.f), FL2FXCONST_DBL(-0.70710678118655f), FL2FXCONST_DBL(-1.f), FL2FXCONST_DBL(-0.70710678118655f), FL2FXCONST_DBL(0.f), FL2FXCONST_DBL(0.70710678118655f), FL2FXCONST_DBL(1.f)}; /* FIR filter. */ /* pre twiddeling with pre-twiddling coefficients c[n] */ /* multiplication with filter coefficients p[n] */ /* hint: (a + ib)*(c + id) = (a*c - b*d) + i(a*d + b*c) */ /* write to fft coefficient n' */ fft[FFT_IDX_R(0)] = (fMult(p[10], (fMultSub(fMultDiv2(cr[2], pQmfReal[pReadIdx[2]]), ci[2], pQmfImag[pReadIdx[2]]))) + fMult(p[6], (fMultSub(fMultDiv2(cr[6], pQmfReal[pReadIdx[6]]), ci[6], pQmfImag[pReadIdx[6]]))) + fMult(p[2], (fMultSub(fMultDiv2(cr[10], pQmfReal[pReadIdx[10]]), ci[10], pQmfImag[pReadIdx[10]])))); fft[FFT_IDX_I(0)] = (fMult(p[10], (fMultAdd(fMultDiv2(ci[2], pQmfReal[pReadIdx[2]]), cr[2], pQmfImag[pReadIdx[2]]))) + fMult(p[6], (fMultAdd(fMultDiv2(ci[6], pQmfReal[pReadIdx[6]]), cr[6], pQmfImag[pReadIdx[6]]))) + fMult(p[2], (fMultAdd(fMultDiv2(ci[10], pQmfReal[pReadIdx[10]]), cr[10], pQmfImag[pReadIdx[10]])))); /* twiddle dee dum */ fft[FFT_IDX_R(1)] = (fMult(p[9], (fMultSub(fMultDiv2(cr[3], pQmfReal[pReadIdx[3]]), ci[3], pQmfImag[pReadIdx[3]]))) + fMult(p[5], (fMultSub(fMultDiv2(cr[7], pQmfReal[pReadIdx[7]]), ci[7], pQmfImag[pReadIdx[7]]))) + fMult(p[1], (fMultSub(fMultDiv2(cr[11], pQmfReal[pReadIdx[11]]), ci[11], pQmfImag[pReadIdx[11]])))); fft[FFT_IDX_I(1)] = (fMult(p[9], (fMultAdd(fMultDiv2(ci[3], pQmfReal[pReadIdx[3]]), cr[3], pQmfImag[pReadIdx[3]]))) + fMult(p[5], (fMultAdd(fMultDiv2(ci[7], pQmfReal[pReadIdx[7]]), cr[7], pQmfImag[pReadIdx[7]]))) + fMult(p[1], (fMultAdd(fMultDiv2(ci[11], pQmfReal[pReadIdx[11]]), cr[11], pQmfImag[pReadIdx[11]])))); /* twiddle dee dee */ fft[FFT_IDX_R(2)] = (fMult(p[12], (fMultSub(fMultDiv2(cr[0], pQmfReal[pReadIdx[0]]), ci[0], pQmfImag[pReadIdx[0]]))) + fMult(p[8], (fMultSub(fMultDiv2(cr[4], pQmfReal[pReadIdx[4]]), ci[4], pQmfImag[pReadIdx[4]]))) + fMult(p[4], (fMultSub(fMultDiv2(cr[8], pQmfReal[pReadIdx[8]]), ci[8], pQmfImag[pReadIdx[8]]))) + fMult(p[0], (fMultSub(fMultDiv2(cr[12], pQmfReal[pReadIdx[12]]), ci[12], pQmfImag[pReadIdx[12]])))); fft[FFT_IDX_I(2)] = (fMult(p[12], (fMultAdd(fMultDiv2(ci[0], pQmfReal[pReadIdx[0]]), cr[0], pQmfImag[pReadIdx[0]]))) + fMult(p[8], (fMultAdd(fMultDiv2(ci[4], pQmfReal[pReadIdx[4]]), cr[4], pQmfImag[pReadIdx[4]]))) + fMult(p[4], (fMultAdd(fMultDiv2(ci[8], pQmfReal[pReadIdx[8]]), cr[8], pQmfImag[pReadIdx[8]]))) + fMult(p[0], (fMultAdd(fMultDiv2(ci[12], pQmfReal[pReadIdx[12]]), cr[12], pQmfImag[pReadIdx[12]])))); fft[FFT_IDX_R(3)] = (fMult(p[11], (fMultSub(fMultDiv2(cr[1], pQmfReal[pReadIdx[1]]), ci[1], pQmfImag[pReadIdx[1]]))) + fMult(p[7], (fMultSub(fMultDiv2(cr[5], pQmfReal[pReadIdx[5]]), ci[5], pQmfImag[pReadIdx[5]]))) + fMult(p[3], (fMultSub(fMultDiv2(cr[9], pQmfReal[pReadIdx[9]]), ci[9], pQmfImag[pReadIdx[9]])))); fft[FFT_IDX_I(3)] = (fMult(p[11], (fMultAdd(fMultDiv2(ci[1], pQmfReal[pReadIdx[1]]), cr[1], pQmfImag[pReadIdx[1]]))) + fMult(p[7], (fMultAdd(fMultDiv2(ci[5], pQmfReal[pReadIdx[5]]), cr[5], pQmfImag[pReadIdx[5]]))) + fMult(p[3], (fMultAdd(fMultDiv2(ci[9], pQmfReal[pReadIdx[9]]), cr[9], pQmfImag[pReadIdx[9]])))); /* fft modulation */ /* here: fast manual fft modulation for a fft of length M=4 */ /* fft_4{x[n]} = x[0]*exp(-i*2*pi/4*m*0) + x[1]*exp(-i*2*pi/4*m*1) + x[2]*exp(-i*2*pi/4*m*2) + x[3]*exp(-i*2*pi/4*m*3) */ /* fft bin m=0: X[0, n] = x[0] + x[1] + x[2] + x[3] */ mHybridReal[0] = fft[FFT_IDX_R(0)] + fft[FFT_IDX_R(1)] + fft[FFT_IDX_R(2)] + fft[FFT_IDX_R(3)]; mHybridImag[0] = fft[FFT_IDX_I(0)] + fft[FFT_IDX_I(1)] + fft[FFT_IDX_I(2)] + fft[FFT_IDX_I(3)]; /* fft bin m=1: X[1, n] = x[0] - i*x[1] - x[2] + i*x[3] */ mHybridReal[1] = fft[FFT_IDX_R(0)] + fft[FFT_IDX_I(1)] - fft[FFT_IDX_R(2)] - fft[FFT_IDX_I(3)]; mHybridImag[1] = fft[FFT_IDX_I(0)] - fft[FFT_IDX_R(1)] - fft[FFT_IDX_I(2)] + fft[FFT_IDX_R(3)]; /* fft bin m=2: X[2, n] = x[0] - x[1] + x[2] - x[3] */ mHybridReal[2] = fft[FFT_IDX_R(0)] - fft[FFT_IDX_R(1)] + fft[FFT_IDX_R(2)] - fft[FFT_IDX_R(3)]; mHybridImag[2] = fft[FFT_IDX_I(0)] - fft[FFT_IDX_I(1)] + fft[FFT_IDX_I(2)] - fft[FFT_IDX_I(3)]; /* fft bin m=3: X[3, n] = x[0] + j*x[1] - x[2] - j*x[3] */ mHybridReal[3] = fft[FFT_IDX_R(0)] - fft[FFT_IDX_I(1)] - fft[FFT_IDX_R(2)] + fft[FFT_IDX_I(3)]; mHybridImag[3] = fft[FFT_IDX_I(0)] + fft[FFT_IDX_R(1)] - fft[FFT_IDX_I(2)] - fft[FFT_IDX_R(3)]; } static void eightChannelFiltering(const FIXP_DBL *const pQmfReal, const FIXP_DBL *const pQmfImag, const INT *const pReadIdx, FIXP_DBL *const mHybridReal, FIXP_DBL *const mHybridImag, const INT invert) { const FIXP_HTP *p = HybFilterCoef8; INT k, sc; FIXP_DBL mfft[16 + ALIGNMENT_DEFAULT]; FIXP_DBL *pfft = (FIXP_DBL *)ALIGN_PTR(mfft); FIXP_DBL accu1, accu2, accu3, accu4; /* pre twiddeling */ pfft[FFT_IDX_R(0)] = pQmfReal[pReadIdx[6]] >> (3 + 1); /* fMultDiv2(p[0].v.re, pQmfReal[pReadIdx[6]]); */ pfft[FFT_IDX_I(0)] = pQmfImag[pReadIdx[6]] >> (3 + 1); /* fMultDiv2(p[0].v.re, pQmfImag[pReadIdx[6]]); */ cplxMultDiv2(&accu1, &accu2, pQmfReal[pReadIdx[7]], pQmfImag[pReadIdx[7]], p[1]); pfft[FFT_IDX_R(1)] = accu1; pfft[FFT_IDX_I(1)] = accu2; cplxMultDiv2(&accu1, &accu2, pQmfReal[pReadIdx[0]], pQmfImag[pReadIdx[0]], p[2]); cplxMultDiv2(&accu3, &accu4, pQmfReal[pReadIdx[8]], pQmfImag[pReadIdx[8]], p[3]); pfft[FFT_IDX_R(2)] = accu1 + accu3; pfft[FFT_IDX_I(2)] = accu2 + accu4; cplxMultDiv2(&accu1, &accu2, pQmfReal[pReadIdx[1]], pQmfImag[pReadIdx[1]], p[4]); cplxMultDiv2(&accu3, &accu4, pQmfReal[pReadIdx[9]], pQmfImag[pReadIdx[9]], p[5]); pfft[FFT_IDX_R(3)] = accu1 + accu3; pfft[FFT_IDX_I(3)] = accu2 + accu4; pfft[FFT_IDX_R(4)] = fMultDiv2(pQmfImag[pReadIdx[10]], p[7].v.im) - fMultDiv2(pQmfImag[pReadIdx[2]], p[6].v.im); pfft[FFT_IDX_I(4)] = fMultDiv2(pQmfReal[pReadIdx[2]], p[6].v.im) - fMultDiv2(pQmfReal[pReadIdx[10]], p[7].v.im); cplxMultDiv2(&accu1, &accu2, pQmfReal[pReadIdx[3]], pQmfImag[pReadIdx[3]], p[8]); cplxMultDiv2(&accu3, &accu4, pQmfReal[pReadIdx[11]], pQmfImag[pReadIdx[11]], p[9]); pfft[FFT_IDX_R(5)] = accu1 + accu3; pfft[FFT_IDX_I(5)] = accu2 + accu4; cplxMultDiv2(&accu1, &accu2, pQmfReal[pReadIdx[4]], pQmfImag[pReadIdx[4]], p[10]); cplxMultDiv2(&accu3, &accu4, pQmfReal[pReadIdx[12]], pQmfImag[pReadIdx[12]], p[11]); pfft[FFT_IDX_R(6)] = accu1 + accu3; pfft[FFT_IDX_I(6)] = accu2 + accu4; cplxMultDiv2(&accu1, &accu2, pQmfReal[pReadIdx[5]], pQmfImag[pReadIdx[5]], p[12]); pfft[FFT_IDX_R(7)] = accu1; pfft[FFT_IDX_I(7)] = accu2; /* fft modulation */ fft_8(pfft); sc = 1 + 2; if (invert) { mHybridReal[0] = pfft[FFT_IDX_R(7)] << sc; mHybridImag[0] = pfft[FFT_IDX_I(7)] << sc; mHybridReal[1] = pfft[FFT_IDX_R(0)] << sc; mHybridImag[1] = pfft[FFT_IDX_I(0)] << sc; mHybridReal[2] = pfft[FFT_IDX_R(6)] << sc; mHybridImag[2] = pfft[FFT_IDX_I(6)] << sc; mHybridReal[3] = pfft[FFT_IDX_R(1)] << sc; mHybridImag[3] = pfft[FFT_IDX_I(1)] << sc; mHybridReal[4] = pfft[FFT_IDX_R(2)] << sc; mHybridReal[4] += pfft[FFT_IDX_R(5)] << sc; mHybridImag[4] = pfft[FFT_IDX_I(2)] << sc; mHybridImag[4] += pfft[FFT_IDX_I(5)] << sc; mHybridReal[5] = pfft[FFT_IDX_R(3)] << sc; mHybridReal[5] += pfft[FFT_IDX_R(4)] << sc; mHybridImag[5] = pfft[FFT_IDX_I(3)] << sc; mHybridImag[5] += pfft[FFT_IDX_I(4)] << sc; } else { for (k = 0; k < 8; k++) { mHybridReal[k] = pfft[FFT_IDX_R(k)] << sc; mHybridImag[k] = pfft[FFT_IDX_I(k)] << sc; } } } static INT kChannelFiltering(const FIXP_DBL *const pQmfReal, const FIXP_DBL *const pQmfImag, const INT *const pReadIdx, FIXP_DBL *const mHybridReal, FIXP_DBL *const mHybridImag, const SCHAR hybridConfig) { INT err = 0; switch (hybridConfig) { case 2: case -2: dualChannelFiltering(pQmfReal, pQmfImag, pReadIdx, mHybridReal, mHybridImag, (hybridConfig < 0) ? 1 : 0); break; case 4: case -4: fourChannelFiltering(pQmfReal, pQmfImag, pReadIdx, mHybridReal, mHybridImag, (hybridConfig < 0) ? 1 : 0); break; case 8: case -8: eightChannelFiltering(pQmfReal, pQmfImag, pReadIdx, mHybridReal, mHybridImag, (hybridConfig < 0) ? 1 : 0); break; default: err = -1; } return err; } fdk-aac-2.0.2/libFDK/src/FDK_lpc.cpp000066400000000000000000000373021404225310700166340ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Manuel Jander Description: LPC related functions *******************************************************************************/ #include "FDK_lpc.h" /* Internal scaling of LPC synthesis to avoid overflow of filte states. This depends on the LPC order, because the LPC order defines the amount of MAC operations. */ static SCHAR order_ld[LPC_MAX_ORDER] = { /* Assume that Synthesis filter output does not clip and filter accu does change no more than 1.0 for each iteration. ceil(0.5*log((1:24))/log(2)) */ 0, 1, 1, 1, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 3, 3, 3, 3, 3, 3, 3, 3}; /* IIRLattice */ #ifndef FUNCTION_CLpc_SynthesisLattice_SGL void CLpc_SynthesisLattice(FIXP_DBL *signal, const int signal_size, const int signal_e, const int signal_e_out, const int inc, const FIXP_SGL *coeff, const int order, FIXP_DBL *state) { int i, j; FIXP_DBL *pSignal; int shift; FDK_ASSERT(order <= LPC_MAX_ORDER); FDK_ASSERT(order > 0); if (inc == -1) pSignal = &signal[signal_size - 1]; else pSignal = &signal[0]; /* tmp = x(k) - K(M)*g(M); for m=M-1:-1:1 tmp = tmp - K(m) * g(m); g(m+1) = g(m) + K(m) * tmp; endfor g(1) = tmp; y(k) = tmp; */ shift = -order_ld[order - 1]; for (i = signal_size; i != 0; i--) { FIXP_DBL *pState = state + order - 1; const FIXP_SGL *pCoeff = coeff + order - 1; FIXP_DBL tmp; tmp = scaleValue(*pSignal, shift + signal_e) - fMultDiv2(*pCoeff--, *pState--); for (j = order - 1; j != 0; j--) { tmp = fMultSubDiv2(tmp, pCoeff[0], pState[0]); pState[1] = pState[0] + (fMultDiv2(*pCoeff--, tmp) << 2); pState--; } *pSignal = scaleValueSaturate(tmp, -shift - signal_e_out); /* exponent of state[] is -1 */ pState[1] = tmp << 1; pSignal += inc; } } #endif #ifndef FUNCTION_CLpc_SynthesisLattice_DBL void CLpc_SynthesisLattice(FIXP_DBL *signal, const int signal_size, const int signal_e, const int signal_e_out, const int inc, const FIXP_DBL *coeff, const int order, FIXP_DBL *state) { int i, j; FIXP_DBL *pSignal; FDK_ASSERT(order <= LPC_MAX_ORDER); FDK_ASSERT(order > 0); if (inc == -1) pSignal = &signal[signal_size - 1]; else pSignal = &signal[0]; FDK_ASSERT(signal_size > 0); for (i = signal_size; i != 0; i--) { FIXP_DBL *pState = state + order - 1; const FIXP_DBL *pCoeff = coeff + order - 1; FIXP_DBL tmp, accu; accu = fMultSubDiv2(scaleValue(*pSignal, signal_e - 1), *pCoeff--, *pState--); tmp = SATURATE_LEFT_SHIFT_ALT(accu, 1, DFRACT_BITS); for (j = order - 1; j != 0; j--) { accu = fMultSubDiv2(tmp >> 1, pCoeff[0], pState[0]); tmp = SATURATE_LEFT_SHIFT_ALT(accu, 1, DFRACT_BITS); accu = fMultAddDiv2(pState[0] >> 1, *pCoeff--, tmp); pState[1] = SATURATE_LEFT_SHIFT_ALT(accu, 1, DFRACT_BITS); pState--; } *pSignal = scaleValue(tmp, -signal_e_out); /* exponent of state[] is 0 */ pState[1] = tmp; pSignal += inc; } } #endif /* LPC_SYNTHESIS_IIR version */ void CLpc_Synthesis(FIXP_DBL *signal, const int signal_size, const int signal_e, const int inc, const FIXP_LPC_TNS *lpcCoeff_m, const int lpcCoeff_e, const int order, FIXP_DBL *state, int *pStateIndex) { int i, j; FIXP_DBL *pSignal; int stateIndex = *pStateIndex; FIXP_LPC_TNS coeff[2 * LPC_MAX_ORDER]; FDKmemcpy(&coeff[0], lpcCoeff_m, order * sizeof(FIXP_LPC_TNS)); FDKmemcpy(&coeff[order], lpcCoeff_m, order * sizeof(FIXP_LPC_TNS)); FDK_ASSERT(order <= LPC_MAX_ORDER); FDK_ASSERT(stateIndex < order); if (inc == -1) pSignal = &signal[signal_size - 1]; else pSignal = &signal[0]; /* y(n) = x(n) - lpc[1]*y(n-1) - ... - lpc[order]*y(n-order) */ for (i = 0; i < signal_size; i++) { FIXP_DBL x; const FIXP_LPC_TNS *pCoeff = coeff + order - stateIndex; x = scaleValue(*pSignal, -(lpcCoeff_e + 1)); for (j = 0; j < order; j++) { x -= fMultDiv2(state[j], pCoeff[j]); } x = SATURATE_SHIFT(x, -lpcCoeff_e - 1, DFRACT_BITS); /* Update states */ stateIndex = ((stateIndex - 1) < 0) ? (order - 1) : (stateIndex - 1); state[stateIndex] = x; *pSignal = scaleValue(x, signal_e); pSignal += inc; } *pStateIndex = stateIndex; } /* default version */ void CLpc_Synthesis(FIXP_DBL *signal, const int signal_size, const int signal_e, const int inc, const FIXP_LPC *lpcCoeff_m, const int lpcCoeff_e, const int order, FIXP_DBL *state, int *pStateIndex) { int i, j; FIXP_DBL *pSignal; int stateIndex = *pStateIndex; FIXP_LPC coeff[2 * LPC_MAX_ORDER]; FDKmemcpy(&coeff[0], lpcCoeff_m, order * sizeof(FIXP_LPC)); FDKmemcpy(&coeff[order], lpcCoeff_m, order * sizeof(FIXP_LPC)); FDK_ASSERT(order <= LPC_MAX_ORDER); FDK_ASSERT(stateIndex < order); if (inc == -1) pSignal = &signal[signal_size - 1]; else pSignal = &signal[0]; /* y(n) = x(n) - lpc[1]*y(n-1) - ... - lpc[order]*y(n-order) */ for (i = 0; i < signal_size; i++) { FIXP_DBL x; const FIXP_LPC *pCoeff = coeff + order - stateIndex; x = scaleValue(*pSignal, -(lpcCoeff_e + 1)); for (j = 0; j < order; j++) { x -= fMultDiv2(state[j], pCoeff[j]); } x = SATURATE_SHIFT(x, -lpcCoeff_e - 1, DFRACT_BITS); /* Update states */ stateIndex = ((stateIndex - 1) < 0) ? (order - 1) : (stateIndex - 1); state[stateIndex] = x; *pSignal = scaleValue(x, signal_e); pSignal += inc; } *pStateIndex = stateIndex; } /* FIR */ void CLpc_Analysis(FIXP_DBL *RESTRICT signal, const int signal_size, const FIXP_LPC lpcCoeff_m[], const int lpcCoeff_e, const int order, FIXP_DBL *RESTRICT filtState, int *filtStateIndex) { int stateIndex; INT i, j, shift = lpcCoeff_e + 1; /* +1, because fMultDiv2 */ FIXP_DBL tmp; if (order <= 0) { return; } if (filtStateIndex != NULL) { stateIndex = *filtStateIndex; } else { stateIndex = 0; } /* keep filter coefficients twice and save memory copy operation in modulo state buffer */ FIXP_LPC coeff[2 * LPC_MAX_ORDER]; FIXP_LPC *pCoeff; FDKmemcpy(&coeff[0], lpcCoeff_m, order * sizeof(FIXP_LPC)); FDKmemcpy(&coeff[order], lpcCoeff_m, order * sizeof(FIXP_LPC)); /* # Analysis filter, obtain residual. for k = 0:BL-1 err(i-BL+k) = a * inputSignal(i-BL+k:-1:i-BL-M+k); endfor */ FDK_ASSERT(shift >= 0); for (j = 0; j < signal_size; j++) { pCoeff = &coeff[(order - stateIndex)]; tmp = signal[j] >> shift; for (i = 0; i < order; i++) { tmp = fMultAddDiv2(tmp, pCoeff[i], filtState[i]); } stateIndex = ((stateIndex - 1) < 0) ? (stateIndex - 1 + order) : (stateIndex - 1); filtState[stateIndex] = signal[j]; signal[j] = tmp << shift; } if (filtStateIndex != NULL) { *filtStateIndex = stateIndex; } } /* For the LPC_SYNTHESIS_IIR version */ INT CLpc_ParcorToLpc(const FIXP_LPC_TNS reflCoeff[], FIXP_LPC_TNS LpcCoeff[], INT numOfCoeff, FIXP_DBL workBuffer[]) { INT i, j; INT shiftval, par2LpcShiftVal = 6; /* 6 should be enough, bec. max(numOfCoeff) = 20 */ FIXP_DBL maxVal = (FIXP_DBL)0; workBuffer[0] = FX_LPC_TNS2FX_DBL(reflCoeff[0]) >> par2LpcShiftVal; for (i = 1; i < numOfCoeff; i++) { for (j = 0; j < i / 2; j++) { FIXP_DBL tmp1, tmp2; tmp1 = workBuffer[j]; tmp2 = workBuffer[i - 1 - j]; workBuffer[j] += fMult(reflCoeff[i], tmp2); workBuffer[i - 1 - j] += fMult(reflCoeff[i], tmp1); } if (i & 1) { workBuffer[j] += fMult(reflCoeff[i], workBuffer[j]); } workBuffer[i] = FX_LPC_TNS2FX_DBL(reflCoeff[i]) >> par2LpcShiftVal; } /* calculate exponent */ for (i = 0; i < numOfCoeff; i++) { maxVal = fMax(maxVal, fAbs(workBuffer[i])); } shiftval = fMin(fNorm(maxVal), par2LpcShiftVal); for (i = 0; i < numOfCoeff; i++) { LpcCoeff[i] = FX_DBL2FX_LPC_TNS(workBuffer[i] << shiftval); } return (par2LpcShiftVal - shiftval); } /* Default version */ INT CLpc_ParcorToLpc(const FIXP_LPC reflCoeff[], FIXP_LPC LpcCoeff[], INT numOfCoeff, FIXP_DBL workBuffer[]) { INT i, j; INT shiftval, par2LpcShiftVal = 6; /* 6 should be enough, bec. max(numOfCoeff) = 20 */ FIXP_DBL maxVal = (FIXP_DBL)0; workBuffer[0] = FX_LPC2FX_DBL(reflCoeff[0]) >> par2LpcShiftVal; for (i = 1; i < numOfCoeff; i++) { for (j = 0; j < i / 2; j++) { FIXP_DBL tmp1, tmp2; tmp1 = workBuffer[j]; tmp2 = workBuffer[i - 1 - j]; workBuffer[j] += fMult(reflCoeff[i], tmp2); workBuffer[i - 1 - j] += fMult(reflCoeff[i], tmp1); } if (i & 1) { workBuffer[j] += fMult(reflCoeff[i], workBuffer[j]); } workBuffer[i] = FX_LPC2FX_DBL(reflCoeff[i]) >> par2LpcShiftVal; } /* calculate exponent */ for (i = 0; i < numOfCoeff; i++) { maxVal = fMax(maxVal, fAbs(workBuffer[i])); } shiftval = fMin(fNorm(maxVal), par2LpcShiftVal); for (i = 0; i < numOfCoeff; i++) { LpcCoeff[i] = FX_DBL2FX_LPC(workBuffer[i] << shiftval); } return (par2LpcShiftVal - shiftval); } void CLpc_AutoToParcor(FIXP_DBL acorr[], const int acorr_e, FIXP_LPC reflCoeff[], const int numOfCoeff, FIXP_DBL *pPredictionGain_m, INT *pPredictionGain_e) { INT i, j, scale = 0; FIXP_DBL parcorWorkBuffer[LPC_MAX_ORDER]; FIXP_DBL *workBuffer = parcorWorkBuffer; FIXP_DBL autoCorr_0 = acorr[0]; FDKmemclear(reflCoeff, numOfCoeff * sizeof(FIXP_LPC)); if (autoCorr_0 == FL2FXCONST_DBL(0.0)) { if (pPredictionGain_m != NULL) { *pPredictionGain_m = FL2FXCONST_DBL(0.5f); *pPredictionGain_e = 1; } return; } FDKmemcpy(workBuffer, acorr + 1, numOfCoeff * sizeof(FIXP_DBL)); for (i = 0; i < numOfCoeff; i++) { LONG sign = ((LONG)workBuffer[0] >> (DFRACT_BITS - 1)); FIXP_DBL tmp = (FIXP_DBL)((LONG)workBuffer[0] ^ sign); /* Check preconditions for division function: num<=denum */ /* For 1st iteration acorr[0] cannot be 0, it is checked before loop */ /* Due to exor operation with "sign", num(=tmp) is greater/equal 0 */ if (acorr[0] < tmp) break; /* tmp = div(num, denum, 16) */ tmp = (FIXP_DBL)((LONG)schur_div(tmp, acorr[0], FRACT_BITS) ^ (~sign)); reflCoeff[i] = FX_DBL2FX_LPC(tmp); for (j = numOfCoeff - i - 1; j >= 0; j--) { FIXP_DBL accu1 = fMult(tmp, acorr[j]); FIXP_DBL accu2 = fMult(tmp, workBuffer[j]); workBuffer[j] += accu1; acorr[j] += accu2; } /* Check preconditions for division function: denum (=acorr[0]) > 0 */ if (acorr[0] == (FIXP_DBL)0) break; workBuffer++; } if (pPredictionGain_m != NULL) { if (acorr[0] > (FIXP_DBL)0) { /* prediction gain = signal power / error (residual) power */ *pPredictionGain_m = fDivNormSigned(autoCorr_0, acorr[0], &scale); *pPredictionGain_e = scale; } else { *pPredictionGain_m = (FIXP_DBL)0; *pPredictionGain_e = 0; } } } fdk-aac-2.0.2/libFDK/src/FDK_matrixCalloc.cpp000066400000000000000000000223441404225310700205000ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Description: matrix memory allocation *******************************************************************************/ #include "FDK_matrixCalloc.h" #include "genericStds.h" void *fdkCallocMatrix1D_aligned(UINT dim1, UINT size) { return FDKaalloc(dim1 * size, ALIGNMENT_DEFAULT); } void *fdkCallocMatrix1D_int(UINT dim, UINT size, MEMORY_SECTION s) { return FDKcalloc_L(dim, size, s); } void *fdkCallocMatrix1D_int_aligned(UINT dim, UINT size, MEMORY_SECTION s) { return FDKaalloc_L(dim * size, ALIGNMENT_DEFAULT, s); } void fdkFreeMatrix1D(void *p) { if (p != NULL) { FDKfree_L(p); } } void fdkFreeMatrix1D_aligned(void *p) { if (p != NULL) { FDKafree_L(p); } } void *fdkCallocMatrix1D(UINT dim1, UINT size) { return FDKcalloc(dim1, size); } /* 2D */ void **fdkCallocMatrix2D(UINT dim1, UINT dim2, UINT size) { void **p1; UINT i; char *p2; if (!dim1 || !dim2) return NULL; if ((p1 = (void **)fdkCallocMatrix1D(dim1, sizeof(void *))) == NULL) { goto bail; } if ((p2 = (char *)fdkCallocMatrix1D(dim1 * dim2, size)) == NULL) { fdkFreeMatrix1D(p1); p1 = NULL; goto bail; } for (i = 0; i < dim1; i++) { p1[i] = p2; p2 += dim2 * size; } bail: return p1; } void **fdkCallocMatrix2D_aligned(UINT dim1, UINT dim2, UINT size) { void **p1; UINT i; char *p2; if (!dim1 || !dim2) return NULL; if ((p1 = (void **)fdkCallocMatrix1D(dim1, sizeof(void *))) == NULL) { goto bail; } if ((p2 = (char *)fdkCallocMatrix1D_aligned(dim1 * dim2, size)) == NULL) { fdkFreeMatrix1D(p1); p1 = NULL; goto bail; } for (i = 0; i < dim1; i++) { p1[i] = p2; p2 += dim2 * size; } bail: return p1; } void fdkFreeMatrix2D(void **p) { if (!p) return; fdkFreeMatrix1D(p[0]); fdkFreeMatrix1D(p); } void fdkFreeMatrix2D_aligned(void **p) { if (!p) return; fdkFreeMatrix1D_aligned(p[0]); fdkFreeMatrix1D(p); } void **fdkCallocMatrix2D_int(UINT dim1, UINT dim2, UINT size, MEMORY_SECTION s) { void **p1; UINT i; char *p2; if (!dim1 || !dim2) return NULL; if ((p1 = (void **)fdkCallocMatrix1D_int(dim1, sizeof(void *), s)) == NULL) { goto bail; } if ((p2 = (char *)fdkCallocMatrix1D_int(dim1 * dim2, size, s)) == NULL) { fdkFreeMatrix1D(p1); p1 = NULL; goto bail; } for (i = 0; i < dim1; i++) { p1[i] = p2; p2 += dim2 * size; } bail: return p1; } void **fdkCallocMatrix2D_int_aligned(UINT dim1, UINT dim2, UINT size, MEMORY_SECTION s) { void **p1; UINT i; char *p2; if (!dim1 || !dim2) return NULL; if ((p1 = (void **)fdkCallocMatrix1D_int(dim1, sizeof(void *), s)) == NULL) { goto bail; } if ((p2 = (char *)fdkCallocMatrix1D_int_aligned(dim1 * dim2, size, s)) == NULL) { fdkFreeMatrix1D(p1); p1 = NULL; goto bail; } for (i = 0; i < dim1; i++) { p1[i] = p2; p2 += dim2 * size; } bail: return p1; } /* 3D */ void ***fdkCallocMatrix3D(UINT dim1, UINT dim2, UINT dim3, UINT size) { void ***p1; UINT i, j; void **p2; char *p3; if (!dim1 || !dim2 || !dim3) return NULL; if ((p1 = (void ***)fdkCallocMatrix1D(dim1, sizeof(void **))) == NULL) { goto bail; } if ((p2 = (void **)fdkCallocMatrix1D(dim1 * dim2, sizeof(void *))) == NULL) { fdkFreeMatrix1D(p1); p1 = NULL; goto bail; } p1[0] = p2; if ((p3 = (char *)fdkCallocMatrix1D(dim1 * dim2 * dim3, size)) == NULL) { fdkFreeMatrix1D(p1); fdkFreeMatrix1D(p2); p1 = NULL; p2 = NULL; goto bail; } for (i = 0; i < dim1; i++) { p1[i] = p2; for (j = 0; j < dim2; j++) { p2[j] = p3; p3 += dim3 * size; } p2 += dim2; } bail: return p1; } void fdkFreeMatrix3D(void ***p) { if (!p) return; if (p[0] != NULL) fdkFreeMatrix1D(p[0][0]); fdkFreeMatrix1D(p[0]); fdkFreeMatrix1D(p); } void ***fdkCallocMatrix3D_int(UINT dim1, UINT dim2, UINT dim3, UINT size, MEMORY_SECTION s) { void ***p1; UINT i, j; void **p2; char *p3; if (!dim1 || !dim2 || !dim3) return NULL; if ((p1 = (void ***)fdkCallocMatrix1D_int(dim1, sizeof(void **), s)) == NULL) { goto bail; } if ((p2 = (void **)fdkCallocMatrix1D_int(dim1 * dim2, sizeof(void *), s)) == NULL) { fdkFreeMatrix1D(p1); p1 = NULL; goto bail; } p1[0] = p2; if ((p3 = (char *)fdkCallocMatrix1D_int(dim1 * dim2 * dim3, size, s)) == NULL) { fdkFreeMatrix1D(p1); fdkFreeMatrix1D(p2); p1 = NULL; p2 = NULL; goto bail; } for (i = 0; i < dim1; i++) { p1[i] = p2; for (j = 0; j < dim2; j++) { p2[j] = p3; p3 += dim3 * size; } p2 += dim2; } bail: return p1; } fdk-aac-2.0.2/libFDK/src/FDK_qmf_domain.cpp000066400000000000000000001134031404225310700201650ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Matthias Hildenbrand Description: Module to efficiently handle QMF data for multiple channels and to share the data between e.g. SBR and MPS *******************************************************************************/ #include "FDK_qmf_domain.h" #include "common_fix.h" #define WORKBUFFER1_TAG 0 #define WORKBUFFER3_TAG 4 #define WORKBUFFER4_TAG 5 #define WORKBUFFER6_TAG 7 #define WORKBUFFER7_TAG 8 C_ALLOC_MEM_OVERLAY(QmfWorkBufferCore1, FIXP_DBL, QMF_WB_SECTION_SIZE, SECT_DATA_L1, WORKBUFFER1_TAG) C_ALLOC_MEM_OVERLAY(QmfWorkBufferCore3, FIXP_DBL, QMF_WB_SECTION_SIZE, SECT_DATA_L2, WORKBUFFER3_TAG) C_ALLOC_MEM_OVERLAY(QmfWorkBufferCore4, FIXP_DBL, QMF_WB_SECTION_SIZE, SECT_DATA_L2, WORKBUFFER4_TAG) C_ALLOC_MEM_OVERLAY(QmfWorkBufferCore6, FIXP_DBL, QMF_WB_SECTION_SIZE, SECT_DATA_L2, WORKBUFFER6_TAG) C_ALLOC_MEM_OVERLAY(QmfWorkBufferCore7, FIXP_DBL, QMF_WB_SECTION_SIZE, SECT_DATA_L2, WORKBUFFER7_TAG) /*! Analysis states buffer.
Dimension: #((8) + (1)) */ C_AALLOC_MEM2(AnaQmfStates, FIXP_DBL, 10 * QMF_DOMAIN_MAX_ANALYSIS_QMF_BANDS, ((8) + (1))) /*! Synthesis states buffer.
Dimension: #((8) + (1)) */ C_AALLOC_MEM2(SynQmfStates, FIXP_QSS, 9 * QMF_DOMAIN_MAX_SYNTHESIS_QMF_BANDS, ((8) + (1))) /*! Pointer to real qmf data for each time slot.
Dimension: #((8) + (1)) */ C_ALLOC_MEM2(QmfSlotsReal, FIXP_DBL *, QMF_DOMAIN_MAX_TIMESLOTS + QMF_DOMAIN_MAX_OV_TIMESLOTS, ((8) + (1))) /*! Pointer to imaginary qmf data for each time slot.
Dimension: #((8) + (1)) */ C_ALLOC_MEM2(QmfSlotsImag, FIXP_DBL *, QMF_DOMAIN_MAX_TIMESLOTS + QMF_DOMAIN_MAX_OV_TIMESLOTS, ((8) + (1))) /*! QMF overlap buffer.
Dimension: #((8) + (1)) */ C_AALLOC_MEM2(QmfOverlapBuffer, FIXP_DBL, 2 * QMF_DOMAIN_MAX_OV_TIMESLOTS * QMF_DOMAIN_MAX_QMF_PROC_BANDS, ((8) + (1))) /*! Analysis states buffer.
Dimension: #((8) + (1)) */ C_AALLOC_MEM2(AnaQmfStates16, FIXP_DBL, 10 * QMF_DOMAIN_ANALYSIS_QMF_BANDS_16, ((8) + (1))) /*! Analysis states buffer.
Dimension: #((8) + (1)) */ C_AALLOC_MEM2(AnaQmfStates24, FIXP_DBL, 10 * QMF_DOMAIN_ANALYSIS_QMF_BANDS_24, ((8) + (1))) /*! Analysis states buffer.
Dimension: #((8) + (1)) */ C_AALLOC_MEM2(AnaQmfStates32, FIXP_DBL, 10 * QMF_DOMAIN_ANALYSIS_QMF_BANDS_32, ((8) + (1))) /*! Pointer to real qmf data for each time slot.
Dimension: #((8) + (1)) */ C_ALLOC_MEM2(QmfSlotsReal16, FIXP_DBL *, QMF_DOMAIN_TIMESLOTS_16 + QMF_DOMAIN_OV_TIMESLOTS_16, ((8) + (1))) /*! Pointer to real qmf data for each time slot.
Dimension: #((8) + (1)) */ C_ALLOC_MEM2(QmfSlotsReal32, FIXP_DBL *, QMF_DOMAIN_TIMESLOTS_32 + QMF_DOMAIN_OV_TIMESLOTS_32, ((8) + (1))) /*! Pointer to imaginary qmf data for each time slot.
Dimension: #((8) + (1)) */ C_ALLOC_MEM2(QmfSlotsImag16, FIXP_DBL *, QMF_DOMAIN_TIMESLOTS_16 + QMF_DOMAIN_OV_TIMESLOTS_16, ((8) + (1))) /*! Pointer to imaginary qmf data for each time slot.
Dimension: #((8) + (1)) */ C_ALLOC_MEM2(QmfSlotsImag32, FIXP_DBL *, QMF_DOMAIN_TIMESLOTS_32 + QMF_DOMAIN_OV_TIMESLOTS_32, ((8) + (1))) /*! QMF overlap buffer.
Dimension: #((8) + (1)) */ C_AALLOC_MEM2(QmfOverlapBuffer16, FIXP_DBL, 2 * QMF_DOMAIN_OV_TIMESLOTS_16 * QMF_DOMAIN_MAX_QMF_PROC_BANDS, ((8) + (1))) /*! QMF overlap buffer.
Dimension: #((8) + (1)) */ C_AALLOC_MEM2(QmfOverlapBuffer32, FIXP_DBL, 2 * QMF_DOMAIN_OV_TIMESLOTS_32 * QMF_DOMAIN_MAX_QMF_PROC_BANDS, ((8) + (1))) static int FDK_QmfDomain_FreePersistentMemory(HANDLE_FDK_QMF_DOMAIN qd) { int err = 0; int ch; for (ch = 0; ch < ((8) + (1)); ch++) { if (qd->QmfDomainIn[ch].pAnaQmfStates) { if (qd->globalConf.nBandsAnalysis == QMF_DOMAIN_ANALYSIS_QMF_BANDS_16) { FreeAnaQmfStates16(&qd->QmfDomainIn[ch].pAnaQmfStates); } else if (qd->globalConf.nBandsAnalysis == QMF_DOMAIN_ANALYSIS_QMF_BANDS_24) { FreeAnaQmfStates24(&qd->QmfDomainIn[ch].pAnaQmfStates); } else if (qd->globalConf.nBandsAnalysis == QMF_DOMAIN_ANALYSIS_QMF_BANDS_32) { FreeAnaQmfStates32(&qd->QmfDomainIn[ch].pAnaQmfStates); } else { FreeAnaQmfStates(&qd->QmfDomainIn[ch].pAnaQmfStates); } } if (qd->QmfDomainIn[ch].pOverlapBuffer) { if (qd->globalConf.nQmfOvTimeSlots == QMF_DOMAIN_OV_TIMESLOTS_16) { FreeQmfOverlapBuffer16(&qd->QmfDomainIn[ch].pOverlapBuffer); } else if (qd->globalConf.nQmfOvTimeSlots == QMF_DOMAIN_OV_TIMESLOTS_32) { FreeQmfOverlapBuffer32(&qd->QmfDomainIn[ch].pOverlapBuffer); } else { FreeQmfOverlapBuffer(&qd->QmfDomainIn[ch].pOverlapBuffer); } } if (qd->QmfDomainIn[ch].hQmfSlotsReal) { if (qd->globalConf.nQmfTimeSlots == QMF_DOMAIN_TIMESLOTS_16) { FreeQmfSlotsReal16(&qd->QmfDomainIn[ch].hQmfSlotsReal); } else if (qd->globalConf.nQmfTimeSlots == QMF_DOMAIN_TIMESLOTS_32) { FreeQmfSlotsReal32(&qd->QmfDomainIn[ch].hQmfSlotsReal); } else { FreeQmfSlotsReal(&qd->QmfDomainIn[ch].hQmfSlotsReal); } } if (qd->QmfDomainIn[ch].hQmfSlotsImag) { if (qd->globalConf.nQmfTimeSlots == QMF_DOMAIN_TIMESLOTS_16) { FreeQmfSlotsImag16(&qd->QmfDomainIn[ch].hQmfSlotsImag); } if (qd->globalConf.nQmfTimeSlots == QMF_DOMAIN_TIMESLOTS_32) { FreeQmfSlotsImag32(&qd->QmfDomainIn[ch].hQmfSlotsImag); } else { FreeQmfSlotsImag(&qd->QmfDomainIn[ch].hQmfSlotsImag); } } } for (ch = 0; ch < ((8) + (1)); ch++) { if (qd->QmfDomainOut[ch].pSynQmfStates) { FreeSynQmfStates(&qd->QmfDomainOut[ch].pSynQmfStates); } } return err; } static int FDK_QmfDomain_AllocatePersistentMemory(HANDLE_FDK_QMF_DOMAIN qd) { int err = 0; int ch; HANDLE_FDK_QMF_DOMAIN_GC gc = &qd->globalConf; if ((gc->nInputChannels > ((8) + (1))) || (gc->nOutputChannels > ((8) + (1)))) return err = 1; for (ch = 0; ch < gc->nInputChannels; ch++) { int size; size = gc->nBandsAnalysis * 10; if (size > 0) { if (gc->nBandsAnalysis == QMF_DOMAIN_ANALYSIS_QMF_BANDS_16) { if (qd->QmfDomainIn[ch].pAnaQmfStates == NULL) { if (NULL == (qd->QmfDomainIn[ch].pAnaQmfStates = GetAnaQmfStates16(ch))) goto bail; } } else if (gc->nBandsAnalysis == QMF_DOMAIN_ANALYSIS_QMF_BANDS_24) { if (qd->QmfDomainIn[ch].pAnaQmfStates == NULL) { if (NULL == (qd->QmfDomainIn[ch].pAnaQmfStates = GetAnaQmfStates24(ch))) goto bail; } } else if (gc->nBandsAnalysis == QMF_DOMAIN_ANALYSIS_QMF_BANDS_32) { if (qd->QmfDomainIn[ch].pAnaQmfStates == NULL) { if (NULL == (qd->QmfDomainIn[ch].pAnaQmfStates = GetAnaQmfStates32(ch))) goto bail; } } else { if (qd->QmfDomainIn[ch].pAnaQmfStates == NULL) { if (NULL == (qd->QmfDomainIn[ch].pAnaQmfStates = GetAnaQmfStates(ch))) goto bail; } } } else { qd->QmfDomainIn[ch].pAnaQmfStates = NULL; } size = gc->nQmfOvTimeSlots + gc->nQmfTimeSlots; if (size > 0) { if (gc->nQmfTimeSlots == QMF_DOMAIN_TIMESLOTS_16) { if (qd->QmfDomainIn[ch].hQmfSlotsReal == NULL) { if (NULL == (qd->QmfDomainIn[ch].hQmfSlotsReal = GetQmfSlotsReal16(ch))) goto bail; } if (qd->QmfDomainIn[ch].hQmfSlotsImag == NULL) { if (NULL == (qd->QmfDomainIn[ch].hQmfSlotsImag = GetQmfSlotsImag16(ch))) goto bail; } } else if (gc->nQmfTimeSlots == QMF_DOMAIN_TIMESLOTS_32) { if (qd->QmfDomainIn[ch].hQmfSlotsReal == NULL) { if (NULL == (qd->QmfDomainIn[ch].hQmfSlotsReal = GetQmfSlotsReal32(ch))) goto bail; } if (qd->QmfDomainIn[ch].hQmfSlotsImag == NULL) { if (NULL == (qd->QmfDomainIn[ch].hQmfSlotsImag = GetQmfSlotsImag32(ch))) goto bail; } } else { if (qd->QmfDomainIn[ch].hQmfSlotsReal == NULL) { if (NULL == (qd->QmfDomainIn[ch].hQmfSlotsReal = GetQmfSlotsReal(ch))) goto bail; } if (qd->QmfDomainIn[ch].hQmfSlotsImag == NULL) { if (NULL == (qd->QmfDomainIn[ch].hQmfSlotsImag = GetQmfSlotsImag(ch))) goto bail; } } } else { qd->QmfDomainIn[ch].hQmfSlotsReal = NULL; qd->QmfDomainIn[ch].hQmfSlotsImag = NULL; } size = gc->nQmfOvTimeSlots * gc->nQmfProcBands * CMPLX_MOD; if (size > 0) { if (gc->nQmfOvTimeSlots == QMF_DOMAIN_OV_TIMESLOTS_16) { if (qd->QmfDomainIn[ch].pOverlapBuffer == NULL) { if (NULL == (qd->QmfDomainIn[ch].pOverlapBuffer = GetQmfOverlapBuffer16(ch))) goto bail; } } else if (gc->nQmfOvTimeSlots == QMF_DOMAIN_OV_TIMESLOTS_32) { if (qd->QmfDomainIn[ch].pOverlapBuffer == NULL) { if (NULL == (qd->QmfDomainIn[ch].pOverlapBuffer = GetQmfOverlapBuffer32(ch))) goto bail; } } else { if (qd->QmfDomainIn[ch].pOverlapBuffer == NULL) { if (NULL == (qd->QmfDomainIn[ch].pOverlapBuffer = GetQmfOverlapBuffer(ch))) goto bail; } } } else { qd->QmfDomainIn[ch].pOverlapBuffer = NULL; } } for (ch = 0; ch < gc->nOutputChannels; ch++) { int size = gc->nBandsSynthesis * 9; if (size > 0) { if (qd->QmfDomainOut[ch].pSynQmfStates == NULL) { if (NULL == (qd->QmfDomainOut[ch].pSynQmfStates = GetSynQmfStates(ch))) goto bail; } } else { qd->QmfDomainOut[ch].pSynQmfStates = NULL; } } return err; bail: FDK_QmfDomain_FreePersistentMemory(qd); return -1; } QMF_DOMAIN_ERROR FDK_QmfDomain_ClearPersistentMemory( HANDLE_FDK_QMF_DOMAIN hqd) { QMF_DOMAIN_ERROR err = QMF_DOMAIN_OK; int ch, size; if (hqd) { HANDLE_FDK_QMF_DOMAIN_GC gc = &hqd->globalConf; size = gc->nQmfOvTimeSlots * gc->nQmfProcBands * CMPLX_MOD; for (ch = 0; ch < gc->nInputChannels; ch++) { if (hqd->QmfDomainIn[ch].pOverlapBuffer) { FDKmemclear(hqd->QmfDomainIn[ch].pOverlapBuffer, size * sizeof(FIXP_DBL)); } } if (FDK_QmfDomain_InitFilterBank(hqd, 0)) { err = QMF_DOMAIN_INIT_ERROR; } } else { err = QMF_DOMAIN_INIT_ERROR; } return err; } /* FDK_getWorkBuffer Parameters: pWorkBuffer i: array of pointers which point to different workbuffer sections workBufferOffset i: offset in the workbuffer to the requested memory memSize i: size of requested memory Function: The functions returns the address to the requested memory in the workbuffer. The overall workbuffer is divided into several sections. There are QMF_MAX_WB_SECTIONS sections of size QMF_WB_SECTION_SIZE. The function selects the workbuffer section with the help of the workBufferOffset and than it verifies whether the requested amount of memory fits into the selected workbuffer section. Returns: address to workbuffer */ static FIXP_DBL *FDK_getWorkBuffer(FIXP_DBL **pWorkBuffer, USHORT workBufferOffset, USHORT workBufferSectSize, USHORT memSize) { int idx1; int idx2; FIXP_DBL *pwb; /* a section must be a multiple of the number of processing bands (currently * always 64) */ FDK_ASSERT((workBufferSectSize % 64) == 0); /* calculate offset within the section */ idx2 = workBufferOffset % workBufferSectSize; /* calculate section number */ idx1 = (workBufferOffset - idx2) / workBufferSectSize; /* maximum sectionnumber is QMF_MAX_WB_SECTIONS */ FDK_ASSERT(idx1 < QMF_MAX_WB_SECTIONS); /* check, whether workbuffer is available */ FDK_ASSERT(pWorkBuffer[idx1] != NULL); /* check, whether buffer fits into selected section */ FDK_ASSERT((idx2 + memSize) <= workBufferSectSize); /* get requested address to workbuffer */ pwb = &pWorkBuffer[idx1][idx2]; return pwb; } static int FDK_QmfDomain_FeedWorkBuffer(HANDLE_FDK_QMF_DOMAIN qd, int ch, FIXP_DBL **pWorkBuffer, USHORT workBufferOffset, USHORT workBufferSectSize, int size) { int err = 0; int mem_needed; mem_needed = qd->QmfDomainIn[ch].workBuf_nBands * qd->QmfDomainIn[ch].workBuf_nTimeSlots * CMPLX_MOD; if (mem_needed > size) { return (err = 1); } qd->QmfDomainIn[ch].pWorkBuffer = pWorkBuffer; qd->QmfDomainIn[ch].workBufferOffset = workBufferOffset; qd->QmfDomainIn[ch].workBufferSectSize = workBufferSectSize; return err; } int FDK_QmfDomain_IsInitialized(const HANDLE_FDK_QMF_DOMAIN qd) { FDK_ASSERT(qd != NULL); return ((qd->QmfDomainIn[0].pAnaQmfStates == NULL) && (qd->QmfDomainOut[0].pSynQmfStates == NULL)) ? 0 : 1; } int FDK_QmfDomain_InitFilterBank(HANDLE_FDK_QMF_DOMAIN qd, UINT extra_flags) { FDK_ASSERT(qd != NULL); int err = 0; int ch, ts; HANDLE_FDK_QMF_DOMAIN_GC gc = &qd->globalConf; int noCols = gc->nQmfTimeSlots; int lsb = gc->nBandsAnalysis; int usb = fMin((INT)gc->nBandsSynthesis, 64); int nProcBands = gc->nQmfProcBands; FDK_ASSERT(nProcBands % ALIGNMENT_DEFAULT == 0); if (extra_flags & QMF_FLAG_MPSLDFB) { gc->flags &= ~QMF_FLAG_CLDFB; gc->flags |= QMF_FLAG_MPSLDFB; } for (ch = 0; ch < gc->nInputChannels; ch++) { /* distribute memory to slots array */ FIXP_DBL *ptrOv = qd->QmfDomainIn[ch].pOverlapBuffer; /* persistent memory for overlap */ if ((ptrOv == NULL) && (gc->nQmfOvTimeSlots != 0)) { err = 1; return err; } /* This assumes the workbuffer defined for ch0 is the big one being used to * hold one full frame of QMF data. */ FIXP_DBL **ptr = qd->QmfDomainIn[fMin(ch, fMax((INT)gc->nQmfProcChannels - 1, 0))] .pWorkBuffer; /* non-persistent workbuffer */ USHORT workBufferOffset = qd->QmfDomainIn[fMin(ch, fMax((INT)gc->nQmfProcChannels - 1, 0))] .workBufferOffset; USHORT workBufferSectSize = qd->QmfDomainIn[fMin(ch, fMax((INT)gc->nQmfProcChannels - 1, 0))] .workBufferSectSize; if ((ptr == NULL) && (gc->nQmfTimeSlots != 0)) { err = 1; return err; } qd->QmfDomainIn[ch].pGlobalConf = gc; for (ts = 0; ts < gc->nQmfOvTimeSlots; ts++) { qd->QmfDomainIn[ch].hQmfSlotsReal[ts] = ptrOv; ptrOv += nProcBands; qd->QmfDomainIn[ch].hQmfSlotsImag[ts] = ptrOv; ptrOv += nProcBands; } for (; ts < (gc->nQmfOvTimeSlots + gc->nQmfTimeSlots); ts++) { qd->QmfDomainIn[ch].hQmfSlotsReal[ts] = FDK_getWorkBuffer( ptr, workBufferOffset, workBufferSectSize, nProcBands); workBufferOffset += nProcBands; qd->QmfDomainIn[ch].hQmfSlotsImag[ts] = FDK_getWorkBuffer( ptr, workBufferOffset, workBufferSectSize, nProcBands); workBufferOffset += nProcBands; } err |= qmfInitAnalysisFilterBank( &qd->QmfDomainIn[ch].fb, qd->QmfDomainIn[ch].pAnaQmfStates, noCols, (qd->QmfDomainIn[ch].fb.lsb == 0) ? lsb : qd->QmfDomainIn[ch].fb.lsb, (qd->QmfDomainIn[ch].fb.usb == 0) ? usb : qd->QmfDomainIn[ch].fb.usb, gc->nBandsAnalysis, gc->flags | extra_flags); } for (ch = 0; ch < gc->nOutputChannels; ch++) { FIXP_DBL outGain_m = qd->QmfDomainOut[ch].fb.outGain_m; int outGain_e = qd->QmfDomainOut[ch].fb.outGain_e; int outScale = qmfGetOutScalefactor(&qd->QmfDomainOut[ch].fb); err |= qmfInitSynthesisFilterBank( &qd->QmfDomainOut[ch].fb, qd->QmfDomainOut[ch].pSynQmfStates, noCols, (qd->QmfDomainOut[ch].fb.lsb == 0) ? lsb : qd->QmfDomainOut[ch].fb.lsb, (qd->QmfDomainOut[ch].fb.usb == 0) ? usb : qd->QmfDomainOut[ch].fb.usb, gc->nBandsSynthesis, gc->flags | extra_flags); if (outGain_m != (FIXP_DBL)0) { qmfChangeOutGain(&qd->QmfDomainOut[ch].fb, outGain_m, outGain_e); } if (outScale) { qmfChangeOutScalefactor(&qd->QmfDomainOut[ch].fb, outScale); } } return err; } void FDK_QmfDomain_SaveOverlap(HANDLE_FDK_QMF_DOMAIN_IN qd_ch, int offset) { FDK_ASSERT(qd_ch != NULL); int ts; HANDLE_FDK_QMF_DOMAIN_GC gc = qd_ch->pGlobalConf; int ovSlots = gc->nQmfOvTimeSlots; int nCols = gc->nQmfTimeSlots; int nProcBands = gc->nQmfProcBands; FIXP_DBL **qmfReal = qd_ch->hQmfSlotsReal; FIXP_DBL **qmfImag = qd_ch->hQmfSlotsImag; QMF_SCALE_FACTOR *pScaling = &qd_ch->scaling; /* for high part it would be enough to save only used part of overlap area */ if (qmfImag != NULL) { for (ts = offset; ts < ovSlots; ts++) { FDKmemcpy(qmfReal[ts], qmfReal[nCols + ts], sizeof(FIXP_DBL) * nProcBands); FDKmemcpy(qmfImag[ts], qmfImag[nCols + ts], sizeof(FIXP_DBL) * nProcBands); } } else { for (ts = 0; ts < ovSlots; ts++) { FDKmemcpy(qmfReal[ts], qmfReal[nCols + ts], sizeof(FIXP_DBL) * nProcBands); } } pScaling->ov_lb_scale = pScaling->lb_scale; } /* Convert headroom bits to exponent */ #define SCALE2EXP(s) (15 - (s)) #define EXP2SCALE(e) (15 - (e)) void FDK_QmfDomain_GetSlot(const HANDLE_FDK_QMF_DOMAIN_IN qd_ch, const int ts, const int start_band, const int stop_band, FIXP_DBL *pQmfOutReal, FIXP_DBL *pQmfOutImag, const int exp_out) { FDK_ASSERT(qd_ch != NULL); FDK_ASSERT(pQmfOutReal != NULL); HANDLE_FDK_QMF_DOMAIN_GC gc = qd_ch->pGlobalConf; const FIXP_DBL *real = qd_ch->hQmfSlotsReal[ts]; const FIXP_DBL *imag = qd_ch->hQmfSlotsImag[ts]; const int ovSlots = gc->nQmfOvTimeSlots; const int exp_lb = SCALE2EXP((ts < ovSlots) ? qd_ch->scaling.ov_lb_scale : qd_ch->scaling.lb_scale); const int exp_hb = SCALE2EXP(qd_ch->scaling.hb_scale); const int lsb = qd_ch->fb.lsb; const int usb = qd_ch->fb.usb; int b = start_band; int lb_sf, hb_sf; int target_exp = ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK + qd_ch->fb.filterScale; FDK_ASSERT(ts < (gc->nQmfTimeSlots + gc->nQmfOvTimeSlots)); FDK_ASSERT(start_band >= 0); FDK_ASSERT(stop_band <= gc->nQmfProcBands); if (qd_ch->fb.no_channels == 24) { target_exp -= 1; } /* Limit scaling factors to maximum negative value to avoid faulty behaviour due to right-shifts. Corresponding asserts were observed during robustness testing. */ lb_sf = fMax(exp_lb - target_exp - exp_out, -31); FDK_ASSERT(lb_sf < 32); hb_sf = fMax(exp_hb - target_exp - exp_out, -31); FDK_ASSERT(hb_sf < 32); if (pQmfOutImag == NULL) { for (; b < fMin(lsb, stop_band); b++) { pQmfOutReal[b] = scaleValueSaturate(real[b], lb_sf); } for (; b < fMin(usb, stop_band); b++) { pQmfOutReal[b] = scaleValueSaturate(real[b], hb_sf); } for (; b < stop_band; b++) { pQmfOutReal[b] = (FIXP_DBL)0; } } else { FDK_ASSERT(imag != NULL); for (; b < fMin(lsb, stop_band); b++) { pQmfOutReal[b] = scaleValueSaturate(real[b], lb_sf); pQmfOutImag[b] = scaleValueSaturate(imag[b], lb_sf); } for (; b < fMin(usb, stop_band); b++) { pQmfOutReal[b] = scaleValueSaturate(real[b], hb_sf); pQmfOutImag[b] = scaleValueSaturate(imag[b], hb_sf); } for (; b < stop_band; b++) { pQmfOutReal[b] = (FIXP_DBL)0; pQmfOutImag[b] = (FIXP_DBL)0; } } } void FDK_QmfDomain_GetWorkBuffer(const HANDLE_FDK_QMF_DOMAIN_IN qd_ch, const int ts, FIXP_DBL **ppQmfReal, FIXP_DBL **ppQmfImag) { FDK_ASSERT(qd_ch != NULL); FDK_ASSERT(ppQmfReal != NULL); FDK_ASSERT(ppQmfImag != NULL); const int bands = qd_ch->workBuf_nBands; FIXP_DBL **pWorkBuf = qd_ch->pWorkBuffer; USHORT workBufferOffset = qd_ch->workBufferOffset; USHORT workBufferSectSize = qd_ch->workBufferSectSize; FDK_ASSERT(bands > 0); FDK_ASSERT(ts < qd_ch->workBuf_nTimeSlots); *ppQmfReal = FDK_getWorkBuffer( pWorkBuf, workBufferOffset + (ts * CMPLX_MOD + 0) * bands, workBufferSectSize, bands); *ppQmfImag = FDK_getWorkBuffer( pWorkBuf, workBufferOffset + (ts * CMPLX_MOD + 1) * bands, workBufferSectSize, bands); } void FDK_QmfDomain_WorkBuffer2ProcChannel( const HANDLE_FDK_QMF_DOMAIN_IN qd_ch) { FDK_ASSERT(qd_ch != NULL); HANDLE_FDK_QMF_DOMAIN_GC gc = qd_ch->pGlobalConf; FIXP_DBL **pWorkBuf = qd_ch->pWorkBuffer; USHORT workBufferOffset = qd_ch->workBufferOffset; USHORT workBufferSectSize = qd_ch->workBufferSectSize; if (FDK_getWorkBuffer(pWorkBuf, workBufferOffset, workBufferSectSize, qd_ch->workBuf_nBands) == qd_ch->hQmfSlotsReal[gc->nQmfOvTimeSlots]) { /* work buffer is part of processing channel => nothing to do */ return; } else { /* copy parked new QMF data to processing channel */ const int bands = qd_ch->workBuf_nBands; const int slots = qd_ch->workBuf_nTimeSlots; int ts; for (ts = 0; ts < slots; ts++) { FDKmemcpy(qd_ch->hQmfSlotsReal[gc->nQmfOvTimeSlots + ts], FDK_getWorkBuffer(pWorkBuf, workBufferOffset, workBufferSectSize, bands), sizeof(FIXP_DBL) * bands); // parkBuf_to_anaMatrix workBufferOffset += bands; FDKmemcpy(qd_ch->hQmfSlotsImag[gc->nQmfOvTimeSlots + ts], FDK_getWorkBuffer(pWorkBuf, workBufferOffset, workBufferSectSize, bands), sizeof(FIXP_DBL) * bands); workBufferOffset += bands; } } } void FDK_QmfDomain_QmfData2HBE(HANDLE_FDK_QMF_DOMAIN_IN qd_ch, FIXP_DBL **ppQmfReal, FIXP_DBL **ppQmfImag) { FDK_ASSERT(qd_ch != NULL); FDK_ASSERT(ppQmfReal != NULL); FDK_ASSERT(ppQmfImag != NULL); HANDLE_FDK_QMF_DOMAIN_GC gc = qd_ch->pGlobalConf; FIXP_DBL **pWorkBuf = qd_ch->pWorkBuffer; USHORT workBufferOffset = qd_ch->workBufferOffset; USHORT workBufferSectSize = qd_ch->workBufferSectSize; if (FDK_getWorkBuffer(pWorkBuf, workBufferOffset, workBufferSectSize, qd_ch->workBuf_nBands) == qd_ch->hQmfSlotsReal[gc->nQmfOvTimeSlots]) { // left channel (anaMatrix) int ts; const int bands = gc->nBandsAnalysis; const int slots = qd_ch->workBuf_nTimeSlots; FDK_ASSERT(bands <= 64); for (ts = 0; ts < slots; ts++) { /* copy current data of processing channel */ FIXP_DBL tmp[64]; // one slot /* real */ FDKmemcpy(tmp, qd_ch->hQmfSlotsReal[gc->nQmfOvTimeSlots + ts], sizeof(FIXP_DBL) * bands); // anaMatrix_to_tmp FDKmemcpy(qd_ch->hQmfSlotsReal[gc->nQmfOvTimeSlots + ts], ppQmfReal[ts], sizeof(FIXP_DBL) * bands); // HBE_to_anaMatrix FDKmemcpy(ppQmfReal[ts], tmp, sizeof(FIXP_DBL) * bands); // tmp_to_HBE /* imag */ FDKmemcpy(tmp, qd_ch->hQmfSlotsImag[gc->nQmfOvTimeSlots + ts], sizeof(FIXP_DBL) * bands); FDKmemcpy(qd_ch->hQmfSlotsImag[gc->nQmfOvTimeSlots + ts], ppQmfImag[ts], sizeof(FIXP_DBL) * bands); FDKmemcpy(ppQmfImag[ts], tmp, sizeof(FIXP_DBL) * bands); } } else { // right channel (parkBuf) const int bands = qd_ch->workBuf_nBands; const int slots = qd_ch->workBuf_nTimeSlots; int ts; FDK_ASSERT(qd_ch->workBuf_nBands == gc->nBandsAnalysis); for (ts = 0; ts < slots; ts++) { /* copy HBE QMF data buffer to processing channel */ FDKmemcpy(qd_ch->hQmfSlotsReal[gc->nQmfOvTimeSlots + ts], ppQmfReal[ts], sizeof(FIXP_DBL) * bands); // HBE_to_anaMatrix FDKmemcpy(qd_ch->hQmfSlotsImag[gc->nQmfOvTimeSlots + ts], ppQmfImag[ts], sizeof(FIXP_DBL) * bands); /* copy parked new QMF data to HBE QMF data buffer */ FDKmemcpy(ppQmfReal[ts], FDK_getWorkBuffer(pWorkBuf, workBufferOffset, workBufferSectSize, bands), sizeof(FIXP_DBL) * bands); // parkBuf_to_HBE workBufferOffset += bands; FDKmemcpy(ppQmfImag[ts], FDK_getWorkBuffer(pWorkBuf, workBufferOffset, workBufferSectSize, bands), sizeof(FIXP_DBL) * bands); workBufferOffset += bands; } } } void FDK_QmfDomain_ClearRequested(HANDLE_FDK_QMF_DOMAIN_GC hgc) { hgc->qmfDomainExplicitConfig = 0; hgc->flags_requested = 0; hgc->nInputChannels_requested = 0; hgc->nOutputChannels_requested = 0; hgc->parkChannel_requested = 0; hgc->nBandsAnalysis_requested = 0; hgc->nBandsSynthesis_requested = 0; hgc->nQmfTimeSlots_requested = 0; hgc->nQmfOvTimeSlots_requested = 0; hgc->nQmfProcBands_requested = 0; hgc->nQmfProcChannels_requested = 0; } static void FDK_QmfDomain_ClearConfigured(HANDLE_FDK_QMF_DOMAIN_GC hgc) { hgc->flags = 0; hgc->nInputChannels = 0; hgc->nOutputChannels = 0; hgc->parkChannel = 0; hgc->nBandsAnalysis = 0; hgc->nBandsSynthesis = 0; hgc->nQmfTimeSlots = 0; hgc->nQmfOvTimeSlots = 0; hgc->nQmfProcBands = 0; hgc->nQmfProcChannels = 0; } static void FDK_QmfDomain_ClearFilterBank(HANDLE_FDK_QMF_DOMAIN hqd) { int ch; for (ch = 0; ch < ((8) + (1)); ch++) { FDKmemclear(&hqd->QmfDomainIn[ch].fb, sizeof(hqd->QmfDomainIn[ch].fb)); } for (ch = 0; ch < ((8) + (1)); ch++) { FDKmemclear(&hqd->QmfDomainOut[ch].fb, sizeof(hqd->QmfDomainIn[ch].fb)); } } QMF_DOMAIN_ERROR FDK_QmfDomain_Configure(HANDLE_FDK_QMF_DOMAIN hqd) { FDK_ASSERT(hqd != NULL); QMF_DOMAIN_ERROR err = QMF_DOMAIN_OK; int i, size_main, size, size_temp = 0; HANDLE_FDK_QMF_DOMAIN_GC hgc = &hqd->globalConf; FIXP_DBL **pWorkBuffer = hgc->pWorkBuffer; int hasChanged = 0; if ((hgc->nQmfProcChannels_requested > 0) && (hgc->nQmfProcBands_requested != 64)) { return QMF_DOMAIN_INIT_ERROR; } if (hgc->nBandsAnalysis_requested > hgc->nQmfProcBands_requested) { /* In general the output of the qmf analysis is written to QMF memory slots which size is defined by nQmfProcBands. nBandsSynthesis may be larger than nQmfProcBands. This is e.g. the case if the QMF based resampler is used. */ return QMF_DOMAIN_INIT_ERROR; } /* 1. adjust change of processing channels by comparison of current and * requested parameters */ if ((hgc->nQmfProcChannels != hgc->nQmfProcChannels_requested) || (hgc->nQmfProcBands != hgc->nQmfProcBands_requested) || (hgc->nQmfTimeSlots != hgc->nQmfTimeSlots_requested)) { for (i = 0; i < hgc->nQmfProcChannels_requested; i++) { hqd->QmfDomainIn[i].workBuf_nBands = hgc->nQmfProcBands_requested; hgc->nQmfProcBands = hgc->nQmfProcBands_requested; hqd->QmfDomainIn[i].workBuf_nTimeSlots = hgc->nQmfTimeSlots_requested; } hgc->nQmfProcChannels = hgc->nQmfProcChannels_requested; /* keep highest value encountered so far as allocated */ hasChanged = 1; } /* 2. reallocate persistent memory if necessary (analysis state-buffers, * timeslot-pointer-array, overlap-buffers, synthesis state-buffers) */ if ((hgc->nInputChannels != hgc->nInputChannels_requested) || (hgc->nBandsAnalysis != hgc->nBandsAnalysis_requested) || (hgc->nQmfTimeSlots != hgc->nQmfTimeSlots_requested) || (hgc->nQmfOvTimeSlots != hgc->nQmfOvTimeSlots_requested) || (hgc->nOutputChannels != hgc->nOutputChannels_requested) || (hgc->nBandsSynthesis != hgc->nBandsSynthesis_requested) || (hgc->parkChannel != hgc->parkChannel_requested)) { hgc->nInputChannels = hgc->nInputChannels_requested; hgc->nBandsAnalysis = hgc->nBandsAnalysis_requested; hgc->nQmfTimeSlots = hgc->nQmfTimeSlots_requested; hgc->nQmfOvTimeSlots = hgc->nQmfOvTimeSlots_requested; hgc->nOutputChannels = hgc->nOutputChannels_requested; hgc->nBandsSynthesis = hgc->nBandsSynthesis_requested; hgc->parkChannel = hgc->parkChannel_requested; if (FDK_QmfDomain_AllocatePersistentMemory(hqd)) { err = QMF_DOMAIN_OUT_OF_MEMORY; goto bail; } /* 3. set request-flag for downsampled SBR */ if ((hgc->nBandsAnalysis == 32) && (hgc->nBandsSynthesis == 32) && !(hgc->flags & (QMF_FLAG_CLDFB | QMF_FLAG_MPSLDFB))) { hgc->flags_requested |= QMF_FLAG_DOWNSAMPLED; } hasChanged = 1; } /* 4. initialize tables and buffer for QMF-resampler */ /* 5. set requested flags */ if (hgc->flags != hgc->flags_requested) { if ((hgc->flags_requested & QMF_FLAG_MPSLDFB) && (hgc->flags_requested & QMF_FLAG_CLDFB)) { hgc->flags_requested &= ~QMF_FLAG_CLDFB; } hgc->flags = hgc->flags_requested; hasChanged = 1; } if (hasChanged) { /* 6. recalculate and check size of required workbuffer-space */ if (hgc->parkChannel && (hqd->globalConf.nQmfProcChannels == 1)) { /* configure temp QMF buffer for parking right channel MPS212 output, * (USAC stereoConfigIndex 3 only) */ hqd->QmfDomainIn[1].workBuf_nBands = hqd->globalConf.nBandsAnalysis; hqd->QmfDomainIn[1].workBuf_nTimeSlots = hqd->globalConf.nQmfTimeSlots; size_temp = hqd->QmfDomainIn[1].workBuf_nBands * hqd->QmfDomainIn[1].workBuf_nTimeSlots * CMPLX_MOD; } size_main = hqd->QmfDomainIn[0].workBuf_nBands * hqd->QmfDomainIn[0].workBuf_nTimeSlots * CMPLX_MOD; size = size_main * hgc->nQmfProcChannels + size_temp; if (size > (QMF_MAX_WB_SECTIONS * QMF_WB_SECTION_SIZE)) { err = QMF_DOMAIN_OUT_OF_MEMORY; goto bail; } /* 7. allocate additional workbuffer if necessary */ if ((size > 0 /* *QMF_WB_SECTION_SIZE */) && (pWorkBuffer[0] == NULL)) { /* get work buffer of size QMF_WB_SECTION_SIZE */ pWorkBuffer[0] = GetQmfWorkBufferCore6(); } if ((size > 1 * QMF_WB_SECTION_SIZE) && (pWorkBuffer[1] == NULL)) { /* get work buffer of size QMF_WB_SECTION_SIZE */ pWorkBuffer[1] = GetQmfWorkBufferCore1(); } if ((size > 2 * QMF_WB_SECTION_SIZE) && (pWorkBuffer[2] == NULL)) { /* get work buffer of size QMF_WB_SECTION_SIZE */ pWorkBuffer[2] = GetQmfWorkBufferCore3(); } if ((size > 3 * QMF_WB_SECTION_SIZE) && (pWorkBuffer[3] == NULL)) { /* get work buffer of size QMF_WB_SECTION_SIZE */ pWorkBuffer[3] = GetQmfWorkBufferCore4(); } if ((size > 4 * QMF_WB_SECTION_SIZE) && (pWorkBuffer[4] == NULL)) { /* get work buffer of size QMF_WB_SECTION_SIZE */ pWorkBuffer[4] = GetQmfWorkBufferCore7(); } /* 8. distribute workbuffer over processing channels */ for (i = 0; i < hgc->nQmfProcChannels; i++) { FDK_QmfDomain_FeedWorkBuffer(hqd, i, pWorkBuffer, size_main * i, QMF_WB_SECTION_SIZE, size_main); } if (hgc->parkChannel) { for (; i < hgc->nInputChannels; i++) { FDK_QmfDomain_FeedWorkBuffer(hqd, 1, pWorkBuffer, size_main * hgc->nQmfProcChannels, QMF_WB_SECTION_SIZE, size_temp); } } /* 9. (re-)init filterbank */ for (i = 0; i < hgc->nOutputChannels; i++) { if ((hqd->QmfDomainOut[i].fb.lsb == 0) && (hqd->QmfDomainOut[i].fb.usb == 0)) { /* Although lsb and usb are set in the SBR module, they are initialized * at this point due to the case of using MPS without SBR. */ hqd->QmfDomainOut[i].fb.lsb = hgc->nBandsAnalysis_requested; hqd->QmfDomainOut[i].fb.usb = fMin((INT)hgc->nBandsSynthesis_requested, 64); } } if (FDK_QmfDomain_InitFilterBank(hqd, 0)) { err = QMF_DOMAIN_INIT_ERROR; } } bail: if (err) { FDK_QmfDomain_FreeMem(hqd); } return err; } static void FDK_QmfDomain_FreeWorkBuffer(HANDLE_FDK_QMF_DOMAIN hqd) { FIXP_DBL **pWorkBuffer = hqd->globalConf.pWorkBuffer; if (pWorkBuffer[0]) FreeQmfWorkBufferCore6(&pWorkBuffer[0]); if (pWorkBuffer[1]) FreeQmfWorkBufferCore1(&pWorkBuffer[1]); if (pWorkBuffer[2]) FreeQmfWorkBufferCore3(&pWorkBuffer[2]); if (pWorkBuffer[3]) FreeQmfWorkBufferCore4(&pWorkBuffer[3]); if (pWorkBuffer[4]) FreeQmfWorkBufferCore7(&pWorkBuffer[4]); } void FDK_QmfDomain_FreeMem(HANDLE_FDK_QMF_DOMAIN hqd) { FDK_QmfDomain_FreeWorkBuffer(hqd); FDK_QmfDomain_FreePersistentMemory(hqd); FDK_QmfDomain_ClearFilterBank(hqd); FDK_QmfDomain_ClearConfigured(&hqd->globalConf); FDK_QmfDomain_ClearRequested(&hqd->globalConf); } void FDK_QmfDomain_Close(HANDLE_FDK_QMF_DOMAIN hqd) { FDK_QmfDomain_FreeWorkBuffer(hqd); FDK_QmfDomain_FreePersistentMemory(hqd); } fdk-aac-2.0.2/libFDK/src/FDK_tools_rom.cpp000066400000000000000000013771541404225310700201100ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Oliver Moser Description: ROM tables used by FDK tools *******************************************************************************/ #include "FDK_tools_rom.h" RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_STP SineTable80[] = { STCP(0x7fffffff, 0x00000000), STCP(0x7ff9af04, 0x02835b5a), STCP(0x7fe6bcb0, 0x05067734), STCP(0x7fc72ae2, 0x07891418), STCP(0x7f9afcb9, 0x0a0af299), STCP(0x7f62368f, 0x0c8bd35e), STCP(0x7f1cde01, 0x0f0b7727), STCP(0x7ecaf9e5, 0x11899ed3), STCP(0x7e6c9251, 0x14060b68), STCP(0x7e01b096, 0x16807e15), STCP(0x7d8a5f40, 0x18f8b83c), STCP(0x7d06aa16, 0x1b6e7b7a), STCP(0x7c769e18, 0x1de189a6), STCP(0x7bda497d, 0x2051a4dd), STCP(0x7b31bbb2, 0x22be8f87), STCP(0x7a7d055b, 0x25280c5e), STCP(0x79bc384d, 0x278dde6e), STCP(0x78ef678f, 0x29efc925), STCP(0x7816a759, 0x2c4d9050), STCP(0x77320d0d, 0x2ea6f827), STCP(0x7641af3d, 0x30fbc54d), STCP(0x7545a5a0, 0x334bbcde), STCP(0x743e0918, 0x3596a46c), STCP(0x732af3a7, 0x37dc420c), STCP(0x720c8075, 0x3a1c5c57), STCP(0x70e2cbc6, 0x3c56ba70), STCP(0x6fadf2fc, 0x3e8b240e), STCP(0x6e6e1492, 0x40b9617d), STCP(0x6d23501b, 0x42e13ba4), STCP(0x6bcdc639, 0x45027c0c), STCP(0x6a6d98a4, 0x471cece7), STCP(0x6902ea1d, 0x4930590f), STCP(0x678dde6e, 0x4b3c8c12), STCP(0x660e9a6a, 0x4d415234), STCP(0x648543e4, 0x4f3e7875), STCP(0x62f201ac, 0x5133cc94), STCP(0x6154fb91, 0x53211d18), STCP(0x5fae5a55, 0x55063951), STCP(0x5dfe47ad, 0x56e2f15d), STCP(0x5c44ee40, 0x58b71632), STCP(0x5a82799a, 0x5a82799a), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_STP SineTable384[] = { STCP(0x7fffffff, 0x00000000), STCP(0x7fffb9d1, 0x00860a79), STCP(0x7ffee744, 0x010c1460), STCP(0x7ffd885a, 0x01921d20), STCP(0x7ffb9d15, 0x02182427), STCP(0x7ff92577, 0x029e28e2), STCP(0x7ff62182, 0x03242abf), STCP(0x7ff2913a, 0x03aa292a), STCP(0x7fee74a2, 0x0430238f), STCP(0x7fe9cbc0, 0x04b6195d), STCP(0x7fe49698, 0x053c0a01), STCP(0x7fded530, 0x05c1f4e7), STCP(0x7fd8878e, 0x0647d97c), STCP(0x7fd1adb9, 0x06cdb72f), STCP(0x7fca47b9, 0x07538d6b), STCP(0x7fc25596, 0x07d95b9e), STCP(0x7fb9d759, 0x085f2137), STCP(0x7fb0cd0a, 0x08e4dda0), STCP(0x7fa736b4, 0x096a9049), STCP(0x7f9d1461, 0x09f0389f), STCP(0x7f92661d, 0x0a75d60e), STCP(0x7f872bf3, 0x0afb6805), STCP(0x7f7b65ef, 0x0b80edf1), STCP(0x7f6f141f, 0x0c066740), STCP(0x7f62368f, 0x0c8bd35e), STCP(0x7f54cd4f, 0x0d1131ba), STCP(0x7f46d86c, 0x0d9681c2), STCP(0x7f3857f6, 0x0e1bc2e4), STCP(0x7f294bfd, 0x0ea0f48c), STCP(0x7f19b491, 0x0f26162a), STCP(0x7f0991c4, 0x0fab272b), STCP(0x7ef8e3a6, 0x103026fe), STCP(0x7ee7aa4c, 0x10b5150f), STCP(0x7ed5e5c6, 0x1139f0cf), STCP(0x7ec3962a, 0x11beb9aa), STCP(0x7eb0bb8a, 0x12436f10), STCP(0x7e9d55fc, 0x12c8106f), STCP(0x7e896595, 0x134c9d34), STCP(0x7e74ea6a, 0x13d114d0), STCP(0x7e5fe493, 0x145576b1), STCP(0x7e4a5426, 0x14d9c245), STCP(0x7e34393b, 0x155df6fc), STCP(0x7e1d93ea, 0x15e21445), STCP(0x7e06644c, 0x1666198d), STCP(0x7deeaa7a, 0x16ea0646), STCP(0x7dd6668f, 0x176dd9de), STCP(0x7dbd98a4, 0x17f193c5), STCP(0x7da440d6, 0x1875336a), STCP(0x7d8a5f40, 0x18f8b83c), STCP(0x7d6ff3fe, 0x197c21ad), STCP(0x7d54ff2e, 0x19ff6f2a), STCP(0x7d3980ec, 0x1a82a026), STCP(0x7d1d7958, 0x1b05b40f), STCP(0x7d00e88f, 0x1b88aa55), STCP(0x7ce3ceb2, 0x1c0b826a), STCP(0x7cc62bdf, 0x1c8e3bbe), STCP(0x7ca80038, 0x1d10d5c2), STCP(0x7c894bde, 0x1d934fe5), STCP(0x7c6a0ef2, 0x1e15a99a), STCP(0x7c4a4996, 0x1e97e251), STCP(0x7c29fbee, 0x1f19f97b), STCP(0x7c09261d, 0x1f9bee8a), STCP(0x7be7c847, 0x201dc0ef), STCP(0x7bc5e290, 0x209f701c), STCP(0x7ba3751d, 0x2120fb83), STCP(0x7b808015, 0x21a26295), STCP(0x7b5d039e, 0x2223a4c5), STCP(0x7b38ffde, 0x22a4c185), STCP(0x7b1474fd, 0x2325b847), STCP(0x7aef6323, 0x23a6887f), STCP(0x7ac9ca7a, 0x2427319d), STCP(0x7aa3ab29, 0x24a7b317), STCP(0x7a7d055b, 0x25280c5e), STCP(0x7a55d93a, 0x25a83ce6), STCP(0x7a2e26f2, 0x26284422), STCP(0x7a05eead, 0x26a82186), STCP(0x79dd3098, 0x2727d486), STCP(0x79b3ece0, 0x27a75c95), STCP(0x798a23b1, 0x2826b928), STCP(0x795fd53a, 0x28a5e9b4), STCP(0x793501a9, 0x2924edac), STCP(0x7909a92d, 0x29a3c485), STCP(0x78ddcbf5, 0x2a226db5), STCP(0x78b16a32, 0x2aa0e8b0), STCP(0x78848414, 0x2b1f34eb), STCP(0x785719cc, 0x2b9d51dd), STCP(0x78292b8d, 0x2c1b3efb), STCP(0x77fab989, 0x2c98fbba), STCP(0x77cbc3f2, 0x2d168792), STCP(0x779c4afc, 0x2d93e1f8), STCP(0x776c4edb, 0x2e110a62), STCP(0x773bcfc4, 0x2e8e0048), STCP(0x770acdec, 0x2f0ac320), STCP(0x76d94989, 0x2f875262), STCP(0x76a742d1, 0x3003ad85), STCP(0x7674b9fa, 0x307fd401), STCP(0x7641af3d, 0x30fbc54d), STCP(0x760e22d1, 0x317780e2), STCP(0x75da14ef, 0x31f30638), STCP(0x75a585cf, 0x326e54c7), STCP(0x757075ac, 0x32e96c09), STCP(0x753ae4c0, 0x33644b76), STCP(0x7504d345, 0x33def287), STCP(0x74ce4177, 0x345960b7), STCP(0x74972f92, 0x34d3957e), STCP(0x745f9dd1, 0x354d9057), STCP(0x74278c72, 0x35c750bc), STCP(0x73eefbb3, 0x3640d627), STCP(0x73b5ebd1, 0x36ba2014), STCP(0x737c5d0b, 0x37332dfd), STCP(0x73424fa0, 0x37abff5d), STCP(0x7307c3d0, 0x382493b0), STCP(0x72ccb9db, 0x389cea72), STCP(0x72913201, 0x3915031f), STCP(0x72552c85, 0x398cdd32), STCP(0x7218a9a7, 0x3a04782a), STCP(0x71dba9ab, 0x3a7bd382), STCP(0x719e2cd2, 0x3af2eeb7), STCP(0x71603361, 0x3b69c947), STCP(0x7121bd9c, 0x3be062b0), STCP(0x70e2cbc6, 0x3c56ba70), STCP(0x70a35e25, 0x3cccd004), STCP(0x706374ff, 0x3d42a2ec), STCP(0x7023109a, 0x3db832a6), STCP(0x6fe2313c, 0x3e2d7eb1), STCP(0x6fa0d72c, 0x3ea2868c), STCP(0x6f5f02b2, 0x3f1749b8), STCP(0x6f1cb416, 0x3f8bc7b4), STCP(0x6ed9eba1, 0x40000000), STCP(0x6e96a99d, 0x4073f21d), STCP(0x6e52ee52, 0x40e79d8c), STCP(0x6e0eba0c, 0x415b01ce), STCP(0x6dca0d14, 0x41ce1e65), STCP(0x6d84e7b7, 0x4240f2d1), STCP(0x6d3f4a40, 0x42b37e96), STCP(0x6cf934fc, 0x4325c135), STCP(0x6cb2a837, 0x4397ba32), STCP(0x6c6ba43e, 0x44096910), STCP(0x6c242960, 0x447acd50), STCP(0x6bdc37eb, 0x44ebe679), STCP(0x6b93d02e, 0x455cb40c), STCP(0x6b4af279, 0x45cd358f), STCP(0x6b019f1a, 0x463d6a87), STCP(0x6ab7d663, 0x46ad5278), STCP(0x6a6d98a4, 0x471cece7), STCP(0x6a22e630, 0x478c395a), STCP(0x69d7bf57, 0x47fb3757), STCP(0x698c246c, 0x4869e665), STCP(0x694015c3, 0x48d84609), STCP(0x68f393ae, 0x494655cc), STCP(0x68a69e81, 0x49b41533), STCP(0x68593691, 0x4a2183c8), STCP(0x680b5c33, 0x4a8ea111), STCP(0x67bd0fbd, 0x4afb6c98), STCP(0x676e5183, 0x4b67e5e4), STCP(0x671f21dc, 0x4bd40c80), STCP(0x66cf8120, 0x4c3fdff4), STCP(0x667f6fa5, 0x4cab5fc9), STCP(0x662eedc3, 0x4d168b8b), STCP(0x65ddfbd3, 0x4d8162c4), STCP(0x658c9a2d, 0x4debe4fe), STCP(0x653ac92b, 0x4e5611c5), STCP(0x64e88926, 0x4ebfe8a5), STCP(0x6495da79, 0x4f296928), STCP(0x6442bd7e, 0x4f9292dc), STCP(0x63ef3290, 0x4ffb654d), STCP(0x639b3a0b, 0x5063e008), STCP(0x6346d44b, 0x50cc029c), STCP(0x62f201ac, 0x5133cc94), STCP(0x629cc28c, 0x519b3d80), STCP(0x62471749, 0x520254ef), STCP(0x61f1003f, 0x5269126e), STCP(0x619a7dce, 0x52cf758f), STCP(0x61439053, 0x53357ddf), STCP(0x60ec3830, 0x539b2af0), STCP(0x609475c3, 0x54007c51), STCP(0x603c496c, 0x54657194), STCP(0x5fe3b38d, 0x54ca0a4b), STCP(0x5f8ab487, 0x552e4605), STCP(0x5f314cba, 0x55922457), STCP(0x5ed77c8a, 0x55f5a4d2), STCP(0x5e7d4458, 0x5658c709), STCP(0x5e22a487, 0x56bb8a90), STCP(0x5dc79d7c, 0x571deefa), STCP(0x5d6c2f99, 0x577ff3da), STCP(0x5d105b44, 0x57e198c7), STCP(0x5cb420e0, 0x5842dd54), STCP(0x5c5780d3, 0x58a3c118), STCP(0x5bfa7b82, 0x590443a7), STCP(0x5b9d1154, 0x59646498), STCP(0x5b3f42ae, 0x59c42381), STCP(0x5ae10ff9, 0x5a237ffa), STCP(0x5a82799a, 0x5a82799a), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_STP SineTable480[] = { STCP(0x7fffffff, 0x00000000), STCP(0x7fffd315, 0x006b3b9b), STCP(0x7fff4c54, 0x00d676eb), STCP(0x7ffe6bbf, 0x0141b1a5), STCP(0x7ffd3154, 0x01aceb7c), STCP(0x7ffb9d15, 0x02182427), STCP(0x7ff9af04, 0x02835b5a), STCP(0x7ff76721, 0x02ee90c8), STCP(0x7ff4c56f, 0x0359c428), STCP(0x7ff1c9ef, 0x03c4f52f), STCP(0x7fee74a2, 0x0430238f), STCP(0x7feac58d, 0x049b4f00), STCP(0x7fe6bcb0, 0x05067734), STCP(0x7fe25a0f, 0x05719be2), STCP(0x7fdd9dad, 0x05dcbcbe), STCP(0x7fd8878e, 0x0647d97c), STCP(0x7fd317b4, 0x06b2f1d2), STCP(0x7fcd4e24, 0x071e0575), STCP(0x7fc72ae2, 0x07891418), STCP(0x7fc0adf2, 0x07f41d72), STCP(0x7fb9d759, 0x085f2137), STCP(0x7fb2a71b, 0x08ca1f1b), STCP(0x7fab1d3d, 0x093516d4), STCP(0x7fa339c5, 0x09a00817), STCP(0x7f9afcb9, 0x0a0af299), STCP(0x7f92661d, 0x0a75d60e), STCP(0x7f8975f9, 0x0ae0b22c), STCP(0x7f802c52, 0x0b4b86a8), STCP(0x7f76892f, 0x0bb65336), STCP(0x7f6c8c96, 0x0c21178c), STCP(0x7f62368f, 0x0c8bd35e), STCP(0x7f578721, 0x0cf68662), STCP(0x7f4c7e54, 0x0d61304e), STCP(0x7f411c2f, 0x0dcbd0d5), STCP(0x7f3560b9, 0x0e3667ad), STCP(0x7f294bfd, 0x0ea0f48c), STCP(0x7f1cde01, 0x0f0b7727), STCP(0x7f1016ce, 0x0f75ef33), STCP(0x7f02f66f, 0x0fe05c64), STCP(0x7ef57cea, 0x104abe71), STCP(0x7ee7aa4c, 0x10b5150f), STCP(0x7ed97e9c, 0x111f5ff4), STCP(0x7ecaf9e5, 0x11899ed3), STCP(0x7ebc1c31, 0x11f3d164), STCP(0x7eace58a, 0x125df75b), STCP(0x7e9d55fc, 0x12c8106f), STCP(0x7e8d6d91, 0x13321c53), STCP(0x7e7d2c54, 0x139c1abf), STCP(0x7e6c9251, 0x14060b68), STCP(0x7e5b9f93, 0x146fee03), STCP(0x7e4a5426, 0x14d9c245), STCP(0x7e38b017, 0x154387e6), STCP(0x7e26b371, 0x15ad3e9a), STCP(0x7e145e42, 0x1616e618), STCP(0x7e01b096, 0x16807e15), STCP(0x7deeaa7a, 0x16ea0646), STCP(0x7ddb4bfc, 0x17537e63), STCP(0x7dc79529, 0x17bce621), STCP(0x7db3860f, 0x18263d36), STCP(0x7d9f1ebd, 0x188f8357), STCP(0x7d8a5f40, 0x18f8b83c), STCP(0x7d7547a7, 0x1961db9b), STCP(0x7d5fd801, 0x19caed29), STCP(0x7d4a105d, 0x1a33ec9c), STCP(0x7d33f0ca, 0x1a9cd9ac), STCP(0x7d1d7958, 0x1b05b40f), STCP(0x7d06aa16, 0x1b6e7b7a), STCP(0x7cef8315, 0x1bd72fa4), STCP(0x7cd80464, 0x1c3fd045), STCP(0x7cc02e15, 0x1ca85d12), STCP(0x7ca80038, 0x1d10d5c2), STCP(0x7c8f7ade, 0x1d793a0b), STCP(0x7c769e18, 0x1de189a6), STCP(0x7c5d69f7, 0x1e49c447), STCP(0x7c43de8e, 0x1eb1e9a7), STCP(0x7c29fbee, 0x1f19f97b), STCP(0x7c0fc22a, 0x1f81f37c), STCP(0x7bf53153, 0x1fe9d75f), STCP(0x7bda497d, 0x2051a4dd), STCP(0x7bbf0aba, 0x20b95bac), STCP(0x7ba3751d, 0x2120fb83), STCP(0x7b8788ba, 0x2188841a), STCP(0x7b6b45a5, 0x21eff528), STCP(0x7b4eabf1, 0x22574e65), STCP(0x7b31bbb2, 0x22be8f87), STCP(0x7b1474fd, 0x2325b847), STCP(0x7af6d7e6, 0x238cc85d), STCP(0x7ad8e482, 0x23f3bf7e), STCP(0x7aba9ae6, 0x245a9d65), STCP(0x7a9bfb27, 0x24c161c7), STCP(0x7a7d055b, 0x25280c5e), STCP(0x7a5db997, 0x258e9ce0), STCP(0x7a3e17f2, 0x25f51307), STCP(0x7a1e2082, 0x265b6e8a), STCP(0x79fdd35c, 0x26c1af22), STCP(0x79dd3098, 0x2727d486), STCP(0x79bc384d, 0x278dde6e), STCP(0x799aea92, 0x27f3cc94), STCP(0x7979477d, 0x28599eb0), STCP(0x79574f28, 0x28bf547b), STCP(0x793501a9, 0x2924edac), STCP(0x79125f19, 0x298a69fc), STCP(0x78ef678f, 0x29efc925), STCP(0x78cc1b26, 0x2a550adf), STCP(0x78a879f4, 0x2aba2ee4), STCP(0x78848414, 0x2b1f34eb), STCP(0x7860399e, 0x2b841caf), STCP(0x783b9aad, 0x2be8e5e8), STCP(0x7816a759, 0x2c4d9050), STCP(0x77f15fbc, 0x2cb21ba0), STCP(0x77cbc3f2, 0x2d168792), STCP(0x77a5d413, 0x2d7ad3de), STCP(0x777f903c, 0x2ddf0040), STCP(0x7758f886, 0x2e430c6f), STCP(0x77320d0d, 0x2ea6f827), STCP(0x770acdec, 0x2f0ac320), STCP(0x76e33b3f, 0x2f6e6d16), STCP(0x76bb5521, 0x2fd1f5c1), STCP(0x76931bae, 0x30355cdd), STCP(0x766a8f04, 0x3098a223), STCP(0x7641af3d, 0x30fbc54d), STCP(0x76187c77, 0x315ec617), STCP(0x75eef6ce, 0x31c1a43b), STCP(0x75c51e61, 0x32245f72), STCP(0x759af34c, 0x3286f779), STCP(0x757075ac, 0x32e96c09), STCP(0x7545a5a0, 0x334bbcde), STCP(0x751a8346, 0x33ade9b3), STCP(0x74ef0ebc, 0x340ff242), STCP(0x74c34820, 0x3471d647), STCP(0x74972f92, 0x34d3957e), STCP(0x746ac52f, 0x35352fa1), STCP(0x743e0918, 0x3596a46c), STCP(0x7410fb6b, 0x35f7f39c), STCP(0x73e39c49, 0x36591cea), STCP(0x73b5ebd1, 0x36ba2014), STCP(0x7387ea23, 0x371afcd5), STCP(0x73599760, 0x377bb2e9), STCP(0x732af3a7, 0x37dc420c), STCP(0x72fbff1b, 0x383ca9fb), STCP(0x72ccb9db, 0x389cea72), STCP(0x729d2409, 0x38fd032d), STCP(0x726d3dc6, 0x395cf3e9), STCP(0x723d0734, 0x39bcbc63), STCP(0x720c8075, 0x3a1c5c57), STCP(0x71dba9ab, 0x3a7bd382), STCP(0x71aa82f7, 0x3adb21a1), STCP(0x71790c7e, 0x3b3a4672), STCP(0x71474660, 0x3b9941b1), STCP(0x711530c2, 0x3bf8131c), STCP(0x70e2cbc6, 0x3c56ba70), STCP(0x70b01790, 0x3cb5376b), STCP(0x707d1443, 0x3d1389cb), STCP(0x7049c203, 0x3d71b14d), STCP(0x701620f5, 0x3dcfadb0), STCP(0x6fe2313c, 0x3e2d7eb1), STCP(0x6fadf2fc, 0x3e8b240e), STCP(0x6f79665b, 0x3ee89d86), STCP(0x6f448b7e, 0x3f45ead8), STCP(0x6f0f6289, 0x3fa30bc1), STCP(0x6ed9eba1, 0x40000000), STCP(0x6ea426ed, 0x405cc754), STCP(0x6e6e1492, 0x40b9617d), STCP(0x6e37b4b6, 0x4115ce38), STCP(0x6e010780, 0x41720d46), STCP(0x6dca0d14, 0x41ce1e65), STCP(0x6d92c59b, 0x422a0154), STCP(0x6d5b313b, 0x4285b5d4), STCP(0x6d23501b, 0x42e13ba4), STCP(0x6ceb2261, 0x433c9283), STCP(0x6cb2a837, 0x4397ba32), STCP(0x6c79e1c2, 0x43f2b271), STCP(0x6c40cf2c, 0x444d7aff), STCP(0x6c07709b, 0x44a8139e), STCP(0x6bcdc639, 0x45027c0c), STCP(0x6b93d02e, 0x455cb40c), STCP(0x6b598ea3, 0x45b6bb5e), STCP(0x6b1f01c0, 0x461091c2), STCP(0x6ae429ae, 0x466a36f9), STCP(0x6aa90697, 0x46c3aac5), STCP(0x6a6d98a4, 0x471cece7), STCP(0x6a31e000, 0x4775fd1f), STCP(0x69f5dcd3, 0x47cedb31), STCP(0x69b98f48, 0x482786dc), STCP(0x697cf78a, 0x487fffe4), STCP(0x694015c3, 0x48d84609), STCP(0x6902ea1d, 0x4930590f), STCP(0x68c574c4, 0x498838b6), STCP(0x6887b5e2, 0x49dfe4c2), STCP(0x6849ada3, 0x4a375cf5), STCP(0x680b5c33, 0x4a8ea111), STCP(0x67ccc1be, 0x4ae5b0da), STCP(0x678dde6e, 0x4b3c8c12), STCP(0x674eb271, 0x4b93327c), STCP(0x670f3df3, 0x4be9a3db), STCP(0x66cf8120, 0x4c3fdff4), STCP(0x668f7c25, 0x4c95e688), STCP(0x664f2f2e, 0x4cebb75c), STCP(0x660e9a6a, 0x4d415234), STCP(0x65cdbe05, 0x4d96b6d3), STCP(0x658c9a2d, 0x4debe4fe), STCP(0x654b2f10, 0x4e40dc79), STCP(0x65097cdb, 0x4e959d08), STCP(0x64c783bd, 0x4eea2670), STCP(0x648543e4, 0x4f3e7875), STCP(0x6442bd7e, 0x4f9292dc), STCP(0x63fff0ba, 0x4fe6756a), STCP(0x63bcddc7, 0x503a1fe5), STCP(0x637984d4, 0x508d9211), STCP(0x6335e611, 0x50e0cbb4), STCP(0x62f201ac, 0x5133cc94), STCP(0x62add7d6, 0x51869476), STCP(0x626968be, 0x51d92321), STCP(0x6224b495, 0x522b7859), STCP(0x61dfbb8a, 0x527d93e6), STCP(0x619a7dce, 0x52cf758f), STCP(0x6154fb91, 0x53211d18), STCP(0x610f3505, 0x53728a4a), STCP(0x60c92a5a, 0x53c3bcea), STCP(0x6082dbc1, 0x5414b4c1), STCP(0x603c496c, 0x54657194), STCP(0x5ff5738d, 0x54b5f32c), STCP(0x5fae5a55, 0x55063951), STCP(0x5f66fdf5, 0x555643c8), STCP(0x5f1f5ea1, 0x55a6125c), STCP(0x5ed77c8a, 0x55f5a4d2), STCP(0x5e8f57e2, 0x5644faf4), STCP(0x5e46f0dd, 0x5694148b), STCP(0x5dfe47ad, 0x56e2f15d), STCP(0x5db55c86, 0x57319135), STCP(0x5d6c2f99, 0x577ff3da), STCP(0x5d22c11c, 0x57ce1917), STCP(0x5cd91140, 0x581c00b3), STCP(0x5c8f203b, 0x5869aa79), STCP(0x5c44ee40, 0x58b71632), STCP(0x5bfa7b82, 0x590443a7), STCP(0x5bafc837, 0x595132a2), STCP(0x5b64d492, 0x599de2ee), STCP(0x5b19a0c8, 0x59ea5454), STCP(0x5ace2d0f, 0x5a36869f), STCP(0x5a82799a, 0x5a82799a), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_STP SineTable512[] = { STCP(0x7fffffff, 0x00000000), STCP(0x7fffd886, 0x006487e3), STCP(0x7fff6216, 0x00c90f88), STCP(0x7ffe9cb2, 0x012d96b1), STCP(0x7ffd885a, 0x01921d20), STCP(0x7ffc250f, 0x01f6a297), STCP(0x7ffa72d1, 0x025b26d7), STCP(0x7ff871a2, 0x02bfa9a4), STCP(0x7ff62182, 0x03242abf), STCP(0x7ff38274, 0x0388a9ea), STCP(0x7ff09478, 0x03ed26e6), STCP(0x7fed5791, 0x0451a177), STCP(0x7fe9cbc0, 0x04b6195d), STCP(0x7fe5f108, 0x051a8e5c), STCP(0x7fe1c76b, 0x057f0035), STCP(0x7fdd4eec, 0x05e36ea9), STCP(0x7fd8878e, 0x0647d97c), STCP(0x7fd37153, 0x06ac406f), STCP(0x7fce0c3e, 0x0710a345), STCP(0x7fc85854, 0x077501be), STCP(0x7fc25596, 0x07d95b9e), STCP(0x7fbc040a, 0x083db0a7), STCP(0x7fb563b3, 0x08a2009a), STCP(0x7fae7495, 0x09064b3a), STCP(0x7fa736b4, 0x096a9049), STCP(0x7f9faa15, 0x09cecf89), STCP(0x7f97cebd, 0x0a3308bd), STCP(0x7f8fa4b0, 0x0a973ba5), STCP(0x7f872bf3, 0x0afb6805), STCP(0x7f7e648c, 0x0b5f8d9f), STCP(0x7f754e80, 0x0bc3ac35), STCP(0x7f6be9d4, 0x0c27c389), STCP(0x7f62368f, 0x0c8bd35e), STCP(0x7f5834b7, 0x0cefdb76), STCP(0x7f4de451, 0x0d53db92), STCP(0x7f434563, 0x0db7d376), STCP(0x7f3857f6, 0x0e1bc2e4), STCP(0x7f2d1c0e, 0x0e7fa99e), STCP(0x7f2191b4, 0x0ee38766), STCP(0x7f15b8ee, 0x0f475bff), STCP(0x7f0991c4, 0x0fab272b), STCP(0x7efd1c3c, 0x100ee8ad), STCP(0x7ef05860, 0x1072a048), STCP(0x7ee34636, 0x10d64dbd), STCP(0x7ed5e5c6, 0x1139f0cf), STCP(0x7ec8371a, 0x119d8941), STCP(0x7eba3a39, 0x120116d5), STCP(0x7eabef2c, 0x1264994e), STCP(0x7e9d55fc, 0x12c8106f), STCP(0x7e8e6eb2, 0x132b7bf9), STCP(0x7e7f3957, 0x138edbb1), STCP(0x7e6fb5f4, 0x13f22f58), STCP(0x7e5fe493, 0x145576b1), STCP(0x7e4fc53e, 0x14b8b17f), STCP(0x7e3f57ff, 0x151bdf86), STCP(0x7e2e9cdf, 0x157f0086), STCP(0x7e1d93ea, 0x15e21445), STCP(0x7e0c3d29, 0x16451a83), STCP(0x7dfa98a8, 0x16a81305), STCP(0x7de8a670, 0x170afd8d), STCP(0x7dd6668f, 0x176dd9de), STCP(0x7dc3d90d, 0x17d0a7bc), STCP(0x7db0fdf8, 0x183366e9), STCP(0x7d9dd55a, 0x18961728), STCP(0x7d8a5f40, 0x18f8b83c), STCP(0x7d769bb5, 0x195b49ea), STCP(0x7d628ac6, 0x19bdcbf3), STCP(0x7d4e2c7f, 0x1a203e1b), STCP(0x7d3980ec, 0x1a82a026), STCP(0x7d24881b, 0x1ae4f1d6), STCP(0x7d0f4218, 0x1b4732ef), STCP(0x7cf9aef0, 0x1ba96335), STCP(0x7ce3ceb2, 0x1c0b826a), STCP(0x7ccda169, 0x1c6d9053), STCP(0x7cb72724, 0x1ccf8cb3), STCP(0x7ca05ff1, 0x1d31774d), STCP(0x7c894bde, 0x1d934fe5), STCP(0x7c71eaf9, 0x1df5163f), STCP(0x7c5a3d50, 0x1e56ca1e), STCP(0x7c4242f2, 0x1eb86b46), STCP(0x7c29fbee, 0x1f19f97b), STCP(0x7c116853, 0x1f7b7481), STCP(0x7bf88830, 0x1fdcdc1b), STCP(0x7bdf5b94, 0x203e300d), STCP(0x7bc5e290, 0x209f701c), STCP(0x7bac1d31, 0x21009c0c), STCP(0x7b920b89, 0x2161b3a0), STCP(0x7b77ada8, 0x21c2b69c), STCP(0x7b5d039e, 0x2223a4c5), STCP(0x7b420d7a, 0x22847de0), STCP(0x7b26cb4f, 0x22e541af), STCP(0x7b0b3d2c, 0x2345eff8), STCP(0x7aef6323, 0x23a6887f), STCP(0x7ad33d45, 0x24070b08), STCP(0x7ab6cba4, 0x24677758), STCP(0x7a9a0e50, 0x24c7cd33), STCP(0x7a7d055b, 0x25280c5e), STCP(0x7a5fb0d8, 0x2588349d), STCP(0x7a4210d8, 0x25e845b6), STCP(0x7a24256f, 0x26483f6c), STCP(0x7a05eead, 0x26a82186), STCP(0x79e76ca7, 0x2707ebc7), STCP(0x79c89f6e, 0x27679df4), STCP(0x79a98715, 0x27c737d3), STCP(0x798a23b1, 0x2826b928), STCP(0x796a7554, 0x288621b9), STCP(0x794a7c12, 0x28e5714b), STCP(0x792a37fe, 0x2944a7a2), STCP(0x7909a92d, 0x29a3c485), STCP(0x78e8cfb2, 0x2a02c7b8), STCP(0x78c7aba2, 0x2a61b101), STCP(0x78a63d11, 0x2ac08026), STCP(0x78848414, 0x2b1f34eb), STCP(0x786280bf, 0x2b7dcf17), STCP(0x78403329, 0x2bdc4e6f), STCP(0x781d9b65, 0x2c3ab2b9), STCP(0x77fab989, 0x2c98fbba), STCP(0x77d78daa, 0x2cf72939), STCP(0x77b417df, 0x2d553afc), STCP(0x7790583e, 0x2db330c7), STCP(0x776c4edb, 0x2e110a62), STCP(0x7747fbce, 0x2e6ec792), STCP(0x77235f2d, 0x2ecc681e), STCP(0x76fe790e, 0x2f29ebcc), STCP(0x76d94989, 0x2f875262), STCP(0x76b3d0b4, 0x2fe49ba7), STCP(0x768e0ea6, 0x3041c761), STCP(0x76680376, 0x309ed556), STCP(0x7641af3d, 0x30fbc54d), STCP(0x761b1211, 0x3158970e), STCP(0x75f42c0b, 0x31b54a5e), STCP(0x75ccfd42, 0x3211df04), STCP(0x75a585cf, 0x326e54c7), STCP(0x757dc5ca, 0x32caab6f), STCP(0x7555bd4c, 0x3326e2c3), STCP(0x752d6c6c, 0x3382fa88), STCP(0x7504d345, 0x33def287), STCP(0x74dbf1ef, 0x343aca87), STCP(0x74b2c884, 0x34968250), STCP(0x7489571c, 0x34f219a8), STCP(0x745f9dd1, 0x354d9057), STCP(0x74359cbd, 0x35a8e625), STCP(0x740b53fb, 0x36041ad9), STCP(0x73e0c3a3, 0x365f2e3b), STCP(0x73b5ebd1, 0x36ba2014), STCP(0x738acc9e, 0x3714f02a), STCP(0x735f6626, 0x376f9e46), STCP(0x7333b883, 0x37ca2a30), STCP(0x7307c3d0, 0x382493b0), STCP(0x72db8828, 0x387eda8e), STCP(0x72af05a7, 0x38d8fe93), STCP(0x72823c67, 0x3932ff87), STCP(0x72552c85, 0x398cdd32), STCP(0x7227d61c, 0x39e6975e), STCP(0x71fa3949, 0x3a402dd2), STCP(0x71cc5626, 0x3a99a057), STCP(0x719e2cd2, 0x3af2eeb7), STCP(0x716fbd68, 0x3b4c18ba), STCP(0x71410805, 0x3ba51e29), STCP(0x71120cc5, 0x3bfdfecd), STCP(0x70e2cbc6, 0x3c56ba70), STCP(0x70b34525, 0x3caf50da), STCP(0x708378ff, 0x3d07c1d6), STCP(0x70536771, 0x3d600d2c), STCP(0x7023109a, 0x3db832a6), STCP(0x6ff27497, 0x3e10320d), STCP(0x6fc19385, 0x3e680b2c), STCP(0x6f906d84, 0x3ebfbdcd), STCP(0x6f5f02b2, 0x3f1749b8), STCP(0x6f2d532c, 0x3f6eaeb8), STCP(0x6efb5f12, 0x3fc5ec98), STCP(0x6ec92683, 0x401d0321), STCP(0x6e96a99d, 0x4073f21d), STCP(0x6e63e87f, 0x40cab958), STCP(0x6e30e34a, 0x4121589b), STCP(0x6dfd9a1c, 0x4177cfb1), STCP(0x6dca0d14, 0x41ce1e65), STCP(0x6d963c54, 0x42244481), STCP(0x6d6227fa, 0x427a41d0), STCP(0x6d2dd027, 0x42d0161e), STCP(0x6cf934fc, 0x4325c135), STCP(0x6cc45698, 0x437b42e1), STCP(0x6c8f351c, 0x43d09aed), STCP(0x6c59d0a9, 0x4425c923), STCP(0x6c242960, 0x447acd50), STCP(0x6bee3f62, 0x44cfa740), STCP(0x6bb812d1, 0x452456bd), STCP(0x6b81a3cd, 0x4578db93), STCP(0x6b4af279, 0x45cd358f), STCP(0x6b13fef5, 0x4621647d), STCP(0x6adcc964, 0x46756828), STCP(0x6aa551e9, 0x46c9405c), STCP(0x6a6d98a4, 0x471cece7), STCP(0x6a359db9, 0x47706d93), STCP(0x69fd614a, 0x47c3c22f), STCP(0x69c4e37a, 0x4816ea86), STCP(0x698c246c, 0x4869e665), STCP(0x69532442, 0x48bcb599), STCP(0x6919e320, 0x490f57ee), STCP(0x68e06129, 0x4961cd33), STCP(0x68a69e81, 0x49b41533), STCP(0x686c9b4b, 0x4a062fbd), STCP(0x683257ab, 0x4a581c9e), STCP(0x67f7d3c5, 0x4aa9dba2), STCP(0x67bd0fbd, 0x4afb6c98), STCP(0x67820bb7, 0x4b4ccf4d), STCP(0x6746c7d8, 0x4b9e0390), STCP(0x670b4444, 0x4bef092d), STCP(0x66cf8120, 0x4c3fdff4), STCP(0x66937e91, 0x4c9087b1), STCP(0x66573cbb, 0x4ce10034), STCP(0x661abbc5, 0x4d31494b), STCP(0x65ddfbd3, 0x4d8162c4), STCP(0x65a0fd0b, 0x4dd14c6e), STCP(0x6563bf92, 0x4e210617), STCP(0x6526438f, 0x4e708f8f), STCP(0x64e88926, 0x4ebfe8a5), STCP(0x64aa907f, 0x4f0f1126), STCP(0x646c59bf, 0x4f5e08e3), STCP(0x642de50d, 0x4faccfab), STCP(0x63ef3290, 0x4ffb654d), STCP(0x63b0426d, 0x5049c999), STCP(0x637114cc, 0x5097fc5e), STCP(0x6331a9d4, 0x50e5fd6d), STCP(0x62f201ac, 0x5133cc94), STCP(0x62b21c7b, 0x518169a5), STCP(0x6271fa69, 0x51ced46e), STCP(0x62319b9d, 0x521c0cc2), STCP(0x61f1003f, 0x5269126e), STCP(0x61b02876, 0x52b5e546), STCP(0x616f146c, 0x53028518), STCP(0x612dc447, 0x534ef1b5), STCP(0x60ec3830, 0x539b2af0), STCP(0x60aa7050, 0x53e73097), STCP(0x60686ccf, 0x5433027d), STCP(0x60262dd6, 0x547ea073), STCP(0x5fe3b38d, 0x54ca0a4b), STCP(0x5fa0fe1f, 0x55153fd4), STCP(0x5f5e0db3, 0x556040e2), STCP(0x5f1ae274, 0x55ab0d46), STCP(0x5ed77c8a, 0x55f5a4d2), STCP(0x5e93dc1f, 0x56400758), STCP(0x5e50015d, 0x568a34a9), STCP(0x5e0bec6e, 0x56d42c99), STCP(0x5dc79d7c, 0x571deefa), STCP(0x5d8314b1, 0x57677b9d), STCP(0x5d3e5237, 0x57b0d256), STCP(0x5cf95638, 0x57f9f2f8), STCP(0x5cb420e0, 0x5842dd54), STCP(0x5c6eb258, 0x588b9140), STCP(0x5c290acc, 0x58d40e8c), STCP(0x5be32a67, 0x591c550e), STCP(0x5b9d1154, 0x59646498), STCP(0x5b56bfbd, 0x59ac3cfd), STCP(0x5b1035cf, 0x59f3de12), STCP(0x5ac973b5, 0x5a3b47ab), STCP(0x5a82799a, 0x5a82799a), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_STP SineTable1024[] = { STCP(0x7fffffff, 0x00000000), STCP(0x7ffff621, 0x003243f5), STCP(0x7fffd886, 0x006487e3), STCP(0x7fffa72c, 0x0096cbc1), STCP(0x7fff6216, 0x00c90f88), STCP(0x7fff0943, 0x00fb5330), STCP(0x7ffe9cb2, 0x012d96b1), STCP(0x7ffe1c65, 0x015fda03), STCP(0x7ffd885a, 0x01921d20), STCP(0x7ffce093, 0x01c45ffe), STCP(0x7ffc250f, 0x01f6a297), STCP(0x7ffb55ce, 0x0228e4e2), STCP(0x7ffa72d1, 0x025b26d7), STCP(0x7ff97c18, 0x028d6870), STCP(0x7ff871a2, 0x02bfa9a4), STCP(0x7ff75370, 0x02f1ea6c), STCP(0x7ff62182, 0x03242abf), STCP(0x7ff4dbd9, 0x03566a96), STCP(0x7ff38274, 0x0388a9ea), STCP(0x7ff21553, 0x03bae8b2), STCP(0x7ff09478, 0x03ed26e6), STCP(0x7feeffe1, 0x041f6480), STCP(0x7fed5791, 0x0451a177), STCP(0x7feb9b85, 0x0483ddc3), STCP(0x7fe9cbc0, 0x04b6195d), STCP(0x7fe7e841, 0x04e8543e), STCP(0x7fe5f108, 0x051a8e5c), STCP(0x7fe3e616, 0x054cc7b1), STCP(0x7fe1c76b, 0x057f0035), STCP(0x7fdf9508, 0x05b137df), STCP(0x7fdd4eec, 0x05e36ea9), STCP(0x7fdaf519, 0x0615a48b), STCP(0x7fd8878e, 0x0647d97c), STCP(0x7fd6064c, 0x067a0d76), STCP(0x7fd37153, 0x06ac406f), STCP(0x7fd0c8a3, 0x06de7262), STCP(0x7fce0c3e, 0x0710a345), STCP(0x7fcb3c23, 0x0742d311), STCP(0x7fc85854, 0x077501be), STCP(0x7fc560cf, 0x07a72f45), STCP(0x7fc25596, 0x07d95b9e), STCP(0x7fbf36aa, 0x080b86c2), STCP(0x7fbc040a, 0x083db0a7), STCP(0x7fb8bdb8, 0x086fd947), STCP(0x7fb563b3, 0x08a2009a), STCP(0x7fb1f5fc, 0x08d42699), STCP(0x7fae7495, 0x09064b3a), STCP(0x7faadf7c, 0x09386e78), STCP(0x7fa736b4, 0x096a9049), STCP(0x7fa37a3c, 0x099cb0a7), STCP(0x7f9faa15, 0x09cecf89), STCP(0x7f9bc640, 0x0a00ece8), STCP(0x7f97cebd, 0x0a3308bd), STCP(0x7f93c38c, 0x0a6522fe), STCP(0x7f8fa4b0, 0x0a973ba5), STCP(0x7f8b7227, 0x0ac952aa), STCP(0x7f872bf3, 0x0afb6805), STCP(0x7f82d214, 0x0b2d7baf), STCP(0x7f7e648c, 0x0b5f8d9f), STCP(0x7f79e35a, 0x0b919dcf), STCP(0x7f754e80, 0x0bc3ac35), STCP(0x7f70a5fe, 0x0bf5b8cb), STCP(0x7f6be9d4, 0x0c27c389), STCP(0x7f671a05, 0x0c59cc68), STCP(0x7f62368f, 0x0c8bd35e), STCP(0x7f5d3f75, 0x0cbdd865), STCP(0x7f5834b7, 0x0cefdb76), STCP(0x7f531655, 0x0d21dc87), STCP(0x7f4de451, 0x0d53db92), STCP(0x7f489eaa, 0x0d85d88f), STCP(0x7f434563, 0x0db7d376), STCP(0x7f3dd87c, 0x0de9cc40), STCP(0x7f3857f6, 0x0e1bc2e4), STCP(0x7f32c3d1, 0x0e4db75b), STCP(0x7f2d1c0e, 0x0e7fa99e), STCP(0x7f2760af, 0x0eb199a4), STCP(0x7f2191b4, 0x0ee38766), STCP(0x7f1baf1e, 0x0f1572dc), STCP(0x7f15b8ee, 0x0f475bff), STCP(0x7f0faf25, 0x0f7942c7), STCP(0x7f0991c4, 0x0fab272b), STCP(0x7f0360cb, 0x0fdd0926), STCP(0x7efd1c3c, 0x100ee8ad), STCP(0x7ef6c418, 0x1040c5bb), STCP(0x7ef05860, 0x1072a048), STCP(0x7ee9d914, 0x10a4784b), STCP(0x7ee34636, 0x10d64dbd), STCP(0x7edc9fc6, 0x11082096), STCP(0x7ed5e5c6, 0x1139f0cf), STCP(0x7ecf1837, 0x116bbe60), STCP(0x7ec8371a, 0x119d8941), STCP(0x7ec14270, 0x11cf516a), STCP(0x7eba3a39, 0x120116d5), STCP(0x7eb31e78, 0x1232d979), STCP(0x7eabef2c, 0x1264994e), STCP(0x7ea4ac58, 0x1296564d), STCP(0x7e9d55fc, 0x12c8106f), STCP(0x7e95ec1a, 0x12f9c7aa), STCP(0x7e8e6eb2, 0x132b7bf9), STCP(0x7e86ddc6, 0x135d2d53), STCP(0x7e7f3957, 0x138edbb1), STCP(0x7e778166, 0x13c0870a), STCP(0x7e6fb5f4, 0x13f22f58), STCP(0x7e67d703, 0x1423d492), STCP(0x7e5fe493, 0x145576b1), STCP(0x7e57dea7, 0x148715ae), STCP(0x7e4fc53e, 0x14b8b17f), STCP(0x7e47985b, 0x14ea4a1f), STCP(0x7e3f57ff, 0x151bdf86), STCP(0x7e37042a, 0x154d71aa), STCP(0x7e2e9cdf, 0x157f0086), STCP(0x7e26221f, 0x15b08c12), STCP(0x7e1d93ea, 0x15e21445), STCP(0x7e14f242, 0x16139918), STCP(0x7e0c3d29, 0x16451a83), STCP(0x7e0374a0, 0x1676987f), STCP(0x7dfa98a8, 0x16a81305), STCP(0x7df1a942, 0x16d98a0c), STCP(0x7de8a670, 0x170afd8d), STCP(0x7ddf9034, 0x173c6d80), STCP(0x7dd6668f, 0x176dd9de), STCP(0x7dcd2981, 0x179f429f), STCP(0x7dc3d90d, 0x17d0a7bc), STCP(0x7dba7534, 0x1802092c), STCP(0x7db0fdf8, 0x183366e9), STCP(0x7da77359, 0x1864c0ea), STCP(0x7d9dd55a, 0x18961728), STCP(0x7d9423fc, 0x18c7699b), STCP(0x7d8a5f40, 0x18f8b83c), STCP(0x7d808728, 0x192a0304), STCP(0x7d769bb5, 0x195b49ea), STCP(0x7d6c9ce9, 0x198c8ce7), STCP(0x7d628ac6, 0x19bdcbf3), STCP(0x7d58654d, 0x19ef0707), STCP(0x7d4e2c7f, 0x1a203e1b), STCP(0x7d43e05e, 0x1a517128), STCP(0x7d3980ec, 0x1a82a026), STCP(0x7d2f0e2b, 0x1ab3cb0d), STCP(0x7d24881b, 0x1ae4f1d6), STCP(0x7d19eebf, 0x1b161479), STCP(0x7d0f4218, 0x1b4732ef), STCP(0x7d048228, 0x1b784d30), STCP(0x7cf9aef0, 0x1ba96335), STCP(0x7ceec873, 0x1bda74f6), STCP(0x7ce3ceb2, 0x1c0b826a), STCP(0x7cd8c1ae, 0x1c3c8b8c), STCP(0x7ccda169, 0x1c6d9053), STCP(0x7cc26de5, 0x1c9e90b8), STCP(0x7cb72724, 0x1ccf8cb3), STCP(0x7cabcd28, 0x1d00843d), STCP(0x7ca05ff1, 0x1d31774d), STCP(0x7c94df83, 0x1d6265dd), STCP(0x7c894bde, 0x1d934fe5), STCP(0x7c7da505, 0x1dc4355e), STCP(0x7c71eaf9, 0x1df5163f), STCP(0x7c661dbc, 0x1e25f282), STCP(0x7c5a3d50, 0x1e56ca1e), STCP(0x7c4e49b7, 0x1e879d0d), STCP(0x7c4242f2, 0x1eb86b46), STCP(0x7c362904, 0x1ee934c3), STCP(0x7c29fbee, 0x1f19f97b), STCP(0x7c1dbbb3, 0x1f4ab968), STCP(0x7c116853, 0x1f7b7481), STCP(0x7c0501d2, 0x1fac2abf), STCP(0x7bf88830, 0x1fdcdc1b), STCP(0x7bebfb70, 0x200d888d), STCP(0x7bdf5b94, 0x203e300d), STCP(0x7bd2a89e, 0x206ed295), STCP(0x7bc5e290, 0x209f701c), STCP(0x7bb9096b, 0x20d0089c), STCP(0x7bac1d31, 0x21009c0c), STCP(0x7b9f1de6, 0x21312a65), STCP(0x7b920b89, 0x2161b3a0), STCP(0x7b84e61f, 0x219237b5), STCP(0x7b77ada8, 0x21c2b69c), STCP(0x7b6a6227, 0x21f3304f), STCP(0x7b5d039e, 0x2223a4c5), STCP(0x7b4f920e, 0x225413f8), STCP(0x7b420d7a, 0x22847de0), STCP(0x7b3475e5, 0x22b4e274), STCP(0x7b26cb4f, 0x22e541af), STCP(0x7b190dbc, 0x23159b88), STCP(0x7b0b3d2c, 0x2345eff8), STCP(0x7afd59a4, 0x23763ef7), STCP(0x7aef6323, 0x23a6887f), STCP(0x7ae159ae, 0x23d6cc87), STCP(0x7ad33d45, 0x24070b08), STCP(0x7ac50dec, 0x243743fa), STCP(0x7ab6cba4, 0x24677758), STCP(0x7aa8766f, 0x2497a517), STCP(0x7a9a0e50, 0x24c7cd33), STCP(0x7a8b9348, 0x24f7efa2), STCP(0x7a7d055b, 0x25280c5e), STCP(0x7a6e648a, 0x2558235f), STCP(0x7a5fb0d8, 0x2588349d), STCP(0x7a50ea47, 0x25b84012), STCP(0x7a4210d8, 0x25e845b6), STCP(0x7a332490, 0x26184581), STCP(0x7a24256f, 0x26483f6c), STCP(0x7a151378, 0x26783370), STCP(0x7a05eead, 0x26a82186), STCP(0x79f6b711, 0x26d809a5), STCP(0x79e76ca7, 0x2707ebc7), STCP(0x79d80f6f, 0x2737c7e3), STCP(0x79c89f6e, 0x27679df4), STCP(0x79b91ca4, 0x27976df1), STCP(0x79a98715, 0x27c737d3), STCP(0x7999dec4, 0x27f6fb92), STCP(0x798a23b1, 0x2826b928), STCP(0x797a55e0, 0x2856708d), STCP(0x796a7554, 0x288621b9), STCP(0x795a820e, 0x28b5cca5), STCP(0x794a7c12, 0x28e5714b), STCP(0x793a6361, 0x29150fa1), STCP(0x792a37fe, 0x2944a7a2), STCP(0x7919f9ec, 0x29743946), STCP(0x7909a92d, 0x29a3c485), STCP(0x78f945c3, 0x29d34958), STCP(0x78e8cfb2, 0x2a02c7b8), STCP(0x78d846fb, 0x2a323f9e), STCP(0x78c7aba2, 0x2a61b101), STCP(0x78b6fda8, 0x2a911bdc), STCP(0x78a63d11, 0x2ac08026), STCP(0x789569df, 0x2aefddd8), STCP(0x78848414, 0x2b1f34eb), STCP(0x78738bb3, 0x2b4e8558), STCP(0x786280bf, 0x2b7dcf17), STCP(0x7851633b, 0x2bad1221), STCP(0x78403329, 0x2bdc4e6f), STCP(0x782ef08b, 0x2c0b83fa), STCP(0x781d9b65, 0x2c3ab2b9), STCP(0x780c33b8, 0x2c69daa6), STCP(0x77fab989, 0x2c98fbba), STCP(0x77e92cd9, 0x2cc815ee), STCP(0x77d78daa, 0x2cf72939), STCP(0x77c5dc01, 0x2d263596), STCP(0x77b417df, 0x2d553afc), STCP(0x77a24148, 0x2d843964), STCP(0x7790583e, 0x2db330c7), STCP(0x777e5cc3, 0x2de2211e), STCP(0x776c4edb, 0x2e110a62), STCP(0x775a2e89, 0x2e3fec8b), STCP(0x7747fbce, 0x2e6ec792), STCP(0x7735b6af, 0x2e9d9b70), STCP(0x77235f2d, 0x2ecc681e), STCP(0x7710f54c, 0x2efb2d95), STCP(0x76fe790e, 0x2f29ebcc), STCP(0x76ebea77, 0x2f58a2be), STCP(0x76d94989, 0x2f875262), STCP(0x76c69647, 0x2fb5fab2), STCP(0x76b3d0b4, 0x2fe49ba7), STCP(0x76a0f8d2, 0x30133539), STCP(0x768e0ea6, 0x3041c761), STCP(0x767b1231, 0x30705217), STCP(0x76680376, 0x309ed556), STCP(0x7654e279, 0x30cd5115), STCP(0x7641af3d, 0x30fbc54d), STCP(0x762e69c4, 0x312a31f8), STCP(0x761b1211, 0x3158970e), STCP(0x7607a828, 0x3186f487), STCP(0x75f42c0b, 0x31b54a5e), STCP(0x75e09dbd, 0x31e39889), STCP(0x75ccfd42, 0x3211df04), STCP(0x75b94a9c, 0x32401dc6), STCP(0x75a585cf, 0x326e54c7), STCP(0x7591aedd, 0x329c8402), STCP(0x757dc5ca, 0x32caab6f), STCP(0x7569ca99, 0x32f8cb07), STCP(0x7555bd4c, 0x3326e2c3), STCP(0x75419de7, 0x3354f29b), STCP(0x752d6c6c, 0x3382fa88), STCP(0x751928e0, 0x33b0fa84), STCP(0x7504d345, 0x33def287), STCP(0x74f06b9e, 0x340ce28b), STCP(0x74dbf1ef, 0x343aca87), STCP(0x74c7663a, 0x3468aa76), STCP(0x74b2c884, 0x34968250), STCP(0x749e18cd, 0x34c4520d), STCP(0x7489571c, 0x34f219a8), STCP(0x74748371, 0x351fd918), STCP(0x745f9dd1, 0x354d9057), STCP(0x744aa63f, 0x357b3f5d), STCP(0x74359cbd, 0x35a8e625), STCP(0x74208150, 0x35d684a6), STCP(0x740b53fb, 0x36041ad9), STCP(0x73f614c0, 0x3631a8b8), STCP(0x73e0c3a3, 0x365f2e3b), STCP(0x73cb60a8, 0x368cab5c), STCP(0x73b5ebd1, 0x36ba2014), STCP(0x73a06522, 0x36e78c5b), STCP(0x738acc9e, 0x3714f02a), STCP(0x73752249, 0x37424b7b), STCP(0x735f6626, 0x376f9e46), STCP(0x73499838, 0x379ce885), STCP(0x7333b883, 0x37ca2a30), STCP(0x731dc70a, 0x37f76341), STCP(0x7307c3d0, 0x382493b0), STCP(0x72f1aed9, 0x3851bb77), STCP(0x72db8828, 0x387eda8e), STCP(0x72c54fc1, 0x38abf0ef), STCP(0x72af05a7, 0x38d8fe93), STCP(0x7298a9dd, 0x39060373), STCP(0x72823c67, 0x3932ff87), STCP(0x726bbd48, 0x395ff2c9), STCP(0x72552c85, 0x398cdd32), STCP(0x723e8a20, 0x39b9bebc), STCP(0x7227d61c, 0x39e6975e), STCP(0x7211107e, 0x3a136712), STCP(0x71fa3949, 0x3a402dd2), STCP(0x71e35080, 0x3a6ceb96), STCP(0x71cc5626, 0x3a99a057), STCP(0x71b54a41, 0x3ac64c0f), STCP(0x719e2cd2, 0x3af2eeb7), STCP(0x7186fdde, 0x3b1f8848), STCP(0x716fbd68, 0x3b4c18ba), STCP(0x71586b74, 0x3b78a007), STCP(0x71410805, 0x3ba51e29), STCP(0x7129931f, 0x3bd19318), STCP(0x71120cc5, 0x3bfdfecd), STCP(0x70fa74fc, 0x3c2a6142), STCP(0x70e2cbc6, 0x3c56ba70), STCP(0x70cb1128, 0x3c830a50), STCP(0x70b34525, 0x3caf50da), STCP(0x709b67c0, 0x3cdb8e09), STCP(0x708378ff, 0x3d07c1d6), STCP(0x706b78e3, 0x3d33ec39), STCP(0x70536771, 0x3d600d2c), STCP(0x703b44ad, 0x3d8c24a8), STCP(0x7023109a, 0x3db832a6), STCP(0x700acb3c, 0x3de4371f), STCP(0x6ff27497, 0x3e10320d), STCP(0x6fda0cae, 0x3e3c2369), STCP(0x6fc19385, 0x3e680b2c), STCP(0x6fa90921, 0x3e93e950), STCP(0x6f906d84, 0x3ebfbdcd), STCP(0x6f77c0b3, 0x3eeb889c), STCP(0x6f5f02b2, 0x3f1749b8), STCP(0x6f463383, 0x3f430119), STCP(0x6f2d532c, 0x3f6eaeb8), STCP(0x6f1461b0, 0x3f9a5290), STCP(0x6efb5f12, 0x3fc5ec98), STCP(0x6ee24b57, 0x3ff17cca), STCP(0x6ec92683, 0x401d0321), STCP(0x6eaff099, 0x40487f94), STCP(0x6e96a99d, 0x4073f21d), STCP(0x6e7d5193, 0x409f5ab6), STCP(0x6e63e87f, 0x40cab958), STCP(0x6e4a6e66, 0x40f60dfb), STCP(0x6e30e34a, 0x4121589b), STCP(0x6e174730, 0x414c992f), STCP(0x6dfd9a1c, 0x4177cfb1), STCP(0x6de3dc11, 0x41a2fc1a), STCP(0x6dca0d14, 0x41ce1e65), STCP(0x6db02d29, 0x41f93689), STCP(0x6d963c54, 0x42244481), STCP(0x6d7c3a98, 0x424f4845), STCP(0x6d6227fa, 0x427a41d0), STCP(0x6d48047e, 0x42a5311b), STCP(0x6d2dd027, 0x42d0161e), STCP(0x6d138afb, 0x42faf0d4), STCP(0x6cf934fc, 0x4325c135), STCP(0x6cdece2f, 0x4350873c), STCP(0x6cc45698, 0x437b42e1), STCP(0x6ca9ce3b, 0x43a5f41e), STCP(0x6c8f351c, 0x43d09aed), STCP(0x6c748b3f, 0x43fb3746), STCP(0x6c59d0a9, 0x4425c923), STCP(0x6c3f055d, 0x4450507e), STCP(0x6c242960, 0x447acd50), STCP(0x6c093cb6, 0x44a53f93), STCP(0x6bee3f62, 0x44cfa740), STCP(0x6bd3316a, 0x44fa0450), STCP(0x6bb812d1, 0x452456bd), STCP(0x6b9ce39b, 0x454e9e80), STCP(0x6b81a3cd, 0x4578db93), STCP(0x6b66536b, 0x45a30df0), STCP(0x6b4af279, 0x45cd358f), STCP(0x6b2f80fb, 0x45f7526b), STCP(0x6b13fef5, 0x4621647d), STCP(0x6af86c6c, 0x464b6bbe), STCP(0x6adcc964, 0x46756828), STCP(0x6ac115e2, 0x469f59b4), STCP(0x6aa551e9, 0x46c9405c), STCP(0x6a897d7d, 0x46f31c1a), STCP(0x6a6d98a4, 0x471cece7), STCP(0x6a51a361, 0x4746b2bc), STCP(0x6a359db9, 0x47706d93), STCP(0x6a1987b0, 0x479a1d67), STCP(0x69fd614a, 0x47c3c22f), STCP(0x69e12a8c, 0x47ed5be6), STCP(0x69c4e37a, 0x4816ea86), STCP(0x69a88c19, 0x48406e08), STCP(0x698c246c, 0x4869e665), STCP(0x696fac78, 0x48935397), STCP(0x69532442, 0x48bcb599), STCP(0x69368bce, 0x48e60c62), STCP(0x6919e320, 0x490f57ee), STCP(0x68fd2a3d, 0x49389836), STCP(0x68e06129, 0x4961cd33), STCP(0x68c387e9, 0x498af6df), STCP(0x68a69e81, 0x49b41533), STCP(0x6889a4f6, 0x49dd282a), STCP(0x686c9b4b, 0x4a062fbd), STCP(0x684f8186, 0x4a2f2be6), STCP(0x683257ab, 0x4a581c9e), STCP(0x68151dbe, 0x4a8101de), STCP(0x67f7d3c5, 0x4aa9dba2), STCP(0x67da79c3, 0x4ad2a9e2), STCP(0x67bd0fbd, 0x4afb6c98), STCP(0x679f95b7, 0x4b2423be), STCP(0x67820bb7, 0x4b4ccf4d), STCP(0x676471c0, 0x4b756f40), STCP(0x6746c7d8, 0x4b9e0390), STCP(0x67290e02, 0x4bc68c36), STCP(0x670b4444, 0x4bef092d), STCP(0x66ed6aa1, 0x4c177a6e), STCP(0x66cf8120, 0x4c3fdff4), STCP(0x66b187c3, 0x4c6839b7), STCP(0x66937e91, 0x4c9087b1), STCP(0x6675658c, 0x4cb8c9dd), STCP(0x66573cbb, 0x4ce10034), STCP(0x66390422, 0x4d092ab0), STCP(0x661abbc5, 0x4d31494b), STCP(0x65fc63a9, 0x4d595bfe), STCP(0x65ddfbd3, 0x4d8162c4), STCP(0x65bf8447, 0x4da95d96), STCP(0x65a0fd0b, 0x4dd14c6e), STCP(0x65826622, 0x4df92f46), STCP(0x6563bf92, 0x4e210617), STCP(0x6545095f, 0x4e48d0dd), STCP(0x6526438f, 0x4e708f8f), STCP(0x65076e25, 0x4e984229), STCP(0x64e88926, 0x4ebfe8a5), STCP(0x64c99498, 0x4ee782fb), STCP(0x64aa907f, 0x4f0f1126), STCP(0x648b7ce0, 0x4f369320), STCP(0x646c59bf, 0x4f5e08e3), STCP(0x644d2722, 0x4f857269), STCP(0x642de50d, 0x4faccfab), STCP(0x640e9386, 0x4fd420a4), STCP(0x63ef3290, 0x4ffb654d), STCP(0x63cfc231, 0x50229da1), STCP(0x63b0426d, 0x5049c999), STCP(0x6390b34a, 0x5070e92f), STCP(0x637114cc, 0x5097fc5e), STCP(0x635166f9, 0x50bf031f), STCP(0x6331a9d4, 0x50e5fd6d), STCP(0x6311dd64, 0x510ceb40), STCP(0x62f201ac, 0x5133cc94), STCP(0x62d216b3, 0x515aa162), STCP(0x62b21c7b, 0x518169a5), STCP(0x6292130c, 0x51a82555), STCP(0x6271fa69, 0x51ced46e), STCP(0x6251d298, 0x51f576ea), STCP(0x62319b9d, 0x521c0cc2), STCP(0x6211557e, 0x524295f0), STCP(0x61f1003f, 0x5269126e), STCP(0x61d09be5, 0x528f8238), STCP(0x61b02876, 0x52b5e546), STCP(0x618fa5f7, 0x52dc3b92), STCP(0x616f146c, 0x53028518), STCP(0x614e73da, 0x5328c1d0), STCP(0x612dc447, 0x534ef1b5), STCP(0x610d05b7, 0x537514c2), STCP(0x60ec3830, 0x539b2af0), STCP(0x60cb5bb7, 0x53c13439), STCP(0x60aa7050, 0x53e73097), STCP(0x60897601, 0x540d2005), STCP(0x60686ccf, 0x5433027d), STCP(0x604754bf, 0x5458d7f9), STCP(0x60262dd6, 0x547ea073), STCP(0x6004f819, 0x54a45be6), STCP(0x5fe3b38d, 0x54ca0a4b), STCP(0x5fc26038, 0x54efab9c), STCP(0x5fa0fe1f, 0x55153fd4), STCP(0x5f7f8d46, 0x553ac6ee), STCP(0x5f5e0db3, 0x556040e2), STCP(0x5f3c7f6b, 0x5585adad), STCP(0x5f1ae274, 0x55ab0d46), STCP(0x5ef936d1, 0x55d05faa), STCP(0x5ed77c8a, 0x55f5a4d2), STCP(0x5eb5b3a2, 0x561adcb9), STCP(0x5e93dc1f, 0x56400758), STCP(0x5e71f606, 0x566524aa), STCP(0x5e50015d, 0x568a34a9), STCP(0x5e2dfe29, 0x56af3750), STCP(0x5e0bec6e, 0x56d42c99), STCP(0x5de9cc33, 0x56f9147e), STCP(0x5dc79d7c, 0x571deefa), STCP(0x5da5604f, 0x5742bc06), STCP(0x5d8314b1, 0x57677b9d), STCP(0x5d60baa7, 0x578c2dba), STCP(0x5d3e5237, 0x57b0d256), STCP(0x5d1bdb65, 0x57d5696d), STCP(0x5cf95638, 0x57f9f2f8), STCP(0x5cd6c2b5, 0x581e6ef1), STCP(0x5cb420e0, 0x5842dd54), STCP(0x5c9170bf, 0x58673e1b), STCP(0x5c6eb258, 0x588b9140), STCP(0x5c4be5b0, 0x58afd6bd), STCP(0x5c290acc, 0x58d40e8c), STCP(0x5c0621b2, 0x58f838a9), STCP(0x5be32a67, 0x591c550e), STCP(0x5bc024f0, 0x594063b5), STCP(0x5b9d1154, 0x59646498), STCP(0x5b79ef96, 0x598857b2), STCP(0x5b56bfbd, 0x59ac3cfd), STCP(0x5b3381ce, 0x59d01475), STCP(0x5b1035cf, 0x59f3de12), STCP(0x5aecdbc5, 0x5a1799d1), STCP(0x5ac973b5, 0x5a3b47ab), STCP(0x5aa5fda5, 0x5a5ee79a), STCP(0x5a82799a, 0x5a82799a), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_STB RotVectorReal6[] = { STC(0x40000000), STC(0xc0000000), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_STB RotVectorImag6[] = { STC(0x6ed9eba1), STC(0x6ed9eba1), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_STB RotVectorReal12[] = { STC(0x6ed9eba1), STC(0x40000000), STC(0x40000000), STC(0xc0000000), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_STB RotVectorImag12[] = { STC(0x40000000), STC(0x6ed9eba1), STC(0x6ed9eba1), STC(0x6ed9eba1), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_STB RotVectorReal24[] = { STC(0x7ba3751d), STC(0x6ed9eba1), STC(0x5a82799a), STC(0x40000000), STC(0x2120fb83), STC(0x00000000), STC(0xdedf047d), STC(0xc0000000), STC(0xa57d8666), STC(0x9126145f), STC(0x845c8ae3), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_STB RotVectorImag24[] = { STC(0x2120fb83), STC(0x40000000), STC(0x5a82799a), STC(0x6ed9eba1), STC(0x7ba3751d), STC(0x7fffffff), STC(0x7ba3751d), STC(0x6ed9eba1), STC(0x5a82799a), STC(0x40000000), STC(0x2120fb83), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_STB RotVectorReal48[] = { STC(0x7ee7aa4c), STC(0x7ba3751d), STC(0x7641af3d), STC(0x7ba3751d), STC(0x6ed9eba1), STC(0x5a82799a), STC(0x7641af3d), STC(0x5a82799a), STC(0x30fbc54d), STC(0x6ed9eba1), STC(0x40000000), STC(0x00000000), STC(0x658c9a2d), STC(0x2120fb83), STC(0xcf043ab3), STC(0x5a82799a), STC(0x00000000), STC(0xa57d8666), STC(0x4debe4fe), STC(0xdedf047d), STC(0x89be50c3), STC(0x40000000), STC(0xc0000000), STC(0x80000000), STC(0x30fbc54d), STC(0xa57d8666), STC(0x89be50c3), STC(0x2120fb83), STC(0x9126145f), STC(0xa57d8666), STC(0x10b5150f), STC(0x845c8ae3), STC(0xcf043ab3), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_STB RotVectorImag48[] = { STC(0x10b5150f), STC(0x2120fb83), STC(0x30fbc54d), STC(0x2120fb83), STC(0x40000000), STC(0x5a82799a), STC(0x30fbc54d), STC(0x5a82799a), STC(0x7641af3d), STC(0x40000000), STC(0x6ed9eba1), STC(0x7fffffff), STC(0x4debe4fe), STC(0x7ba3751d), STC(0x7641af3d), STC(0x5a82799a), STC(0x7fffffff), STC(0x5a82799a), STC(0x658c9a2d), STC(0x7ba3751d), STC(0x30fbc54d), STC(0x6ed9eba1), STC(0x6ed9eba1), STC(0x00000000), STC(0x7641af3d), STC(0x5a82799a), STC(0xcf043ab3), STC(0x7ba3751d), STC(0x40000000), STC(0xa57d8666), STC(0x7ee7aa4c), STC(0x2120fb83), STC(0x89be50c3), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_STB RotVectorReal80[] = { STC(0x7f9afcb9), STC(0x7e6c9251), STC(0x7c769e18), STC(0x79bc384d), STC(0x7e6c9251), STC(0x79bc384d), STC(0x720c8075), STC(0x678dde6e), STC(0x7c769e18), STC(0x720c8075), STC(0x6154fb91), STC(0x4b3c8c12), STC(0x79bc384d), STC(0x678dde6e), STC(0x4b3c8c12), STC(0x278dde6e), STC(0x7641af3d), STC(0x5a82799a), STC(0x30fbc54d), STC(0x00000000), STC(0x720c8075), STC(0x4b3c8c12), STC(0x14060b68), STC(0xd8722192), STC(0x6d23501b), STC(0x3a1c5c57), STC(0xf5f50d67), STC(0xb4c373ee), STC(0x678dde6e), STC(0x278dde6e), STC(0xd8722192), STC(0x98722192), STC(0x6154fb91), STC(0x14060b68), STC(0xbd1ec45c), STC(0x8643c7b3), STC(0x5a82799a), STC(0x00000000), STC(0xa57d8666), STC(0x80000000), STC(0x53211d18), STC(0xebf9f498), STC(0x92dcafe5), STC(0x8643c7b3), STC(0x4b3c8c12), STC(0xd8722192), STC(0x8643c7b3), STC(0x98722192), STC(0x42e13ba4), STC(0xc5e3a3a9), STC(0x80650347), STC(0xb4c373ee), STC(0x3a1c5c57), STC(0xb4c373ee), STC(0x81936daf), STC(0xd8722192), STC(0x30fbc54d), STC(0xa57d8666), STC(0x89be50c3), STC(0x00000000), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_STB RotVectorImag80[] = { STC(0x0a0af299), STC(0x14060b68), STC(0x1de189a6), STC(0x278dde6e), STC(0x14060b68), STC(0x278dde6e), STC(0x3a1c5c57), STC(0x4b3c8c12), STC(0x1de189a6), STC(0x3a1c5c57), STC(0x53211d18), STC(0x678dde6e), STC(0x278dde6e), STC(0x4b3c8c12), STC(0x678dde6e), STC(0x79bc384d), STC(0x30fbc54d), STC(0x5a82799a), STC(0x7641af3d), STC(0x7fffffff), STC(0x3a1c5c57), STC(0x678dde6e), STC(0x7e6c9251), STC(0x79bc384d), STC(0x42e13ba4), STC(0x720c8075), STC(0x7f9afcb9), STC(0x678dde6e), STC(0x4b3c8c12), STC(0x79bc384d), STC(0x79bc384d), STC(0x4b3c8c12), STC(0x53211d18), STC(0x7e6c9251), STC(0x6d23501b), STC(0x278dde6e), STC(0x5a82799a), STC(0x7fffffff), STC(0x5a82799a), STC(0x00000000), STC(0x6154fb91), STC(0x7e6c9251), STC(0x42e13ba4), STC(0xd8722192), STC(0x678dde6e), STC(0x79bc384d), STC(0x278dde6e), STC(0xb4c373ee), STC(0x6d23501b), STC(0x720c8075), STC(0x0a0af299), STC(0x98722192), STC(0x720c8075), STC(0x678dde6e), STC(0xebf9f498), STC(0x8643c7b3), STC(0x7641af3d), STC(0x5a82799a), STC(0xcf043ab3), STC(0x80000000), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_STB RotVectorReal96[] = { STC(0x7fb9d759), STC(0x7ee7aa4c), STC(0x7ee7aa4c), STC(0x7ba3751d), STC(0x7d8a5f40), STC(0x7641af3d), STC(0x7ba3751d), STC(0x6ed9eba1), STC(0x793501a9), STC(0x658c9a2d), STC(0x7641af3d), STC(0x5a82799a), STC(0x72ccb9db), STC(0x4debe4fe), STC(0x6ed9eba1), STC(0x40000000), STC(0x6a6d98a4), STC(0x30fbc54d), STC(0x658c9a2d), STC(0x2120fb83), STC(0x603c496c), STC(0x10b5150f), STC(0x5a82799a), STC(0x00000000), STC(0x54657194), STC(0xef4aeaf1), STC(0x4debe4fe), STC(0xdedf047d), STC(0x471cece7), STC(0xcf043ab3), STC(0x40000000), STC(0xc0000000), STC(0x389cea72), STC(0xb2141b02), STC(0x30fbc54d), STC(0xa57d8666), STC(0x2924edac), STC(0x9a7365d3), STC(0x2120fb83), STC(0x9126145f), STC(0x18f8b83c), STC(0x89be50c3), STC(0x10b5150f), STC(0x845c8ae3), STC(0x085f2137), STC(0x811855b4), STC(0x00000000), STC(0x80000000), STC(0xf7a0dec9), STC(0x811855b4), STC(0xef4aeaf1), STC(0x845c8ae3), STC(0xe70747c4), STC(0x89be50c3), STC(0xdedf047d), STC(0x9126145f), STC(0xd6db1254), STC(0x9a7365d3), STC(0xcf043ab3), STC(0xa57d8666), STC(0xc763158e), STC(0xb2141b02), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_STB RotVectorImag96[] = { STC(0x085f2137), STC(0x10b5150f), STC(0x10b5150f), STC(0x2120fb83), STC(0x18f8b83c), STC(0x30fbc54d), STC(0x2120fb83), STC(0x40000000), STC(0x2924edac), STC(0x4debe4fe), STC(0x30fbc54d), STC(0x5a82799a), STC(0x389cea72), STC(0x658c9a2d), STC(0x40000000), STC(0x6ed9eba1), STC(0x471cece7), STC(0x7641af3d), STC(0x4debe4fe), STC(0x7ba3751d), STC(0x54657194), STC(0x7ee7aa4c), STC(0x5a82799a), STC(0x7fffffff), STC(0x603c496c), STC(0x7ee7aa4c), STC(0x658c9a2d), STC(0x7ba3751d), STC(0x6a6d98a4), STC(0x7641af3d), STC(0x6ed9eba1), STC(0x6ed9eba1), STC(0x72ccb9db), STC(0x658c9a2d), STC(0x7641af3d), STC(0x5a82799a), STC(0x793501a9), STC(0x4debe4fe), STC(0x7ba3751d), STC(0x40000000), STC(0x7d8a5f40), STC(0x30fbc54d), STC(0x7ee7aa4c), STC(0x2120fb83), STC(0x7fb9d759), STC(0x10b5150f), STC(0x7fffffff), STC(0x00000000), STC(0x7fb9d759), STC(0xef4aeaf1), STC(0x7ee7aa4c), STC(0xdedf047d), STC(0x7d8a5f40), STC(0xcf043ab3), STC(0x7ba3751d), STC(0xc0000000), STC(0x793501a9), STC(0xb2141b02), STC(0x7641af3d), STC(0xa57d8666), STC(0x72ccb9db), STC(0x9a7365d3), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_STB RotVectorReal384[] = { STC(0x7ffb9d15), STC(0x7fee74a2), STC(0x7fd8878e), STC(0x7fb9d759), STC(0x7f92661d), STC(0x7f62368f), STC(0x7f294bfd), STC(0x7ee7aa4c), STC(0x7e9d55fc), STC(0x7e4a5426), STC(0x7deeaa7a), STC(0x7fee74a2), STC(0x7fb9d759), STC(0x7f62368f), STC(0x7ee7aa4c), STC(0x7e4a5426), STC(0x7d8a5f40), STC(0x7ca80038), STC(0x7ba3751d), STC(0x7a7d055b), STC(0x793501a9), STC(0x77cbc3f2), STC(0x7fd8878e), STC(0x7f62368f), STC(0x7e9d55fc), STC(0x7d8a5f40), STC(0x7c29fbee), STC(0x7a7d055b), STC(0x78848414), STC(0x7641af3d), STC(0x73b5ebd1), STC(0x70e2cbc6), STC(0x6dca0d14), STC(0x7fb9d759), STC(0x7ee7aa4c), STC(0x7d8a5f40), STC(0x7ba3751d), STC(0x793501a9), STC(0x7641af3d), STC(0x72ccb9db), STC(0x6ed9eba1), STC(0x6a6d98a4), STC(0x658c9a2d), STC(0x603c496c), STC(0x7f92661d), STC(0x7e4a5426), STC(0x7c29fbee), STC(0x793501a9), STC(0x757075ac), STC(0x70e2cbc6), STC(0x6b93d02e), STC(0x658c9a2d), STC(0x5ed77c8a), STC(0x577ff3da), STC(0x4f9292dc), STC(0x7f62368f), STC(0x7d8a5f40), STC(0x7a7d055b), STC(0x7641af3d), STC(0x70e2cbc6), STC(0x6a6d98a4), STC(0x62f201ac), STC(0x5a82799a), STC(0x5133cc94), STC(0x471cece7), STC(0x3c56ba70), STC(0x7f294bfd), STC(0x7ca80038), STC(0x78848414), STC(0x72ccb9db), STC(0x6b93d02e), STC(0x62f201ac), STC(0x590443a7), STC(0x4debe4fe), STC(0x41ce1e65), STC(0x34d3957e), STC(0x2727d486), STC(0x7ee7aa4c), STC(0x7ba3751d), STC(0x7641af3d), STC(0x6ed9eba1), STC(0x658c9a2d), STC(0x5a82799a), STC(0x4debe4fe), STC(0x40000000), STC(0x30fbc54d), STC(0x2120fb83), STC(0x10b5150f), STC(0x7e9d55fc), STC(0x7a7d055b), STC(0x73b5ebd1), STC(0x6a6d98a4), STC(0x5ed77c8a), STC(0x5133cc94), STC(0x41ce1e65), STC(0x30fbc54d), STC(0x1f19f97b), STC(0x0c8bd35e), STC(0xf9b82684), STC(0x7e4a5426), STC(0x793501a9), STC(0x70e2cbc6), STC(0x658c9a2d), STC(0x577ff3da), STC(0x471cece7), STC(0x34d3957e), STC(0x2120fb83), STC(0x0c8bd35e), STC(0xf7a0dec9), STC(0xe2ef2a3e), STC(0x7deeaa7a), STC(0x77cbc3f2), STC(0x6dca0d14), STC(0x603c496c), STC(0x4f9292dc), STC(0x3c56ba70), STC(0x2727d486), STC(0x10b5150f), STC(0xf9b82684), STC(0xe2ef2a3e), STC(0xcd1693f7), STC(0x7d8a5f40), STC(0x7641af3d), STC(0x6a6d98a4), STC(0x5a82799a), STC(0x471cece7), STC(0x30fbc54d), STC(0x18f8b83c), STC(0x00000000), STC(0xe70747c4), STC(0xcf043ab3), STC(0xb8e31319), STC(0x7d1d7958), STC(0x74972f92), STC(0x66cf8120), STC(0x54657194), STC(0x3e2d7eb1), STC(0x25280c5e), STC(0x0a75d60e), STC(0xef4aeaf1), STC(0xd4e0cb15), STC(0xbc6845ce), STC(0xa6fbbc59), STC(0x7ca80038), STC(0x72ccb9db), STC(0x62f201ac), STC(0x4debe4fe), STC(0x34d3957e), STC(0x18f8b83c), STC(0xfbcfdc71), STC(0xdedf047d), STC(0xc3a94590), STC(0xab9a8e6c), STC(0x97f4a3cd), STC(0x7c29fbee), STC(0x70e2cbc6), STC(0x5ed77c8a), STC(0x471cece7), STC(0x2b1f34eb), STC(0x0c8bd35e), STC(0xed37ef91), STC(0xcf043ab3), STC(0xb3c0200c), STC(0x9d0dfe54), STC(0x8c4a142f), STC(0x7ba3751d), STC(0x6ed9eba1), STC(0x5a82799a), STC(0x40000000), STC(0x2120fb83), STC(0x00000000), STC(0xdedf047d), STC(0xc0000000), STC(0xa57d8666), STC(0x9126145f), STC(0x845c8ae3), STC(0x7b1474fd), STC(0x6cb2a837), STC(0x55f5a4d2), STC(0x389cea72), STC(0x16ea0646), STC(0xf3742ca2), STC(0xd0f53ce0), STC(0xb2141b02), STC(0x99307ee0), STC(0x88343c0e), STC(0x806d99e3), STC(0x7a7d055b), STC(0x6a6d98a4), STC(0x5133cc94), STC(0x30fbc54d), STC(0x0c8bd35e), STC(0xe70747c4), STC(0xc3a94590), STC(0xa57d8666), STC(0x8f1d343a), STC(0x8275a0c0), STC(0x809dc971), STC(0x79dd3098), STC(0x680b5c33), STC(0x4c3fdff4), STC(0x2924edac), STC(0x02182427), STC(0xdad7f3a2), STC(0xb727b9f7), STC(0x9a7365d3), STC(0x877b7bec), STC(0x80118b5e), STC(0x84eb8b03), STC(0x793501a9), STC(0x658c9a2d), STC(0x471cece7), STC(0x2120fb83), STC(0xf7a0dec9), STC(0xcf043ab3), STC(0xab9a8e6c), STC(0x9126145f), STC(0x8275a0c0), STC(0x811855b4), STC(0x8d334625), STC(0x78848414), STC(0x62f201ac), STC(0x41ce1e65), STC(0x18f8b83c), STC(0xed37ef91), STC(0xc3a94590), STC(0xa1288376), STC(0x89be50c3), STC(0x80277872), STC(0x8582faa5), STC(0x99307ee0), STC(0x77cbc3f2), STC(0x603c496c), STC(0x3c56ba70), STC(0x10b5150f), STC(0xe2ef2a3e), STC(0xb8e31319), STC(0x97f4a3cd), STC(0x845c8ae3), STC(0x809dc971), STC(0x8d334625), STC(0xa8800c26), STC(0x770acdec), STC(0x5d6c2f99), STC(0x36ba2014), STC(0x085f2137), STC(0xd8d82b7a), STC(0xaecc336c), STC(0x901dcec4), STC(0x811855b4), STC(0x83d60412), STC(0x97f4a3cd), STC(0xbaa34bf4), STC(0x7641af3d), STC(0x5a82799a), STC(0x30fbc54d), STC(0x00000000), STC(0xcf043ab3), STC(0xa57d8666), STC(0x89be50c3), STC(0x80000000), STC(0x89be50c3), STC(0xa57d8666), STC(0xcf043ab3), STC(0x757075ac), STC(0x577ff3da), STC(0x2b1f34eb), STC(0xf7a0dec9), STC(0xc5842c7e), STC(0x9d0dfe54), STC(0x84eb8b03), STC(0x811855b4), STC(0x9235f2ec), STC(0xb5715eef), STC(0xe4fa4bf1), STC(0x74972f92), STC(0x54657194), STC(0x25280c5e), STC(0xef4aeaf1), STC(0xbc6845ce), STC(0x9592675c), STC(0x81b5abda), STC(0x845c8ae3), STC(0x9d0dfe54), STC(0xc763158e), STC(0xfbcfdc71), STC(0x73b5ebd1), STC(0x5133cc94), STC(0x1f19f97b), STC(0xe70747c4), STC(0xb3c0200c), STC(0x8f1d343a), STC(0x80277872), STC(0x89be50c3), STC(0xaa0a5b2e), STC(0xdad7f3a2), STC(0x12c8106f), STC(0x72ccb9db), STC(0x4debe4fe), STC(0x18f8b83c), STC(0xdedf047d), STC(0xab9a8e6c), STC(0x89be50c3), STC(0x804628a7), STC(0x9126145f), STC(0xb8e31319), STC(0xef4aeaf1), STC(0x2924edac), STC(0x71dba9ab), STC(0x4a8ea111), STC(0x12c8106f), STC(0xd6db1254), STC(0xa405847e), STC(0x8582faa5), STC(0x82115586), STC(0x9a7365d3), STC(0xc945dfec), STC(0x0430238f), STC(0x3e2d7eb1), STC(0x70e2cbc6), STC(0x471cece7), STC(0x0c8bd35e), STC(0xcf043ab3), STC(0x9d0dfe54), STC(0x8275a0c0), STC(0x8582faa5), STC(0xa57d8666), STC(0xdad7f3a2), STC(0x18f8b83c), STC(0x5133cc94), STC(0x6fe2313c), STC(0x4397ba32), STC(0x0647d97c), STC(0xc763158e), STC(0x96bfea3d), STC(0x809dc971), STC(0x8a8f8a54), STC(0xb2141b02), STC(0xed37ef91), STC(0x2d168792), STC(0x619a7dce), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_STB RotVectorImag384[] = { STC(0x02182427), STC(0x0430238f), STC(0x0647d97c), STC(0x085f2137), STC(0x0a75d60e), STC(0x0c8bd35e), STC(0x0ea0f48c), STC(0x10b5150f), STC(0x12c8106f), STC(0x14d9c245), STC(0x16ea0646), STC(0x0430238f), STC(0x085f2137), STC(0x0c8bd35e), STC(0x10b5150f), STC(0x14d9c245), STC(0x18f8b83c), STC(0x1d10d5c2), STC(0x2120fb83), STC(0x25280c5e), STC(0x2924edac), STC(0x2d168792), STC(0x0647d97c), STC(0x0c8bd35e), STC(0x12c8106f), STC(0x18f8b83c), STC(0x1f19f97b), STC(0x25280c5e), STC(0x2b1f34eb), STC(0x30fbc54d), STC(0x36ba2014), STC(0x3c56ba70), STC(0x41ce1e65), STC(0x085f2137), STC(0x10b5150f), STC(0x18f8b83c), STC(0x2120fb83), STC(0x2924edac), STC(0x30fbc54d), STC(0x389cea72), STC(0x40000000), STC(0x471cece7), STC(0x4debe4fe), STC(0x54657194), STC(0x0a75d60e), STC(0x14d9c245), STC(0x1f19f97b), STC(0x2924edac), STC(0x32e96c09), STC(0x3c56ba70), STC(0x455cb40c), STC(0x4debe4fe), STC(0x55f5a4d2), STC(0x5d6c2f99), STC(0x6442bd7e), STC(0x0c8bd35e), STC(0x18f8b83c), STC(0x25280c5e), STC(0x30fbc54d), STC(0x3c56ba70), STC(0x471cece7), STC(0x5133cc94), STC(0x5a82799a), STC(0x62f201ac), STC(0x6a6d98a4), STC(0x70e2cbc6), STC(0x0ea0f48c), STC(0x1d10d5c2), STC(0x2b1f34eb), STC(0x389cea72), STC(0x455cb40c), STC(0x5133cc94), STC(0x5bfa7b82), STC(0x658c9a2d), STC(0x6dca0d14), STC(0x74972f92), STC(0x79dd3098), STC(0x10b5150f), STC(0x2120fb83), STC(0x30fbc54d), STC(0x40000000), STC(0x4debe4fe), STC(0x5a82799a), STC(0x658c9a2d), STC(0x6ed9eba1), STC(0x7641af3d), STC(0x7ba3751d), STC(0x7ee7aa4c), STC(0x12c8106f), STC(0x25280c5e), STC(0x36ba2014), STC(0x471cece7), STC(0x55f5a4d2), STC(0x62f201ac), STC(0x6dca0d14), STC(0x7641af3d), STC(0x7c29fbee), STC(0x7f62368f), STC(0x7fd8878e), STC(0x14d9c245), STC(0x2924edac), STC(0x3c56ba70), STC(0x4debe4fe), STC(0x5d6c2f99), STC(0x6a6d98a4), STC(0x74972f92), STC(0x7ba3751d), STC(0x7f62368f), STC(0x7fb9d759), STC(0x7ca80038), STC(0x16ea0646), STC(0x2d168792), STC(0x41ce1e65), STC(0x54657194), STC(0x6442bd7e), STC(0x70e2cbc6), STC(0x79dd3098), STC(0x7ee7aa4c), STC(0x7fd8878e), STC(0x7ca80038), STC(0x757075ac), STC(0x18f8b83c), STC(0x30fbc54d), STC(0x471cece7), STC(0x5a82799a), STC(0x6a6d98a4), STC(0x7641af3d), STC(0x7d8a5f40), STC(0x7fffffff), STC(0x7d8a5f40), STC(0x7641af3d), STC(0x6a6d98a4), STC(0x1b05b40f), STC(0x34d3957e), STC(0x4c3fdff4), STC(0x603c496c), STC(0x6fe2313c), STC(0x7a7d055b), STC(0x7f92661d), STC(0x7ee7aa4c), STC(0x78848414), STC(0x6cb2a837), STC(0x5bfa7b82), STC(0x1d10d5c2), STC(0x389cea72), STC(0x5133cc94), STC(0x658c9a2d), STC(0x74972f92), STC(0x7d8a5f40), STC(0x7fee74a2), STC(0x7ba3751d), STC(0x70e2cbc6), STC(0x603c496c), STC(0x4a8ea111), STC(0x1f19f97b), STC(0x3c56ba70), STC(0x55f5a4d2), STC(0x6a6d98a4), STC(0x78848414), STC(0x7f62368f), STC(0x7e9d55fc), STC(0x7641af3d), STC(0x66cf8120), STC(0x5133cc94), STC(0x36ba2014), STC(0x2120fb83), STC(0x40000000), STC(0x5a82799a), STC(0x6ed9eba1), STC(0x7ba3751d), STC(0x7fffffff), STC(0x7ba3751d), STC(0x6ed9eba1), STC(0x5a82799a), STC(0x40000000), STC(0x2120fb83), STC(0x2325b847), STC(0x4397ba32), STC(0x5ed77c8a), STC(0x72ccb9db), STC(0x7deeaa7a), STC(0x7f62368f), STC(0x770acdec), STC(0x658c9a2d), STC(0x4c3fdff4), STC(0x2d168792), STC(0x0a75d60e), STC(0x25280c5e), STC(0x471cece7), STC(0x62f201ac), STC(0x7641af3d), STC(0x7f62368f), STC(0x7d8a5f40), STC(0x70e2cbc6), STC(0x5a82799a), STC(0x3c56ba70), STC(0x18f8b83c), STC(0xf3742ca2), STC(0x2727d486), STC(0x4a8ea111), STC(0x66cf8120), STC(0x793501a9), STC(0x7ffb9d15), STC(0x7a7d055b), STC(0x694015c3), STC(0x4debe4fe), STC(0x2b1f34eb), STC(0x0430238f), STC(0xdcda47b9), STC(0x2924edac), STC(0x4debe4fe), STC(0x6a6d98a4), STC(0x7ba3751d), STC(0x7fb9d759), STC(0x7641af3d), STC(0x603c496c), STC(0x40000000), STC(0x18f8b83c), STC(0xef4aeaf1), STC(0xc763158e), STC(0x2b1f34eb), STC(0x5133cc94), STC(0x6dca0d14), STC(0x7d8a5f40), STC(0x7e9d55fc), STC(0x70e2cbc6), STC(0x55f5a4d2), STC(0x30fbc54d), STC(0x0647d97c), STC(0xdad7f3a2), STC(0xb3c0200c), STC(0x2d168792), STC(0x54657194), STC(0x70e2cbc6), STC(0x7ee7aa4c), STC(0x7ca80038), STC(0x6a6d98a4), STC(0x4a8ea111), STC(0x2120fb83), STC(0xf3742ca2), STC(0xc763158e), STC(0xa293d067), STC(0x2f0ac320), STC(0x577ff3da), STC(0x73b5ebd1), STC(0x7fb9d759), STC(0x79dd3098), STC(0x62f201ac), STC(0x3e2d7eb1), STC(0x10b5150f), STC(0xe0e60685), STC(0xb5715eef), STC(0x946c2fd2), STC(0x30fbc54d), STC(0x5a82799a), STC(0x7641af3d), STC(0x7fffffff), STC(0x7641af3d), STC(0x5a82799a), STC(0x30fbc54d), STC(0x00000000), STC(0xcf043ab3), STC(0xa57d8666), STC(0x89be50c3), STC(0x32e96c09), STC(0x5d6c2f99), STC(0x78848414), STC(0x7fb9d759), STC(0x71dba9ab), STC(0x5133cc94), STC(0x2325b847), STC(0xef4aeaf1), STC(0xbe31e19b), STC(0x97f4a3cd), STC(0x82e286a8), STC(0x34d3957e), STC(0x603c496c), STC(0x7a7d055b), STC(0x7ee7aa4c), STC(0x6cb2a837), STC(0x471cece7), STC(0x14d9c245), STC(0xdedf047d), STC(0xaecc336c), STC(0x8d334625), STC(0x80118b5e), STC(0x36ba2014), STC(0x62f201ac), STC(0x7c29fbee), STC(0x7d8a5f40), STC(0x66cf8120), STC(0x3c56ba70), STC(0x0647d97c), STC(0xcf043ab3), STC(0xa1288376), STC(0x8582faa5), STC(0x8162aa04), STC(0x389cea72), STC(0x658c9a2d), STC(0x7d8a5f40), STC(0x7ba3751d), STC(0x603c496c), STC(0x30fbc54d), STC(0xf7a0dec9), STC(0xc0000000), STC(0x9592675c), STC(0x811855b4), STC(0x86cafe57), STC(0x3a7bd382), STC(0x680b5c33), STC(0x7e9d55fc), STC(0x793501a9), STC(0x590443a7), STC(0x25280c5e), STC(0xe915f9ba), STC(0xb2141b02), STC(0x8c4a142f), STC(0x80118b5e), STC(0x901dcec4), STC(0x3c56ba70), STC(0x6a6d98a4), STC(0x7f62368f), STC(0x7641af3d), STC(0x5133cc94), STC(0x18f8b83c), STC(0xdad7f3a2), STC(0xa57d8666), STC(0x8582faa5), STC(0x8275a0c0), STC(0x9d0dfe54), STC(0x3e2d7eb1), STC(0x6cb2a837), STC(0x7fd8878e), STC(0x72ccb9db), STC(0x48d84609), STC(0x0c8bd35e), STC(0xcd1693f7), STC(0x9a7365d3), STC(0x8162aa04), STC(0x88343c0e), STC(0xad308a71), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_STB RotVectorReal60[] = { STC(0x7f4c7e54), STC(0x7d33f0ca), STC(0x79bc384d), STC(0x7d33f0ca), STC(0x74ef0ebc), STC(0x678dde6e), STC(0x79bc384d), STC(0x678dde6e), STC(0x4b3c8c12), STC(0x74ef0ebc), STC(0x55a6125c), STC(0x278dde6e), STC(0x6ed9eba1), STC(0x40000000), STC(0x00000000), STC(0x678dde6e), STC(0x278dde6e), STC(0xd8722192), STC(0x5f1f5ea1), STC(0x0d61304e), STC(0xb4c373ee), STC(0x55a6125c), STC(0xf29ecfb2), STC(0x98722192), STC(0x4b3c8c12), STC(0xd8722192), STC(0x8643c7b3), STC(0x40000000), STC(0xc0000000), STC(0x80000000), STC(0x340ff242), STC(0xaa59eda4), STC(0x8643c7b3), STC(0x278dde6e), STC(0x98722192), STC(0x98722192), STC(0x1a9cd9ac), STC(0x8b10f144), STC(0xb4c373ee), STC(0x0d61304e), STC(0x82cc0f36), STC(0xd8722192), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_STB RotVectorImag60[] = { STC(0x0d61304e), STC(0x1a9cd9ac), STC(0x278dde6e), STC(0x1a9cd9ac), STC(0x340ff242), STC(0x4b3c8c12), STC(0x278dde6e), STC(0x4b3c8c12), STC(0x678dde6e), STC(0x340ff242), STC(0x5f1f5ea1), STC(0x79bc384d), STC(0x40000000), STC(0x6ed9eba1), STC(0x7fffffff), STC(0x4b3c8c12), STC(0x79bc384d), STC(0x79bc384d), STC(0x55a6125c), STC(0x7f4c7e54), STC(0x678dde6e), STC(0x5f1f5ea1), STC(0x7f4c7e54), STC(0x4b3c8c12), STC(0x678dde6e), STC(0x79bc384d), STC(0x278dde6e), STC(0x6ed9eba1), STC(0x6ed9eba1), STC(0x00000000), STC(0x74ef0ebc), STC(0x5f1f5ea1), STC(0xd8722192), STC(0x79bc384d), STC(0x4b3c8c12), STC(0xb4c373ee), STC(0x7d33f0ca), STC(0x340ff242), STC(0x98722192), STC(0x7f4c7e54), STC(0x1a9cd9ac), STC(0x8643c7b3), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_STB RotVectorReal120[] = { STC(0x7fd317b4), STC(0x7f4c7e54), STC(0x7e6c9251), STC(0x7d33f0ca), STC(0x7ba3751d), STC(0x79bc384d), STC(0x777f903c), STC(0x7f4c7e54), STC(0x7d33f0ca), STC(0x79bc384d), STC(0x74ef0ebc), STC(0x6ed9eba1), STC(0x678dde6e), STC(0x5f1f5ea1), STC(0x7e6c9251), STC(0x79bc384d), STC(0x720c8075), STC(0x678dde6e), STC(0x5a82799a), STC(0x4b3c8c12), STC(0x3a1c5c57), STC(0x7d33f0ca), STC(0x74ef0ebc), STC(0x678dde6e), STC(0x55a6125c), STC(0x40000000), STC(0x278dde6e), STC(0x0d61304e), STC(0x7ba3751d), STC(0x6ed9eba1), STC(0x5a82799a), STC(0x40000000), STC(0x2120fb83), STC(0x00000000), STC(0xdedf047d), STC(0x79bc384d), STC(0x678dde6e), STC(0x4b3c8c12), STC(0x278dde6e), STC(0x00000000), STC(0xd8722192), STC(0xb4c373ee), STC(0x777f903c), STC(0x5f1f5ea1), STC(0x3a1c5c57), STC(0x0d61304e), STC(0xdedf047d), STC(0xb4c373ee), STC(0x94a6715d), STC(0x74ef0ebc), STC(0x55a6125c), STC(0x278dde6e), STC(0xf29ecfb2), STC(0xc0000000), STC(0x98722192), STC(0x82cc0f36), STC(0x720c8075), STC(0x4b3c8c12), STC(0x14060b68), STC(0xd8722192), STC(0xa57d8666), STC(0x8643c7b3), STC(0x81936daf), STC(0x6ed9eba1), STC(0x40000000), STC(0x00000000), STC(0xc0000000), STC(0x9126145f), STC(0x80000000), STC(0x9126145f), STC(0x6b598ea3), STC(0x340ff242), STC(0xebf9f498), STC(0xaa59eda4), STC(0x845c8ae3), STC(0x8643c7b3), STC(0xaf726def), STC(0x678dde6e), STC(0x278dde6e), STC(0xd8722192), STC(0x98722192), STC(0x80000000), STC(0x98722192), STC(0xd8722192), STC(0x637984d4), STC(0x1a9cd9ac), STC(0xc5e3a3a9), STC(0x8b10f144), STC(0x845c8ae3), STC(0xb4c373ee), STC(0x06b2f1d2), STC(0x5f1f5ea1), STC(0x0d61304e), STC(0xb4c373ee), STC(0x82cc0f36), STC(0x9126145f), STC(0xd8722192), STC(0x340ff242), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_STB RotVectorImag120[] = { STC(0x06b2f1d2), STC(0x0d61304e), STC(0x14060b68), STC(0x1a9cd9ac), STC(0x2120fb83), STC(0x278dde6e), STC(0x2ddf0040), STC(0x0d61304e), STC(0x1a9cd9ac), STC(0x278dde6e), STC(0x340ff242), STC(0x40000000), STC(0x4b3c8c12), STC(0x55a6125c), STC(0x14060b68), STC(0x278dde6e), STC(0x3a1c5c57), STC(0x4b3c8c12), STC(0x5a82799a), STC(0x678dde6e), STC(0x720c8075), STC(0x1a9cd9ac), STC(0x340ff242), STC(0x4b3c8c12), STC(0x5f1f5ea1), STC(0x6ed9eba1), STC(0x79bc384d), STC(0x7f4c7e54), STC(0x2120fb83), STC(0x40000000), STC(0x5a82799a), STC(0x6ed9eba1), STC(0x7ba3751d), STC(0x7fffffff), STC(0x7ba3751d), STC(0x278dde6e), STC(0x4b3c8c12), STC(0x678dde6e), STC(0x79bc384d), STC(0x7fffffff), STC(0x79bc384d), STC(0x678dde6e), STC(0x2ddf0040), STC(0x55a6125c), STC(0x720c8075), STC(0x7f4c7e54), STC(0x7ba3751d), STC(0x678dde6e), STC(0x45b6bb5e), STC(0x340ff242), STC(0x5f1f5ea1), STC(0x79bc384d), STC(0x7f4c7e54), STC(0x6ed9eba1), STC(0x4b3c8c12), STC(0x1a9cd9ac), STC(0x3a1c5c57), STC(0x678dde6e), STC(0x7e6c9251), STC(0x79bc384d), STC(0x5a82799a), STC(0x278dde6e), STC(0xebf9f498), STC(0x40000000), STC(0x6ed9eba1), STC(0x7fffffff), STC(0x6ed9eba1), STC(0x40000000), STC(0x00000000), STC(0xc0000000), STC(0x45b6bb5e), STC(0x74ef0ebc), STC(0x7e6c9251), STC(0x5f1f5ea1), STC(0x2120fb83), STC(0xd8722192), STC(0x9c867b2c), STC(0x4b3c8c12), STC(0x79bc384d), STC(0x79bc384d), STC(0x4b3c8c12), STC(0x00000000), STC(0xb4c373ee), STC(0x8643c7b3), STC(0x508d9211), STC(0x7d33f0ca), STC(0x720c8075), STC(0x340ff242), STC(0xdedf047d), STC(0x98722192), STC(0x802ce84c), STC(0x55a6125c), STC(0x7f4c7e54), STC(0x678dde6e), STC(0x1a9cd9ac), STC(0xc0000000), STC(0x8643c7b3), STC(0x8b10f144), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_STB RotVectorReal192[] = { STC(0x7fee74a2), STC(0x7fb9d759), STC(0x7f62368f), STC(0x7ee7aa4c), STC(0x7e4a5426), STC(0x7d8a5f40), STC(0x7ca80038), STC(0x7ba3751d), STC(0x7a7d055b), STC(0x793501a9), STC(0x77cbc3f2), STC(0x7641af3d), STC(0x74972f92), STC(0x72ccb9db), STC(0x70e2cbc6), STC(0x7fb9d759), STC(0x7ee7aa4c), STC(0x7d8a5f40), STC(0x7ba3751d), STC(0x793501a9), STC(0x7641af3d), STC(0x72ccb9db), STC(0x6ed9eba1), STC(0x6a6d98a4), STC(0x658c9a2d), STC(0x603c496c), STC(0x5a82799a), STC(0x54657194), STC(0x4debe4fe), STC(0x471cece7), STC(0x7f62368f), STC(0x7d8a5f40), STC(0x7a7d055b), STC(0x7641af3d), STC(0x70e2cbc6), STC(0x6a6d98a4), STC(0x62f201ac), STC(0x5a82799a), STC(0x5133cc94), STC(0x471cece7), STC(0x3c56ba70), STC(0x30fbc54d), STC(0x25280c5e), STC(0x18f8b83c), STC(0x0c8bd35e), STC(0x7ee7aa4c), STC(0x7ba3751d), STC(0x7641af3d), STC(0x6ed9eba1), STC(0x658c9a2d), STC(0x5a82799a), STC(0x4debe4fe), STC(0x40000000), STC(0x30fbc54d), STC(0x2120fb83), STC(0x10b5150f), STC(0x00000000), STC(0xef4aeaf1), STC(0xdedf047d), STC(0xcf043ab3), STC(0x7e4a5426), STC(0x793501a9), STC(0x70e2cbc6), STC(0x658c9a2d), STC(0x577ff3da), STC(0x471cece7), STC(0x34d3957e), STC(0x2120fb83), STC(0x0c8bd35e), STC(0xf7a0dec9), STC(0xe2ef2a3e), STC(0xcf043ab3), STC(0xbc6845ce), STC(0xab9a8e6c), STC(0x9d0dfe54), STC(0x7d8a5f40), STC(0x7641af3d), STC(0x6a6d98a4), STC(0x5a82799a), STC(0x471cece7), STC(0x30fbc54d), STC(0x18f8b83c), STC(0x00000000), STC(0xe70747c4), STC(0xcf043ab3), STC(0xb8e31319), STC(0xa57d8666), STC(0x9592675c), STC(0x89be50c3), STC(0x8275a0c0), STC(0x7ca80038), STC(0x72ccb9db), STC(0x62f201ac), STC(0x4debe4fe), STC(0x34d3957e), STC(0x18f8b83c), STC(0xfbcfdc71), STC(0xdedf047d), STC(0xc3a94590), STC(0xab9a8e6c), STC(0x97f4a3cd), STC(0x89be50c3), STC(0x81b5abda), STC(0x804628a7), STC(0x8582faa5), STC(0x7ba3751d), STC(0x6ed9eba1), STC(0x5a82799a), STC(0x40000000), STC(0x2120fb83), STC(0x00000000), STC(0xdedf047d), STC(0xc0000000), STC(0xa57d8666), STC(0x9126145f), STC(0x845c8ae3), STC(0x80000000), STC(0x845c8ae3), STC(0x9126145f), STC(0xa57d8666), STC(0x7a7d055b), STC(0x6a6d98a4), STC(0x5133cc94), STC(0x30fbc54d), STC(0x0c8bd35e), STC(0xe70747c4), STC(0xc3a94590), STC(0xa57d8666), STC(0x8f1d343a), STC(0x8275a0c0), STC(0x809dc971), STC(0x89be50c3), STC(0x9d0dfe54), STC(0xb8e31319), STC(0xdad7f3a2), STC(0x793501a9), STC(0x658c9a2d), STC(0x471cece7), STC(0x2120fb83), STC(0xf7a0dec9), STC(0xcf043ab3), STC(0xab9a8e6c), STC(0x9126145f), STC(0x8275a0c0), STC(0x811855b4), STC(0x8d334625), STC(0xa57d8666), STC(0xc763158e), STC(0xef4aeaf1), STC(0x18f8b83c), STC(0x77cbc3f2), STC(0x603c496c), STC(0x3c56ba70), STC(0x10b5150f), STC(0xe2ef2a3e), STC(0xb8e31319), STC(0x97f4a3cd), STC(0x845c8ae3), STC(0x809dc971), STC(0x8d334625), STC(0xa8800c26), STC(0xcf043ab3), STC(0xfbcfdc71), STC(0x2924edac), STC(0x5133cc94), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_STB RotVectorImag192[] = { STC(0x0430238f), STC(0x085f2137), STC(0x0c8bd35e), STC(0x10b5150f), STC(0x14d9c245), STC(0x18f8b83c), STC(0x1d10d5c2), STC(0x2120fb83), STC(0x25280c5e), STC(0x2924edac), STC(0x2d168792), STC(0x30fbc54d), STC(0x34d3957e), STC(0x389cea72), STC(0x3c56ba70), STC(0x085f2137), STC(0x10b5150f), STC(0x18f8b83c), STC(0x2120fb83), STC(0x2924edac), STC(0x30fbc54d), STC(0x389cea72), STC(0x40000000), STC(0x471cece7), STC(0x4debe4fe), STC(0x54657194), STC(0x5a82799a), STC(0x603c496c), STC(0x658c9a2d), STC(0x6a6d98a4), STC(0x0c8bd35e), STC(0x18f8b83c), STC(0x25280c5e), STC(0x30fbc54d), STC(0x3c56ba70), STC(0x471cece7), STC(0x5133cc94), STC(0x5a82799a), STC(0x62f201ac), STC(0x6a6d98a4), STC(0x70e2cbc6), STC(0x7641af3d), STC(0x7a7d055b), STC(0x7d8a5f40), STC(0x7f62368f), STC(0x10b5150f), STC(0x2120fb83), STC(0x30fbc54d), STC(0x40000000), STC(0x4debe4fe), STC(0x5a82799a), STC(0x658c9a2d), STC(0x6ed9eba1), STC(0x7641af3d), STC(0x7ba3751d), STC(0x7ee7aa4c), STC(0x7fffffff), STC(0x7ee7aa4c), STC(0x7ba3751d), STC(0x7641af3d), STC(0x14d9c245), STC(0x2924edac), STC(0x3c56ba70), STC(0x4debe4fe), STC(0x5d6c2f99), STC(0x6a6d98a4), STC(0x74972f92), STC(0x7ba3751d), STC(0x7f62368f), STC(0x7fb9d759), STC(0x7ca80038), STC(0x7641af3d), STC(0x6cb2a837), STC(0x603c496c), STC(0x5133cc94), STC(0x18f8b83c), STC(0x30fbc54d), STC(0x471cece7), STC(0x5a82799a), STC(0x6a6d98a4), STC(0x7641af3d), STC(0x7d8a5f40), STC(0x7fffffff), STC(0x7d8a5f40), STC(0x7641af3d), STC(0x6a6d98a4), STC(0x5a82799a), STC(0x471cece7), STC(0x30fbc54d), STC(0x18f8b83c), STC(0x1d10d5c2), STC(0x389cea72), STC(0x5133cc94), STC(0x658c9a2d), STC(0x74972f92), STC(0x7d8a5f40), STC(0x7fee74a2), STC(0x7ba3751d), STC(0x70e2cbc6), STC(0x603c496c), STC(0x4a8ea111), STC(0x30fbc54d), STC(0x14d9c245), STC(0xf7a0dec9), STC(0xdad7f3a2), STC(0x2120fb83), STC(0x40000000), STC(0x5a82799a), STC(0x6ed9eba1), STC(0x7ba3751d), STC(0x7fffffff), STC(0x7ba3751d), STC(0x6ed9eba1), STC(0x5a82799a), STC(0x40000000), STC(0x2120fb83), STC(0x00000000), STC(0xdedf047d), STC(0xc0000000), STC(0xa57d8666), STC(0x25280c5e), STC(0x471cece7), STC(0x62f201ac), STC(0x7641af3d), STC(0x7f62368f), STC(0x7d8a5f40), STC(0x70e2cbc6), STC(0x5a82799a), STC(0x3c56ba70), STC(0x18f8b83c), STC(0xf3742ca2), STC(0xcf043ab3), STC(0xaecc336c), STC(0x9592675c), STC(0x8582faa5), STC(0x2924edac), STC(0x4debe4fe), STC(0x6a6d98a4), STC(0x7ba3751d), STC(0x7fb9d759), STC(0x7641af3d), STC(0x603c496c), STC(0x40000000), STC(0x18f8b83c), STC(0xef4aeaf1), STC(0xc763158e), STC(0xa57d8666), STC(0x8d334625), STC(0x811855b4), STC(0x8275a0c0), STC(0x2d168792), STC(0x54657194), STC(0x70e2cbc6), STC(0x7ee7aa4c), STC(0x7ca80038), STC(0x6a6d98a4), STC(0x4a8ea111), STC(0x2120fb83), STC(0xf3742ca2), STC(0xc763158e), STC(0xa293d067), STC(0x89be50c3), STC(0x80118b5e), STC(0x86cafe57), STC(0x9d0dfe54), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_STB RotVectorReal240[] = { STC(0x7ff4c56f), STC(0x7fd317b4), STC(0x7f9afcb9), STC(0x7f4c7e54), STC(0x7ee7aa4c), STC(0x7e6c9251), STC(0x7ddb4bfc), STC(0x7d33f0ca), STC(0x7c769e18), STC(0x7ba3751d), STC(0x7aba9ae6), STC(0x79bc384d), STC(0x78a879f4), STC(0x777f903c), STC(0x7641af3d), STC(0x7fd317b4), STC(0x7f4c7e54), STC(0x7e6c9251), STC(0x7d33f0ca), STC(0x7ba3751d), STC(0x79bc384d), STC(0x777f903c), STC(0x74ef0ebc), STC(0x720c8075), STC(0x6ed9eba1), STC(0x6b598ea3), STC(0x678dde6e), STC(0x637984d4), STC(0x5f1f5ea1), STC(0x5a82799a), STC(0x7f9afcb9), STC(0x7e6c9251), STC(0x7c769e18), STC(0x79bc384d), STC(0x7641af3d), STC(0x720c8075), STC(0x6d23501b), STC(0x678dde6e), STC(0x6154fb91), STC(0x5a82799a), STC(0x53211d18), STC(0x4b3c8c12), STC(0x42e13ba4), STC(0x3a1c5c57), STC(0x30fbc54d), STC(0x7f4c7e54), STC(0x7d33f0ca), STC(0x79bc384d), STC(0x74ef0ebc), STC(0x6ed9eba1), STC(0x678dde6e), STC(0x5f1f5ea1), STC(0x55a6125c), STC(0x4b3c8c12), STC(0x40000000), STC(0x340ff242), STC(0x278dde6e), STC(0x1a9cd9ac), STC(0x0d61304e), STC(0x00000000), STC(0x7ee7aa4c), STC(0x7ba3751d), STC(0x7641af3d), STC(0x6ed9eba1), STC(0x658c9a2d), STC(0x5a82799a), STC(0x4debe4fe), STC(0x40000000), STC(0x30fbc54d), STC(0x2120fb83), STC(0x10b5150f), STC(0x00000000), STC(0xef4aeaf1), STC(0xdedf047d), STC(0xcf043ab3), STC(0x7e6c9251), STC(0x79bc384d), STC(0x720c8075), STC(0x678dde6e), STC(0x5a82799a), STC(0x4b3c8c12), STC(0x3a1c5c57), STC(0x278dde6e), STC(0x14060b68), STC(0x00000000), STC(0xebf9f498), STC(0xd8722192), STC(0xc5e3a3a9), STC(0xb4c373ee), STC(0xa57d8666), STC(0x7ddb4bfc), STC(0x777f903c), STC(0x6d23501b), STC(0x5f1f5ea1), STC(0x4debe4fe), STC(0x3a1c5c57), STC(0x245a9d65), STC(0x0d61304e), STC(0xf5f50d67), STC(0xdedf047d), STC(0xc8e5032b), STC(0xb4c373ee), STC(0xa326eec0), STC(0x94a6715d), STC(0x89be50c3), STC(0x7d33f0ca), STC(0x74ef0ebc), STC(0x678dde6e), STC(0x55a6125c), STC(0x40000000), STC(0x278dde6e), STC(0x0d61304e), STC(0xf29ecfb2), STC(0xd8722192), STC(0xc0000000), STC(0xaa59eda4), STC(0x98722192), STC(0x8b10f144), STC(0x82cc0f36), STC(0x80000000), STC(0x7c769e18), STC(0x720c8075), STC(0x6154fb91), STC(0x4b3c8c12), STC(0x30fbc54d), STC(0x14060b68), STC(0xf5f50d67), STC(0xd8722192), STC(0xbd1ec45c), STC(0xa57d8666), STC(0x92dcafe5), STC(0x8643c7b3), STC(0x80650347), STC(0x81936daf), STC(0x89be50c3), STC(0x7ba3751d), STC(0x6ed9eba1), STC(0x5a82799a), STC(0x40000000), STC(0x2120fb83), STC(0x00000000), STC(0xdedf047d), STC(0xc0000000), STC(0xa57d8666), STC(0x9126145f), STC(0x845c8ae3), STC(0x80000000), STC(0x845c8ae3), STC(0x9126145f), STC(0xa57d8666), STC(0x7aba9ae6), STC(0x6b598ea3), STC(0x53211d18), STC(0x340ff242), STC(0x10b5150f), STC(0xebf9f498), STC(0xc8e5032b), STC(0xaa59eda4), STC(0x92dcafe5), STC(0x845c8ae3), STC(0x800b3a91), STC(0x8643c7b3), STC(0x96830876), STC(0xaf726def), STC(0xcf043ab3), STC(0x79bc384d), STC(0x678dde6e), STC(0x4b3c8c12), STC(0x278dde6e), STC(0x00000000), STC(0xd8722192), STC(0xb4c373ee), STC(0x98722192), STC(0x8643c7b3), STC(0x80000000), STC(0x8643c7b3), STC(0x98722192), STC(0xb4c373ee), STC(0xd8722192), STC(0x00000000), STC(0x78a879f4), STC(0x637984d4), STC(0x42e13ba4), STC(0x1a9cd9ac), STC(0xef4aeaf1), STC(0xc5e3a3a9), STC(0xa326eec0), STC(0x8b10f144), STC(0x80650347), STC(0x845c8ae3), STC(0x96830876), STC(0xb4c373ee), STC(0xdba5629b), STC(0x06b2f1d2), STC(0x30fbc54d), STC(0x777f903c), STC(0x5f1f5ea1), STC(0x3a1c5c57), STC(0x0d61304e), STC(0xdedf047d), STC(0xb4c373ee), STC(0x94a6715d), STC(0x82cc0f36), STC(0x81936daf), STC(0x9126145f), STC(0xaf726def), STC(0xd8722192), STC(0x06b2f1d2), STC(0x340ff242), STC(0x5a82799a), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_STB RotVectorImag240[] = { STC(0x0359c428), STC(0x06b2f1d2), STC(0x0a0af299), STC(0x0d61304e), STC(0x10b5150f), STC(0x14060b68), STC(0x17537e63), STC(0x1a9cd9ac), STC(0x1de189a6), STC(0x2120fb83), STC(0x245a9d65), STC(0x278dde6e), STC(0x2aba2ee4), STC(0x2ddf0040), STC(0x30fbc54d), STC(0x06b2f1d2), STC(0x0d61304e), STC(0x14060b68), STC(0x1a9cd9ac), STC(0x2120fb83), STC(0x278dde6e), STC(0x2ddf0040), STC(0x340ff242), STC(0x3a1c5c57), STC(0x40000000), STC(0x45b6bb5e), STC(0x4b3c8c12), STC(0x508d9211), STC(0x55a6125c), STC(0x5a82799a), STC(0x0a0af299), STC(0x14060b68), STC(0x1de189a6), STC(0x278dde6e), STC(0x30fbc54d), STC(0x3a1c5c57), STC(0x42e13ba4), STC(0x4b3c8c12), STC(0x53211d18), STC(0x5a82799a), STC(0x6154fb91), STC(0x678dde6e), STC(0x6d23501b), STC(0x720c8075), STC(0x7641af3d), STC(0x0d61304e), STC(0x1a9cd9ac), STC(0x278dde6e), STC(0x340ff242), STC(0x40000000), STC(0x4b3c8c12), STC(0x55a6125c), STC(0x5f1f5ea1), STC(0x678dde6e), STC(0x6ed9eba1), STC(0x74ef0ebc), STC(0x79bc384d), STC(0x7d33f0ca), STC(0x7f4c7e54), STC(0x7fffffff), STC(0x10b5150f), STC(0x2120fb83), STC(0x30fbc54d), STC(0x40000000), STC(0x4debe4fe), STC(0x5a82799a), STC(0x658c9a2d), STC(0x6ed9eba1), STC(0x7641af3d), STC(0x7ba3751d), STC(0x7ee7aa4c), STC(0x7fffffff), STC(0x7ee7aa4c), STC(0x7ba3751d), STC(0x7641af3d), STC(0x14060b68), STC(0x278dde6e), STC(0x3a1c5c57), STC(0x4b3c8c12), STC(0x5a82799a), STC(0x678dde6e), STC(0x720c8075), STC(0x79bc384d), STC(0x7e6c9251), STC(0x7fffffff), STC(0x7e6c9251), STC(0x79bc384d), STC(0x720c8075), STC(0x678dde6e), STC(0x5a82799a), STC(0x17537e63), STC(0x2ddf0040), STC(0x42e13ba4), STC(0x55a6125c), STC(0x658c9a2d), STC(0x720c8075), STC(0x7aba9ae6), STC(0x7f4c7e54), STC(0x7f9afcb9), STC(0x7ba3751d), STC(0x7387ea23), STC(0x678dde6e), STC(0x581c00b3), STC(0x45b6bb5e), STC(0x30fbc54d), STC(0x1a9cd9ac), STC(0x340ff242), STC(0x4b3c8c12), STC(0x5f1f5ea1), STC(0x6ed9eba1), STC(0x79bc384d), STC(0x7f4c7e54), STC(0x7f4c7e54), STC(0x79bc384d), STC(0x6ed9eba1), STC(0x5f1f5ea1), STC(0x4b3c8c12), STC(0x340ff242), STC(0x1a9cd9ac), STC(0x00000000), STC(0x1de189a6), STC(0x3a1c5c57), STC(0x53211d18), STC(0x678dde6e), STC(0x7641af3d), STC(0x7e6c9251), STC(0x7f9afcb9), STC(0x79bc384d), STC(0x6d23501b), STC(0x5a82799a), STC(0x42e13ba4), STC(0x278dde6e), STC(0x0a0af299), STC(0xebf9f498), STC(0xcf043ab3), STC(0x2120fb83), STC(0x40000000), STC(0x5a82799a), STC(0x6ed9eba1), STC(0x7ba3751d), STC(0x7fffffff), STC(0x7ba3751d), STC(0x6ed9eba1), STC(0x5a82799a), STC(0x40000000), STC(0x2120fb83), STC(0x00000000), STC(0xdedf047d), STC(0xc0000000), STC(0xa57d8666), STC(0x245a9d65), STC(0x45b6bb5e), STC(0x6154fb91), STC(0x74ef0ebc), STC(0x7ee7aa4c), STC(0x7e6c9251), STC(0x7387ea23), STC(0x5f1f5ea1), STC(0x42e13ba4), STC(0x2120fb83), STC(0xfca63bd8), STC(0xd8722192), STC(0xb780001c), STC(0x9c867b2c), STC(0x89be50c3), STC(0x278dde6e), STC(0x4b3c8c12), STC(0x678dde6e), STC(0x79bc384d), STC(0x7fffffff), STC(0x79bc384d), STC(0x678dde6e), STC(0x4b3c8c12), STC(0x278dde6e), STC(0x00000000), STC(0xd8722192), STC(0xb4c373ee), STC(0x98722192), STC(0x8643c7b3), STC(0x80000000), STC(0x2aba2ee4), STC(0x508d9211), STC(0x6d23501b), STC(0x7d33f0ca), STC(0x7ee7aa4c), STC(0x720c8075), STC(0x581c00b3), STC(0x340ff242), STC(0x0a0af299), STC(0xdedf047d), STC(0xb780001c), STC(0x98722192), STC(0x8545651a), STC(0x802ce84c), STC(0x89be50c3), STC(0x2ddf0040), STC(0x55a6125c), STC(0x720c8075), STC(0x7f4c7e54), STC(0x7ba3751d), STC(0x678dde6e), STC(0x45b6bb5e), STC(0x1a9cd9ac), STC(0xebf9f498), STC(0xc0000000), STC(0x9c867b2c), STC(0x8643c7b3), STC(0x802ce84c), STC(0x8b10f144), STC(0xa57d8666), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_STB RotVectorReal480[] = { STC(0x7ffd3154), STC(0x7ff4c56f), STC(0x7fe6bcb0), STC(0x7fd317b4), STC(0x7fb9d759), STC(0x7f9afcb9), STC(0x7f76892f), STC(0x7f4c7e54), STC(0x7f1cde01), STC(0x7ee7aa4c), STC(0x7eace58a), STC(0x7e6c9251), STC(0x7e26b371), STC(0x7ddb4bfc), STC(0x7d8a5f40), STC(0x7d33f0ca), STC(0x7cd80464), STC(0x7c769e18), STC(0x7c0fc22a), STC(0x7ba3751d), STC(0x7b31bbb2), STC(0x7aba9ae6), STC(0x7a3e17f2), STC(0x79bc384d), STC(0x793501a9), STC(0x78a879f4), STC(0x7816a759), STC(0x777f903c), STC(0x76e33b3f), STC(0x7641af3d), STC(0x759af34c), STC(0x7ff4c56f), STC(0x7fd317b4), STC(0x7f9afcb9), STC(0x7f4c7e54), STC(0x7ee7aa4c), STC(0x7e6c9251), STC(0x7ddb4bfc), STC(0x7d33f0ca), STC(0x7c769e18), STC(0x7ba3751d), STC(0x7aba9ae6), STC(0x79bc384d), STC(0x78a879f4), STC(0x777f903c), STC(0x7641af3d), STC(0x74ef0ebc), STC(0x7387ea23), STC(0x720c8075), STC(0x707d1443), STC(0x6ed9eba1), STC(0x6d23501b), STC(0x6b598ea3), STC(0x697cf78a), STC(0x678dde6e), STC(0x658c9a2d), STC(0x637984d4), STC(0x6154fb91), STC(0x5f1f5ea1), STC(0x5cd91140), STC(0x5a82799a), STC(0x581c00b3), STC(0x7fe6bcb0), STC(0x7f9afcb9), STC(0x7f1cde01), STC(0x7e6c9251), STC(0x7d8a5f40), STC(0x7c769e18), STC(0x7b31bbb2), STC(0x79bc384d), STC(0x7816a759), STC(0x7641af3d), STC(0x743e0918), STC(0x720c8075), STC(0x6fadf2fc), STC(0x6d23501b), STC(0x6a6d98a4), STC(0x678dde6e), STC(0x648543e4), STC(0x6154fb91), STC(0x5dfe47ad), STC(0x5a82799a), STC(0x56e2f15d), STC(0x53211d18), STC(0x4f3e7875), STC(0x4b3c8c12), STC(0x471cece7), STC(0x42e13ba4), STC(0x3e8b240e), STC(0x3a1c5c57), STC(0x3596a46c), STC(0x30fbc54d), STC(0x2c4d9050), STC(0x7fd317b4), STC(0x7f4c7e54), STC(0x7e6c9251), STC(0x7d33f0ca), STC(0x7ba3751d), STC(0x79bc384d), STC(0x777f903c), STC(0x74ef0ebc), STC(0x720c8075), STC(0x6ed9eba1), STC(0x6b598ea3), STC(0x678dde6e), STC(0x637984d4), STC(0x5f1f5ea1), STC(0x5a82799a), STC(0x55a6125c), STC(0x508d9211), STC(0x4b3c8c12), STC(0x45b6bb5e), STC(0x40000000), STC(0x3a1c5c57), STC(0x340ff242), STC(0x2ddf0040), STC(0x278dde6e), STC(0x2120fb83), STC(0x1a9cd9ac), STC(0x14060b68), STC(0x0d61304e), STC(0x06b2f1d2), STC(0x00000000), STC(0xf94d0e2e), STC(0x7fb9d759), STC(0x7ee7aa4c), STC(0x7d8a5f40), STC(0x7ba3751d), STC(0x793501a9), STC(0x7641af3d), STC(0x72ccb9db), STC(0x6ed9eba1), STC(0x6a6d98a4), STC(0x658c9a2d), STC(0x603c496c), STC(0x5a82799a), STC(0x54657194), STC(0x4debe4fe), STC(0x471cece7), STC(0x40000000), STC(0x389cea72), STC(0x30fbc54d), STC(0x2924edac), STC(0x2120fb83), STC(0x18f8b83c), STC(0x10b5150f), STC(0x085f2137), STC(0x00000000), STC(0xf7a0dec9), STC(0xef4aeaf1), STC(0xe70747c4), STC(0xdedf047d), STC(0xd6db1254), STC(0xcf043ab3), STC(0xc763158e), STC(0x7f9afcb9), STC(0x7e6c9251), STC(0x7c769e18), STC(0x79bc384d), STC(0x7641af3d), STC(0x720c8075), STC(0x6d23501b), STC(0x678dde6e), STC(0x6154fb91), STC(0x5a82799a), STC(0x53211d18), STC(0x4b3c8c12), STC(0x42e13ba4), STC(0x3a1c5c57), STC(0x30fbc54d), STC(0x278dde6e), STC(0x1de189a6), STC(0x14060b68), STC(0x0a0af299), STC(0x00000000), STC(0xf5f50d67), STC(0xebf9f498), STC(0xe21e765a), STC(0xd8722192), STC(0xcf043ab3), STC(0xc5e3a3a9), STC(0xbd1ec45c), STC(0xb4c373ee), STC(0xacdee2e8), STC(0xa57d8666), STC(0x9eab046f), STC(0x7f76892f), STC(0x7ddb4bfc), STC(0x7b31bbb2), STC(0x777f903c), STC(0x72ccb9db), STC(0x6d23501b), STC(0x668f7c25), STC(0x5f1f5ea1), STC(0x56e2f15d), STC(0x4debe4fe), STC(0x444d7aff), STC(0x3a1c5c57), STC(0x2f6e6d16), STC(0x245a9d65), STC(0x18f8b83c), STC(0x0d61304e), STC(0x01aceb7c), STC(0xf5f50d67), STC(0xea52c166), STC(0xdedf047d), STC(0xd3b26fb0), STC(0xc8e5032b), STC(0xbe8df2ba), STC(0xb4c373ee), STC(0xab9a8e6c), STC(0xa326eec0), STC(0x9b7abc1c), STC(0x94a6715d), STC(0x8eb8b9a0), STC(0x89be50c3), STC(0x85c1e80e), STC(0x7f4c7e54), STC(0x7d33f0ca), STC(0x79bc384d), STC(0x74ef0ebc), STC(0x6ed9eba1), STC(0x678dde6e), STC(0x5f1f5ea1), STC(0x55a6125c), STC(0x4b3c8c12), STC(0x40000000), STC(0x340ff242), STC(0x278dde6e), STC(0x1a9cd9ac), STC(0x0d61304e), STC(0x00000000), STC(0xf29ecfb2), STC(0xe5632654), STC(0xd8722192), STC(0xcbf00dbe), STC(0xc0000000), STC(0xb4c373ee), STC(0xaa59eda4), STC(0xa0e0a15f), STC(0x98722192), STC(0x9126145f), STC(0x8b10f144), STC(0x8643c7b3), STC(0x82cc0f36), STC(0x80b381ac), STC(0x80000000), STC(0x80b381ac), STC(0x7f1cde01), STC(0x7c769e18), STC(0x7816a759), STC(0x720c8075), STC(0x6a6d98a4), STC(0x6154fb91), STC(0x56e2f15d), STC(0x4b3c8c12), STC(0x3e8b240e), STC(0x30fbc54d), STC(0x22be8f87), STC(0x14060b68), STC(0x05067734), STC(0xf5f50d67), STC(0xe70747c4), STC(0xd8722192), STC(0xca695b94), STC(0xbd1ec45c), STC(0xb0c1878b), STC(0xa57d8666), STC(0x9b7abc1c), STC(0x92dcafe5), STC(0x8bc1f6e8), STC(0x8643c7b3), STC(0x8275a0c0), STC(0x80650347), STC(0x80194350), STC(0x81936daf), STC(0x84ce444e), STC(0x89be50c3), STC(0x90520d04), STC(0x7ee7aa4c), STC(0x7ba3751d), STC(0x7641af3d), STC(0x6ed9eba1), STC(0x658c9a2d), STC(0x5a82799a), STC(0x4debe4fe), STC(0x40000000), STC(0x30fbc54d), STC(0x2120fb83), STC(0x10b5150f), STC(0x00000000), STC(0xef4aeaf1), STC(0xdedf047d), STC(0xcf043ab3), STC(0xc0000000), STC(0xb2141b02), STC(0xa57d8666), STC(0x9a7365d3), STC(0x9126145f), STC(0x89be50c3), STC(0x845c8ae3), STC(0x811855b4), STC(0x80000000), STC(0x811855b4), STC(0x845c8ae3), STC(0x89be50c3), STC(0x9126145f), STC(0x9a7365d3), STC(0xa57d8666), STC(0xb2141b02), STC(0x7eace58a), STC(0x7aba9ae6), STC(0x743e0918), STC(0x6b598ea3), STC(0x603c496c), STC(0x53211d18), STC(0x444d7aff), STC(0x340ff242), STC(0x22be8f87), STC(0x10b5150f), STC(0xfe531484), STC(0xebf9f498), STC(0xda0aecf9), STC(0xc8e5032b), STC(0xb8e31319), STC(0xaa59eda4), STC(0x9d969742), STC(0x92dcafe5), STC(0x8a650cb4), STC(0x845c8ae3), STC(0x80e321ff), STC(0x800b3a91), STC(0x81d94c8f), STC(0x8643c7b3), STC(0x8d334625), STC(0x96830876), STC(0xa201b853), STC(0xaf726def), STC(0xbe8df2ba), STC(0xcf043ab3), STC(0xe07e0c84), STC(0x7e6c9251), STC(0x79bc384d), STC(0x720c8075), STC(0x678dde6e), STC(0x5a82799a), STC(0x4b3c8c12), STC(0x3a1c5c57), STC(0x278dde6e), STC(0x14060b68), STC(0x00000000), STC(0xebf9f498), STC(0xd8722192), STC(0xc5e3a3a9), STC(0xb4c373ee), STC(0xa57d8666), STC(0x98722192), STC(0x8df37f8b), STC(0x8643c7b3), STC(0x81936daf), STC(0x80000000), STC(0x81936daf), STC(0x8643c7b3), STC(0x8df37f8b), STC(0x98722192), STC(0xa57d8666), STC(0xb4c373ee), STC(0xc5e3a3a9), STC(0xd8722192), STC(0xebf9f498), STC(0x00000000), STC(0x14060b68), STC(0x7e26b371), STC(0x78a879f4), STC(0x6fadf2fc), STC(0x637984d4), STC(0x54657194), STC(0x42e13ba4), STC(0x2f6e6d16), STC(0x1a9cd9ac), STC(0x05067734), STC(0xef4aeaf1), STC(0xda0aecf9), STC(0xc5e3a3a9), STC(0xb36a1978), STC(0xa326eec0), STC(0x9592675c), STC(0x8b10f144), STC(0x83f03dd6), STC(0x80650347), STC(0x808976d1), STC(0x845c8ae3), STC(0x8bc1f6e8), STC(0x96830876), STC(0xa45037c9), STC(0xb4c373ee), STC(0xc763158e), STC(0xdba5629b), STC(0xf0f488d9), STC(0x06b2f1d2), STC(0x1c3fd045), STC(0x30fbc54d), STC(0x444d7aff), STC(0x7ddb4bfc), STC(0x777f903c), STC(0x6d23501b), STC(0x5f1f5ea1), STC(0x4debe4fe), STC(0x3a1c5c57), STC(0x245a9d65), STC(0x0d61304e), STC(0xf5f50d67), STC(0xdedf047d), STC(0xc8e5032b), STC(0xb4c373ee), STC(0xa326eec0), STC(0x94a6715d), STC(0x89be50c3), STC(0x82cc0f36), STC(0x800b3a91), STC(0x81936daf), STC(0x8757860c), STC(0x9126145f), STC(0x9eab046f), STC(0xaf726def), STC(0xc2ec7635), STC(0xd8722192), STC(0xef4aeaf1), STC(0x06b2f1d2), STC(0x1de189a6), STC(0x340ff242), STC(0x487fffe4), STC(0x5a82799a), STC(0x697cf78a), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_STB RotVectorImag480[] = { STC(0x01aceb7c), STC(0x0359c428), STC(0x05067734), STC(0x06b2f1d2), STC(0x085f2137), STC(0x0a0af299), STC(0x0bb65336), STC(0x0d61304e), STC(0x0f0b7727), STC(0x10b5150f), STC(0x125df75b), STC(0x14060b68), STC(0x15ad3e9a), STC(0x17537e63), STC(0x18f8b83c), STC(0x1a9cd9ac), STC(0x1c3fd045), STC(0x1de189a6), STC(0x1f81f37c), STC(0x2120fb83), STC(0x22be8f87), STC(0x245a9d65), STC(0x25f51307), STC(0x278dde6e), STC(0x2924edac), STC(0x2aba2ee4), STC(0x2c4d9050), STC(0x2ddf0040), STC(0x2f6e6d16), STC(0x30fbc54d), STC(0x3286f779), STC(0x0359c428), STC(0x06b2f1d2), STC(0x0a0af299), STC(0x0d61304e), STC(0x10b5150f), STC(0x14060b68), STC(0x17537e63), STC(0x1a9cd9ac), STC(0x1de189a6), STC(0x2120fb83), STC(0x245a9d65), STC(0x278dde6e), STC(0x2aba2ee4), STC(0x2ddf0040), STC(0x30fbc54d), STC(0x340ff242), STC(0x371afcd5), STC(0x3a1c5c57), STC(0x3d1389cb), STC(0x40000000), STC(0x42e13ba4), STC(0x45b6bb5e), STC(0x487fffe4), STC(0x4b3c8c12), STC(0x4debe4fe), STC(0x508d9211), STC(0x53211d18), STC(0x55a6125c), STC(0x581c00b3), STC(0x5a82799a), STC(0x5cd91140), STC(0x05067734), STC(0x0a0af299), STC(0x0f0b7727), STC(0x14060b68), STC(0x18f8b83c), STC(0x1de189a6), STC(0x22be8f87), STC(0x278dde6e), STC(0x2c4d9050), STC(0x30fbc54d), STC(0x3596a46c), STC(0x3a1c5c57), STC(0x3e8b240e), STC(0x42e13ba4), STC(0x471cece7), STC(0x4b3c8c12), STC(0x4f3e7875), STC(0x53211d18), STC(0x56e2f15d), STC(0x5a82799a), STC(0x5dfe47ad), STC(0x6154fb91), STC(0x648543e4), STC(0x678dde6e), STC(0x6a6d98a4), STC(0x6d23501b), STC(0x6fadf2fc), STC(0x720c8075), STC(0x743e0918), STC(0x7641af3d), STC(0x7816a759), STC(0x06b2f1d2), STC(0x0d61304e), STC(0x14060b68), STC(0x1a9cd9ac), STC(0x2120fb83), STC(0x278dde6e), STC(0x2ddf0040), STC(0x340ff242), STC(0x3a1c5c57), STC(0x40000000), STC(0x45b6bb5e), STC(0x4b3c8c12), STC(0x508d9211), STC(0x55a6125c), STC(0x5a82799a), STC(0x5f1f5ea1), STC(0x637984d4), STC(0x678dde6e), STC(0x6b598ea3), STC(0x6ed9eba1), STC(0x720c8075), STC(0x74ef0ebc), STC(0x777f903c), STC(0x79bc384d), STC(0x7ba3751d), STC(0x7d33f0ca), STC(0x7e6c9251), STC(0x7f4c7e54), STC(0x7fd317b4), STC(0x7fffffff), STC(0x7fd317b4), STC(0x085f2137), STC(0x10b5150f), STC(0x18f8b83c), STC(0x2120fb83), STC(0x2924edac), STC(0x30fbc54d), STC(0x389cea72), STC(0x40000000), STC(0x471cece7), STC(0x4debe4fe), STC(0x54657194), STC(0x5a82799a), STC(0x603c496c), STC(0x658c9a2d), STC(0x6a6d98a4), STC(0x6ed9eba1), STC(0x72ccb9db), STC(0x7641af3d), STC(0x793501a9), STC(0x7ba3751d), STC(0x7d8a5f40), STC(0x7ee7aa4c), STC(0x7fb9d759), STC(0x7fffffff), STC(0x7fb9d759), STC(0x7ee7aa4c), STC(0x7d8a5f40), STC(0x7ba3751d), STC(0x793501a9), STC(0x7641af3d), STC(0x72ccb9db), STC(0x0a0af299), STC(0x14060b68), STC(0x1de189a6), STC(0x278dde6e), STC(0x30fbc54d), STC(0x3a1c5c57), STC(0x42e13ba4), STC(0x4b3c8c12), STC(0x53211d18), STC(0x5a82799a), STC(0x6154fb91), STC(0x678dde6e), STC(0x6d23501b), STC(0x720c8075), STC(0x7641af3d), STC(0x79bc384d), STC(0x7c769e18), STC(0x7e6c9251), STC(0x7f9afcb9), STC(0x7fffffff), STC(0x7f9afcb9), STC(0x7e6c9251), STC(0x7c769e18), STC(0x79bc384d), STC(0x7641af3d), STC(0x720c8075), STC(0x6d23501b), STC(0x678dde6e), STC(0x6154fb91), STC(0x5a82799a), STC(0x53211d18), STC(0x0bb65336), STC(0x17537e63), STC(0x22be8f87), STC(0x2ddf0040), STC(0x389cea72), STC(0x42e13ba4), STC(0x4c95e688), STC(0x55a6125c), STC(0x5dfe47ad), STC(0x658c9a2d), STC(0x6c40cf2c), STC(0x720c8075), STC(0x76e33b3f), STC(0x7aba9ae6), STC(0x7d8a5f40), STC(0x7f4c7e54), STC(0x7ffd3154), STC(0x7f9afcb9), STC(0x7e26b371), STC(0x7ba3751d), STC(0x7816a759), STC(0x7387ea23), STC(0x6e010780), STC(0x678dde6e), STC(0x603c496c), STC(0x581c00b3), STC(0x4f3e7875), STC(0x45b6bb5e), STC(0x3b9941b1), STC(0x30fbc54d), STC(0x25f51307), STC(0x0d61304e), STC(0x1a9cd9ac), STC(0x278dde6e), STC(0x340ff242), STC(0x40000000), STC(0x4b3c8c12), STC(0x55a6125c), STC(0x5f1f5ea1), STC(0x678dde6e), STC(0x6ed9eba1), STC(0x74ef0ebc), STC(0x79bc384d), STC(0x7d33f0ca), STC(0x7f4c7e54), STC(0x7fffffff), STC(0x7f4c7e54), STC(0x7d33f0ca), STC(0x79bc384d), STC(0x74ef0ebc), STC(0x6ed9eba1), STC(0x678dde6e), STC(0x5f1f5ea1), STC(0x55a6125c), STC(0x4b3c8c12), STC(0x40000000), STC(0x340ff242), STC(0x278dde6e), STC(0x1a9cd9ac), STC(0x0d61304e), STC(0x00000000), STC(0xf29ecfb2), STC(0x0f0b7727), STC(0x1de189a6), STC(0x2c4d9050), STC(0x3a1c5c57), STC(0x471cece7), STC(0x53211d18), STC(0x5dfe47ad), STC(0x678dde6e), STC(0x6fadf2fc), STC(0x7641af3d), STC(0x7b31bbb2), STC(0x7e6c9251), STC(0x7fe6bcb0), STC(0x7f9afcb9), STC(0x7d8a5f40), STC(0x79bc384d), STC(0x743e0918), STC(0x6d23501b), STC(0x648543e4), STC(0x5a82799a), STC(0x4f3e7875), STC(0x42e13ba4), STC(0x3596a46c), STC(0x278dde6e), STC(0x18f8b83c), STC(0x0a0af299), STC(0xfaf988cc), STC(0xebf9f498), STC(0xdd417079), STC(0xcf043ab3), STC(0xc174dbf2), STC(0x10b5150f), STC(0x2120fb83), STC(0x30fbc54d), STC(0x40000000), STC(0x4debe4fe), STC(0x5a82799a), STC(0x658c9a2d), STC(0x6ed9eba1), STC(0x7641af3d), STC(0x7ba3751d), STC(0x7ee7aa4c), STC(0x7fffffff), STC(0x7ee7aa4c), STC(0x7ba3751d), STC(0x7641af3d), STC(0x6ed9eba1), STC(0x658c9a2d), STC(0x5a82799a), STC(0x4debe4fe), STC(0x40000000), STC(0x30fbc54d), STC(0x2120fb83), STC(0x10b5150f), STC(0x00000000), STC(0xef4aeaf1), STC(0xdedf047d), STC(0xcf043ab3), STC(0xc0000000), STC(0xb2141b02), STC(0xa57d8666), STC(0x9a7365d3), STC(0x125df75b), STC(0x245a9d65), STC(0x3596a46c), STC(0x45b6bb5e), STC(0x54657194), STC(0x6154fb91), STC(0x6c40cf2c), STC(0x74ef0ebc), STC(0x7b31bbb2), STC(0x7ee7aa4c), STC(0x7ffd3154), STC(0x7e6c9251), STC(0x7a3e17f2), STC(0x7387ea23), STC(0x6a6d98a4), STC(0x5f1f5ea1), STC(0x51d92321), STC(0x42e13ba4), STC(0x3286f779), STC(0x2120fb83), STC(0x0f0b7727), STC(0xfca63bd8), STC(0xea52c166), STC(0xd8722192), STC(0xc763158e), STC(0xb780001c), STC(0xa91d0ea3), STC(0x9c867b2c), STC(0x91fef880), STC(0x89be50c3), STC(0x83f03dd6), STC(0x14060b68), STC(0x278dde6e), STC(0x3a1c5c57), STC(0x4b3c8c12), STC(0x5a82799a), STC(0x678dde6e), STC(0x720c8075), STC(0x79bc384d), STC(0x7e6c9251), STC(0x7fffffff), STC(0x7e6c9251), STC(0x79bc384d), STC(0x720c8075), STC(0x678dde6e), STC(0x5a82799a), STC(0x4b3c8c12), STC(0x3a1c5c57), STC(0x278dde6e), STC(0x14060b68), STC(0x00000000), STC(0xebf9f498), STC(0xd8722192), STC(0xc5e3a3a9), STC(0xb4c373ee), STC(0xa57d8666), STC(0x98722192), STC(0x8df37f8b), STC(0x8643c7b3), STC(0x81936daf), STC(0x80000000), STC(0x81936daf), STC(0x15ad3e9a), STC(0x2aba2ee4), STC(0x3e8b240e), STC(0x508d9211), STC(0x603c496c), STC(0x6d23501b), STC(0x76e33b3f), STC(0x7d33f0ca), STC(0x7fe6bcb0), STC(0x7ee7aa4c), STC(0x7a3e17f2), STC(0x720c8075), STC(0x668f7c25), STC(0x581c00b3), STC(0x471cece7), STC(0x340ff242), STC(0x1f81f37c), STC(0x0a0af299), STC(0xf449acca), STC(0xdedf047d), STC(0xca695b94), STC(0xb780001c), STC(0xa6aecd5e), STC(0x98722192), STC(0x8d334625), STC(0x8545651a), STC(0x80e321ff), STC(0x802ce84c), STC(0x8327fb9c), STC(0x89be50c3), STC(0x93bf30d4), STC(0x17537e63), STC(0x2ddf0040), STC(0x42e13ba4), STC(0x55a6125c), STC(0x658c9a2d), STC(0x720c8075), STC(0x7aba9ae6), STC(0x7f4c7e54), STC(0x7f9afcb9), STC(0x7ba3751d), STC(0x7387ea23), STC(0x678dde6e), STC(0x581c00b3), STC(0x45b6bb5e), STC(0x30fbc54d), STC(0x1a9cd9ac), STC(0x0359c428), STC(0xebf9f498), STC(0xd545d11c), STC(0xc0000000), STC(0xacdee2e8), STC(0x9c867b2c), STC(0x8f82ebbd), STC(0x8643c7b3), STC(0x811855b4), STC(0x802ce84c), STC(0x838961e8), STC(0x8b10f144), STC(0x96830876), STC(0xa57d8666), STC(0xb780001c), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_STB RotVectorReal20[] = { STC(0x79bc384d), STC(0x678dde6e), STC(0x4b3c8c12), STC(0x678dde6e), STC(0x278dde6e), STC(0xd8722192), STC(0x4b3c8c12), STC(0xd8722192), STC(0x8643c7b3), STC(0x278dde6e), STC(0x98722192), STC(0x98722192), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_STB RotVectorImag20[] = { STC(0x278dde6e), STC(0x4b3c8c12), STC(0x678dde6e), STC(0x4b3c8c12), STC(0x79bc384d), STC(0x79bc384d), STC(0x678dde6e), STC(0x79bc384d), STC(0x278dde6e), STC(0x79bc384d), STC(0x4b3c8c12), STC(0xb4c373ee), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_WTP SineWindow8[] = { WTCP(0x7f62368f, 0x0c8bd35e), WTCP(0x7a7d055b, 0x25280c5e), WTCP(0x70e2cbc6, 0x3c56ba70), WTCP(0x62f201ac, 0x5133cc94), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_WTP SineWindow12[] = { WTCP(0x7fb9d759, 0x085f2137), WTCP(0x7d8a5f40, 0x18f8b83c), WTCP(0x793501a9, 0x2924edac), WTCP(0x72ccb9db, 0x389cea72), WTCP(0x6a6d98a4, 0x471cece7), WTCP(0x603c496c, 0x54657194), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_WTP SineWindow16[] = { WTCP(0x7fd8878e, 0x0647d97c), WTCP(0x7e9d55fc, 0x12c8106f), WTCP(0x7c29fbee, 0x1f19f97b), WTCP(0x78848414, 0x2b1f34eb), WTCP(0x73b5ebd1, 0x36ba2014), WTCP(0x6dca0d14, 0x41ce1e65), WTCP(0x66cf8120, 0x4c3fdff4), WTCP(0x5ed77c8a, 0x55f5a4d2), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_WTP SineWindow20[] = { WTCP(0x7fe6bcb0, 0x05067734), WTCP(0x7f1cde01, 0x0f0b7727), WTCP(0x7d8a5f40, 0x18f8b83c), WTCP(0x7b31bbb2, 0x22be8f87), WTCP(0x7816a759, 0x2c4d9050), WTCP(0x743e0918, 0x3596a46c), WTCP(0x6fadf2fc, 0x3e8b240e), WTCP(0x6a6d98a4, 0x471cece7), WTCP(0x648543e4, 0x4f3e7875), WTCP(0x5dfe47ad, 0x56e2f15d), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_WTP SineWindow24[] = { WTCP(0x7fee74a2, 0x0430238f), WTCP(0x7f62368f, 0x0c8bd35e), WTCP(0x7e4a5426, 0x14d9c245), WTCP(0x7ca80038, 0x1d10d5c2), WTCP(0x7a7d055b, 0x25280c5e), WTCP(0x77cbc3f2, 0x2d168792), WTCP(0x74972f92, 0x34d3957e), WTCP(0x70e2cbc6, 0x3c56ba70), WTCP(0x6cb2a837, 0x4397ba32), WTCP(0x680b5c33, 0x4a8ea111), WTCP(0x62f201ac, 0x5133cc94), WTCP(0x5d6c2f99, 0x577ff3da), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_WTP SineWindow32[] = { WTCP(0x7ff62182, 0x03242abf), WTCP(0x7fa736b4, 0x096a9049), WTCP(0x7f0991c4, 0x0fab272b), WTCP(0x7e1d93ea, 0x15e21445), WTCP(0x7ce3ceb2, 0x1c0b826a), WTCP(0x7b5d039e, 0x2223a4c5), WTCP(0x798a23b1, 0x2826b928), WTCP(0x776c4edb, 0x2e110a62), WTCP(0x7504d345, 0x33def287), WTCP(0x72552c85, 0x398cdd32), WTCP(0x6f5f02b2, 0x3f1749b8), WTCP(0x6c242960, 0x447acd50), WTCP(0x68a69e81, 0x49b41533), WTCP(0x64e88926, 0x4ebfe8a5), WTCP(0x60ec3830, 0x539b2af0), WTCP(0x5cb420e0, 0x5842dd54), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_WTP SineWindow40[] = { WTCP(0x7ff9af04, 0x02835b5a), WTCP(0x7fc72ae2, 0x07891418), WTCP(0x7f62368f, 0x0c8bd35e), WTCP(0x7ecaf9e5, 0x11899ed3), WTCP(0x7e01b096, 0x16807e15), WTCP(0x7d06aa16, 0x1b6e7b7a), WTCP(0x7bda497d, 0x2051a4dd), WTCP(0x7a7d055b, 0x25280c5e), WTCP(0x78ef678f, 0x29efc925), WTCP(0x77320d0d, 0x2ea6f827), WTCP(0x7545a5a0, 0x334bbcde), WTCP(0x732af3a7, 0x37dc420c), WTCP(0x70e2cbc6, 0x3c56ba70), WTCP(0x6e6e1492, 0x40b9617d), WTCP(0x6bcdc639, 0x45027c0c), WTCP(0x6902ea1d, 0x4930590f), WTCP(0x660e9a6a, 0x4d415234), WTCP(0x62f201ac, 0x5133cc94), WTCP(0x5fae5a55, 0x55063951), WTCP(0x5c44ee40, 0x58b71632), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_WTP SineWindow48[] = { WTCP(0x7ffb9d15, 0x02182427), WTCP(0x7fd8878e, 0x0647d97c), WTCP(0x7f92661d, 0x0a75d60e), WTCP(0x7f294bfd, 0x0ea0f48c), WTCP(0x7e9d55fc, 0x12c8106f), WTCP(0x7deeaa7a, 0x16ea0646), WTCP(0x7d1d7958, 0x1b05b40f), WTCP(0x7c29fbee, 0x1f19f97b), WTCP(0x7b1474fd, 0x2325b847), WTCP(0x79dd3098, 0x2727d486), WTCP(0x78848414, 0x2b1f34eb), WTCP(0x770acdec, 0x2f0ac320), WTCP(0x757075ac, 0x32e96c09), WTCP(0x73b5ebd1, 0x36ba2014), WTCP(0x71dba9ab, 0x3a7bd382), WTCP(0x6fe2313c, 0x3e2d7eb1), WTCP(0x6dca0d14, 0x41ce1e65), WTCP(0x6b93d02e, 0x455cb40c), WTCP(0x694015c3, 0x48d84609), WTCP(0x66cf8120, 0x4c3fdff4), WTCP(0x6442bd7e, 0x4f9292dc), WTCP(0x619a7dce, 0x52cf758f), WTCP(0x5ed77c8a, 0x55f5a4d2), WTCP(0x5bfa7b82, 0x590443a7), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_WTP SineWindow64[] = { WTCP(0x7ffd885a, 0x01921d20), WTCP(0x7fe9cbc0, 0x04b6195d), WTCP(0x7fc25596, 0x07d95b9e), WTCP(0x7f872bf3, 0x0afb6805), WTCP(0x7f3857f6, 0x0e1bc2e4), WTCP(0x7ed5e5c6, 0x1139f0cf), WTCP(0x7e5fe493, 0x145576b1), WTCP(0x7dd6668f, 0x176dd9de), WTCP(0x7d3980ec, 0x1a82a026), WTCP(0x7c894bde, 0x1d934fe5), WTCP(0x7bc5e290, 0x209f701c), WTCP(0x7aef6323, 0x23a6887f), WTCP(0x7a05eead, 0x26a82186), WTCP(0x7909a92d, 0x29a3c485), WTCP(0x77fab989, 0x2c98fbba), WTCP(0x76d94989, 0x2f875262), WTCP(0x75a585cf, 0x326e54c7), WTCP(0x745f9dd1, 0x354d9057), WTCP(0x7307c3d0, 0x382493b0), WTCP(0x719e2cd2, 0x3af2eeb7), WTCP(0x7023109a, 0x3db832a6), WTCP(0x6e96a99d, 0x4073f21d), WTCP(0x6cf934fc, 0x4325c135), WTCP(0x6b4af279, 0x45cd358f), WTCP(0x698c246c, 0x4869e665), WTCP(0x67bd0fbd, 0x4afb6c98), WTCP(0x65ddfbd3, 0x4d8162c4), WTCP(0x63ef3290, 0x4ffb654d), WTCP(0x61f1003f, 0x5269126e), WTCP(0x5fe3b38d, 0x54ca0a4b), WTCP(0x5dc79d7c, 0x571deefa), WTCP(0x5b9d1154, 0x59646498), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_WTP SineWindow96[] = { WTCP(0x7ffee744, 0x010c1460), WTCP(0x7ff62182, 0x03242abf), WTCP(0x7fe49698, 0x053c0a01), WTCP(0x7fca47b9, 0x07538d6b), WTCP(0x7fa736b4, 0x096a9049), WTCP(0x7f7b65ef, 0x0b80edf1), WTCP(0x7f46d86c, 0x0d9681c2), WTCP(0x7f0991c4, 0x0fab272b), WTCP(0x7ec3962a, 0x11beb9aa), WTCP(0x7e74ea6a, 0x13d114d0), WTCP(0x7e1d93ea, 0x15e21445), WTCP(0x7dbd98a4, 0x17f193c5), WTCP(0x7d54ff2e, 0x19ff6f2a), WTCP(0x7ce3ceb2, 0x1c0b826a), WTCP(0x7c6a0ef2, 0x1e15a99a), WTCP(0x7be7c847, 0x201dc0ef), WTCP(0x7b5d039e, 0x2223a4c5), WTCP(0x7ac9ca7a, 0x2427319d), WTCP(0x7a2e26f2, 0x26284422), WTCP(0x798a23b1, 0x2826b928), WTCP(0x78ddcbf5, 0x2a226db5), WTCP(0x78292b8d, 0x2c1b3efb), WTCP(0x776c4edb, 0x2e110a62), WTCP(0x76a742d1, 0x3003ad85), WTCP(0x75da14ef, 0x31f30638), WTCP(0x7504d345, 0x33def287), WTCP(0x74278c72, 0x35c750bc), WTCP(0x73424fa0, 0x37abff5d), WTCP(0x72552c85, 0x398cdd32), WTCP(0x71603361, 0x3b69c947), WTCP(0x706374ff, 0x3d42a2ec), WTCP(0x6f5f02b2, 0x3f1749b8), WTCP(0x6e52ee52, 0x40e79d8c), WTCP(0x6d3f4a40, 0x42b37e96), WTCP(0x6c242960, 0x447acd50), WTCP(0x6b019f1a, 0x463d6a87), WTCP(0x69d7bf57, 0x47fb3757), WTCP(0x68a69e81, 0x49b41533), WTCP(0x676e5183, 0x4b67e5e4), WTCP(0x662eedc3, 0x4d168b8b), WTCP(0x64e88926, 0x4ebfe8a5), WTCP(0x639b3a0b, 0x5063e008), WTCP(0x62471749, 0x520254ef), WTCP(0x60ec3830, 0x539b2af0), WTCP(0x5f8ab487, 0x552e4605), WTCP(0x5e22a487, 0x56bb8a90), WTCP(0x5cb420e0, 0x5842dd54), WTCP(0x5b3f42ae, 0x59c42381), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_WTP SineWindow120[] = { WTCP(0x7fff4c54, 0x00d676eb), WTCP(0x7ff9af04, 0x02835b5a), WTCP(0x7fee74a2, 0x0430238f), WTCP(0x7fdd9dad, 0x05dcbcbe), WTCP(0x7fc72ae2, 0x07891418), WTCP(0x7fab1d3d, 0x093516d4), WTCP(0x7f8975f9, 0x0ae0b22c), WTCP(0x7f62368f, 0x0c8bd35e), WTCP(0x7f3560b9, 0x0e3667ad), WTCP(0x7f02f66f, 0x0fe05c64), WTCP(0x7ecaf9e5, 0x11899ed3), WTCP(0x7e8d6d91, 0x13321c53), WTCP(0x7e4a5426, 0x14d9c245), WTCP(0x7e01b096, 0x16807e15), WTCP(0x7db3860f, 0x18263d36), WTCP(0x7d5fd801, 0x19caed29), WTCP(0x7d06aa16, 0x1b6e7b7a), WTCP(0x7ca80038, 0x1d10d5c2), WTCP(0x7c43de8e, 0x1eb1e9a7), WTCP(0x7bda497d, 0x2051a4dd), WTCP(0x7b6b45a5, 0x21eff528), WTCP(0x7af6d7e6, 0x238cc85d), WTCP(0x7a7d055b, 0x25280c5e), WTCP(0x79fdd35c, 0x26c1af22), WTCP(0x7979477d, 0x28599eb0), WTCP(0x78ef678f, 0x29efc925), WTCP(0x7860399e, 0x2b841caf), WTCP(0x77cbc3f2, 0x2d168792), WTCP(0x77320d0d, 0x2ea6f827), WTCP(0x76931bae, 0x30355cdd), WTCP(0x75eef6ce, 0x31c1a43b), WTCP(0x7545a5a0, 0x334bbcde), WTCP(0x74972f92, 0x34d3957e), WTCP(0x73e39c49, 0x36591cea), WTCP(0x732af3a7, 0x37dc420c), WTCP(0x726d3dc6, 0x395cf3e9), WTCP(0x71aa82f7, 0x3adb21a1), WTCP(0x70e2cbc6, 0x3c56ba70), WTCP(0x701620f5, 0x3dcfadb0), WTCP(0x6f448b7e, 0x3f45ead8), WTCP(0x6e6e1492, 0x40b9617d), WTCP(0x6d92c59b, 0x422a0154), WTCP(0x6cb2a837, 0x4397ba32), WTCP(0x6bcdc639, 0x45027c0c), WTCP(0x6ae429ae, 0x466a36f9), WTCP(0x69f5dcd3, 0x47cedb31), WTCP(0x6902ea1d, 0x4930590f), WTCP(0x680b5c33, 0x4a8ea111), WTCP(0x670f3df3, 0x4be9a3db), WTCP(0x660e9a6a, 0x4d415234), WTCP(0x65097cdb, 0x4e959d08), WTCP(0x63fff0ba, 0x4fe6756a), WTCP(0x62f201ac, 0x5133cc94), WTCP(0x61dfbb8a, 0x527d93e6), WTCP(0x60c92a5a, 0x53c3bcea), WTCP(0x5fae5a55, 0x55063951), WTCP(0x5e8f57e2, 0x5644faf4), WTCP(0x5d6c2f99, 0x577ff3da), WTCP(0x5c44ee40, 0x58b71632), WTCP(0x5b19a0c8, 0x59ea5454), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_WTP SineWindow128[] = { WTCP(0x7fff6216, 0x00c90f88), WTCP(0x7ffa72d1, 0x025b26d7), WTCP(0x7ff09478, 0x03ed26e6), WTCP(0x7fe1c76b, 0x057f0035), WTCP(0x7fce0c3e, 0x0710a345), WTCP(0x7fb563b3, 0x08a2009a), WTCP(0x7f97cebd, 0x0a3308bd), WTCP(0x7f754e80, 0x0bc3ac35), WTCP(0x7f4de451, 0x0d53db92), WTCP(0x7f2191b4, 0x0ee38766), WTCP(0x7ef05860, 0x1072a048), WTCP(0x7eba3a39, 0x120116d5), WTCP(0x7e7f3957, 0x138edbb1), WTCP(0x7e3f57ff, 0x151bdf86), WTCP(0x7dfa98a8, 0x16a81305), WTCP(0x7db0fdf8, 0x183366e9), WTCP(0x7d628ac6, 0x19bdcbf3), WTCP(0x7d0f4218, 0x1b4732ef), WTCP(0x7cb72724, 0x1ccf8cb3), WTCP(0x7c5a3d50, 0x1e56ca1e), WTCP(0x7bf88830, 0x1fdcdc1b), WTCP(0x7b920b89, 0x2161b3a0), WTCP(0x7b26cb4f, 0x22e541af), WTCP(0x7ab6cba4, 0x24677758), WTCP(0x7a4210d8, 0x25e845b6), WTCP(0x79c89f6e, 0x27679df4), WTCP(0x794a7c12, 0x28e5714b), WTCP(0x78c7aba2, 0x2a61b101), WTCP(0x78403329, 0x2bdc4e6f), WTCP(0x77b417df, 0x2d553afc), WTCP(0x77235f2d, 0x2ecc681e), WTCP(0x768e0ea6, 0x3041c761), WTCP(0x75f42c0b, 0x31b54a5e), WTCP(0x7555bd4c, 0x3326e2c3), WTCP(0x74b2c884, 0x34968250), WTCP(0x740b53fb, 0x36041ad9), WTCP(0x735f6626, 0x376f9e46), WTCP(0x72af05a7, 0x38d8fe93), WTCP(0x71fa3949, 0x3a402dd2), WTCP(0x71410805, 0x3ba51e29), WTCP(0x708378ff, 0x3d07c1d6), WTCP(0x6fc19385, 0x3e680b2c), WTCP(0x6efb5f12, 0x3fc5ec98), WTCP(0x6e30e34a, 0x4121589b), WTCP(0x6d6227fa, 0x427a41d0), WTCP(0x6c8f351c, 0x43d09aed), WTCP(0x6bb812d1, 0x452456bd), WTCP(0x6adcc964, 0x46756828), WTCP(0x69fd614a, 0x47c3c22f), WTCP(0x6919e320, 0x490f57ee), WTCP(0x683257ab, 0x4a581c9e), WTCP(0x6746c7d8, 0x4b9e0390), WTCP(0x66573cbb, 0x4ce10034), WTCP(0x6563bf92, 0x4e210617), WTCP(0x646c59bf, 0x4f5e08e3), WTCP(0x637114cc, 0x5097fc5e), WTCP(0x6271fa69, 0x51ced46e), WTCP(0x616f146c, 0x53028518), WTCP(0x60686ccf, 0x5433027d), WTCP(0x5f5e0db3, 0x556040e2), WTCP(0x5e50015d, 0x568a34a9), WTCP(0x5d3e5237, 0x57b0d256), WTCP(0x5c290acc, 0x58d40e8c), WTCP(0x5b1035cf, 0x59f3de12), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_WTP SineWindow160[] = { WTCP(0x7fff9aef, 0x00a0d951), WTCP(0x7ffc726f, 0x01e287fc), WTCP(0x7ff62182, 0x03242abf), WTCP(0x7feca851, 0x0465b9aa), WTCP(0x7fe00716, 0x05a72ccf), WTCP(0x7fd03e23, 0x06e87c3f), WTCP(0x7fbd4dda, 0x0829a00c), WTCP(0x7fa736b4, 0x096a9049), WTCP(0x7f8df93c, 0x0aab450d), WTCP(0x7f719611, 0x0bebb66c), WTCP(0x7f520de6, 0x0d2bdc80), WTCP(0x7f2f6183, 0x0e6baf61), WTCP(0x7f0991c4, 0x0fab272b), WTCP(0x7ee09f95, 0x10ea3bfd), WTCP(0x7eb48bfb, 0x1228e5f8), WTCP(0x7e85580c, 0x13671d3d), WTCP(0x7e5304f2, 0x14a4d9f4), WTCP(0x7e1d93ea, 0x15e21445), WTCP(0x7de50646, 0x171ec45c), WTCP(0x7da95d6c, 0x185ae269), WTCP(0x7d6a9ad5, 0x199666a0), WTCP(0x7d28c00c, 0x1ad14938), WTCP(0x7ce3ceb2, 0x1c0b826a), WTCP(0x7c9bc87a, 0x1d450a78), WTCP(0x7c50af2b, 0x1e7dd9a4), WTCP(0x7c02849f, 0x1fb5e836), WTCP(0x7bb14ac5, 0x20ed2e7b), WTCP(0x7b5d039e, 0x2223a4c5), WTCP(0x7b05b13d, 0x2359436c), WTCP(0x7aab55ca, 0x248e02cb), WTCP(0x7a4df380, 0x25c1db44), WTCP(0x79ed8cad, 0x26f4c53e), WTCP(0x798a23b1, 0x2826b928), WTCP(0x7923bb01, 0x2957af74), WTCP(0x78ba5524, 0x2a87a09d), WTCP(0x784df4b3, 0x2bb68522), WTCP(0x77de9c5b, 0x2ce45589), WTCP(0x776c4edb, 0x2e110a62), WTCP(0x76f70f05, 0x2f3c9c40), WTCP(0x767edfbe, 0x306703bf), WTCP(0x7603c3fd, 0x31903982), WTCP(0x7585becb, 0x32b83634), WTCP(0x7504d345, 0x33def287), WTCP(0x74810499, 0x35046736), WTCP(0x73fa5607, 0x36288d03), WTCP(0x7370cae2, 0x374b5cb9), WTCP(0x72e4668f, 0x386ccf2a), WTCP(0x72552c85, 0x398cdd32), WTCP(0x71c3204c, 0x3aab7fb7), WTCP(0x712e457f, 0x3bc8afa5), WTCP(0x70969fca, 0x3ce465f3), WTCP(0x6ffc32eb, 0x3dfe9ba1), WTCP(0x6f5f02b2, 0x3f1749b8), WTCP(0x6ebf12ff, 0x402e694c), WTCP(0x6e1c67c4, 0x4143f379), WTCP(0x6d770506, 0x4257e166), WTCP(0x6cceeed8, 0x436a2c45), WTCP(0x6c242960, 0x447acd50), WTCP(0x6b76b8d6, 0x4589bdcf), WTCP(0x6ac6a180, 0x4696f710), WTCP(0x6a13e7b8, 0x47a27271), WTCP(0x695e8fe5, 0x48ac2957), WTCP(0x68a69e81, 0x49b41533), WTCP(0x67ec1817, 0x4aba2f84), WTCP(0x672f013f, 0x4bbe71d1), WTCP(0x666f5ea6, 0x4cc0d5ae), WTCP(0x65ad3505, 0x4dc154bb), WTCP(0x64e88926, 0x4ebfe8a5), WTCP(0x64215fe5, 0x4fbc8b22), WTCP(0x6357be2a, 0x50b735f8), WTCP(0x628ba8ef, 0x51afe2f6), WTCP(0x61bd253f, 0x52a68bfb), WTCP(0x60ec3830, 0x539b2af0), WTCP(0x6018e6eb, 0x548db9cb), WTCP(0x5f4336a7, 0x557e3292), WTCP(0x5e6b2ca8, 0x566c8f55), WTCP(0x5d90ce45, 0x5758ca31), WTCP(0x5cb420e0, 0x5842dd54), WTCP(0x5bd529eb, 0x592ac2f7), WTCP(0x5af3eee6, 0x5a107561), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_WTP SineWindow192[] = { WTCP(0x7fffb9d1, 0x00860a79), WTCP(0x7ffd885a, 0x01921d20), WTCP(0x7ff92577, 0x029e28e2), WTCP(0x7ff2913a, 0x03aa292a), WTCP(0x7fe9cbc0, 0x04b6195d), WTCP(0x7fded530, 0x05c1f4e7), WTCP(0x7fd1adb9, 0x06cdb72f), WTCP(0x7fc25596, 0x07d95b9e), WTCP(0x7fb0cd0a, 0x08e4dda0), WTCP(0x7f9d1461, 0x09f0389f), WTCP(0x7f872bf3, 0x0afb6805), WTCP(0x7f6f141f, 0x0c066740), WTCP(0x7f54cd4f, 0x0d1131ba), WTCP(0x7f3857f6, 0x0e1bc2e4), WTCP(0x7f19b491, 0x0f26162a), WTCP(0x7ef8e3a6, 0x103026fe), WTCP(0x7ed5e5c6, 0x1139f0cf), WTCP(0x7eb0bb8a, 0x12436f10), WTCP(0x7e896595, 0x134c9d34), WTCP(0x7e5fe493, 0x145576b1), WTCP(0x7e34393b, 0x155df6fc), WTCP(0x7e06644c, 0x1666198d), WTCP(0x7dd6668f, 0x176dd9de), WTCP(0x7da440d6, 0x1875336a), WTCP(0x7d6ff3fe, 0x197c21ad), WTCP(0x7d3980ec, 0x1a82a026), WTCP(0x7d00e88f, 0x1b88aa55), WTCP(0x7cc62bdf, 0x1c8e3bbe), WTCP(0x7c894bde, 0x1d934fe5), WTCP(0x7c4a4996, 0x1e97e251), WTCP(0x7c09261d, 0x1f9bee8a), WTCP(0x7bc5e290, 0x209f701c), WTCP(0x7b808015, 0x21a26295), WTCP(0x7b38ffde, 0x22a4c185), WTCP(0x7aef6323, 0x23a6887f), WTCP(0x7aa3ab29, 0x24a7b317), WTCP(0x7a55d93a, 0x25a83ce6), WTCP(0x7a05eead, 0x26a82186), WTCP(0x79b3ece0, 0x27a75c95), WTCP(0x795fd53a, 0x28a5e9b4), WTCP(0x7909a92d, 0x29a3c485), WTCP(0x78b16a32, 0x2aa0e8b0), WTCP(0x785719cc, 0x2b9d51dd), WTCP(0x77fab989, 0x2c98fbba), WTCP(0x779c4afc, 0x2d93e1f8), WTCP(0x773bcfc4, 0x2e8e0048), WTCP(0x76d94989, 0x2f875262), WTCP(0x7674b9fa, 0x307fd401), WTCP(0x760e22d1, 0x317780e2), WTCP(0x75a585cf, 0x326e54c7), WTCP(0x753ae4c0, 0x33644b76), WTCP(0x74ce4177, 0x345960b7), WTCP(0x745f9dd1, 0x354d9057), WTCP(0x73eefbb3, 0x3640d627), WTCP(0x737c5d0b, 0x37332dfd), WTCP(0x7307c3d0, 0x382493b0), WTCP(0x72913201, 0x3915031f), WTCP(0x7218a9a7, 0x3a04782a), WTCP(0x719e2cd2, 0x3af2eeb7), WTCP(0x7121bd9c, 0x3be062b0), WTCP(0x70a35e25, 0x3cccd004), WTCP(0x7023109a, 0x3db832a6), WTCP(0x6fa0d72c, 0x3ea2868c), WTCP(0x6f1cb416, 0x3f8bc7b4), WTCP(0x6e96a99d, 0x4073f21d), WTCP(0x6e0eba0c, 0x415b01ce), WTCP(0x6d84e7b7, 0x4240f2d1), WTCP(0x6cf934fc, 0x4325c135), WTCP(0x6c6ba43e, 0x44096910), WTCP(0x6bdc37eb, 0x44ebe679), WTCP(0x6b4af279, 0x45cd358f), WTCP(0x6ab7d663, 0x46ad5278), WTCP(0x6a22e630, 0x478c395a), WTCP(0x698c246c, 0x4869e665), WTCP(0x68f393ae, 0x494655cc), WTCP(0x68593691, 0x4a2183c8), WTCP(0x67bd0fbd, 0x4afb6c98), WTCP(0x671f21dc, 0x4bd40c80), WTCP(0x667f6fa5, 0x4cab5fc9), WTCP(0x65ddfbd3, 0x4d8162c4), WTCP(0x653ac92b, 0x4e5611c5), WTCP(0x6495da79, 0x4f296928), WTCP(0x63ef3290, 0x4ffb654d), WTCP(0x6346d44b, 0x50cc029c), WTCP(0x629cc28c, 0x519b3d80), WTCP(0x61f1003f, 0x5269126e), WTCP(0x61439053, 0x53357ddf), WTCP(0x609475c3, 0x54007c51), WTCP(0x5fe3b38d, 0x54ca0a4b), WTCP(0x5f314cba, 0x55922457), WTCP(0x5e7d4458, 0x5658c709), WTCP(0x5dc79d7c, 0x571deefa), WTCP(0x5d105b44, 0x57e198c7), WTCP(0x5c5780d3, 0x58a3c118), WTCP(0x5b9d1154, 0x59646498), WTCP(0x5ae10ff9, 0x5a237ffa), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_WTP SineWindow240[] = { WTCP(0x7fffd315, 0x006b3b9b), WTCP(0x7ffe6bbf, 0x0141b1a5), WTCP(0x7ffb9d15, 0x02182427), WTCP(0x7ff76721, 0x02ee90c8), WTCP(0x7ff1c9ef, 0x03c4f52f), WTCP(0x7feac58d, 0x049b4f00), WTCP(0x7fe25a0f, 0x05719be2), WTCP(0x7fd8878e, 0x0647d97c), WTCP(0x7fcd4e24, 0x071e0575), WTCP(0x7fc0adf2, 0x07f41d72), WTCP(0x7fb2a71b, 0x08ca1f1b), WTCP(0x7fa339c5, 0x09a00817), WTCP(0x7f92661d, 0x0a75d60e), WTCP(0x7f802c52, 0x0b4b86a8), WTCP(0x7f6c8c96, 0x0c21178c), WTCP(0x7f578721, 0x0cf68662), WTCP(0x7f411c2f, 0x0dcbd0d5), WTCP(0x7f294bfd, 0x0ea0f48c), WTCP(0x7f1016ce, 0x0f75ef33), WTCP(0x7ef57cea, 0x104abe71), WTCP(0x7ed97e9c, 0x111f5ff4), WTCP(0x7ebc1c31, 0x11f3d164), WTCP(0x7e9d55fc, 0x12c8106f), WTCP(0x7e7d2c54, 0x139c1abf), WTCP(0x7e5b9f93, 0x146fee03), WTCP(0x7e38b017, 0x154387e6), WTCP(0x7e145e42, 0x1616e618), WTCP(0x7deeaa7a, 0x16ea0646), WTCP(0x7dc79529, 0x17bce621), WTCP(0x7d9f1ebd, 0x188f8357), WTCP(0x7d7547a7, 0x1961db9b), WTCP(0x7d4a105d, 0x1a33ec9c), WTCP(0x7d1d7958, 0x1b05b40f), WTCP(0x7cef8315, 0x1bd72fa4), WTCP(0x7cc02e15, 0x1ca85d12), WTCP(0x7c8f7ade, 0x1d793a0b), WTCP(0x7c5d69f7, 0x1e49c447), WTCP(0x7c29fbee, 0x1f19f97b), WTCP(0x7bf53153, 0x1fe9d75f), WTCP(0x7bbf0aba, 0x20b95bac), WTCP(0x7b8788ba, 0x2188841a), WTCP(0x7b4eabf1, 0x22574e65), WTCP(0x7b1474fd, 0x2325b847), WTCP(0x7ad8e482, 0x23f3bf7e), WTCP(0x7a9bfb27, 0x24c161c7), WTCP(0x7a5db997, 0x258e9ce0), WTCP(0x7a1e2082, 0x265b6e8a), WTCP(0x79dd3098, 0x2727d486), WTCP(0x799aea92, 0x27f3cc94), WTCP(0x79574f28, 0x28bf547b), WTCP(0x79125f19, 0x298a69fc), WTCP(0x78cc1b26, 0x2a550adf), WTCP(0x78848414, 0x2b1f34eb), WTCP(0x783b9aad, 0x2be8e5e8), WTCP(0x77f15fbc, 0x2cb21ba0), WTCP(0x77a5d413, 0x2d7ad3de), WTCP(0x7758f886, 0x2e430c6f), WTCP(0x770acdec, 0x2f0ac320), WTCP(0x76bb5521, 0x2fd1f5c1), WTCP(0x766a8f04, 0x3098a223), WTCP(0x76187c77, 0x315ec617), WTCP(0x75c51e61, 0x32245f72), WTCP(0x757075ac, 0x32e96c09), WTCP(0x751a8346, 0x33ade9b3), WTCP(0x74c34820, 0x3471d647), WTCP(0x746ac52f, 0x35352fa1), WTCP(0x7410fb6b, 0x35f7f39c), WTCP(0x73b5ebd1, 0x36ba2014), WTCP(0x73599760, 0x377bb2e9), WTCP(0x72fbff1b, 0x383ca9fb), WTCP(0x729d2409, 0x38fd032d), WTCP(0x723d0734, 0x39bcbc63), WTCP(0x71dba9ab, 0x3a7bd382), WTCP(0x71790c7e, 0x3b3a4672), WTCP(0x711530c2, 0x3bf8131c), WTCP(0x70b01790, 0x3cb5376b), WTCP(0x7049c203, 0x3d71b14d), WTCP(0x6fe2313c, 0x3e2d7eb1), WTCP(0x6f79665b, 0x3ee89d86), WTCP(0x6f0f6289, 0x3fa30bc1), WTCP(0x6ea426ed, 0x405cc754), WTCP(0x6e37b4b6, 0x4115ce38), WTCP(0x6dca0d14, 0x41ce1e65), WTCP(0x6d5b313b, 0x4285b5d4), WTCP(0x6ceb2261, 0x433c9283), WTCP(0x6c79e1c2, 0x43f2b271), WTCP(0x6c07709b, 0x44a8139e), WTCP(0x6b93d02e, 0x455cb40c), WTCP(0x6b1f01c0, 0x461091c2), WTCP(0x6aa90697, 0x46c3aac5), WTCP(0x6a31e000, 0x4775fd1f), WTCP(0x69b98f48, 0x482786dc), WTCP(0x694015c3, 0x48d84609), WTCP(0x68c574c4, 0x498838b6), WTCP(0x6849ada3, 0x4a375cf5), WTCP(0x67ccc1be, 0x4ae5b0da), WTCP(0x674eb271, 0x4b93327c), WTCP(0x66cf8120, 0x4c3fdff4), WTCP(0x664f2f2e, 0x4cebb75c), WTCP(0x65cdbe05, 0x4d96b6d3), WTCP(0x654b2f10, 0x4e40dc79), WTCP(0x64c783bd, 0x4eea2670), WTCP(0x6442bd7e, 0x4f9292dc), WTCP(0x63bcddc7, 0x503a1fe5), WTCP(0x6335e611, 0x50e0cbb4), WTCP(0x62add7d6, 0x51869476), WTCP(0x6224b495, 0x522b7859), WTCP(0x619a7dce, 0x52cf758f), WTCP(0x610f3505, 0x53728a4a), WTCP(0x6082dbc1, 0x5414b4c1), WTCP(0x5ff5738d, 0x54b5f32c), WTCP(0x5f66fdf5, 0x555643c8), WTCP(0x5ed77c8a, 0x55f5a4d2), WTCP(0x5e46f0dd, 0x5694148b), WTCP(0x5db55c86, 0x57319135), WTCP(0x5d22c11c, 0x57ce1917), WTCP(0x5c8f203b, 0x5869aa79), WTCP(0x5bfa7b82, 0x590443a7), WTCP(0x5b64d492, 0x599de2ee), WTCP(0x5ace2d0f, 0x5a36869f), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_WTP SineWindow256[] = { WTCP(0x7fffd886, 0x006487e3), WTCP(0x7ffe9cb2, 0x012d96b1), WTCP(0x7ffc250f, 0x01f6a297), WTCP(0x7ff871a2, 0x02bfa9a4), WTCP(0x7ff38274, 0x0388a9ea), WTCP(0x7fed5791, 0x0451a177), WTCP(0x7fe5f108, 0x051a8e5c), WTCP(0x7fdd4eec, 0x05e36ea9), WTCP(0x7fd37153, 0x06ac406f), WTCP(0x7fc85854, 0x077501be), WTCP(0x7fbc040a, 0x083db0a7), WTCP(0x7fae7495, 0x09064b3a), WTCP(0x7f9faa15, 0x09cecf89), WTCP(0x7f8fa4b0, 0x0a973ba5), WTCP(0x7f7e648c, 0x0b5f8d9f), WTCP(0x7f6be9d4, 0x0c27c389), WTCP(0x7f5834b7, 0x0cefdb76), WTCP(0x7f434563, 0x0db7d376), WTCP(0x7f2d1c0e, 0x0e7fa99e), WTCP(0x7f15b8ee, 0x0f475bff), WTCP(0x7efd1c3c, 0x100ee8ad), WTCP(0x7ee34636, 0x10d64dbd), WTCP(0x7ec8371a, 0x119d8941), WTCP(0x7eabef2c, 0x1264994e), WTCP(0x7e8e6eb2, 0x132b7bf9), WTCP(0x7e6fb5f4, 0x13f22f58), WTCP(0x7e4fc53e, 0x14b8b17f), WTCP(0x7e2e9cdf, 0x157f0086), WTCP(0x7e0c3d29, 0x16451a83), WTCP(0x7de8a670, 0x170afd8d), WTCP(0x7dc3d90d, 0x17d0a7bc), WTCP(0x7d9dd55a, 0x18961728), WTCP(0x7d769bb5, 0x195b49ea), WTCP(0x7d4e2c7f, 0x1a203e1b), WTCP(0x7d24881b, 0x1ae4f1d6), WTCP(0x7cf9aef0, 0x1ba96335), WTCP(0x7ccda169, 0x1c6d9053), WTCP(0x7ca05ff1, 0x1d31774d), WTCP(0x7c71eaf9, 0x1df5163f), WTCP(0x7c4242f2, 0x1eb86b46), WTCP(0x7c116853, 0x1f7b7481), WTCP(0x7bdf5b94, 0x203e300d), WTCP(0x7bac1d31, 0x21009c0c), WTCP(0x7b77ada8, 0x21c2b69c), WTCP(0x7b420d7a, 0x22847de0), WTCP(0x7b0b3d2c, 0x2345eff8), WTCP(0x7ad33d45, 0x24070b08), WTCP(0x7a9a0e50, 0x24c7cd33), WTCP(0x7a5fb0d8, 0x2588349d), WTCP(0x7a24256f, 0x26483f6c), WTCP(0x79e76ca7, 0x2707ebc7), WTCP(0x79a98715, 0x27c737d3), WTCP(0x796a7554, 0x288621b9), WTCP(0x792a37fe, 0x2944a7a2), WTCP(0x78e8cfb2, 0x2a02c7b8), WTCP(0x78a63d11, 0x2ac08026), WTCP(0x786280bf, 0x2b7dcf17), WTCP(0x781d9b65, 0x2c3ab2b9), WTCP(0x77d78daa, 0x2cf72939), WTCP(0x7790583e, 0x2db330c7), WTCP(0x7747fbce, 0x2e6ec792), WTCP(0x76fe790e, 0x2f29ebcc), WTCP(0x76b3d0b4, 0x2fe49ba7), WTCP(0x76680376, 0x309ed556), WTCP(0x761b1211, 0x3158970e), WTCP(0x75ccfd42, 0x3211df04), WTCP(0x757dc5ca, 0x32caab6f), WTCP(0x752d6c6c, 0x3382fa88), WTCP(0x74dbf1ef, 0x343aca87), WTCP(0x7489571c, 0x34f219a8), WTCP(0x74359cbd, 0x35a8e625), WTCP(0x73e0c3a3, 0x365f2e3b), WTCP(0x738acc9e, 0x3714f02a), WTCP(0x7333b883, 0x37ca2a30), WTCP(0x72db8828, 0x387eda8e), WTCP(0x72823c67, 0x3932ff87), WTCP(0x7227d61c, 0x39e6975e), WTCP(0x71cc5626, 0x3a99a057), WTCP(0x716fbd68, 0x3b4c18ba), WTCP(0x71120cc5, 0x3bfdfecd), WTCP(0x70b34525, 0x3caf50da), WTCP(0x70536771, 0x3d600d2c), WTCP(0x6ff27497, 0x3e10320d), WTCP(0x6f906d84, 0x3ebfbdcd), WTCP(0x6f2d532c, 0x3f6eaeb8), WTCP(0x6ec92683, 0x401d0321), WTCP(0x6e63e87f, 0x40cab958), WTCP(0x6dfd9a1c, 0x4177cfb1), WTCP(0x6d963c54, 0x42244481), WTCP(0x6d2dd027, 0x42d0161e), WTCP(0x6cc45698, 0x437b42e1), WTCP(0x6c59d0a9, 0x4425c923), WTCP(0x6bee3f62, 0x44cfa740), WTCP(0x6b81a3cd, 0x4578db93), WTCP(0x6b13fef5, 0x4621647d), WTCP(0x6aa551e9, 0x46c9405c), WTCP(0x6a359db9, 0x47706d93), WTCP(0x69c4e37a, 0x4816ea86), WTCP(0x69532442, 0x48bcb599), WTCP(0x68e06129, 0x4961cd33), WTCP(0x686c9b4b, 0x4a062fbd), WTCP(0x67f7d3c5, 0x4aa9dba2), WTCP(0x67820bb7, 0x4b4ccf4d), WTCP(0x670b4444, 0x4bef092d), WTCP(0x66937e91, 0x4c9087b1), WTCP(0x661abbc5, 0x4d31494b), WTCP(0x65a0fd0b, 0x4dd14c6e), WTCP(0x6526438f, 0x4e708f8f), WTCP(0x64aa907f, 0x4f0f1126), WTCP(0x642de50d, 0x4faccfab), WTCP(0x63b0426d, 0x5049c999), WTCP(0x6331a9d4, 0x50e5fd6d), WTCP(0x62b21c7b, 0x518169a5), WTCP(0x62319b9d, 0x521c0cc2), WTCP(0x61b02876, 0x52b5e546), WTCP(0x612dc447, 0x534ef1b5), WTCP(0x60aa7050, 0x53e73097), WTCP(0x60262dd6, 0x547ea073), WTCP(0x5fa0fe1f, 0x55153fd4), WTCP(0x5f1ae274, 0x55ab0d46), WTCP(0x5e93dc1f, 0x56400758), WTCP(0x5e0bec6e, 0x56d42c99), WTCP(0x5d8314b1, 0x57677b9d), WTCP(0x5cf95638, 0x57f9f2f8), WTCP(0x5c6eb258, 0x588b9140), WTCP(0x5be32a67, 0x591c550e), WTCP(0x5b56bfbd, 0x59ac3cfd), WTCP(0x5ac973b5, 0x5a3b47ab), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_WTP SineWindow384[] = { WTCP(0x7fffee74, 0x00430546), WTCP(0x7fff6216, 0x00c90f88), WTCP(0x7ffe495b, 0x014f18ee), WTCP(0x7ffca443, 0x01d520e4), WTCP(0x7ffa72d1, 0x025b26d7), WTCP(0x7ff7b507, 0x02e12a36), WTCP(0x7ff46ae8, 0x03672a6c), WTCP(0x7ff09478, 0x03ed26e6), WTCP(0x7fec31ba, 0x04731f13), WTCP(0x7fe742b4, 0x04f9125e), WTCP(0x7fe1c76b, 0x057f0035), WTCP(0x7fdbbfe6, 0x0604e805), WTCP(0x7fd52c29, 0x068ac93b), WTCP(0x7fce0c3e, 0x0710a345), WTCP(0x7fc6602c, 0x0796758f), WTCP(0x7fbe27fa, 0x081c3f87), WTCP(0x7fb563b3, 0x08a2009a), WTCP(0x7fac135f, 0x0927b836), WTCP(0x7fa2370a, 0x09ad65c8), WTCP(0x7f97cebd, 0x0a3308bd), WTCP(0x7f8cda84, 0x0ab8a082), WTCP(0x7f815a6b, 0x0b3e2c86), WTCP(0x7f754e80, 0x0bc3ac35), WTCP(0x7f68b6ce, 0x0c491efe), WTCP(0x7f5b9364, 0x0cce844e), WTCP(0x7f4de451, 0x0d53db92), WTCP(0x7f3fa9a2, 0x0dd92439), WTCP(0x7f30e369, 0x0e5e5db0), WTCP(0x7f2191b4, 0x0ee38766), WTCP(0x7f11b495, 0x0f68a0c8), WTCP(0x7f014c1e, 0x0feda943), WTCP(0x7ef05860, 0x1072a048), WTCP(0x7eded96d, 0x10f78543), WTCP(0x7ecccf5a, 0x117c57a2), WTCP(0x7eba3a39, 0x120116d5), WTCP(0x7ea71a20, 0x1285c249), WTCP(0x7e936f22, 0x130a596e), WTCP(0x7e7f3957, 0x138edbb1), WTCP(0x7e6a78d3, 0x14134881), WTCP(0x7e552dae, 0x14979f4e), WTCP(0x7e3f57ff, 0x151bdf86), WTCP(0x7e28f7de, 0x15a00897), WTCP(0x7e120d63, 0x162419f2), WTCP(0x7dfa98a8, 0x16a81305), WTCP(0x7de299c6, 0x172bf33f), WTCP(0x7dca10d8, 0x17afba11), WTCP(0x7db0fdf8, 0x183366e9), WTCP(0x7d976142, 0x18b6f936), WTCP(0x7d7d3ad3, 0x193a706a), WTCP(0x7d628ac6, 0x19bdcbf3), WTCP(0x7d475139, 0x1a410b41), WTCP(0x7d2b8e4a, 0x1ac42dc5), WTCP(0x7d0f4218, 0x1b4732ef), WTCP(0x7cf26cc1, 0x1bca1a2f), WTCP(0x7cd50e65, 0x1c4ce2f6), WTCP(0x7cb72724, 0x1ccf8cb3), WTCP(0x7c98b71f, 0x1d5216d8), WTCP(0x7c79be78, 0x1dd480d6), WTCP(0x7c5a3d50, 0x1e56ca1e), WTCP(0x7c3a33ca, 0x1ed8f220), WTCP(0x7c19a209, 0x1f5af84f), WTCP(0x7bf88830, 0x1fdcdc1b), WTCP(0x7bd6e665, 0x205e9cf6), WTCP(0x7bb4bccb, 0x20e03a51), WTCP(0x7b920b89, 0x2161b3a0), WTCP(0x7b6ed2c5, 0x21e30853), WTCP(0x7b4b12a4, 0x226437dc), WTCP(0x7b26cb4f, 0x22e541af), WTCP(0x7b01fced, 0x2366253d), WTCP(0x7adca7a6, 0x23e6e1fa), WTCP(0x7ab6cba4, 0x24677758), WTCP(0x7a90690f, 0x24e7e4c9), WTCP(0x7a698012, 0x256829c2), WTCP(0x7a4210d8, 0x25e845b6), WTCP(0x7a1a1b8c, 0x26683818), WTCP(0x79f1a05a, 0x26e8005b), WTCP(0x79c89f6e, 0x27679df4), WTCP(0x799f18f4, 0x27e71057), WTCP(0x79750d1c, 0x286656f8), WTCP(0x794a7c12, 0x28e5714b), WTCP(0x791f6605, 0x29645ec5), WTCP(0x78f3cb25, 0x29e31edb), WTCP(0x78c7aba2, 0x2a61b101), WTCP(0x789b07ab, 0x2ae014ae), WTCP(0x786ddf72, 0x2b5e4956), WTCP(0x78403329, 0x2bdc4e6f), WTCP(0x78120300, 0x2c5a236f), WTCP(0x77e34f2c, 0x2cd7c7cc), WTCP(0x77b417df, 0x2d553afc), WTCP(0x77845d4e, 0x2dd27c75), WTCP(0x77541fab, 0x2e4f8bae), WTCP(0x77235f2d, 0x2ecc681e), WTCP(0x76f21c09, 0x2f49113d), WTCP(0x76c05674, 0x2fc58680), WTCP(0x768e0ea6, 0x3041c761), WTCP(0x765b44d5, 0x30bdd356), WTCP(0x7627f939, 0x3139a9d7), WTCP(0x75f42c0b, 0x31b54a5e), WTCP(0x75bfdd83, 0x3230b461), WTCP(0x758b0ddb, 0x32abe75a), WTCP(0x7555bd4c, 0x3326e2c3), WTCP(0x751fec11, 0x33a1a612), WTCP(0x74e99a65, 0x341c30c4), WTCP(0x74b2c884, 0x34968250), WTCP(0x747b76a9, 0x35109a31), WTCP(0x7443a512, 0x358a77e0), WTCP(0x740b53fb, 0x36041ad9), WTCP(0x73d283a2, 0x367d8296), WTCP(0x73993447, 0x36f6ae91), WTCP(0x735f6626, 0x376f9e46), WTCP(0x73251981, 0x37e85130), WTCP(0x72ea4e96, 0x3860c6cb), WTCP(0x72af05a7, 0x38d8fe93), WTCP(0x72733ef3, 0x3950f804), WTCP(0x7236fabe, 0x39c8b29a), WTCP(0x71fa3949, 0x3a402dd2), WTCP(0x71bcfad6, 0x3ab76929), WTCP(0x717f3fa8, 0x3b2e641c), WTCP(0x71410805, 0x3ba51e29), WTCP(0x7102542f, 0x3c1b96ce), WTCP(0x70c3246b, 0x3c91cd88), WTCP(0x708378ff, 0x3d07c1d6), WTCP(0x70435230, 0x3d7d7337), WTCP(0x7002b045, 0x3df2e129), WTCP(0x6fc19385, 0x3e680b2c), WTCP(0x6f7ffc37, 0x3edcf0c0), WTCP(0x6f3deaa4, 0x3f519164), WTCP(0x6efb5f12, 0x3fc5ec98), WTCP(0x6eb859cc, 0x403a01dc), WTCP(0x6e74db1c, 0x40add0b2), WTCP(0x6e30e34a, 0x4121589b), WTCP(0x6dec72a2, 0x41949917), WTCP(0x6da7896e, 0x420791a8), WTCP(0x6d6227fa, 0x427a41d0), WTCP(0x6d1c4e93, 0x42eca912), WTCP(0x6cd5fd85, 0x435ec6f0), WTCP(0x6c8f351c, 0x43d09aed), WTCP(0x6c47f5a7, 0x4442248b), WTCP(0x6c003f74, 0x44b3634f), WTCP(0x6bb812d1, 0x452456bd), WTCP(0x6b6f700e, 0x4594fe58), WTCP(0x6b265779, 0x460559a4), WTCP(0x6adcc964, 0x46756828), WTCP(0x6a92c61f, 0x46e52967), WTCP(0x6a484dfc, 0x47549ce7), WTCP(0x69fd614a, 0x47c3c22f), WTCP(0x69b2005e, 0x483298c4), WTCP(0x69662b8a, 0x48a1202c), WTCP(0x6919e320, 0x490f57ee), WTCP(0x68cd2775, 0x497d3f93), WTCP(0x687ff8dc, 0x49ead6a0), WTCP(0x683257ab, 0x4a581c9e), WTCP(0x67e44436, 0x4ac51114), WTCP(0x6795bed3, 0x4b31b38d), WTCP(0x6746c7d8, 0x4b9e0390), WTCP(0x66f75f9b, 0x4c0a00a6), WTCP(0x66a78675, 0x4c75aa5a), WTCP(0x66573cbb, 0x4ce10034), WTCP(0x660682c7, 0x4d4c01c0), WTCP(0x65b558f1, 0x4db6ae88), WTCP(0x6563bf92, 0x4e210617), WTCP(0x6511b703, 0x4e8b07f9), WTCP(0x64bf3f9f, 0x4ef4b3b9), WTCP(0x646c59bf, 0x4f5e08e3), WTCP(0x641905bf, 0x4fc70704), WTCP(0x63c543fa, 0x502fada9), WTCP(0x637114cc, 0x5097fc5e), WTCP(0x631c7892, 0x50fff2b2), WTCP(0x62c76fa7, 0x51679033), WTCP(0x6271fa69, 0x51ced46e), WTCP(0x621c1937, 0x5235bef4), WTCP(0x61c5cc6d, 0x529c4f51), WTCP(0x616f146c, 0x53028518), WTCP(0x6117f191, 0x53685fd6), WTCP(0x60c0643d, 0x53cddf1d), WTCP(0x60686ccf, 0x5433027d), WTCP(0x60100ba8, 0x5497c988), WTCP(0x5fb74129, 0x54fc33ce), WTCP(0x5f5e0db3, 0x556040e2), WTCP(0x5f0471a8, 0x55c3f056), WTCP(0x5eaa6d6b, 0x562741bd), WTCP(0x5e50015d, 0x568a34a9), WTCP(0x5df52de3, 0x56ecc8af), WTCP(0x5d99f35f, 0x574efd62), WTCP(0x5d3e5237, 0x57b0d256), WTCP(0x5ce24acd, 0x58124720), WTCP(0x5c85dd88, 0x58735b56), WTCP(0x5c290acc, 0x58d40e8c), WTCP(0x5bcbd300, 0x5934605a), WTCP(0x5b6e3689, 0x59945054), WTCP(0x5b1035cf, 0x59f3de12), WTCP(0x5ab1d138, 0x5a53092c), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_WTP SineWindow480[] = { WTCP(0x7ffff4c5, 0x00359dd2), WTCP(0x7fff9aef, 0x00a0d951), WTCP(0x7ffee744, 0x010c1460), WTCP(0x7ffdd9c4, 0x01774eb2), WTCP(0x7ffc726f, 0x01e287fc), WTCP(0x7ffab147, 0x024dbff4), WTCP(0x7ff8964d, 0x02b8f64e), WTCP(0x7ff62182, 0x03242abf), WTCP(0x7ff352e8, 0x038f5cfb), WTCP(0x7ff02a82, 0x03fa8cb8), WTCP(0x7feca851, 0x0465b9aa), WTCP(0x7fe8cc57, 0x04d0e386), WTCP(0x7fe49698, 0x053c0a01), WTCP(0x7fe00716, 0x05a72ccf), WTCP(0x7fdb1dd5, 0x06124ba5), WTCP(0x7fd5dad8, 0x067d6639), WTCP(0x7fd03e23, 0x06e87c3f), WTCP(0x7fca47b9, 0x07538d6b), WTCP(0x7fc3f7a0, 0x07be9973), WTCP(0x7fbd4dda, 0x0829a00c), WTCP(0x7fb64a6e, 0x0894a0ea), WTCP(0x7faeed5f, 0x08ff9bc2), WTCP(0x7fa736b4, 0x096a9049), WTCP(0x7f9f2671, 0x09d57e35), WTCP(0x7f96bc9c, 0x0a40653a), WTCP(0x7f8df93c, 0x0aab450d), WTCP(0x7f84dc55, 0x0b161d63), WTCP(0x7f7b65ef, 0x0b80edf1), WTCP(0x7f719611, 0x0bebb66c), WTCP(0x7f676cc0, 0x0c56768a), WTCP(0x7f5cea05, 0x0cc12dff), WTCP(0x7f520de6, 0x0d2bdc80), WTCP(0x7f46d86c, 0x0d9681c2), WTCP(0x7f3b499d, 0x0e011d7c), WTCP(0x7f2f6183, 0x0e6baf61), WTCP(0x7f232026, 0x0ed63727), WTCP(0x7f16858e, 0x0f40b483), WTCP(0x7f0991c4, 0x0fab272b), WTCP(0x7efc44d0, 0x10158ed4), WTCP(0x7eee9ebe, 0x107feb33), WTCP(0x7ee09f95, 0x10ea3bfd), WTCP(0x7ed24761, 0x115480e9), WTCP(0x7ec3962a, 0x11beb9aa), WTCP(0x7eb48bfb, 0x1228e5f8), WTCP(0x7ea528e0, 0x12930586), WTCP(0x7e956ce1, 0x12fd180b), WTCP(0x7e85580c, 0x13671d3d), WTCP(0x7e74ea6a, 0x13d114d0), WTCP(0x7e642408, 0x143afe7b), WTCP(0x7e5304f2, 0x14a4d9f4), WTCP(0x7e418d32, 0x150ea6ef), WTCP(0x7e2fbcd6, 0x15786522), WTCP(0x7e1d93ea, 0x15e21445), WTCP(0x7e0b127a, 0x164bb40b), WTCP(0x7df83895, 0x16b5442b), WTCP(0x7de50646, 0x171ec45c), WTCP(0x7dd17b9c, 0x17883452), WTCP(0x7dbd98a4, 0x17f193c5), WTCP(0x7da95d6c, 0x185ae269), WTCP(0x7d94ca03, 0x18c41ff6), WTCP(0x7d7fde76, 0x192d4c21), WTCP(0x7d6a9ad5, 0x199666a0), WTCP(0x7d54ff2e, 0x19ff6f2a), WTCP(0x7d3f0b90, 0x1a686575), WTCP(0x7d28c00c, 0x1ad14938), WTCP(0x7d121cb0, 0x1b3a1a28), WTCP(0x7cfb218c, 0x1ba2d7fc), WTCP(0x7ce3ceb2, 0x1c0b826a), WTCP(0x7ccc2430, 0x1c74192a), WTCP(0x7cb42217, 0x1cdc9bf2), WTCP(0x7c9bc87a, 0x1d450a78), WTCP(0x7c831767, 0x1dad6473), WTCP(0x7c6a0ef2, 0x1e15a99a), WTCP(0x7c50af2b, 0x1e7dd9a4), WTCP(0x7c36f824, 0x1ee5f447), WTCP(0x7c1ce9ef, 0x1f4df93a), WTCP(0x7c02849f, 0x1fb5e836), WTCP(0x7be7c847, 0x201dc0ef), WTCP(0x7bccb4f8, 0x2085831f), WTCP(0x7bb14ac5, 0x20ed2e7b), WTCP(0x7b9589c3, 0x2154c2bb), WTCP(0x7b797205, 0x21bc3f97), WTCP(0x7b5d039e, 0x2223a4c5), WTCP(0x7b403ea2, 0x228af1fe), WTCP(0x7b232325, 0x22f226f8), WTCP(0x7b05b13d, 0x2359436c), WTCP(0x7ae7e8fc, 0x23c04710), WTCP(0x7ac9ca7a, 0x2427319d), WTCP(0x7aab55ca, 0x248e02cb), WTCP(0x7a8c8b01, 0x24f4ba50), WTCP(0x7a6d6a37, 0x255b57e6), WTCP(0x7a4df380, 0x25c1db44), WTCP(0x7a2e26f2, 0x26284422), WTCP(0x7a0e04a4, 0x268e9238), WTCP(0x79ed8cad, 0x26f4c53e), WTCP(0x79ccbf22, 0x275adcee), WTCP(0x79ab9c1c, 0x27c0d8fe), WTCP(0x798a23b1, 0x2826b928), WTCP(0x796855f9, 0x288c7d24), WTCP(0x7946330c, 0x28f224ab), WTCP(0x7923bb01, 0x2957af74), WTCP(0x7900edf2, 0x29bd1d3a), WTCP(0x78ddcbf5, 0x2a226db5), WTCP(0x78ba5524, 0x2a87a09d), WTCP(0x78968998, 0x2aecb5ac), WTCP(0x7872696a, 0x2b51ac9a), WTCP(0x784df4b3, 0x2bb68522), WTCP(0x78292b8d, 0x2c1b3efb), WTCP(0x78040e12, 0x2c7fd9e0), WTCP(0x77de9c5b, 0x2ce45589), WTCP(0x77b8d683, 0x2d48b1b1), WTCP(0x7792bca5, 0x2dacee11), WTCP(0x776c4edb, 0x2e110a62), WTCP(0x77458d40, 0x2e75065e), WTCP(0x771e77f0, 0x2ed8e1c0), WTCP(0x76f70f05, 0x2f3c9c40), WTCP(0x76cf529c, 0x2fa03599), WTCP(0x76a742d1, 0x3003ad85), WTCP(0x767edfbe, 0x306703bf), WTCP(0x76562982, 0x30ca3800), WTCP(0x762d2038, 0x312d4a03), WTCP(0x7603c3fd, 0x31903982), WTCP(0x75da14ef, 0x31f30638), WTCP(0x75b01329, 0x3255afe0), WTCP(0x7585becb, 0x32b83634), WTCP(0x755b17f2, 0x331a98ef), WTCP(0x75301ebb, 0x337cd7cd), WTCP(0x7504d345, 0x33def287), WTCP(0x74d935ae, 0x3440e8da), WTCP(0x74ad4615, 0x34a2ba81), WTCP(0x74810499, 0x35046736), WTCP(0x74547158, 0x3565eeb6), WTCP(0x74278c72, 0x35c750bc), WTCP(0x73fa5607, 0x36288d03), WTCP(0x73ccce36, 0x3689a348), WTCP(0x739ef51f, 0x36ea9346), WTCP(0x7370cae2, 0x374b5cb9), WTCP(0x73424fa0, 0x37abff5d), WTCP(0x73138379, 0x380c7aee), WTCP(0x72e4668f, 0x386ccf2a), WTCP(0x72b4f902, 0x38ccfbcb), WTCP(0x72853af3, 0x392d008f), WTCP(0x72552c85, 0x398cdd32), WTCP(0x7224cdd8, 0x39ec9172), WTCP(0x71f41f0f, 0x3a4c1d09), WTCP(0x71c3204c, 0x3aab7fb7), WTCP(0x7191d1b1, 0x3b0ab937), WTCP(0x71603361, 0x3b69c947), WTCP(0x712e457f, 0x3bc8afa5), WTCP(0x70fc082d, 0x3c276c0d), WTCP(0x70c97b90, 0x3c85fe3d), WTCP(0x70969fca, 0x3ce465f3), WTCP(0x706374ff, 0x3d42a2ec), WTCP(0x702ffb54, 0x3da0b4e7), WTCP(0x6ffc32eb, 0x3dfe9ba1), WTCP(0x6fc81bea, 0x3e5c56d8), WTCP(0x6f93b676, 0x3eb9e64b), WTCP(0x6f5f02b2, 0x3f1749b8), WTCP(0x6f2a00c4, 0x3f7480dd), WTCP(0x6ef4b0d1, 0x3fd18b7a), WTCP(0x6ebf12ff, 0x402e694c), WTCP(0x6e892772, 0x408b1a12), WTCP(0x6e52ee52, 0x40e79d8c), WTCP(0x6e1c67c4, 0x4143f379), WTCP(0x6de593ee, 0x41a01b97), WTCP(0x6dae72f7, 0x41fc15a6), WTCP(0x6d770506, 0x4257e166), WTCP(0x6d3f4a40, 0x42b37e96), WTCP(0x6d0742cf, 0x430eecf6), WTCP(0x6cceeed8, 0x436a2c45), WTCP(0x6c964e83, 0x43c53c44), WTCP(0x6c5d61f9, 0x44201cb2), WTCP(0x6c242960, 0x447acd50), WTCP(0x6beaa4e2, 0x44d54ddf), WTCP(0x6bb0d4a7, 0x452f9e1e), WTCP(0x6b76b8d6, 0x4589bdcf), WTCP(0x6b3c519a, 0x45e3acb1), WTCP(0x6b019f1a, 0x463d6a87), WTCP(0x6ac6a180, 0x4696f710), WTCP(0x6a8b58f6, 0x46f0520f), WTCP(0x6a4fc5a6, 0x47497b44), WTCP(0x6a13e7b8, 0x47a27271), WTCP(0x69d7bf57, 0x47fb3757), WTCP(0x699b4cad, 0x4853c9b9), WTCP(0x695e8fe5, 0x48ac2957), WTCP(0x69218929, 0x490455f4), WTCP(0x68e438a4, 0x495c4f52), WTCP(0x68a69e81, 0x49b41533), WTCP(0x6868baec, 0x4a0ba75b), WTCP(0x682a8e0f, 0x4a63058a), WTCP(0x67ec1817, 0x4aba2f84), WTCP(0x67ad592f, 0x4b11250c), WTCP(0x676e5183, 0x4b67e5e4), WTCP(0x672f013f, 0x4bbe71d1), WTCP(0x66ef6891, 0x4c14c894), WTCP(0x66af87a4, 0x4c6ae9f2), WTCP(0x666f5ea6, 0x4cc0d5ae), WTCP(0x662eedc3, 0x4d168b8b), WTCP(0x65ee3529, 0x4d6c0b4e), WTCP(0x65ad3505, 0x4dc154bb), WTCP(0x656bed84, 0x4e166795), WTCP(0x652a5ed6, 0x4e6b43a2), WTCP(0x64e88926, 0x4ebfe8a5), WTCP(0x64a66ca5, 0x4f145662), WTCP(0x6464097f, 0x4f688ca0), WTCP(0x64215fe5, 0x4fbc8b22), WTCP(0x63de7003, 0x501051ae), WTCP(0x639b3a0b, 0x5063e008), WTCP(0x6357be2a, 0x50b735f8), WTCP(0x6313fc90, 0x510a5340), WTCP(0x62cff56c, 0x515d37a9), WTCP(0x628ba8ef, 0x51afe2f6), WTCP(0x62471749, 0x520254ef), WTCP(0x620240a8, 0x52548d59), WTCP(0x61bd253f, 0x52a68bfb), WTCP(0x6177c53c, 0x52f8509b), WTCP(0x613220d2, 0x5349daff), WTCP(0x60ec3830, 0x539b2af0), WTCP(0x60a60b88, 0x53ec4032), WTCP(0x605f9b0b, 0x543d1a8e), WTCP(0x6018e6eb, 0x548db9cb), WTCP(0x5fd1ef59, 0x54de1db1), WTCP(0x5f8ab487, 0x552e4605), WTCP(0x5f4336a7, 0x557e3292), WTCP(0x5efb75ea, 0x55cde31e), WTCP(0x5eb37285, 0x561d5771), WTCP(0x5e6b2ca8, 0x566c8f55), WTCP(0x5e22a487, 0x56bb8a90), WTCP(0x5dd9da55, 0x570a48ec), WTCP(0x5d90ce45, 0x5758ca31), WTCP(0x5d47808a, 0x57a70e29), WTCP(0x5cfdf157, 0x57f5149d), WTCP(0x5cb420e0, 0x5842dd54), WTCP(0x5c6a0f59, 0x5890681a), WTCP(0x5c1fbcf6, 0x58ddb4b8), WTCP(0x5bd529eb, 0x592ac2f7), WTCP(0x5b8a566c, 0x597792a1), WTCP(0x5b3f42ae, 0x59c42381), WTCP(0x5af3eee6, 0x5a107561), WTCP(0x5aa85b48, 0x5a5c880a), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_WTP SineWindow512[] = { WTCP(0x7ffff621, 0x003243f5), WTCP(0x7fffa72c, 0x0096cbc1), WTCP(0x7fff0943, 0x00fb5330), WTCP(0x7ffe1c65, 0x015fda03), WTCP(0x7ffce093, 0x01c45ffe), WTCP(0x7ffb55ce, 0x0228e4e2), WTCP(0x7ff97c18, 0x028d6870), WTCP(0x7ff75370, 0x02f1ea6c), WTCP(0x7ff4dbd9, 0x03566a96), WTCP(0x7ff21553, 0x03bae8b2), WTCP(0x7feeffe1, 0x041f6480), WTCP(0x7feb9b85, 0x0483ddc3), WTCP(0x7fe7e841, 0x04e8543e), WTCP(0x7fe3e616, 0x054cc7b1), WTCP(0x7fdf9508, 0x05b137df), WTCP(0x7fdaf519, 0x0615a48b), WTCP(0x7fd6064c, 0x067a0d76), WTCP(0x7fd0c8a3, 0x06de7262), WTCP(0x7fcb3c23, 0x0742d311), WTCP(0x7fc560cf, 0x07a72f45), WTCP(0x7fbf36aa, 0x080b86c2), WTCP(0x7fb8bdb8, 0x086fd947), WTCP(0x7fb1f5fc, 0x08d42699), WTCP(0x7faadf7c, 0x09386e78), WTCP(0x7fa37a3c, 0x099cb0a7), WTCP(0x7f9bc640, 0x0a00ece8), WTCP(0x7f93c38c, 0x0a6522fe), WTCP(0x7f8b7227, 0x0ac952aa), WTCP(0x7f82d214, 0x0b2d7baf), WTCP(0x7f79e35a, 0x0b919dcf), WTCP(0x7f70a5fe, 0x0bf5b8cb), WTCP(0x7f671a05, 0x0c59cc68), WTCP(0x7f5d3f75, 0x0cbdd865), WTCP(0x7f531655, 0x0d21dc87), WTCP(0x7f489eaa, 0x0d85d88f), WTCP(0x7f3dd87c, 0x0de9cc40), WTCP(0x7f32c3d1, 0x0e4db75b), WTCP(0x7f2760af, 0x0eb199a4), WTCP(0x7f1baf1e, 0x0f1572dc), WTCP(0x7f0faf25, 0x0f7942c7), WTCP(0x7f0360cb, 0x0fdd0926), WTCP(0x7ef6c418, 0x1040c5bb), WTCP(0x7ee9d914, 0x10a4784b), WTCP(0x7edc9fc6, 0x11082096), WTCP(0x7ecf1837, 0x116bbe60), WTCP(0x7ec14270, 0x11cf516a), WTCP(0x7eb31e78, 0x1232d979), WTCP(0x7ea4ac58, 0x1296564d), WTCP(0x7e95ec1a, 0x12f9c7aa), WTCP(0x7e86ddc6, 0x135d2d53), WTCP(0x7e778166, 0x13c0870a), WTCP(0x7e67d703, 0x1423d492), WTCP(0x7e57dea7, 0x148715ae), WTCP(0x7e47985b, 0x14ea4a1f), WTCP(0x7e37042a, 0x154d71aa), WTCP(0x7e26221f, 0x15b08c12), WTCP(0x7e14f242, 0x16139918), WTCP(0x7e0374a0, 0x1676987f), WTCP(0x7df1a942, 0x16d98a0c), WTCP(0x7ddf9034, 0x173c6d80), WTCP(0x7dcd2981, 0x179f429f), WTCP(0x7dba7534, 0x1802092c), WTCP(0x7da77359, 0x1864c0ea), WTCP(0x7d9423fc, 0x18c7699b), WTCP(0x7d808728, 0x192a0304), WTCP(0x7d6c9ce9, 0x198c8ce7), WTCP(0x7d58654d, 0x19ef0707), WTCP(0x7d43e05e, 0x1a517128), WTCP(0x7d2f0e2b, 0x1ab3cb0d), WTCP(0x7d19eebf, 0x1b161479), WTCP(0x7d048228, 0x1b784d30), WTCP(0x7ceec873, 0x1bda74f6), WTCP(0x7cd8c1ae, 0x1c3c8b8c), WTCP(0x7cc26de5, 0x1c9e90b8), WTCP(0x7cabcd28, 0x1d00843d), WTCP(0x7c94df83, 0x1d6265dd), WTCP(0x7c7da505, 0x1dc4355e), WTCP(0x7c661dbc, 0x1e25f282), WTCP(0x7c4e49b7, 0x1e879d0d), WTCP(0x7c362904, 0x1ee934c3), WTCP(0x7c1dbbb3, 0x1f4ab968), WTCP(0x7c0501d2, 0x1fac2abf), WTCP(0x7bebfb70, 0x200d888d), WTCP(0x7bd2a89e, 0x206ed295), WTCP(0x7bb9096b, 0x20d0089c), WTCP(0x7b9f1de6, 0x21312a65), WTCP(0x7b84e61f, 0x219237b5), WTCP(0x7b6a6227, 0x21f3304f), WTCP(0x7b4f920e, 0x225413f8), WTCP(0x7b3475e5, 0x22b4e274), WTCP(0x7b190dbc, 0x23159b88), WTCP(0x7afd59a4, 0x23763ef7), WTCP(0x7ae159ae, 0x23d6cc87), WTCP(0x7ac50dec, 0x243743fa), WTCP(0x7aa8766f, 0x2497a517), WTCP(0x7a8b9348, 0x24f7efa2), WTCP(0x7a6e648a, 0x2558235f), WTCP(0x7a50ea47, 0x25b84012), WTCP(0x7a332490, 0x26184581), WTCP(0x7a151378, 0x26783370), WTCP(0x79f6b711, 0x26d809a5), WTCP(0x79d80f6f, 0x2737c7e3), WTCP(0x79b91ca4, 0x27976df1), WTCP(0x7999dec4, 0x27f6fb92), WTCP(0x797a55e0, 0x2856708d), WTCP(0x795a820e, 0x28b5cca5), WTCP(0x793a6361, 0x29150fa1), WTCP(0x7919f9ec, 0x29743946), WTCP(0x78f945c3, 0x29d34958), WTCP(0x78d846fb, 0x2a323f9e), WTCP(0x78b6fda8, 0x2a911bdc), WTCP(0x789569df, 0x2aefddd8), WTCP(0x78738bb3, 0x2b4e8558), WTCP(0x7851633b, 0x2bad1221), WTCP(0x782ef08b, 0x2c0b83fa), WTCP(0x780c33b8, 0x2c69daa6), WTCP(0x77e92cd9, 0x2cc815ee), WTCP(0x77c5dc01, 0x2d263596), WTCP(0x77a24148, 0x2d843964), WTCP(0x777e5cc3, 0x2de2211e), WTCP(0x775a2e89, 0x2e3fec8b), WTCP(0x7735b6af, 0x2e9d9b70), WTCP(0x7710f54c, 0x2efb2d95), WTCP(0x76ebea77, 0x2f58a2be), WTCP(0x76c69647, 0x2fb5fab2), WTCP(0x76a0f8d2, 0x30133539), WTCP(0x767b1231, 0x30705217), WTCP(0x7654e279, 0x30cd5115), WTCP(0x762e69c4, 0x312a31f8), WTCP(0x7607a828, 0x3186f487), WTCP(0x75e09dbd, 0x31e39889), WTCP(0x75b94a9c, 0x32401dc6), WTCP(0x7591aedd, 0x329c8402), WTCP(0x7569ca99, 0x32f8cb07), WTCP(0x75419de7, 0x3354f29b), WTCP(0x751928e0, 0x33b0fa84), WTCP(0x74f06b9e, 0x340ce28b), WTCP(0x74c7663a, 0x3468aa76), WTCP(0x749e18cd, 0x34c4520d), WTCP(0x74748371, 0x351fd918), WTCP(0x744aa63f, 0x357b3f5d), WTCP(0x74208150, 0x35d684a6), WTCP(0x73f614c0, 0x3631a8b8), WTCP(0x73cb60a8, 0x368cab5c), WTCP(0x73a06522, 0x36e78c5b), WTCP(0x73752249, 0x37424b7b), WTCP(0x73499838, 0x379ce885), WTCP(0x731dc70a, 0x37f76341), WTCP(0x72f1aed9, 0x3851bb77), WTCP(0x72c54fc1, 0x38abf0ef), WTCP(0x7298a9dd, 0x39060373), WTCP(0x726bbd48, 0x395ff2c9), WTCP(0x723e8a20, 0x39b9bebc), WTCP(0x7211107e, 0x3a136712), WTCP(0x71e35080, 0x3a6ceb96), WTCP(0x71b54a41, 0x3ac64c0f), WTCP(0x7186fdde, 0x3b1f8848), WTCP(0x71586b74, 0x3b78a007), WTCP(0x7129931f, 0x3bd19318), WTCP(0x70fa74fc, 0x3c2a6142), WTCP(0x70cb1128, 0x3c830a50), WTCP(0x709b67c0, 0x3cdb8e09), WTCP(0x706b78e3, 0x3d33ec39), WTCP(0x703b44ad, 0x3d8c24a8), WTCP(0x700acb3c, 0x3de4371f), WTCP(0x6fda0cae, 0x3e3c2369), WTCP(0x6fa90921, 0x3e93e950), WTCP(0x6f77c0b3, 0x3eeb889c), WTCP(0x6f463383, 0x3f430119), WTCP(0x6f1461b0, 0x3f9a5290), WTCP(0x6ee24b57, 0x3ff17cca), WTCP(0x6eaff099, 0x40487f94), WTCP(0x6e7d5193, 0x409f5ab6), WTCP(0x6e4a6e66, 0x40f60dfb), WTCP(0x6e174730, 0x414c992f), WTCP(0x6de3dc11, 0x41a2fc1a), WTCP(0x6db02d29, 0x41f93689), WTCP(0x6d7c3a98, 0x424f4845), WTCP(0x6d48047e, 0x42a5311b), WTCP(0x6d138afb, 0x42faf0d4), WTCP(0x6cdece2f, 0x4350873c), WTCP(0x6ca9ce3b, 0x43a5f41e), WTCP(0x6c748b3f, 0x43fb3746), WTCP(0x6c3f055d, 0x4450507e), WTCP(0x6c093cb6, 0x44a53f93), WTCP(0x6bd3316a, 0x44fa0450), WTCP(0x6b9ce39b, 0x454e9e80), WTCP(0x6b66536b, 0x45a30df0), WTCP(0x6b2f80fb, 0x45f7526b), WTCP(0x6af86c6c, 0x464b6bbe), WTCP(0x6ac115e2, 0x469f59b4), WTCP(0x6a897d7d, 0x46f31c1a), WTCP(0x6a51a361, 0x4746b2bc), WTCP(0x6a1987b0, 0x479a1d67), WTCP(0x69e12a8c, 0x47ed5be6), WTCP(0x69a88c19, 0x48406e08), WTCP(0x696fac78, 0x48935397), WTCP(0x69368bce, 0x48e60c62), WTCP(0x68fd2a3d, 0x49389836), WTCP(0x68c387e9, 0x498af6df), WTCP(0x6889a4f6, 0x49dd282a), WTCP(0x684f8186, 0x4a2f2be6), WTCP(0x68151dbe, 0x4a8101de), WTCP(0x67da79c3, 0x4ad2a9e2), WTCP(0x679f95b7, 0x4b2423be), WTCP(0x676471c0, 0x4b756f40), WTCP(0x67290e02, 0x4bc68c36), WTCP(0x66ed6aa1, 0x4c177a6e), WTCP(0x66b187c3, 0x4c6839b7), WTCP(0x6675658c, 0x4cb8c9dd), WTCP(0x66390422, 0x4d092ab0), WTCP(0x65fc63a9, 0x4d595bfe), WTCP(0x65bf8447, 0x4da95d96), WTCP(0x65826622, 0x4df92f46), WTCP(0x6545095f, 0x4e48d0dd), WTCP(0x65076e25, 0x4e984229), WTCP(0x64c99498, 0x4ee782fb), WTCP(0x648b7ce0, 0x4f369320), WTCP(0x644d2722, 0x4f857269), WTCP(0x640e9386, 0x4fd420a4), WTCP(0x63cfc231, 0x50229da1), WTCP(0x6390b34a, 0x5070e92f), WTCP(0x635166f9, 0x50bf031f), WTCP(0x6311dd64, 0x510ceb40), WTCP(0x62d216b3, 0x515aa162), WTCP(0x6292130c, 0x51a82555), WTCP(0x6251d298, 0x51f576ea), WTCP(0x6211557e, 0x524295f0), WTCP(0x61d09be5, 0x528f8238), WTCP(0x618fa5f7, 0x52dc3b92), WTCP(0x614e73da, 0x5328c1d0), WTCP(0x610d05b7, 0x537514c2), WTCP(0x60cb5bb7, 0x53c13439), WTCP(0x60897601, 0x540d2005), WTCP(0x604754bf, 0x5458d7f9), WTCP(0x6004f819, 0x54a45be6), WTCP(0x5fc26038, 0x54efab9c), WTCP(0x5f7f8d46, 0x553ac6ee), WTCP(0x5f3c7f6b, 0x5585adad), WTCP(0x5ef936d1, 0x55d05faa), WTCP(0x5eb5b3a2, 0x561adcb9), WTCP(0x5e71f606, 0x566524aa), WTCP(0x5e2dfe29, 0x56af3750), WTCP(0x5de9cc33, 0x56f9147e), WTCP(0x5da5604f, 0x5742bc06), WTCP(0x5d60baa7, 0x578c2dba), WTCP(0x5d1bdb65, 0x57d5696d), WTCP(0x5cd6c2b5, 0x581e6ef1), WTCP(0x5c9170bf, 0x58673e1b), WTCP(0x5c4be5b0, 0x58afd6bd), WTCP(0x5c0621b2, 0x58f838a9), WTCP(0x5bc024f0, 0x594063b5), WTCP(0x5b79ef96, 0x598857b2), WTCP(0x5b3381ce, 0x59d01475), WTCP(0x5aecdbc5, 0x5a1799d1), WTCP(0x5aa5fda5, 0x5a5ee79a), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_WTP SineWindow768[] = { WTCP(0x7ffffb9d, 0x002182a4), WTCP(0x7fffd886, 0x006487e3), WTCP(0x7fff9257, 0x00a78d06), WTCP(0x7fff2910, 0x00ea91fc), WTCP(0x7ffe9cb2, 0x012d96b1), WTCP(0x7ffded3d, 0x01709b13), WTCP(0x7ffd1ab2, 0x01b39f11), WTCP(0x7ffc250f, 0x01f6a297), WTCP(0x7ffb0c56, 0x0239a593), WTCP(0x7ff9d087, 0x027ca7f3), WTCP(0x7ff871a2, 0x02bfa9a4), WTCP(0x7ff6efa7, 0x0302aa95), WTCP(0x7ff54a98, 0x0345aab2), WTCP(0x7ff38274, 0x0388a9ea), WTCP(0x7ff1973b, 0x03cba829), WTCP(0x7fef88ef, 0x040ea55e), WTCP(0x7fed5791, 0x0451a177), WTCP(0x7feb031f, 0x04949c60), WTCP(0x7fe88b9c, 0x04d79608), WTCP(0x7fe5f108, 0x051a8e5c), WTCP(0x7fe33364, 0x055d854a), WTCP(0x7fe052af, 0x05a07abf), WTCP(0x7fdd4eec, 0x05e36ea9), WTCP(0x7fda281b, 0x062660f6), WTCP(0x7fd6de3d, 0x06695194), WTCP(0x7fd37153, 0x06ac406f), WTCP(0x7fcfe15d, 0x06ef2d76), WTCP(0x7fcc2e5d, 0x07321897), WTCP(0x7fc85854, 0x077501be), WTCP(0x7fc45f42, 0x07b7e8da), WTCP(0x7fc04329, 0x07facdd9), WTCP(0x7fbc040a, 0x083db0a7), WTCP(0x7fb7a1e6, 0x08809133), WTCP(0x7fb31cbf, 0x08c36f6a), WTCP(0x7fae7495, 0x09064b3a), WTCP(0x7fa9a96a, 0x09492491), WTCP(0x7fa4bb3f, 0x098bfb5c), WTCP(0x7f9faa15, 0x09cecf89), WTCP(0x7f9a75ef, 0x0a11a106), WTCP(0x7f951ecc, 0x0a546fc0), WTCP(0x7f8fa4b0, 0x0a973ba5), WTCP(0x7f8a079a, 0x0ada04a3), WTCP(0x7f84478e, 0x0b1ccaa7), WTCP(0x7f7e648c, 0x0b5f8d9f), WTCP(0x7f785e96, 0x0ba24d79), WTCP(0x7f7235ad, 0x0be50a23), WTCP(0x7f6be9d4, 0x0c27c389), WTCP(0x7f657b0c, 0x0c6a799b), WTCP(0x7f5ee957, 0x0cad2c45), WTCP(0x7f5834b7, 0x0cefdb76), WTCP(0x7f515d2d, 0x0d32871a), WTCP(0x7f4a62bb, 0x0d752f20), WTCP(0x7f434563, 0x0db7d376), WTCP(0x7f3c0528, 0x0dfa7409), WTCP(0x7f34a20b, 0x0e3d10c7), WTCP(0x7f2d1c0e, 0x0e7fa99e), WTCP(0x7f257334, 0x0ec23e7b), WTCP(0x7f1da77e, 0x0f04cf4c), WTCP(0x7f15b8ee, 0x0f475bff), WTCP(0x7f0da787, 0x0f89e482), WTCP(0x7f05734b, 0x0fcc68c2), WTCP(0x7efd1c3c, 0x100ee8ad), WTCP(0x7ef4a25d, 0x10516432), WTCP(0x7eec05af, 0x1093db3d), WTCP(0x7ee34636, 0x10d64dbd), WTCP(0x7eda63f3, 0x1118bb9f), WTCP(0x7ed15ee9, 0x115b24d1), WTCP(0x7ec8371a, 0x119d8941), WTCP(0x7ebeec89, 0x11dfe8dc), WTCP(0x7eb57f39, 0x12224392), WTCP(0x7eabef2c, 0x1264994e), WTCP(0x7ea23c65, 0x12a6ea00), WTCP(0x7e9866e6, 0x12e93594), WTCP(0x7e8e6eb2, 0x132b7bf9), WTCP(0x7e8453cc, 0x136dbd1d), WTCP(0x7e7a1636, 0x13aff8ed), WTCP(0x7e6fb5f4, 0x13f22f58), WTCP(0x7e653308, 0x1434604a), WTCP(0x7e5a8d75, 0x14768bb3), WTCP(0x7e4fc53e, 0x14b8b17f), WTCP(0x7e44da66, 0x14fad19e), WTCP(0x7e39ccf0, 0x153cebfb), WTCP(0x7e2e9cdf, 0x157f0086), WTCP(0x7e234a36, 0x15c10f2d), WTCP(0x7e17d4f8, 0x160317dc), WTCP(0x7e0c3d29, 0x16451a83), WTCP(0x7e0082cb, 0x1687170f), WTCP(0x7df4a5e2, 0x16c90d6e), WTCP(0x7de8a670, 0x170afd8d), WTCP(0x7ddc847a, 0x174ce75b), WTCP(0x7dd04003, 0x178ecac6), WTCP(0x7dc3d90d, 0x17d0a7bc), WTCP(0x7db74f9d, 0x18127e2a), WTCP(0x7daaa3b5, 0x18544dff), WTCP(0x7d9dd55a, 0x18961728), WTCP(0x7d90e48f, 0x18d7d993), WTCP(0x7d83d156, 0x1919952f), WTCP(0x7d769bb5, 0x195b49ea), WTCP(0x7d6943ae, 0x199cf7b0), WTCP(0x7d5bc946, 0x19de9e72), WTCP(0x7d4e2c7f, 0x1a203e1b), WTCP(0x7d406d5e, 0x1a61d69b), WTCP(0x7d328be6, 0x1aa367df), WTCP(0x7d24881b, 0x1ae4f1d6), WTCP(0x7d166201, 0x1b26746d), WTCP(0x7d08199c, 0x1b67ef93), WTCP(0x7cf9aef0, 0x1ba96335), WTCP(0x7ceb2201, 0x1beacf42), WTCP(0x7cdc72d3, 0x1c2c33a7), WTCP(0x7ccda169, 0x1c6d9053), WTCP(0x7cbeadc8, 0x1caee534), WTCP(0x7caf97f4, 0x1cf03238), WTCP(0x7ca05ff1, 0x1d31774d), WTCP(0x7c9105c3, 0x1d72b461), WTCP(0x7c81896f, 0x1db3e962), WTCP(0x7c71eaf9, 0x1df5163f), WTCP(0x7c622a64, 0x1e363ae5), WTCP(0x7c5247b6, 0x1e775743), WTCP(0x7c4242f2, 0x1eb86b46), WTCP(0x7c321c1e, 0x1ef976de), WTCP(0x7c21d33c, 0x1f3a79f7), WTCP(0x7c116853, 0x1f7b7481), WTCP(0x7c00db66, 0x1fbc6669), WTCP(0x7bf02c7b, 0x1ffd4f9e), WTCP(0x7bdf5b94, 0x203e300d), WTCP(0x7bce68b8, 0x207f07a6), WTCP(0x7bbd53eb, 0x20bfd656), WTCP(0x7bac1d31, 0x21009c0c), WTCP(0x7b9ac490, 0x214158b5), WTCP(0x7b894a0b, 0x21820c41), WTCP(0x7b77ada8, 0x21c2b69c), WTCP(0x7b65ef6c, 0x220357b6), WTCP(0x7b540f5b, 0x2243ef7d), WTCP(0x7b420d7a, 0x22847de0), WTCP(0x7b2fe9cf, 0x22c502cb), WTCP(0x7b1da45e, 0x23057e2e), WTCP(0x7b0b3d2c, 0x2345eff8), WTCP(0x7af8b43f, 0x23865816), WTCP(0x7ae6099b, 0x23c6b676), WTCP(0x7ad33d45, 0x24070b08), WTCP(0x7ac04f44, 0x244755b9), WTCP(0x7aad3f9b, 0x24879678), WTCP(0x7a9a0e50, 0x24c7cd33), WTCP(0x7a86bb68, 0x2507f9d8), WTCP(0x7a7346e9, 0x25481c57), WTCP(0x7a5fb0d8, 0x2588349d), WTCP(0x7a4bf93a, 0x25c84299), WTCP(0x7a382015, 0x26084639), WTCP(0x7a24256f, 0x26483f6c), WTCP(0x7a10094c, 0x26882e21), WTCP(0x79fbcbb2, 0x26c81245), WTCP(0x79e76ca7, 0x2707ebc7), WTCP(0x79d2ec30, 0x2747ba95), WTCP(0x79be4a53, 0x27877e9f), WTCP(0x79a98715, 0x27c737d3), WTCP(0x7994a27d, 0x2806e61f), WTCP(0x797f9c90, 0x28468971), WTCP(0x796a7554, 0x288621b9), WTCP(0x79552cce, 0x28c5aee5), WTCP(0x793fc305, 0x290530e3), WTCP(0x792a37fe, 0x2944a7a2), WTCP(0x79148bbf, 0x29841311), WTCP(0x78febe4e, 0x29c3731e), WTCP(0x78e8cfb2, 0x2a02c7b8), WTCP(0x78d2bfef, 0x2a4210ce), WTCP(0x78bc8f0d, 0x2a814e4d), WTCP(0x78a63d11, 0x2ac08026), WTCP(0x788fca01, 0x2affa646), WTCP(0x787935e4, 0x2b3ec09c), WTCP(0x786280bf, 0x2b7dcf17), WTCP(0x784baa9a, 0x2bbcd1a6), WTCP(0x7834b37a, 0x2bfbc837), WTCP(0x781d9b65, 0x2c3ab2b9), WTCP(0x78066262, 0x2c79911b), WTCP(0x77ef0877, 0x2cb8634b), WTCP(0x77d78daa, 0x2cf72939), WTCP(0x77bff203, 0x2d35e2d3), WTCP(0x77a83587, 0x2d749008), WTCP(0x7790583e, 0x2db330c7), WTCP(0x77785a2d, 0x2df1c4fe), WTCP(0x77603b5a, 0x2e304c9d), WTCP(0x7747fbce, 0x2e6ec792), WTCP(0x772f9b8e, 0x2ead35cd), WTCP(0x77171aa1, 0x2eeb973b), WTCP(0x76fe790e, 0x2f29ebcc), WTCP(0x76e5b6dc, 0x2f68336f), WTCP(0x76ccd411, 0x2fa66e13), WTCP(0x76b3d0b4, 0x2fe49ba7), WTCP(0x769aaccc, 0x3022bc19), WTCP(0x7681685f, 0x3060cf59), WTCP(0x76680376, 0x309ed556), WTCP(0x764e7e17, 0x30dccdfe), WTCP(0x7634d848, 0x311ab941), WTCP(0x761b1211, 0x3158970e), WTCP(0x76012b79, 0x31966753), WTCP(0x75e72487, 0x31d42a00), WTCP(0x75ccfd42, 0x3211df04), WTCP(0x75b2b5b2, 0x324f864e), WTCP(0x75984ddc, 0x328d1fcc), WTCP(0x757dc5ca, 0x32caab6f), WTCP(0x75631d82, 0x33082925), WTCP(0x7548550b, 0x334598de), WTCP(0x752d6c6c, 0x3382fa88), WTCP(0x751263ae, 0x33c04e13), WTCP(0x74f73ad7, 0x33fd936e), WTCP(0x74dbf1ef, 0x343aca87), WTCP(0x74c088fe, 0x3477f350), WTCP(0x74a5000a, 0x34b50db5), WTCP(0x7489571c, 0x34f219a8), WTCP(0x746d8e3a, 0x352f1716), WTCP(0x7451a56e, 0x356c05f0), WTCP(0x74359cbd, 0x35a8e625), WTCP(0x74197431, 0x35e5b7a3), WTCP(0x73fd2bd0, 0x36227a5b), WTCP(0x73e0c3a3, 0x365f2e3b), WTCP(0x73c43bb1, 0x369bd334), WTCP(0x73a79402, 0x36d86934), WTCP(0x738acc9e, 0x3714f02a), WTCP(0x736de58d, 0x37516807), WTCP(0x7350ded7, 0x378dd0b9), WTCP(0x7333b883, 0x37ca2a30), WTCP(0x7316729a, 0x3806745c), WTCP(0x72f90d24, 0x3842af2b), WTCP(0x72db8828, 0x387eda8e), WTCP(0x72bde3af, 0x38baf674), WTCP(0x72a01fc2, 0x38f702cd), WTCP(0x72823c67, 0x3932ff87), WTCP(0x726439a8, 0x396eec93), WTCP(0x7246178c, 0x39aac9e0), WTCP(0x7227d61c, 0x39e6975e), WTCP(0x72097560, 0x3a2254fc), WTCP(0x71eaf561, 0x3a5e02aa), WTCP(0x71cc5626, 0x3a99a057), WTCP(0x71ad97b9, 0x3ad52df4), WTCP(0x718eba22, 0x3b10ab70), WTCP(0x716fbd68, 0x3b4c18ba), WTCP(0x7150a195, 0x3b8775c2), WTCP(0x713166b1, 0x3bc2c279), WTCP(0x71120cc5, 0x3bfdfecd), WTCP(0x70f293d9, 0x3c392aaf), WTCP(0x70d2fbf6, 0x3c74460e), WTCP(0x70b34525, 0x3caf50da), WTCP(0x70936f6e, 0x3cea4b04), WTCP(0x70737ad9, 0x3d253479), WTCP(0x70536771, 0x3d600d2c), WTCP(0x7033353d, 0x3d9ad50b), WTCP(0x7012e447, 0x3dd58c06), WTCP(0x6ff27497, 0x3e10320d), WTCP(0x6fd1e635, 0x3e4ac711), WTCP(0x6fb1392c, 0x3e854b01), WTCP(0x6f906d84, 0x3ebfbdcd), WTCP(0x6f6f8346, 0x3efa1f65), WTCP(0x6f4e7a7b, 0x3f346fb8), WTCP(0x6f2d532c, 0x3f6eaeb8), WTCP(0x6f0c0d62, 0x3fa8dc54), WTCP(0x6eeaa927, 0x3fe2f87c), WTCP(0x6ec92683, 0x401d0321), WTCP(0x6ea7857f, 0x4056fc31), WTCP(0x6e85c626, 0x4090e39e), WTCP(0x6e63e87f, 0x40cab958), WTCP(0x6e41ec95, 0x41047d4e), WTCP(0x6e1fd271, 0x413e2f71), WTCP(0x6dfd9a1c, 0x4177cfb1), WTCP(0x6ddb439f, 0x41b15dfe), WTCP(0x6db8cf04, 0x41eada49), WTCP(0x6d963c54, 0x42244481), WTCP(0x6d738b99, 0x425d9c97), WTCP(0x6d50bcdc, 0x4296e27b), WTCP(0x6d2dd027, 0x42d0161e), WTCP(0x6d0ac584, 0x43093770), WTCP(0x6ce79cfc, 0x43424661), WTCP(0x6cc45698, 0x437b42e1), WTCP(0x6ca0f262, 0x43b42ce1), WTCP(0x6c7d7065, 0x43ed0452), WTCP(0x6c59d0a9, 0x4425c923), WTCP(0x6c361339, 0x445e7b46), WTCP(0x6c12381e, 0x44971aaa), WTCP(0x6bee3f62, 0x44cfa740), WTCP(0x6bca2910, 0x450820f8), WTCP(0x6ba5f530, 0x454087c4), WTCP(0x6b81a3cd, 0x4578db93), WTCP(0x6b5d34f1, 0x45b11c57), WTCP(0x6b38a8a6, 0x45e949ff), WTCP(0x6b13fef5, 0x4621647d), WTCP(0x6aef37e9, 0x46596bc1), WTCP(0x6aca538c, 0x46915fbb), WTCP(0x6aa551e9, 0x46c9405c), WTCP(0x6a803308, 0x47010d96), WTCP(0x6a5af6f5, 0x4738c758), WTCP(0x6a359db9, 0x47706d93), WTCP(0x6a102760, 0x47a80039), WTCP(0x69ea93f2, 0x47df7f3a), WTCP(0x69c4e37a, 0x4816ea86), WTCP(0x699f1604, 0x484e420f), WTCP(0x69792b98, 0x488585c5), WTCP(0x69532442, 0x48bcb599), WTCP(0x692d000c, 0x48f3d17c), WTCP(0x6906bf00, 0x492ad95f), WTCP(0x68e06129, 0x4961cd33), WTCP(0x68b9e692, 0x4998ace9), WTCP(0x68934f44, 0x49cf7871), WTCP(0x686c9b4b, 0x4a062fbd), WTCP(0x6845cab1, 0x4a3cd2be), WTCP(0x681edd81, 0x4a736165), WTCP(0x67f7d3c5, 0x4aa9dba2), WTCP(0x67d0ad88, 0x4ae04167), WTCP(0x67a96ad5, 0x4b1692a5), WTCP(0x67820bb7, 0x4b4ccf4d), WTCP(0x675a9038, 0x4b82f750), WTCP(0x6732f863, 0x4bb90aa0), WTCP(0x670b4444, 0x4bef092d), WTCP(0x66e373e4, 0x4c24f2e9), WTCP(0x66bb8750, 0x4c5ac7c4), WTCP(0x66937e91, 0x4c9087b1), WTCP(0x666b59b3, 0x4cc632a0), WTCP(0x664318c0, 0x4cfbc883), WTCP(0x661abbc5, 0x4d31494b), WTCP(0x65f242cc, 0x4d66b4e9), WTCP(0x65c9addf, 0x4d9c0b4f), WTCP(0x65a0fd0b, 0x4dd14c6e), WTCP(0x6578305a, 0x4e067837), WTCP(0x654f47d7, 0x4e3b8e9d), WTCP(0x6526438f, 0x4e708f8f), WTCP(0x64fd238b, 0x4ea57b01), WTCP(0x64d3e7d7, 0x4eda50e2), WTCP(0x64aa907f, 0x4f0f1126), WTCP(0x64811d8e, 0x4f43bbbd), WTCP(0x64578f0f, 0x4f785099), WTCP(0x642de50d, 0x4faccfab), WTCP(0x64041f95, 0x4fe138e5), WTCP(0x63da3eb1, 0x50158c39), WTCP(0x63b0426d, 0x5049c999), WTCP(0x63862ad5, 0x507df0f6), WTCP(0x635bf7f3, 0x50b20241), WTCP(0x6331a9d4, 0x50e5fd6d), WTCP(0x63074084, 0x5119e26b), WTCP(0x62dcbc0d, 0x514db12d), WTCP(0x62b21c7b, 0x518169a5), WTCP(0x628761db, 0x51b50bc4), WTCP(0x625c8c38, 0x51e8977d), WTCP(0x62319b9d, 0x521c0cc2), WTCP(0x62069017, 0x524f6b83), WTCP(0x61db69b1, 0x5282b3b4), WTCP(0x61b02876, 0x52b5e546), WTCP(0x6184cc74, 0x52e9002a), WTCP(0x615955b6, 0x531c0454), WTCP(0x612dc447, 0x534ef1b5), WTCP(0x61021834, 0x5381c83f), WTCP(0x60d65188, 0x53b487e5), WTCP(0x60aa7050, 0x53e73097), WTCP(0x607e7497, 0x5419c249), WTCP(0x60525e6b, 0x544c3cec), WTCP(0x60262dd6, 0x547ea073), WTCP(0x5ff9e2e5, 0x54b0ecd0), WTCP(0x5fcd7da4, 0x54e321f5), WTCP(0x5fa0fe1f, 0x55153fd4), WTCP(0x5f746462, 0x55474660), WTCP(0x5f47b07a, 0x5579358b), WTCP(0x5f1ae274, 0x55ab0d46), WTCP(0x5eedfa5a, 0x55dccd86), WTCP(0x5ec0f839, 0x560e763b), WTCP(0x5e93dc1f, 0x56400758), WTCP(0x5e66a617, 0x567180d0), WTCP(0x5e39562d, 0x56a2e295), WTCP(0x5e0bec6e, 0x56d42c99), WTCP(0x5dde68e7, 0x57055ed0), WTCP(0x5db0cba4, 0x5736792b), WTCP(0x5d8314b1, 0x57677b9d), WTCP(0x5d55441b, 0x57986619), WTCP(0x5d2759ee, 0x57c93891), WTCP(0x5cf95638, 0x57f9f2f8), WTCP(0x5ccb3905, 0x582a9540), WTCP(0x5c9d0260, 0x585b1f5c), WTCP(0x5c6eb258, 0x588b9140), WTCP(0x5c4048f9, 0x58bbeadd), WTCP(0x5c11c64f, 0x58ec2c26), WTCP(0x5be32a67, 0x591c550e), WTCP(0x5bb4754e, 0x594c6588), WTCP(0x5b85a711, 0x597c5d87), WTCP(0x5b56bfbd, 0x59ac3cfd), WTCP(0x5b27bf5e, 0x59dc03de), WTCP(0x5af8a602, 0x5a0bb21c), WTCP(0x5ac973b5, 0x5a3b47ab), WTCP(0x5a9a2884, 0x5a6ac47c), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_WTP SineWindow960[] = { WTCP(0x7ffffd31, 0x001aceea), WTCP(0x7fffe6bc, 0x00506cb9), WTCP(0x7fffb9d1, 0x00860a79), WTCP(0x7fff7671, 0x00bba822), WTCP(0x7fff1c9b, 0x00f145ab), WTCP(0x7ffeac50, 0x0126e309), WTCP(0x7ffe2590, 0x015c8033), WTCP(0x7ffd885a, 0x01921d20), WTCP(0x7ffcd4b0, 0x01c7b9c6), WTCP(0x7ffc0a91, 0x01fd561d), WTCP(0x7ffb29fd, 0x0232f21a), WTCP(0x7ffa32f4, 0x02688db4), WTCP(0x7ff92577, 0x029e28e2), WTCP(0x7ff80186, 0x02d3c39b), WTCP(0x7ff6c720, 0x03095dd5), WTCP(0x7ff57647, 0x033ef786), WTCP(0x7ff40efa, 0x037490a5), WTCP(0x7ff2913a, 0x03aa292a), WTCP(0x7ff0fd07, 0x03dfc109), WTCP(0x7fef5260, 0x0415583b), WTCP(0x7fed9148, 0x044aeeb5), WTCP(0x7febb9bd, 0x0480846e), WTCP(0x7fe9cbc0, 0x04b6195d), WTCP(0x7fe7c752, 0x04ebad79), WTCP(0x7fe5ac72, 0x052140b7), WTCP(0x7fe37b22, 0x0556d30f), WTCP(0x7fe13361, 0x058c6478), WTCP(0x7fded530, 0x05c1f4e7), WTCP(0x7fdc608f, 0x05f78453), WTCP(0x7fd9d57f, 0x062d12b4), WTCP(0x7fd73401, 0x06629ffe), WTCP(0x7fd47c14, 0x06982c2b), WTCP(0x7fd1adb9, 0x06cdb72f), WTCP(0x7fcec8f1, 0x07034101), WTCP(0x7fcbcdbc, 0x0738c998), WTCP(0x7fc8bc1b, 0x076e50eb), WTCP(0x7fc5940e, 0x07a3d6f0), WTCP(0x7fc25596, 0x07d95b9e), WTCP(0x7fbf00b3, 0x080edeec), WTCP(0x7fbb9567, 0x084460cf), WTCP(0x7fb813b0, 0x0879e140), WTCP(0x7fb47b91, 0x08af6033), WTCP(0x7fb0cd0a, 0x08e4dda0), WTCP(0x7fad081b, 0x091a597e), WTCP(0x7fa92cc5, 0x094fd3c3), WTCP(0x7fa53b09, 0x09854c66), WTCP(0x7fa132e8, 0x09bac35d), WTCP(0x7f9d1461, 0x09f0389f), WTCP(0x7f98df77, 0x0a25ac23), WTCP(0x7f949429, 0x0a5b1dde), WTCP(0x7f903279, 0x0a908dc9), WTCP(0x7f8bba66, 0x0ac5fbd9), WTCP(0x7f872bf3, 0x0afb6805), WTCP(0x7f82871f, 0x0b30d244), WTCP(0x7f7dcbec, 0x0b663a8c), WTCP(0x7f78fa5b, 0x0b9ba0d5), WTCP(0x7f74126b, 0x0bd10513), WTCP(0x7f6f141f, 0x0c066740), WTCP(0x7f69ff76, 0x0c3bc74f), WTCP(0x7f64d473, 0x0c71253a), WTCP(0x7f5f9315, 0x0ca680f5), WTCP(0x7f5a3b5e, 0x0cdbda79), WTCP(0x7f54cd4f, 0x0d1131ba), WTCP(0x7f4f48e8, 0x0d4686b1), WTCP(0x7f49ae2a, 0x0d7bd954), WTCP(0x7f43fd18, 0x0db12999), WTCP(0x7f3e35b0, 0x0de67776), WTCP(0x7f3857f6, 0x0e1bc2e4), WTCP(0x7f3263e9, 0x0e510bd8), WTCP(0x7f2c598a, 0x0e865248), WTCP(0x7f2638db, 0x0ebb962c), WTCP(0x7f2001dd, 0x0ef0d77b), WTCP(0x7f19b491, 0x0f26162a), WTCP(0x7f1350f8, 0x0f5b5231), WTCP(0x7f0cd712, 0x0f908b86), WTCP(0x7f0646e2, 0x0fc5c220), WTCP(0x7effa069, 0x0ffaf5f6), WTCP(0x7ef8e3a6, 0x103026fe), WTCP(0x7ef2109d, 0x1065552e), WTCP(0x7eeb274d, 0x109a807e), WTCP(0x7ee427b9, 0x10cfa8e5), WTCP(0x7edd11e1, 0x1104ce58), WTCP(0x7ed5e5c6, 0x1139f0cf), WTCP(0x7ecea36b, 0x116f1040), WTCP(0x7ec74acf, 0x11a42ca2), WTCP(0x7ebfdbf5, 0x11d945eb), WTCP(0x7eb856de, 0x120e5c13), WTCP(0x7eb0bb8a, 0x12436f10), WTCP(0x7ea909fc, 0x12787ed8), WTCP(0x7ea14235, 0x12ad8b63), WTCP(0x7e996436, 0x12e294a7), WTCP(0x7e917000, 0x13179a9b), WTCP(0x7e896595, 0x134c9d34), WTCP(0x7e8144f6, 0x13819c6c), WTCP(0x7e790e25, 0x13b69836), WTCP(0x7e70c124, 0x13eb908c), WTCP(0x7e685df2, 0x14208563), WTCP(0x7e5fe493, 0x145576b1), WTCP(0x7e575508, 0x148a646e), WTCP(0x7e4eaf51, 0x14bf4e91), WTCP(0x7e45f371, 0x14f43510), WTCP(0x7e3d2169, 0x152917e1), WTCP(0x7e34393b, 0x155df6fc), WTCP(0x7e2b3ae8, 0x1592d257), WTCP(0x7e222672, 0x15c7a9ea), WTCP(0x7e18fbda, 0x15fc7daa), WTCP(0x7e0fbb22, 0x16314d8e), WTCP(0x7e06644c, 0x1666198d), WTCP(0x7dfcf759, 0x169ae19f), WTCP(0x7df3744b, 0x16cfa5b9), WTCP(0x7de9db23, 0x170465d2), WTCP(0x7de02be4, 0x173921e2), WTCP(0x7dd6668f, 0x176dd9de), WTCP(0x7dcc8b25, 0x17a28dbe), WTCP(0x7dc299a9, 0x17d73d79), WTCP(0x7db8921c, 0x180be904), WTCP(0x7dae747f, 0x18409058), WTCP(0x7da440d6, 0x1875336a), WTCP(0x7d99f721, 0x18a9d231), WTCP(0x7d8f9762, 0x18de6ca5), WTCP(0x7d85219c, 0x191302bc), WTCP(0x7d7a95cf, 0x1947946c), WTCP(0x7d6ff3fe, 0x197c21ad), WTCP(0x7d653c2b, 0x19b0aa75), WTCP(0x7d5a6e57, 0x19e52ebb), WTCP(0x7d4f8a85, 0x1a19ae76), WTCP(0x7d4490b6, 0x1a4e299d), WTCP(0x7d3980ec, 0x1a82a026), WTCP(0x7d2e5b2a, 0x1ab71208), WTCP(0x7d231f70, 0x1aeb7f3a), WTCP(0x7d17cdc2, 0x1b1fe7b3), WTCP(0x7d0c6621, 0x1b544b6a), WTCP(0x7d00e88f, 0x1b88aa55), WTCP(0x7cf5550e, 0x1bbd046c), WTCP(0x7ce9aba1, 0x1bf159a4), WTCP(0x7cddec48, 0x1c25a9f6), WTCP(0x7cd21707, 0x1c59f557), WTCP(0x7cc62bdf, 0x1c8e3bbe), WTCP(0x7cba2ad3, 0x1cc27d23), WTCP(0x7cae13e4, 0x1cf6b97c), WTCP(0x7ca1e715, 0x1d2af0c1), WTCP(0x7c95a467, 0x1d5f22e7), WTCP(0x7c894bde, 0x1d934fe5), WTCP(0x7c7cdd7b, 0x1dc777b3), WTCP(0x7c705940, 0x1dfb9a48), WTCP(0x7c63bf2f, 0x1e2fb79a), WTCP(0x7c570f4b, 0x1e63cfa0), WTCP(0x7c4a4996, 0x1e97e251), WTCP(0x7c3d6e13, 0x1ecbefa4), WTCP(0x7c307cc2, 0x1efff78f), WTCP(0x7c2375a8, 0x1f33fa0a), WTCP(0x7c1658c5, 0x1f67f70b), WTCP(0x7c09261d, 0x1f9bee8a), WTCP(0x7bfbddb1, 0x1fcfe07d), WTCP(0x7bee7f85, 0x2003ccdb), WTCP(0x7be10b99, 0x2037b39b), WTCP(0x7bd381f1, 0x206b94b4), WTCP(0x7bc5e290, 0x209f701c), WTCP(0x7bb82d76, 0x20d345cc), WTCP(0x7baa62a8, 0x210715b8), WTCP(0x7b9c8226, 0x213adfda), WTCP(0x7b8e8bf5, 0x216ea426), WTCP(0x7b808015, 0x21a26295), WTCP(0x7b725e8a, 0x21d61b1e), WTCP(0x7b642756, 0x2209cdb6), WTCP(0x7b55da7c, 0x223d7a55), WTCP(0x7b4777fe, 0x227120f3), WTCP(0x7b38ffde, 0x22a4c185), WTCP(0x7b2a721f, 0x22d85c04), WTCP(0x7b1bcec4, 0x230bf065), WTCP(0x7b0d15d0, 0x233f7ea0), WTCP(0x7afe4744, 0x237306ab), WTCP(0x7aef6323, 0x23a6887f), WTCP(0x7ae06971, 0x23da0411), WTCP(0x7ad15a2f, 0x240d7958), WTCP(0x7ac23561, 0x2440e84d), WTCP(0x7ab2fb09, 0x247450e4), WTCP(0x7aa3ab29, 0x24a7b317), WTCP(0x7a9445c5, 0x24db0edb), WTCP(0x7a84cade, 0x250e6427), WTCP(0x7a753a79, 0x2541b2f3), WTCP(0x7a659496, 0x2574fb36), WTCP(0x7a55d93a, 0x25a83ce6), WTCP(0x7a460867, 0x25db77fa), WTCP(0x7a362220, 0x260eac6a), WTCP(0x7a262668, 0x2641da2d), WTCP(0x7a161540, 0x26750139), WTCP(0x7a05eead, 0x26a82186), WTCP(0x79f5b2b1, 0x26db3b0a), WTCP(0x79e5614f, 0x270e4dbd), WTCP(0x79d4fa89, 0x27415996), WTCP(0x79c47e63, 0x27745e8c), WTCP(0x79b3ece0, 0x27a75c95), WTCP(0x79a34602, 0x27da53a9), WTCP(0x799289cc, 0x280d43bf), WTCP(0x7981b841, 0x28402cce), WTCP(0x7970d165, 0x28730ecd), WTCP(0x795fd53a, 0x28a5e9b4), WTCP(0x794ec3c3, 0x28d8bd78), WTCP(0x793d9d03, 0x290b8a12), WTCP(0x792c60fe, 0x293e4f78), WTCP(0x791b0fb5, 0x29710da1), WTCP(0x7909a92d, 0x29a3c485), WTCP(0x78f82d68, 0x29d6741b), WTCP(0x78e69c69, 0x2a091c59), WTCP(0x78d4f634, 0x2a3bbd37), WTCP(0x78c33acb, 0x2a6e56ac), WTCP(0x78b16a32, 0x2aa0e8b0), WTCP(0x789f846b, 0x2ad37338), WTCP(0x788d897b, 0x2b05f63d), WTCP(0x787b7963, 0x2b3871b5), WTCP(0x78695428, 0x2b6ae598), WTCP(0x785719cc, 0x2b9d51dd), WTCP(0x7844ca53, 0x2bcfb67b), WTCP(0x783265c0, 0x2c021369), WTCP(0x781fec15, 0x2c34689e), WTCP(0x780d5d57, 0x2c66b611), WTCP(0x77fab989, 0x2c98fbba), WTCP(0x77e800ad, 0x2ccb3990), WTCP(0x77d532c7, 0x2cfd6f8a), WTCP(0x77c24fdb, 0x2d2f9d9f), WTCP(0x77af57eb, 0x2d61c3c7), WTCP(0x779c4afc, 0x2d93e1f8), WTCP(0x77892910, 0x2dc5f829), WTCP(0x7775f22a, 0x2df80653), WTCP(0x7762a64f, 0x2e2a0c6c), WTCP(0x774f4581, 0x2e5c0a6b), WTCP(0x773bcfc4, 0x2e8e0048), WTCP(0x7728451c, 0x2ebfedfa), WTCP(0x7714a58b, 0x2ef1d377), WTCP(0x7700f115, 0x2f23b0b9), WTCP(0x76ed27be, 0x2f5585b5), WTCP(0x76d94989, 0x2f875262), WTCP(0x76c55679, 0x2fb916b9), WTCP(0x76b14e93, 0x2fead2b0), WTCP(0x769d31d9, 0x301c863f), WTCP(0x76890050, 0x304e315d), WTCP(0x7674b9fa, 0x307fd401), WTCP(0x76605edb, 0x30b16e23), WTCP(0x764beef8, 0x30e2ffb9), WTCP(0x76376a52, 0x311488bc), WTCP(0x7622d0ef, 0x31460922), WTCP(0x760e22d1, 0x317780e2), WTCP(0x75f95ffc, 0x31a8eff5), WTCP(0x75e48874, 0x31da5651), WTCP(0x75cf9c3d, 0x320bb3ee), WTCP(0x75ba9b5a, 0x323d08c3), WTCP(0x75a585cf, 0x326e54c7), WTCP(0x75905ba0, 0x329f97f3), WTCP(0x757b1ccf, 0x32d0d23c), WTCP(0x7565c962, 0x3302039b), WTCP(0x7550615c, 0x33332c06), WTCP(0x753ae4c0, 0x33644b76), WTCP(0x75255392, 0x339561e1), WTCP(0x750fadd7, 0x33c66f40), WTCP(0x74f9f391, 0x33f77388), WTCP(0x74e424c5, 0x34286eb3), WTCP(0x74ce4177, 0x345960b7), WTCP(0x74b849aa, 0x348a498b), WTCP(0x74a23d62, 0x34bb2927), WTCP(0x748c1ca4, 0x34ebff83), WTCP(0x7475e772, 0x351ccc96), WTCP(0x745f9dd1, 0x354d9057), WTCP(0x74493fc5, 0x357e4abe), WTCP(0x7432cd51, 0x35aefbc2), WTCP(0x741c467b, 0x35dfa35a), WTCP(0x7405ab45, 0x3610417f), WTCP(0x73eefbb3, 0x3640d627), WTCP(0x73d837ca, 0x3671614b), WTCP(0x73c15f8d, 0x36a1e2e0), WTCP(0x73aa7301, 0x36d25ae0), WTCP(0x7393722a, 0x3702c942), WTCP(0x737c5d0b, 0x37332dfd), WTCP(0x736533a9, 0x37638908), WTCP(0x734df607, 0x3793da5b), WTCP(0x7336a42b, 0x37c421ee), WTCP(0x731f3e17, 0x37f45fb7), WTCP(0x7307c3d0, 0x382493b0), WTCP(0x72f0355a, 0x3854bdcf), WTCP(0x72d892ba, 0x3884de0b), WTCP(0x72c0dbf3, 0x38b4f45d), WTCP(0x72a91109, 0x38e500bc), WTCP(0x72913201, 0x3915031f), WTCP(0x72793edf, 0x3944fb7e), WTCP(0x726137a8, 0x3974e9d0), WTCP(0x72491c5e, 0x39a4ce0e), WTCP(0x7230ed07, 0x39d4a82f), WTCP(0x7218a9a7, 0x3a04782a), WTCP(0x72005242, 0x3a343df7), WTCP(0x71e7e6dc, 0x3a63f98d), WTCP(0x71cf677a, 0x3a93aae5), WTCP(0x71b6d420, 0x3ac351f6), WTCP(0x719e2cd2, 0x3af2eeb7), WTCP(0x71857195, 0x3b228120), WTCP(0x716ca26c, 0x3b52092a), WTCP(0x7153bf5d, 0x3b8186ca), WTCP(0x713ac86b, 0x3bb0f9fa), WTCP(0x7121bd9c, 0x3be062b0), WTCP(0x71089ef2, 0x3c0fc0e6), WTCP(0x70ef6c74, 0x3c3f1491), WTCP(0x70d62625, 0x3c6e5daa), WTCP(0x70bccc09, 0x3c9d9c28), WTCP(0x70a35e25, 0x3cccd004), WTCP(0x7089dc7e, 0x3cfbf935), WTCP(0x70704718, 0x3d2b17b3), WTCP(0x70569df8, 0x3d5a2b75), WTCP(0x703ce122, 0x3d893474), WTCP(0x7023109a, 0x3db832a6), WTCP(0x70092c65, 0x3de72604), WTCP(0x6fef3488, 0x3e160e85), WTCP(0x6fd52907, 0x3e44ec22), WTCP(0x6fbb09e7, 0x3e73bed2), WTCP(0x6fa0d72c, 0x3ea2868c), WTCP(0x6f8690db, 0x3ed14349), WTCP(0x6f6c36f8, 0x3efff501), WTCP(0x6f51c989, 0x3f2e9bab), WTCP(0x6f374891, 0x3f5d373e), WTCP(0x6f1cb416, 0x3f8bc7b4), WTCP(0x6f020c1c, 0x3fba4d03), WTCP(0x6ee750a8, 0x3fe8c724), WTCP(0x6ecc81be, 0x4017360e), WTCP(0x6eb19f64, 0x404599b9), WTCP(0x6e96a99d, 0x4073f21d), WTCP(0x6e7ba06f, 0x40a23f32), WTCP(0x6e6083de, 0x40d080f0), WTCP(0x6e4553ef, 0x40feb74f), WTCP(0x6e2a10a8, 0x412ce246), WTCP(0x6e0eba0c, 0x415b01ce), WTCP(0x6df35020, 0x418915de), WTCP(0x6dd7d2ea, 0x41b71e6f), WTCP(0x6dbc426e, 0x41e51b77), WTCP(0x6da09eb1, 0x42130cf0), WTCP(0x6d84e7b7, 0x4240f2d1), WTCP(0x6d691d87, 0x426ecd12), WTCP(0x6d4d4023, 0x429c9bab), WTCP(0x6d314f93, 0x42ca5e94), WTCP(0x6d154bd9, 0x42f815c5), WTCP(0x6cf934fc, 0x4325c135), WTCP(0x6cdd0b00, 0x435360de), WTCP(0x6cc0cdea, 0x4380f4b7), WTCP(0x6ca47dbf, 0x43ae7cb7), WTCP(0x6c881a84, 0x43dbf8d7), WTCP(0x6c6ba43e, 0x44096910), WTCP(0x6c4f1af2, 0x4436cd58), WTCP(0x6c327ea6, 0x446425a8), WTCP(0x6c15cf5d, 0x449171f8), WTCP(0x6bf90d1d, 0x44beb240), WTCP(0x6bdc37eb, 0x44ebe679), WTCP(0x6bbf4fcd, 0x45190e99), WTCP(0x6ba254c7, 0x45462a9a), WTCP(0x6b8546de, 0x45733a73), WTCP(0x6b682617, 0x45a03e1d), WTCP(0x6b4af279, 0x45cd358f), WTCP(0x6b2dac06, 0x45fa20c2), WTCP(0x6b1052c6, 0x4626ffae), WTCP(0x6af2e6bc, 0x4653d24b), WTCP(0x6ad567ef, 0x46809891), WTCP(0x6ab7d663, 0x46ad5278), WTCP(0x6a9a321d, 0x46d9fff8), WTCP(0x6a7c7b23, 0x4706a10a), WTCP(0x6a5eb17a, 0x473335a5), WTCP(0x6a40d527, 0x475fbdc3), WTCP(0x6a22e630, 0x478c395a), WTCP(0x6a04e499, 0x47b8a864), WTCP(0x69e6d067, 0x47e50ad8), WTCP(0x69c8a9a1, 0x481160ae), WTCP(0x69aa704c, 0x483da9e0), WTCP(0x698c246c, 0x4869e665), WTCP(0x696dc607, 0x48961635), WTCP(0x694f5523, 0x48c23949), WTCP(0x6930d1c4, 0x48ee4f98), WTCP(0x69123bf1, 0x491a591c), WTCP(0x68f393ae, 0x494655cc), WTCP(0x68d4d900, 0x497245a1), WTCP(0x68b60bee, 0x499e2892), WTCP(0x68972c7d, 0x49c9fe99), WTCP(0x68783ab1, 0x49f5c7ae), WTCP(0x68593691, 0x4a2183c8), WTCP(0x683a2022, 0x4a4d32e1), WTCP(0x681af76a, 0x4a78d4f0), WTCP(0x67fbbc6d, 0x4aa469ee), WTCP(0x67dc6f31, 0x4acff1d3), WTCP(0x67bd0fbd, 0x4afb6c98), WTCP(0x679d9e14, 0x4b26da35), WTCP(0x677e1a3e, 0x4b523aa2), WTCP(0x675e843e, 0x4b7d8dd8), WTCP(0x673edc1c, 0x4ba8d3cf), WTCP(0x671f21dc, 0x4bd40c80), WTCP(0x66ff5584, 0x4bff37e2), WTCP(0x66df771a, 0x4c2a55ef), WTCP(0x66bf86a3, 0x4c55669f), WTCP(0x669f8425, 0x4c8069ea), WTCP(0x667f6fa5, 0x4cab5fc9), WTCP(0x665f4929, 0x4cd64834), WTCP(0x663f10b7, 0x4d012324), WTCP(0x661ec654, 0x4d2bf091), WTCP(0x65fe6a06, 0x4d56b073), WTCP(0x65ddfbd3, 0x4d8162c4), WTCP(0x65bd7bc0, 0x4dac077b), WTCP(0x659ce9d4, 0x4dd69e92), WTCP(0x657c4613, 0x4e012800), WTCP(0x655b9083, 0x4e2ba3be), WTCP(0x653ac92b, 0x4e5611c5), WTCP(0x6519f010, 0x4e80720e), WTCP(0x64f90538, 0x4eaac490), WTCP(0x64d808a8, 0x4ed50945), WTCP(0x64b6fa66, 0x4eff4025), WTCP(0x6495da79, 0x4f296928), WTCP(0x6474a8e5, 0x4f538448), WTCP(0x645365b2, 0x4f7d917c), WTCP(0x643210e4, 0x4fa790be), WTCP(0x6410aa81, 0x4fd18206), WTCP(0x63ef3290, 0x4ffb654d), WTCP(0x63cda916, 0x50253a8b), WTCP(0x63ac0e19, 0x504f01ba), WTCP(0x638a619e, 0x5078bad1), WTCP(0x6368a3ad, 0x50a265c9), WTCP(0x6346d44b, 0x50cc029c), WTCP(0x6324f37d, 0x50f59141), WTCP(0x6303014a, 0x511f11b2), WTCP(0x62e0fdb8, 0x514883e7), WTCP(0x62bee8cc, 0x5171e7d9), WTCP(0x629cc28c, 0x519b3d80), WTCP(0x627a8b00, 0x51c484d6), WTCP(0x6258422c, 0x51edbdd4), WTCP(0x6235e816, 0x5216e871), WTCP(0x62137cc5, 0x524004a7), WTCP(0x61f1003f, 0x5269126e), WTCP(0x61ce7289, 0x529211c0), WTCP(0x61abd3ab, 0x52bb0295), WTCP(0x618923a9, 0x52e3e4e6), WTCP(0x61666289, 0x530cb8ac), WTCP(0x61439053, 0x53357ddf), WTCP(0x6120ad0d, 0x535e3479), WTCP(0x60fdb8bb, 0x5386dc72), WTCP(0x60dab365, 0x53af75c3), WTCP(0x60b79d10, 0x53d80065), WTCP(0x609475c3, 0x54007c51), WTCP(0x60713d84, 0x5428e980), WTCP(0x604df459, 0x545147eb), WTCP(0x602a9a48, 0x5479978a), WTCP(0x60072f57, 0x54a1d857), WTCP(0x5fe3b38d, 0x54ca0a4b), WTCP(0x5fc026f0, 0x54f22d5d), WTCP(0x5f9c8987, 0x551a4189), WTCP(0x5f78db56, 0x554246c6), WTCP(0x5f551c65, 0x556a3d0d), WTCP(0x5f314cba, 0x55922457), WTCP(0x5f0d6c5b, 0x55b9fc9e), WTCP(0x5ee97b4f, 0x55e1c5da), WTCP(0x5ec5799b, 0x56098005), WTCP(0x5ea16747, 0x56312b17), WTCP(0x5e7d4458, 0x5658c709), WTCP(0x5e5910d4, 0x568053d5), WTCP(0x5e34ccc3, 0x56a7d174), WTCP(0x5e10782b, 0x56cf3fde), WTCP(0x5dec1311, 0x56f69f0d), WTCP(0x5dc79d7c, 0x571deefa), WTCP(0x5da31773, 0x57452f9d), WTCP(0x5d7e80fc, 0x576c60f1), WTCP(0x5d59da1e, 0x579382ee), WTCP(0x5d3522de, 0x57ba958d), WTCP(0x5d105b44, 0x57e198c7), WTCP(0x5ceb8355, 0x58088c96), WTCP(0x5cc69b19, 0x582f70f3), WTCP(0x5ca1a295, 0x585645d7), WTCP(0x5c7c99d1, 0x587d0b3b), WTCP(0x5c5780d3, 0x58a3c118), WTCP(0x5c3257a0, 0x58ca6767), WTCP(0x5c0d1e41, 0x58f0fe23), WTCP(0x5be7d4ba, 0x59178543), WTCP(0x5bc27b14, 0x593dfcc2), WTCP(0x5b9d1154, 0x59646498), WTCP(0x5b779780, 0x598abcbe), WTCP(0x5b520da1, 0x59b1052f), WTCP(0x5b2c73bb, 0x59d73de3), WTCP(0x5b06c9d6, 0x59fd66d4), WTCP(0x5ae10ff9, 0x5a237ffa), WTCP(0x5abb4629, 0x5a498950), WTCP(0x5a956c6e, 0x5a6f82ce), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_WTP SineWindow1024[] = { WTCP(0x7ffffd88, 0x001921fb), WTCP(0x7fffe9cb, 0x004b65ee), WTCP(0x7fffc251, 0x007da9d4), WTCP(0x7fff8719, 0x00afeda8), WTCP(0x7fff3824, 0x00e23160), WTCP(0x7ffed572, 0x011474f6), WTCP(0x7ffe5f03, 0x0146b860), WTCP(0x7ffdd4d7, 0x0178fb99), WTCP(0x7ffd36ee, 0x01ab3e97), WTCP(0x7ffc8549, 0x01dd8154), WTCP(0x7ffbbfe6, 0x020fc3c6), WTCP(0x7ffae6c7, 0x024205e8), WTCP(0x7ff9f9ec, 0x027447b0), WTCP(0x7ff8f954, 0x02a68917), WTCP(0x7ff7e500, 0x02d8ca16), WTCP(0x7ff6bcf0, 0x030b0aa4), WTCP(0x7ff58125, 0x033d4abb), WTCP(0x7ff4319d, 0x036f8a51), WTCP(0x7ff2ce5b, 0x03a1c960), WTCP(0x7ff1575d, 0x03d407df), WTCP(0x7fefcca4, 0x040645c7), WTCP(0x7fee2e30, 0x04388310), WTCP(0x7fec7c02, 0x046abfb3), WTCP(0x7feab61a, 0x049cfba7), WTCP(0x7fe8dc78, 0x04cf36e5), WTCP(0x7fe6ef1c, 0x05017165), WTCP(0x7fe4ee06, 0x0533ab20), WTCP(0x7fe2d938, 0x0565e40d), WTCP(0x7fe0b0b1, 0x05981c26), WTCP(0x7fde7471, 0x05ca5361), WTCP(0x7fdc247a, 0x05fc89b8), WTCP(0x7fd9c0ca, 0x062ebf22), WTCP(0x7fd74964, 0x0660f398), WTCP(0x7fd4be46, 0x06932713), WTCP(0x7fd21f72, 0x06c5598a), WTCP(0x7fcf6ce8, 0x06f78af6), WTCP(0x7fcca6a7, 0x0729bb4e), WTCP(0x7fc9ccb2, 0x075bea8c), WTCP(0x7fc6df08, 0x078e18a7), WTCP(0x7fc3dda9, 0x07c04598), WTCP(0x7fc0c896, 0x07f27157), WTCP(0x7fbd9fd0, 0x08249bdd), WTCP(0x7fba6357, 0x0856c520), WTCP(0x7fb7132b, 0x0888ed1b), WTCP(0x7fb3af4e, 0x08bb13c5), WTCP(0x7fb037bf, 0x08ed3916), WTCP(0x7facac7f, 0x091f5d06), WTCP(0x7fa90d8e, 0x09517f8f), WTCP(0x7fa55aee, 0x0983a0a7), WTCP(0x7fa1949e, 0x09b5c048), WTCP(0x7f9dbaa0, 0x09e7de6a), WTCP(0x7f99ccf4, 0x0a19fb04), WTCP(0x7f95cb9a, 0x0a4c1610), WTCP(0x7f91b694, 0x0a7e2f85), WTCP(0x7f8d8de1, 0x0ab0475c), WTCP(0x7f895182, 0x0ae25d8d), WTCP(0x7f850179, 0x0b147211), WTCP(0x7f809dc5, 0x0b4684df), WTCP(0x7f7c2668, 0x0b7895f0), WTCP(0x7f779b62, 0x0baaa53b), WTCP(0x7f72fcb4, 0x0bdcb2bb), WTCP(0x7f6e4a5e, 0x0c0ebe66), WTCP(0x7f698461, 0x0c40c835), WTCP(0x7f64aabf, 0x0c72d020), WTCP(0x7f5fbd77, 0x0ca4d620), WTCP(0x7f5abc8a, 0x0cd6da2d), WTCP(0x7f55a7fa, 0x0d08dc3f), WTCP(0x7f507fc7, 0x0d3adc4e), WTCP(0x7f4b43f2, 0x0d6cda53), WTCP(0x7f45f47b, 0x0d9ed646), WTCP(0x7f409164, 0x0dd0d01f), WTCP(0x7f3b1aad, 0x0e02c7d7), WTCP(0x7f359057, 0x0e34bd66), WTCP(0x7f2ff263, 0x0e66b0c3), WTCP(0x7f2a40d2, 0x0e98a1e9), WTCP(0x7f247ba5, 0x0eca90ce), WTCP(0x7f1ea2dc, 0x0efc7d6b), WTCP(0x7f18b679, 0x0f2e67b8), WTCP(0x7f12b67c, 0x0f604faf), WTCP(0x7f0ca2e7, 0x0f923546), WTCP(0x7f067bba, 0x0fc41876), WTCP(0x7f0040f6, 0x0ff5f938), WTCP(0x7ef9f29d, 0x1027d784), WTCP(0x7ef390ae, 0x1059b352), WTCP(0x7eed1b2c, 0x108b8c9b), WTCP(0x7ee69217, 0x10bd6356), WTCP(0x7edff570, 0x10ef377d), WTCP(0x7ed94538, 0x11210907), WTCP(0x7ed28171, 0x1152d7ed), WTCP(0x7ecbaa1a, 0x1184a427), WTCP(0x7ec4bf36, 0x11b66dad), WTCP(0x7ebdc0c6, 0x11e83478), WTCP(0x7eb6aeca, 0x1219f880), WTCP(0x7eaf8943, 0x124bb9be), WTCP(0x7ea85033, 0x127d7829), WTCP(0x7ea1039b, 0x12af33ba), WTCP(0x7e99a37c, 0x12e0ec6a), WTCP(0x7e922fd6, 0x1312a230), WTCP(0x7e8aa8ac, 0x13445505), WTCP(0x7e830dff, 0x137604e2), WTCP(0x7e7b5fce, 0x13a7b1bf), WTCP(0x7e739e1d, 0x13d95b93), WTCP(0x7e6bc8eb, 0x140b0258), WTCP(0x7e63e03b, 0x143ca605), WTCP(0x7e5be40c, 0x146e4694), WTCP(0x7e53d462, 0x149fe3fc), WTCP(0x7e4bb13c, 0x14d17e36), WTCP(0x7e437a9c, 0x1503153a), WTCP(0x7e3b3083, 0x1534a901), WTCP(0x7e32d2f4, 0x15663982), WTCP(0x7e2a61ed, 0x1597c6b7), WTCP(0x7e21dd73, 0x15c95097), WTCP(0x7e194584, 0x15fad71b), WTCP(0x7e109a24, 0x162c5a3b), WTCP(0x7e07db52, 0x165dd9f0), WTCP(0x7dff0911, 0x168f5632), WTCP(0x7df62362, 0x16c0cef9), WTCP(0x7ded2a47, 0x16f2443e), WTCP(0x7de41dc0, 0x1723b5f9), WTCP(0x7ddafdce, 0x17552422), WTCP(0x7dd1ca75, 0x17868eb3), WTCP(0x7dc883b4, 0x17b7f5a3), WTCP(0x7dbf298d, 0x17e958ea), WTCP(0x7db5bc02, 0x181ab881), WTCP(0x7dac3b15, 0x184c1461), WTCP(0x7da2a6c6, 0x187d6c82), WTCP(0x7d98ff17, 0x18aec0db), WTCP(0x7d8f4409, 0x18e01167), WTCP(0x7d85759f, 0x19115e1c), WTCP(0x7d7b93da, 0x1942a6f3), WTCP(0x7d719eba, 0x1973ebe6), WTCP(0x7d679642, 0x19a52ceb), WTCP(0x7d5d7a74, 0x19d669fc), WTCP(0x7d534b50, 0x1a07a311), WTCP(0x7d4908d9, 0x1a38d823), WTCP(0x7d3eb30f, 0x1a6a0929), WTCP(0x7d3449f5, 0x1a9b361d), WTCP(0x7d29cd8c, 0x1acc5ef6), WTCP(0x7d1f3dd6, 0x1afd83ad), WTCP(0x7d149ad5, 0x1b2ea43a), WTCP(0x7d09e489, 0x1b5fc097), WTCP(0x7cff1af5, 0x1b90d8bb), WTCP(0x7cf43e1a, 0x1bc1ec9e), WTCP(0x7ce94dfb, 0x1bf2fc3a), WTCP(0x7cde4a98, 0x1c240786), WTCP(0x7cd333f3, 0x1c550e7c), WTCP(0x7cc80a0f, 0x1c861113), WTCP(0x7cbcccec, 0x1cb70f43), WTCP(0x7cb17c8d, 0x1ce80906), WTCP(0x7ca618f3, 0x1d18fe54), WTCP(0x7c9aa221, 0x1d49ef26), WTCP(0x7c8f1817, 0x1d7adb73), WTCP(0x7c837ad8, 0x1dabc334), WTCP(0x7c77ca65, 0x1ddca662), WTCP(0x7c6c06c0, 0x1e0d84f5), WTCP(0x7c602fec, 0x1e3e5ee5), WTCP(0x7c5445e9, 0x1e6f342c), WTCP(0x7c4848ba, 0x1ea004c1), WTCP(0x7c3c3860, 0x1ed0d09d), WTCP(0x7c3014de, 0x1f0197b8), WTCP(0x7c23de35, 0x1f325a0b), WTCP(0x7c179467, 0x1f63178f), WTCP(0x7c0b3777, 0x1f93d03c), WTCP(0x7bfec765, 0x1fc4840a), WTCP(0x7bf24434, 0x1ff532f2), WTCP(0x7be5ade6, 0x2025dcec), WTCP(0x7bd9047c, 0x205681f1), WTCP(0x7bcc47fa, 0x208721f9), WTCP(0x7bbf7860, 0x20b7bcfe), WTCP(0x7bb295b0, 0x20e852f6), WTCP(0x7ba59fee, 0x2118e3dc), WTCP(0x7b989719, 0x21496fa7), WTCP(0x7b8b7b36, 0x2179f64f), WTCP(0x7b7e4c45, 0x21aa77cf), WTCP(0x7b710a49, 0x21daf41d), WTCP(0x7b63b543, 0x220b6b32), WTCP(0x7b564d36, 0x223bdd08), WTCP(0x7b48d225, 0x226c4996), WTCP(0x7b3b4410, 0x229cb0d5), WTCP(0x7b2da2fa, 0x22cd12bd), WTCP(0x7b1feee5, 0x22fd6f48), WTCP(0x7b1227d3, 0x232dc66d), WTCP(0x7b044dc7, 0x235e1826), WTCP(0x7af660c2, 0x238e646a), WTCP(0x7ae860c7, 0x23beab33), WTCP(0x7ada4dd8, 0x23eeec78), WTCP(0x7acc27f7, 0x241f2833), WTCP(0x7abdef25, 0x244f5e5c), WTCP(0x7aafa367, 0x247f8eec), WTCP(0x7aa144bc, 0x24afb9da), WTCP(0x7a92d329, 0x24dfdf20), WTCP(0x7a844eae, 0x250ffeb7), WTCP(0x7a75b74f, 0x25401896), WTCP(0x7a670d0d, 0x25702cb7), WTCP(0x7a584feb, 0x25a03b11), WTCP(0x7a497feb, 0x25d0439f), WTCP(0x7a3a9d0f, 0x26004657), WTCP(0x7a2ba75a, 0x26304333), WTCP(0x7a1c9ece, 0x26603a2c), WTCP(0x7a0d836d, 0x26902b39), WTCP(0x79fe5539, 0x26c01655), WTCP(0x79ef1436, 0x26effb76), WTCP(0x79dfc064, 0x271fda96), WTCP(0x79d059c8, 0x274fb3ae), WTCP(0x79c0e062, 0x277f86b5), WTCP(0x79b15435, 0x27af53a6), WTCP(0x79a1b545, 0x27df1a77), WTCP(0x79920392, 0x280edb23), WTCP(0x79823f20, 0x283e95a1), WTCP(0x797267f2, 0x286e49ea), WTCP(0x79627e08, 0x289df7f8), WTCP(0x79528167, 0x28cd9fc1), WTCP(0x79427210, 0x28fd4140), WTCP(0x79325006, 0x292cdc6d), WTCP(0x79221b4b, 0x295c7140), WTCP(0x7911d3e2, 0x298bffb2), WTCP(0x790179cd, 0x29bb87bc), WTCP(0x78f10d0f, 0x29eb0957), WTCP(0x78e08dab, 0x2a1a847b), WTCP(0x78cffba3, 0x2a49f920), WTCP(0x78bf56f9, 0x2a796740), WTCP(0x78ae9fb0, 0x2aa8ced3), WTCP(0x789dd5cb, 0x2ad82fd2), WTCP(0x788cf94c, 0x2b078a36), WTCP(0x787c0a36, 0x2b36ddf7), WTCP(0x786b088c, 0x2b662b0e), WTCP(0x7859f44f, 0x2b957173), WTCP(0x7848cd83, 0x2bc4b120), WTCP(0x7837942b, 0x2bf3ea0d), WTCP(0x78264849, 0x2c231c33), WTCP(0x7814e9df, 0x2c52478a), WTCP(0x780378f1, 0x2c816c0c), WTCP(0x77f1f581, 0x2cb089b1), WTCP(0x77e05f91, 0x2cdfa071), WTCP(0x77ceb725, 0x2d0eb046), WTCP(0x77bcfc3f, 0x2d3db928), WTCP(0x77ab2ee2, 0x2d6cbb10), WTCP(0x77994f11, 0x2d9bb5f6), WTCP(0x77875cce, 0x2dcaa9d5), WTCP(0x7775581d, 0x2df996a3), WTCP(0x776340ff, 0x2e287c5a), WTCP(0x77511778, 0x2e575af3), WTCP(0x773edb8b, 0x2e863267), WTCP(0x772c8d3a, 0x2eb502ae), WTCP(0x771a2c88, 0x2ee3cbc1), WTCP(0x7707b979, 0x2f128d99), WTCP(0x76f5340e, 0x2f41482e), WTCP(0x76e29c4b, 0x2f6ffb7a), WTCP(0x76cff232, 0x2f9ea775), WTCP(0x76bd35c7, 0x2fcd4c19), WTCP(0x76aa670d, 0x2ffbe95d), WTCP(0x76978605, 0x302a7f3a), WTCP(0x768492b4, 0x30590dab), WTCP(0x76718d1c, 0x308794a6), WTCP(0x765e7540, 0x30b61426), WTCP(0x764b4b23, 0x30e48c22), WTCP(0x76380ec8, 0x3112fc95), WTCP(0x7624c031, 0x31416576), WTCP(0x76115f63, 0x316fc6be), WTCP(0x75fdec60, 0x319e2067), WTCP(0x75ea672a, 0x31cc7269), WTCP(0x75d6cfc5, 0x31fabcbd), WTCP(0x75c32634, 0x3228ff5c), WTCP(0x75af6a7b, 0x32573a3f), WTCP(0x759b9c9b, 0x32856d5e), WTCP(0x7587bc98, 0x32b398b3), WTCP(0x7573ca75, 0x32e1bc36), WTCP(0x755fc635, 0x330fd7e1), WTCP(0x754bafdc, 0x333debab), WTCP(0x7537876c, 0x336bf78f), WTCP(0x75234ce8, 0x3399fb85), WTCP(0x750f0054, 0x33c7f785), WTCP(0x74faa1b3, 0x33f5eb89), WTCP(0x74e63108, 0x3423d78a), WTCP(0x74d1ae55, 0x3451bb81), WTCP(0x74bd199f, 0x347f9766), WTCP(0x74a872e8, 0x34ad6b32), WTCP(0x7493ba34, 0x34db36df), WTCP(0x747eef85, 0x3508fa66), WTCP(0x746a12df, 0x3536b5be), WTCP(0x74552446, 0x356468e2), WTCP(0x744023bc, 0x359213c9), WTCP(0x742b1144, 0x35bfb66e), WTCP(0x7415ece2, 0x35ed50c9), WTCP(0x7400b69a, 0x361ae2d3), WTCP(0x73eb6e6e, 0x36486c86), WTCP(0x73d61461, 0x3675edd9), WTCP(0x73c0a878, 0x36a366c6), WTCP(0x73ab2ab4, 0x36d0d746), WTCP(0x73959b1b, 0x36fe3f52), WTCP(0x737ff9ae, 0x372b9ee3), WTCP(0x736a4671, 0x3758f5f2), WTCP(0x73548168, 0x37864477), WTCP(0x733eaa96, 0x37b38a6d), WTCP(0x7328c1ff, 0x37e0c7cc), WTCP(0x7312c7a5, 0x380dfc8d), WTCP(0x72fcbb8c, 0x383b28a9), WTCP(0x72e69db7, 0x38684c19), WTCP(0x72d06e2b, 0x389566d6), WTCP(0x72ba2cea, 0x38c278d9), WTCP(0x72a3d9f7, 0x38ef821c), WTCP(0x728d7557, 0x391c8297), WTCP(0x7276ff0d, 0x39497a43), WTCP(0x7260771b, 0x39766919), WTCP(0x7249dd86, 0x39a34f13), WTCP(0x72333251, 0x39d02c2a), WTCP(0x721c7580, 0x39fd0056), WTCP(0x7205a716, 0x3a29cb91), WTCP(0x71eec716, 0x3a568dd4), WTCP(0x71d7d585, 0x3a834717), WTCP(0x71c0d265, 0x3aaff755), WTCP(0x71a9bdba, 0x3adc9e86), WTCP(0x71929789, 0x3b093ca3), WTCP(0x717b5fd3, 0x3b35d1a5), WTCP(0x7164169d, 0x3b625d86), WTCP(0x714cbbeb, 0x3b8ee03e), WTCP(0x71354fc0, 0x3bbb59c7), WTCP(0x711dd220, 0x3be7ca1a), WTCP(0x7106430e, 0x3c143130), WTCP(0x70eea28e, 0x3c408f03), WTCP(0x70d6f0a4, 0x3c6ce38a), WTCP(0x70bf2d53, 0x3c992ec0), WTCP(0x70a7589f, 0x3cc5709e), WTCP(0x708f728b, 0x3cf1a91c), WTCP(0x70777b1c, 0x3d1dd835), WTCP(0x705f7255, 0x3d49fde1), WTCP(0x70475839, 0x3d761a19), WTCP(0x702f2ccd, 0x3da22cd7), WTCP(0x7016f014, 0x3dce3614), WTCP(0x6ffea212, 0x3dfa35c8), WTCP(0x6fe642ca, 0x3e262bee), WTCP(0x6fcdd241, 0x3e52187f), WTCP(0x6fb5507a, 0x3e7dfb73), WTCP(0x6f9cbd79, 0x3ea9d4c3), WTCP(0x6f841942, 0x3ed5a46b), WTCP(0x6f6b63d8, 0x3f016a61), WTCP(0x6f529d40, 0x3f2d26a0), WTCP(0x6f39c57d, 0x3f58d921), WTCP(0x6f20dc92, 0x3f8481dd), WTCP(0x6f07e285, 0x3fb020ce), WTCP(0x6eeed758, 0x3fdbb5ec), WTCP(0x6ed5bb10, 0x40074132), WTCP(0x6ebc8db0, 0x4032c297), WTCP(0x6ea34f3d, 0x405e3a16), WTCP(0x6e89ffb9, 0x4089a7a8), WTCP(0x6e709f2a, 0x40b50b46), WTCP(0x6e572d93, 0x40e064ea), WTCP(0x6e3daaf8, 0x410bb48c), WTCP(0x6e24175c, 0x4136fa27), WTCP(0x6e0a72c5, 0x416235b2), WTCP(0x6df0bd35, 0x418d6729), WTCP(0x6dd6f6b1, 0x41b88e84), WTCP(0x6dbd1f3c, 0x41e3abbc), WTCP(0x6da336dc, 0x420ebecb), WTCP(0x6d893d93, 0x4239c7aa), WTCP(0x6d6f3365, 0x4264c653), WTCP(0x6d551858, 0x428fbabe), WTCP(0x6d3aec6e, 0x42baa4e6), WTCP(0x6d20afac, 0x42e584c3), WTCP(0x6d066215, 0x43105a50), WTCP(0x6cec03af, 0x433b2585), WTCP(0x6cd1947c, 0x4365e65b), WTCP(0x6cb71482, 0x43909ccd), WTCP(0x6c9c83c3, 0x43bb48d4), WTCP(0x6c81e245, 0x43e5ea68), WTCP(0x6c67300b, 0x44108184), WTCP(0x6c4c6d1a, 0x443b0e21), WTCP(0x6c319975, 0x44659039), WTCP(0x6c16b521, 0x449007c4), WTCP(0x6bfbc021, 0x44ba74bd), WTCP(0x6be0ba7b, 0x44e4d71c), WTCP(0x6bc5a431, 0x450f2edb), WTCP(0x6baa7d49, 0x45397bf4), WTCP(0x6b8f45c7, 0x4563be60), WTCP(0x6b73fdae, 0x458df619), WTCP(0x6b58a503, 0x45b82318), WTCP(0x6b3d3bcb, 0x45e24556), WTCP(0x6b21c208, 0x460c5cce), WTCP(0x6b0637c1, 0x46366978), WTCP(0x6aea9cf8, 0x46606b4e), WTCP(0x6acef1b2, 0x468a624a), WTCP(0x6ab335f4, 0x46b44e65), WTCP(0x6a9769c1, 0x46de2f99), WTCP(0x6a7b8d1e, 0x470805df), WTCP(0x6a5fa010, 0x4731d131), WTCP(0x6a43a29a, 0x475b9188), WTCP(0x6a2794c1, 0x478546de), WTCP(0x6a0b7689, 0x47aef12c), WTCP(0x69ef47f6, 0x47d8906d), WTCP(0x69d3090e, 0x48022499), WTCP(0x69b6b9d3, 0x482badab), WTCP(0x699a5a4c, 0x48552b9b), WTCP(0x697dea7b, 0x487e9e64), WTCP(0x69616a65, 0x48a805ff), WTCP(0x6944da10, 0x48d16265), WTCP(0x6928397e, 0x48fab391), WTCP(0x690b88b5, 0x4923f97b), WTCP(0x68eec7b9, 0x494d341e), WTCP(0x68d1f68f, 0x49766373), WTCP(0x68b5153a, 0x499f8774), WTCP(0x689823bf, 0x49c8a01b), WTCP(0x687b2224, 0x49f1ad61), WTCP(0x685e106c, 0x4a1aaf3f), WTCP(0x6840ee9b, 0x4a43a5b0), WTCP(0x6823bcb7, 0x4a6c90ad), WTCP(0x68067ac3, 0x4a957030), WTCP(0x67e928c5, 0x4abe4433), WTCP(0x67cbc6c0, 0x4ae70caf), WTCP(0x67ae54ba, 0x4b0fc99d), WTCP(0x6790d2b6, 0x4b387af9), WTCP(0x677340ba, 0x4b6120bb), WTCP(0x67559eca, 0x4b89badd), WTCP(0x6737ecea, 0x4bb24958), WTCP(0x671a2b20, 0x4bdacc28), WTCP(0x66fc596f, 0x4c034345), WTCP(0x66de77dc, 0x4c2baea9), WTCP(0x66c0866d, 0x4c540e4e), WTCP(0x66a28524, 0x4c7c622d), WTCP(0x66847408, 0x4ca4aa41), WTCP(0x6666531d, 0x4ccce684), WTCP(0x66482267, 0x4cf516ee), WTCP(0x6629e1ec, 0x4d1d3b7a), WTCP(0x660b91af, 0x4d455422), WTCP(0x65ed31b5, 0x4d6d60df), WTCP(0x65cec204, 0x4d9561ac), WTCP(0x65b0429f, 0x4dbd5682), WTCP(0x6591b38c, 0x4de53f5a), WTCP(0x657314cf, 0x4e0d1c30), WTCP(0x6554666d, 0x4e34ecfc), WTCP(0x6535a86b, 0x4e5cb1b9), WTCP(0x6516dacd, 0x4e846a60), WTCP(0x64f7fd98, 0x4eac16eb), WTCP(0x64d910d1, 0x4ed3b755), WTCP(0x64ba147d, 0x4efb4b96), WTCP(0x649b08a0, 0x4f22d3aa), WTCP(0x647bed3f, 0x4f4a4f89), WTCP(0x645cc260, 0x4f71bf2e), WTCP(0x643d8806, 0x4f992293), WTCP(0x641e3e38, 0x4fc079b1), WTCP(0x63fee4f8, 0x4fe7c483), WTCP(0x63df7c4d, 0x500f0302), WTCP(0x63c0043b, 0x50363529), WTCP(0x63a07cc7, 0x505d5af1), WTCP(0x6380e5f6, 0x50847454), WTCP(0x63613fcd, 0x50ab814d), WTCP(0x63418a50, 0x50d281d5), WTCP(0x6321c585, 0x50f975e6), WTCP(0x6301f171, 0x51205d7b), WTCP(0x62e20e17, 0x5147388c), WTCP(0x62c21b7e, 0x516e0715), WTCP(0x62a219aa, 0x5194c910), WTCP(0x628208a1, 0x51bb7e75), WTCP(0x6261e866, 0x51e22740), WTCP(0x6241b8ff, 0x5208c36a), WTCP(0x62217a72, 0x522f52ee), WTCP(0x62012cc2, 0x5255d5c5), WTCP(0x61e0cff5, 0x527c4bea), WTCP(0x61c06410, 0x52a2b556), WTCP(0x619fe918, 0x52c91204), WTCP(0x617f5f12, 0x52ef61ee), WTCP(0x615ec603, 0x5315a50e), WTCP(0x613e1df0, 0x533bdb5d), WTCP(0x611d66de, 0x536204d7), WTCP(0x60fca0d2, 0x53882175), WTCP(0x60dbcbd1, 0x53ae3131), WTCP(0x60bae7e1, 0x53d43406), WTCP(0x6099f505, 0x53fa29ed), WTCP(0x6078f344, 0x542012e1), WTCP(0x6057e2a2, 0x5445eedb), WTCP(0x6036c325, 0x546bbdd7), WTCP(0x601594d1, 0x54917fce), WTCP(0x5ff457ad, 0x54b734ba), WTCP(0x5fd30bbc, 0x54dcdc96), WTCP(0x5fb1b104, 0x5502775c), WTCP(0x5f90478a, 0x55280505), WTCP(0x5f6ecf53, 0x554d858d), WTCP(0x5f4d4865, 0x5572f8ed), WTCP(0x5f2bb2c5, 0x55985f20), WTCP(0x5f0a0e77, 0x55bdb81f), WTCP(0x5ee85b82, 0x55e303e6), WTCP(0x5ec699e9, 0x5608426e), WTCP(0x5ea4c9b3, 0x562d73b2), WTCP(0x5e82eae5, 0x565297ab), WTCP(0x5e60fd84, 0x5677ae54), WTCP(0x5e3f0194, 0x569cb7a8), WTCP(0x5e1cf71c, 0x56c1b3a1), WTCP(0x5dfade20, 0x56e6a239), WTCP(0x5dd8b6a7, 0x570b8369), WTCP(0x5db680b4, 0x5730572e), WTCP(0x5d943c4e, 0x57551d80), WTCP(0x5d71e979, 0x5779d65b), WTCP(0x5d4f883b, 0x579e81b8), WTCP(0x5d2d189a, 0x57c31f92), WTCP(0x5d0a9a9a, 0x57e7afe4), WTCP(0x5ce80e41, 0x580c32a7), WTCP(0x5cc57394, 0x5830a7d6), WTCP(0x5ca2ca99, 0x58550f6c), WTCP(0x5c801354, 0x58796962), WTCP(0x5c5d4dcc, 0x589db5b3), WTCP(0x5c3a7a05, 0x58c1f45b), WTCP(0x5c179806, 0x58e62552), WTCP(0x5bf4a7d2, 0x590a4893), WTCP(0x5bd1a971, 0x592e5e19), WTCP(0x5bae9ce7, 0x595265df), WTCP(0x5b8b8239, 0x59765fde), WTCP(0x5b68596d, 0x599a4c12), WTCP(0x5b452288, 0x59be2a74), WTCP(0x5b21dd90, 0x59e1faff), WTCP(0x5afe8a8b, 0x5a05bdae), WTCP(0x5adb297d, 0x5a29727b), WTCP(0x5ab7ba6c, 0x5a4d1960), WTCP(0x5a943d5e, 0x5a70b258), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_WTP KBDWindow96[] = { WTCP(0x7ffffffd, 0x0001a838), WTCP(0x7fffffe2, 0x00056e83), WTCP(0x7fffff79, 0x000b9fda), WTCP(0x7ffffe45, 0x00150e8e), WTCP(0x7ffffb4d, 0x0022aeeb), WTCP(0x7ffff4c6, 0x00359b36), WTCP(0x7fffe792, 0x004f14ff), WTCP(0x7fffce8b, 0x0070858c), WTCP(0x7fffa18f, 0x009b7d75), WTCP(0x7fff5439, 0x00d1b353), WTCP(0x7ffed442, 0x0115018f), WTCP(0x7ffe0775, 0x01676335), WTCP(0x7ffcc937, 0x01caefcb), WTCP(0x7ffae79f, 0x0241d62e), WTCP(0x7ff82019, 0x02ce567f), WTCP(0x7ff41ba4, 0x0372bb25), WTCP(0x7fee6ac3, 0x043150fc), WTCP(0x7fe68129, 0x050c5ec8), WTCP(0x7fdbb164, 0x06061c0f), WTCP(0x7fcd2894, 0x0720a779), WTCP(0x7fb9ea80, 0x085dfce2), WTCP(0x7fa0ce2e, 0x09bfeb4d), WTCP(0x7f807b45, 0x0b480ae2), WTCP(0x7f576880, 0x0cf7b339), WTCP(0x7f23db4e, 0x0ecff212), WTCP(0x7ee3e8ee, 0x10d182c0), WTCP(0x7e95791f, 0x12fcc670), WTCP(0x7e364a74, 0x1551bd88), WTCP(0x7dc3f864, 0x17d00238), WTCP(0x7d3c02fd, 0x1a76c47e), WTCP(0x7c9bd82a, 0x1d44c7ad), WTCP(0x7be0de56, 0x203861a1), WTCP(0x7b08803d, 0x234f7ba6), WTCP(0x7a103993, 0x26879530), WTCP(0x78f5a442, 0x29ddc854), WTCP(0x77b685de, 0x2d4ed00f), WTCP(0x7650dcf5, 0x30d7103d), WTCP(0x74c2ede4, 0x34729f2d), WTCP(0x730b4edb, 0x381d50ad), WTCP(0x7128f2c1, 0x3bd2c273), WTCP(0x6f1b32a9, 0x3f8e698f), WTCP(0x6ce1d5a0, 0x434ba0d6), WTCP(0x6a7d16a3, 0x4705b7e5), WTCP(0x67eda890, 0x4ab80288), WTCP(0x6534b7f8, 0x4e5de842), WTCP(0x6253eacd, 0x51f2f39a), WTCP(0x5f4d5de1, 0x5572e0f7), WTCP(0x5c23a04a, 0x58d9acb9), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_WTP KBDWindow120[] = { WTCP(0x7ffffffe, 0x00017b6f), WTCP(0x7fffffef, 0x00042d2f), WTCP(0x7fffffbb, 0x000849d0), WTCP(0x7fffff36, 0x000e3494), WTCP(0x7ffffe0c, 0x00165efd), WTCP(0x7ffffbac, 0x002149be), WTCP(0x7ffff72e, 0x002f854c), WTCP(0x7fffef24, 0x0041b235), WTCP(0x7fffe167, 0x0058814f), WTCP(0x7fffcacd, 0x0074b3af), WTCP(0x7fffa6d0, 0x00971a67), WTCP(0x7fff6f1e, 0x00c0960e), WTCP(0x7fff1b12, 0x00f21602), WTCP(0x7ffe9f0b, 0x012c9775), WTCP(0x7ffdebb2, 0x01712428), WTCP(0x7ffced1b, 0x01c0d0f7), WTCP(0x7ffb89c2, 0x021cbc12), WTCP(0x7ff9a17c, 0x02860b05), WTCP(0x7ff70c39, 0x02fde875), WTCP(0x7ff398bc, 0x038581b3), WTCP(0x7fef0b3b, 0x041e040c), WTCP(0x7fe91bf3, 0x04c899f4), WTCP(0x7fe175ba, 0x05866803), WTCP(0x7fd7b493, 0x065889d5), WTCP(0x7fcb6459, 0x07400ed4), WTCP(0x7fbbff82, 0x083df6e9), WTCP(0x7fa8ee09, 0x09532f37), WTCP(0x7f91849a, 0x0a808ed1), WTCP(0x7f7503f2, 0x0bc6d381), WTCP(0x7f52989a, 0x0d269eb0), WTCP(0x7f295af4, 0x0ea07270), WTCP(0x7ef84fb6, 0x1034aeb6), WTCP(0x7ebe68c5, 0x11e38ed2), WTCP(0x7e7a8686, 0x13ad2733), WTCP(0x7e2b79a3, 0x1591636d), WTCP(0x7dd0053c, 0x179004a7), WTCP(0x7d66e18b, 0x19a8a05f), WTCP(0x7ceebef0, 0x1bda9fa2), WTCP(0x7c664953, 0x1e253ea1), WTCP(0x7bcc2be8, 0x20878cce), WTCP(0x7b1f1526, 0x23006d5d), WTCP(0x7a5dbb01, 0x258e9848), WTCP(0x7986df3e, 0x28309bc6), WTCP(0x789953e0, 0x2ae4de3e), WTCP(0x7793ff88, 0x2da9a0a8), WTCP(0x7675e1cc, 0x307d0163), WTCP(0x753e1763, 0x335cff72), WTCP(0x73ebde10, 0x36477e1f), WTCP(0x727e984e, 0x393a48f1), WTCP(0x70f5d09b, 0x3c3317f9), WTCP(0x6f513c60, 0x3f2f945c), WTCP(0x6d90be61, 0x422d5d18), WTCP(0x6bb468b1, 0x452a0bf3), WTCP(0x69bc7e1e, 0x48233a81), WTCP(0x67a97317, 0x4b16873e), WTCP(0x657bedfa, 0x4e019a9d), WTCP(0x6334c6d2, 0x50e22c0b), WTCP(0x60d50689, 0x53b606cb), WTCP(0x5e5de588, 0x567b0ea7), WTCP(0x5bd0c9c6, 0x592f4460), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_WTP KBDWindow128[] = { WTCP(0x7ffffffe, 0x00016f63), WTCP(0x7ffffff1, 0x0003e382), WTCP(0x7fffffc7, 0x00078f64), WTCP(0x7fffff5d, 0x000cc323), WTCP(0x7ffffe76, 0x0013d9ed), WTCP(0x7ffffcaa, 0x001d3a9d), WTCP(0x7ffff953, 0x0029581f), WTCP(0x7ffff372, 0x0038b1bd), WTCP(0x7fffe98b, 0x004bd34d), WTCP(0x7fffd975, 0x00635538), WTCP(0x7fffc024, 0x007fdc64), WTCP(0x7fff995b, 0x00a219f1), WTCP(0x7fff5f5b, 0x00cacad0), WTCP(0x7fff0a75, 0x00fab72d), WTCP(0x7ffe9091, 0x0132b1af), WTCP(0x7ffde49e, 0x01739689), WTCP(0x7ffcf5ef, 0x01be4a63), WTCP(0x7ffbaf84, 0x0213b910), WTCP(0x7ff9f73a, 0x0274d41e), WTCP(0x7ff7acf1, 0x02e2913a), WTCP(0x7ff4a99a, 0x035de86c), WTCP(0x7ff0be3d, 0x03e7d233), WTCP(0x7febb2f1, 0x0481457c), WTCP(0x7fe545d4, 0x052b357c), WTCP(0x7fdd2a02, 0x05e68f77), WTCP(0x7fd30695, 0x06b4386f), WTCP(0x7fc675b4, 0x07950acb), WTCP(0x7fb703be, 0x0889d3ef), WTCP(0x7fa42e89, 0x099351e0), WTCP(0x7f8d64d8, 0x0ab230e0), WTCP(0x7f7205f8, 0x0be70923), WTCP(0x7f516195, 0x0d325c93), WTCP(0x7f2ab7d0, 0x0e9494ae), WTCP(0x7efd3997, 0x100e0085), WTCP(0x7ec8094a, 0x119ed2ef), WTCP(0x7e8a3ba7, 0x134720d8), WTCP(0x7e42d906, 0x1506dfdc), WTCP(0x7df0dee4, 0x16dde50b), WTCP(0x7d9341b4, 0x18cbe3f7), WTCP(0x7d28ef02, 0x1ad06e07), WTCP(0x7cb0cfcc, 0x1ceaf215), WTCP(0x7c29cb20, 0x1f1abc4f), WTCP(0x7b92c8eb, 0x215ef677), WTCP(0x7aeab4ec, 0x23b6a867), WTCP(0x7a3081d0, 0x2620b8ec), WTCP(0x79632c5a, 0x289beef5), WTCP(0x7881be95, 0x2b26f30b), WTCP(0x778b5304, 0x2dc0511f), WTCP(0x767f17c0, 0x30667aa2), WTCP(0x755c5178, 0x3317c8dd), WTCP(0x74225e50, 0x35d27f98), WTCP(0x72d0b887, 0x3894cff3), WTCP(0x7166f8e7, 0x3b5cdb7b), WTCP(0x6fe4d8e8, 0x3e28b770), WTCP(0x6e4a3491, 0x40f6702a), WTCP(0x6c970bfc, 0x43c40caa), WTCP(0x6acb8483, 0x468f9231), WTCP(0x68e7e994, 0x495707f5), WTCP(0x66ecad1c, 0x4c187ac7), WTCP(0x64da6797, 0x4ed200c5), WTCP(0x62b1d7b7, 0x5181bcea), WTCP(0x6073e1ae, 0x5425e28e), WTCP(0x5e218e16, 0x56bcb8c2), WTCP(0x5bbc0875, 0x59449d76), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_WTP KBDWindow256[] = { WTCP(0x7fffffff, 0x000103c8), WTCP(0x7ffffffc, 0x000203ad), WTCP(0x7ffffff5, 0x0003410a), WTCP(0x7fffffe9, 0x0004c6ce), WTCP(0x7fffffd4, 0x00069ee0), WTCP(0x7fffffb2, 0x0008d376), WTCP(0x7fffff7d, 0x000b6f5a), WTCP(0x7fffff2e, 0x000e7dfd), WTCP(0x7ffffeba, 0x00120b83), WTCP(0x7ffffe16, 0x001624cd), WTCP(0x7ffffd30, 0x001ad778), WTCP(0x7ffffbf3, 0x002031e2), WTCP(0x7ffffa48, 0x00264330), WTCP(0x7ffff80d, 0x002d1b4b), WTCP(0x7ffff51d, 0x0034cae6), WTCP(0x7ffff147, 0x003d637c), WTCP(0x7fffec54, 0x0046f751), WTCP(0x7fffe5fe, 0x00519974), WTCP(0x7fffddf3, 0x005d5dba), WTCP(0x7fffd3d2, 0x006a58c1), WTCP(0x7fffc72a, 0x00789feb), WTCP(0x7fffb772, 0x0088495d), WTCP(0x7fffa40e, 0x00996bfb), WTCP(0x7fff8c46, 0x00ac1f63), WTCP(0x7fff6f46, 0x00c07bec), WTCP(0x7fff4c19, 0x00d69a9b), WTCP(0x7fff21a6, 0x00ee9523), WTCP(0x7ffeeeab, 0x010885d9), WTCP(0x7ffeb1b8, 0x012487b1), WTCP(0x7ffe692f, 0x0142b631), WTCP(0x7ffe1335, 0x01632d6f), WTCP(0x7ffdadb8, 0x01860a00), WTCP(0x7ffd3661, 0x01ab68f3), WTCP(0x7ffcaa91, 0x01d367c5), WTCP(0x7ffc075b, 0x01fe2453), WTCP(0x7ffb497e, 0x022bbcd0), WTCP(0x7ffa6d59, 0x025c4fba), WTCP(0x7ff96eeb, 0x028ffbc7), WTCP(0x7ff849c6, 0x02c6dfdb), WTCP(0x7ff6f90b, 0x03011afc), WTCP(0x7ff57760, 0x033ecc3a), WTCP(0x7ff3bee7, 0x038012a8), WTCP(0x7ff1c939, 0x03c50d47), WTCP(0x7fef8f5a, 0x040ddaf6), WTCP(0x7fed09b4, 0x045a9a64), WTCP(0x7fea300e, 0x04ab69f9), WTCP(0x7fe6f980, 0x050067c7), WTCP(0x7fe35c70, 0x0559b17b), WTCP(0x7fdf4e88, 0x05b76443), WTCP(0x7fdac4ad, 0x06199cc4), WTCP(0x7fd5b2f8, 0x068076fe), WTCP(0x7fd00caf, 0x06ec0e41), WTCP(0x7fc9c441, 0x075c7d16), WTCP(0x7fc2cb3b, 0x07d1dd2c), WTCP(0x7fbb1242, 0x084c4745), WTCP(0x7fb28915, 0x08cbd323), WTCP(0x7fa91e7e, 0x09509778), WTCP(0x7f9ec059, 0x09daa9cc), WTCP(0x7f935b87, 0x0a6a1e74), WTCP(0x7f86dbf2, 0x0aff0877), WTCP(0x7f792c8a, 0x0b997983), WTCP(0x7f6a3746, 0x0c3981d6), WTCP(0x7f59e520, 0x0cdf3030), WTCP(0x7f481e1c, 0x0d8a91c3), WTCP(0x7f34c949, 0x0e3bb222), WTCP(0x7f1fccc3, 0x0ef29b30), WTCP(0x7f090dbc, 0x0faf5513), WTCP(0x7ef0707d, 0x1071e629), WTCP(0x7ed5d872, 0x113a52f4), WTCP(0x7eb92831, 0x12089e14), WTCP(0x7e9a4183, 0x12dcc836), WTCP(0x7e790571, 0x13b6d010), WTCP(0x7e55544e, 0x1496b24f), WTCP(0x7e2f0dc8, 0x157c6998), WTCP(0x7e0610f1, 0x1667ee77), WTCP(0x7dda3c54, 0x17593760), WTCP(0x7dab6e06, 0x185038a3), WTCP(0x7d7983b3, 0x194ce46e), WTCP(0x7d445ab5, 0x1a4f2ac4), WTCP(0x7d0bd028, 0x1b56f981), WTCP(0x7ccfc0fd, 0x1c643c54), WTCP(0x7c900a11, 0x1d76dcc2), WTCP(0x7c4c8844, 0x1e8ec227), WTCP(0x7c05188d, 0x1fabd1bb), WTCP(0x7bb99817, 0x20cdee92), WTCP(0x7b69e455, 0x21f4f9a6), WTCP(0x7b15db1a, 0x2320d1dc), WTCP(0x7abd5ab8, 0x2451540c), WTCP(0x7a604213, 0x25865b09), WTCP(0x79fe70bf, 0x26bfbfaf), WTCP(0x7997c716, 0x27fd58ed), WTCP(0x792c2654, 0x293efbd0), WTCP(0x78bb70b0, 0x2a847b97), WTCP(0x78458976, 0x2bcda9bb), WTCP(0x77ca551d, 0x2d1a5608), WTCP(0x7749b965, 0x2e6a4ea6), WTCP(0x76c39d68, 0x2fbd6036), WTCP(0x7637e9b8, 0x311355dc), WTCP(0x75a68873, 0x326bf95a), WTCP(0x750f6559, 0x33c71326), WTCP(0x74726de1, 0x35246a7e), WTCP(0x73cf914f, 0x3683c582), WTCP(0x7326c0c8, 0x37e4e94b), WTCP(0x7277ef5f, 0x39479a08), WTCP(0x71c3122f, 0x3aab9b14), WTCP(0x71082063, 0x3c10af11), WTCP(0x7047134a, 0x3d769807), WTCP(0x6f7fe661, 0x3edd177c), WTCP(0x6eb29763, 0x4043ee92), WTCP(0x6ddf2651, 0x41aade26), WTCP(0x6d05957c, 0x4311a6e8), WTCP(0x6c25e98f, 0x4478097b), WTCP(0x6b402991, 0x45ddc693), WTCP(0x6a545ef0, 0x47429f13), WTCP(0x6962957f, 0x48a65427), WTCP(0x686adb7c, 0x4a08a764), WTCP(0x676d418d, 0x4b695ae8), WTCP(0x6669dac2, 0x4cc83171), WTCP(0x6560bc90, 0x4e24ee7d), WTCP(0x6451fecf, 0x4f7f5668), WTCP(0x633dbbb1, 0x50d72e85), WTCP(0x62240fbd, 0x522c3d3b), WTCP(0x610519c7, 0x537e4a1f), WTCP(0x5fe0fae3, 0x54cd1e10), WTCP(0x5eb7d65c, 0x5618834c), WTCP(0x5d89d1a5, 0x57604590), WTCP(0x5c57144b, 0x58a43227), WTCP(0x5b1fc7e6, 0x59e41808), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_WTP KBDWindow768[] = { WTCP(0x7fffff85, 0x000b11d9), WTCP(0x7ffffef0, 0x00107aa9), WTCP(0x7ffffe3e, 0x0015351c), WTCP(0x7ffffd6c, 0x0019b0a1), WTCP(0x7ffffc77, 0x001e1656), WTCP(0x7ffffb5b, 0x00227a80), WTCP(0x7ffffa16, 0x0026e8d3), WTCP(0x7ffff8a4, 0x002b68c9), WTCP(0x7ffff700, 0x002fff8a), WTCP(0x7ffff528, 0x0034b0d9), WTCP(0x7ffff316, 0x00397f9c), WTCP(0x7ffff0c6, 0x003e6e22), WTCP(0x7fffee35, 0x00437e53), WTCP(0x7fffeb5b, 0x0048b1d0), WTCP(0x7fffe836, 0x004e0a05), WTCP(0x7fffe4be, 0x00538837), WTCP(0x7fffe0ef, 0x00592d8e), WTCP(0x7fffdcc3, 0x005efb1a), WTCP(0x7fffd832, 0x0064f1da), WTCP(0x7fffd337, 0x006b12c1), WTCP(0x7fffcdcb, 0x00715eb4), WTCP(0x7fffc7e7, 0x0077d692), WTCP(0x7fffc182, 0x007e7b30), WTCP(0x7fffba96, 0x00854d61), WTCP(0x7fffb31b, 0x008c4df0), WTCP(0x7fffab06, 0x00937da6), WTCP(0x7fffa251, 0x009add48), WTCP(0x7fff98f1, 0x00a26d98), WTCP(0x7fff8edd, 0x00aa2f57), WTCP(0x7fff840b, 0x00b22343), WTCP(0x7fff7870, 0x00ba4a19), WTCP(0x7fff6c02, 0x00c2a495), WTCP(0x7fff5eb5, 0x00cb3371), WTCP(0x7fff507e, 0x00d3f767), WTCP(0x7fff4150, 0x00dcf130), WTCP(0x7fff311f, 0x00e62183), WTCP(0x7fff1fde, 0x00ef8919), WTCP(0x7fff0d7f, 0x00f928a7), WTCP(0x7ffef9f4, 0x010300e5), WTCP(0x7ffee52f, 0x010d1288), WTCP(0x7ffecf20, 0x01175e47), WTCP(0x7ffeb7b8, 0x0121e4d6), WTCP(0x7ffe9ee6, 0x012ca6eb), WTCP(0x7ffe849b, 0x0137a53b), WTCP(0x7ffe68c4, 0x0142e07a), WTCP(0x7ffe4b50, 0x014e595c), WTCP(0x7ffe2c2c, 0x015a1095), WTCP(0x7ffe0b45, 0x016606da), WTCP(0x7ffde888, 0x01723cde), WTCP(0x7ffdc3df, 0x017eb353), WTCP(0x7ffd9d37, 0x018b6aed), WTCP(0x7ffd7479, 0x0198645f), WTCP(0x7ffd4990, 0x01a5a05b), WTCP(0x7ffd1c63, 0x01b31f92), WTCP(0x7ffcecdc, 0x01c0e2b8), WTCP(0x7ffcbae2, 0x01ceea7d), WTCP(0x7ffc865c, 0x01dd3793), WTCP(0x7ffc4f2f, 0x01ebcaaa), WTCP(0x7ffc1542, 0x01faa472), WTCP(0x7ffbd879, 0x0209c59c), WTCP(0x7ffb98b7, 0x02192ed7), WTCP(0x7ffb55e0, 0x0228e0d2), WTCP(0x7ffb0fd6, 0x0238dc3c), WTCP(0x7ffac679, 0x024921c3), WTCP(0x7ffa79ac, 0x0259b215), WTCP(0x7ffa294d, 0x026a8dde), WTCP(0x7ff9d53b, 0x027bb5cc), WTCP(0x7ff97d54, 0x028d2a8a), WTCP(0x7ff92175, 0x029eecc3), WTCP(0x7ff8c17a, 0x02b0fd23), WTCP(0x7ff85d3f, 0x02c35c53), WTCP(0x7ff7f49d, 0x02d60afd), WTCP(0x7ff7876e, 0x02e909ca), WTCP(0x7ff7158b, 0x02fc5960), WTCP(0x7ff69eca, 0x030ffa69), WTCP(0x7ff62303, 0x0323ed89), WTCP(0x7ff5a20a, 0x03383367), WTCP(0x7ff51bb3, 0x034ccca7), WTCP(0x7ff48fd3, 0x0361b9ed), WTCP(0x7ff3fe3c, 0x0376fbdd), WTCP(0x7ff366be, 0x038c9317), WTCP(0x7ff2c929, 0x03a2803e), WTCP(0x7ff2254e, 0x03b8c3f2), WTCP(0x7ff17afa, 0x03cf5ed1), WTCP(0x7ff0c9f9, 0x03e6517a), WTCP(0x7ff01218, 0x03fd9c8a), WTCP(0x7fef5321, 0x0415409c), WTCP(0x7fee8cde, 0x042d3e4d), WTCP(0x7fedbf17, 0x04459634), WTCP(0x7fece993, 0x045e48ec), WTCP(0x7fec0c18, 0x0477570a), WTCP(0x7feb266a, 0x0490c127), WTCP(0x7fea384e, 0x04aa87d5), WTCP(0x7fe94186, 0x04c4abaa), WTCP(0x7fe841d3, 0x04df2d37), WTCP(0x7fe738f4, 0x04fa0d0d), WTCP(0x7fe626a9, 0x05154bbc), WTCP(0x7fe50aaf, 0x0530e9d3), WTCP(0x7fe3e4c1, 0x054ce7dd), WTCP(0x7fe2b49b, 0x05694667), WTCP(0x7fe179f6, 0x058605fa), WTCP(0x7fe0348b, 0x05a3271e), WTCP(0x7fdee410, 0x05c0aa5c), WTCP(0x7fdd883b, 0x05de9038), WTCP(0x7fdc20c1, 0x05fcd935), WTCP(0x7fdaad53, 0x061b85d6), WTCP(0x7fd92da5, 0x063a969c), WTCP(0x7fd7a166, 0x065a0c06), WTCP(0x7fd60844, 0x0679e690), WTCP(0x7fd461ee, 0x069a26b6), WTCP(0x7fd2ae10, 0x06baccf2), WTCP(0x7fd0ec55, 0x06dbd9bd), WTCP(0x7fcf1c65, 0x06fd4d8c), WTCP(0x7fcd3de9, 0x071f28d3), WTCP(0x7fcb5088, 0x07416c06), WTCP(0x7fc953e6, 0x07641794), WTCP(0x7fc747a8, 0x07872bee), WTCP(0x7fc52b70, 0x07aaa97f), WTCP(0x7fc2fedf, 0x07ce90b4), WTCP(0x7fc0c195, 0x07f2e1f4), WTCP(0x7fbe732f, 0x08179da7), WTCP(0x7fbc134b, 0x083cc431), WTCP(0x7fb9a183, 0x086255f7), WTCP(0x7fb71d72, 0x08885359), WTCP(0x7fb486af, 0x08aebcb5), WTCP(0x7fb1dcd3, 0x08d59269), WTCP(0x7faf1f72, 0x08fcd4cf), WTCP(0x7fac4e21, 0x09248440), WTCP(0x7fa96873, 0x094ca111), WTCP(0x7fa66df8, 0x09752b98), WTCP(0x7fa35e40, 0x099e2425), WTCP(0x7fa038db, 0x09c78b09), WTCP(0x7f9cfd54, 0x09f16090), WTCP(0x7f99ab38, 0x0a1ba507), WTCP(0x7f964210, 0x0a4658b6), WTCP(0x7f92c165, 0x0a717be2), WTCP(0x7f8f28bf, 0x0a9d0ed1), WTCP(0x7f8b77a4, 0x0ac911c4), WTCP(0x7f87ad97, 0x0af584fb), WTCP(0x7f83ca1d, 0x0b2268b2), WTCP(0x7f7fccb5, 0x0b4fbd23), WTCP(0x7f7bb4e2, 0x0b7d8288), WTCP(0x7f778221, 0x0babb915), WTCP(0x7f7333f1, 0x0bda60fd), WTCP(0x7f6ec9cd, 0x0c097a72), WTCP(0x7f6a4330, 0x0c3905a1), WTCP(0x7f659f94, 0x0c6902b6), WTCP(0x7f60de70, 0x0c9971d9), WTCP(0x7f5bff3b, 0x0cca5331), WTCP(0x7f57016b, 0x0cfba6e3), WTCP(0x7f51e474, 0x0d2d6d0e), WTCP(0x7f4ca7c8, 0x0d5fa5d2), WTCP(0x7f474ad9, 0x0d92514a), WTCP(0x7f41cd17, 0x0dc56f90), WTCP(0x7f3c2df1, 0x0df900bb), WTCP(0x7f366cd5, 0x0e2d04de), WTCP(0x7f30892e, 0x0e617c0a), WTCP(0x7f2a8269, 0x0e96664e), WTCP(0x7f2457ef, 0x0ecbc3b5), WTCP(0x7f1e0929, 0x0f019449), WTCP(0x7f17957e, 0x0f37d80f), WTCP(0x7f10fc55, 0x0f6e8f0c), WTCP(0x7f0a3d14, 0x0fa5b940), WTCP(0x7f03571d, 0x0fdd56a8), WTCP(0x7efc49d4, 0x10156740), WTCP(0x7ef5149b, 0x104deb00), WTCP(0x7eedb6d2, 0x1086e1dd), WTCP(0x7ee62fda, 0x10c04bca), WTCP(0x7ede7f11, 0x10fa28b7), WTCP(0x7ed6a3d5, 0x11347890), WTCP(0x7ece9d81, 0x116f3b3f), WTCP(0x7ec66b73, 0x11aa70ac), WTCP(0x7ebe0d04, 0x11e618ba), WTCP(0x7eb5818d, 0x1222334c), WTCP(0x7eacc869, 0x125ec03e), WTCP(0x7ea3e0ef, 0x129bbf6e), WTCP(0x7e9aca75, 0x12d930b2), WTCP(0x7e918452, 0x131713e2), WTCP(0x7e880ddb, 0x135568cf), WTCP(0x7e7e6665, 0x13942f49), WTCP(0x7e748d43, 0x13d3671e), WTCP(0x7e6a81c8, 0x14131017), WTCP(0x7e604347, 0x145329fa), WTCP(0x7e55d111, 0x1493b48c), WTCP(0x7e4b2a76, 0x14d4af8e), WTCP(0x7e404ec8, 0x15161abe), WTCP(0x7e353d55, 0x1557f5d7), WTCP(0x7e29f56c, 0x159a4090), WTCP(0x7e1e765c, 0x15dcfaa0), WTCP(0x7e12bf72, 0x162023b7), WTCP(0x7e06cffc, 0x1663bb86), WTCP(0x7dfaa746, 0x16a7c1b9), WTCP(0x7dee449e, 0x16ec35f7), WTCP(0x7de1a74e, 0x173117e9), WTCP(0x7dd4cea3, 0x17766731), WTCP(0x7dc7b9e7, 0x17bc236f), WTCP(0x7dba6865, 0x18024c40), WTCP(0x7dacd968, 0x1848e13f), WTCP(0x7d9f0c3a, 0x188fe204), WTCP(0x7d910025, 0x18d74e22), WTCP(0x7d82b472, 0x191f252c), WTCP(0x7d74286c, 0x196766ae), WTCP(0x7d655b5b, 0x19b01236), WTCP(0x7d564c8a, 0x19f9274b), WTCP(0x7d46fb40, 0x1a42a574), WTCP(0x7d3766c8, 0x1a8c8c32), WTCP(0x7d278e6a, 0x1ad6db06), WTCP(0x7d17716f, 0x1b21916c), WTCP(0x7d070f22, 0x1b6caedf), WTCP(0x7cf666cb, 0x1bb832d5), WTCP(0x7ce577b3, 0x1c041cc2), WTCP(0x7cd44124, 0x1c506c17), WTCP(0x7cc2c269, 0x1c9d2044), WTCP(0x7cb0faca, 0x1cea38b2), WTCP(0x7c9ee992, 0x1d37b4cc), WTCP(0x7c8c8e0c, 0x1d8593f5), WTCP(0x7c79e782, 0x1dd3d592), WTCP(0x7c66f541, 0x1e227903), WTCP(0x7c53b692, 0x1e717da3), WTCP(0x7c402ac3, 0x1ec0e2cf), WTCP(0x7c2c5120, 0x1f10a7dc), WTCP(0x7c1828f6, 0x1f60cc21), WTCP(0x7c03b193, 0x1fb14eef), WTCP(0x7beeea44, 0x20022f96), WTCP(0x7bd9d259, 0x20536d61), WTCP(0x7bc46921, 0x20a5079a), WTCP(0x7baeadec, 0x20f6fd8a), WTCP(0x7b98a00b, 0x21494e73), WTCP(0x7b823ecf, 0x219bf998), WTCP(0x7b6b898b, 0x21eefe37), WTCP(0x7b547f93, 0x22425b8d), WTCP(0x7b3d203a, 0x229610d4), WTCP(0x7b256ad5, 0x22ea1d42), WTCP(0x7b0d5ebb, 0x233e800c), WTCP(0x7af4fb42, 0x23933864), WTCP(0x7adc3fc2, 0x23e8457a), WTCP(0x7ac32b95, 0x243da679), WTCP(0x7aa9be14, 0x24935a8d), WTCP(0x7a8ff69a, 0x24e960dd), WTCP(0x7a75d485, 0x253fb88e), WTCP(0x7a5b5731, 0x259660c3), WTCP(0x7a407dfe, 0x25ed589c), WTCP(0x7a25484c, 0x26449f38), WTCP(0x7a09b57c, 0x269c33b1), WTCP(0x79edc4f1, 0x26f41522), WTCP(0x79d1760e, 0x274c42a0), WTCP(0x79b4c83b, 0x27a4bb40), WTCP(0x7997badd, 0x27fd7e15), WTCP(0x797a4d5e, 0x28568a2f), WTCP(0x795c7f26, 0x28afde9a), WTCP(0x793e4fa3, 0x29097a63), WTCP(0x791fbe40, 0x29635c92), WTCP(0x7900ca6e, 0x29bd842e), WTCP(0x78e1739c, 0x2a17f03e), WTCP(0x78c1b93d, 0x2a729fc2), WTCP(0x78a19ac4, 0x2acd91bc), WTCP(0x788117a7, 0x2b28c52a), WTCP(0x78602f5e, 0x2b843909), WTCP(0x783ee163, 0x2bdfec54), WTCP(0x781d2d2f, 0x2c3bde02), WTCP(0x77fb1241, 0x2c980d0a), WTCP(0x77d89017, 0x2cf47862), WTCP(0x77b5a632, 0x2d511efb), WTCP(0x77925416, 0x2dadffc6), WTCP(0x776e9947, 0x2e0b19b3), WTCP(0x774a754d, 0x2e686bae), WTCP(0x7725e7b0, 0x2ec5f4a4), WTCP(0x7700effd, 0x2f23b37d), WTCP(0x76db8dbf, 0x2f81a721), WTCP(0x76b5c088, 0x2fdfce77), WTCP(0x768f87e8, 0x303e2863), WTCP(0x7668e375, 0x309cb3c8), WTCP(0x7641d2c4, 0x30fb6f88), WTCP(0x761a556e, 0x315a5a82), WTCP(0x75f26b0e, 0x31b97394), WTCP(0x75ca1341, 0x3218b99c), WTCP(0x75a14da8, 0x32782b74), WTCP(0x757819e4, 0x32d7c7f6), WTCP(0x754e779a, 0x33378dfc), WTCP(0x75246671, 0x33977c5b), WTCP(0x74f9e613, 0x33f791e9), WTCP(0x74cef62b, 0x3457cd7c), WTCP(0x74a3966a, 0x34b82de6), WTCP(0x7477c67f, 0x3518b1f9), WTCP(0x744b861e, 0x35795887), WTCP(0x741ed4ff, 0x35da205e), WTCP(0x73f1b2da, 0x363b084e), WTCP(0x73c41f6b, 0x369c0f24), WTCP(0x73961a71, 0x36fd33ac), WTCP(0x7367a3ac, 0x375e74b1), WTCP(0x7338bae1, 0x37bfd0ff), WTCP(0x73095fd7, 0x3821475f), WTCP(0x72d99257, 0x3882d699), WTCP(0x72a9522d, 0x38e47d75), WTCP(0x72789f28, 0x39463aba), WTCP(0x7247791b, 0x39a80d2e), WTCP(0x7215dfda, 0x3a09f397), WTCP(0x71e3d33d, 0x3a6becba), WTCP(0x71b1531f, 0x3acdf75a), WTCP(0x717e5f5d, 0x3b30123b), WTCP(0x714af7d7, 0x3b923c20), WTCP(0x71171c72, 0x3bf473cc), WTCP(0x70e2cd14, 0x3c56b7ff), WTCP(0x70ae09a6, 0x3cb9077b), WTCP(0x7078d215, 0x3d1b6101), WTCP(0x7043264f, 0x3d7dc353), WTCP(0x700d0648, 0x3de02d2e), WTCP(0x6fd671f5, 0x3e429d55), WTCP(0x6f9f694f, 0x3ea51285), WTCP(0x6f67ec52, 0x3f078b7f), WTCP(0x6f2ffafb, 0x3f6a0701), WTCP(0x6ef7954e, 0x3fcc83ca), WTCP(0x6ebebb4e, 0x402f009a), WTCP(0x6e856d05, 0x40917c2e), WTCP(0x6e4baa7e, 0x40f3f546), WTCP(0x6e1173c6, 0x41566aa1), WTCP(0x6dd6c8ef, 0x41b8dafc), WTCP(0x6d9baa0f, 0x421b4518), WTCP(0x6d60173d, 0x427da7b1), WTCP(0x6d241094, 0x42e00189), WTCP(0x6ce79632, 0x4342515e), WTCP(0x6caaa839, 0x43a495ef), WTCP(0x6c6d46ce, 0x4406cdfd), WTCP(0x6c2f7218, 0x4468f848), WTCP(0x6bf12a42, 0x44cb138f), WTCP(0x6bb26f7b, 0x452d1e94), WTCP(0x6b7341f5, 0x458f1818), WTCP(0x6b33a1e3, 0x45f0fede), WTCP(0x6af38f7e, 0x4652d1a6), WTCP(0x6ab30b01, 0x46b48f34), WTCP(0x6a7214ab, 0x4716364c), WTCP(0x6a30acbd, 0x4777c5b2), WTCP(0x69eed37c, 0x47d93c2a), WTCP(0x69ac8930, 0x483a987a), WTCP(0x6969ce24, 0x489bd968), WTCP(0x6926a2a8, 0x48fcfdbb), WTCP(0x68e3070c, 0x495e043b), WTCP(0x689efba7, 0x49beebb0), WTCP(0x685a80cf, 0x4a1fb2e5), WTCP(0x681596e1, 0x4a8058a4), WTCP(0x67d03e3b, 0x4ae0dbb8), WTCP(0x678a773f, 0x4b413aee), WTCP(0x67444253, 0x4ba17514), WTCP(0x66fd9fde, 0x4c0188f8), WTCP(0x66b6904c, 0x4c61756b), WTCP(0x666f140d, 0x4cc1393d), WTCP(0x66272b91, 0x4d20d341), WTCP(0x65ded74d, 0x4d80424a), WTCP(0x659617bb, 0x4ddf852d), WTCP(0x654ced55, 0x4e3e9ac1), WTCP(0x6503589b, 0x4e9d81dc), WTCP(0x64b95a0d, 0x4efc3959), WTCP(0x646ef230, 0x4f5ac010), WTCP(0x6424218d, 0x4fb914df), WTCP(0x63d8e8ae, 0x501736a1), WTCP(0x638d4822, 0x50752438), WTCP(0x6341407a, 0x50d2dc82), WTCP(0x62f4d24b, 0x51305e61), WTCP(0x62a7fe2b, 0x518da8bb), WTCP(0x625ac4b5, 0x51eaba74), WTCP(0x620d2686, 0x52479273), WTCP(0x61bf2440, 0x52a42fa2), WTCP(0x6170be85, 0x530090ea), WTCP(0x6121f5fb, 0x535cb53a), WTCP(0x60d2cb4e, 0x53b89b7e), WTCP(0x60833f28, 0x541442a8), WTCP(0x60335239, 0x546fa9a9), WTCP(0x5fe30533, 0x54cacf77), WTCP(0x5f9258cc, 0x5525b306), WTCP(0x5f414dbb, 0x55805350), WTCP(0x5eefe4bc, 0x55daaf4e), WTCP(0x5e9e1e8c, 0x5634c5fe), WTCP(0x5e4bfbec, 0x568e965c), WTCP(0x5df97d9e, 0x56e81f6c), WTCP(0x5da6a46a, 0x5741602e), WTCP(0x5d537118, 0x579a57a8), WTCP(0x5cffe474, 0x57f304e2), WTCP(0x5cabff4c, 0x584b66e4), WTCP(0x5c57c271, 0x58a37cbb), WTCP(0x5c032eb7, 0x58fb4576), WTCP(0x5bae44f4, 0x5952c024), WTCP(0x5b590602, 0x59a9ebd8), WTCP(0x5b0372bb, 0x5a00c7a8), WTCP(0x5aad8bfe, 0x5a5752ac), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_WTP KBDWindow960[] = { WTCP(0x7fffff9e, 0x0009e6ac), WTCP(0x7fffff2b, 0x000e96d5), WTCP(0x7ffffea6, 0x0012987e), WTCP(0x7ffffe0e, 0x001652b6), WTCP(0x7ffffd60, 0x0019ebce), WTCP(0x7ffffc9c, 0x001d76bf), WTCP(0x7ffffbbf, 0x0020fe79), WTCP(0x7ffffac9, 0x002489ef), WTCP(0x7ffff9b7, 0x00281de2), WTCP(0x7ffff887, 0x002bbdbb), WTCP(0x7ffff737, 0x002f6c0d), WTCP(0x7ffff5c6, 0x00332ad8), WTCP(0x7ffff431, 0x0036fbb9), WTCP(0x7ffff276, 0x003ae004), WTCP(0x7ffff092, 0x003ed8d8), WTCP(0x7fffee84, 0x0042e72f), WTCP(0x7fffec48, 0x00470be3), WTCP(0x7fffe9dd, 0x004b47b8), WTCP(0x7fffe73f, 0x004f9b5f), WTCP(0x7fffe46b, 0x0054077a), WTCP(0x7fffe15f, 0x00588ca1), WTCP(0x7fffde17, 0x005d2b61), WTCP(0x7fffda91, 0x0061e442), WTCP(0x7fffd6c9, 0x0066b7c2), WTCP(0x7fffd2bb, 0x006ba65c), WTCP(0x7fffce65, 0x0070b087), WTCP(0x7fffc9c2, 0x0075d6b5), WTCP(0x7fffc4cf, 0x007b1955), WTCP(0x7fffbf87, 0x008078d5), WTCP(0x7fffb9e7, 0x0085f5a0), WTCP(0x7fffb3ea, 0x008b901d), WTCP(0x7fffad8c, 0x009148b4), WTCP(0x7fffa6c9, 0x00971fcb), WTCP(0x7fff9f9c, 0x009d15c7), WTCP(0x7fff9800, 0x00a32b0b), WTCP(0x7fff8ff0, 0x00a95ff9), WTCP(0x7fff8767, 0x00afb4f4), WTCP(0x7fff7e5f, 0x00b62a5c), WTCP(0x7fff74d4, 0x00bcc093), WTCP(0x7fff6ac0, 0x00c377f8), WTCP(0x7fff601c, 0x00ca50eb), WTCP(0x7fff54e3, 0x00d14bcb), WTCP(0x7fff490e, 0x00d868f7), WTCP(0x7fff3c98, 0x00dfa8ce), WTCP(0x7fff2f79, 0x00e70bad), WTCP(0x7fff21ac, 0x00ee91f3), WTCP(0x7fff1328, 0x00f63bfe), WTCP(0x7fff03e7, 0x00fe0a2c), WTCP(0x7ffef3e1, 0x0105fcd9), WTCP(0x7ffee310, 0x010e1462), WTCP(0x7ffed16a, 0x01165126), WTCP(0x7ffebee9, 0x011eb381), WTCP(0x7ffeab83, 0x01273bd0), WTCP(0x7ffe9731, 0x012fea6f), WTCP(0x7ffe81ea, 0x0138bfbc), WTCP(0x7ffe6ba4, 0x0141bc12), WTCP(0x7ffe5457, 0x014adfce), WTCP(0x7ffe3bfa, 0x01542b4d), WTCP(0x7ffe2282, 0x015d9ee9), WTCP(0x7ffe07e6, 0x01673b01), WTCP(0x7ffdec1b, 0x0170ffee), WTCP(0x7ffdcf17, 0x017aee0e), WTCP(0x7ffdb0d0, 0x018505bc), WTCP(0x7ffd913b, 0x018f4754), WTCP(0x7ffd704b, 0x0199b330), WTCP(0x7ffd4df7, 0x01a449ad), WTCP(0x7ffd2a31, 0x01af0b25), WTCP(0x7ffd04ef, 0x01b9f7f4), WTCP(0x7ffcde23, 0x01c51074), WTCP(0x7ffcb5c1, 0x01d05501), WTCP(0x7ffc8bbc, 0x01dbc5f5), WTCP(0x7ffc6006, 0x01e763ab), WTCP(0x7ffc3293, 0x01f32e7d), WTCP(0x7ffc0354, 0x01ff26c5), WTCP(0x7ffbd23b, 0x020b4cde), WTCP(0x7ffb9f3a, 0x0217a120), WTCP(0x7ffb6a41, 0x022423e6), WTCP(0x7ffb3342, 0x0230d58a), WTCP(0x7ffafa2d, 0x023db664), WTCP(0x7ffabef2, 0x024ac6ce), WTCP(0x7ffa8180, 0x02580720), WTCP(0x7ffa41c9, 0x026577b3), WTCP(0x7ff9ffb9, 0x027318e0), WTCP(0x7ff9bb41, 0x0280eaff), WTCP(0x7ff9744e, 0x028eee68), WTCP(0x7ff92acf, 0x029d2371), WTCP(0x7ff8deb1, 0x02ab8a74), WTCP(0x7ff88fe2, 0x02ba23c7), WTCP(0x7ff83e4d, 0x02c8efc0), WTCP(0x7ff7e9e1, 0x02d7eeb7), WTCP(0x7ff79288, 0x02e72101), WTCP(0x7ff7382f, 0x02f686f5), WTCP(0x7ff6dac1, 0x030620e9), WTCP(0x7ff67a29, 0x0315ef31), WTCP(0x7ff61651, 0x0325f224), WTCP(0x7ff5af23, 0x03362a14), WTCP(0x7ff5448a, 0x03469758), WTCP(0x7ff4d66d, 0x03573a42), WTCP(0x7ff464b7, 0x03681327), WTCP(0x7ff3ef4f, 0x0379225a), WTCP(0x7ff3761d, 0x038a682e), WTCP(0x7ff2f90a, 0x039be4f4), WTCP(0x7ff277fb, 0x03ad9900), WTCP(0x7ff1f2d8, 0x03bf84a3), WTCP(0x7ff16986, 0x03d1a82e), WTCP(0x7ff0dbec, 0x03e403f3), WTCP(0x7ff049ef, 0x03f69840), WTCP(0x7fefb373, 0x04096568), WTCP(0x7fef185d, 0x041c6bb8), WTCP(0x7fee7890, 0x042fab81), WTCP(0x7fedd3f1, 0x04432510), WTCP(0x7fed2a61, 0x0456d8b4), WTCP(0x7fec7bc4, 0x046ac6ba), WTCP(0x7febc7fb, 0x047eef70), WTCP(0x7feb0ee8, 0x04935322), WTCP(0x7fea506b, 0x04a7f21d), WTCP(0x7fe98c65, 0x04bcccab), WTCP(0x7fe8c2b7, 0x04d1e318), WTCP(0x7fe7f33e, 0x04e735af), WTCP(0x7fe71ddb, 0x04fcc4ba), WTCP(0x7fe6426c, 0x05129081), WTCP(0x7fe560ce, 0x0528994d), WTCP(0x7fe478df, 0x053edf68), WTCP(0x7fe38a7c, 0x05556318), WTCP(0x7fe29581, 0x056c24a5), WTCP(0x7fe199ca, 0x05832455), WTCP(0x7fe09733, 0x059a626e), WTCP(0x7fdf8d95, 0x05b1df35), WTCP(0x7fde7ccb, 0x05c99aef), WTCP(0x7fdd64af, 0x05e195e0), WTCP(0x7fdc451a, 0x05f9d04b), WTCP(0x7fdb1de4, 0x06124a73), WTCP(0x7fd9eee5, 0x062b0499), WTCP(0x7fd8b7f5, 0x0643ff00), WTCP(0x7fd778ec, 0x065d39e7), WTCP(0x7fd6319e, 0x0676b58f), WTCP(0x7fd4e1e2, 0x06907237), WTCP(0x7fd3898d, 0x06aa701d), WTCP(0x7fd22873, 0x06c4af80), WTCP(0x7fd0be6a, 0x06df309c), WTCP(0x7fcf4b44, 0x06f9f3ad), WTCP(0x7fcdced4, 0x0714f8f0), WTCP(0x7fcc48ed, 0x0730409f), WTCP(0x7fcab960, 0x074bcaf5), WTCP(0x7fc91fff, 0x0767982a), WTCP(0x7fc77c9a, 0x0783a877), WTCP(0x7fc5cf02, 0x079ffc14), WTCP(0x7fc41705, 0x07bc9338), WTCP(0x7fc25474, 0x07d96e19), WTCP(0x7fc0871b, 0x07f68ced), WTCP(0x7fbeaeca, 0x0813efe7), WTCP(0x7fbccb4c, 0x0831973d), WTCP(0x7fbadc70, 0x084f8320), WTCP(0x7fb8e200, 0x086db3c3), WTCP(0x7fb6dbc8, 0x088c2957), WTCP(0x7fb4c993, 0x08aae40c), WTCP(0x7fb2ab2b, 0x08c9e412), WTCP(0x7fb0805a, 0x08e92997), WTCP(0x7fae48e9, 0x0908b4c9), WTCP(0x7fac04a0, 0x092885d6), WTCP(0x7fa9b347, 0x09489ce8), WTCP(0x7fa754a6, 0x0968fa2c), WTCP(0x7fa4e884, 0x09899dcb), WTCP(0x7fa26ea6, 0x09aa87ee), WTCP(0x7f9fe6d1, 0x09cbb8be), WTCP(0x7f9d50cc, 0x09ed3062), WTCP(0x7f9aac5a, 0x0a0eef00), WTCP(0x7f97f93f, 0x0a30f4bf), WTCP(0x7f95373e, 0x0a5341c2), WTCP(0x7f92661b, 0x0a75d62e), WTCP(0x7f8f8596, 0x0a98b224), WTCP(0x7f8c9572, 0x0abbd5c7), WTCP(0x7f89956f, 0x0adf4137), WTCP(0x7f86854d, 0x0b02f494), WTCP(0x7f8364cd, 0x0b26effd), WTCP(0x7f8033ae, 0x0b4b338f), WTCP(0x7f7cf1ae, 0x0b6fbf67), WTCP(0x7f799e8b, 0x0b9493a0), WTCP(0x7f763a03, 0x0bb9b056), WTCP(0x7f72c3d2, 0x0bdf15a2), WTCP(0x7f6f3bb5, 0x0c04c39c), WTCP(0x7f6ba168, 0x0c2aba5d), WTCP(0x7f67f4a6, 0x0c50f9fa), WTCP(0x7f643529, 0x0c77828a), WTCP(0x7f6062ac, 0x0c9e5420), WTCP(0x7f5c7ce8, 0x0cc56ed1), WTCP(0x7f588397, 0x0cecd2ae), WTCP(0x7f547670, 0x0d147fc8), WTCP(0x7f50552c, 0x0d3c7630), WTCP(0x7f4c1f83, 0x0d64b5f6), WTCP(0x7f47d52a, 0x0d8d3f26), WTCP(0x7f4375d9, 0x0db611ce), WTCP(0x7f3f0144, 0x0ddf2dfa), WTCP(0x7f3a7723, 0x0e0893b4), WTCP(0x7f35d729, 0x0e324306), WTCP(0x7f31210a, 0x0e5c3bf9), WTCP(0x7f2c547b, 0x0e867e94), WTCP(0x7f27712e, 0x0eb10add), WTCP(0x7f2276d8, 0x0edbe0da), WTCP(0x7f1d6529, 0x0f07008e), WTCP(0x7f183bd3, 0x0f3269fc), WTCP(0x7f12fa89, 0x0f5e1d27), WTCP(0x7f0da0fb, 0x0f8a1a0e), WTCP(0x7f082ed8, 0x0fb660b1), WTCP(0x7f02a3d2, 0x0fe2f10f), WTCP(0x7efcff98, 0x100fcb25), WTCP(0x7ef741d9, 0x103ceeee), WTCP(0x7ef16a42, 0x106a5c66), WTCP(0x7eeb7884, 0x10981386), WTCP(0x7ee56c4a, 0x10c61447), WTCP(0x7edf4543, 0x10f45ea0), WTCP(0x7ed9031b, 0x1122f288), WTCP(0x7ed2a57f, 0x1151cff3), WTCP(0x7ecc2c1a, 0x1180f6d5), WTCP(0x7ec59699, 0x11b06720), WTCP(0x7ebee4a6, 0x11e020c8), WTCP(0x7eb815ed, 0x121023ba), WTCP(0x7eb12a18, 0x12406fe8), WTCP(0x7eaa20d1, 0x1271053e), WTCP(0x7ea2f9c2, 0x12a1e3a9), WTCP(0x7e9bb494, 0x12d30b15), WTCP(0x7e9450f0, 0x13047b6c), WTCP(0x7e8cce7f, 0x13363497), WTCP(0x7e852ce9, 0x1368367f), WTCP(0x7e7d6bd6, 0x139a8109), WTCP(0x7e758aee, 0x13cd141b), WTCP(0x7e6d89d9, 0x13ffef99), WTCP(0x7e65683d, 0x14331368), WTCP(0x7e5d25c1, 0x14667f67), WTCP(0x7e54c20b, 0x149a3379), WTCP(0x7e4c3cc3, 0x14ce2f7c), WTCP(0x7e43958e, 0x1502734f), WTCP(0x7e3acc11, 0x1536fece), WTCP(0x7e31dff2, 0x156bd1d6), WTCP(0x7e28d0d7, 0x15a0ec41), WTCP(0x7e1f9e63, 0x15d64de9), WTCP(0x7e16483d, 0x160bf6a5), WTCP(0x7e0cce08, 0x1641e64c), WTCP(0x7e032f6a, 0x16781cb4), WTCP(0x7df96c05, 0x16ae99b2), WTCP(0x7def837e, 0x16e55d18), WTCP(0x7de57579, 0x171c66ba), WTCP(0x7ddb419a, 0x1753b667), WTCP(0x7dd0e784, 0x178b4bef), WTCP(0x7dc666d9, 0x17c32721), WTCP(0x7dbbbf3e, 0x17fb47ca), WTCP(0x7db0f056, 0x1833adb5), WTCP(0x7da5f9c3, 0x186c58ae), WTCP(0x7d9adb29, 0x18a5487d), WTCP(0x7d8f9429, 0x18de7cec), WTCP(0x7d842467, 0x1917f5c1), WTCP(0x7d788b86, 0x1951b2c2), WTCP(0x7d6cc927, 0x198bb3b4), WTCP(0x7d60dced, 0x19c5f85a), WTCP(0x7d54c67c, 0x1a008077), WTCP(0x7d488574, 0x1a3b4bcb), WTCP(0x7d3c1979, 0x1a765a17), WTCP(0x7d2f822d, 0x1ab1ab18), WTCP(0x7d22bf32, 0x1aed3e8d), WTCP(0x7d15d02b, 0x1b291432), WTCP(0x7d08b4ba, 0x1b652bc1), WTCP(0x7cfb6c82, 0x1ba184f5), WTCP(0x7cedf725, 0x1bde1f86), WTCP(0x7ce05445, 0x1c1afb2c), WTCP(0x7cd28386, 0x1c58179c), WTCP(0x7cc48489, 0x1c95748d), WTCP(0x7cb656f3, 0x1cd311b1), WTCP(0x7ca7fa65, 0x1d10eebd), WTCP(0x7c996e83, 0x1d4f0b60), WTCP(0x7c8ab2f0, 0x1d8d674c), WTCP(0x7c7bc74f, 0x1dcc0230), WTCP(0x7c6cab44, 0x1e0adbbb), WTCP(0x7c5d5e71, 0x1e49f398), WTCP(0x7c4de07c, 0x1e894973), WTCP(0x7c3e3108, 0x1ec8dcf8), WTCP(0x7c2e4fb9, 0x1f08add0), WTCP(0x7c1e3c34, 0x1f48bba3), WTCP(0x7c0df61d, 0x1f890618), WTCP(0x7bfd7d18, 0x1fc98cd6), WTCP(0x7becd0cc, 0x200a4f80), WTCP(0x7bdbf0dd, 0x204b4dbc), WTCP(0x7bcadcf1, 0x208c872c), WTCP(0x7bb994ae, 0x20cdfb71), WTCP(0x7ba817b9, 0x210faa2c), WTCP(0x7b9665bb, 0x215192fc), WTCP(0x7b847e58, 0x2193b57f), WTCP(0x7b726139, 0x21d61153), WTCP(0x7b600e05, 0x2218a614), WTCP(0x7b4d8463, 0x225b735d), WTCP(0x7b3ac3fc, 0x229e78c7), WTCP(0x7b27cc79, 0x22e1b5eb), WTCP(0x7b149d82, 0x23252a62), WTCP(0x7b0136c1, 0x2368d5c2), WTCP(0x7aed97df, 0x23acb7a0), WTCP(0x7ad9c087, 0x23f0cf92), WTCP(0x7ac5b063, 0x24351d2a), WTCP(0x7ab1671e, 0x24799ffc), WTCP(0x7a9ce464, 0x24be5799), WTCP(0x7a8827e1, 0x25034391), WTCP(0x7a733142, 0x25486375), WTCP(0x7a5e0033, 0x258db6d2), WTCP(0x7a489461, 0x25d33d35), WTCP(0x7a32ed7c, 0x2618f62c), WTCP(0x7a1d0b31, 0x265ee143), WTCP(0x7a06ed2f, 0x26a4fe02), WTCP(0x79f09327, 0x26eb4bf5), WTCP(0x79d9fcc8, 0x2731caa3), WTCP(0x79c329c2, 0x27787995), WTCP(0x79ac19c9, 0x27bf5850), WTCP(0x7994cc8d, 0x2806665c), WTCP(0x797d41c1, 0x284da33c), WTCP(0x79657918, 0x28950e74), WTCP(0x794d7247, 0x28dca788), WTCP(0x79352d01, 0x29246dfa), WTCP(0x791ca8fc, 0x296c614a), WTCP(0x7903e5ee, 0x29b480f9), WTCP(0x78eae38d, 0x29fccc87), WTCP(0x78d1a191, 0x2a454372), WTCP(0x78b81fb1, 0x2a8de537), WTCP(0x789e5da6, 0x2ad6b155), WTCP(0x78845b29, 0x2b1fa745), WTCP(0x786a17f5, 0x2b68c684), WTCP(0x784f93c4, 0x2bb20e8c), WTCP(0x7834ce53, 0x2bfb7ed7), WTCP(0x7819c75c, 0x2c4516dc), WTCP(0x77fe7e9e, 0x2c8ed615), WTCP(0x77e2f3d7, 0x2cd8bbf7), WTCP(0x77c726c5, 0x2d22c7fa), WTCP(0x77ab1728, 0x2d6cf993), WTCP(0x778ec4c0, 0x2db75037), WTCP(0x77722f4e, 0x2e01cb59), WTCP(0x77555695, 0x2e4c6a6d), WTCP(0x77383a58, 0x2e972ce6), WTCP(0x771ada5a, 0x2ee21235), WTCP(0x76fd3660, 0x2f2d19cc), WTCP(0x76df4e30, 0x2f78431a), WTCP(0x76c12190, 0x2fc38d91), WTCP(0x76a2b047, 0x300ef89d), WTCP(0x7683fa1e, 0x305a83af), WTCP(0x7664fede, 0x30a62e34), WTCP(0x7645be51, 0x30f1f798), WTCP(0x76263842, 0x313ddf49), WTCP(0x76066c7e, 0x3189e4b1), WTCP(0x75e65ad1, 0x31d6073d), WTCP(0x75c60309, 0x32224657), WTCP(0x75a564f6, 0x326ea168), WTCP(0x75848067, 0x32bb17da), WTCP(0x7563552d, 0x3307a917), WTCP(0x7541e31a, 0x33545486), WTCP(0x75202a02, 0x33a1198e), WTCP(0x74fe29b8, 0x33edf798), WTCP(0x74dbe211, 0x343aee09), WTCP(0x74b952e3, 0x3487fc48), WTCP(0x74967c06, 0x34d521bb), WTCP(0x74735d51, 0x35225dc7), WTCP(0x744ff69f, 0x356fafcf), WTCP(0x742c47c9, 0x35bd173a), WTCP(0x740850ab, 0x360a9369), WTCP(0x73e41121, 0x365823c1), WTCP(0x73bf8909, 0x36a5c7a4), WTCP(0x739ab842, 0x36f37e75), WTCP(0x73759eab, 0x37414796), WTCP(0x73503c26, 0x378f2268), WTCP(0x732a9095, 0x37dd0e4c), WTCP(0x73049bda, 0x382b0aa4), WTCP(0x72de5ddb, 0x387916d0), WTCP(0x72b7d67d, 0x38c73230), WTCP(0x729105a6, 0x39155c24), WTCP(0x7269eb3f, 0x3963940c), WTCP(0x72428730, 0x39b1d946), WTCP(0x721ad964, 0x3a002b31), WTCP(0x71f2e1c5, 0x3a4e892c), WTCP(0x71caa042, 0x3a9cf296), WTCP(0x71a214c7, 0x3aeb66cc), WTCP(0x71793f43, 0x3b39e52c), WTCP(0x71501fa6, 0x3b886d14), WTCP(0x7126b5e3, 0x3bd6fde1), WTCP(0x70fd01eb, 0x3c2596f1), WTCP(0x70d303b2, 0x3c74379f), WTCP(0x70a8bb2e, 0x3cc2df49), WTCP(0x707e2855, 0x3d118d4c), WTCP(0x70534b1e, 0x3d604103), WTCP(0x70282381, 0x3daef9cc), WTCP(0x6ffcb17a, 0x3dfdb702), WTCP(0x6fd0f504, 0x3e4c7800), WTCP(0x6fa4ee1a, 0x3e9b3c25), WTCP(0x6f789cbb, 0x3eea02ca), WTCP(0x6f4c00e5, 0x3f38cb4b), WTCP(0x6f1f1a9a, 0x3f879505), WTCP(0x6ef1e9da, 0x3fd65f53), WTCP(0x6ec46ea9, 0x40252990), WTCP(0x6e96a90b, 0x4073f318), WTCP(0x6e689905, 0x40c2bb46), WTCP(0x6e3a3e9d, 0x41118176), WTCP(0x6e0b99dd, 0x41604504), WTCP(0x6ddcaacc, 0x41af054a), WTCP(0x6dad7177, 0x41fdc1a5), WTCP(0x6d7dede8, 0x424c7970), WTCP(0x6d4e202e, 0x429b2c06), WTCP(0x6d1e0855, 0x42e9d8c4), WTCP(0x6ceda66f, 0x43387f05), WTCP(0x6cbcfa8d, 0x43871e26), WTCP(0x6c8c04c0, 0x43d5b581), WTCP(0x6c5ac51d, 0x44244474), WTCP(0x6c293bb8, 0x4472ca5a), WTCP(0x6bf768a8, 0x44c14690), WTCP(0x6bc54c06, 0x450fb873), WTCP(0x6b92e5e9, 0x455e1f5f), WTCP(0x6b60366c, 0x45ac7ab2), WTCP(0x6b2d3dab, 0x45fac9c8), WTCP(0x6af9fbc2, 0x46490bff), WTCP(0x6ac670d1, 0x469740b5), WTCP(0x6a929cf6, 0x46e56747), WTCP(0x6a5e8053, 0x47337f13), WTCP(0x6a2a1b0a, 0x47818779), WTCP(0x69f56d3e, 0x47cf7fd6), WTCP(0x69c07715, 0x481d678a), WTCP(0x698b38b4, 0x486b3df3), WTCP(0x6955b243, 0x48b90272), WTCP(0x691fe3ec, 0x4906b466), WTCP(0x68e9cdd8, 0x49545330), WTCP(0x68b37033, 0x49a1de30), WTCP(0x687ccb29, 0x49ef54c8), WTCP(0x6845dee9, 0x4a3cb657), WTCP(0x680eaba3, 0x4a8a0242), WTCP(0x67d73187, 0x4ad737e9), WTCP(0x679f70c7, 0x4b2456af), WTCP(0x67676997, 0x4b715df7), WTCP(0x672f1c2b, 0x4bbe4d25), WTCP(0x66f688ba, 0x4c0b239c), WTCP(0x66bdaf7b, 0x4c57e0c2), WTCP(0x668490a6, 0x4ca483fa), WTCP(0x664b2c76, 0x4cf10cac), WTCP(0x66118326, 0x4d3d7a3b), WTCP(0x65d794f3, 0x4d89cc0f), WTCP(0x659d621a, 0x4dd6018f), WTCP(0x6562eada, 0x4e221a22), WTCP(0x65282f74, 0x4e6e1530), WTCP(0x64ed302b, 0x4eb9f222), WTCP(0x64b1ed40, 0x4f05b061), WTCP(0x647666f8, 0x4f514f57), WTCP(0x643a9d99, 0x4f9cce6f), WTCP(0x63fe916a, 0x4fe82d13), WTCP(0x63c242b2, 0x50336aaf), WTCP(0x6385b1bc, 0x507e86b0), WTCP(0x6348ded1, 0x50c98082), WTCP(0x630bca3f, 0x51145793), WTCP(0x62ce7451, 0x515f0b51), WTCP(0x6290dd57, 0x51a99b2b), WTCP(0x625305a0, 0x51f40692), WTCP(0x6214ed7d, 0x523e4cf5), WTCP(0x61d69541, 0x52886dc5), WTCP(0x6197fd3e, 0x52d26875), WTCP(0x615925c9, 0x531c3c77), WTCP(0x611a0f39, 0x5365e93e), WTCP(0x60dab9e3, 0x53af6e3e), WTCP(0x609b2621, 0x53f8caed), WTCP(0x605b544c, 0x5441fec0), WTCP(0x601b44bf, 0x548b092e), WTCP(0x5fdaf7d5, 0x54d3e9ae), WTCP(0x5f9a6deb, 0x551c9fb7), WTCP(0x5f59a761, 0x55652ac3), WTCP(0x5f18a494, 0x55ad8a4d), WTCP(0x5ed765e6, 0x55f5bdcd), WTCP(0x5e95ebb8, 0x563dc4c1), WTCP(0x5e54366d, 0x56859ea3), WTCP(0x5e12466a, 0x56cd4af3), WTCP(0x5dd01c13, 0x5714c92d), WTCP(0x5d8db7cf, 0x575c18d0), WTCP(0x5d4b1a05, 0x57a3395e), WTCP(0x5d08431e, 0x57ea2a56), WTCP(0x5cc53384, 0x5830eb3a), WTCP(0x5c81eba0, 0x58777b8e), WTCP(0x5c3e6bdf, 0x58bddad5), WTCP(0x5bfab4af, 0x59040893), WTCP(0x5bb6c67c, 0x594a044f), WTCP(0x5b72a1b6, 0x598fcd8e), WTCP(0x5b2e46ce, 0x59d563d9), WTCP(0x5ae9b634, 0x5a1ac6b8), WTCP(0x5aa4f05a, 0x5a5ff5b5), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_WTP KBDWindow1024[] = { WTCP(0x7fffffa4, 0x0009962f), WTCP(0x7fffff39, 0x000e16fb), WTCP(0x7ffffebf, 0x0011ea65), WTCP(0x7ffffe34, 0x0015750e), WTCP(0x7ffffd96, 0x0018dc74), WTCP(0x7ffffce5, 0x001c332e), WTCP(0x7ffffc1f, 0x001f83f5), WTCP(0x7ffffb43, 0x0022d59a), WTCP(0x7ffffa4f, 0x00262cc2), WTCP(0x7ffff942, 0x00298cc4), WTCP(0x7ffff81a, 0x002cf81f), WTCP(0x7ffff6d6, 0x003070c4), WTCP(0x7ffff573, 0x0033f840), WTCP(0x7ffff3f1, 0x00378fd9), WTCP(0x7ffff24d, 0x003b38a1), WTCP(0x7ffff085, 0x003ef381), WTCP(0x7fffee98, 0x0042c147), WTCP(0x7fffec83, 0x0046a2a8), WTCP(0x7fffea44, 0x004a9847), WTCP(0x7fffe7d8, 0x004ea2b7), WTCP(0x7fffe53f, 0x0052c283), WTCP(0x7fffe274, 0x0056f829), WTCP(0x7fffdf76, 0x005b4422), WTCP(0x7fffdc43, 0x005fa6dd), WTCP(0x7fffd8d6, 0x006420c8), WTCP(0x7fffd52f, 0x0068b249), WTCP(0x7fffd149, 0x006d5bc4), WTCP(0x7fffcd22, 0x00721d9a), WTCP(0x7fffc8b6, 0x0076f828), WTCP(0x7fffc404, 0x007bebca), WTCP(0x7fffbf06, 0x0080f8d9), WTCP(0x7fffb9bb, 0x00861fae), WTCP(0x7fffb41e, 0x008b609e), WTCP(0x7fffae2c, 0x0090bbff), WTCP(0x7fffa7e1, 0x00963224), WTCP(0x7fffa13a, 0x009bc362), WTCP(0x7fff9a32, 0x00a17009), WTCP(0x7fff92c5, 0x00a7386c), WTCP(0x7fff8af0, 0x00ad1cdc), WTCP(0x7fff82ad, 0x00b31da8), WTCP(0x7fff79f9, 0x00b93b21), WTCP(0x7fff70cf, 0x00bf7596), WTCP(0x7fff672a, 0x00c5cd57), WTCP(0x7fff5d05, 0x00cc42b1), WTCP(0x7fff525c, 0x00d2d5f3), WTCP(0x7fff4729, 0x00d9876c), WTCP(0x7fff3b66, 0x00e05769), WTCP(0x7fff2f10, 0x00e74638), WTCP(0x7fff221f, 0x00ee5426), WTCP(0x7fff148e, 0x00f58182), WTCP(0x7fff0658, 0x00fcce97), WTCP(0x7ffef776, 0x01043bb3), WTCP(0x7ffee7e2, 0x010bc923), WTCP(0x7ffed795, 0x01137733), WTCP(0x7ffec68a, 0x011b4631), WTCP(0x7ffeb4ba, 0x01233669), WTCP(0x7ffea21d, 0x012b4827), WTCP(0x7ffe8eac, 0x01337bb8), WTCP(0x7ffe7a61, 0x013bd167), WTCP(0x7ffe6533, 0x01444982), WTCP(0x7ffe4f1c, 0x014ce454), WTCP(0x7ffe3813, 0x0155a229), WTCP(0x7ffe2011, 0x015e834d), WTCP(0x7ffe070d, 0x0167880c), WTCP(0x7ffdecff, 0x0170b0b2), WTCP(0x7ffdd1df, 0x0179fd8b), WTCP(0x7ffdb5a2, 0x01836ee1), WTCP(0x7ffd9842, 0x018d0500), WTCP(0x7ffd79b3, 0x0196c035), WTCP(0x7ffd59ee, 0x01a0a0ca), WTCP(0x7ffd38e8, 0x01aaa70a), WTCP(0x7ffd1697, 0x01b4d341), WTCP(0x7ffcf2f2, 0x01bf25b9), WTCP(0x7ffccdee, 0x01c99ebd), WTCP(0x7ffca780, 0x01d43e99), WTCP(0x7ffc7f9e, 0x01df0597), WTCP(0x7ffc563d, 0x01e9f401), WTCP(0x7ffc2b51, 0x01f50a22), WTCP(0x7ffbfecf, 0x02004844), WTCP(0x7ffbd0ab, 0x020baeb1), WTCP(0x7ffba0da, 0x02173db4), WTCP(0x7ffb6f4f, 0x0222f596), WTCP(0x7ffb3bfd, 0x022ed6a1), WTCP(0x7ffb06d8, 0x023ae11f), WTCP(0x7ffacfd3, 0x02471558), WTCP(0x7ffa96e0, 0x02537397), WTCP(0x7ffa5bf2, 0x025ffc25), WTCP(0x7ffa1efc, 0x026caf4a), WTCP(0x7ff9dfee, 0x02798d4f), WTCP(0x7ff99ebb, 0x0286967c), WTCP(0x7ff95b55, 0x0293cb1b), WTCP(0x7ff915ab, 0x02a12b72), WTCP(0x7ff8cdaf, 0x02aeb7cb), WTCP(0x7ff88351, 0x02bc706d), WTCP(0x7ff83682, 0x02ca559f), WTCP(0x7ff7e731, 0x02d867a9), WTCP(0x7ff7954e, 0x02e6a6d2), WTCP(0x7ff740c8, 0x02f51361), WTCP(0x7ff6e98e, 0x0303ad9c), WTCP(0x7ff68f8f, 0x031275ca), WTCP(0x7ff632ba, 0x03216c30), WTCP(0x7ff5d2fb, 0x03309116), WTCP(0x7ff57042, 0x033fe4bf), WTCP(0x7ff50a7a, 0x034f6773), WTCP(0x7ff4a192, 0x035f1975), WTCP(0x7ff43576, 0x036efb0a), WTCP(0x7ff3c612, 0x037f0c78), WTCP(0x7ff35353, 0x038f4e02), WTCP(0x7ff2dd24, 0x039fbfeb), WTCP(0x7ff26370, 0x03b06279), WTCP(0x7ff1e623, 0x03c135ed), WTCP(0x7ff16527, 0x03d23a8b), WTCP(0x7ff0e067, 0x03e37095), WTCP(0x7ff057cc, 0x03f4d84e), WTCP(0x7fefcb40, 0x040671f7), WTCP(0x7fef3aad, 0x04183dd3), WTCP(0x7feea5fa, 0x042a3c22), WTCP(0x7fee0d11, 0x043c6d25), WTCP(0x7fed6fda, 0x044ed11d), WTCP(0x7fecce3d, 0x04616849), WTCP(0x7fec2821, 0x047432eb), WTCP(0x7feb7d6c, 0x04873140), WTCP(0x7feace07, 0x049a6388), WTCP(0x7fea19d6, 0x04adca01), WTCP(0x7fe960c0, 0x04c164ea), WTCP(0x7fe8a2aa, 0x04d53481), WTCP(0x7fe7df79, 0x04e93902), WTCP(0x7fe71712, 0x04fd72aa), WTCP(0x7fe6495a, 0x0511e1b6), WTCP(0x7fe57634, 0x05268663), WTCP(0x7fe49d83, 0x053b60eb), WTCP(0x7fe3bf2b, 0x05507189), WTCP(0x7fe2db0f, 0x0565b879), WTCP(0x7fe1f110, 0x057b35f4), WTCP(0x7fe10111, 0x0590ea35), WTCP(0x7fe00af3, 0x05a6d574), WTCP(0x7fdf0e97, 0x05bcf7ea), WTCP(0x7fde0bdd, 0x05d351cf), WTCP(0x7fdd02a6, 0x05e9e35c), WTCP(0x7fdbf2d2, 0x0600acc8), WTCP(0x7fdadc40, 0x0617ae48), WTCP(0x7fd9becf, 0x062ee814), WTCP(0x7fd89a5e, 0x06465a62), WTCP(0x7fd76eca, 0x065e0565), WTCP(0x7fd63bf1, 0x0675e954), WTCP(0x7fd501b0, 0x068e0662), WTCP(0x7fd3bfe4, 0x06a65cc3), WTCP(0x7fd2766a, 0x06beecaa), WTCP(0x7fd1251e, 0x06d7b648), WTCP(0x7fcfcbda, 0x06f0b9d1), WTCP(0x7fce6a7a, 0x0709f775), WTCP(0x7fcd00d8, 0x07236f65), WTCP(0x7fcb8ecf, 0x073d21d2), WTCP(0x7fca1439, 0x07570eea), WTCP(0x7fc890ed, 0x077136dd), WTCP(0x7fc704c7, 0x078b99da), WTCP(0x7fc56f9d, 0x07a6380d), WTCP(0x7fc3d147, 0x07c111a4), WTCP(0x7fc2299e, 0x07dc26cc), WTCP(0x7fc07878, 0x07f777b1), WTCP(0x7fbebdac, 0x0813047d), WTCP(0x7fbcf90f, 0x082ecd5b), WTCP(0x7fbb2a78, 0x084ad276), WTCP(0x7fb951bc, 0x086713f7), WTCP(0x7fb76eaf, 0x08839206), WTCP(0x7fb58126, 0x08a04ccb), WTCP(0x7fb388f4, 0x08bd446e), WTCP(0x7fb185ee, 0x08da7915), WTCP(0x7faf77e5, 0x08f7eae7), WTCP(0x7fad5ead, 0x09159a09), WTCP(0x7fab3a17, 0x0933869f), WTCP(0x7fa909f6, 0x0951b0cd), WTCP(0x7fa6ce1a, 0x097018b7), WTCP(0x7fa48653, 0x098ebe7f), WTCP(0x7fa23273, 0x09ada248), WTCP(0x7f9fd249, 0x09ccc431), WTCP(0x7f9d65a4, 0x09ec245b), WTCP(0x7f9aec53, 0x0a0bc2e7), WTCP(0x7f986625, 0x0a2b9ff3), WTCP(0x7f95d2e7, 0x0a4bbb9e), WTCP(0x7f933267, 0x0a6c1604), WTCP(0x7f908472, 0x0a8caf43), WTCP(0x7f8dc8d5, 0x0aad8776), WTCP(0x7f8aff5c, 0x0ace9eb9), WTCP(0x7f8827d3, 0x0aeff526), WTCP(0x7f854204, 0x0b118ad8), WTCP(0x7f824dbb, 0x0b335fe6), WTCP(0x7f7f4ac3, 0x0b557469), WTCP(0x7f7c38e4, 0x0b77c879), WTCP(0x7f7917e9, 0x0b9a5c2b), WTCP(0x7f75e79b, 0x0bbd2f97), WTCP(0x7f72a7c3, 0x0be042d0), WTCP(0x7f6f5828, 0x0c0395ec), WTCP(0x7f6bf892, 0x0c2728fd), WTCP(0x7f6888c9, 0x0c4afc16), WTCP(0x7f650894, 0x0c6f0f4a), WTCP(0x7f6177b9, 0x0c9362a8), WTCP(0x7f5dd5ff, 0x0cb7f642), WTCP(0x7f5a232a, 0x0cdcca26), WTCP(0x7f565f00, 0x0d01de63), WTCP(0x7f528947, 0x0d273307), WTCP(0x7f4ea1c2, 0x0d4cc81f), WTCP(0x7f4aa835, 0x0d729db7), WTCP(0x7f469c65, 0x0d98b3da), WTCP(0x7f427e13, 0x0dbf0a92), WTCP(0x7f3e4d04, 0x0de5a1e9), WTCP(0x7f3a08f9, 0x0e0c79e7), WTCP(0x7f35b1b4, 0x0e339295), WTCP(0x7f3146f8, 0x0e5aebfa), WTCP(0x7f2cc884, 0x0e82861a), WTCP(0x7f28361b, 0x0eaa60fd), WTCP(0x7f238f7c, 0x0ed27ca5), WTCP(0x7f1ed467, 0x0efad917), WTCP(0x7f1a049d, 0x0f237656), WTCP(0x7f151fdc, 0x0f4c5462), WTCP(0x7f1025e3, 0x0f75733d), WTCP(0x7f0b1672, 0x0f9ed2e6), WTCP(0x7f05f146, 0x0fc8735e), WTCP(0x7f00b61d, 0x0ff254a1), WTCP(0x7efb64b4, 0x101c76ae), WTCP(0x7ef5fcca, 0x1046d981), WTCP(0x7ef07e19, 0x10717d15), WTCP(0x7eeae860, 0x109c6165), WTCP(0x7ee53b5b, 0x10c7866a), WTCP(0x7edf76c4, 0x10f2ec1e), WTCP(0x7ed99a58, 0x111e9279), WTCP(0x7ed3a5d1, 0x114a7971), WTCP(0x7ecd98eb, 0x1176a0fc), WTCP(0x7ec77360, 0x11a30910), WTCP(0x7ec134eb, 0x11cfb1a1), WTCP(0x7ebadd44, 0x11fc9aa2), WTCP(0x7eb46c27, 0x1229c406), WTCP(0x7eade14c, 0x12572dbf), WTCP(0x7ea73c6c, 0x1284d7bc), WTCP(0x7ea07d41, 0x12b2c1ed), WTCP(0x7e99a382, 0x12e0ec42), WTCP(0x7e92aee7, 0x130f56a8), WTCP(0x7e8b9f2a, 0x133e010b), WTCP(0x7e847402, 0x136ceb59), WTCP(0x7e7d2d25, 0x139c157b), WTCP(0x7e75ca4c, 0x13cb7f5d), WTCP(0x7e6e4b2d, 0x13fb28e6), WTCP(0x7e66af7f, 0x142b1200), WTCP(0x7e5ef6f8, 0x145b3a92), WTCP(0x7e572150, 0x148ba281), WTCP(0x7e4f2e3b, 0x14bc49b4), WTCP(0x7e471d70, 0x14ed300f), WTCP(0x7e3eeea5, 0x151e5575), WTCP(0x7e36a18e, 0x154fb9c9), WTCP(0x7e2e35e2, 0x15815ced), WTCP(0x7e25ab56, 0x15b33ec1), WTCP(0x7e1d019e, 0x15e55f25), WTCP(0x7e14386e, 0x1617bdf9), WTCP(0x7e0b4f7d, 0x164a5b19), WTCP(0x7e02467e, 0x167d3662), WTCP(0x7df91d25, 0x16b04fb2), WTCP(0x7defd327, 0x16e3a6e2), WTCP(0x7de66837, 0x17173bce), WTCP(0x7ddcdc0a, 0x174b0e4d), WTCP(0x7dd32e53, 0x177f1e39), WTCP(0x7dc95ec6, 0x17b36b69), WTCP(0x7dbf6d17, 0x17e7f5b3), WTCP(0x7db558f9, 0x181cbcec), WTCP(0x7dab221f, 0x1851c0e9), WTCP(0x7da0c83c, 0x1887017d), WTCP(0x7d964b05, 0x18bc7e7c), WTCP(0x7d8baa2b, 0x18f237b6), WTCP(0x7d80e563, 0x19282cfd), WTCP(0x7d75fc5e, 0x195e5e20), WTCP(0x7d6aeed0, 0x1994caee), WTCP(0x7d5fbc6d, 0x19cb7335), WTCP(0x7d5464e6, 0x1a0256c2), WTCP(0x7d48e7ef, 0x1a397561), WTCP(0x7d3d453b, 0x1a70cede), WTCP(0x7d317c7c, 0x1aa86301), WTCP(0x7d258d65, 0x1ae03195), WTCP(0x7d1977aa, 0x1b183a63), WTCP(0x7d0d3afc, 0x1b507d30), WTCP(0x7d00d710, 0x1b88f9c5), WTCP(0x7cf44b97, 0x1bc1afe6), WTCP(0x7ce79846, 0x1bfa9f58), WTCP(0x7cdabcce, 0x1c33c7e0), WTCP(0x7ccdb8e4, 0x1c6d293f), WTCP(0x7cc08c39, 0x1ca6c337), WTCP(0x7cb33682, 0x1ce0958a), WTCP(0x7ca5b772, 0x1d1a9ff8), WTCP(0x7c980ebd, 0x1d54e240), WTCP(0x7c8a3c14, 0x1d8f5c21), WTCP(0x7c7c3f2e, 0x1dca0d56), WTCP(0x7c6e17bc, 0x1e04f59f), WTCP(0x7c5fc573, 0x1e4014b4), WTCP(0x7c514807, 0x1e7b6a53), WTCP(0x7c429f2c, 0x1eb6f633), WTCP(0x7c33ca96, 0x1ef2b80f), WTCP(0x7c24c9fa, 0x1f2eaf9e), WTCP(0x7c159d0d, 0x1f6adc98), WTCP(0x7c064383, 0x1fa73eb2), WTCP(0x7bf6bd11, 0x1fe3d5a3), WTCP(0x7be7096c, 0x2020a11e), WTCP(0x7bd7284a, 0x205da0d8), WTCP(0x7bc71960, 0x209ad483), WTCP(0x7bb6dc65, 0x20d83bd1), WTCP(0x7ba6710d, 0x2115d674), WTCP(0x7b95d710, 0x2153a41b), WTCP(0x7b850e24, 0x2191a476), WTCP(0x7b7415ff, 0x21cfd734), WTCP(0x7b62ee59, 0x220e3c02), WTCP(0x7b5196e9, 0x224cd28d), WTCP(0x7b400f67, 0x228b9a82), WTCP(0x7b2e578a, 0x22ca938a), WTCP(0x7b1c6f0b, 0x2309bd52), WTCP(0x7b0a55a1, 0x23491783), WTCP(0x7af80b07, 0x2388a1c4), WTCP(0x7ae58ef5, 0x23c85bbf), WTCP(0x7ad2e124, 0x2408451a), WTCP(0x7ac0014e, 0x24485d7c), WTCP(0x7aacef2e, 0x2488a48a), WTCP(0x7a99aa7e, 0x24c919e9), WTCP(0x7a8632f8, 0x2509bd3d), WTCP(0x7a728858, 0x254a8e29), WTCP(0x7a5eaa5a, 0x258b8c50), WTCP(0x7a4a98b9, 0x25ccb753), WTCP(0x7a365333, 0x260e0ed3), WTCP(0x7a21d983, 0x264f9271), WTCP(0x7a0d2b68, 0x269141cb), WTCP(0x79f8489e, 0x26d31c80), WTCP(0x79e330e4, 0x2715222f), WTCP(0x79cde3f8, 0x27575273), WTCP(0x79b8619a, 0x2799acea), WTCP(0x79a2a989, 0x27dc3130), WTCP(0x798cbb85, 0x281ededf), WTCP(0x7976974e, 0x2861b591), WTCP(0x79603ca5, 0x28a4b4e0), WTCP(0x7949ab4c, 0x28e7dc65), WTCP(0x7932e304, 0x292b2bb8), WTCP(0x791be390, 0x296ea270), WTCP(0x7904acb3, 0x29b24024), WTCP(0x78ed3e30, 0x29f6046b), WTCP(0x78d597cc, 0x2a39eed8), WTCP(0x78bdb94a, 0x2a7dff02), WTCP(0x78a5a270, 0x2ac2347c), WTCP(0x788d5304, 0x2b068eda), WTCP(0x7874cacb, 0x2b4b0dae), WTCP(0x785c098d, 0x2b8fb08a), WTCP(0x78430f11, 0x2bd47700), WTCP(0x7829db1f, 0x2c1960a1), WTCP(0x78106d7f, 0x2c5e6cfd), WTCP(0x77f6c5fb, 0x2ca39ba3), WTCP(0x77dce45c, 0x2ce8ec23), WTCP(0x77c2c86e, 0x2d2e5e0b), WTCP(0x77a871fa, 0x2d73f0e8), WTCP(0x778de0cd, 0x2db9a449), WTCP(0x777314b2, 0x2dff77b8), WTCP(0x77580d78, 0x2e456ac4), WTCP(0x773ccaeb, 0x2e8b7cf6), WTCP(0x77214cdb, 0x2ed1addb), WTCP(0x77059315, 0x2f17fcfb), WTCP(0x76e99d69, 0x2f5e69e2), WTCP(0x76cd6ba9, 0x2fa4f419), WTCP(0x76b0fda4, 0x2feb9b27), WTCP(0x7694532e, 0x30325e96), WTCP(0x76776c17, 0x30793dee), WTCP(0x765a4834, 0x30c038b5), WTCP(0x763ce759, 0x31074e72), WTCP(0x761f4959, 0x314e7eab), WTCP(0x76016e0b, 0x3195c8e6), WTCP(0x75e35545, 0x31dd2ca9), WTCP(0x75c4fedc, 0x3224a979), WTCP(0x75a66aab, 0x326c3ed8), WTCP(0x75879887, 0x32b3ec4d), WTCP(0x7568884b, 0x32fbb159), WTCP(0x754939d1, 0x33438d81), WTCP(0x7529acf4, 0x338b8045), WTCP(0x7509e18e, 0x33d3892a), WTCP(0x74e9d77d, 0x341ba7b1), WTCP(0x74c98e9e, 0x3463db5a), WTCP(0x74a906cd, 0x34ac23a7), WTCP(0x74883fec, 0x34f48019), WTCP(0x746739d8, 0x353cf02f), WTCP(0x7445f472, 0x3585736a), WTCP(0x74246f9c, 0x35ce0949), WTCP(0x7402ab37, 0x3616b14c), WTCP(0x73e0a727, 0x365f6af0), WTCP(0x73be6350, 0x36a835b5), WTCP(0x739bdf95, 0x36f11118), WTCP(0x73791bdd, 0x3739fc98), WTCP(0x7356180e, 0x3782f7b2), WTCP(0x7332d410, 0x37cc01e3), WTCP(0x730f4fc9, 0x38151aa8), WTCP(0x72eb8b24, 0x385e417e), WTCP(0x72c7860a, 0x38a775e1), WTCP(0x72a34066, 0x38f0b74d), WTCP(0x727eba24, 0x393a053e), WTCP(0x7259f331, 0x39835f30), WTCP(0x7234eb79, 0x39ccc49e), WTCP(0x720fa2eb, 0x3a163503), WTCP(0x71ea1977, 0x3a5fafda), WTCP(0x71c44f0c, 0x3aa9349e), WTCP(0x719e439d, 0x3af2c2ca), WTCP(0x7177f71a, 0x3b3c59d7), WTCP(0x71516978, 0x3b85f940), WTCP(0x712a9aaa, 0x3bcfa07e), WTCP(0x71038aa4, 0x3c194f0d), WTCP(0x70dc395e, 0x3c630464), WTCP(0x70b4a6cd, 0x3cacbfff), WTCP(0x708cd2e9, 0x3cf68155), WTCP(0x7064bdab, 0x3d4047e1), WTCP(0x703c670d, 0x3d8a131c), WTCP(0x7013cf0a, 0x3dd3e27e), WTCP(0x6feaf59c, 0x3e1db580), WTCP(0x6fc1dac1, 0x3e678b9b), WTCP(0x6f987e76, 0x3eb16449), WTCP(0x6f6ee0b9, 0x3efb3f01), WTCP(0x6f45018b, 0x3f451b3d), WTCP(0x6f1ae0eb, 0x3f8ef874), WTCP(0x6ef07edb, 0x3fd8d620), WTCP(0x6ec5db5d, 0x4022b3b9), WTCP(0x6e9af675, 0x406c90b7), WTCP(0x6e6fd027, 0x40b66c93), WTCP(0x6e446879, 0x410046c5), WTCP(0x6e18bf71, 0x414a1ec6), WTCP(0x6decd517, 0x4193f40d), WTCP(0x6dc0a972, 0x41ddc615), WTCP(0x6d943c8d, 0x42279455), WTCP(0x6d678e71, 0x42715e45), WTCP(0x6d3a9f2a, 0x42bb235f), WTCP(0x6d0d6ec5, 0x4304e31a), WTCP(0x6cdffd4f, 0x434e9cf1), WTCP(0x6cb24ad6, 0x4398505b), WTCP(0x6c84576b, 0x43e1fcd1), WTCP(0x6c56231c, 0x442ba1cd), WTCP(0x6c27adfd, 0x44753ec7), WTCP(0x6bf8f81e, 0x44bed33a), WTCP(0x6bca0195, 0x45085e9d), WTCP(0x6b9aca75, 0x4551e06b), WTCP(0x6b6b52d5, 0x459b581e), WTCP(0x6b3b9ac9, 0x45e4c52f), WTCP(0x6b0ba26b, 0x462e2717), WTCP(0x6adb69d3, 0x46777d52), WTCP(0x6aaaf11b, 0x46c0c75a), WTCP(0x6a7a385c, 0x470a04a9), WTCP(0x6a493fb3, 0x475334b9), WTCP(0x6a18073d, 0x479c5707), WTCP(0x69e68f17, 0x47e56b0c), WTCP(0x69b4d761, 0x482e7045), WTCP(0x6982e039, 0x4877662c), WTCP(0x6950a9c0, 0x48c04c3f), WTCP(0x691e341a, 0x490921f8), WTCP(0x68eb7f67, 0x4951e6d5), WTCP(0x68b88bcd, 0x499a9a51), WTCP(0x68855970, 0x49e33beb), WTCP(0x6851e875, 0x4a2bcb1f), WTCP(0x681e3905, 0x4a74476b), WTCP(0x67ea4b47, 0x4abcb04c), WTCP(0x67b61f63, 0x4b050541), WTCP(0x6781b585, 0x4b4d45c9), WTCP(0x674d0dd6, 0x4b957162), WTCP(0x67182883, 0x4bdd878c), WTCP(0x66e305b8, 0x4c2587c6), WTCP(0x66ada5a5, 0x4c6d7190), WTCP(0x66780878, 0x4cb5446a), WTCP(0x66422e60, 0x4cfcffd5), WTCP(0x660c1790, 0x4d44a353), WTCP(0x65d5c439, 0x4d8c2e64), WTCP(0x659f348e, 0x4dd3a08c), WTCP(0x656868c3, 0x4e1af94b), WTCP(0x6531610d, 0x4e623825), WTCP(0x64fa1da3, 0x4ea95c9d), WTCP(0x64c29ebb, 0x4ef06637), WTCP(0x648ae48d, 0x4f375477), WTCP(0x6452ef53, 0x4f7e26e1), WTCP(0x641abf46, 0x4fc4dcfb), WTCP(0x63e254a2, 0x500b7649), WTCP(0x63a9afa2, 0x5051f253), WTCP(0x6370d083, 0x5098509f), WTCP(0x6337b784, 0x50de90b3), WTCP(0x62fe64e3, 0x5124b218), WTCP(0x62c4d8e0, 0x516ab455), WTCP(0x628b13bc, 0x51b096f3), WTCP(0x625115b8, 0x51f6597b), WTCP(0x6216df18, 0x523bfb78), WTCP(0x61dc701f, 0x52817c72), WTCP(0x61a1c912, 0x52c6dbf5), WTCP(0x6166ea36, 0x530c198d), WTCP(0x612bd3d2, 0x535134c5), WTCP(0x60f0862d, 0x53962d2a), WTCP(0x60b50190, 0x53db024a), WTCP(0x60794644, 0x541fb3b1), WTCP(0x603d5494, 0x546440ef), WTCP(0x60012cca, 0x54a8a992), WTCP(0x5fc4cf33, 0x54eced2b), WTCP(0x5f883c1c, 0x55310b48), WTCP(0x5f4b73d2, 0x5575037c), WTCP(0x5f0e76a5, 0x55b8d558), WTCP(0x5ed144e5, 0x55fc806f), WTCP(0x5e93dee1, 0x56400452), WTCP(0x5e5644ec, 0x56836096), WTCP(0x5e187757, 0x56c694cf), WTCP(0x5dda7677, 0x5709a092), WTCP(0x5d9c429f, 0x574c8374), WTCP(0x5d5ddc24, 0x578f3d0d), WTCP(0x5d1f435d, 0x57d1ccf2), WTCP(0x5ce078a0, 0x581432bd), WTCP(0x5ca17c45, 0x58566e04), WTCP(0x5c624ea4, 0x58987e63), WTCP(0x5c22f016, 0x58da6372), WTCP(0x5be360f6, 0x591c1ccc), WTCP(0x5ba3a19f, 0x595daa0d), WTCP(0x5b63b26c, 0x599f0ad1), WTCP(0x5b2393ba, 0x59e03eb6), WTCP(0x5ae345e7, 0x5a214558), WTCP(0x5aa2c951, 0x5a621e56), }; /** * \brief Helper table containing the length, rasterand shape mapping to * individual window slope tables. [0: sine ][0: radix2 raster * ][ceil(log2(length)) length 4 .. 1024 ] [1: 10ms raster * ][ceil(log2(length)) length 3.25 .. 960 ] [2: 3/4 of radix 2 * raster][ceil(log2(length)) length 3 .. 768 ] [1: KBD ][0: * radix2 raster ][ceil(log2(length)) length 128 .. 1024 ] [1: 10ms * raster ][ceil(log2(length)) length 120 .. 960 ] [2: * 3/4 of radix 2 raster][ceil(log2(length)) length 96 .. 768 ] */ const FIXP_WTP *const windowSlopes[2][4][9] = { { /* Sine */ {/* Radix 2 */ NULL, SineWindow8, SineWindow16, SineWindow32, SineWindow64, SineWindow128, SineWindow256, SineWindow512, SineWindow1024}, { /* 10ms raster */ NULL, /* 3.25 */ NULL, /* 7.5 */ NULL, NULL, NULL, SineWindow120, SineWindow240, SineWindow480, SineWindow960}, { /* 3/4 radix2 raster */ NULL, /* 3 */ NULL, /* 6 */ SineWindow12, SineWindow24, SineWindow48, SineWindow96, SineWindow192, SineWindow384, SineWindow768}, { /* 3/4 radix2 raster */ NULL, NULL, /* 3 */ NULL, /* 6 */ SineWindow20, SineWindow40, NULL, SineWindow160, NULL, NULL, }}, { /* KBD */ {/* Radix 2 */ NULL, KBDWindow128, KBDWindow256, SineWindow512, KBDWindow1024}, {/* 10ms raster */ NULL, KBDWindow120, NULL, SineWindow480, KBDWindow960}, {/* 3/4 radix2 raster */ NULL, KBDWindow96, SineWindow192, /* This entry might be accessed for erred bit streams. */ NULL, KBDWindow768}, {NULL, NULL, NULL, NULL}}}; const FIXP_WTP *FDKgetWindowSlope(int length, int shape) { const FIXP_WTP *w = NULL; int raster, ld2_length; /* Get ld2 of length - 2 + 1 -2: because first table entry is window of size 4 +1: because we already include +1 because of ceil(log2(length)) */ ld2_length = DFRACT_BITS - 1 - fNormz((FIXP_DBL)length) - 1; /* Extract sort of "eigenvalue" (the 4 left most bits) of length. */ switch ((length) >> (ld2_length - 2)) { case 0x8: /* radix 2 */ raster = 0; ld2_length--; /* revert + 1 because of ceil(log2(length)) from above. */ break; case 0xf: /* 10 ms */ raster = 1; break; case 0xc: /* 3/4 of radix 2 */ raster = 2; break; default: raster = 0; break; } /* The table for sine windows (shape == 0) is 4 entries longer. */ if (shape == 1) { ld2_length -= 4; } /* Look up table */ w = windowSlopes[shape & 1][raster][ld2_length]; FDK_ASSERT(w != NULL); return w; } /* * QMF filter and twiddle tables */ #ifdef QMF_COEFF_16BIT #define QFC(x) FX_DBL2FXCONST_SGL(x) #define QTCFL(x) FL2FXCONST_SGL(x) #define QTC(x) FX_DBL2FXCONST_SGL(x) #else #define QFC(x) ((FIXP_DBL)(x)) #define QTCFL(x) FL2FXCONST_DBL(x) #define QTC(x) ((FIXP_DBL)(x)) #endif /* ARCH_PREFER_MULT_32x16 */ /*! \name QMF \brief QMF-Table 64 channels, N = 640, optimized by PE 010516 The coeffs are rearranged compared with the reference in the following way, exploiting symmetry : sbr_qmf_64[5] = p_64_640_qmf[0]; sbr_qmf_64[6] = p_64_640_qmf[128]; sbr_qmf_64[7] = p_64_640_qmf[256]; sbr_qmf_64[8] = p_64_640_qmf[384]; sbr_qmf_64[9] = p_64_640_qmf[512]; sbr_qmf_64[10] = p_64_640_qmf[1]; sbr_qmf_64[11] = p_64_640_qmf[129]; sbr_qmf_64[12] = p_64_640_qmf[257]; sbr_qmf_64[13] = p_64_640_qmf[385]; sbr_qmf_64[14] = p_64_640_qmf[513]; . . . sbr_qmf_64_640_qmf[315] = p_64_640_qmf[62]; sbr_qmf_64_640_qmf[316] = p_64_640_qmf[190]; sbr_qmf_64_640_qmf[317] = p_64_640_qmf[318]; sbr_qmf_64_640_qmf[318] = p_64_640_qmf[446]; sbr_qmf_64_640_qmf[319] = p_64_640_qmf[574]; sbr_qmf_64_640_qmf[320] = p_64_640_qmf[63]; sbr_qmf_64_640_qmf[321] = p_64_640_qmf[191]; sbr_qmf_64_640_qmf[322] = p_64_640_qmf[319]; sbr_qmf_64_640_qmf[323] = p_64_640_qmf[447]; sbr_qmf_64_640_qmf[324] = p_64_640_qmf[575]; sbr_qmf_64_640_qmf[319] = p_64_640_qmf[64]; sbr_qmf_64_640_qmf[318] = p_64_640_qmf[192]; sbr_qmf_64_640_qmf[317] = p_64_640_qmf[320]; sbr_qmf_64_640_qmf[316] = p_64_640_qmf[448]; sbr_qmf_64_640_qmf[315] = p_64_640_qmf[576]; sbr_qmf_64_640_qmf[314] = p_64_640_qmf[65]; sbr_qmf_64_640_qmf[313] = p_64_640_qmf[193]; sbr_qmf_64_640_qmf[312] = p_64_640_qmf[321]; sbr_qmf_64_640_qmf[311] = p_64_640_qmf[449]; sbr_qmf_64_640_qmf[310] = p_64_640_qmf[577]; . . . sbr_qmf_64[9] = p_64_640_qmf[126] sbr_qmf_64[8] = p_64_640_qmf[254]; sbr_qmf_64[7] = p_64_640_qmf[382]; sbr_qmf_64[6] = p_64_640_qmf[510]; sbr_qmf_64[5] = p_64_640_qmf[638]; sbr_qmf_64[4] = p_64_640_qmf[127] sbr_qmf_64[3] = p_64_640_qmf[255]; sbr_qmf_64[2] = p_64_640_qmf[383]; sbr_qmf_64[1] = p_64_640_qmf[511]; sbr_qmf_64[0] = p_64_640_qmf[639]; Max sum of all FIR filter absolute coefficients is: 0x7FF5B201 thus, the filter output is not required to be scaled. \showinitializer */ RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_PFT qmf_pfilt120[] = { QFC(0x00000000), QFC(0x01b2e41d), QFC(0x2e3a7532), QFC(0xd1c58ace), QFC(0xfe4d1be3), QFC(0xffefcdb5), QFC(0x02828e13), QFC(0x35eecfd1), QFC(0xd94e53e3), QFC(0xfefdfe42), QFC(0xffec30b0), QFC(0x036b8e20), QFC(0x3daa7c5c), QFC(0xe08b3fa6), QFC(0xff8f33fc), QFC(0xffe88ba8), QFC(0x04694101), QFC(0x4547daeb), QFC(0xe75f8bb7), QFC(0x0000e790), QFC(0xffe69150), QFC(0x057341bc), QFC(0x4c9ef50f), QFC(0xedb0fdbd), QFC(0x00549c76), QFC(0xffe6db43), QFC(0x067ef951), QFC(0x5389d1bb), QFC(0xf36dbfe6), QFC(0x008cbe92), QFC(0xffea353a), QFC(0x077fedb3), QFC(0x59e2f69e), QFC(0xf887507c), QFC(0x00acbd2f), QFC(0xfff176e1), QFC(0x086685a4), QFC(0x5f845914), QFC(0xfcf2b6c8), QFC(0x00b881db), QFC(0xfffd1253), QFC(0x09233c49), QFC(0x64504658), QFC(0x00adb69e), QFC(0x00b4790a), QFC(0x000d31b5), QFC(0x09a3e163), QFC(0x682b39a4), QFC(0x03b8f8dc), QFC(0x00a520bb), QFC(0x0021e26b), QFC(0x09d536b4), QFC(0x6afb0c80), QFC(0x06186566), QFC(0x008db1f0), QFC(0x003a81c0), QFC(0x09a505f2), QFC(0x6cb28145), QFC(0x07d6e67c), QFC(0x00728512), QFC(0x0055dba1), QFC(0x09015651), QFC(0x6d474e1d), QFC(0x09015651), QFC(0x0055dba1), QFC(0xfe4d1be3), QFC(0xd1c58ace), QFC(0x2e3a7532), QFC(0x01b2e41d), QFC(0x00000000), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_PFT qmf_pfilt200[] = { QFC(0x00000000), QFC(0x01b2e41d), QFC(0x2e3a7532), QFC(0xd1c58ace), QFC(0xfe4d1be3), QFC(0xffefd5d9), QFC(0x022c39a4), QFC(0x32d6e6f6), QFC(0xd652421f), QFC(0xfebafd64), QFC(0xffef3d2e), QFC(0x02af2a39), QFC(0x377b44a6), QFC(0xdac7ff47), QFC(0xff1d9e1f), QFC(0xffed03e9), QFC(0x033b07ff), QFC(0x3c1fc4e4), QFC(0xdf2029d5), QFC(0xff74a37e), QFC(0xffeab7cc), QFC(0x03cf3ade), QFC(0x40bc12f6), QFC(0xe3546cf8), QFC(0xffc070af), QFC(0xffe88ba8), QFC(0x04694101), QFC(0x4547daeb), QFC(0xe75f8bb7), QFC(0x0000e790), QFC(0xffe7546d), QFC(0x050826e6), QFC(0x49ba0a48), QFC(0xeb3ac63a), QFC(0x0036aa5d), QFC(0xffe6665c), QFC(0x05a92d73), QFC(0x4e0b0602), QFC(0xeee323fd), QFC(0x0061fdf9), QFC(0xffe6858d), QFC(0x0649e26b), QFC(0x523225cf), QFC(0xf2549ca7), QFC(0x00838276), QFC(0xffe7e0bd), QFC(0x06e7cba4), QFC(0x5627597c), QFC(0xf58c23ae), QFC(0x009c49df), QFC(0xffea353a), QFC(0x077fedb3), QFC(0x59e2f69e), QFC(0xf887507c), QFC(0x00acbd2f), QFC(0xffee0a64), QFC(0x080e83ac), QFC(0x5d5bac5e), QFC(0xfb432a8a), QFC(0x00b5e294), QFC(0xfff35c0f), QFC(0x08905893), QFC(0x608bf7c1), QFC(0xfdbfe2d8), QFC(0x00b8dcd6), QFC(0xfffa67ed), QFC(0x0901a70f), QFC(0x636d2657), QFC(0xfffccdc7), QFC(0x00b66387), QFC(0x0002f512), QFC(0x095eb98e), QFC(0x65f9595d), QFC(0x01fa380f), QFC(0x00afb0f3), QFC(0x000d31b5), QFC(0x09a3e163), QFC(0x682b39a4), QFC(0x03b8f8dc), QFC(0x00a520bb), QFC(0x00193141), QFC(0x09cc1a7d), QFC(0x69fbfee3), QFC(0x05395430), QFC(0x0097ce05), QFC(0x00269ad4), QFC(0x09d3fe14), QFC(0x6b69bfaf), QFC(0x067e12f2), QFC(0x00889924), QFC(0x003567de), QFC(0x09b75cca), QFC(0x6c716eb9), QFC(0x0789e850), QFC(0x00781556), QFC(0x0045436a), QFC(0x097277a9), QFC(0x6d110fe4), QFC(0x085f29c6), QFC(0x00670cb6), QFC(0x0055dba1), QFC(0x09015651), QFC(0x6d474e1d), QFC(0x09015651), QFC(0x0055dba1), QFC(0xfe4d1be3), QFC(0xd1c58ace), QFC(0x2e3a7532), QFC(0x01b2e41d), QFC(0x00000000), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_QTW qmf_phaseshift_cos40[] = { QTC(0x7fef5260), QTC(0x7f69ff76), QTC(0x7e5fe493), QTC(0x7cd21707), QTC(0x7ac23561), QTC(0x783265c0), QTC(0x75255392), QTC(0x719e2cd2), QTC(0x6da09eb1), QTC(0x6930d1c4), QTC(0x645365b2), QTC(0x5f0d6c5b), QTC(0x59646498), QTC(0x535e3479), QTC(0x4d012324), QTC(0x4653d24b), QTC(0x3f5d373e), QTC(0x382493b0), QTC(0x30b16e23), QTC(0x290b8a12), QTC(0x213adfda), QTC(0x1947946c), QTC(0x1139f0cf), QTC(0x091a597e), QTC(0x00f145ab), QTC(0xf8c73668), QTC(0xf0a4adcf), QTC(0xe8922622), QTC(0xe09808f5), QTC(0xd8bea66a), QTC(0xd10e2c89), QTC(0xc98e9eb5), QTC(0xc247cd5a), QTC(0xbb414dc0), QTC(0xb4827228), QTC(0xae12422c), QTC(0xa7f7736a), QTC(0xa2386284), QTC(0x9cdb0c83), QTC(0x97e50896), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_QTW qmf_phaseshift_sin40[] = { QTC(0x0415583b), QTC(0x0c3bc74f), QTC(0x145576b1), QTC(0x1c59f557), QTC(0x2440e84d), QTC(0x2c021369), QTC(0x339561e1), QTC(0x3af2eeb7), QTC(0x42130cf0), QTC(0x48ee4f98), QTC(0x4f7d917c), QTC(0x55b9fc9e), QTC(0x5b9d1154), QTC(0x6120ad0d), QTC(0x663f10b7), QTC(0x6af2e6bc), QTC(0x6f374891), QTC(0x7307c3d0), QTC(0x76605edb), QTC(0x793d9d03), QTC(0x7b9c8226), QTC(0x7d7a95cf), QTC(0x7ed5e5c6), QTC(0x7fad081b), QTC(0x7fff1c9b), QTC(0x7fcbcdbc), QTC(0x7f1350f8), QTC(0x7dd6668f), QTC(0x7c1658c5), QTC(0x79d4fa89), QTC(0x7714a58b), QTC(0x73d837ca), QTC(0x7023109a), QTC(0x6bf90d1d), QTC(0x675e843e), QTC(0x6258422c), QTC(0x5ceb8355), QTC(0x571deefa), QTC(0x50f59141), QTC(0x4a78d4f0), }; /* This filter is scaled (0.8*pfilt) */ RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_PFT qmf_pfilt400[] = { QFC(0x00000000), QFC(0x015be9b1), QFC(0x24fb90f5), QFC(0xdb046f0b), QFC(0xfea4164f), QFC(0xfff15ed6), QFC(0x018b53a8), QFC(0x26d2bd4e), QFC(0xdcd812f9), QFC(0xfed12595), QFC(0xfff3117b), QFC(0x01bcfae9), QFC(0x28abebf8), QFC(0xdea834e5), QFC(0xfefbfdea), QFC(0xfff32e53), QFC(0x01f075de), QFC(0x2a86e540), QFC(0xe07383c3), QFC(0xff24936e), QFC(0xfff29758), QFC(0x0225bb61), QFC(0x2c629d51), QFC(0xe2399905), QFC(0xff4ae4e6), QFC(0xfff1ab73), QFC(0x025cb6d7), QFC(0x2e3e69f9), QFC(0xe3fa13fc), QFC(0xff6eefd4), QFC(0xfff0cfed), QFC(0x0295a000), QFC(0x30196a50), QFC(0xe5b354ab), QFC(0xff9082cb), QFC(0xffefd442), QFC(0x02d01d61), QFC(0x31f2b6ac), QFC(0xe765dadc), QFC(0xffb0037f), QFC(0xffeef970), QFC(0x030c2f18), QFC(0x33c9a8c5), QFC(0xe910572d), QFC(0xffcd26f2), QFC(0xffee0f91), QFC(0x03494088), QFC(0x359ce8be), QFC(0xeab28265), QFC(0xffe8133f), QFC(0xffed3c86), QFC(0x03876734), QFC(0x376caf22), QFC(0xec4c6fc6), QFC(0x0000b940), QFC(0xffecb05f), QFC(0x03c6b32b), QFC(0x3936c186), QFC(0xeddbfa4a), QFC(0x00174372), QFC(0xffec438a), QFC(0x04068585), QFC(0x3afb3b6d), QFC(0xef62382f), QFC(0x002bbb7e), QFC(0xffebc5c7), QFC(0x0446af4f), QFC(0x3cb9159f), QFC(0xf0de3518), QFC(0x003e0713), QFC(0xffeb8517), QFC(0x0487578f), QFC(0x3e6f3802), QFC(0xf24f4ffd), QFC(0x004e64c7), QFC(0xffeb8b0d), QFC(0x04c7cd0d), QFC(0x401d78d8), QFC(0xf3b6114c), QFC(0x005ccd60), QFC(0xffeb9e0a), QFC(0x0507e855), QFC(0x41c1b7d9), QFC(0xf5107d52), QFC(0x0069352b), QFC(0xffec0c97), QFC(0x054789e4), QFC(0x435c76d2), QFC(0xf6600380), QFC(0x0073ff44), QFC(0xffecb3ca), QFC(0x05863c83), QFC(0x44ec4796), QFC(0xf7a34fbf), QFC(0x007d07e5), QFC(0xffed65ae), QFC(0x05c3bdde), QFC(0x46702a28), QFC(0xf8da6b28), QFC(0x008444ef), QFC(0xffee90fb), QFC(0x05fff15c), QFC(0x47e8c54c), QFC(0xfa05d9fc), QFC(0x008a30f2), QFC(0xffefff78), QFC(0x0639db53), QFC(0x4952ab1e), QFC(0xfb23d977), QFC(0x008e9313), QFC(0xfff1a1ea), QFC(0x067202f0), QFC(0x4aafbd18), QFC(0xfc35bba2), QFC(0x00918210), QFC(0xfff3a45f), QFC(0x06a741b7), QFC(0x4bfdfb06), QFC(0xfd3aee85), QFC(0x009350b6), QFC(0xfff5e33f), QFC(0x06d9e076), QFC(0x4d3cc634), QFC(0xfe331be0), QFC(0x0093e3de), QFC(0xfff867de), QFC(0x07090b4f), QFC(0x4e6cc1b3), QFC(0xff1f4fd2), QFC(0x00936109), QFC(0xfffb8658), QFC(0x073485a5), QFC(0x4f8a8512), QFC(0xfffd716c), QFC(0x0091e939), QFC(0xfffec6af), QFC(0x075c2159), QFC(0x50986228), QFC(0x00cfb536), QFC(0x008f7f85), QFC(0x00025da8), QFC(0x077efad8), QFC(0x5194477e), QFC(0x0194f9a6), QFC(0x008c8d8f), QFC(0x00064e63), QFC(0x079d423f), QFC(0x527db75e), QFC(0x024d9e1c), QFC(0x00886b36), QFC(0x000a8e2a), QFC(0x07b64de9), QFC(0x5355c7b6), QFC(0x02fa60b0), QFC(0x00841a2f), QFC(0x000f2b4f), QFC(0x07c95704), QFC(0x5418bd4a), QFC(0x0399eb6f), QFC(0x007eea79), QFC(0x00142767), QFC(0x07d67b97), QFC(0x54c998b6), QFC(0x042ddcf3), QFC(0x0079719e), QFC(0x00193ee8), QFC(0x07dd27cf), QFC(0x55662c93), QFC(0x04b5da5c), QFC(0x007369b7), QFC(0x001ee243), QFC(0x07dccb44), QFC(0x55ee32f2), QFC(0x0531a8c2), QFC(0x006d4750), QFC(0x002471ce), QFC(0x07d588d9), QFC(0x566317ad), QFC(0x05a2ff7a), QFC(0x0066c7aa), QFC(0x002ab97f), QFC(0x07c5e3d5), QFC(0x56c12561), QFC(0x0607ed0d), QFC(0x00601112), QFC(0x0030e1af), QFC(0x07ae9698), QFC(0x570be9e8), QFC(0x0662a78a), QFC(0x005958bb), QFC(0x00376922), QFC(0x078ec621), QFC(0x5740d984), QFC(0x06b287d1), QFC(0x00527092), QFC(0x003e065c), QFC(0x0765b74d), QFC(0x57607ccb), QFC(0x06f819ec), QFC(0x004b9363), QFC(0x0044afb4), QFC(0x0734450e), QFC(0x576c3e7e), QFC(0x0734450e), QFC(0x0044afb4), QFC(0xfea4164f), QFC(0xdb046f0b), QFC(0x24fb90f5), QFC(0x015be9b1), QFC(0x00000000), }; const FIXP_QTW qmf_phaseshift_cos16[] = { QTC(0x7fc25596), QTC(0x7dd6668f), QTC(0x7a05eead), QTC(0x745f9dd1), QTC(0x6cf934fc), QTC(0x63ef3290), QTC(0x59646498), QTC(0x4d8162c4), QTC(0x4073f21d), QTC(0x326e54c7), QTC(0x23a6887f), QTC(0x145576b1), QTC(0x04b6195d), QTC(0xf50497fb), QTC(0xe57d5fda), QTC(0xd65c3b7b), }; const FIXP_QTW qmf_phaseshift_sin16[] = { QTC(0x07d95b9e), QTC(0x176dd9de), QTC(0x26a82186), QTC(0x354d9057), QTC(0x4325c135), QTC(0x4ffb654d), QTC(0x5b9d1154), QTC(0x65ddfbd3), QTC(0x6e96a99d), QTC(0x75a585cf), QTC(0x7aef6323), QTC(0x7e5fe493), QTC(0x7fe9cbc0), QTC(0x7f872bf3), QTC(0x7d3980ec), QTC(0x7909a92d), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_PFT qmf_pfilt240[] = { /* FP filter implementation */ QFC(0x00000000), QFC(0x0121ed68), QFC(0x1ed1a380), QFC(0xe12e5c80), QFC(0xfede1298), QFC(0xfff4b438), QFC(0x0164d8de), QFC(0x21610064), QFC(0xe3b64ef2), QFC(0xff1ba1be), QFC(0xfff533ce), QFC(0x01ac5eb8), QFC(0x23f48a8e), QFC(0xe63437e4), QFC(0xff53fed7), QFC(0xfff40ee0), QFC(0x01f7edb3), QFC(0x26895855), QFC(0xe8a5bb55), QFC(0xff871d30), QFC(0xfff2cb20), QFC(0x0247b415), QFC(0x291c52e4), QFC(0xeb077fc7), QFC(0xffb4cd53), QFC(0xfff18a22), QFC(0x029b070e), QFC(0x2baa29ab), QFC(0xed57da15), QFC(0xffdd4df1), QFC(0xfff05d1b), QFC(0x02f0d600), QFC(0x2e2fe755), QFC(0xef9507d5), QFC(0x00009a60), QFC(0xffefac36), QFC(0x0348fcbc), QFC(0x30a98c1c), QFC(0xf1bba8f2), QFC(0x001eed4c), QFC(0xffef0b8b), QFC(0x03a22bd2), QFC(0x3314a372), QFC(0xf3cb53d5), QFC(0x0038684e), QFC(0xffeef3e0), QFC(0x03fbd58b), QFC(0x356de4ab), QFC(0xf5c263c0), QFC(0x004d55d0), QFC(0xffef3cd8), QFC(0x0454a637), QFC(0x37b13672), QFC(0xf79e7feb), QFC(0x005dd461), QFC(0xfff01619), QFC(0x04abb9c0), QFC(0x39dc5c00), QFC(0xf95f9279), QFC(0x006a5b4d), QFC(0xfff178d2), QFC(0x04fff3cb), QFC(0x3beca455), QFC(0xfb04e050), QFC(0x007328ca), QFC(0xfff390f0), QFC(0x054fa1dc), QFC(0x3ddd668e), QFC(0xfc8c7550), QFC(0x00788f16), QFC(0xfff64f40), QFC(0x0599ae6b), QFC(0x3fad90c7), QFC(0xfdf72485), QFC(0x007b013c), QFC(0xfff9abe4), QFC(0x05dcdec0), QFC(0x415aa155), QFC(0xff44c284), QFC(0x007ad0dd), QFC(0xfffe0c37), QFC(0x06177d87), QFC(0x42e02f00), QFC(0x0073cf14), QFC(0x007850b2), QFC(0x000314dd), QFC(0x0647fe8b), QFC(0x443e0472), QFC(0x0185ddb7), QFC(0x00741328), QFC(0x0008cbce), QFC(0x066d40eb), QFC(0x45722655), QFC(0x027b5093), QFC(0x006e15d2), QFC(0x000f67a8), QFC(0x0684f772), QFC(0x46789539), QFC(0x03537bc9), QFC(0x0066c76d), QFC(0x001696f2), QFC(0x068e247c), QFC(0x47520855), QFC(0x04104399), QFC(0x005e76a0), QFC(0x001e5ed7), QFC(0x06874760), QFC(0x47fd3e55), QFC(0x04b27f90), QFC(0x0055a663), QFC(0x0027012b), QFC(0x066e03f9), QFC(0x487700c7), QFC(0x0539eefc), QFC(0x004c58b7), QFC(0x0030042f), QFC(0x0641b0ab), QFC(0x48c0afc7), QFC(0x05a90172), QFC(0x0042c9e7), QFC(0x00393d16), QFC(0x0600e435), QFC(0x48da3400), QFC(0x0600e435), QFC(0x00393d16), QFC(0xfede1298), QFC(0xe12e5c80), QFC(0x1ed1a380), QFC(0x0121ed68), QFC(0x00000000), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_QTW qmf_phaseshift_cos24[] = { QTC(0x7fded530), QTC(0x7ed5e5c6), QTC(0x7cc62bdf), QTC(0x79b3ece0), QTC(0x75a585cf), QTC(0x70a35e25), QTC(0x6ab7d663), QTC(0x63ef3290), QTC(0x5c5780d3), QTC(0x54007c51), QTC(0x4afb6c98), QTC(0x415b01ce), QTC(0x37332dfd), QTC(0x2c98fbba), QTC(0x21a26295), QTC(0x1666198d), QTC(0x0afb6805), QTC(0xff79f587), QTC(0xf3f998c0), QTC(0xe8922622), QTC(0xdd5b3e7b), QTC(0xd26c1e08), QTC(0xc7db6c50), QTC(0xbdbf0d2f), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_QTW qmf_phaseshift_sin24[] = { QTC(0x05c1f4e7), QTC(0x1139f0cf), QTC(0x1c8e3bbe), QTC(0x27a75c95), QTC(0x326e54c7), QTC(0x3cccd004), QTC(0x46ad5278), QTC(0x4ffb654d), QTC(0x58a3c118), QTC(0x609475c3), QTC(0x67bd0fbd), QTC(0x6e0eba0c), QTC(0x737c5d0b), QTC(0x77fab989), QTC(0x7b808015), QTC(0x7e06644c), QTC(0x7f872bf3), QTC(0x7fffb9d1), QTC(0x7f6f141f), QTC(0x7dd6668f), QTC(0x7b38ffde), QTC(0x779c4afc), QTC(0x7307c3d0), QTC(0x6d84e7b7), }; /* qmf_pfilt640 is used with stride 2 instead of qmf_pfilt320[] */ RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_QTW qmf_phaseshift_cos32[] = { QTC(0x7fe9cbc0), QTC(0x7f3857f6), QTC(0x7dd6668f), QTC(0x7bc5e290), QTC(0x7909a92d), QTC(0x75a585cf), QTC(0x719e2cd2), QTC(0x6cf934fc), QTC(0x67bd0fbd), QTC(0x61f1003f), QTC(0x5b9d1154), QTC(0x54ca0a4b), QTC(0x4d8162c4), QTC(0x45cd358f), QTC(0x3db832a6), QTC(0x354d9057), QTC(0x2c98fbba), QTC(0x23a6887f), QTC(0x1a82a026), QTC(0x1139f0cf), QTC(0x07d95b9e), QTC(0xfe6de2e0), QTC(0xf50497fb), QTC(0xebaa894f), QTC(0xe26cb01b), QTC(0xd957de7a), QTC(0xd078ad9e), QTC(0xc7db6c50), QTC(0xbf8c0de3), QTC(0xb796199b), QTC(0xb0049ab3), QTC(0xa8e21106), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_QTW qmf_phaseshift_sin32[] = { QTC(0x04b6195d), QTC(0x0e1bc2e4), QTC(0x176dd9de), QTC(0x209f701c), QTC(0x29a3c485), QTC(0x326e54c7), QTC(0x3af2eeb7), QTC(0x4325c135), QTC(0x4afb6c98), QTC(0x5269126e), QTC(0x59646498), QTC(0x5fe3b38d), QTC(0x65ddfbd3), QTC(0x6b4af279), QTC(0x7023109a), QTC(0x745f9dd1), QTC(0x77fab989), QTC(0x7aef6323), QTC(0x7d3980ec), QTC(0x7ed5e5c6), QTC(0x7fc25596), QTC(0x7ffd885a), QTC(0x7f872bf3), QTC(0x7e5fe493), QTC(0x7c894bde), QTC(0x7a05eead), QTC(0x76d94989), QTC(0x7307c3d0), QTC(0x6e96a99d), QTC(0x698c246c), QTC(0x63ef3290), QTC(0x5dc79d7c), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_QTW qmf_phaseshift_cos_downsamp32[] = { QTC(0x7fd8878e), QTC(0x7e9d55fc), QTC(0x7c29fbee), QTC(0x78848414), QTC(0x73b5ebd1), QTC(0x6dca0d14), QTC(0x66cf8120), QTC(0x5ed77c8a), QTC(0x55f5a4d2), QTC(0x4c3fdff4), QTC(0x41ce1e65), QTC(0x36ba2014), QTC(0x2b1f34eb), QTC(0x1f19f97b), QTC(0x12c8106f), QTC(0x0647d97c), QTC(0xf9b82684), QTC(0xed37ef91), QTC(0xe0e60685), QTC(0xd4e0cb15), QTC(0xc945dfec), QTC(0xbe31e19b), QTC(0xb3c0200c), QTC(0xaa0a5b2e), QTC(0xa1288376), QTC(0x99307ee0), QTC(0x9235f2ec), QTC(0x8c4a142f), QTC(0x877b7bec), QTC(0x83d60412), QTC(0x8162aa04), QTC(0x80277872), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_QTW qmf_phaseshift_sin_downsamp32[] = { QTC(0x0647d97c), QTC(0x12c8106f), QTC(0x1f19f97b), QTC(0x2b1f34eb), QTC(0x36ba2014), QTC(0x41ce1e65), QTC(0x4c3fdff4), QTC(0x55f5a4d2), QTC(0x5ed77c8a), QTC(0x66cf8120), QTC(0x6dca0d14), QTC(0x73b5ebd1), QTC(0x78848414), QTC(0x7c29fbee), QTC(0x7e9d55fc), QTC(0x7fd8878e), QTC(0x7fd8878e), QTC(0x7e9d55fc), QTC(0x7c29fbee), QTC(0x78848414), QTC(0x73b5ebd1), QTC(0x6dca0d14), QTC(0x66cf8120), QTC(0x5ed77c8a), QTC(0x55f5a4d2), QTC(0x4c3fdff4), QTC(0x41ce1e65), QTC(0x36ba2014), QTC(0x2b1f34eb), QTC(0x1f19f97b), QTC(0x12c8106f), QTC(0x0647d97c), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_PFT qmf_pfilt640[] = { QFC(0x00000000), QFC(0x01b2e41d), QFC(0x2e3a7532), QFC(0xd1c58ace), QFC(0xfe4d1be3), QFC(0xffede50e), QFC(0x01d78bfc), QFC(0x2faa221c), QFC(0xd3337b3d), QFC(0xfe70b8d1), QFC(0xffed978a), QFC(0x01fd3ba0), QFC(0x311af3a4), QFC(0xd49fd55f), QFC(0xfe933dc0), QFC(0xffefc9b9), QFC(0x02244a25), QFC(0x328cc6f0), QFC(0xd60a46e5), QFC(0xfeb48d0d), QFC(0xfff0065d), QFC(0x024bf7a1), QFC(0x33ff670e), QFC(0xd7722f04), QFC(0xfed4bec3), QFC(0xffeff6ca), QFC(0x0274ba43), QFC(0x3572ec70), QFC(0xd8d7f21f), QFC(0xfef3f6ab), QFC(0xffef7b8b), QFC(0x029e35b4), QFC(0x36e69691), QFC(0xda3b176a), QFC(0xff120d70), QFC(0xffeedfa4), QFC(0x02c89901), QFC(0x385a49c4), QFC(0xdb9b5b12), QFC(0xff2ef725), QFC(0xffee1650), QFC(0x02f3e48d), QFC(0x39ce0477), QFC(0xdcf898fb), QFC(0xff4aabc8), QFC(0xffed651d), QFC(0x03201116), QFC(0x3b415115), QFC(0xde529086), QFC(0xff6542d1), QFC(0xffecc31b), QFC(0x034d01f1), QFC(0x3cb41219), QFC(0xdfa93ab5), QFC(0xff7ee3f1), QFC(0xffebe77b), QFC(0x037ad438), QFC(0x3e25b17e), QFC(0xe0fc421e), QFC(0xff975c01), QFC(0xffeb50b2), QFC(0x03a966bc), QFC(0x3f962fb8), QFC(0xe24b8f66), QFC(0xffaea5d6), QFC(0xffea9192), QFC(0x03d8afe6), QFC(0x41058bc6), QFC(0xe396a45d), QFC(0xffc4e365), QFC(0xffe9ca76), QFC(0x04083fec), QFC(0x4272a385), QFC(0xe4de0cb0), QFC(0xffda17f2), QFC(0xffe940f4), QFC(0x043889c6), QFC(0x43de620a), QFC(0xe620c476), QFC(0xffee183b), QFC(0xffe88ba8), QFC(0x04694101), QFC(0x4547daeb), QFC(0xe75f8bb7), QFC(0x0000e790), QFC(0xffe83a07), QFC(0x049aa82f), QFC(0x46aea856), QFC(0xe89971b7), QFC(0x00131c75), QFC(0xffe79e16), QFC(0x04cc2fcf), QFC(0x4812f848), QFC(0xe9cea84a), QFC(0x0023b989), QFC(0xffe7746e), QFC(0x04fe20be), QFC(0x4973fef2), QFC(0xeafee7f1), QFC(0x0033b927), QFC(0xffe6d466), QFC(0x05303f88), QFC(0x4ad237a2), QFC(0xec2a3f5f), QFC(0x00426f36), QFC(0xffe6afed), QFC(0x05626209), QFC(0x4c2ca3df), QFC(0xed50a31d), QFC(0x00504f41), QFC(0xffe65416), QFC(0x05950122), QFC(0x4d83976d), QFC(0xee71b2fe), QFC(0x005d36df), QFC(0xffe681c6), QFC(0x05c76fed), QFC(0x4ed62be3), QFC(0xef8d4d7b), QFC(0x006928a0), QFC(0xffe66dd0), QFC(0x05f9c051), QFC(0x5024d70e), QFC(0xf0a3959f), QFC(0x007400b8), QFC(0xffe66fab), QFC(0x062bf5ec), QFC(0x516eefb9), QFC(0xf1b461ab), QFC(0x007e0393), QFC(0xffe69423), QFC(0x065dd56a), QFC(0x52b449de), QFC(0xf2bf6ea4), QFC(0x00872c63), QFC(0xffe6fed4), QFC(0x068f8b44), QFC(0x53f495aa), QFC(0xf3c4e887), QFC(0x008f87aa), QFC(0xffe75361), QFC(0x06c0f0c0), QFC(0x552f8ff7), QFC(0xf4c473c5), QFC(0x0096dcc2), QFC(0xffe80414), QFC(0x06f1825d), QFC(0x56654bdd), QFC(0xf5be0fa9), QFC(0x009da526), QFC(0xffe85b4a), QFC(0x0721bf22), QFC(0x579505f5), QFC(0xf6b1f3c3), QFC(0x00a3508f), QFC(0xffe954d0), QFC(0x075112a2), QFC(0x58befacd), QFC(0xf79fa13a), QFC(0x00a85e94), QFC(0xffea353a), QFC(0x077fedb3), QFC(0x59e2f69e), QFC(0xf887507c), QFC(0x00acbd2f), QFC(0xffeb3849), QFC(0x07ad8c26), QFC(0x5b001db8), QFC(0xf96916f5), QFC(0x00b06b68), QFC(0xffec8409), QFC(0x07da2b7f), QFC(0x5c16d0ae), QFC(0xfa44a069), QFC(0x00b36acd), QFC(0xffedc418), QFC(0x08061671), QFC(0x5d26be9b), QFC(0xfb19b7bd), QFC(0x00b58c8d), QFC(0xffef2395), QFC(0x08303897), QFC(0x5e2f6367), QFC(0xfbe8f5bd), QFC(0x00b73ab0), QFC(0xfff0e7ef), QFC(0x08594888), QFC(0x5f30ff5f), QFC(0xfcb1d740), QFC(0x00b85f70), QFC(0xfff294c3), QFC(0x0880ffdd), QFC(0x602b0c7f), QFC(0xfd7475d8), QFC(0x00b8c6b0), QFC(0xfff48700), QFC(0x08a75da4), QFC(0x611d58a3), QFC(0xfe310657), QFC(0x00b8fe0d), QFC(0xfff681d6), QFC(0x08cb4e23), QFC(0x6207f220), QFC(0xfee723c6), QFC(0x00b8394b), QFC(0xfff91fc9), QFC(0x08edfeaa), QFC(0x62ea6474), QFC(0xff96db8f), QFC(0x00b74c37), QFC(0xfffb42b0), QFC(0x090ec1fd), QFC(0x63c45243), QFC(0x0040c497), QFC(0x00b5c867), QFC(0xfffdfa24), QFC(0x092d7970), QFC(0x64964063), QFC(0x00e42fa2), QFC(0x00b3d15c), QFC(0x00007134), QFC(0x0949eaac), QFC(0x655f63f2), QFC(0x01816e06), QFC(0x00b1978d), QFC(0x00039609), QFC(0x0963ed46), QFC(0x661fd6b8), QFC(0x02186a92), QFC(0x00af374c), QFC(0x0006b1cf), QFC(0x097c1ee9), QFC(0x66d76725), QFC(0x02a99097), QFC(0x00abe79e), QFC(0x0009aa3f), QFC(0x099140a7), QFC(0x6785c24d), QFC(0x03343534), QFC(0x00a8739d), QFC(0x000d31b5), QFC(0x09a3e163), QFC(0x682b39a4), QFC(0x03b8f8dc), QFC(0x00a520bb), QFC(0x0010bc63), QFC(0x09b3d780), QFC(0x68c7269c), QFC(0x0437fb0a), QFC(0x00a1039c), QFC(0x001471f8), QFC(0x09c0e59f), QFC(0x6959709d), QFC(0x04b0adcb), QFC(0x009d10bf), QFC(0x0018703f), QFC(0x09cab9f2), QFC(0x69e29784), QFC(0x05237f9d), QFC(0x0098b855), QFC(0x001c3549), QFC(0x09d19ca9), QFC(0x6a619c5e), QFC(0x0590a67d), QFC(0x009424c6), QFC(0x002064f8), QFC(0x09d52709), QFC(0x6ad73e8e), QFC(0x05f7fb90), QFC(0x008f4bfd), QFC(0x0024dd50), QFC(0x09d5560b), QFC(0x6b42a864), QFC(0x06593912), QFC(0x008a7dd7), QFC(0x00293718), QFC(0x09d1fa23), QFC(0x6ba4629f), QFC(0x06b559c3), QFC(0x0085c217), QFC(0x002d8e42), QFC(0x09caeb0f), QFC(0x6bfbdd98), QFC(0x070bbf58), QFC(0x00807994), QFC(0x00329ab6), QFC(0x09c018cf), QFC(0x6c492217), QFC(0x075ca90c), QFC(0x007b3875), QFC(0x003745f9), QFC(0x09b18a1d), QFC(0x6c8c4c7a), QFC(0x07a8127d), QFC(0x0075fded), QFC(0x003c1fa4), QFC(0x099ec3dc), QFC(0x6cc59bab), QFC(0x07ee507c), QFC(0x0070c8a5), QFC(0x004103f5), QFC(0x09881dc5), QFC(0x6cf4073e), QFC(0x082f552e), QFC(0x006b47fa), QFC(0x00465348), QFC(0x096d0e22), QFC(0x6d18520e), QFC(0x086b1eec), QFC(0x0065fde5), QFC(0x004b6c46), QFC(0x094d7ec2), QFC(0x6d32730f), QFC(0x08a24899), QFC(0x006090c4), QFC(0x0050b177), QFC(0x09299ead), QFC(0x6d41d964), QFC(0x08d3e41b), QFC(0x005b5371), QFC(0x0055dba1), QFC(0x09015651), QFC(0x6d474e1d), QFC(0x09015651), QFC(0x0055dba1), QFC(0xfe4d1be3), QFC(0xd1c58ace), QFC(0x2e3a7532), QFC(0x01b2e41d), QFC(0x00000000), }; /* This variant of the table above is used on platforms, that have vectorized access to the table reading 4 filter sets (each of 5 coefficients) in a block. Format: 1st row flt[0] of 4 sets (e.g. set 0, 1, 2, 3) 2nd row flt[1] of 4 sets (e.g. set 0, 1, 2, 3) 3rd row flt[2] of 4 sets (e.g. set 0, 1, 2, 3) 4th row flt[3] of 4 sets (e.g. set 0, 1, 2, 3) 5th row flt[4] of 4 sets (e.g. set 0, 1, 2, 3) There are 32 blocks of 20 coefficients, in total 640. Each of the rows must be at least 64-bit aligned (see: RAM_ALIGN). */ RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_PFT qmf_pfilt640_vector[] = { /*------------- 1 .. 4 ---------------*/ QFC(0xFFEDE50E), QFC(0xFFED978A), QFC(0xFFEFC9B9), QFC(0xFFF0065D), QFC(0x01D78BFC), QFC(0x01FD3BA0), QFC(0x02244A25), QFC(0x024BF7A1), QFC(0x2FAA221C), QFC(0x311AF3A4), QFC(0x328CC6F0), QFC(0x33FF670E), QFC(0xD3337B3D), QFC(0xD49FD55F), QFC(0xD60A46E5), QFC(0xD7722F04), QFC(0xFE70B8D1), QFC(0xFE933DC0), QFC(0xFEB48D0D), QFC(0xFED4BEC3), /*------------- 5 .. 8 ---------------*/ QFC(0xFFEFF6CA), QFC(0xFFEF7B8B), QFC(0xFFEEDFA4), QFC(0xFFEE1650), QFC(0x0274BA43), QFC(0x029E35B4), QFC(0x02C89901), QFC(0x02F3E48D), QFC(0x3572EC70), QFC(0x36E69691), QFC(0x385A49C4), QFC(0x39CE0477), QFC(0xD8D7F21F), QFC(0xDA3B176A), QFC(0xDB9B5B12), QFC(0xDCF898FB), QFC(0xFEF3F6AB), QFC(0xFF120D70), QFC(0xFF2EF725), QFC(0xFF4AABC8), /*------------- 9 .. 12 ---------------*/ QFC(0xFFED651D), QFC(0xFFECC31B), QFC(0xFFEBE77B), QFC(0xFFEB50B2), QFC(0x03201116), QFC(0x034D01F1), QFC(0x037AD438), QFC(0x03A966BC), QFC(0x3B415115), QFC(0x3CB41219), QFC(0x3E25B17E), QFC(0x3F962FB8), QFC(0xDE529086), QFC(0xDFA93AB5), QFC(0xE0FC421E), QFC(0xE24B8F66), QFC(0xFF6542D1), QFC(0xFF7EE3F1), QFC(0xFF975C01), QFC(0xFFAEA5D6), /*------------- 13 .. 16 ---------------*/ QFC(0xFFEA9192), QFC(0xFFE9CA76), QFC(0xFFE940F4), QFC(0xFFE88BA8), QFC(0x03D8AFE6), QFC(0x04083FEC), QFC(0x043889C6), QFC(0x04694101), QFC(0x41058BC6), QFC(0x4272A385), QFC(0x43DE620A), QFC(0x4547DAEB), QFC(0xE396A45D), QFC(0xE4DE0CB0), QFC(0xE620C476), QFC(0xE75F8BB7), QFC(0xFFC4E365), QFC(0xFFDA17F2), QFC(0xFFEE183B), QFC(0x0000E790), /*------------- 17 .. 20 ---------------*/ QFC(0xFFE83A07), QFC(0xFFE79E16), QFC(0xFFE7746E), QFC(0xFFE6D466), QFC(0x049AA82F), QFC(0x04CC2FCF), QFC(0x04FE20BE), QFC(0x05303F88), QFC(0x46AEA856), QFC(0x4812F848), QFC(0x4973FEF2), QFC(0x4AD237A2), QFC(0xE89971B7), QFC(0xE9CEA84A), QFC(0xEAFEE7F1), QFC(0xEC2A3F5F), QFC(0x00131C75), QFC(0x0023B989), QFC(0x0033B927), QFC(0x00426F36), /*------------- 21 .. 24 ---------------*/ QFC(0xFFE6AFED), QFC(0xFFE65416), QFC(0xFFE681C6), QFC(0xFFE66DD0), QFC(0x05626209), QFC(0x05950122), QFC(0x05C76FED), QFC(0x05F9C051), QFC(0x4C2CA3DF), QFC(0x4D83976D), QFC(0x4ED62BE3), QFC(0x5024D70E), QFC(0xED50A31D), QFC(0xEE71B2FE), QFC(0xEF8D4D7B), QFC(0xF0A3959F), QFC(0x00504F41), QFC(0x005D36DF), QFC(0x006928A0), QFC(0x007400B8), /*------------- 25 .. 28 ---------------*/ QFC(0xFFE66FAB), QFC(0xFFE69423), QFC(0xFFE6FED4), QFC(0xFFE75361), QFC(0x062BF5EC), QFC(0x065DD56A), QFC(0x068F8B44), QFC(0x06C0F0C0), QFC(0x516EEFB9), QFC(0x52B449DE), QFC(0x53F495AA), QFC(0x552F8FF7), QFC(0xF1B461AB), QFC(0xF2BF6EA4), QFC(0xF3C4E887), QFC(0xF4C473C5), QFC(0x007E0393), QFC(0x00872C63), QFC(0x008F87AA), QFC(0x0096DCC2), /*------------- 29 .. 32 ---------------*/ QFC(0xFFE80414), QFC(0xFFE85B4A), QFC(0xFFE954D0), QFC(0xFFEA353A), QFC(0x06F1825D), QFC(0x0721BF22), QFC(0x075112A2), QFC(0x077FEDB3), QFC(0x56654BDD), QFC(0x579505F5), QFC(0x58BEFACD), QFC(0x59E2F69E), QFC(0xF5BE0FA9), QFC(0xF6B1F3C3), QFC(0xF79FA13A), QFC(0xF887507C), QFC(0x009DA526), QFC(0x00A3508F), QFC(0x00A85E94), QFC(0x00ACBD2F), /*------------- 33 .. 36 ---------------*/ QFC(0xFFEB3849), QFC(0xFFEC8409), QFC(0xFFEDC418), QFC(0xFFEF2395), QFC(0x07AD8C26), QFC(0x07DA2B7F), QFC(0x08061671), QFC(0x08303897), QFC(0x5B001DB8), QFC(0x5C16D0AE), QFC(0x5D26BE9B), QFC(0x5E2F6367), QFC(0xF96916F5), QFC(0xFA44A069), QFC(0xFB19B7BD), QFC(0xFBE8F5BD), QFC(0x00B06B68), QFC(0x00B36ACD), QFC(0x00B58C8D), QFC(0x00B73AB0), /*------------- 37 .. 40 ---------------*/ QFC(0xFFF0E7EF), QFC(0xFFF294C3), QFC(0xFFF48700), QFC(0xFFF681D6), QFC(0x08594888), QFC(0x0880FFDD), QFC(0x08A75DA4), QFC(0x08CB4E23), QFC(0x5F30FF5F), QFC(0x602B0C7F), QFC(0x611D58A3), QFC(0x6207F220), QFC(0xFCB1D740), QFC(0xFD7475D8), QFC(0xFE310657), QFC(0xFEE723C6), QFC(0x00B85F70), QFC(0x00B8C6B0), QFC(0x00B8FE0D), QFC(0x00B8394B), /*------------- 41 .. 44 ---------------*/ QFC(0xFFF91FC9), QFC(0xFFFB42B0), QFC(0xFFFDFA24), QFC(0x00007134), QFC(0x08EDFEAA), QFC(0x090EC1FD), QFC(0x092D7970), QFC(0x0949EAAC), QFC(0x62EA6474), QFC(0x63C45243), QFC(0x64964063), QFC(0x655F63F2), QFC(0xFF96DB8F), QFC(0x0040C497), QFC(0x00E42FA2), QFC(0x01816E06), QFC(0x00B74C37), QFC(0x00B5C867), QFC(0x00B3D15C), QFC(0x00B1978D), /*------------- 45 .. 48 ---------------*/ QFC(0x00039609), QFC(0x0006B1CF), QFC(0x0009AA3F), QFC(0x000D31B5), QFC(0x0963ED46), QFC(0x097C1EE9), QFC(0x099140A7), QFC(0x09A3E163), QFC(0x661FD6B8), QFC(0x66D76725), QFC(0x6785C24D), QFC(0x682B39A4), QFC(0x02186A92), QFC(0x02A99097), QFC(0x03343534), QFC(0x03B8F8DC), QFC(0x00AF374C), QFC(0x00ABE79E), QFC(0x00A8739D), QFC(0x00A520BB), /*------------- 49 .. 52 ---------------*/ QFC(0x0010BC63), QFC(0x001471F8), QFC(0x0018703F), QFC(0x001C3549), QFC(0x09B3D780), QFC(0x09C0E59F), QFC(0x09CAB9F2), QFC(0x09D19CA9), QFC(0x68C7269C), QFC(0x6959709D), QFC(0x69E29784), QFC(0x6A619C5E), QFC(0x0437FB0A), QFC(0x04B0ADCB), QFC(0x05237F9D), QFC(0x0590A67D), QFC(0x00A1039C), QFC(0x009D10BF), QFC(0x0098B855), QFC(0x009424C6), /*------------- 53 .. 56 ---------------*/ QFC(0x002064F8), QFC(0x0024DD50), QFC(0x00293718), QFC(0x002D8E42), QFC(0x09D52709), QFC(0x09D5560B), QFC(0x09D1FA23), QFC(0x09CAEB0F), QFC(0x6AD73E8E), QFC(0x6B42A864), QFC(0x6BA4629F), QFC(0x6BFBDD98), QFC(0x05F7FB90), QFC(0x06593912), QFC(0x06B559C3), QFC(0x070BBF58), QFC(0x008F4BFD), QFC(0x008A7DD7), QFC(0x0085C217), QFC(0x00807994), /*------------- 57 .. 60 ---------------*/ QFC(0x00329AB6), QFC(0x003745F9), QFC(0x003C1FA4), QFC(0x004103F5), QFC(0x09C018CF), QFC(0x09B18A1D), QFC(0x099EC3DC), QFC(0x09881DC5), QFC(0x6C492217), QFC(0x6C8C4C7A), QFC(0x6CC59BAB), QFC(0x6CF4073E), QFC(0x075CA90C), QFC(0x07A8127D), QFC(0x07EE507C), QFC(0x082F552E), QFC(0x007B3875), QFC(0x0075FDED), QFC(0x0070C8A5), QFC(0x006B47FA), /*------------- 61 .. 64 ---------------*/ QFC(0x00465348), QFC(0x004B6C46), QFC(0x0050B177), QFC(0x0055DBA1), QFC(0x096D0E22), QFC(0x094D7EC2), QFC(0x09299EAD), QFC(0x09015651), QFC(0x6D18520E), QFC(0x6D32730F), QFC(0x6D41D964), QFC(0x6D474E1D), QFC(0x086B1EEC), QFC(0x08A24899), QFC(0x08D3E41B), QFC(0x09015651), QFC(0x0065FDE5), QFC(0x006090C4), QFC(0x005B5371), QFC(0x0055DBA1), /*------------- 63 .. 60 ---------------*/ QFC(0x005B5371), QFC(0x006090C4), QFC(0x0065FDE5), QFC(0x006B47FA), QFC(0x08D3E41B), QFC(0x08A24899), QFC(0x086B1EEC), QFC(0x082F552E), QFC(0x6D41D964), QFC(0x6D32730F), QFC(0x6D18520E), QFC(0x6CF4073E), QFC(0x09299EAD), QFC(0x094D7EC2), QFC(0x096D0E22), QFC(0x09881DC5), QFC(0x0050B177), QFC(0x004B6C46), QFC(0x00465348), QFC(0x004103F5), /*------------- 59 .. 56 ---------------*/ QFC(0x0070C8A5), QFC(0x0075FDED), QFC(0x007B3875), QFC(0x00807994), QFC(0x07EE507C), QFC(0x07A8127D), QFC(0x075CA90C), QFC(0x070BBF58), QFC(0x6CC59BAB), QFC(0x6C8C4C7A), QFC(0x6C492217), QFC(0x6BFBDD98), QFC(0x099EC3DC), QFC(0x09B18A1D), QFC(0x09C018CF), QFC(0x09CAEB0F), QFC(0x003C1FA4), QFC(0x003745F9), QFC(0x00329AB6), QFC(0x002D8E42), /*------------- 55 .. 52 ---------------*/ QFC(0x0085C217), QFC(0x008A7DD7), QFC(0x008F4BFD), QFC(0x009424C6), QFC(0x06B559C3), QFC(0x06593912), QFC(0x05F7FB90), QFC(0x0590A67D), QFC(0x6BA4629F), QFC(0x6B42A864), QFC(0x6AD73E8E), QFC(0x6A619C5E), QFC(0x09D1FA23), QFC(0x09D5560B), QFC(0x09D52709), QFC(0x09D19CA9), QFC(0x00293718), QFC(0x0024DD50), QFC(0x002064F8), QFC(0x001C3549), /*------------- 51 .. 48 ---------------*/ QFC(0x0098B855), QFC(0x009D10BF), QFC(0x00A1039C), QFC(0x00A520BB), QFC(0x05237F9D), QFC(0x04B0ADCB), QFC(0x0437FB0A), QFC(0x03B8F8DC), QFC(0x69E29784), QFC(0x6959709D), QFC(0x68C7269C), QFC(0x682B39A4), QFC(0x09CAB9F2), QFC(0x09C0E59F), QFC(0x09B3D780), QFC(0x09A3E163), QFC(0x0018703F), QFC(0x001471F8), QFC(0x0010BC63), QFC(0x000D31B5), /*------------- 47 .. 44 ---------------*/ QFC(0x00A8739D), QFC(0x00ABE79E), QFC(0x00AF374C), QFC(0x00B1978D), QFC(0x03343534), QFC(0x02A99097), QFC(0x02186A92), QFC(0x01816E06), QFC(0x6785C24D), QFC(0x66D76725), QFC(0x661FD6B8), QFC(0x655F63F2), QFC(0x099140A7), QFC(0x097C1EE9), QFC(0x0963ED46), QFC(0x0949EAAC), QFC(0x0009AA3F), QFC(0x0006B1CF), QFC(0x00039609), QFC(0x00007134), /*------------- 43 .. 40 ---------------*/ QFC(0x00B3D15C), QFC(0x00B5C867), QFC(0x00B74C37), QFC(0x00B8394B), QFC(0x00E42FA2), QFC(0x0040C497), QFC(0xFF96DB8F), QFC(0xFEE723C6), QFC(0x64964063), QFC(0x63C45243), QFC(0x62EA6474), QFC(0x6207F220), QFC(0x092D7970), QFC(0x090EC1FD), QFC(0x08EDFEAA), QFC(0x08CB4E23), QFC(0xFFFDFA24), QFC(0xFFFB42B0), QFC(0xFFF91FC9), QFC(0xFFF681D6), /*------------- 39 .. 36 ---------------*/ QFC(0x00B8FE0D), QFC(0x00B8C6B0), QFC(0x00B85F70), QFC(0x00B73AB0), QFC(0xFE310657), QFC(0xFD7475D8), QFC(0xFCB1D740), QFC(0xFBE8F5BD), QFC(0x611D58A3), QFC(0x602B0C7F), QFC(0x5F30FF5F), QFC(0x5E2F6367), QFC(0x08A75DA4), QFC(0x0880FFDD), QFC(0x08594888), QFC(0x08303897), QFC(0xFFF48700), QFC(0xFFF294C3), QFC(0xFFF0E7EF), QFC(0xFFEF2395), /*------------- 35 .. 32 ---------------*/ QFC(0x00B58C8D), QFC(0x00B36ACD), QFC(0x00B06B68), QFC(0x00ACBD2F), QFC(0xFB19B7BD), QFC(0xFA44A069), QFC(0xF96916F5), QFC(0xF887507C), QFC(0x5D26BE9B), QFC(0x5C16D0AE), QFC(0x5B001DB8), QFC(0x59E2F69E), QFC(0x08061671), QFC(0x07DA2B7F), QFC(0x07AD8C26), QFC(0x077FEDB3), QFC(0xFFEDC418), QFC(0xFFEC8409), QFC(0xFFEB3849), QFC(0xFFEA353A), /*------------- 31 .. 28 ---------------*/ QFC(0x00A85E94), QFC(0x00A3508F), QFC(0x009DA526), QFC(0x0096DCC2), QFC(0xF79FA13A), QFC(0xF6B1F3C3), QFC(0xF5BE0FA9), QFC(0xF4C473C5), QFC(0x58BEFACD), QFC(0x579505F5), QFC(0x56654BDD), QFC(0x552F8FF7), QFC(0x075112A2), QFC(0x0721BF22), QFC(0x06F1825D), QFC(0x06C0F0C0), QFC(0xFFE954D0), QFC(0xFFE85B4A), QFC(0xFFE80414), QFC(0xFFE75361), /*------------- 27 .. 24 ---------------*/ QFC(0x008F87AA), QFC(0x00872C63), QFC(0x007E0393), QFC(0x007400B8), QFC(0xF3C4E887), QFC(0xF2BF6EA4), QFC(0xF1B461AB), QFC(0xF0A3959F), QFC(0x53F495AA), QFC(0x52B449DE), QFC(0x516EEFB9), QFC(0x5024D70E), QFC(0x068F8B44), QFC(0x065DD56A), QFC(0x062BF5EC), QFC(0x05F9C051), QFC(0xFFE6FED4), QFC(0xFFE69423), QFC(0xFFE66FAB), QFC(0xFFE66DD0), /*------------- 23 .. 20 ---------------*/ QFC(0x006928A0), QFC(0x005D36DF), QFC(0x00504F41), QFC(0x00426F36), QFC(0xEF8D4D7B), QFC(0xEE71B2FE), QFC(0xED50A31D), QFC(0xEC2A3F5F), QFC(0x4ED62BE3), QFC(0x4D83976D), QFC(0x4C2CA3DF), QFC(0x4AD237A2), QFC(0x05C76FED), QFC(0x05950122), QFC(0x05626209), QFC(0x05303F88), QFC(0xFFE681C6), QFC(0xFFE65416), QFC(0xFFE6AFED), QFC(0xFFE6D466), /*------------- 19 .. 16 ---------------*/ QFC(0x0033B927), QFC(0x0023B989), QFC(0x00131C75), QFC(0x0000E790), QFC(0xEAFEE7F1), QFC(0xE9CEA84A), QFC(0xE89971B7), QFC(0xE75F8BB7), QFC(0x4973FEF2), QFC(0x4812F848), QFC(0x46AEA856), QFC(0x4547DAEB), QFC(0x04FE20BE), QFC(0x04CC2FCF), QFC(0x049AA82F), QFC(0x04694101), QFC(0xFFE7746E), QFC(0xFFE79E16), QFC(0xFFE83A07), QFC(0xFFE88BA8), /*------------- 15 .. 12 ---------------*/ QFC(0xFFEE183B), QFC(0xFFDA17F2), QFC(0xFFC4E365), QFC(0xFFAEA5D6), QFC(0xE620C476), QFC(0xE4DE0CB0), QFC(0xE396A45D), QFC(0xE24B8F66), QFC(0x43DE620A), QFC(0x4272A385), QFC(0x41058BC6), QFC(0x3F962FB8), QFC(0x043889C6), QFC(0x04083FEC), QFC(0x03D8AFE6), QFC(0x03A966BC), QFC(0xFFE940F4), QFC(0xFFE9CA76), QFC(0xFFEA9192), QFC(0xFFEB50B2), /*------------- 11 .. 8 ---------------*/ QFC(0xFF975C01), QFC(0xFF7EE3F1), QFC(0xFF6542D1), QFC(0xFF4AABC8), QFC(0xE0FC421E), QFC(0xDFA93AB5), QFC(0xDE529086), QFC(0xDCF898FB), QFC(0x3E25B17E), QFC(0x3CB41219), QFC(0x3B415115), QFC(0x39CE0477), QFC(0x037AD438), QFC(0x034D01F1), QFC(0x03201116), QFC(0x02F3E48D), QFC(0xFFEBE77B), QFC(0xFFECC31B), QFC(0xFFED651D), QFC(0xFFEE1650), /*------------- 7 .. 4 ---------------*/ QFC(0xFF2EF725), QFC(0xFF120D70), QFC(0xFEF3F6AB), QFC(0xFED4BEC3), QFC(0xDB9B5B12), QFC(0xDA3B176A), QFC(0xD8D7F21F), QFC(0xD7722F04), QFC(0x385A49C4), QFC(0x36E69691), QFC(0x3572EC70), QFC(0x33FF670E), QFC(0x02C89901), QFC(0x029E35B4), QFC(0x0274BA43), QFC(0x024BF7A1), QFC(0xFFEEDFA4), QFC(0xFFEF7B8B), QFC(0xFFEFF6CA), QFC(0xFFF0065D), /*------------- 3 .. 0 ---------------*/ QFC(0xFEB48D0D), QFC(0xFE933DC0), QFC(0xFE70B8D1), QFC(0xFE4D1BE3), QFC(0xD60A46E5), QFC(0xD49FD55F), QFC(0xD3337B3D), QFC(0xD1C58ACE), QFC(0x328CC6F0), QFC(0x311AF3A4), QFC(0x2FAA221C), QFC(0x2E3A7532), QFC(0x02244A25), QFC(0x01FD3BA0), QFC(0x01D78BFC), QFC(0x01B2E41D), QFC(0xFFEFC9B9), QFC(0xFFED978A), QFC(0xFFEDE50E), QFC(0x00000000), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_QTW qmf_phaseshift_cos64[] = { QTC(0x7ff62182), QTC(0x7fa736b4), QTC(0x7f0991c4), QTC(0x7e1d93ea), QTC(0x7ce3ceb2), QTC(0x7b5d039e), QTC(0x798a23b1), QTC(0x776c4edb), QTC(0x7504d345), QTC(0x72552c85), QTC(0x6f5f02b2), QTC(0x6c242960), QTC(0x68a69e81), QTC(0x64e88926), QTC(0x60ec3830), QTC(0x5cb420e0), QTC(0x5842dd54), QTC(0x539b2af0), QTC(0x4ebfe8a5), QTC(0x49b41533), QTC(0x447acd50), QTC(0x3f1749b8), QTC(0x398cdd32), QTC(0x33def287), QTC(0x2e110a62), QTC(0x2826b928), QTC(0x2223a4c5), QTC(0x1c0b826a), QTC(0x15e21445), QTC(0x0fab272b), QTC(0x096a9049), QTC(0x03242abf), QTC(0xfcdbd541), QTC(0xf6956fb7), QTC(0xf054d8d5), QTC(0xea1debbb), QTC(0xe3f47d96), QTC(0xdddc5b3b), QTC(0xd7d946d8), QTC(0xd1eef59e), QTC(0xcc210d79), QTC(0xc67322ce), QTC(0xc0e8b648), QTC(0xbb8532b0), QTC(0xb64beacd), QTC(0xb140175b), QTC(0xac64d510), QTC(0xa7bd22ac), QTC(0xa34bdf20), QTC(0x9f13c7d0), QTC(0x9b1776da), QTC(0x9759617f), QTC(0x93dbd6a0), QTC(0x90a0fd4e), QTC(0x8daad37b), QTC(0x8afb2cbb), QTC(0x8893b125), QTC(0x8675dc4f), QTC(0x84a2fc62), QTC(0x831c314e), QTC(0x81e26c16), QTC(0x80f66e3c), QTC(0x8058c94c), QTC(0x8009de7e), }; RAM_ALIGN LNK_SECTION_CONSTDATA const FIXP_QTW qmf_phaseshift_sin64[] = { QTC(0x03242abf), QTC(0x096a9049), QTC(0x0fab272b), QTC(0x15e21445), QTC(0x1c0b826a), QTC(0x2223a4c5), QTC(0x2826b928), QTC(0x2e110a62), QTC(0x33def287), QTC(0x398cdd32), QTC(0x3f1749b8), QTC(0x447acd50), QTC(0x49b41533), QTC(0x4ebfe8a5), QTC(0x539b2af0), QTC(0x5842dd54), QTC(0x5cb420e0), QTC(0x60ec3830), QTC(0x64e88926), QTC(0x68a69e81), QTC(0x6c242960), QTC(0x6f5f02b2), QTC(0x72552c85), QTC(0x7504d345), QTC(0x776c4edb), QTC(0x798a23b1), QTC(0x7b5d039e), QTC(0x7ce3ceb2), QTC(0x7e1d93ea), QTC(0x7f0991c4), QTC(0x7fa736b4), QTC(0x7ff62182), QTC(0x7ff62182), QTC(0x7fa736b4), QTC(0x7f0991c4), QTC(0x7e1d93ea), QTC(0x7ce3ceb2), QTC(0x7b5d039e), QTC(0x798a23b1), QTC(0x776c4edb), QTC(0x7504d345), QTC(0x72552c85), QTC(0x6f5f02b2), QTC(0x6c242960), QTC(0x68a69e81), QTC(0x64e88926), QTC(0x60ec3830), QTC(0x5cb420e0), QTC(0x5842dd54), QTC(0x539b2af0), QTC(0x4ebfe8a5), QTC(0x49b41533), QTC(0x447acd50), QTC(0x3f1749b8), QTC(0x398cdd32), QTC(0x33def287), QTC(0x2e110a62), QTC(0x2826b928), QTC(0x2223a4c5), QTC(0x1c0b826a), QTC(0x15e21445), QTC(0x0fab272b), QTC(0x096a9049), QTC(0x03242abf), }; /* * Low Delay QMF aka CLDFB */ #if defined(QMF_COEFF_16BIT) #define QTCFLLD(x) FL2FXCONST_SGL(x / (float)(1 << QMF_CLDFB_PFT_SCALE)) #define QTCFLLDT(x) FL2FXCONST_SGL(x) #else #define QTCFLLD(x) FL2FXCONST_DBL(x / (float)(1 << QMF_CLDFB_PFT_SCALE)) #define QTCFLLDT(x) FL2FXCONST_DBL(x) #endif #ifndef LOW_POWER_SBR_ONLY /*! \name QMF-Twiddle \brief QMF twiddle factors L=32, gain=2.0, angle = 0.75 */ /* sin/cos (angle) / 2 */ const FIXP_QTW qmf_phaseshift_cos32_cldfb_ana[32] = { /* analysis twiddle table */ QTCFLLDT(-7.071067e-01), QTCFLLDT(7.071070e-01), QTCFLLDT(7.071064e-01), QTCFLLDT(-7.071073e-01), QTCFLLDT(-7.071061e-01), QTCFLLDT(7.071076e-01), QTCFLLDT(7.071058e-01), QTCFLLDT(-7.071080e-01), QTCFLLDT(-7.071055e-01), QTCFLLDT(7.071083e-01), QTCFLLDT(7.071052e-01), QTCFLLDT(-7.071086e-01), QTCFLLDT(-7.071049e-01), QTCFLLDT(7.071089e-01), QTCFLLDT(7.071046e-01), QTCFLLDT(-7.071092e-01), QTCFLLDT(-7.071042e-01), QTCFLLDT(7.071095e-01), QTCFLLDT(7.071039e-01), QTCFLLDT(-7.071098e-01), QTCFLLDT(-7.071036e-01), QTCFLLDT(7.071101e-01), QTCFLLDT(7.071033e-01), QTCFLLDT(-7.071104e-01), QTCFLLDT(-7.071030e-01), QTCFLLDT(7.071107e-01), QTCFLLDT(7.071027e-01), QTCFLLDT(-7.071111e-01), QTCFLLDT(-7.071024e-01), QTCFLLDT(7.071114e-01), QTCFLLDT(7.071021e-01), QTCFLLDT(-7.071117e-01), }; const FIXP_QTW qmf_phaseshift_cos32_cldfb_syn[32] = { /* synthesis twiddle table */ QTCFLLDT(7.071067e-01), QTCFLLDT(-7.071070e-01), QTCFLLDT(-7.071064e-01), QTCFLLDT(7.071073e-01), QTCFLLDT(7.071061e-01), QTCFLLDT(-7.071076e-01), QTCFLLDT(-7.071058e-01), QTCFLLDT(7.071080e-01), QTCFLLDT(7.071055e-01), QTCFLLDT(-7.071083e-01), QTCFLLDT(-7.071052e-01), QTCFLLDT(7.071086e-01), QTCFLLDT(7.071049e-01), QTCFLLDT(-7.071089e-01), QTCFLLDT(-7.071046e-01), QTCFLLDT(7.071092e-01), QTCFLLDT(7.071042e-01), QTCFLLDT(-7.071095e-01), QTCFLLDT(-7.071039e-01), QTCFLLDT(7.071098e-01), QTCFLLDT(7.071036e-01), QTCFLLDT(-7.071101e-01), QTCFLLDT(-7.071033e-01), QTCFLLDT(7.071104e-01), QTCFLLDT(7.071030e-01), QTCFLLDT(-7.071107e-01), QTCFLLDT(-7.071027e-01), QTCFLLDT(7.071111e-01), QTCFLLDT(7.071024e-01), QTCFLLDT(-7.071114e-01), QTCFLLDT(-7.071021e-01), QTCFLLDT(7.071117e-01), }; const FIXP_QTW qmf_phaseshift_sin32_cldfb[32] = { QTCFLLDT(7.071068e-01), QTCFLLDT(7.071065e-01), QTCFLLDT(-7.071072e-01), QTCFLLDT(-7.071062e-01), QTCFLLDT(7.071075e-01), QTCFLLDT(7.071059e-01), QTCFLLDT(-7.071078e-01), QTCFLLDT(-7.071056e-01), QTCFLLDT(7.071081e-01), QTCFLLDT(7.071053e-01), QTCFLLDT(-7.071084e-01), QTCFLLDT(-7.071050e-01), QTCFLLDT(7.071087e-01), QTCFLLDT(7.071047e-01), QTCFLLDT(-7.071090e-01), QTCFLLDT(-7.071044e-01), QTCFLLDT(7.071093e-01), QTCFLLDT(7.071041e-01), QTCFLLDT(-7.071096e-01), QTCFLLDT(-7.071038e-01), QTCFLLDT(7.071099e-01), QTCFLLDT(7.071034e-01), QTCFLLDT(-7.071103e-01), QTCFLLDT(-7.071031e-01), QTCFLLDT(7.071106e-01), QTCFLLDT(7.071028e-01), QTCFLLDT(-7.071109e-01), QTCFLLDT(-7.071025e-01), QTCFLLDT(7.071112e-01), QTCFLLDT(7.071022e-01), QTCFLLDT(-7.071115e-01), QTCFLLDT(-7.071019e-01), }; /* twiddles for X=(8,16) band qmf are copied from float simpleplayer * implementation: qmf_phaseshift_cosX_cldfb_ana = * QMFlib_twiddle3RealX_SBRLD_A qmf_phaseshift_cosX_cldfb_syn = * -(QMFlib_twiddle3RealX_SBRLD_A) qmf_phaseshift_sinX_cldfb = * QMFlib_twiddle3ImagX_SBRLD_A */ /* cos ((n + 0.5)*pi*angle/L) , order = 159, L=16 */ const FIXP_QTW qmf_phaseshift_cos16_cldfb_ana[16] = { QTCFLLDT(-0.7071067812), QTCFLLDT(0.7071067812), QTCFLLDT(0.7071067812), QTCFLLDT(-0.7071067812), QTCFLLDT(-0.7071067812), QTCFLLDT(0.7071067812), QTCFLLDT(0.7071067812), QTCFLLDT(-0.7071067812), QTCFLLDT(-0.7071067812), QTCFLLDT(0.7071067812), QTCFLLDT(0.7071067812), QTCFLLDT(-0.7071067812), QTCFLLDT(-0.7071067812), QTCFLLDT(0.7071067812), QTCFLLDT(0.7071067812), QTCFLLDT(-0.7071067812), }; /* cos ((n + 0.5)*pi*angle/L) , order = 159, L=16 */ const FIXP_QTW qmf_phaseshift_cos16_cldfb_syn[16] = { QTCFLLDT(0.7071067812), QTCFLLDT(-0.7071067812), QTCFLLDT(-0.7071067812), QTCFLLDT(0.7071067812), QTCFLLDT(0.7071067812), QTCFLLDT(-0.7071067812), QTCFLLDT(-0.7071067812), QTCFLLDT(0.7071067812), QTCFLLDT(0.7071067812), QTCFLLDT(-0.7071067812), QTCFLLDT(-0.7071067812), QTCFLLDT(0.7071067812), QTCFLLDT(0.7071067812), QTCFLLDT(-0.7071067812), QTCFLLDT(-0.7071067812), QTCFLLDT(0.7071067812), }; /* sin ((n + 0.5)*pi*angle/L) , order = 159, L=16 */ const FIXP_QTW qmf_phaseshift_sin16_cldfb[16] = { QTCFLLDT(0.7071067812), QTCFLLDT(0.7071067812), QTCFLLDT(-0.7071067812), QTCFLLDT(-0.7071067812), QTCFLLDT(0.7071067812), QTCFLLDT(0.7071067812), QTCFLLDT(-0.7071067812), QTCFLLDT(-0.7071067812), QTCFLLDT(0.7071067812), QTCFLLDT(0.7071067812), QTCFLLDT(-0.7071067812), QTCFLLDT(-0.7071067812), QTCFLLDT(0.7071067812), QTCFLLDT(0.7071067812), QTCFLLDT(-0.7071067812), QTCFLLDT(-0.7071067812), }; /* cos ((n + 0.5)*pi*angle/L) , order = 79, L=8 */ const FIXP_QTW qmf_phaseshift_cos8_cldfb_ana[8] = { QTCFLLDT(-0.7071067812), QTCFLLDT(0.7071067812), QTCFLLDT(0.7071067812), QTCFLLDT(-0.7071067812), QTCFLLDT(-0.7071067812), QTCFLLDT(0.7071067812), QTCFLLDT(0.7071067812), QTCFLLDT(-0.7071067812), }; const FIXP_QTW qmf_phaseshift_cos8_cldfb_syn[8] = { QTCFLLDT(0.7071067812), QTCFLLDT(-0.7071067812), QTCFLLDT(-0.7071067812), QTCFLLDT(0.7071067812), QTCFLLDT(0.7071067812), QTCFLLDT(-0.7071067812), QTCFLLDT(-0.7071067812), QTCFLLDT(0.7071067812), }; /* sin ((n + 0.5)*pi*angle/L) , order = 79, L=8 */ const FIXP_QTW qmf_phaseshift_sin8_cldfb[8] = { QTCFLLDT(0.7071067812), QTCFLLDT(0.7071067812), QTCFLLDT(-0.7071067812), QTCFLLDT(-0.7071067812), QTCFLLDT(0.7071067812), QTCFLLDT(0.7071067812), QTCFLLDT(-0.7071067812), QTCFLLDT(-0.7071067812), }; /* sin/cos (angle) / 128 */ const FIXP_QTW qmf_phaseshift_cos64_cldfb[64] = { QTCFLLDT(7.071068e-01), QTCFLLDT(-7.071066e-01), QTCFLLDT(-7.071070e-01), QTCFLLDT(7.071065e-01), QTCFLLDT(7.071072e-01), QTCFLLDT(-7.071063e-01), QTCFLLDT(-7.071074e-01), QTCFLLDT(7.071061e-01), QTCFLLDT(7.071075e-01), QTCFLLDT(-7.071059e-01), QTCFLLDT(-7.071078e-01), QTCFLLDT(7.071057e-01), QTCFLLDT(7.071080e-01), QTCFLLDT(-7.071055e-01), QTCFLLDT(-7.071081e-01), QTCFLLDT(7.071053e-01), QTCFLLDT(7.071083e-01), QTCFLLDT(-7.071052e-01), QTCFLLDT(-7.071085e-01), QTCFLLDT(7.071050e-01), QTCFLLDT(7.071087e-01), QTCFLLDT(-7.071048e-01), QTCFLLDT(-7.071089e-01), QTCFLLDT(7.071046e-01), QTCFLLDT(7.071090e-01), QTCFLLDT(-7.071044e-01), QTCFLLDT(-7.071092e-01), QTCFLLDT(7.071042e-01), QTCFLLDT(7.071095e-01), QTCFLLDT(-7.071040e-01), QTCFLLDT(-7.071096e-01), QTCFLLDT(7.071038e-01), QTCFLLDT(7.071098e-01), QTCFLLDT(-7.071037e-01), QTCFLLDT(-7.071100e-01), QTCFLLDT(7.071035e-01), QTCFLLDT(7.071102e-01), QTCFLLDT(-7.071033e-01), QTCFLLDT(-7.071103e-01), QTCFLLDT(7.071031e-01), QTCFLLDT(7.071105e-01), QTCFLLDT(-7.071030e-01), QTCFLLDT(-7.071107e-01), QTCFLLDT(7.071028e-01), QTCFLLDT(7.071109e-01), QTCFLLDT(-7.071025e-01), QTCFLLDT(-7.071111e-01), QTCFLLDT(7.071024e-01), QTCFLLDT(7.071113e-01), QTCFLLDT(-7.071022e-01), QTCFLLDT(-7.071115e-01), QTCFLLDT(7.071020e-01), QTCFLLDT(7.071117e-01), QTCFLLDT(-7.071018e-01), QTCFLLDT(-7.071118e-01), QTCFLLDT(7.071016e-01), QTCFLLDT(7.071120e-01), QTCFLLDT(-7.071015e-01), QTCFLLDT(-7.071122e-01), QTCFLLDT(7.071013e-01), QTCFLLDT(7.071124e-01), QTCFLLDT(-7.071011e-01), QTCFLLDT(-7.071126e-01), QTCFLLDT(7.071009e-01), }; const FIXP_QTW qmf_phaseshift_sin64_cldfb[64] = { QTCFLLDT(7.071067e-01), QTCFLLDT(7.071069e-01), QTCFLLDT(-7.071065e-01), QTCFLLDT(-7.071071e-01), QTCFLLDT(7.071064e-01), QTCFLLDT(7.071073e-01), QTCFLLDT(-7.071062e-01), QTCFLLDT(-7.071075e-01), QTCFLLDT(7.071060e-01), QTCFLLDT(7.071077e-01), QTCFLLDT(-7.071058e-01), QTCFLLDT(-7.071078e-01), QTCFLLDT(7.071056e-01), QTCFLLDT(7.071080e-01), QTCFLLDT(-7.071055e-01), QTCFLLDT(-7.071082e-01), QTCFLLDT(7.071053e-01), QTCFLLDT(7.071084e-01), QTCFLLDT(-7.071050e-01), QTCFLLDT(-7.071086e-01), QTCFLLDT(7.071049e-01), QTCFLLDT(7.071088e-01), QTCFLLDT(-7.071047e-01), QTCFLLDT(-7.071090e-01), QTCFLLDT(7.071045e-01), QTCFLLDT(7.071092e-01), QTCFLLDT(-7.071043e-01), QTCFLLDT(-7.071093e-01), QTCFLLDT(7.071041e-01), QTCFLLDT(7.071095e-01), QTCFLLDT(-7.071040e-01), QTCFLLDT(-7.071097e-01), QTCFLLDT(7.071038e-01), QTCFLLDT(7.071099e-01), QTCFLLDT(-7.071036e-01), QTCFLLDT(-7.071100e-01), QTCFLLDT(7.071034e-01), QTCFLLDT(7.071103e-01), QTCFLLDT(-7.071032e-01), QTCFLLDT(-7.071105e-01), QTCFLLDT(7.071030e-01), QTCFLLDT(7.071106e-01), QTCFLLDT(-7.071028e-01), QTCFLLDT(-7.071108e-01), QTCFLLDT(7.071027e-01), QTCFLLDT(7.071110e-01), QTCFLLDT(-7.071025e-01), QTCFLLDT(-7.071112e-01), QTCFLLDT(7.071023e-01), QTCFLLDT(7.071114e-01), QTCFLLDT(-7.071021e-01), QTCFLLDT(-7.071115e-01), QTCFLLDT(7.071019e-01), QTCFLLDT(7.071117e-01), QTCFLLDT(-7.071017e-01), QTCFLLDT(-7.071120e-01), QTCFLLDT(7.071015e-01), QTCFLLDT(7.071121e-01), QTCFLLDT(-7.071013e-01), QTCFLLDT(-7.071123e-01), QTCFLLDT(7.071012e-01), QTCFLLDT(7.071125e-01), QTCFLLDT(-7.071010e-01), QTCFLLDT(-7.071127e-01), }; //@} #endif /* #ifdef LOW_POWER_SBR_ONLY */ /*! \name QMF \brief QMF-Table 64 channels, N = 640, optimized by PE 010516 The coeffs are rearranged compared with the reference in the following way: sbr_qmf_64[0] = sbr_qmf_64_reference[0]; sbr_qmf_64[1] = sbr_qmf_64_reference[128]; sbr_qmf_64[2] = sbr_qmf_64_reference[256]; sbr_qmf_64[3] = sbr_qmf_64_reference[384]; sbr_qmf_64[4] = sbr_qmf_64_reference[512]; sbr_qmf_64[5] = sbr_qmf_64_reference[1]; sbr_qmf_64[6] = sbr_qmf_64_reference[129]; sbr_qmf_64[7] = sbr_qmf_64_reference[257]; sbr_qmf_64[8] = sbr_qmf_64_reference[385]; sbr_qmf_64[9] = sbr_qmf_64_reference[513]; . . . sbr_qmf_64[635] = sbr_qmf_64_reference[127] sbr_qmf_64[636] = sbr_qmf_64_reference[255]; sbr_qmf_64[637] = sbr_qmf_64_reference[383]; sbr_qmf_64[638] = sbr_qmf_64_reference[511]; sbr_qmf_64[639] = sbr_qmf_64_reference[639]; Symmetric properties of qmf coeffs: Use point symmetry: sbr_qmf_64_640_qmf[320..634] = p_64_640_qmf[314..0] Max sum of all FIR filter absolute coefficients is: 0x7FF5B201 thus, the filter output is not required to be scaled. \showinitializer */ //@{ LNK_SECTION_CONSTDATA_L1 RAM_ALIGN const FIXP_PFT qmf_cldfb_640[QMF640_CLDFB_PFT_TABLE_SIZE] = { QTCFLLD(6.571760e-07), QTCFLLD(-8.010079e-06), QTCFLLD(-1.250743e-03), QTCFLLD(8.996371e-03), QTCFLLD(5.128557e-01), QTCFLLD(4.118360e-07), QTCFLLD(-1.469933e-05), QTCFLLD(-1.194743e-03), QTCFLLD(9.640299e-03), QTCFLLD(5.299510e-01), QTCFLLD(8.109952e-07), QTCFLLD(4.840578e-06), QTCFLLD(-1.151796e-03), QTCFLLD(1.033126e-02), QTCFLLD(5.470652e-01), QTCFLLD(7.099633e-07), QTCFLLD(7.167101e-06), QTCFLLD(-1.099001e-03), QTCFLLD(1.106959e-02), QTCFLLD(5.641523e-01), QTCFLLD(6.834210e-07), QTCFLLD(1.088325e-05), QTCFLLD(-1.047655e-03), QTCFLLD(1.186211e-02), QTCFLLD(5.811993e-01), QTCFLLD(4.292862e-07), QTCFLLD(1.013260e-05), QTCFLLD(-9.862027e-04), QTCFLLD(1.270747e-02), QTCFLLD(5.981877e-01), QTCFLLD(-5.426597e-09), QTCFLLD(5.869707e-06), QTCFLLD(-9.294665e-04), QTCFLLD(1.361072e-02), QTCFLLD(6.151031e-01), QTCFLLD(6.355303e-08), QTCFLLD(1.125135e-05), QTCFLLD(-9.767709e-04), QTCFLLD(1.456209e-02), QTCFLLD(6.319284e-01), QTCFLLD(5.490570e-07), QTCFLLD(2.015445e-05), QTCFLLD(-1.040598e-03), QTCFLLD(1.557759e-02), QTCFLLD(6.486438e-01), QTCFLLD(1.620171e-06), QTCFLLD(2.800456e-05), QTCFLLD(-1.146268e-03), QTCFLLD(1.665188e-02), QTCFLLD(6.652304e-01), QTCFLLD(-6.025110e-10), QTCFLLD(8.975978e-06), QTCFLLD(-1.292866e-03), QTCFLLD(1.778249e-02), QTCFLLD(6.816668e-01), QTCFLLD(-6.325664e-10), QTCFLLD(8.563820e-06), QTCFLLD(-1.196638e-03), QTCFLLD(1.897506e-02), QTCFLLD(6.979337e-01), QTCFLLD(-4.013525e-09), QTCFLLD(1.168895e-05), QTCFLLD(-9.726699e-04), QTCFLLD(2.023525e-02), QTCFLLD(7.140087e-01), QTCFLLD(-4.244091e-09), QTCFLLD(7.300589e-06), QTCFLLD(-8.029620e-04), QTCFLLD(2.156305e-02), QTCFLLD(7.298746e-01), QTCFLLD(-1.846548e-08), QTCFLLD(3.965364e-06), QTCFLLD(-6.754936e-04), QTCFLLD(2.296471e-02), QTCFLLD(7.455112e-01), QTCFLLD(-3.870537e-09), QTCFLLD(1.374896e-06), QTCFLLD(-5.791145e-04), QTCFLLD(2.443434e-02), QTCFLLD(7.609051e-01), QTCFLLD(-8.883499e-10), QTCFLLD(3.798520e-07), QTCFLLD(-4.733148e-04), QTCFLLD(2.597957e-02), QTCFLLD(7.760386e-01), QTCFLLD(5.303528e-08), QTCFLLD(4.469729e-06), QTCFLLD(-2.998740e-04), QTCFLLD(2.760091e-02), QTCFLLD(7.908995e-01), QTCFLLD(7.391974e-08), QTCFLLD(2.461877e-05), QTCFLLD(7.882620e-05), QTCFLLD(2.931526e-02), QTCFLLD(8.054701e-01), QTCFLLD(1.723217e-09), QTCFLLD(4.005269e-05), QTCFLLD(4.708010e-04), QTCFLLD(3.110861e-02), QTCFLLD(8.197387e-01), QTCFLLD(2.443085e-07), QTCFLLD(5.272982e-05), QTCFLLD(8.089812e-04), QTCFLLD(3.298151e-02), QTCFLLD(8.336864e-01), QTCFLLD(1.387567e-08), QTCFLLD(4.939392e-05), QTCFLLD(1.127142e-03), QTCFLLD(3.493300e-02), QTCFLLD(8.472987e-01), QTCFLLD(-5.690531e-06), QTCFLLD(-4.256442e-05), QTCFLLD(1.417367e-03), QTCFLLD(3.696343e-02), QTCFLLD(8.605543e-01), QTCFLLD(3.629067e-06), QTCFLLD(6.582328e-05), QTCFLLD(1.725030e-03), QTCFLLD(3.907138e-02), QTCFLLD(8.734367e-01), QTCFLLD(-5.393556e-08), QTCFLLD(6.481921e-05), QTCFLLD(1.948069e-03), QTCFLLD(4.125570e-02), QTCFLLD(8.859232e-01), QTCFLLD(1.349944e-07), QTCFLLD(3.367998e-05), QTCFLLD(2.033465e-03), QTCFLLD(4.355568e-02), QTCFLLD(8.979959e-01), QTCFLLD(7.326611e-09), QTCFLLD(4.694252e-05), QTCFLLD(2.239143e-03), QTCFLLD(4.599068e-02), QTCFLLD(9.096311e-01), QTCFLLD(2.399696e-07), QTCFLLD(6.904415e-05), QTCFLLD(2.470456e-03), QTCFLLD(4.849285e-02), QTCFLLD(9.208195e-01), QTCFLLD(3.330982e-07), QTCFLLD(5.643103e-05), QTCFLLD(2.630472e-03), QTCFLLD(5.105621e-02), QTCFLLD(9.315442e-01), QTCFLLD(4.767794e-07), QTCFLLD(7.095887e-05), QTCFLLD(2.703019e-03), QTCFLLD(5.368313e-02), QTCFLLD(9.417976e-01), QTCFLLD(3.428661e-07), QTCFLLD(7.872593e-05), QTCFLLD(2.729137e-03), QTCFLLD(5.637219e-02), QTCFLLD(9.515675e-01), QTCFLLD(8.676848e-06), QTCFLLD(2.666445e-04), QTCFLLD(2.719749e-03), QTCFLLD(5.911363e-02), QTCFLLD(9.608520e-01), QTCFLLD(2.722296e-05), QTCFLLD(5.822201e-04), QTCFLLD(2.530907e-03), QTCFLLD(6.192693e-02), QTCFLLD(9.696426e-01), QTCFLLD(3.575651e-07), QTCFLLD(7.870355e-05), QTCFLLD(2.225524e-03), QTCFLLD(6.480449e-02), QTCFLLD(9.779405e-01), QTCFLLD(6.293002e-07), QTCFLLD(7.245096e-05), QTCFLLD(1.891972e-03), QTCFLLD(6.771675e-02), QTCFLLD(9.857388e-01), QTCFLLD(1.070243e-06), QTCFLLD(7.194151e-05), QTCFLLD(1.557112e-03), QTCFLLD(7.064948e-02), QTCFLLD(9.930380e-01), QTCFLLD(-3.225913e-07), QTCFLLD(-7.679955e-05), QTCFLLD(1.194731e-03), QTCFLLD(7.360559e-02), QTCFLLD(9.998286e-01), QTCFLLD(-9.597516e-09), QTCFLLD(-6.093373e-05), QTCFLLD(6.415402e-04), QTCFLLD(7.657650e-02), QTCFLLD(1.006109e+00), QTCFLLD(-8.908041e-08), QTCFLLD(-1.721347e-05), QTCFLLD(1.092526e-04), QTCFLLD(7.955571e-02), QTCFLLD(1.011868e+00), QTCFLLD(-2.285563e-05), QTCFLLD(-8.882305e-05), QTCFLLD(2.934876e-04), QTCFLLD(8.251962e-02), QTCFLLD(1.017100e+00), QTCFLLD(1.013575e-05), QTCFLLD(6.418658e-05), QTCFLLD(5.721223e-04), QTCFLLD(8.547716e-02), QTCFLLD(1.021799e+00), QTCFLLD(-1.706941e-05), QTCFLLD(1.769262e-04), QTCFLLD(6.976561e-04), QTCFLLD(8.841813e-02), QTCFLLD(1.025967e+00), QTCFLLD(1.356728e-06), QTCFLLD(2.206341e-05), QTCFLLD(7.376101e-04), QTCFLLD(9.133591e-02), QTCFLLD(1.029601e+00), QTCFLLD(-1.398913e-08), QTCFLLD(-6.538879e-06), QTCFLLD(7.154124e-04), QTCFLLD(9.421624e-02), QTCFLLD(1.032713e+00), QTCFLLD(3.552992e-08), QTCFLLD(-1.052707e-05), QTCFLLD(7.139920e-04), QTCFLLD(9.705240e-02), QTCFLLD(1.035312e+00), QTCFLLD(4.211177e-07), QTCFLLD(-9.075431e-06), QTCFLLD(6.944123e-04), QTCFLLD(9.982958e-02), QTCFLLD(1.037422e+00), QTCFLLD(5.433719e-07), QTCFLLD(-1.748285e-05), QTCFLLD(6.766320e-04), QTCFLLD(1.025398e-01), QTCFLLD(1.039062e+00), QTCFLLD(8.226600e-08), QTCFLLD(-3.498286e-05), QTCFLLD(6.887784e-04), QTCFLLD(1.051642e-01), QTCFLLD(1.040262e+00), QTCFLLD(1.272705e-07), QTCFLLD(-4.489491e-05), QTCFLLD(6.673250e-04), QTCFLLD(1.076972e-01), QTCFLLD(1.041043e+00), QTCFLLD(2.542598e-07), QTCFLLD(-5.449816e-05), QTCFLLD(5.970697e-04), QTCFLLD(1.101216e-01), QTCFLLD(1.041434e+00), QTCFLLD(6.322770e-07), QTCFLLD(-5.874199e-05), QTCFLLD(4.749931e-04), QTCFLLD(1.124296e-01), QTCFLLD(1.041443e+00), QTCFLLD(2.801882e-08), QTCFLLD(-7.934510e-05), QTCFLLD(3.189336e-04), QTCFLLD(1.146042e-01), QTCFLLD(1.041087e+00), QTCFLLD(5.891904e-07), QTCFLLD(-8.039232e-05), QTCFLLD(1.218226e-04), QTCFLLD(1.166399e-01), QTCFLLD(1.040350e+00), QTCFLLD(7.301957e-07), QTCFLLD(-9.907631e-05), QTCFLLD(-1.324292e-04), QTCFLLD(1.185243e-01), QTCFLLD(1.039228e+00), QTCFLLD(-4.518603e-06), QTCFLLD(-2.217025e-04), QTCFLLD(-4.268575e-04), QTCFLLD(1.202546e-01), QTCFLLD(1.037683e+00), QTCFLLD(-3.561585e-06), QTCFLLD(-2.415166e-04), QTCFLLD(-7.804546e-04), QTCFLLD(1.218184e-01), QTCFLLD(1.035694e+00), QTCFLLD(-1.074717e-07), QTCFLLD(-2.123672e-04), QTCFLLD(-1.156680e-03), QTCFLLD(1.232132e-01), QTCFLLD(1.033206e+00), QTCFLLD(1.323268e-06), QTCFLLD(-2.078299e-04), QTCFLLD(-1.525819e-03), QTCFLLD(1.244270e-01), QTCFLLD(1.030199e+00), QTCFLLD(3.377815e-06), QTCFLLD(-1.885286e-04), QTCFLLD(-1.914115e-03), QTCFLLD(1.254605e-01), QTCFLLD(1.026616e+00), QTCFLLD(5.161607e-06), QTCFLLD(-1.728673e-04), QTCFLLD(-2.292814e-03), QTCFLLD(1.262996e-01), QTCFLLD(1.022470e+00), QTCFLLD(5.924001e-06), QTCFLLD(-1.744842e-04), QTCFLLD(-2.658042e-03), QTCFLLD(1.269416e-01), QTCFLLD(1.017729e+00), QTCFLLD(6.310208e-06), QTCFLLD(-1.784193e-04), QTCFLLD(-3.000423e-03), QTCFLLD(1.273648e-01), QTCFLLD(1.012508e+00), QTCFLLD(3.357219e-06), QTCFLLD(-2.131406e-04), QTCFLLD(-3.318858e-03), QTCFLLD(1.275561e-01), QTCFLLD(1.006893e+00), QTCFLLD(5.189087e-06), QTCFLLD(-2.078886e-04), QTCFLLD(-3.597476e-03), QTCFLLD(1.274568e-01), QTCFLLD(1.001463e+00), QTCFLLD(4.178050e-06), QTCFLLD(-4.663778e-05), QTCFLLD(-3.870852e-03), QTCFLLD(1.273591e-01), QTCFLLD(9.927544e-01), QTCFLLD(5.364807e-06), QTCFLLD(-5.889277e-06), QTCFLLD(-4.135130e-03), QTCFLLD(1.272499e-01), QTCFLLD(9.807692e-01), QTCFLLD(4.083719e-06), QTCFLLD(-1.774108e-05), QTCFLLD(-4.351668e-03), QTCFLLD(1.268281e-01), QTCFLLD(9.690017e-01), QTCFLLD(3.567581e-06), QTCFLLD(-2.599468e-08), QTCFLLD(-4.517190e-03), QTCFLLD(1.261262e-01), QTCFLLD(9.568886e-01), QTCFLLD(3.262754e-06), QTCFLLD(1.260640e-05), QTCFLLD(-4.636228e-03), QTCFLLD(1.251477e-01), QTCFLLD(9.443803e-01), QTCFLLD(2.041128e-06), QTCFLLD(2.364519e-05), QTCFLLD(-4.704321e-03), QTCFLLD(1.238869e-01), QTCFLLD(9.313874e-01), QTCFLLD(-2.567965e-08), QTCFLLD(2.806963e-05), QTCFLLD(-4.722568e-03), QTCFLLD(1.223371e-01), QTCFLLD(9.179666e-01), QTCFLLD(2.714879e-07), QTCFLLD(4.493916e-05), QTCFLLD(-4.663276e-03), QTCFLLD(1.204854e-01), QTCFLLD(9.041286e-01), QTCFLLD(2.150884e-06), QTCFLLD(5.408155e-05), QTCFLLD(-4.554811e-03), QTCFLLD(1.183233e-01), QTCFLLD(8.899474e-01), QTCFLLD(5.818595e-06), QTCFLLD(3.759630e-05), QTCFLLD(-4.369554e-03), QTCFLLD(1.158359e-01), QTCFLLD(8.754641e-01), QTCFLLD(-1.686137e-09), QTCFLLD(2.515118e-05), QTCFLLD(-4.091033e-03), QTCFLLD(1.130180e-01), QTCFLLD(8.607492e-01), QTCFLLD(-1.775191e-09), QTCFLLD(2.406517e-05), QTCFLLD(-3.794425e-03), QTCFLLD(1.098551e-01), QTCFLLD(8.458450e-01), QTCFLLD(-2.222072e-09), QTCFLLD(3.628511e-05), QTCFLLD(-3.460363e-03), QTCFLLD(1.063455e-01), QTCFLLD(8.308040e-01), QTCFLLD(-1.280675e-08), QTCFLLD(2.241546e-05), QTCFLLD(-3.064311e-03), QTCFLLD(1.024805e-01), QTCFLLD(8.156523e-01), QTCFLLD(-6.977078e-08), QTCFLLD(1.499170e-05), QTCFLLD(-2.621537e-03), QTCFLLD(9.826251e-02), QTCFLLD(8.004165e-01), QTCFLLD(-1.409927e-08), QTCFLLD(5.009913e-06), QTCFLLD(-2.124648e-03), QTCFLLD(9.368652e-02), QTCFLLD(7.851012e-01), QTCFLLD(-2.986489e-09), QTCFLLD(1.277184e-06), QTCFLLD(-1.594861e-03), QTCFLLD(8.875756e-02), QTCFLLD(7.697093e-01), QTCFLLD(1.876022e-07), QTCFLLD(1.580189e-05), QTCFLLD(-1.061499e-03), QTCFLLD(8.347151e-02), QTCFLLD(7.542294e-01), QTCFLLD(1.737277e-07), QTCFLLD(5.533953e-05), QTCFLLD(-6.169855e-04), QTCFLLD(7.783300e-02), QTCFLLD(7.386515e-01), QTCFLLD(3.818589e-09), QTCFLLD(8.870182e-05), QTCFLLD(-2.004823e-04), QTCFLLD(7.184074e-02), QTCFLLD(7.229599e-01), QTCFLLD(5.143615e-07), QTCFLLD(1.035783e-04), QTCFLLD(2.048499e-04), QTCFLLD(6.550209e-02), QTCFLLD(7.071448e-01), QTCFLLD(2.820292e-08), QTCFLLD(9.990758e-05), QTCFLLD(5.621721e-04), QTCFLLD(5.881297e-02), QTCFLLD(6.911982e-01), QTCFLLD(4.677016e-06), QTCFLLD(1.181078e-04), QTCFLLD(9.373975e-04), QTCFLLD(5.177965e-02), QTCFLLD(6.751199e-01), QTCFLLD(3.361682e-06), QTCFLLD(2.126365e-05), QTCFLLD(1.344657e-03), QTCFLLD(4.439684e-02), QTCFLLD(6.589149e-01), QTCFLLD(-4.880845e-08), QTCFLLD(5.861800e-05), QTCFLLD(1.812176e-03), QTCFLLD(3.666943e-02), QTCFLLD(6.425940e-01), QTCFLLD(2.267731e-07), QTCFLLD(5.021906e-05), QTCFLLD(2.172866e-03), QTCFLLD(2.857528e-02), QTCFLLD(6.261725e-01), QTCFLLD(5.158213e-09), QTCFLLD(4.150075e-05), QTCFLLD(1.985825e-03), QTCFLLD(2.012237e-02), QTCFLLD(6.096690e-01), QTCFLLD(-2.066962e-07), QTCFLLD(3.799972e-05), QTCFLLD(1.697653e-03), QTCFLLD(1.132324e-02), QTCFLLD(5.930982e-01), QTCFLLD(4.883305e-07), QTCFLLD(6.606462e-05), QTCFLLD(1.471167e-03), QTCFLLD(2.184257e-03), QTCFLLD(5.764735e-01), QTCFLLD(8.254430e-07), QTCFLLD(9.755685e-05), QTCFLLD(1.232134e-03), QTCFLLD(-7.298198e-03), QTCFLLD(5.598052e-01), QTCFLLD(9.464783e-07), QTCFLLD(1.831121e-04), QTCFLLD(8.990256e-04), QTCFLLD(-1.711324e-02), QTCFLLD(5.430990e-01), QTCFLLD(-1.232693e-05), QTCFLLD(-5.901618e-07), QTCFLLD(6.150317e-04), QTCFLLD(-2.726484e-02), QTCFLLD(5.263554e-01), QTCFLLD(3.867483e-05), QTCFLLD(-3.595054e-04), QTCFLLD(6.307841e-04), QTCFLLD(-3.775928e-02), QTCFLLD(5.095721e-01), QTCFLLD(-9.870548e-07), QTCFLLD(-1.815837e-04), QTCFLLD(4.366447e-04), QTCFLLD(-4.859006e-02), QTCFLLD(4.927464e-01), QTCFLLD(-1.089501e-06), QTCFLLD(-9.204876e-05), QTCFLLD(1.498232e-04), QTCFLLD(-5.973742e-02), QTCFLLD(4.758754e-01), QTCFLLD(-1.569003e-06), QTCFLLD(-5.192444e-05), QTCFLLD(-9.099723e-05), QTCFLLD(-7.120357e-02), QTCFLLD(4.589583e-01), QTCFLLD(-2.778618e-07), QTCFLLD(6.487880e-05), QTCFLLD(-3.337967e-04), QTCFLLD(-8.298103e-02), QTCFLLD(4.420014e-01), QTCFLLD(6.757015e-09), QTCFLLD(5.397065e-05), QTCFLLD(-5.599348e-04), QTCFLLD(-9.506967e-02), QTCFLLD(4.250144e-01), QTCFLLD(1.496436e-07), QTCFLLD(2.472024e-05), QTCFLLD(-7.677634e-04), QTCFLLD(-1.074631e-01), QTCFLLD(4.080155e-01), QTCFLLD(2.068297e-05), QTCFLLD(9.711682e-05), QTCFLLD(-9.730460e-04), QTCFLLD(-1.201629e-01), QTCFLLD(3.910244e-01), QTCFLLD(-9.388963e-06), QTCFLLD(5.144969e-05), QTCFLLD(-1.131860e-03), QTCFLLD(-1.331545e-01), QTCFLLD(3.740644e-01), QTCFLLD(-1.402925e-05), QTCFLLD(-1.039264e-04), QTCFLLD(-1.283281e-03), QTCFLLD(-1.464389e-01), QTCFLLD(3.571528e-01), QTCFLLD(-2.757611e-06), QTCFLLD(2.853437e-06), QTCFLLD(-1.480543e-03), QTCFLLD(-1.600062e-01), QTCFLLD(3.403074e-01), QTCFLLD(2.945239e-08), QTCFLLD(1.334091e-05), QTCFLLD(-1.699161e-03), QTCFLLD(-1.738542e-01), QTCFLLD(3.235299e-01), QTCFLLD(-7.873304e-08), QTCFLLD(2.443161e-05), QTCFLLD(-1.924845e-03), QTCFLLD(-1.879712e-01), QTCFLLD(3.068187e-01), QTCFLLD(-9.897194e-07), QTCFLLD(3.568555e-05), QTCFLLD(-2.152380e-03), QTCFLLD(-2.023548e-01), QTCFLLD(2.901491e-01), QTCFLLD(-1.922074e-06), QTCFLLD(6.193370e-05), QTCFLLD(-2.396404e-03), QTCFLLD(-2.169926e-01), QTCFLLD(2.734977e-01), QTCFLLD(-2.765650e-07), QTCFLLD(1.176237e-04), QTCFLLD(-2.653819e-03), QTCFLLD(-2.318815e-01), QTCFLLD(2.568176e-01), QTCFLLD(-4.636105e-07), QTCFLLD(1.635906e-04), QTCFLLD(-2.927159e-03), QTCFLLD(-2.470098e-01), QTCFLLD(2.400768e-01), QTCFLLD(-9.607069e-07), QTCFLLD(2.060394e-04), QTCFLLD(-3.209093e-03), QTCFLLD(-2.623749e-01), QTCFLLD(2.232277e-01), QTCFLLD(-1.907927e-06), QTCFLLD(2.346981e-04), QTCFLLD(-3.505531e-03), QTCFLLD(-2.779638e-01), QTCFLLD(2.062605e-01), QTCFLLD(-1.551251e-08), QTCFLLD(2.520607e-04), QTCFLLD(-3.811612e-03), QTCFLLD(-2.937725e-01), QTCFLLD(1.891590e-01), QTCFLLD(-1.653464e-06), QTCFLLD(2.556450e-04), QTCFLLD(-4.133640e-03), QTCFLLD(-3.097862e-01), QTCFLLD(1.719726e-01), QTCFLLD(-2.043464e-06), QTCFLLD(3.157664e-04), QTCFLLD(-4.448993e-03), QTCFLLD(-3.259994e-01), QTCFLLD(1.547461e-01), QTCFLLD(1.622786e-05), QTCFLLD(6.205676e-04), QTCFLLD(-4.754192e-03), QTCFLLD(-3.423942e-01), QTCFLLD(1.376150e-01), QTCFLLD(1.395221e-05), QTCFLLD(7.847840e-04), QTCFLLD(-5.063851e-03), QTCFLLD(-3.589627e-01), QTCFLLD(1.206924e-01), QTCFLLD(4.591010e-07), QTCFLLD(9.019129e-04), QTCFLLD(-5.394570e-03), QTCFLLD(-3.756822e-01), QTCFLLD(1.042033e-01), QTCFLLD(-6.261944e-06), QTCFLLD(1.054963e-03), QTCFLLD(-5.741103e-03), QTCFLLD(-3.925409e-01), QTCFLLD(8.829745e-02), QTCFLLD(-1.606051e-05), QTCFLLD(1.089429e-03), QTCFLLD(-6.109179e-03), QTCFLLD(-4.095160e-01), QTCFLLD(7.325979e-02), QTCFLLD(-2.464228e-05), QTCFLLD(1.122503e-03), QTCFLLD(-6.500503e-03), QTCFLLD(-4.265950e-01), QTCFLLD(5.918678e-02), QTCFLLD(-2.976824e-05), QTCFLLD(1.177515e-03), QTCFLLD(-6.925141e-03), QTCFLLD(-4.437530e-01), QTCFLLD(4.634696e-02), QTCFLLD(-3.177468e-05), QTCFLLD(1.226113e-03), QTCFLLD(-7.380544e-03), QTCFLLD(-4.609829e-01), QTCFLLD(3.450719e-02), QTCFLLD(-4.373302e-05), QTCFLLD(1.263569e-03), QTCFLLD(-7.876393e-03), QTCFLLD(-4.782650e-01), QTCFLLD(2.353060e-02), QTCFLLD(-3.299004e-05), QTCFLLD(1.287819e-03), QTCFLLD(-8.407749e-03), QTCFLLD(-4.956175e-01), QTCFLLD(1.129580e-02), }; RAM_ALIGN const FIXP_PFT qmf_cldfb_320[QMF320_CLDFB_PFT_TABLE_SIZE] = { QTCFLLD(5.345060e-07), QTCFLLD(-1.135471e-05), QTCFLLD(-1.222743e-03), QTCFLLD(9.318335e-03), QTCFLLD(5.214033e-01), QTCFLLD(7.604792e-07), QTCFLLD(6.003839e-06), QTCFLLD(-1.125398e-03), QTCFLLD(1.070043e-02), QTCFLLD(5.556087e-01), QTCFLLD(5.563536e-07), QTCFLLD(1.050792e-05), QTCFLLD(-1.016929e-03), QTCFLLD(1.228479e-02), QTCFLLD(5.896935e-01), QTCFLLD(2.906322e-08), QTCFLLD(8.560527e-06), QTCFLLD(-9.531187e-04), QTCFLLD(1.408640e-02), QTCFLLD(6.235157e-01), QTCFLLD(1.084614e-06), QTCFLLD(2.407951e-05), QTCFLLD(-1.093433e-03), QTCFLLD(1.611474e-02), QTCFLLD(6.569371e-01), QTCFLLD(-6.175387e-10), QTCFLLD(8.769899e-06), QTCFLLD(-1.244752e-03), QTCFLLD(1.837877e-02), QTCFLLD(6.898003e-01), QTCFLLD(-4.128808e-09), QTCFLLD(9.494767e-06), QTCFLLD(-8.878160e-04), QTCFLLD(2.089915e-02), QTCFLLD(7.219416e-01), QTCFLLD(-1.116801e-08), QTCFLLD(2.670130e-06), QTCFLLD(-6.273041e-04), QTCFLLD(2.369952e-02), QTCFLLD(7.532082e-01), QTCFLLD(2.607347e-08), QTCFLLD(2.424790e-06), QTCFLLD(-3.865944e-04), QTCFLLD(2.679024e-02), QTCFLLD(7.834691e-01), QTCFLLD(3.782148e-08), QTCFLLD(3.233573e-05), QTCFLLD(2.748136e-04), QTCFLLD(3.021193e-02), QTCFLLD(8.126044e-01), QTCFLLD(1.290921e-07), QTCFLLD(5.106187e-05), QTCFLLD(9.680615e-04), QTCFLLD(3.395726e-02), QTCFLLD(8.404925e-01), QTCFLLD(-1.030732e-06), QTCFLLD(1.162943e-05), QTCFLLD(1.571198e-03), QTCFLLD(3.801740e-02), QTCFLLD(8.669955e-01), QTCFLLD(4.052940e-08), QTCFLLD(4.924960e-05), QTCFLLD(1.990767e-03), QTCFLLD(4.240569e-02), QTCFLLD(8.919595e-01), QTCFLLD(1.236481e-07), QTCFLLD(5.799333e-05), QTCFLLD(2.354800e-03), QTCFLLD(4.724177e-02), QTCFLLD(9.152253e-01), QTCFLLD(4.049388e-07), QTCFLLD(6.369496e-05), QTCFLLD(2.666746e-03), QTCFLLD(5.236967e-02), QTCFLLD(9.366709e-01), QTCFLLD(4.509857e-06), QTCFLLD(1.726852e-04), QTCFLLD(2.724443e-03), QTCFLLD(5.774291e-02), QTCFLLD(9.562097e-01), QTCFLLD(1.379026e-05), QTCFLLD(3.304619e-04), QTCFLLD(2.378216e-03), QTCFLLD(6.336571e-02), QTCFLLD(9.737916e-01), QTCFLLD(8.497715e-07), QTCFLLD(7.219624e-05), QTCFLLD(1.724542e-03), QTCFLLD(6.918311e-02), QTCFLLD(9.893883e-01), QTCFLLD(-1.660944e-07), QTCFLLD(-6.886664e-05), QTCFLLD(9.181354e-04), QTCFLLD(7.509105e-02), QTCFLLD(1.002969e+00), QTCFLLD(-1.147235e-05), QTCFLLD(-5.301826e-05), QTCFLLD(2.013701e-04), QTCFLLD(8.103766e-02), QTCFLLD(1.014484e+00), QTCFLLD(-3.466829e-06), QTCFLLD(1.205564e-04), QTCFLLD(6.348892e-04), QTCFLLD(8.694765e-02), QTCFLLD(1.023883e+00), QTCFLLD(6.713692e-07), QTCFLLD(7.762268e-06), QTCFLLD(7.265112e-04), QTCFLLD(9.277608e-02), QTCFLLD(1.031157e+00), QTCFLLD(2.283238e-07), QTCFLLD(-9.801253e-06), QTCFLLD(7.042022e-04), QTCFLLD(9.844099e-02), QTCFLLD(1.036367e+00), QTCFLLD(3.128189e-07), QTCFLLD(-2.623285e-05), QTCFLLD(6.827052e-04), QTCFLLD(1.038520e-01), QTCFLLD(1.039662e+00), QTCFLLD(1.907652e-07), QTCFLLD(-4.969654e-05), QTCFLLD(6.321974e-04), QTCFLLD(1.089094e-01), QTCFLLD(1.041239e+00), QTCFLLD(3.301479e-07), QTCFLLD(-6.904354e-05), QTCFLLD(3.969634e-04), QTCFLLD(1.135169e-01), QTCFLLD(1.041265e+00), QTCFLLD(6.596931e-07), QTCFLLD(-8.973431e-05), QTCFLLD(-5.303260e-06), QTCFLLD(1.175821e-01), QTCFLLD(1.039789e+00), QTCFLLD(-4.040094e-06), QTCFLLD(-2.316096e-04), QTCFLLD(-6.036561e-04), QTCFLLD(1.210365e-01), QTCFLLD(1.036689e+00), QTCFLLD(6.078980e-07), QTCFLLD(-2.100985e-04), QTCFLLD(-1.341249e-03), QTCFLLD(1.238201e-01), QTCFLLD(1.031702e+00), QTCFLLD(4.269711e-06), QTCFLLD(-1.806979e-04), QTCFLLD(-2.103464e-03), QTCFLLD(1.258800e-01), QTCFLLD(1.024543e+00), QTCFLLD(6.117105e-06), QTCFLLD(-1.764517e-04), QTCFLLD(-2.829232e-03), QTCFLLD(1.271532e-01), QTCFLLD(1.015119e+00), QTCFLLD(4.273153e-06), QTCFLLD(-2.105146e-04), QTCFLLD(-3.458167e-03), QTCFLLD(1.275064e-01), QTCFLLD(1.004178e+00), QTCFLLD(4.771428e-06), QTCFLLD(-2.626353e-05), QTCFLLD(-4.002991e-03), QTCFLLD(1.273045e-01), QTCFLLD(9.867618e-01), QTCFLLD(3.825650e-06), QTCFLLD(-8.883540e-06), QTCFLLD(-4.434429e-03), QTCFLLD(1.264771e-01), QTCFLLD(9.629451e-01), QTCFLLD(2.651941e-06), QTCFLLD(1.812579e-05), QTCFLLD(-4.670274e-03), QTCFLLD(1.245173e-01), QTCFLLD(9.378839e-01), QTCFLLD(1.229041e-07), QTCFLLD(3.650440e-05), QTCFLLD(-4.692922e-03), QTCFLLD(1.214113e-01), QTCFLLD(9.110476e-01), QTCFLLD(3.984739e-06), QTCFLLD(4.583892e-05), QTCFLLD(-4.462183e-03), QTCFLLD(1.170796e-01), QTCFLLD(8.827057e-01), QTCFLLD(-1.730664e-09), QTCFLLD(2.460818e-05), QTCFLLD(-3.942729e-03), QTCFLLD(1.114366e-01), QTCFLLD(8.532971e-01), QTCFLLD(-7.514413e-09), QTCFLLD(2.935029e-05), QTCFLLD(-3.262337e-03), QTCFLLD(1.044130e-01), QTCFLLD(8.232281e-01), QTCFLLD(-4.193503e-08), QTCFLLD(1.000081e-05), QTCFLLD(-2.373092e-03), QTCFLLD(9.597452e-02), QTCFLLD(7.927589e-01), QTCFLLD(9.230786e-08), QTCFLLD(8.539538e-06), QTCFLLD(-1.328180e-03), QTCFLLD(8.611453e-02), QTCFLLD(7.619694e-01), QTCFLLD(8.877312e-08), QTCFLLD(7.202067e-05), QTCFLLD(-4.087339e-04), QTCFLLD(7.483687e-02), QTCFLLD(7.308058e-01), QTCFLLD(2.712822e-07), QTCFLLD(1.017429e-04), QTCFLLD(3.835110e-04), QTCFLLD(6.215753e-02), QTCFLLD(6.991715e-01), QTCFLLD(4.019349e-06), QTCFLLD(6.968570e-05), QTCFLLD(1.141027e-03), QTCFLLD(4.808825e-02), QTCFLLD(6.670174e-01), QTCFLLD(8.898233e-08), QTCFLLD(5.441853e-05), QTCFLLD(1.992521e-03), QTCFLLD(3.262236e-02), QTCFLLD(6.343833e-01), QTCFLLD(-1.007690e-07), QTCFLLD(3.975024e-05), QTCFLLD(1.841739e-03), QTCFLLD(1.572281e-02), QTCFLLD(6.013836e-01), QTCFLLD(6.568868e-07), QTCFLLD(8.181074e-05), QTCFLLD(1.351651e-03), QTCFLLD(-2.556970e-03), QTCFLLD(5.681393e-01), QTCFLLD(-5.690228e-06), QTCFLLD(9.126098e-05), QTCFLLD(7.570286e-04), QTCFLLD(-2.218904e-02), QTCFLLD(5.347272e-01), QTCFLLD(1.884389e-05), QTCFLLD(-2.705446e-04), QTCFLLD(5.337144e-04), QTCFLLD(-4.317467e-02), QTCFLLD(5.011593e-01), QTCFLLD(-1.329252e-06), QTCFLLD(-7.198660e-05), QTCFLLD(2.941296e-05), QTCFLLD(-6.547049e-02), QTCFLLD(4.674168e-01), QTCFLLD(-1.355524e-07), QTCFLLD(5.942472e-05), QTCFLLD(-4.468657e-04), QTCFLLD(-8.902535e-02), QTCFLLD(4.335079e-01), QTCFLLD(1.041631e-05), QTCFLLD(6.091853e-05), QTCFLLD(-8.704047e-04), QTCFLLD(-1.138130e-01), QTCFLLD(3.995200e-01), QTCFLLD(-1.170911e-05), QTCFLLD(-2.623833e-05), QTCFLLD(-1.207570e-03), QTCFLLD(-1.397967e-01), QTCFLLD(3.656086e-01), QTCFLLD(-1.364079e-06), QTCFLLD(8.097173e-06), QTCFLLD(-1.589852e-03), QTCFLLD(-1.669302e-01), QTCFLLD(3.319187e-01), QTCFLLD(-5.342262e-07), QTCFLLD(3.005858e-05), QTCFLLD(-2.038612e-03), QTCFLLD(-1.951630e-01), QTCFLLD(2.984839e-01), QTCFLLD(-1.099320e-06), QTCFLLD(8.977871e-05), QTCFLLD(-2.525111e-03), QTCFLLD(-2.244371e-01), QTCFLLD(2.651577e-01), QTCFLLD(-7.121587e-07), QTCFLLD(1.848150e-04), QTCFLLD(-3.068126e-03), QTCFLLD(-2.546924e-01), QTCFLLD(2.316523e-01), QTCFLLD(-9.617199e-07), QTCFLLD(2.433794e-04), QTCFLLD(-3.658572e-03), QTCFLLD(-2.858681e-01), QTCFLLD(1.977098e-01), QTCFLLD(-1.848464e-06), QTCFLLD(2.857057e-04), QTCFLLD(-4.291316e-03), QTCFLLD(-3.178928e-01), QTCFLLD(1.633594e-01), QTCFLLD(1.509004e-05), QTCFLLD(7.026758e-04), QTCFLLD(-4.909021e-03), QTCFLLD(-3.506784e-01), QTCFLLD(1.291537e-01), QTCFLLD(-2.901422e-06), QTCFLLD(9.784381e-04), QTCFLLD(-5.567837e-03), QTCFLLD(-3.841116e-01), QTCFLLD(9.625038e-02), QTCFLLD(-2.035140e-05), QTCFLLD(1.105966e-03), QTCFLLD(-6.304841e-03), QTCFLLD(-4.180555e-01), QTCFLLD(6.622328e-02), QTCFLLD(-3.077146e-05), QTCFLLD(1.201814e-03), QTCFLLD(-7.152842e-03), QTCFLLD(-4.523680e-01), QTCFLLD(4.042707e-02), QTCFLLD(-3.836153e-05), QTCFLLD(1.275694e-03), QTCFLLD(-8.142071e-03), QTCFLLD(-4.869413e-01), QTCFLLD(1.741320e-02), }; RAM_ALIGN const FIXP_PFT qmf_cldfb_160[QMF160_CLDFB_PFT_TABLE_SIZE] = { QTCFLLD(6.114156e-07), QTCFLLD(-4.929378e-06), QTCFLLD(-1.173270e-03), QTCFLLD(9.985781e-03), QTCFLLD(5.385081e-01), QTCFLLD(2.119298e-07), QTCFLLD(8.001152e-06), QTCFLLD(-9.578346e-04), QTCFLLD(1.315910e-02), QTCFLLD(6.066454e-01), QTCFLLD(8.097845e-07), QTCFLLD(1.849027e-05), QTCFLLD(-1.219567e-03), QTCFLLD(1.721718e-02), QTCFLLD(6.734486e-01), QTCFLLD(-1.135478e-08), QTCFLLD(5.632976e-06), QTCFLLD(-7.392278e-04), QTCFLLD(2.226388e-02), QTCFLLD(7.376929e-01), QTCFLLD(6.347751e-08), QTCFLLD(1.454425e-05), QTCFLLD(-1.105239e-04), QTCFLLD(2.845808e-02), QTCFLLD(7.981848e-01), QTCFLLD(-2.838328e-06), QTCFLLD(3.414749e-06), QTCFLLD(1.272254e-03), QTCFLLD(3.594821e-02), QTCFLLD(8.539265e-01), QTCFLLD(7.116049e-08), QTCFLLD(4.031125e-05), QTCFLLD(2.136304e-03), QTCFLLD(4.477318e-02), QTCFLLD(9.038135e-01), QTCFLLD(4.098227e-07), QTCFLLD(7.484240e-05), QTCFLLD(2.716078e-03), QTCFLLD(5.502766e-02), QTCFLLD(9.466825e-01), QTCFLLD(4.934327e-07), QTCFLLD(7.557725e-05), QTCFLLD(2.058748e-03), QTCFLLD(6.626062e-02), QTCFLLD(9.818396e-01), QTCFLLD(-4.933896e-08), QTCFLLD(-3.907360e-05), QTCFLLD(3.753964e-04), QTCFLLD(7.806610e-02), QTCFLLD(1.008988e+00), QTCFLLD(-7.856341e-06), QTCFLLD(9.949480e-05), QTCFLLD(7.176331e-04), QTCFLLD(8.987702e-02), QTCFLLD(1.027784e+00), QTCFLLD(4.822448e-07), QTCFLLD(-1.327914e-05), QTCFLLD(6.855222e-04), QTCFLLD(1.011847e-01), QTCFLLD(1.038242e+00), QTCFLLD(4.432684e-07), QTCFLLD(-5.662008e-05), QTCFLLD(5.360314e-04), QTCFLLD(1.112756e-01), QTCFLLD(1.041439e+00), QTCFLLD(-1.894204e-06), QTCFLLD(-1.603894e-04), QTCFLLD(-2.796433e-04), QTCFLLD(1.193894e-01), QTCFLLD(1.038456e+00), QTCFLLD(2.350541e-06), QTCFLLD(-1.981793e-04), QTCFLLD(-1.719967e-03), QTCFLLD(1.249437e-01), QTCFLLD(1.028407e+00), QTCFLLD(4.833713e-06), QTCFLLD(-1.957799e-04), QTCFLLD(-3.159640e-03), QTCFLLD(1.274605e-01), QTCFLLD(1.009701e+00), QTCFLLD(4.724263e-06), QTCFLLD(-1.181518e-05), QTCFLLD(-4.243399e-03), QTCFLLD(1.270390e-01), QTCFLLD(9.748854e-01), QTCFLLD(1.007724e-06), QTCFLLD(2.585741e-05), QTCFLLD(-4.713445e-03), QTCFLLD(1.231120e-01), QTCFLLD(9.246770e-01), QTCFLLD(2.908454e-06), QTCFLLD(3.137374e-05), QTCFLLD(-4.230293e-03), QTCFLLD(1.144269e-01), QTCFLLD(8.681067e-01), QTCFLLD(-4.128877e-08), QTCFLLD(1.870358e-05), QTCFLLD(-2.842924e-03), QTCFLLD(1.003715e-01), QTCFLLD(8.080344e-01), QTCFLLD(1.806649e-07), QTCFLLD(3.557071e-05), QTCFLLD(-8.392422e-04), QTCFLLD(8.065225e-02), QTCFLLD(7.464405e-01), QTCFLLD(2.352609e-06), QTCFLLD(1.090077e-04), QTCFLLD(7.497848e-04), QTCFLLD(5.529631e-02), QTCFLLD(6.831591e-01), QTCFLLD(1.159657e-07), QTCFLLD(4.585990e-05), QTCFLLD(2.079346e-03), QTCFLLD(2.434883e-02), QTCFLLD(6.179208e-01), QTCFLLD(8.859606e-07), QTCFLLD(1.403345e-04), QTCFLLD(1.065580e-03), QTCFLLD(-1.220572e-02), QTCFLLD(5.514521e-01), QTCFLLD(-1.038278e-06), QTCFLLD(-1.368162e-04), QTCFLLD(2.932339e-04), QTCFLLD(-5.416374e-02), QTCFLLD(4.843109e-01), QTCFLLD(7.820030e-08), QTCFLLD(3.934544e-05), QTCFLLD(-6.638491e-04), QTCFLLD(-1.012664e-01), QTCFLLD(4.165150e-01), QTCFLLD(-8.393432e-06), QTCFLLD(-5.053646e-05), QTCFLLD(-1.381912e-03), QTCFLLD(-1.532225e-01), QTCFLLD(3.487301e-01), QTCFLLD(-1.455897e-06), QTCFLLD(4.880962e-05), QTCFLLD(-2.274392e-03), QTCFLLD(-2.096737e-01), QTCFLLD(2.818234e-01), QTCFLLD(-1.434317e-06), QTCFLLD(2.203687e-04), QTCFLLD(-3.357312e-03), QTCFLLD(-2.701693e-01), QTCFLLD(2.147441e-01), QTCFLLD(7.092199e-06), QTCFLLD(4.681670e-04), QTCFLLD(-4.601593e-03), QTCFLLD(-3.341968e-01), QTCFLLD(1.461805e-01), QTCFLLD(-1.116123e-05), QTCFLLD(1.072196e-03), QTCFLLD(-5.925141e-03), QTCFLLD(-4.010285e-01), QTCFLLD(8.077862e-02), QTCFLLD(-3.775385e-05), QTCFLLD(1.244841e-03), QTCFLLD(-7.628469e-03), QTCFLLD(-4.696240e-01), QTCFLLD(2.901889e-02), }; RAM_ALIGN const FIXP_PFT qmf_cldfb_80[QMF80_CLDFB_PFT_TABLE_SIZE] = { QTCFLLD(6.966921e-07), QTCFLLD(9.025176e-06), QTCFLLD(-1.073328e-03), QTCFLLD(1.146585e-02), QTCFLLD(5.726758e-01), QTCFLLD(-2.323046e-09), QTCFLLD(1.012638e-05), QTCFLLD(-1.084654e-03), QTCFLLD(1.960515e-02), QTCFLLD(7.059712e-01), QTCFLLD(1.230159e-07), QTCFLLD(4.639126e-05), QTCFLLD(6.398911e-04), QTCFLLD(3.204506e-02), QTCFLLD(8.267125e-01), QTCFLLD(2.865339e-07), QTCFLLD(6.273759e-05), QTCFLLD(2.550464e-03), QTCFLLD(4.977453e-02), QTCFLLD(9.261818e-01), QTCFLLD(3.738257e-07), QTCFLLD(-2.429021e-06), QTCFLLD(1.375921e-03), QTCFLLD(7.212754e-02), QTCFLLD(9.964333e-01), QTCFLLD(1.077039e-08), QTCFLLD(-8.532976e-06), QTCFLLD(7.147022e-04), QTCFLLD(9.563432e-02), QTCFLLD(1.034012e+00), QTCFLLD(3.086046e-07), QTCFLLD(-7.986870e-05), QTCFLLD(2.203781e-04), QTCFLLD(1.156221e-01), QTCFLLD(1.040718e+00), QTCFLLD(5.542804e-06), QTCFLLD(-1.736757e-04), QTCFLLD(-2.475428e-03), QTCFLLD(1.266206e-01), QTCFLLD(1.020100e+00), QTCFLLD(3.415168e-06), QTCFLLD(6.290201e-06), QTCFLLD(-4.576709e-03), QTCFLLD(1.256370e-01), QTCFLLD(9.506344e-01), QTCFLLD(-1.998632e-09), QTCFLLD(3.017514e-05), QTCFLLD(-3.627394e-03), QTCFLLD(1.081003e-01), QTCFLLD(8.383245e-01), QTCFLLD(2.590900e-07), QTCFLLD(9.614004e-05), QTCFLLD(2.183786e-06), QTCFLLD(6.867141e-02), QTCFLLD(7.150523e-01), QTCFLLD(1.408172e-07), QTCFLLD(5.203217e-05), QTCFLLD(1.584410e-03), QTCFLLD(6.753749e-03), QTCFLLD(5.847858e-01), QTCFLLD(-9.234326e-07), QTCFLLD(6.477183e-06), QTCFLLD(-2.123969e-04), QTCFLLD(-7.709230e-02), QTCFLLD(4.504798e-01), QTCFLLD(-2.464033e-08), QTCFLLD(1.888626e-05), QTCFLLD(-1.812003e-03), QTCFLLD(-1.809127e-01), QTCFLLD(3.151743e-01), QTCFLLD(-8.344882e-07), QTCFLLD(2.538528e-04), QTCFLLD(-3.972626e-03), QTCFLLD(-3.017793e-01), QTCFLLD(1.805658e-01), QTCFLLD(-2.720526e-05), QTCFLLD(1.150009e-03), QTCFLLD(-6.712822e-03), QTCFLLD(-4.351740e-01), QTCFLLD(5.276687e-02), }; #if defined(QMF_COEFF_16BIT) #define QTMFLLD(x) FL2FXCONST_SGL(x / (float)(1 << QMF_MPSLDFB_PFT_SCALE)) #define QTMFLLDT(x) FX_DBL2FXCONST_SGL(x) #else #define QTMFLLD(x) FL2FXCONST_DBL(x / (float)(1 << QMF_MPSLDFB_PFT_SCALE)) #define QTMFLLDT(x) (FIXP_DBL)(x) #endif /*! \name QMF \brief QMF-Table 32 channels, N = 320, The coefficients are derived from the MPS Low Delay coefficient set with 640 samples. The coefficients are interpolated and rearranged in the following way compared to the reference: qmf_mpsldfb_320[0] = (qmf_64_reference[ 0] + qmf_64_reference[ 1])/2.0; qmf_mpsldfb_320[1] = (qmf_64_reference[128] + qmf_64_reference[129])/2.0; qmf_mpsldfb_320[2] = (qmf_64_reference[256] + qmf_64_reference[257])/2.0; qmf_mpsldfb_320[3] = (qmf_64_reference[384] + qmf_64_reference[385])/2.0; qmf_mpsldfb_320[4] = (qmf_64_reference[512] + qmf_64_reference[513])/2.0; qmf_mpsldfb_320[5] = (qmf_64_reference[ 2] + qmf_64_reference[ 3])/2.0; qmf_mpsldfb_320[6] = (qmf_64_reference[130] + qmf_64_reference[131])/2.0; qmf_mpsldfb_320[7] = (qmf_64_reference[258] + qmf_64_reference[259])/2.0; qmf_mpsldfb_320[8] = (qmf_64_reference[386] + qmf_64_reference[387])/2.0; qmf_mpsldfb_320[9] = (qmf_64_reference[514] + qmf_64_reference[515])/2.0; . . . qmf_mpsldfb_320[315] = (qmf_64_reference[126] + qmf_64_reference[127])/2.0; qmf_mpsldfb_320[316] = (qmf_64_reference[254] + qmf_64_reference[255])/2.0; qmf_mpsldfb_320[317] = (qmf_64_reference[382] + qmf_64_reference[383])/2.0; qmf_mpsldfb_320[318] = (qmf_64_reference[510] + qmf_64_reference[511])/2.0; qmf_mpsldfb_320[319] = (qmf_64_reference[638] + qmf_64_reference[639])/2.0; The filter output is required to be scaled by 1 bit. \showinitializer */ //@{ const FIXP_PFT qmf_mpsldfb_320[QMF320_MPSLDFB_PFT_TABLE_SIZE] = { QTMFLLD(1.0777725402e-004), QTMFLLD(-9.4703806099e-004), QTMFLLD(6.1286436394e-003), QTMFLLD(-9.0161964297e-002), QTMFLLD(5.5554401875e-001), QTMFLLD(1.2731316383e-004), QTMFLLD(-1.2311334722e-003), QTMFLLD(4.9468209036e-003), QTMFLLD(-1.1305026710e-001), QTMFLLD(5.2990418673e-001), QTMFLLD(1.1927412561e-004), QTMFLLD(-1.5128203668e-003), QTMFLLD(3.5794533323e-003), QTMFLLD(-1.3681203127e-001), QTMFLLD(5.0423312187e-001), QTMFLLD(1.0006380762e-004), QTMFLLD(-1.7925058492e-003), QTMFLLD(2.0164034795e-003), QTMFLLD(-1.6139641404e-001), QTMFLLD(4.7861024737e-001), QTMFLLD(7.2826202086e-005), QTMFLLD(-2.0697340369e-003), QTMFLLD(2.4838969694e-004), QTMFLLD(-1.8674756587e-001), QTMFLLD(4.5311337709e-001), QTMFLLD(3.8808015233e-005), QTMFLLD(-2.3429044522e-003), QTMFLLD(-1.7331546405e-003), QTMFLLD(-2.1280488372e-001), QTMFLLD(4.2781800032e-001), QTMFLLD(-5.4359588830e-007), QTMFLLD(-2.6112669148e-003), QTMFLLD(-3.9357249625e-003), QTMFLLD(-2.3950359225e-001), QTMFLLD(4.0279802680e-001), QTMFLLD(-4.3614549213e-005), QTMFLLD(-2.8741455171e-003), QTMFLLD(-6.3655078411e-003), QTMFLLD(-2.6677471399e-001), QTMFLLD(3.7812507153e-001), QTMFLLD(-8.9040157036e-005), QTMFLLD(-3.1308881007e-003), QTMFLLD(-9.0275555849e-003), QTMFLLD(-2.9454550147e-001), QTMFLLD(3.5386830568e-001), QTMFLLD(-1.3519046479e-004), QTMFLLD(-3.3808732405e-003), QTMFLLD(-1.1925406754e-002), QTMFLLD(-3.2273942232e-001), QTMFLLD(3.3009397984e-001), QTMFLLD(-1.8045579782e-004), QTMFLLD(-3.6236830056e-003), QTMFLLD(-1.5061311424e-002), QTMFLLD(-3.5127705336e-001), QTMFLLD(3.0686509609e-001), QTMFLLD(-2.2396800341e-004), QTMFLLD(-3.8587960880e-003), QTMFLLD(-1.8435835838e-002), QTMFLLD(-3.8007527590e-001), QTMFLLD(2.8424069285e-001), QTMFLLD(-2.6416976471e-004), QTMFLLD(-4.0859002620e-003), QTMFLLD(-2.2048022598e-002), QTMFLLD(-4.0904915333e-001), QTMFLLD(2.6227575541e-001), QTMFLLD(-3.0001887353e-004), QTMFLLD(-4.3045589700e-003), QTMFLLD(-2.5894984603e-002), QTMFLLD(-4.3811064959e-001), QTMFLLD(2.4102044106e-001), QTMFLLD(-3.3083156450e-004), QTMFLLD(-4.5145484619e-003), QTMFLLD(-2.9972121119e-002), QTMFLLD(-4.6717000008e-001), QTMFLLD(2.2052007914e-001), QTMFLLD(-3.5614447552e-004), QTMFLLD(-4.7155953944e-003), QTMFLLD(-3.4272894263e-002), QTMFLLD(-4.9613577127e-001), QTMFLLD(2.0081442595e-001), QTMFLLD(-3.7579826312e-004), QTMFLLD(-4.9072988331e-003), QTMFLLD(-3.8788780570e-002), QTMFLLD(-5.2491527796e-001), QTMFLLD(1.8193808198e-001), QTMFLLD(-3.8993739872e-004), QTMFLLD(-5.0893351436e-003), QTMFLLD(-4.3509010226e-002), QTMFLLD(-5.5341482162e-001), QTMFLLD(1.6391974688e-001), QTMFLLD(-3.9912899956e-004), QTMFLLD(-5.2615385503e-003), QTMFLLD(-4.8421185464e-002), QTMFLLD(-5.8154034615e-001), QTMFLLD(1.4678207040e-001), QTMFLLD(-4.0421969607e-004), QTMFLLD(-5.4236799479e-003), QTMFLLD(-5.3510606289e-002), QTMFLLD(-6.0919785500e-001), QTMFLLD(1.3054165244e-001), QTMFLLD(-4.0645478293e-004), QTMFLLD(-5.5756671354e-003), QTMFLLD(-5.8760054410e-002), QTMFLLD(-6.3629388809e-001), QTMFLLD(1.1520925164e-001), QTMFLLD(-4.0720938705e-004), QTMFLLD(-5.7173836976e-003), QTMFLLD(-6.4149998128e-002), QTMFLLD(-6.6273581982e-001), QTMFLLD(1.0078965127e-001), QTMFLLD(-4.0812738007e-004), QTMFLLD(-5.8488911018e-003), QTMFLLD(-6.9658569992e-002), QTMFLLD(-6.8843221664e-001), QTMFLLD(8.7281554937e-002), QTMFLLD(-4.1120912647e-004), QTMFLLD(-5.9703430161e-003), QTMFLLD(-7.5261354446e-002), QTMFLLD(-7.1329379082e-001), QTMFLLD(7.4678033590e-002), QTMFLLD(-4.1838851757e-004), QTMFLLD(-6.0821287334e-003), QTMFLLD(-8.0931767821e-002), QTMFLLD(-7.3723363876e-001), QTMFLLD(6.2966249883e-002), QTMFLLD(-4.3148122495e-004), QTMFLLD(-6.1847940087e-003), QTMFLLD(-8.6640790105e-002), QTMFLLD(-7.6016783714e-001), QTMFLLD(5.2128262818e-002), QTMFLLD(-4.5229538227e-004), QTMFLLD(-6.2791546807e-003), QTMFLLD(-9.2357128859e-002), QTMFLLD(-7.8201586008e-001), QTMFLLD(4.2139917612e-002), QTMFLLD(-4.8211280955e-004), QTMFLLD(-6.3661932945e-003), QTMFLLD(-9.8047181964e-002), QTMFLLD(-8.0270123482e-001), QTMFLLD(3.2972395420e-002), QTMFLLD(-5.2196672186e-004), QTMFLLD(-6.4471233636e-003), QTMFLLD(-1.0367526114e-001), QTMFLLD(-8.2215231657e-001), QTMFLLD(2.4589803070e-002), QTMFLLD(-5.7247944642e-004), QTMFLLD(-6.5232971683e-003), QTMFLLD(-1.0920339823e-001), QTMFLLD(-8.4030228853e-001), QTMFLLD(1.6952158883e-002), QTMFLLD(-6.3343788497e-004), QTMFLLD(-6.5963375382e-003), QTMFLLD(-1.1459194124e-001), QTMFLLD(-8.5709118843e-001), QTMFLLD(1.0006074794e-002), QTMFLLD(-7.0449430496e-004), QTMFLLD(-6.6681848839e-003), QTMFLLD(-1.1979964375e-001), QTMFLLD(-8.7246519327e-001), QTMFLLD(3.6968050990e-003), QTMFLLD(-7.9609593377e-004), QTMFLLD(-6.7403013818e-003), QTMFLLD(-1.2478165329e-001), QTMFLLD(-8.8632321358e-001), QTMFLLD(-1.6344460892e-003), QTMFLLD(-9.0200459817e-004), QTMFLLD(-6.8151149899e-003), QTMFLLD(-1.2949258089e-001), QTMFLLD(-8.9860773087e-001), QTMFLLD(-5.9283543378e-003), QTMFLLD(-1.0116943158e-003), QTMFLLD(-6.8955891766e-003), QTMFLLD(-1.3388808072e-001), QTMFLLD(-9.0933418274e-001), QTMFLLD(-9.6466485411e-003), QTMFLLD(-1.1244935449e-003), QTMFLLD(-6.9835213944e-003), QTMFLLD(-1.3791990280e-001), QTMFLLD(-9.1846722364e-001), QTMFLLD(-1.2838950381e-002), QTMFLLD(-1.2393904617e-003), QTMFLLD(-7.0809246972e-003), QTMFLLD(-1.4153905213e-001), QTMFLLD(-9.2597639561e-001), QTMFLLD(-1.5539921820e-002), QTMFLLD(-1.3542033266e-003), QTMFLLD(-7.1895248257e-003), QTMFLLD(-1.4469626546e-001), QTMFLLD(-9.3183851242e-001), QTMFLLD(-1.7783239484e-002), QTMFLLD(-1.4669501688e-003), QTMFLLD(-7.3110014200e-003), QTMFLLD(-1.4734169841e-001), QTMFLLD(-9.3603670597e-001), QTMFLLD(-1.9597738981e-002), QTMFLLD(-1.5753224725e-003), QTMFLLD(-7.4466220103e-003), QTMFLLD(-1.4942565560e-001), QTMFLLD(-9.3856132030e-001), QTMFLLD(-2.1011535078e-002), QTMFLLD(-1.6771152150e-003), QTMFLLD(-7.5972955674e-003), QTMFLLD(-1.5089863539e-001), QTMFLLD(-9.3940949440e-001), QTMFLLD(-2.2049814463e-002), QTMFLLD(-1.7698677257e-003), QTMFLLD(-7.7634919435e-003), QTMFLLD(-1.5171185136e-001), QTMFLLD(-9.3858534098e-001), QTMFLLD(-2.2738276049e-002), QTMFLLD(-1.8512960523e-003), QTMFLLD(-7.9450644553e-003), QTMFLLD(-1.5181747079e-001), QTMFLLD(-9.3610012531e-001), QTMFLLD(-2.3101080209e-002), QTMFLLD(-1.9192657201e-003), QTMFLLD(-8.1413704902e-003), QTMFLLD(-1.5116891265e-001), QTMFLLD(-9.3197190762e-001), QTMFLLD(-2.3163486272e-002), QTMFLLD(-1.9716904499e-003), QTMFLLD(-8.3509404212e-003), QTMFLLD(-1.4972095191e-001), QTMFLLD(-9.2622530460e-001), QTMFLLD(-2.2950030863e-002), QTMFLLD(-2.0066620782e-003), QTMFLLD(-8.5715763271e-003), QTMFLLD(-1.4743055403e-001), QTMFLLD(-9.1889131069e-001), QTMFLLD(-2.2486699745e-002), QTMFLLD(-2.0227057394e-003), QTMFLLD(-8.8005559519e-003), QTMFLLD(-1.4425669611e-001), QTMFLLD(-9.1000711918e-001), QTMFLLD(-2.1799135953e-002), QTMFLLD(-2.0185527392e-003), QTMFLLD(-9.0341167524e-003), QTMFLLD(-1.4016106725e-001), QTMFLLD(-8.9961612225e-001), QTMFLLD(-2.0914383233e-002), QTMFLLD(-1.9932338037e-003), QTMFLLD(-9.2674419284e-003), QTMFLLD(-1.3510815799e-001), QTMFLLD(-8.8776648045e-001), QTMFLLD(-1.9859094173e-002), QTMFLLD(-1.9461065531e-003), QTMFLLD(-9.4948727638e-003), QTMFLLD(-1.2906542420e-001), QTMFLLD(-8.7451159954e-001), QTMFLLD(-1.8660902977e-002), QTMFLLD(-1.8770052120e-003), QTMFLLD(-9.7100129351e-003), QTMFLLD(-1.2200380862e-001), QTMFLLD(-8.5991013050e-001), QTMFLLD(-1.7346922308e-002), QTMFLLD(-1.7859865911e-003), QTMFLLD(-9.9056493491e-003), QTMFLLD(-1.1389782280e-001), QTMFLLD(-8.4402561188e-001), QTMFLLD(-1.5944939107e-002), QTMFLLD(-1.6734169330e-003), QTMFLLD(-1.0073989630e-002), QTMFLLD(-1.0472598672e-001), QTMFLLD(-8.2692527771e-001), QTMFLLD(-1.4481747523e-002), QTMFLLD(-1.5399802942e-003), QTMFLLD(-1.0205906816e-002), QTMFLLD(-9.4470888376e-002), QTMFLLD(-8.0868041515e-001), QTMFLLD(-1.2984249741e-002), QTMFLLD(-1.3865872752e-003), QTMFLLD(-1.0291703977e-002), QTMFLLD(-8.3119556308e-002), QTMFLLD(-7.8936588764e-001), QTMFLLD(-1.1477986351e-002), QTMFLLD(-1.2144348584e-003), QTMFLLD(-1.0320962407e-002), QTMFLLD(-7.0663399994e-002), QTMFLLD(-7.6905936003e-001), QTMFLLD(-9.9884867668e-003), QTMFLLD(-1.0248266626e-003), QTMFLLD(-1.0282764211e-002), QTMFLLD(-5.7098604739e-002), QTMFLLD(-7.4784147739e-001), QTMFLLD(-8.5393209010e-003), QTMFLLD(-8.1919803051e-004), QTMFLLD(-1.0165717453e-002), QTMFLLD(-4.2426198721e-002), QTMFLLD(-7.2579479218e-001), QTMFLLD(-7.1533406153e-003), QTMFLLD(-5.9914286248e-004), QTMFLLD(-9.9579729140e-003), QTMFLLD(-2.6652012020e-002), QTMFLLD(-7.0300412178e-001), QTMFLLD(-5.8508114889e-003), QTMFLLD(-3.6626873771e-004), QTMFLLD(-9.6475090832e-003), QTMFLLD(-9.7871217877e-003), QTMFLLD(-6.7955517769e-001), QTMFLLD(-4.6512838453e-003), QTMFLLD(-1.2227181287e-004), QTMFLLD(-9.2221321538e-003), QTMFLLD(8.1523396075e-003), QTMFLLD(-6.5553492308e-001), QTMFLLD(-3.5699680448e-003), QTMFLLD(1.3090072025e-004), QTMFLLD(-8.6695179343e-003), QTMFLLD(2.7145106345e-002), QTMFLLD(-6.3103044033e-001), QTMFLLD(-2.6181070134e-003), QTMFLLD(3.9128778735e-004), QTMFLLD(-7.9773496836e-003), QTMFLLD(4.7164849937e-002), QTMFLLD(-6.0613000393e-001), QTMFLLD(-1.7908872105e-003), QTMFLLD(6.5761915175e-004), QTMFLLD(-7.1337916888e-003), QTMFLLD(6.8181537092e-002), QTMFLLD(-5.8092808723e-001), QTMFLLD(-1.0135001503e-003)}; /*! \name QMF \brief QMF-Table 64 channels, N = 640, The coeffs are rearranged compared with the reference in the following way: qmf_64[0] = qmf_64_reference[0]; qmf_64[1] = qmf_64_reference[128]; qmf_64[2] = qmf_64_reference[256]; qmf_64[3] = qmf_64_reference[384]; qmf_64[4] = qmf_64_reference[512]; qmf_64[5] = qmf_64_reference[1]; qmf_64[6] = qmf_64_reference[129]; qmf_64[7] = qmf_64_reference[257]; qmf_64[8] = qmf_64_reference[385]; qmf_64[9] = qmf_64_reference[513]; . . . qmf_64[635] = qmf_64_reference[127] qmf_64[636] = qmf_64_reference[255]; qmf_64[637] = qmf_64_reference[383]; qmf_64[638] = qmf_64_reference[511]; qmf_64[639] = qmf_64_reference[639]; The filter output is required to be scaled by 1 bit. \showinitializer */ //@{ LNK_SECTION_CONSTDATA_L1 RAM_ALIGN const FIXP_PFT qmf_mpsldfb_640[QMF640_MPSLDFB_PFT_TABLE_SIZE] = { QTMFLLD(9.3863010989e-005), QTMFLLD(-8.7536586216e-004), QTMFLLD(6.4016343094e-003), QTMFLLD(-8.4552817047e-002), QTMFLLD(5.6194400787e-001), QTMFLLD(1.2169149704e-004), QTMFLLD(-1.0187102016e-003), QTMFLLD(5.8556534350e-003), QTMFLLD(-9.5771118999e-002), QTMFLLD(5.4914402962e-001), QTMFLLD(1.2793767382e-004), QTMFLLD(-1.1605311884e-003), QTMFLLD(5.2649765275e-003), QTMFLLD(-1.0721673071e-001), QTMFLLD(5.3632181883e-001), QTMFLLD(1.2668863928e-004), QTMFLLD(-1.3017356396e-003), QTMFLLD(4.6286652796e-003), QTMFLLD(-1.1888379604e-001), QTMFLLD(5.2348655462e-001), QTMFLLD(1.2296593923e-004), QTMFLLD(-1.4426353155e-003), QTMFLLD(3.9453012869e-003), QTMFLLD(-1.3076621294e-001), QTMFLLD(5.1064836979e-001), QTMFLLD(1.1558231199e-004), QTMFLLD(-1.5830053017e-003), QTMFLLD(3.2136053778e-003), QTMFLLD(-1.4285783470e-001), QTMFLLD(4.9781781435e-001), QTMFLLD(1.0582985124e-004), QTMFLLD(-1.7228506040e-003), QTMFLLD(2.4323666003e-003), QTMFLLD(-1.5515175462e-001), QTMFLLD(4.8500382900e-001), QTMFLLD(9.4297764008e-005), QTMFLLD(-1.8621610943e-003), QTMFLLD(1.6004402423e-003), QTMFLLD(-1.6764105856e-001), QTMFLLD(4.7221666574e-001), QTMFLLD(8.0514568253e-005), QTMFLLD(-2.0008818246e-003), QTMFLLD(7.1672687773e-004), QTMFLLD(-1.8031860888e-001), QTMFLLD(4.5946595073e-001), QTMFLLD(6.5137835918e-005), QTMFLLD(-2.1385864820e-003), QTMFLLD(-2.1994746930e-004), QTMFLLD(-1.9317652285e-001), QTMFLLD(4.4676083326e-001), QTMFLLD(4.8101064749e-005), QTMFLLD(-2.2751907818e-003), QTMFLLD(-1.2104592752e-003), QTMFLLD(-2.0620720088e-001), QTMFLLD(4.3411090970e-001), QTMFLLD(2.9514967537e-005), QTMFLLD(-2.4106178898e-003), QTMFLLD(-2.2558500059e-003), QTMFLLD(-2.1940255165e-001), QTMFLLD(4.2152509093e-001), QTMFLLD(9.8814107332e-006), QTMFLLD(-2.5448307861e-003), QTMFLLD(-3.3569468651e-003), QTMFLLD(-2.3275400698e-001), QTMFLLD(4.0901294351e-001), QTMFLLD(-1.0968602510e-005), QTMFLLD(-2.6777030434e-003), QTMFLLD(-4.5145032927e-003), QTMFLLD(-2.4625316262e-001), QTMFLLD(3.9658311009e-001), QTMFLLD(-3.2559255487e-005), QTMFLLD(-2.8091520071e-003), QTMFLLD(-5.7292259298e-003), QTMFLLD(-2.5989097357e-001), QTMFLLD(3.8424444199e-001), QTMFLLD(-5.4669842939e-005), QTMFLLD(-2.9391390271e-003), QTMFLLD(-7.0017897524e-003), QTMFLLD(-2.7365845442e-001), QTMFLLD(3.7200567126e-001), QTMFLLD(-7.7506563684e-005), QTMFLLD(-3.0675258022e-003), QTMFLLD(-8.3327051252e-003), QTMFLLD(-2.8754624724e-001), QTMFLLD(3.5987523198e-001), QTMFLLD(-1.0057374311e-004), QTMFLLD(-3.1942503992e-003), QTMFLLD(-9.7224051133e-003), QTMFLLD(-3.0154475570e-001), QTMFLLD(3.4786140919e-001), QTMFLLD(-1.2368557509e-004), QTMFLLD(-3.3192564733e-003), QTMFLLD(-1.1171258055e-002), QTMFLLD(-3.1564420462e-001), QTMFLLD(3.3597227931e-001), QTMFLLD(-1.4669535449e-004), QTMFLLD(-3.4424900077e-003), QTMFLLD(-1.2679555453e-002), QTMFLLD(-3.2983466983e-001), QTMFLLD(3.2421571016e-001), QTMFLLD(-1.6928518016e-004), QTMFLLD(-3.5639149137e-003), QTMFLLD(-1.4247507788e-002), QTMFLLD(-3.4410607815e-001), QTMFLLD(3.1259948015e-001), QTMFLLD(-1.9162640092e-004), QTMFLLD(-3.6834510975e-003), QTMFLLD(-1.5875114128e-002), QTMFLLD(-3.5844799876e-001), QTMFLLD(3.0113074183e-001), QTMFLLD(-2.1345751884e-004), QTMFLLD(-3.8009947166e-003), QTMFLLD(-1.7562393099e-002), QTMFLLD(-3.7284970284e-001), QTMFLLD(2.8981682658e-001), QTMFLLD(-2.3447850253e-004), QTMFLLD(-3.9165974595e-003), QTMFLLD(-1.9309276715e-002), QTMFLLD(-3.8730087876e-001), QTMFLLD(2.7866455913e-001), QTMFLLD(-2.5462667691e-004), QTMFLLD(-4.0301652625e-003), QTMFLLD(-2.1115457639e-002), QTMFLLD(-4.0179058909e-001), QTMFLLD(2.6768052578e-001), QTMFLLD(-2.7371285250e-004), QTMFLLD(-4.1416347958e-003), QTMFLLD(-2.2980585694e-002), QTMFLLD(-4.1630774736e-001), QTMFLLD(2.5687095523e-001), QTMFLLD(-2.9165804153e-004), QTMFLLD(-4.2509674095e-003), QTMFLLD(-2.4904217571e-002), QTMFLLD(-4.3084129691e-001), QTMFLLD(2.4624188244e-001), QTMFLLD(-3.0837973463e-004), QTMFLLD(-4.3581505306e-003), QTMFLLD(-2.6885753497e-002), QTMFLLD(-4.4538003206e-001), QTMFLLD(2.3579898477e-001), QTMFLLD(-3.2378203468e-004), QTMFLLD(-4.4631510973e-003), QTMFLLD(-2.8924530372e-002), QTMFLLD(-4.5991250873e-001), QTMFLLD(2.2554755211e-001), QTMFLLD(-3.3788106521e-004), QTMFLLD(-4.5659458265e-003), QTMFLLD(-3.1019711867e-002), QTMFLLD(-4.7442746162e-001), QTMFLLD(2.1549259126e-001), QTMFLLD(-3.5053401371e-004), QTMFLLD(-4.6664695255e-003), QTMFLLD(-3.3170353621e-002), QTMFLLD(-4.8891320825e-001), QTMFLLD(2.0563863218e-001), QTMFLLD(-3.6175493733e-004), QTMFLLD(-4.7647207975e-003), QTMFLLD(-3.5375438631e-002), QTMFLLD(-5.0335830450e-001), QTMFLLD(1.9599021971e-001), QTMFLLD(-3.7159718340e-004), QTMFLLD(-4.8605888151e-003), QTMFLLD(-3.7633713335e-002), QTMFLLD(-5.1775097847e-001), QTMFLLD(1.8655113876e-001), QTMFLLD(-3.7999937194e-004), QTMFLLD(-4.9540083855e-003), QTMFLLD(-3.9943847805e-002), QTMFLLD(-5.3207957745e-001), QTMFLLD(1.7732504010e-001), QTMFLLD(-3.8705617771e-004), QTMFLLD(-5.0450465642e-003), QTMFLLD(-4.2304381728e-002), QTMFLLD(-5.4633224010e-001), QTMFLLD(1.6831515729e-001), QTMFLLD(-3.9281861973e-004), QTMFLLD(-5.1336232573e-003), QTMFLLD(-4.4713638723e-002), QTMFLLD(-5.6049734354e-001), QTMFLLD(1.5952435136e-001), QTMFLLD(-3.9737694897e-004), QTMFLLD(-5.2197398618e-003), QTMFLLD(-4.7170232981e-002), QTMFLLD(-5.7456302643e-001), QTMFLLD(1.5095503628e-001), QTMFLLD(-4.0088107926e-004), QTMFLLD(-5.3033372387e-003), QTMFLLD(-4.9672137946e-002), QTMFLLD(-5.8851766586e-001), QTMFLLD(1.4260910451e-001), QTMFLLD(-4.0338383405e-004), QTMFLLD(-5.3843962960e-003), QTMFLLD(-5.2217379212e-002), QTMFLLD(-6.0234934092e-001), QTMFLLD(1.3448855281e-001), QTMFLLD(-4.0505555808e-004), QTMFLLD(-5.4629631341e-003), QTMFLLD(-5.4803829640e-002), QTMFLLD(-6.1604642868e-001), QTMFLLD(1.2659475207e-001), QTMFLLD(-4.0614881436e-004), QTMFLLD(-5.5389581248e-003), QTMFLLD(-5.7429198176e-002), QTMFLLD(-6.2959736586e-001), QTMFLLD(1.1892842501e-001), QTMFLLD(-4.0676075150e-004), QTMFLLD(-5.6123761460e-003), QTMFLLD(-6.0090914369e-002), QTMFLLD(-6.4299046993e-001), QTMFLLD(1.1149007827e-001), QTMFLLD(-4.0709332097e-004), QTMFLLD(-5.6832311675e-003), QTMFLLD(-6.2786586583e-002), QTMFLLD(-6.5621429682e-001), QTMFLLD(1.0428040475e-001), QTMFLLD(-4.0732545312e-004), QTMFLLD(-5.7515366934e-003), QTMFLLD(-6.5513409674e-002), QTMFLLD(-6.6925734282e-001), QTMFLLD(9.7298897803e-002), QTMFLLD(-4.0770808118e-004), QTMFLLD(-5.8172862045e-003), QTMFLLD(-6.8268470466e-002), QTMFLLD(-6.8210834265e-001), QTMFLLD(9.0545162559e-002), QTMFLLD(-4.0854664985e-004), QTMFLLD(-5.8804959990e-003), QTMFLLD(-7.1048669517e-002), QTMFLLD(-6.9475615025e-001), QTMFLLD(8.4017947316e-002), QTMFLLD(-4.1002241778e-004), QTMFLLD(-5.9412117116e-003), QTMFLLD(-7.3850922287e-002), QTMFLLD(-7.0718955994e-001), QTMFLLD(7.7716566622e-002), QTMFLLD(-4.1239586426e-004), QTMFLLD(-5.9994738549e-003), QTMFLLD(-7.6671779156e-002), QTMFLLD(-7.1939796209e-001), QTMFLLD(7.1639508009e-002), QTMFLLD(-4.1594370850e-004), QTMFLLD(-6.0553550720e-003), QTMFLLD(-7.9507902265e-002), QTMFLLD(-7.3137050867e-001), QTMFLLD(6.5784148872e-002), QTMFLLD(-4.2083335575e-004), QTMFLLD(-6.1089023948e-003), QTMFLLD(-8.2355625927e-002), QTMFLLD(-7.4309676886e-001), QTMFLLD(6.0148354620e-002), QTMFLLD(-4.2732476140e-004), QTMFLLD(-6.1602159403e-003), QTMFLLD(-8.5211075842e-002), QTMFLLD(-7.5456637144e-001), QTMFLLD(5.4730266333e-002), QTMFLLD(-4.3563771760e-004), QTMFLLD(-6.2093720771e-003), QTMFLLD(-8.8070511818e-002), QTMFLLD(-7.6576924324e-001), QTMFLLD(4.9526259303e-002), QTMFLLD(-4.4600359979e-004), QTMFLLD(-6.2565426342e-003), QTMFLLD(-9.0929701924e-002), QTMFLLD(-7.7669566870e-001), QTMFLLD(4.4533081353e-002), QTMFLLD(-4.5858716476e-004), QTMFLLD(-6.3017667271e-003), QTMFLLD(-9.3784548342e-002), QTMFLLD(-7.8733605146e-001), QTMFLLD(3.9746750146e-002), QTMFLLD(-4.7345875646e-004), QTMFLLD(-6.3452622853e-003), QTMFLLD(-9.6630692482e-002), QTMFLLD(-7.9768097401e-001), QTMFLLD(3.5163912922e-002), QTMFLLD(-4.9076689174e-004), QTMFLLD(-6.3871243037e-003), QTMFLLD(-9.9463671446e-002), QTMFLLD(-8.0772149563e-001), QTMFLLD(3.0780877918e-002), QTMFLLD(-5.1067111781e-004), QTMFLLD(-6.4275567420e-003), QTMFLLD(-1.0227891803e-001), QTMFLLD(-8.1744915247e-001), QTMFLLD(2.6590615511e-002), QTMFLLD(-5.3326232592e-004), QTMFLLD(-6.4666904509e-003), QTMFLLD(-1.0507161170e-001), QTMFLLD(-8.2685548067e-001), QTMFLLD(2.2588992491e-002), QTMFLLD(-5.5855646497e-004), QTMFLLD(-6.5047293901e-003), QTMFLLD(-1.0783691704e-001), QTMFLLD(-8.3593225479e-001), QTMFLLD(1.8772648647e-002), QTMFLLD(-5.8640236966e-004), QTMFLLD(-6.5418654121e-003), QTMFLLD(-1.1056987941e-001), QTMFLLD(-8.4467232227e-001), QTMFLLD(1.5131668188e-002), QTMFLLD(-6.1692652525e-004), QTMFLLD(-6.5783206373e-003), QTMFLLD(-1.1326543987e-001), QTMFLLD(-8.5306841135e-001), QTMFLLD(1.1661184952e-002), QTMFLLD(-6.4994930290e-004), QTMFLLD(-6.6143544391e-003), QTMFLLD(-1.1591844261e-001), QTMFLLD(-8.6111402512e-001), QTMFLLD(8.3509646356e-003), QTMFLLD(-6.8494328298e-004), QTMFLLD(-6.6502285190e-003), QTMFLLD(-1.1852371693e-001), QTMFLLD(-8.6880439520e-001), QTMFLLD(5.1832948811e-003), QTMFLLD(-7.2404538514e-004), QTMFLLD(-6.6861407831e-003), QTMFLLD(-1.2107557058e-001), QTMFLLD(-8.7612599134e-001), QTMFLLD(2.2103153169e-003), QTMFLLD(-7.7061145566e-004), QTMFLLD(-6.7221261561e-003), QTMFLLD(-1.2356808037e-001), QTMFLLD(-8.8305824995e-001), QTMFLLD(-4.6855807886e-004), QTMFLLD(-8.2158041187e-004), QTMFLLD(-6.7584766075e-003), QTMFLLD(-1.2599521875e-001), QTMFLLD(-8.8958823681e-001), QTMFLLD(-2.8003340121e-003), QTMFLLD(-8.7498105131e-004), QTMFLLD(-6.7957863212e-003), QTMFLLD(-1.2835204601e-001), QTMFLLD(-8.9572954178e-001), QTMFLLD(-4.9293786287e-003), QTMFLLD(-9.2902814504e-004), QTMFLLD(-6.8344431929e-003), QTMFLLD(-1.3063311577e-001), QTMFLLD(-9.0148586035e-001), QTMFLLD(-6.9273295812e-003), QTMFLLD(-9.8383461591e-004), QTMFLLD(-6.8746237084e-003), QTMFLLD(-1.3283239305e-001), QTMFLLD(-9.0685033798e-001), QTMFLLD(-8.7857460603e-003), QTMFLLD(-1.0395538993e-003), QTMFLLD(-6.9165546447e-003), QTMFLLD(-1.3494376838e-001), QTMFLLD(-9.1181802750e-001), QTMFLLD(-1.0507551953e-002), QTMFLLD(-1.0959620122e-003), QTMFLLD(-6.9604511373e-003), QTMFLLD(-1.3696120679e-001), QTMFLLD(-9.1638565063e-001), QTMFLLD(-1.2103702873e-002), QTMFLLD(-1.1530250777e-003), QTMFLLD(-7.0065916516e-003), QTMFLLD(-1.3887859881e-001), QTMFLLD(-9.2054879665e-001), QTMFLLD(-1.3574197888e-002), QTMFLLD(-1.2105966453e-003), QTMFLLD(-7.0552495308e-003), QTMFLLD(-1.4068968594e-001), QTMFLLD(-9.2430406809e-001), QTMFLLD(-1.4923358336e-002), QTMFLLD(-1.2681842782e-003), QTMFLLD(-7.1066003293e-003), QTMFLLD(-1.4238841832e-001), QTMFLLD(-9.2764878273e-001), QTMFLLD(-1.6156485304e-002), QTMFLLD(-1.3256429229e-003), QTMFLLD(-7.1608433500e-003), QTMFLLD(-1.4396859705e-001), QTMFLLD(-9.3058031797e-001), QTMFLLD(-1.7277117819e-002), QTMFLLD(-1.3827638468e-003), QTMFLLD(-7.2182063013e-003), QTMFLLD(-1.4542391896e-001), QTMFLLD(-9.3309664726e-001), QTMFLLD(-1.8289361149e-002), QTMFLLD(-1.4391905861e-003), QTMFLLD(-7.2789187543e-003), QTMFLLD(-1.4674818516e-001), QTMFLLD(-9.3519610167e-001), QTMFLLD(-1.9195662811e-002), QTMFLLD(-1.4947097516e-003), QTMFLLD(-7.3430840857e-003), QTMFLLD(-1.4793521166e-001), QTMFLLD(-9.3687731028e-001), QTMFLLD(-1.9999813288e-002), QTMFLLD(-1.5489540529e-003), QTMFLLD(-7.4108825065e-003), QTMFLLD(-1.4897871017e-001), QTMFLLD(-9.3813979626e-001), QTMFLLD(-2.0706148818e-002), QTMFLLD(-1.6016908921e-003), QTMFLLD(-7.4823615141e-003), QTMFLLD(-1.4987260103e-001), QTMFLLD(-9.3898290396e-001), QTMFLLD(-2.1316919476e-002), QTMFLLD(-1.6526894178e-003), QTMFLLD(-7.5576924719e-003), QTMFLLD(-1.5061059594e-001), QTMFLLD(-9.3940681219e-001), QTMFLLD(-2.1835187450e-002), QTMFLLD(-1.7015410122e-003), QTMFLLD(-7.6368991286e-003), QTMFLLD(-1.5118667483e-001), QTMFLLD(-9.3941211700e-001), QTMFLLD(-2.2264443338e-002), QTMFLLD(-1.7479787348e-003), QTMFLLD(-7.7200052328e-003), QTMFLLD(-1.5159477293e-001), QTMFLLD(-9.3899971247e-001), QTMFLLD(-2.2607907653e-002), QTMFLLD(-1.7917567166e-003), QTMFLLD(-7.8069791198e-003), QTMFLLD(-1.5182891488e-001), QTMFLLD(-9.3817096949e-001), QTMFLLD(-2.2868644446e-002), QTMFLLD(-1.8325200072e-003), QTMFLLD(-7.8977877274e-003), QTMFLLD(-1.5188319981e-001), QTMFLLD(-9.3692785501e-001), QTMFLLD(-2.3049183190e-002), QTMFLLD(-1.8700722139e-003), QTMFLLD(-7.9923402518e-003), QTMFLLD(-1.5175175667e-001), QTMFLLD(-9.3527245522e-001), QTMFLLD(-2.3152977228e-002), QTMFLLD(-1.9041235792e-003), QTMFLLD(-8.0905584618e-003), QTMFLLD(-1.5142890811e-001), QTMFLLD(-9.3320751190e-001), QTMFLLD(-2.3183524609e-002), QTMFLLD(-1.9344078610e-003), QTMFLLD(-8.1921815872e-003), QTMFLLD(-1.5090890229e-001), QTMFLLD(-9.3073624372e-001), QTMFLLD(-2.3143447936e-002), QTMFLLD(-1.9606938586e-003), QTMFLLD(-8.2970457152e-003), QTMFLLD(-1.5018628538e-001), QTMFLLD(-9.2786192894e-001), QTMFLLD(-2.3035895079e-002), QTMFLLD(-1.9826870412e-003), QTMFLLD(-8.4048351273e-003), QTMFLLD(-1.4925561845e-001), QTMFLLD(-9.2458862066e-001), QTMFLLD(-2.2864164785e-002), QTMFLLD(-2.0002126694e-003), QTMFLLD(-8.5152359679e-003), QTMFLLD(-1.4811170101e-001), QTMFLLD(-9.2092043161e-001), QTMFLLD(-2.2631708533e-002), QTMFLLD(-2.0131117199e-003), QTMFLLD(-8.6279176176e-003), QTMFLLD(-1.4674940705e-001), QTMFLLD(-9.1686213017e-001), QTMFLLD(-2.2341690958e-002), QTMFLLD(-2.0211567171e-003), QTMFLLD(-8.7425475940e-003), QTMFLLD(-1.4516362548e-001), QTMFLLD(-9.1241872311e-001), QTMFLLD(-2.1996961907e-002), QTMFLLD(-2.0242547616e-003), QTMFLLD(-8.8585643098e-003), QTMFLLD(-1.4334976673e-001), QTMFLLD(-9.0759557486e-001), QTMFLLD(-2.1601308137e-002), QTMFLLD(-2.0221893210e-003), QTMFLLD(-8.9755039662e-003), QTMFLLD(-1.4130303264e-001), QTMFLLD(-9.0239852667e-001), QTMFLLD(-2.1158147603e-002), QTMFLLD(-2.0149163902e-003), QTMFLLD(-9.0927295387e-003), QTMFLLD(-1.3901908696e-001), QTMFLLD(-8.9683371782e-001), QTMFLLD(-2.0670616999e-002), QTMFLLD(-2.0022888202e-003), QTMFLLD(-9.2095714062e-003), QTMFLLD(-1.3649365306e-001), QTMFLLD(-8.9090716839e-001), QTMFLLD(-2.0142132416e-002), QTMFLLD(-1.9841785543e-003), QTMFLLD(-9.3253115192e-003), QTMFLLD(-1.3372266293e-001), QTMFLLD(-8.8462579250e-001), QTMFLLD(-1.9576057792e-002), QTMFLLD(-1.9606270362e-003), QTMFLLD(-9.4392402098e-003), QTMFLLD(-1.3070219755e-001), QTMFLLD(-8.7799650431e-001), QTMFLLD(-1.8976125866e-002), QTMFLLD(-1.9315859536e-003), QTMFLLD(-9.5505062491e-003), QTMFLLD(-1.2742865086e-001), QTMFLLD(-8.7102663517e-001), QTMFLLD(-1.8345680088e-002), QTMFLLD(-1.8970289966e-003), QTMFLLD(-9.6583357081e-003), QTMFLLD(-1.2389861047e-001), QTMFLLD(-8.6372399330e-001), QTMFLLD(-1.7687706277e-002), QTMFLLD(-1.8569815438e-003), QTMFLLD(-9.7616901621e-003), QTMFLLD(-1.2010899931e-001), QTMFLLD(-8.5609632730e-001), QTMFLLD(-1.7006140202e-002), QTMFLLD(-1.8114587292e-003), QTMFLLD(-9.8597351462e-003), QTMFLLD(-1.1605655402e-001), QTMFLLD(-8.4815198183e-001), QTMFLLD(-1.6304368153e-002), QTMFLLD(-1.7605143366e-003), QTMFLLD(-9.9515644833e-003), QTMFLLD(-1.1173909158e-001), QTMFLLD(-8.3989918232e-001), QTMFLLD(-1.5585509129e-002), QTMFLLD(-1.7042002873e-003), QTMFLLD(-1.0036026128e-002), QTMFLLD(-1.0715358704e-001), QTMFLLD(-8.3134686947e-001), QTMFLLD(-1.4853162691e-002), QTMFLLD(-1.6426335787e-003), QTMFLLD(-1.0111952201e-002), QTMFLLD(-1.0229838639e-001), QTMFLLD(-8.2250368595e-001), QTMFLLD(-1.4110331424e-002), QTMFLLD(-1.5758809168e-003), QTMFLLD(-1.0178210214e-002), QTMFLLD(-9.7171187401e-002), QTMFLLD(-8.1337898970e-001), QTMFLLD(-1.3360806741e-002), QTMFLLD(-1.5040797880e-003), QTMFLLD(-1.0233603418e-002), QTMFLLD(-9.1770596802e-002), QTMFLLD(-8.0398184061e-001), QTMFLLD(-1.2607692741e-002), QTMFLLD(-1.4273397392e-003), QTMFLLD(-1.0276827961e-002), QTMFLLD(-8.6095176637e-002), QTMFLLD(-7.9432225227e-001), QTMFLLD(-1.1853585951e-002), QTMFLLD(-1.3458349276e-003), QTMFLLD(-1.0306579992e-002), QTMFLLD(-8.0143928528e-002), QTMFLLD(-7.8440952301e-001), QTMFLLD(-1.1102385819e-002), QTMFLLD(-1.2597256573e-003), QTMFLLD(-1.0321546346e-002), QTMFLLD(-7.3915921152e-002), QTMFLLD(-7.7425378561e-001), QTMFLLD(-1.0356968269e-002), QTMFLLD(-1.1691439431e-003), QTMFLLD(-1.0320378467e-002), QTMFLLD(-6.7410878837e-002), QTMFLLD(-7.6386493444e-001), QTMFLLD(-9.6200043336e-003), QTMFLLD(-1.0743001476e-003), QTMFLLD(-1.0301630013e-002), QTMFLLD(-6.0628447682e-002), QTMFLLD(-7.5325345993e-001), QTMFLLD(-8.8949296623e-003), QTMFLLD(-9.7535311943e-004), QTMFLLD(-1.0263898410e-002), QTMFLLD(-5.3568758070e-002), QTMFLLD(-7.4242949486e-001), QTMFLLD(-8.1837112084e-003), QTMFLLD(-8.7248592172e-004), QTMFLLD(-1.0205759667e-002), QTMFLLD(-4.6232450753e-002), QTMFLLD(-7.3140352964e-001), QTMFLLD(-7.4901022017e-003), QTMFLLD(-7.6591013931e-004), QTMFLLD(-1.0125675239e-002), QTMFLLD(-3.8619950414e-002), QTMFLLD(-7.2018599510e-001), QTMFLLD(-6.8165790290e-003), QTMFLLD(-6.5580842784e-004), QTMFLLD(-1.0022218339e-002), QTMFLLD(-3.0732547864e-002), QTMFLLD(-7.0878815651e-001), QTMFLLD(-6.1642420478e-003), QTMFLLD(-5.4247735534e-004), QTMFLLD(-9.8937284201e-003), QTMFLLD(-2.2571478039e-002), QTMFLLD(-6.9722014666e-001), QTMFLLD(-5.5373813957e-003), QTMFLLD(-4.2596619460e-004), QTMFLLD(-9.7389295697e-003), QTMFLLD(-1.4138570987e-002), QTMFLLD(-6.8549299240e-001), QTMFLLD(-4.9372608773e-003), QTMFLLD(-3.0657128082e-004), QTMFLLD(-9.5560895279e-003), QTMFLLD(-5.4356725886e-003), QTMFLLD(-6.7361742258e-001), QTMFLLD(-4.3653072789e-003), QTMFLLD(-1.8451632059e-004), QTMFLLD(-9.3438196927e-003), QTMFLLD(3.5346730147e-003), QTMFLLD(-6.6160440445e-001), QTMFLLD(-3.8251809310e-003), QTMFLLD(-6.0027297877e-005), QTMFLLD(-9.1004446149e-003), QTMFLLD(1.2770005502e-002), QTMFLLD(-6.4946544170e-001), QTMFLLD(-3.3147553913e-003), QTMFLLD(6.6618180426e-005), QTMFLLD(-8.8245263323e-003), QTMFLLD(2.2267201915e-002), QTMFLLD(-6.3721030951e-001), QTMFLLD(-2.8387091588e-003), QTMFLLD(1.9518326735e-004), QTMFLLD(-8.5145104676e-003), QTMFLLD(3.2023012638e-002), QTMFLLD(-6.2485051155e-001), QTMFLLD(-2.3975048680e-003), QTMFLLD(3.2545044087e-004), QTMFLLD(-8.1687811762e-003), QTMFLLD(4.2033810169e-002), QTMFLLD(-6.1239802837e-001), QTMFLLD(-1.9807203207e-003), QTMFLLD(4.5712510473e-004), QTMFLLD(-7.7859172598e-003), QTMFLLD(5.2295893431e-002), QTMFLLD(-5.9986191988e-001), QTMFLLD(-1.6010539839e-003), QTMFLLD(5.9015140869e-004), QTMFLLD(-7.3645371012e-003), QTMFLLD(6.2805138528e-002), QTMFLLD(-5.8725595474e-001), QTMFLLD(-1.2320743408e-003), QTMFLLD(7.2508689482e-004), QTMFLLD(-6.9030462764e-003), QTMFLLD(7.3557935655e-002), QTMFLLD(-5.7460016012e-001), QTMFLLD(-7.9492607620e-004)}; //@{ /*! \name DCT_II twiddle factors, L=64 */ /*! sin (3.14159265358979323 / (2*L) * n) , L=64*/ LNK_SECTION_CONSTDATA RAM_ALIGN const FIXP_WTP sin_twiddle_L64[] = { WTCP(0x7fffffff, 0x00000000), WTCP(0x7ff62182, 0x03242abf), WTCP(0x7fd8878e, 0x0647d97c), WTCP(0x7fa736b4, 0x096a9049), WTCP(0x7f62368f, 0x0c8bd35e), WTCP(0x7f0991c4, 0x0fab272b), WTCP(0x7e9d55fc, 0x12c8106f), WTCP(0x7e1d93ea, 0x15e21445), WTCP(0x7d8a5f40, 0x18f8b83c), WTCP(0x7ce3ceb2, 0x1c0b826a), WTCP(0x7c29fbee, 0x1f19f97b), WTCP(0x7b5d039e, 0x2223a4c5), WTCP(0x7a7d055b, 0x25280c5e), WTCP(0x798a23b1, 0x2826b928), WTCP(0x78848414, 0x2b1f34eb), WTCP(0x776c4edb, 0x2e110a62), WTCP(0x7641af3d, 0x30fbc54d), WTCP(0x7504d345, 0x33def287), WTCP(0x73b5ebd1, 0x36ba2014), WTCP(0x72552c85, 0x398cdd32), WTCP(0x70e2cbc6, 0x3c56ba70), WTCP(0x6f5f02b2, 0x3f1749b8), WTCP(0x6dca0d14, 0x41ce1e65), WTCP(0x6c242960, 0x447acd50), WTCP(0x6a6d98a4, 0x471cece7), WTCP(0x68a69e81, 0x49b41533), WTCP(0x66cf8120, 0x4c3fdff4), WTCP(0x64e88926, 0x4ebfe8a5), WTCP(0x62f201ac, 0x5133cc94), WTCP(0x60ec3830, 0x539b2af0), WTCP(0x5ed77c8a, 0x55f5a4d2), WTCP(0x5cb420e0, 0x5842dd54), WTCP(0x5a82799a, 0x5a82799a), WTCP(0x5842dd54, 0x5cb420e0), WTCP(0x55f5a4d2, 0x5ed77c8a), WTCP(0x539b2af0, 0x60ec3830), WTCP(0x5133cc94, 0x62f201ac), WTCP(0x4ebfe8a5, 0x64e88926), WTCP(0x4c3fdff4, 0x66cf8120), WTCP(0x49b41533, 0x68a69e81), WTCP(0x471cece7, 0x6a6d98a4), WTCP(0x447acd50, 0x6c242960), WTCP(0x41ce1e65, 0x6dca0d14), WTCP(0x3f1749b8, 0x6f5f02b2), WTCP(0x3c56ba70, 0x70e2cbc6), WTCP(0x398cdd32, 0x72552c85), WTCP(0x36ba2014, 0x73b5ebd1), WTCP(0x33def287, 0x7504d345), WTCP(0x30fbc54d, 0x7641af3d), WTCP(0x2e110a62, 0x776c4edb), WTCP(0x2b1f34eb, 0x78848414), WTCP(0x2826b928, 0x798a23b1), WTCP(0x25280c5e, 0x7a7d055b), WTCP(0x2223a4c5, 0x7b5d039e), WTCP(0x1f19f97b, 0x7c29fbee), WTCP(0x1c0b826a, 0x7ce3ceb2), WTCP(0x18f8b83c, 0x7d8a5f40), WTCP(0x15e21445, 0x7e1d93ea), WTCP(0x12c8106f, 0x7e9d55fc), WTCP(0x0fab272b, 0x7f0991c4), WTCP(0x0c8bd35e, 0x7f62368f), WTCP(0x096a9049, 0x7fa736b4), WTCP(0x0647d97c, 0x7fd8878e), WTCP(0x03242abf, 0x7ff62182)}; const USHORT sqrt_tab[49] = { 0x5a82, 0x5d4b, 0x6000, 0x62a1, 0x6531, 0x67b1, 0x6a21, 0x6c84, 0x6ed9, 0x7123, 0x7360, 0x7593, 0x77bb, 0x79da, 0x7bef, 0x7dfb, 0x8000, 0x81fc, 0x83f0, 0x85dd, 0x87c3, 0x89a3, 0x8b7c, 0x8d4e, 0x8f1b, 0x90e2, 0x92a4, 0x9460, 0x9617, 0x97ca, 0x9977, 0x9b20, 0x9cc4, 0x9e64, 0xa000, 0xa197, 0xa32b, 0xa4ba, 0xa646, 0xa7cf, 0xa953, 0xaad5, 0xac53, 0xadcd, 0xaf45, 0xb0b9, 0xb22b, 0xb399, 0xb504}; LNK_SECTION_CONSTDATA_L1 const FIXP_DBL invCount[80] = /* This could be 16-bit wide */ {0x00000000, 0x7fffffff, 0x40000000, 0x2aaaaaab, 0x20000000, 0x1999999a, 0x15555555, 0x12492492, 0x10000000, 0x0e38e38e, 0x0ccccccd, 0x0ba2e8ba, 0x0aaaaaab, 0x09d89d8a, 0x09249249, 0x08888889, 0x08000000, 0x07878788, 0x071c71c7, 0x06bca1af, 0x06666666, 0x06186186, 0x05d1745d, 0x0590b216, 0x05555555, 0x051eb852, 0x04ec4ec5, 0x04bda12f, 0x04924925, 0x0469ee58, 0x04444444, 0x04210842, 0x04000000, 0x03e0f83e, 0x03c3c3c4, 0x03a83a84, 0x038e38e4, 0x03759f23, 0x035e50d8, 0x03483483, 0x03333333, 0x031f3832, 0x030c30c3, 0x02fa0be8, 0x02e8ba2f, 0x02d82d83, 0x02c8590b, 0x02b93105, 0x02aaaaab, 0x029cbc15, 0x028f5c29, 0x02828283, 0x02762762, 0x026a439f, 0x025ed098, 0x0253c825, 0x02492492, 0x023ee090, 0x0234f72c, 0x022b63cc, 0x02222222, 0x02192e2a, 0x02108421, 0x02082082, 0x02000000, 0x01f81f82, 0x01f07c1f, 0x01e9131b, 0x01e1e1e2, 0x01dae607, 0x01d41d42, 0x01cd8569, 0x01c71c72, 0x01c0e070, 0x01bacf91, 0x01b4e81b, 0x01af286c, 0x01a98ef6, 0x01a41a42, 0x019ec8e9}; /* * Bitstream data lists */ /* * AOT {2,5,29} * epConfig = -1 */ static const rbd_id_t el_aac_sce[] = { adtscrc_start_reg1, element_instance_tag, global_gain, ics_info, section_data, scale_factor_data, pulse, tns_data_present, tns_data, gain_control_data_present, /* gain_control_data, */ spectral_data, adtscrc_end_reg1, end_of_sequence}; static const struct element_list node_aac_sce = {el_aac_sce, {NULL, NULL}}; /* CCE */ static const rbd_id_t el_aac_cce[] = { adtscrc_start_reg1, element_instance_tag, coupled_elements, /* CCE specific */ global_gain, ics_info, section_data, scale_factor_data, pulse, tns_data_present, tns_data, gain_control_data_present, /* gain_control_data, */ spectral_data, gain_element_lists, /* CCE specific */ adtscrc_end_reg1, end_of_sequence}; static const struct element_list node_aac_cce = {el_aac_cce, {NULL, NULL}}; static const rbd_id_t el_aac_cpe[] = {adtscrc_start_reg1, element_instance_tag, common_window, link_sequence}; static const rbd_id_t el_aac_cpe0[] = { /*common_window = 0*/ global_gain, ics_info, section_data, scale_factor_data, pulse, tns_data_present, tns_data, gain_control_data_present, /*gain_control_data,*/ spectral_data, next_channel, adtscrc_start_reg2, global_gain, ics_info, section_data, scale_factor_data, pulse, tns_data_present, tns_data, gain_control_data_present, /*gain_control_data,*/ spectral_data, adtscrc_end_reg1, adtscrc_end_reg2, end_of_sequence}; static const rbd_id_t el_aac_cpe1[] = { /* common_window = 1 */ ics_info, ms, global_gain, section_data, scale_factor_data, pulse, tns_data_present, tns_data, gain_control_data_present, /*gain_control_data,*/ spectral_data, next_channel, adtscrc_start_reg2, global_gain, section_data, scale_factor_data, pulse, tns_data_present, tns_data, gain_control_data_present, /*gain_control_data,*/ spectral_data, adtscrc_end_reg1, adtscrc_end_reg2, end_of_sequence}; static const struct element_list node_aac_cpe0 = {el_aac_cpe0, {NULL, NULL}}; static const struct element_list node_aac_cpe1 = {el_aac_cpe1, {NULL, NULL}}; static const element_list_t node_aac_cpe = {el_aac_cpe, {&node_aac_cpe0, &node_aac_cpe1}}; /* * AOT C- {17,23} * epConfig = 0,1 */ static const rbd_id_t el_aac_sce_epc0[] = { element_instance_tag, global_gain, ics_info, section_data, scale_factor_data, pulse, tns_data_present, gain_control_data_present, gain_control_data, esc1_hcr, /*length_of_rvlc_escapes, length_of_rvlc_sf */ esc2_rvlc, /* rvlc_cod_sf, rvlc_esc_sf */ tns_data, spectral_data, end_of_sequence}; static const struct element_list node_aac_sce_epc0 = {el_aac_sce_epc0, {NULL, NULL}}; static const rbd_id_t el_aac_sce_epc1[] = { element_instance_tag, global_gain, ics_info, section_data, scale_factor_data, pulse, tns_data_present, gain_control_data_present, /*gain_control_data,*/ esc1_hcr, /*length_of_rvlc_escapes, length_of_rvlc_sf */ esc2_rvlc, /* rvlc_cod_sf, rvlc_esc_sf */ tns_data, spectral_data, end_of_sequence}; static const struct element_list node_aac_sce_epc1 = {el_aac_sce_epc1, {NULL, NULL}}; static const rbd_id_t el_aac_cpe_epc0[] = {element_instance_tag, common_window, link_sequence}; static const rbd_id_t el_aac_cpe0_epc0[] = { /* common_window = 0 */ /* ESC 1: */ global_gain, ics_info, /* ltp_data_present, ltp_data, */ section_data, scale_factor_data, pulse, tns_data_present, gain_control_data_present, /*gain_control_data,*/ esc1_hcr, /*length_of_rvlc_escapes, length_of_rvlc_sf */ /* ESC 2: */ esc2_rvlc, /* rvlc_cod_sf, rvlc_esc_sf */ /* ESC 3: */ tns_data, /* ESC 4: */ spectral_data, next_channel, /* ESC 1: */ global_gain, ics_info, /* ltp_data_present, ltp_data, */ section_data, scale_factor_data, pulse, tns_data_present, gain_control_data_present, /*gain_control_data,*/ esc1_hcr, /*length_of_rvlc_escapes, length_of_rvlc_sf */ /* ESC 2: */ esc2_rvlc, /* rvlc_cod_sf, rvlc_esc_sf */ /* ESC 3: */ tns_data, /* ESC 4: */ spectral_data, end_of_sequence}; static const rbd_id_t el_aac_cpe1_epc0[] = { /* common_window = 1 */ /* ESC 0: */ ics_info, /* ltp_data_present, ltp_data, next_channel, ltp_data_present, ltp_data, next_channel, */ ms, /* ESC 1: */ global_gain, section_data, scale_factor_data, pulse, tns_data_present, gain_control_data_present, /*gain_control_data,*/ esc1_hcr, /* length_of_reordered_spectral_data, length_of_longest_codeword */ /* ESC 2: */ esc2_rvlc, /* rvlc_cod_sf, rvlc_esc_sf */ /* ESC 3: */ tns_data, /* ESC 4: */ spectral_data, next_channel, /* ESC 1: */ global_gain, section_data, scale_factor_data, pulse, tns_data_present, gain_control_data_present, /*gain_control_data,*/ esc1_hcr, /* length_of_reordered_spectral_data, length_of_longest_codeword */ /* ESC 2: */ esc2_rvlc, /* rvlc_cod_sf, rvlc_esc_sf */ /* ESC 3: */ tns_data, /* ESC 4: */ spectral_data, end_of_sequence}; static const struct element_list node_aac_cpe0_epc0 = {el_aac_cpe0_epc0, {NULL, NULL}}; static const struct element_list node_aac_cpe1_epc0 = {el_aac_cpe1_epc0, {NULL, NULL}}; static const element_list_t node_aac_cpe_epc0 = { el_aac_cpe_epc0, {&node_aac_cpe0_epc0, &node_aac_cpe1_epc0}}; static const rbd_id_t el_aac_cpe0_epc1[] = { global_gain, ics_info, section_data, scale_factor_data, pulse, tns_data_present, gain_control_data_present, /*gain_control_data,*/ next_channel, global_gain, ics_info, section_data, scale_factor_data, pulse, tns_data_present, gain_control_data_present, /*gain_control_data,*/ next_channel, esc1_hcr, /*length_of_rvlc_escapes, length_of_rvlc_sf */ next_channel, esc1_hcr, /*length_of_rvlc_escapes, length_of_rvlc_sf */ next_channel, esc2_rvlc, /* rvlc_cod_sf, rvlc_esc_sf */ next_channel, esc2_rvlc, /* rvlc_cod_sf, rvlc_esc_sf */ next_channel, tns_data, next_channel, tns_data, next_channel, spectral_data, next_channel, spectral_data, end_of_sequence}; static const rbd_id_t el_aac_cpe1_epc1[] = { ics_info, ms, ltp_data_present, /* ltp_data, */ global_gain, section_data, scale_factor_data, pulse, tns_data_present, gain_control_data_present, /*gain_control_data,*/ next_channel, ltp_data_present, /* ltp_data, */ global_gain, section_data, scale_factor_data, pulse, tns_data_present, gain_control_data_present, /*gain_control_data,*/ next_channel, esc1_hcr, /*length_of_rvlc_escapes, length_of_rvlc_sf */ next_channel, esc1_hcr, /*length_of_rvlc_escapes, length_of_rvlc_sf */ next_channel, esc2_rvlc, /* rvlc_cod_sf, rvlc_esc_sf */ next_channel, esc2_rvlc, /* rvlc_cod_sf, rvlc_esc_sf */ next_channel, tns_data, next_channel, tns_data, next_channel, spectral_data, next_channel, spectral_data, end_of_sequence}; static const struct element_list node_aac_cpe0_epc1 = {el_aac_cpe0_epc1, {NULL, NULL}}; static const struct element_list node_aac_cpe1_epc1 = {el_aac_cpe1_epc1, {NULL, NULL}}; static const element_list_t node_aac_cpe_epc1 = { el_aac_cpe, {&node_aac_cpe0_epc1, &node_aac_cpe1_epc1}}; /* * AOT = 20 * epConfig = 0 */ static const rbd_id_t el_scal_sce_epc0[] = {ics_info, /* ESC 1 */ tns_data_present, ltp_data_present, /* ltp_data, */ global_gain, section_data, scale_factor_data, esc1_hcr, esc2_rvlc, /* ESC 2 */ tns_data, /* ESC 3 */ spectral_data, /* ESC 4 */ end_of_sequence}; static const struct element_list node_scal_sce_epc0 = {el_scal_sce_epc0, {NULL, NULL}}; static const rbd_id_t el_scal_cpe_epc0[] = { ics_info, /* ESC 0 */ ms, tns_data_present, /* ESC 1 (ch 0) */ ltp_data_present, /* ltp_data, */ global_gain, section_data, scale_factor_data, esc1_hcr, esc2_rvlc, /* ESC 2 (ch 0) */ tns_data, /* ESC 3 (ch 0) */ spectral_data, /* ESC 4 (ch 0) */ next_channel, tns_data_present, /* ESC 1 (ch 1) */ ltp_data_present, global_gain, section_data, scale_factor_data, esc1_hcr, esc2_rvlc, /* ESC 2 (ch 1) */ tns_data, /* ESC 3 (ch 1) */ spectral_data, /* ESC 4 (ch 1) */ end_of_sequence}; static const struct element_list node_scal_cpe_epc0 = {el_scal_cpe_epc0, {NULL, NULL}}; /* * AOT = 20 * epConfig = 1 */ static const rbd_id_t el_scal_sce_epc1[] = { ics_info, tns_data_present, ltp_data_present, /* ltp_data, */ global_gain, section_data, scale_factor_data, esc1_hcr, tns_data, spectral_data, end_of_sequence}; static const struct element_list node_scal_sce_epc1 = {el_scal_sce_epc1, {NULL, NULL}}; static const rbd_id_t el_scal_cpe_epc1[] = { ics_info, ms, tns_data_present, ltp_data_present, /* ltp_data, */ global_gain, section_data, scale_factor_data, esc1_hcr, next_channel, tns_data_present, ltp_data_present, /* ltp_data, */ global_gain, section_data, scale_factor_data, esc1_hcr, next_channel, tns_data, next_channel, tns_data, next_channel, spectral_data, next_channel, spectral_data, end_of_sequence}; static const struct element_list node_scal_cpe_epc1 = {el_scal_cpe_epc1, {NULL, NULL}}; /* * Pseudo AOT for DRM/DRM+ (similar to AOT 20) */ static const rbd_id_t el_drm_sce[] = { drmcrc_start_reg, ics_info, tns_data_present, ltp_data_present, /* ltp_data, */ global_gain, section_data, scale_factor_data, esc1_hcr, tns_data, drmcrc_end_reg, spectral_data, end_of_sequence}; static const struct element_list node_drm_sce = {el_drm_sce, {NULL, NULL}}; static const rbd_id_t el_drm_cpe[] = { drmcrc_start_reg, ics_info, ms, tns_data_present, ltp_data_present, /* ltp_data, */ global_gain, section_data, scale_factor_data, esc1_hcr, next_channel, tns_data_present, ltp_data_present, /* ltp_data, */ global_gain, section_data, scale_factor_data, esc1_hcr, next_channel, tns_data, next_channel, tns_data, drmcrc_end_reg, next_channel, spectral_data, next_channel, spectral_data, end_of_sequence}; static const struct element_list node_drm_cpe = {el_drm_cpe, {NULL, NULL}}; /* * AOT = 39 * epConfig = 0 */ static const rbd_id_t el_eld_sce_epc0[] = { global_gain, ics_info, section_data, scale_factor_data, tns_data_present, tns_data, esc1_hcr, esc2_rvlc, spectral_data, end_of_sequence}; static const struct element_list node_eld_sce_epc0 = {el_eld_sce_epc0, {NULL, NULL}}; #define node_eld_sce_epc1 node_eld_sce_epc0 static const rbd_id_t el_eld_cpe_epc0[] = {ics_info, ms, global_gain, section_data, scale_factor_data, tns_data_present, tns_data, esc1_hcr, esc2_rvlc, spectral_data, next_channel, global_gain, section_data, scale_factor_data, tns_data_present, tns_data, esc1_hcr, esc2_rvlc, spectral_data, end_of_sequence}; static const rbd_id_t el_eld_cpe_epc1[] = {ics_info, ms, global_gain, section_data, scale_factor_data, tns_data_present, next_channel, global_gain, section_data, scale_factor_data, tns_data_present, next_channel, tns_data, next_channel, tns_data, next_channel, esc1_hcr, esc2_rvlc, spectral_data, next_channel, esc1_hcr, esc2_rvlc, spectral_data, end_of_sequence}; static const struct element_list node_eld_cpe_epc0 = {el_eld_cpe_epc0, {NULL, NULL}}; static const struct element_list node_eld_cpe_epc1 = {el_eld_cpe_epc1, {NULL, NULL}}; /* * AOT = 42 * epConfig = 0 */ static const rbd_id_t el_usac_coremode[] = {core_mode, next_channel, link_sequence}; static const rbd_id_t el_usac_sce0_epc0[] = { tns_data_present, /* fd_channel_stream */ global_gain, noise, ics_info, tw_data, scale_factor_data_usac, tns_data, ac_spectral_data, fac_data, end_of_sequence}; static const rbd_id_t el_usac_lfe_epc0[] = { /* fd_channel_stream */ global_gain, ics_info, scale_factor_data_usac, ac_spectral_data, fac_data, end_of_sequence}; static const rbd_id_t el_usac_lpd_epc0[] = {lpd_channel_stream, end_of_sequence}; static const struct element_list node_usac_sce0_epc0 = {el_usac_sce0_epc0, {NULL, NULL}}; static const struct element_list node_usac_sce1_epc0 = {el_usac_lpd_epc0, {NULL, NULL}}; static const struct element_list node_usac_sce_epc0 = { el_usac_coremode, {&node_usac_sce0_epc0, &node_usac_sce1_epc0}}; static const rbd_id_t list_usac_cpe00_epc0[] = {tns_active, common_window, link_sequence}; static const rbd_id_t el_usac_common_tw[] = {common_tw, link_sequence}; static const rbd_id_t list_usac_cpe0000_epc0[] = { /* core_mode0 = 0 */ /* core_mode1 = 0 */ /* common_window = 0 */ /* common_tw = 0 */ tns_data_present_usac, global_gain, noise, ics_info, tw_data, scale_factor_data_usac, tns_data, ac_spectral_data, fac_data, next_channel, global_gain, noise, ics_info, tw_data, scale_factor_data_usac, tns_data, ac_spectral_data, fac_data, end_of_sequence}; static const rbd_id_t list_usac_cpe0001_epc0[] = { /* core_mode0 = 0 core_mode1 = 0 common_window = 0 common_tw = 1 */ tw_data, tns_data_present_usac, global_gain, noise, ics_info, scale_factor_data_usac, tns_data, ac_spectral_data, fac_data, next_channel, global_gain, noise, ics_info, scale_factor_data_usac, tns_data, ac_spectral_data, fac_data, end_of_sequence}; static const rbd_id_t list_usac_cpe001_epc0[] = { /* core_mode0 = 0 */ /* core_mode1 = 0 */ /* common_window = 1 */ ics_info, common_max_sfb, ms, common_tw, link_sequence}; static const rbd_id_t list_usac_cpe0010_epc0[] = { /* core_mode0 = 0 */ /* core_mode1 = 0 */ /* common_window = 1 */ /* common_tw = 0 */ tns_data_present_usac, global_gain, noise, tw_data, scale_factor_data_usac, tns_data, ac_spectral_data, fac_data, next_channel, global_gain, noise, tw_data, scale_factor_data_usac, tns_data, ac_spectral_data, fac_data, end_of_sequence}; static const rbd_id_t list_usac_cpe0011_epc0[] = { /* core_mode0 = 0 */ /* core_mode1 = 0 */ /* common_window = 1 */ /* common_tw = 1 */ tw_data, tns_data_present_usac, global_gain, noise, scale_factor_data_usac, tns_data, ac_spectral_data, fac_data, next_channel, global_gain, noise, scale_factor_data_usac, tns_data, ac_spectral_data, fac_data, end_of_sequence}; static const rbd_id_t list_usac_cpe10_epc0[] = { /* core_mode0 = 1 */ /* core_mode1 = 0 */ lpd_channel_stream, next_channel, tns_data_present, global_gain, noise, ics_info, tw_data, scale_factor_data_usac, tns_data, ac_spectral_data, fac_data, end_of_sequence}; static const rbd_id_t list_usac_cpe01_epc0[] = { /* core_mode0 = 0 */ /* core_mode1 = 1 */ tns_data_present, global_gain, noise, ics_info, tw_data, scale_factor_data_usac, tns_data, ac_spectral_data, fac_data, next_channel, lpd_channel_stream, end_of_sequence}; static const rbd_id_t list_usac_cpe11_epc0[] = { /* core_mode0 = 1 */ /* core_mode1 = 1 */ lpd_channel_stream, next_channel, lpd_channel_stream, end_of_sequence}; static const struct element_list node_usac_cpe0000_epc0 = { /* core_mode0 = 0 */ /* core_mode1 = 0 */ /* common_window = 0 */ /* common_tw = 0 */ list_usac_cpe0000_epc0, {NULL, NULL}}; static const struct element_list node_usac_cpe0010_epc0 = { /* core_mode0 = 0 */ /* core_mode1 = 0 */ /* common_window = 1 */ /* common_tw = 0 */ list_usac_cpe0010_epc0, {NULL, NULL}}; static const struct element_list node_usac_cpe0001_epc0 = { /* core_mode0 = 0 */ /* core_mode1 = 0 */ /* common_window = 0 */ /* common_tw = 1 */ list_usac_cpe0001_epc0, {NULL, NULL}}; static const struct element_list node_usac_cpe0011_epc0 = { /* core_mode0 = 0 */ /* core_mode1 = 0 */ /* common_window = 1 */ /* common_tw = 1 */ list_usac_cpe0011_epc0, {NULL, NULL}}; static const struct element_list node_usac_cpe000_epc0 = { /* core_mode0 = 0 */ /* core_mode1 = 0 */ /* common_window = 0 */ el_usac_common_tw, {&node_usac_cpe0000_epc0, &node_usac_cpe0001_epc0}}; static const struct element_list node_usac_cpe001_epc0 = { list_usac_cpe001_epc0, {&node_usac_cpe0010_epc0, &node_usac_cpe0011_epc0}}; static const struct element_list node_usac_cpe00_epc0 = { /* core_mode0 = 0 */ /* core_mode1 = 0 */ list_usac_cpe00_epc0, {&node_usac_cpe000_epc0, &node_usac_cpe001_epc0}}; static const struct element_list node_usac_cpe10_epc0 = { /* core_mode0 = 1 */ /* core_mode1 = 0 */ list_usac_cpe10_epc0, {NULL, NULL}}; static const struct element_list node_usac_cpe01_epc0 = {list_usac_cpe01_epc0, {NULL, NULL}}; static const struct element_list node_usac_cpe11_epc0 = {list_usac_cpe11_epc0, {NULL, NULL}}; static const struct element_list node_usac_cpe0_epc0 = { /* core_mode0 = 0 */ el_usac_coremode, {&node_usac_cpe00_epc0, &node_usac_cpe01_epc0}}; static const struct element_list node_usac_cpe1_epc0 = { /* core_mode0 = 1 */ el_usac_coremode, {&node_usac_cpe10_epc0, &node_usac_cpe11_epc0}}; static const struct element_list node_usac_cpe_epc0 = { el_usac_coremode, {&node_usac_cpe0_epc0, &node_usac_cpe1_epc0}}; static const struct element_list node_usac_lfe_epc0 = {el_usac_lfe_epc0, {NULL, NULL}}; const element_list_t *getBitstreamElementList(AUDIO_OBJECT_TYPE aot, SCHAR epConfig, UCHAR nChannels, UCHAR layer, UINT elFlags) { switch (aot) { case AOT_AAC_LC: case AOT_SBR: case AOT_PS: FDK_ASSERT(epConfig == -1); if (elFlags & AC_EL_GA_CCE) { return &node_aac_cce; } else { if (nChannels == 1) { return &node_aac_sce; } else { return &node_aac_cpe; } } case AOT_ER_AAC_LC: case AOT_ER_AAC_LD: if (nChannels == 1) { if (epConfig == 0) { return &node_aac_sce_epc0; } else { return &node_aac_sce_epc1; } } else { if (epConfig == 0) return &node_aac_cpe_epc0; else return &node_aac_cpe_epc1; } case AOT_USAC: if (elFlags & AC_EL_USAC_LFE) { FDK_ASSERT(nChannels == 1); return &node_usac_lfe_epc0; } if (nChannels == 1) { return &node_usac_sce_epc0; } else { return &node_usac_cpe_epc0; } case AOT_ER_AAC_SCAL: if (nChannels == 1) { if (epConfig <= 0) return &node_scal_sce_epc0; else return &node_scal_sce_epc1; } else { if (epConfig <= 0) return &node_scal_cpe_epc0; else return &node_scal_cpe_epc1; } case AOT_ER_AAC_ELD: if (nChannels == 1) { if (epConfig <= 0) return &node_eld_sce_epc0; else return &node_eld_sce_epc1; } else { if (epConfig <= 0) return &node_eld_cpe_epc0; else return &node_eld_cpe_epc1; } case AOT_DRM_AAC: case AOT_DRM_SBR: case AOT_DRM_MPEG_PS: case AOT_DRM_SURROUND: FDK_ASSERT(epConfig == 1); if (nChannels == 1) { return &node_drm_sce; } else { return &node_drm_cpe; } default: break; } return NULL; } /* Inverse square root table for operands running from 0.5 to ~1.0 */ /* (INT) (0.5 + 1.0/sqrt((op)/FDKpow(2.0,31))); */ /* Note: First value is rnot rounded for accuracy reasons */ /* Implicit exponent is 1. */ /* Examples: 0x5A82799A = invSqrtNorm2 (0x4000.0000), exp=1 */ /* 0x5A82799A = invSqrtNorm2 (0x4000.0000), exp=1 */ LNK_SECTION_CONSTDATA_L1 const FIXP_DBL invSqrtTab[SQRT_VALUES] = { 0x5A827999, 0x5A287E03, 0x59CF8CBC, 0x5977A0AC, 0x5920B4DF, 0x58CAC480, 0x5875CADE, 0x5821C364, 0x57CEA99D, 0x577C7930, 0x572B2DE0, 0x56DAC38E, 0x568B3632, 0x563C81E0, 0x55EEA2C4, 0x55A19522, 0x55555555, 0x5509DFD0, 0x54BF311A, 0x547545D0, 0x542C1AA4, 0x53E3AC5B, 0x539BF7CD, 0x5354F9E7, 0x530EAFA5, 0x52C91618, 0x52842A5F, 0x523FE9AC, 0x51FC5140, 0x51B95E6B, 0x51770E8F, 0x51355F1A, 0x50F44D89, 0x50B3D768, 0x5073FA50, 0x5034B3E7, 0x4FF601E0, 0x4FB7E1FA, 0x4F7A5202, 0x4F3D4FCF, 0x4F00D944, 0x4EC4EC4F, 0x4E8986EA, 0x4E4EA718, 0x4E144AE9, 0x4DDA7073, 0x4DA115DA, 0x4D683948, 0x4D2FD8F4, 0x4CF7F31B, 0x4CC08605, 0x4C899000, 0x4C530F65, 0x4C1D0294, 0x4BE767F5, 0x4BB23DF9, 0x4B7D8317, 0x4B4935CF, 0x4B1554A6, 0x4AE1DE2A, 0x4AAED0F0, 0x4A7C2B93, 0x4A49ECB3, 0x4A1812FA, 0x49E69D16, 0x49B589BB, 0x4984D7A4, 0x49548592, 0x49249249, 0x48F4FC97, 0x48C5C34B, 0x4896E53D, 0x48686148, 0x483A364D, 0x480C6332, 0x47DEE6E1, 0x47B1C049, 0x4784EE60, 0x4758701C, 0x472C447C, 0x47006A81, 0x46D4E130, 0x46A9A794, 0x467EBCBA, 0x46541FB4, 0x4629CF98, 0x45FFCB80, 0x45D6128A, 0x45ACA3D5, 0x45837E88, 0x455AA1CB, 0x45320CC8, 0x4509BEB0, 0x44E1B6B4, 0x44B9F40B, 0x449275ED, 0x446B3B96, 0x44444444, 0x441D8F3B, 0x43F71BBF, 0x43D0E917, 0x43AAF68F, 0x43854374, 0x435FCF15, 0x433A98C6, 0x43159FDC, 0x42F0E3AE, 0x42CC6398, 0x42A81EF6, 0x42841527, 0x4260458E, 0x423CAF8D, 0x4219528B, 0x41F62DF2, 0x41D3412A, 0x41B08BA2, 0x418E0CC8, 0x416BC40D, 0x4149B0E5, 0x4127D2C3, 0x41062920, 0x40E4B374, 0x40C3713B, 0x40A261EF, 0x40818512, 0x4060DA22, 0x404060A1, 0x40201814, 0x40000000, 0x3FE017EC /* , 0x3FC05F61 */ }; /* number of channels of the formats */ const INT format_nchan[FDK_NFORMATS + 9 - 2] = { 0, /* any set-up, ChConfIdx = 0 */ 1, /* mono ChConfIdx = 1 */ 2, /* stereo ChConfIdx = 2 */ 3, /* 3/0.0 ChConfIdx = 3 */ 4, /* 3/1.0 ChConfIdx = 4 */ 5, /* 3/2.0 ChConfIdx = 5 */ 6, /* 5.1 ChConfIdx = 6 */ 8, /* 5/2.1 ALT ChConfIdx = 7 */ 0, /* Empty n.a. ChConfIdx = 8 */ 3, /* 2/1.0 ChConfIdx = 9 */ 4, /* 2/2.0 ChConfIdx = 10 */ 7, /* 3/3.1 ChConfIdx = 11 */ 8, /* 3/4.1 ChConfIdx = 12 */ 24, /* 22.2 ChConfIdx = 13 */ 8, /* 5/2.1 ChConfIdx = 14 */ 12, /* 5/5.2 ChConfIdx = 15 */ 10, /* 5/4.1 ChConfIdx = 16 */ 12, /* 6/5.1 ChConfIdx = 17 */ 14, /* 6/7.1 ChConfIdx = 18 */ 12, /* 5/6.1 ChConfIdx = 19 */ 14 /* 7/6.1 ChConfIdx = 20 */ }; fdk-aac-2.0.2/libFDK/src/FDK_trigFcts.cpp000066400000000000000000000304131404225310700176370ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Haricharan Lakshman, Manuel Jander Description: Trigonometric functions fixed point fractional implementation. *******************************************************************************/ #include "FDK_trigFcts.h" #include "fixpoint_math.h" #define IMPROVE_ATAN2_ACCURACY 1 /* 0 --> 59 dB SNR 1 --> 65 dB SNR */ #define MINSFTAB 7 #define MAXSFTAB 25 #if IMPROVE_ATAN2_ACCURACY static const FIXP_DBL f_atan_expand_range[MAXSFTAB - (MINSFTAB - 1)] = { /***************************************************************************** * * Table holds fixp_atan() output values which are outside of input range * of fixp_atan() to improve SNR of fixp_atan2(). * * This Table might also be used in fixp_atan() so there a wider input * range can be covered, too. * *****************************************************************************/ FL2FXCONST_DBL(7.775862990872099e-001), FL2FXCONST_DBL(7.814919928673978e-001), FL2FXCONST_DBL(7.834450483314648e-001), FL2FXCONST_DBL(7.844216021392089e-001), FL2FXCONST_DBL(7.849098823026687e-001), FL2FXCONST_DBL(7.851540227918509e-001), FL2FXCONST_DBL(7.852760930873737e-001), FL2FXCONST_DBL(7.853371282415015e-001), FL2FXCONST_DBL(7.853676458193612e-001), FL2FXCONST_DBL(7.853829046083906e-001), FL2FXCONST_DBL(7.853905340029177e-001), FL2FXCONST_DBL(7.853943487001828e-001), FL2FXCONST_DBL(7.853962560488155e-001), FL2FXCONST_DBL(7.853972097231319e-001), FL2FXCONST_DBL(7.853976865602901e-001), FL2FXCONST_DBL(7.853979249788692e-001), FL2FXCONST_DBL(7.853980441881587e-001), FL2FXCONST_DBL(7.853981037928035e-001), FL2FXCONST_DBL(7.853981335951259e-001) /* pi/4 = 0.785398163397448 = pi/2/ATO_SCALE */ }; #endif FIXP_DBL fixp_atan2(FIXP_DBL y, FIXP_DBL x) { FIXP_DBL q; FIXP_DBL at; /* atan out */ FIXP_DBL at2; /* atan2 out */ FIXP_DBL ret = FL2FXCONST_DBL(-1.0f); INT sf, sfo, stf; /* --- division */ if (y > FL2FXCONST_DBL(0.0f)) { if (x > FL2FXCONST_DBL(0.0f)) { q = fDivNormHighPrec(y, x, &sf); /* both pos. */ } else if (x < FL2FXCONST_DBL(0.0f)) { q = -fDivNormHighPrec(y, -x, &sf); /* x neg. */ } else { /* (x == FL2FXCONST_DBL(0.0f)) */ q = FL2FXCONST_DBL(+1.0f); /* y/x = pos/zero = +Inf */ sf = 0; } } else if (y < FL2FXCONST_DBL(0.0f)) { if (x > FL2FXCONST_DBL(0.0f)) { q = -fDivNormHighPrec(-y, x, &sf); /* y neg. */ } else if (x < FL2FXCONST_DBL(0.0f)) { q = fDivNormHighPrec(-y, -x, &sf); /* both neg. */ } else { /* (x == FL2FXCONST_DBL(0.0f)) */ q = FL2FXCONST_DBL(-1.0f); /* y/x = neg/zero = -Inf */ sf = 0; } } else { /* (y == FL2FXCONST_DBL(0.0f)) */ q = FL2FXCONST_DBL(0.0f); sf = 0; } sfo = sf; /* --- atan() */ if (sfo > ATI_SF) { /* --- could not calc fixp_atan() here bec of input data out of range */ /* ==> therefore give back boundary values */ #if IMPROVE_ATAN2_ACCURACY if (sfo > MAXSFTAB) sfo = MAXSFTAB; #endif if (q > FL2FXCONST_DBL(0.0f)) { #if IMPROVE_ATAN2_ACCURACY at = +f_atan_expand_range[sfo - ATI_SF - 1]; #else at = FL2FXCONST_DBL(+M_PI / 2 / ATO_SCALE); #endif } else if (q < FL2FXCONST_DBL(0.0f)) { #if IMPROVE_ATAN2_ACCURACY at = -f_atan_expand_range[sfo - ATI_SF - 1]; #else at = FL2FXCONST_DBL(-M_PI / 2 / ATO_SCALE); #endif } else { /* q == FL2FXCONST_DBL(0.0f) */ at = FL2FXCONST_DBL(0.0f); } } else { /* --- calc of fixp_atan() is possible; input data within range */ /* ==> set q on fixed scale level as desired from fixp_atan() */ stf = sfo - ATI_SF; if (stf > 0) q = q << (INT)fMin(stf, DFRACT_BITS - 1); else q = q >> (INT)fMin(-stf, DFRACT_BITS - 1); at = fixp_atan(q); /* ATO_SF */ } // --- atan2() at2 = at >> (AT2O_SF - ATO_SF); // now AT2O_SF for atan2 if (x > FL2FXCONST_DBL(0.0f)) { ret = at2; } else if (x < FL2FXCONST_DBL(0.0f)) { if (y >= FL2FXCONST_DBL(0.0f)) { ret = at2 + FL2FXCONST_DBL(M_PI / AT2O_SCALE); } else { ret = at2 - FL2FXCONST_DBL(M_PI / AT2O_SCALE); } } else { // x == 0 if (y > FL2FXCONST_DBL(0.0f)) { ret = FL2FXCONST_DBL(+M_PI / 2 / AT2O_SCALE); } else if (y < FL2FXCONST_DBL(0.0f)) { ret = FL2FXCONST_DBL(-M_PI / 2 / AT2O_SCALE); } else if (y == FL2FXCONST_DBL(0.0f)) { ret = FL2FXCONST_DBL(0.0f); } } return ret; } FIXP_DBL fixp_atan(FIXP_DBL x) { INT sign; FIXP_DBL result, temp; /* SNR of fixp_atan() = 56 dB */ FIXP_DBL P281 = (FIXP_DBL)0x00013000; // 0.281 in q18 FIXP_DBL ONEP571 = (FIXP_DBL)0x6487ef00; // 1.571 in q30 if (x < FIXP_DBL(0)) { sign = 1; x = -x; } else { sign = 0; } FDK_ASSERT(FL2FXCONST_DBL(1.0 / 64.0) == Q(Q_ATANINP)); /* calc of arctan */ if (x < FL2FXCONST_DBL(1.0 / 64.0)) /* Chebyshev polynomial approximation of atan(x) 5th-order approximation: atan(x) = a1*x + a2*x^3 + a3*x^5 = x(a1 + x^2*(a2 + a3*x^2)); a1 = 0.9949493661166540f, a2 = 0.2870606355326520f, a3 = 0.0780371764464410f; 7th-order approximation: atan(x) = a1*x + a2*x^3 + a3*x^5 + a3*x^7 = x(a1 + x^2*(a2 + x^2*(a3 + a4*x^2))); a1 = 0.9991334482227801, a2 = -0.3205332923816640, a3 = 0.1449824901444650, a4 = -0.0382544649702990; 7th-order approximation in use (the most accurate solution) */ { x <<= ATI_SF; FIXP_DBL x2 = fPow2(x); temp = fMultAddDiv2((FL2FXCONST_DBL(0.1449824901444650f) >> 1), x2, FL2FXCONST_DBL(-0.0382544649702990)); temp = fMultAddDiv2((FL2FXCONST_DBL(-0.3205332923816640f) >> 2), x2, temp); temp = fMultAddDiv2((FL2FXCONST_DBL(0.9991334482227801f) >> 3), x2, temp); result = fMult(x, (temp << 2)); } else if (x < FL2FXCONST_DBL(1.28 / 64.0)) { FIXP_DBL delta_fix; FIXP_DBL PI_BY_4 = FL2FXCONST_DBL(3.1415926 / 4.0) >> 1; /* pi/4 in q30 */ delta_fix = (x - FL2FXCONST_DBL(1.0 / 64.0)) << 5; /* q30 */ result = PI_BY_4 + (delta_fix >> 1) - (fPow2Div2(delta_fix)); } else { /* Other approximation for |x| > 1.28 */ INT res_e; temp = fPow2Div2(x); /* q25 * q25 - (DFRACT_BITS-1) - 1 = q18 */ temp = temp + P281; /* q18 + q18 = q18 */ result = fDivNorm(x, temp, &res_e); result = scaleValue(result, (Q_ATANOUT - Q_ATANINP + 18 - DFRACT_BITS + 1) + res_e); result = ONEP571 - result; /* q30 + q30 = q30 */ } if (sign) { result = -result; } return (result); } #include "FDK_tools_rom.h" FIXP_DBL fixp_cos(FIXP_DBL x, int scale) { FIXP_DBL residual, error, sine, cosine; residual = fixp_sin_cos_residual_inline(x, scale, &sine, &cosine); error = fMult(sine, residual); #ifdef SINETABLE_16BIT return cosine - error; #else /* Undo downscaling by 1 which was done at fixp_sin_cos_residual_inline */ return SATURATE_LEFT_SHIFT(cosine - error, 1, DFRACT_BITS); #endif } FIXP_DBL fixp_sin(FIXP_DBL x, int scale) { FIXP_DBL residual, error, sine, cosine; residual = fixp_sin_cos_residual_inline(x, scale, &sine, &cosine); error = fMult(cosine, residual); #ifdef SINETABLE_16BIT return sine + error; #else return SATURATE_LEFT_SHIFT(sine + error, 1, DFRACT_BITS); #endif } void fixp_cos_sin(FIXP_DBL x, int scale, FIXP_DBL *cos, FIXP_DBL *sin) { FIXP_DBL residual, error0, error1, sine, cosine; residual = fixp_sin_cos_residual_inline(x, scale, &sine, &cosine); error0 = fMult(sine, residual); error1 = fMult(cosine, residual); #ifdef SINETABLE_16BIT *cos = cosine - error0; *sin = sine + error1; #else *cos = SATURATE_LEFT_SHIFT(cosine - error0, 1, DFRACT_BITS); *sin = SATURATE_LEFT_SHIFT(sine + error1, 1, DFRACT_BITS); #endif } fdk-aac-2.0.2/libFDK/src/arm/000077500000000000000000000000001404225310700154405ustar00rootroot00000000000000fdk-aac-2.0.2/libFDK/src/arm/fft_rad2_arm.cpp000066400000000000000000000232451404225310700205000ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Description: dit_fft ARM assembler replacements. *******************************************************************************/ #ifndef __FFT_RAD2_CPP__ #error \ "Do not compile this file separately. It is included on demand from fft_rad2.cpp" #endif #ifndef FUNCTION_dit_fft #if defined(SINETABLE_16BIT) #define FUNCTION_dit_fft #if defined(FUNCTION_dit_fft) void dit_fft(FIXP_DBL *x, const INT ldn, const FIXP_STP *trigdata, const INT trigDataSize) { const INT n = 1 << ldn; INT i; scramble(x, n); /* * 1+2 stage radix 4 */ for (i = 0; i < n * 2; i += 8) { FIXP_DBL a00, a10, a20, a30; a00 = (x[i + 0] + x[i + 2]) >> 1; /* Re A + Re B */ a10 = (x[i + 4] + x[i + 6]) >> 1; /* Re C + Re D */ a20 = (x[i + 1] + x[i + 3]) >> 1; /* Im A + Im B */ a30 = (x[i + 5] + x[i + 7]) >> 1; /* Im C + Im D */ x[i + 0] = a00 + a10; /* Re A' = Re A + Re B + Re C + Re D */ x[i + 4] = a00 - a10; /* Re C' = Re A + Re B - Re C - Re D */ x[i + 1] = a20 + a30; /* Im A' = Im A + Im B + Im C + Im D */ x[i + 5] = a20 - a30; /* Im C' = Im A + Im B - Im C - Im D */ a00 = a00 - x[i + 2]; /* Re A - Re B */ a10 = a10 - x[i + 6]; /* Re C - Re D */ a20 = a20 - x[i + 3]; /* Im A - Im B */ a30 = a30 - x[i + 7]; /* Im C - Im D */ x[i + 2] = a00 + a30; /* Re B' = Re A - Re B + Im C - Im D */ x[i + 6] = a00 - a30; /* Re D' = Re A - Re B - Im C + Im D */ x[i + 3] = a20 - a10; /* Im B' = Im A - Im B - Re C + Re D */ x[i + 7] = a20 + a10; /* Im D' = Im A - Im B + Re C - Re D */ } INT mh = 1 << 1; INT ldm = ldn - 2; INT trigstep = trigDataSize; do { const FIXP_STP *pTrigData = trigdata; INT j; mh <<= 1; trigstep >>= 1; FDK_ASSERT(trigstep > 0); /* Do first iteration with c=1.0 and s=0.0 separately to avoid loosing to much precision. Beware: The impact on the overal FFT precision is rather large. */ { FIXP_DBL *xt1 = x; int r = n; do { FIXP_DBL *xt2 = xt1 + (mh << 1); /* FIXP_DBL *xt1 = x+ ((r)<<1); FIXP_DBL *xt2 = xt1 + (mh<<1); */ FIXP_DBL vr, vi, ur, ui; // cplxMultDiv2(&vi, &vr, x[t2+1], x[t2], (FIXP_SGL)1.0, (FIXP_SGL)0.0); vi = xt2[1] >> 1; vr = xt2[0] >> 1; ur = xt1[0] >> 1; ui = xt1[1] >> 1; xt1[0] = ur + vr; xt1[1] = ui + vi; xt2[0] = ur - vr; xt2[1] = ui - vi; xt1 += mh; xt2 += mh; // cplxMultDiv2(&vr, &vi, x[t2+1], x[t2], (FIXP_SGL)1.0, (FIXP_SGL)0.0); vr = xt2[1] >> 1; vi = xt2[0] >> 1; ur = xt1[0] >> 1; ui = xt1[1] >> 1; xt1[0] = ur + vr; xt1[1] = ui - vi; xt2[0] = ur - vr; xt2[1] = ui + vi; xt1 = xt2 + mh; } while ((r = r - (mh << 1)) != 0); } for (j = 4; j < mh; j += 4) { FIXP_DBL *xt1 = x + (j >> 1); FIXP_SPK cs; int r = n; pTrigData += trigstep; cs = *pTrigData; do { FIXP_DBL *xt2 = xt1 + (mh << 1); FIXP_DBL vr, vi, ur, ui; cplxMultDiv2(&vi, &vr, xt2[1], xt2[0], cs); ur = xt1[0] >> 1; ui = xt1[1] >> 1; xt1[0] = ur + vr; xt1[1] = ui + vi; xt2[0] = ur - vr; xt2[1] = ui - vi; xt1 += mh; xt2 += mh; cplxMultDiv2(&vr, &vi, xt2[1], xt2[0], cs); ur = xt1[0] >> 1; ui = xt1[1] >> 1; xt1[0] = ur + vr; xt1[1] = ui - vi; xt2[0] = ur - vr; xt2[1] = ui + vi; /* Same as above but for t1,t2 with j>mh/4 and thus cs swapped */ xt1 = xt1 - (j); xt2 = xt1 + (mh << 1); cplxMultDiv2(&vi, &vr, xt2[0], xt2[1], cs); ur = xt1[0] >> 1; ui = xt1[1] >> 1; xt1[0] = ur + vr; xt1[1] = ui - vi; xt2[0] = ur - vr; xt2[1] = ui + vi; xt1 += mh; xt2 += mh; cplxMultDiv2(&vr, &vi, xt2[0], xt2[1], cs); ur = xt1[0] >> 1; ui = xt1[1] >> 1; xt1[0] = ur - vr; xt1[1] = ui - vi; xt2[0] = ur + vr; xt2[1] = ui + vi; xt1 = xt2 + (j); } while ((r = r - (mh << 1)) != 0); } { FIXP_DBL *xt1 = x + (mh >> 1); int r = n; do { FIXP_DBL *xt2 = xt1 + (mh << 1); FIXP_DBL vr, vi, ur, ui; cplxMultDiv2(&vi, &vr, xt2[1], xt2[0], STC(0x5a82799a), STC(0x5a82799a)); ur = xt1[0] >> 1; ui = xt1[1] >> 1; xt1[0] = ur + vr; xt1[1] = ui + vi; xt2[0] = ur - vr; xt2[1] = ui - vi; xt1 += mh; xt2 += mh; cplxMultDiv2(&vr, &vi, xt2[1], xt2[0], STC(0x5a82799a), STC(0x5a82799a)); ur = xt1[0] >> 1; ui = xt1[1] >> 1; xt1[0] = ur + vr; xt1[1] = ui - vi; xt2[0] = ur - vr; xt2[1] = ui + vi; xt1 = xt2 + mh; } while ((r = r - (mh << 1)) != 0); } } while (--ldm != 0); } #endif /* if defined(FUNCTION_dit_fft) */ #endif /* if defined(SINETABLE_16BIT) */ #endif /* ifndef FUNCTION_dit_fft */ fdk-aac-2.0.2/libFDK/src/arm/scale_arm.cpp000066400000000000000000000154351404225310700201020ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Arthur Tritthart Description: Scaling operations for ARM *******************************************************************************/ /* prevent multiple inclusion with re-definitions */ #ifndef __INCLUDE_SCALE_ARM__ #define __INCLUDE_SCALE_ARM__ #if !defined(FUNCTION_scaleValuesWithFactor_DBL) #define FUNCTION_scaleValuesWithFactor_DBL SCALE_INLINE void scaleValuesWithFactor(FIXP_DBL *vector, FIXP_DBL factor, INT len, INT scalefactor) { /* This code combines the fMult with the scaling */ /* It performs a fMultDiv2 and increments shift by 1 */ int shift = scalefactor + 1; FIXP_DBL *mySpec = vector; shift = fixmin_I(shift, (INT)DFRACT_BITS - 1); if (shift >= 0) { for (int i = 0; i < (len >> 2); i++) { FIXP_DBL tmp0 = mySpec[0]; FIXP_DBL tmp1 = mySpec[1]; FIXP_DBL tmp2 = mySpec[2]; FIXP_DBL tmp3 = mySpec[3]; tmp0 = fMultDiv2(tmp0, factor); tmp1 = fMultDiv2(tmp1, factor); tmp2 = fMultDiv2(tmp2, factor); tmp3 = fMultDiv2(tmp3, factor); tmp0 <<= shift; tmp1 <<= shift; tmp2 <<= shift; tmp3 <<= shift; *mySpec++ = tmp0; *mySpec++ = tmp1; *mySpec++ = tmp2; *mySpec++ = tmp3; } for (int i = len & 3; i--;) { FIXP_DBL tmp0 = mySpec[0]; tmp0 = fMultDiv2(tmp0, factor); tmp0 <<= shift; *mySpec++ = tmp0; } } else { shift = -shift; for (int i = 0; i < (len >> 2); i++) { FIXP_DBL tmp0 = mySpec[0]; FIXP_DBL tmp1 = mySpec[1]; FIXP_DBL tmp2 = mySpec[2]; FIXP_DBL tmp3 = mySpec[3]; tmp0 = fMultDiv2(tmp0, factor); tmp1 = fMultDiv2(tmp1, factor); tmp2 = fMultDiv2(tmp2, factor); tmp3 = fMultDiv2(tmp3, factor); tmp0 >>= shift; tmp1 >>= shift; tmp2 >>= shift; tmp3 >>= shift; *mySpec++ = tmp0; *mySpec++ = tmp1; *mySpec++ = tmp2; *mySpec++ = tmp3; } for (int i = len & 3; i--;) { FIXP_DBL tmp0 = mySpec[0]; tmp0 = fMultDiv2(tmp0, factor); tmp0 >>= shift; *mySpec++ = tmp0; } } } #endif /* #if !defined(FUNCTION_scaleValuesWithFactor_DBL) */ #endif /* #ifndef __INCLUDE_SCALE_ARM__ */ fdk-aac-2.0.2/libFDK/src/autocorr2nd.cpp000066400000000000000000000247001404225310700176320ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): M. Lohwasser Description: auto-correlation functions *******************************************************************************/ #include "autocorr2nd.h" /* If the accumulator does not provide enough overflow bits, products have to be shifted down in the autocorrelation below. */ #define SHIFT_FACTOR (5) #define SHIFT >> (SHIFT_FACTOR) /*! * * \brief Calculate second order autocorrelation using 2 accumulators * */ #if !defined(FUNCTION_autoCorr2nd_real) INT autoCorr2nd_real( ACORR_COEFS *ac, /*!< Pointer to autocorrelation coeffs */ const FIXP_DBL *reBuffer, /*!< Pointer to to real part of input samples */ const int len /*!< Number input samples */ ) { int j, autoCorrScaling, mScale; FIXP_DBL accu1, accu2, accu3, accu4, accu5; const FIXP_DBL *pReBuf; const FIXP_DBL *realBuf = reBuffer; /* r11r,r22r r01r,r12r r02r */ pReBuf = realBuf - 2; accu5 = ((fMultDiv2(pReBuf[0], pReBuf[2]) + fMultDiv2(pReBuf[1], pReBuf[3])) SHIFT); pReBuf++; /* len must be even */ accu1 = fPow2Div2(pReBuf[0]) SHIFT; accu3 = fMultDiv2(pReBuf[0], pReBuf[1]) SHIFT; pReBuf++; for (j = (len - 2) >> 1; j != 0; j--, pReBuf += 2) { accu1 += ((fPow2Div2(pReBuf[0]) + fPow2Div2(pReBuf[1])) SHIFT); accu3 += ((fMultDiv2(pReBuf[0], pReBuf[1]) + fMultDiv2(pReBuf[1], pReBuf[2])) SHIFT); accu5 += ((fMultDiv2(pReBuf[0], pReBuf[2]) + fMultDiv2(pReBuf[1], pReBuf[3])) SHIFT); } accu2 = (fPow2Div2(realBuf[-2]) SHIFT); accu2 += accu1; accu1 += (fPow2Div2(realBuf[len - 2]) SHIFT); accu4 = (fMultDiv2(realBuf[-1], realBuf[-2]) SHIFT); accu4 += accu3; accu3 += (fMultDiv2(realBuf[len - 1], realBuf[len - 2]) SHIFT); mScale = CntLeadingZeros( (accu1 | accu2 | fAbs(accu3) | fAbs(accu4) | fAbs(accu5))) - 1; autoCorrScaling = mScale - 1 - SHIFT_FACTOR; /* -1 because of fMultDiv2*/ /* Scale to common scale factor */ ac->r11r = accu1 << mScale; ac->r22r = accu2 << mScale; ac->r01r = accu3 << mScale; ac->r12r = accu4 << mScale; ac->r02r = accu5 << mScale; ac->det = (fMultDiv2(ac->r11r, ac->r22r) - fMultDiv2(ac->r12r, ac->r12r)); mScale = CountLeadingBits(fAbs(ac->det)); ac->det <<= mScale; ac->det_scale = mScale - 1; return autoCorrScaling; } #endif #if !defined(FUNCTION_autoCorr2nd_cplx) INT autoCorr2nd_cplx( ACORR_COEFS *ac, /*!< Pointer to autocorrelation coeffs */ const FIXP_DBL *reBuffer, /*!< Pointer to real part of input samples */ const FIXP_DBL *imBuffer, /*!< Pointer to imag part of input samples */ const int len /*!< Number of input samples (should be smaller than 128) */ ) { int j, autoCorrScaling, mScale, len_scale; FIXP_DBL accu0, accu1, accu2, accu3, accu4, accu5, accu6, accu7, accu8; const FIXP_DBL *pReBuf, *pImBuf; const FIXP_DBL *realBuf = reBuffer; const FIXP_DBL *imagBuf = imBuffer; (len > 64) ? (len_scale = 6) : (len_scale = 5); /* r00r, r11r,r22r r01r,r12r r01i,r12i r02r,r02i */ accu1 = accu3 = accu5 = accu7 = accu8 = FL2FXCONST_DBL(0.0f); pReBuf = realBuf - 2, pImBuf = imagBuf - 2; accu7 += ((fMultDiv2(pReBuf[2], pReBuf[0]) + fMultDiv2(pImBuf[2], pImBuf[0])) >> len_scale); accu8 += ((fMultDiv2(pImBuf[2], pReBuf[0]) - fMultDiv2(pReBuf[2], pImBuf[0])) >> len_scale); pReBuf = realBuf - 1, pImBuf = imagBuf - 1; for (j = (len - 1); j != 0; j--, pReBuf++, pImBuf++) { accu1 += ((fPow2Div2(pReBuf[0]) + fPow2Div2(pImBuf[0])) >> len_scale); accu3 += ((fMultDiv2(pReBuf[0], pReBuf[1]) + fMultDiv2(pImBuf[0], pImBuf[1])) >> len_scale); accu5 += ((fMultDiv2(pImBuf[1], pReBuf[0]) - fMultDiv2(pReBuf[1], pImBuf[0])) >> len_scale); accu7 += ((fMultDiv2(pReBuf[2], pReBuf[0]) + fMultDiv2(pImBuf[2], pImBuf[0])) >> len_scale); accu8 += ((fMultDiv2(pImBuf[2], pReBuf[0]) - fMultDiv2(pReBuf[2], pImBuf[0])) >> len_scale); } accu2 = ((fPow2Div2(realBuf[-2]) + fPow2Div2(imagBuf[-2])) >> len_scale); accu2 += accu1; accu1 += ((fPow2Div2(realBuf[len - 2]) + fPow2Div2(imagBuf[len - 2])) >> len_scale); accu0 = ((fPow2Div2(realBuf[len - 1]) + fPow2Div2(imagBuf[len - 1])) >> len_scale) - ((fPow2Div2(realBuf[-1]) + fPow2Div2(imagBuf[-1])) >> len_scale); accu0 += accu1; accu4 = ((fMultDiv2(realBuf[-1], realBuf[-2]) + fMultDiv2(imagBuf[-1], imagBuf[-2])) >> len_scale); accu4 += accu3; accu3 += ((fMultDiv2(realBuf[len - 1], realBuf[len - 2]) + fMultDiv2(imagBuf[len - 1], imagBuf[len - 2])) >> len_scale); accu6 = ((fMultDiv2(imagBuf[-1], realBuf[-2]) - fMultDiv2(realBuf[-1], imagBuf[-2])) >> len_scale); accu6 += accu5; accu5 += ((fMultDiv2(imagBuf[len - 1], realBuf[len - 2]) - fMultDiv2(realBuf[len - 1], imagBuf[len - 2])) >> len_scale); mScale = CntLeadingZeros((accu0 | accu1 | accu2 | fAbs(accu3) | fAbs(accu4) | fAbs(accu5) | fAbs(accu6) | fAbs(accu7) | fAbs(accu8))) - 1; autoCorrScaling = mScale - 1 - len_scale; /* -1 because of fMultDiv2*/ /* Scale to common scale factor */ ac->r00r = (FIXP_DBL)accu0 << mScale; ac->r11r = (FIXP_DBL)accu1 << mScale; ac->r22r = (FIXP_DBL)accu2 << mScale; ac->r01r = (FIXP_DBL)accu3 << mScale; ac->r12r = (FIXP_DBL)accu4 << mScale; ac->r01i = (FIXP_DBL)accu5 << mScale; ac->r12i = (FIXP_DBL)accu6 << mScale; ac->r02r = (FIXP_DBL)accu7 << mScale; ac->r02i = (FIXP_DBL)accu8 << mScale; ac->det = (fMultDiv2(ac->r11r, ac->r22r) >> 1) - ((fMultDiv2(ac->r12r, ac->r12r) + fMultDiv2(ac->r12i, ac->r12i)) >> 1); mScale = CntLeadingZeros(fAbs(ac->det)) - 1; ac->det <<= mScale; ac->det_scale = mScale - 2; return autoCorrScaling; } #endif /* FUNCTION_autoCorr2nd_cplx */ fdk-aac-2.0.2/libFDK/src/dct.cpp000066400000000000000000000407121404225310700161430ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Description: *******************************************************************************/ /*! \file dct.cpp \brief DCT Implementations Library functions to calculate standard DCTs. This will most likely be replaced by hand-optimized functions for the specific target processor. Three different implementations of the dct type II and the dct type III transforms are provided. By default implementations which are based on a single, standard complex FFT-kernel are used (dctII_f() and dctIII_f()). These are specifically helpful in cases where optimized FFT libraries are already available. The FFT used in these implementation is FFT rad2 from FDK_tools. Of course, one might also use DCT-libraries should they be available. The DCT and DST type IV implementations are only available in a version based on a complex FFT kernel. */ #include "dct.h" #include "FDK_tools_rom.h" #include "fft.h" void dct_getTables(const FIXP_WTP **ptwiddle, const FIXP_STP **sin_twiddle, int *sin_step, int length) { const FIXP_WTP *twiddle; int ld2_length; /* Get ld2 of length - 2 + 1 -2: because first table entry is window of size 4 +1: because we already include +1 because of ceil(log2(length)) */ ld2_length = DFRACT_BITS - 1 - fNormz((FIXP_DBL)length) - 1; /* Extract sort of "eigenvalue" (the 4 left most bits) of length. */ switch ((length) >> (ld2_length - 1)) { case 0x4: /* radix 2 */ *sin_twiddle = SineTable1024; *sin_step = 1 << (10 - ld2_length); twiddle = windowSlopes[0][0][ld2_length - 1]; break; case 0x7: /* 10 ms */ *sin_twiddle = SineTable480; *sin_step = 1 << (8 - ld2_length); twiddle = windowSlopes[0][1][ld2_length]; break; case 0x6: /* 3/4 of radix 2 */ *sin_twiddle = SineTable384; *sin_step = 1 << (8 - ld2_length); twiddle = windowSlopes[0][2][ld2_length]; break; case 0x5: /* 5/16 of radix 2*/ *sin_twiddle = SineTable80; *sin_step = 1 << (6 - ld2_length); twiddle = windowSlopes[0][3][ld2_length]; break; default: *sin_twiddle = NULL; *sin_step = 0; twiddle = NULL; break; } if (ptwiddle != NULL) { FDK_ASSERT(twiddle != NULL); *ptwiddle = twiddle; } FDK_ASSERT(*sin_step > 0); } #if !defined(FUNCTION_dct_III) void dct_III(FIXP_DBL *pDat, /*!< pointer to input/output */ FIXP_DBL *tmp, /*!< pointer to temporal working buffer */ int L, /*!< lenght of transform */ int *pDat_e) { const FIXP_WTP *sin_twiddle; int i; FIXP_DBL xr, accu1, accu2; int inc, index; int M = L >> 1; FDK_ASSERT(L % 4 == 0); dct_getTables(NULL, &sin_twiddle, &inc, L); inc >>= 1; FIXP_DBL *pTmp_0 = &tmp[2]; FIXP_DBL *pTmp_1 = &tmp[(M - 1) * 2]; index = 4 * inc; /* This loop performs multiplication for index i (i*inc) */ for (i = 1; i> 1; i++, pTmp_0 += 2, pTmp_1 -= 2) { FIXP_DBL accu3, accu4, accu5, accu6; cplxMultDiv2(&accu2, &accu1, pDat[L - i], pDat[i], sin_twiddle[i * inc]); cplxMultDiv2(&accu4, &accu3, pDat[M + i], pDat[M - i], sin_twiddle[(M - i) * inc]); accu3 >>= 1; accu4 >>= 1; /* This method is better for ARM926, that uses operand2 shifted right by 1 * always */ if (2 * i < (M / 2)) { cplxMultDiv2(&accu6, &accu5, (accu3 - (accu1 >> 1)), ((accu2 >> 1) + accu4), sin_twiddle[index]); } else { cplxMultDiv2(&accu6, &accu5, ((accu2 >> 1) + accu4), (accu3 - (accu1 >> 1)), sin_twiddle[index]); accu6 = -accu6; } xr = (accu1 >> 1) + accu3; pTmp_0[0] = (xr >> 1) - accu5; pTmp_1[0] = (xr >> 1) + accu5; xr = (accu2 >> 1) - accu4; pTmp_0[1] = (xr >> 1) - accu6; pTmp_1[1] = -((xr >> 1) + accu6); /* Create index helper variables for (4*i)*inc indexed equivalent values of * short tables. */ if (2 * i < ((M / 2) - 1)) { index += 4 * inc; } else if (2 * i >= ((M / 2))) { index -= 4 * inc; } } xr = fMultDiv2(pDat[M], sin_twiddle[M * inc].v.re); /* cos((PI/(2*L))*M); */ tmp[0] = ((pDat[0] >> 1) + xr) >> 1; tmp[1] = ((pDat[0] >> 1) - xr) >> 1; cplxMultDiv2(&accu2, &accu1, pDat[L - (M / 2)], pDat[M / 2], sin_twiddle[M * inc / 2]); tmp[M] = accu1 >> 1; tmp[M + 1] = accu2 >> 1; /* dit_fft expects 1 bit scaled input values */ fft(M, tmp, pDat_e); /* ARM926: 12 cycles per 2-iteration, no overhead code by compiler */ pTmp_1 = &tmp[L]; for (i = M >> 1; i--;) { FIXP_DBL tmp1, tmp2, tmp3, tmp4; tmp1 = *tmp++; tmp2 = *tmp++; tmp3 = *--pTmp_1; tmp4 = *--pTmp_1; *pDat++ = tmp1; *pDat++ = tmp3; *pDat++ = tmp2; *pDat++ = tmp4; } *pDat_e += 2; } void dst_III(FIXP_DBL *pDat, /*!< pointer to input/output */ FIXP_DBL *tmp, /*!< pointer to temporal working buffer */ int L, /*!< lenght of transform */ int *pDat_e) { int L2 = L >> 1; int i; FIXP_DBL t; /* note: DCT III is reused here, direct DST III implementation might be more * efficient */ /* mirror input */ for (i = 0; i < L2; i++) { t = pDat[i]; pDat[i] = pDat[L - 1 - i]; pDat[L - 1 - i] = t; } /* DCT-III */ dct_III(pDat, tmp, L, pDat_e); /* flip signs at odd indices */ for (i = 1; i < L; i += 2) pDat[i] = -pDat[i]; } #endif #if !defined(FUNCTION_dct_II) void dct_II( FIXP_DBL *pDat, /*!< pointer to input/output */ FIXP_DBL *tmp, /*!< pointer to temporal working buffer */ int L, /*!< lenght of transform (has to be a multiple of 8 (or 4 in case DCT_II_L_MULTIPLE_OF_4_SUPPORT is defined) */ int *pDat_e) { const FIXP_WTP *sin_twiddle; FIXP_DBL accu1, accu2; FIXP_DBL *pTmp_0, *pTmp_1; int i; int inc, index = 0; int M = L >> 1; FDK_ASSERT(L % 4 == 0); dct_getTables(NULL, &sin_twiddle, &inc, L); inc >>= 1; { for (i = 0; i < M; i++) { tmp[i] = pDat[2 * i] >> 1; /* dit_fft expects 1 bit scaled input values */ tmp[L - 1 - i] = pDat[2 * i + 1] >> 1; /* dit_fft expects 1 bit scaled input values */ } } fft(M, tmp, pDat_e); pTmp_0 = &tmp[2]; pTmp_1 = &tmp[(M - 1) * 2]; index = inc * 4; for (i = 1; i> 1; i++, pTmp_0 += 2, pTmp_1 -= 2) { FIXP_DBL a1, a2; FIXP_DBL accu3, accu4; a1 = ((pTmp_0[1] >> 1) + (pTmp_1[1] >> 1)); a2 = ((pTmp_1[0] >> 1) - (pTmp_0[0] >> 1)); if (2 * i < (M / 2)) { cplxMultDiv2(&accu1, &accu2, a2, a1, sin_twiddle[index]); } else { cplxMultDiv2(&accu1, &accu2, a1, a2, sin_twiddle[index]); accu1 = -accu1; } accu1 <<= 1; accu2 <<= 1; a1 = ((pTmp_0[0] >> 1) + (pTmp_1[0] >> 1)); a2 = ((pTmp_0[1] >> 1) - (pTmp_1[1] >> 1)); cplxMultDiv2(&accu3, &accu4, (a1 + accu2), -(accu1 + a2), sin_twiddle[i * inc]); pDat[L - i] = accu4; pDat[i] = accu3; cplxMultDiv2(&accu3, &accu4, (a1 - accu2), -(accu1 - a2), sin_twiddle[(M - i) * inc]); pDat[M + i] = accu4; pDat[M - i] = accu3; /* Create index helper variables for (4*i)*inc indexed equivalent values of * short tables. */ if (2 * i < ((M / 2) - 1)) { index += 4 * inc; } else if (2 * i >= ((M / 2))) { index -= 4 * inc; } } cplxMultDiv2(&accu1, &accu2, tmp[M], tmp[M + 1], sin_twiddle[(M / 2) * inc]); pDat[L - (M / 2)] = accu2; pDat[M / 2] = accu1; pDat[0] = (tmp[0] >> 1) + (tmp[1] >> 1); pDat[M] = fMult(((tmp[0] >> 1) - (tmp[1] >> 1)), sin_twiddle[M * inc].v.re); /* cos((PI/(2*L))*M); */ *pDat_e += 2; } #endif #if !defined(FUNCTION_dct_IV) void dct_IV(FIXP_DBL *pDat, int L, int *pDat_e) { int sin_step = 0; int M = L >> 1; const FIXP_WTP *twiddle; const FIXP_STP *sin_twiddle; FDK_ASSERT(L >= 4); FDK_ASSERT(L >= 4); dct_getTables(&twiddle, &sin_twiddle, &sin_step, L); { FIXP_DBL *RESTRICT pDat_0 = &pDat[0]; FIXP_DBL *RESTRICT pDat_1 = &pDat[L - 2]; int i; /* 29 cycles on ARM926 */ for (i = 0; i < M - 1; i += 2, pDat_0 += 2, pDat_1 -= 2) { FIXP_DBL accu1, accu2, accu3, accu4; accu1 = pDat_1[1]; accu2 = pDat_0[0]; accu3 = pDat_0[1]; accu4 = pDat_1[0]; cplxMultDiv2(&accu1, &accu2, accu1, accu2, twiddle[i]); cplxMultDiv2(&accu3, &accu4, accu4, accu3, twiddle[i + 1]); pDat_0[0] = accu2 >> 1; pDat_0[1] = accu1 >> 1; pDat_1[0] = accu4 >> 1; pDat_1[1] = -(accu3 >> 1); } if (M & 1) { FIXP_DBL accu1, accu2; accu1 = pDat_1[1]; accu2 = pDat_0[0]; cplxMultDiv2(&accu1, &accu2, accu1, accu2, twiddle[i]); pDat_0[0] = accu2 >> 1; pDat_0[1] = accu1 >> 1; } } fft(M, pDat, pDat_e); { FIXP_DBL *RESTRICT pDat_0 = &pDat[0]; FIXP_DBL *RESTRICT pDat_1 = &pDat[L - 2]; FIXP_DBL accu1, accu2, accu3, accu4; int idx, i; /* Sin and Cos values are 0.0f and 1.0f */ accu1 = pDat_1[0]; accu2 = pDat_1[1]; pDat_1[1] = -pDat_0[1]; /* 28 cycles for ARM926 */ for (idx = sin_step, i = 1; i<(M + 1)>> 1; i++, idx += sin_step) { FIXP_STP twd = sin_twiddle[idx]; cplxMult(&accu3, &accu4, accu1, accu2, twd); pDat_0[1] = accu3; pDat_1[0] = accu4; pDat_0 += 2; pDat_1 -= 2; cplxMult(&accu3, &accu4, pDat_0[1], pDat_0[0], twd); accu1 = pDat_1[0]; accu2 = pDat_1[1]; pDat_1[1] = -accu3; pDat_0[0] = accu4; } if ((M & 1) == 0) { /* Last Sin and Cos value pair are the same */ accu1 = fMult(accu1, WTC(0x5a82799a)); accu2 = fMult(accu2, WTC(0x5a82799a)); pDat_1[0] = accu1 + accu2; pDat_0[1] = accu1 - accu2; } } /* Add twiddeling scale. */ *pDat_e += 2; } #endif /* defined (FUNCTION_dct_IV) */ #if !defined(FUNCTION_dst_IV) void dst_IV(FIXP_DBL *pDat, int L, int *pDat_e) { int sin_step = 0; int M = L >> 1; const FIXP_WTP *twiddle; const FIXP_STP *sin_twiddle; FDK_ASSERT(L >= 4); FDK_ASSERT(L >= 4); dct_getTables(&twiddle, &sin_twiddle, &sin_step, L); { FIXP_DBL *RESTRICT pDat_0 = &pDat[0]; FIXP_DBL *RESTRICT pDat_1 = &pDat[L - 2]; int i; /* 34 cycles on ARM926 */ for (i = 0; i < M - 1; i += 2, pDat_0 += 2, pDat_1 -= 2) { FIXP_DBL accu1, accu2, accu3, accu4; accu1 = pDat_1[1] >> 1; accu2 = -(pDat_0[0] >> 1); accu3 = pDat_0[1] >> 1; accu4 = -(pDat_1[0] >> 1); cplxMultDiv2(&accu1, &accu2, accu1, accu2, twiddle[i]); cplxMultDiv2(&accu3, &accu4, accu4, accu3, twiddle[i + 1]); pDat_0[0] = accu2; pDat_0[1] = accu1; pDat_1[0] = accu4; pDat_1[1] = -accu3; } if (M & 1) { FIXP_DBL accu1, accu2; accu1 = pDat_1[1]; accu2 = -pDat_0[0]; cplxMultDiv2(&accu1, &accu2, accu1, accu2, twiddle[i]); pDat_0[0] = accu2 >> 1; pDat_0[1] = accu1 >> 1; } } fft(M, pDat, pDat_e); { FIXP_DBL *RESTRICT pDat_0; FIXP_DBL *RESTRICT pDat_1; FIXP_DBL accu1, accu2, accu3, accu4; int idx, i; pDat_0 = &pDat[0]; pDat_1 = &pDat[L - 2]; /* Sin and Cos values are 0.0f and 1.0f */ accu1 = pDat_1[0]; accu2 = pDat_1[1]; pDat_1[1] = -pDat_0[0]; pDat_0[0] = pDat_0[1]; for (idx = sin_step, i = 1; i<(M + 1)>> 1; i++, idx += sin_step) { FIXP_STP twd = sin_twiddle[idx]; cplxMult(&accu3, &accu4, accu1, accu2, twd); pDat_1[0] = -accu3; pDat_0[1] = -accu4; pDat_0 += 2; pDat_1 -= 2; cplxMult(&accu3, &accu4, pDat_0[1], pDat_0[0], twd); accu1 = pDat_1[0]; accu2 = pDat_1[1]; pDat_0[0] = accu3; pDat_1[1] = -accu4; } if ((M & 1) == 0) { /* Last Sin and Cos value pair are the same */ accu1 = fMult(accu1, WTC(0x5a82799a)); accu2 = fMult(accu2, WTC(0x5a82799a)); pDat_0[1] = -accu1 - accu2; pDat_1[0] = accu2 - accu1; } } /* Add twiddeling scale. */ *pDat_e += 2; } #endif /* !defined(FUNCTION_dst_IV) */ fdk-aac-2.0.2/libFDK/src/fft.cpp000066400000000000000000001603771404225310700161620ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Josef Hoepfl, DSP Solutions Description: Fix point FFT *******************************************************************************/ #include "fft_rad2.h" #include "FDK_tools_rom.h" #define W_PiFOURTH STC(0x5a82799a) //#define W_PiFOURTH ((FIXP_DBL)(0x5a82799a)) #ifndef SUMDIFF_PIFOURTH #define SUMDIFF_PIFOURTH(diff, sum, a, b) \ { \ FIXP_DBL wa, wb; \ wa = fMultDiv2(a, W_PiFOURTH); \ wb = fMultDiv2(b, W_PiFOURTH); \ diff = wb - wa; \ sum = wb + wa; \ } #define SUMDIFF_PIFOURTH16(diff, sum, a, b) \ { \ FIXP_SGL wa, wb; \ wa = FX_DBL2FX_SGL(fMultDiv2(a, W_PiFOURTH)); \ wb = FX_DBL2FX_SGL(fMultDiv2(b, W_PiFOURTH)); \ diff = wb - wa; \ sum = wb + wa; \ } #endif #define SCALEFACTOR2048 10 #define SCALEFACTOR1024 9 #define SCALEFACTOR512 8 #define SCALEFACTOR256 7 #define SCALEFACTOR128 6 #define SCALEFACTOR64 5 #define SCALEFACTOR32 4 #define SCALEFACTOR16 3 #define SCALEFACTOR8 2 #define SCALEFACTOR4 1 #define SCALEFACTOR2 1 #define SCALEFACTOR3 1 #define SCALEFACTOR5 1 #define SCALEFACTOR6 (SCALEFACTOR2 + SCALEFACTOR3 + 2) #define SCALEFACTOR7 2 #define SCALEFACTOR9 2 #define SCALEFACTOR10 5 #define SCALEFACTOR12 3 #define SCALEFACTOR15 3 #define SCALEFACTOR18 (SCALEFACTOR2 + SCALEFACTOR9 + 2) #define SCALEFACTOR20 (SCALEFACTOR4 + SCALEFACTOR5 + 2) #define SCALEFACTOR21 (SCALEFACTOR3 + SCALEFACTOR7 + 2) #define SCALEFACTOR24 (SCALEFACTOR2 + SCALEFACTOR12 + 2) #define SCALEFACTOR30 (SCALEFACTOR2 + SCALEFACTOR15 + 2) #define SCALEFACTOR40 (SCALEFACTOR5 + SCALEFACTOR8 + 2) #define SCALEFACTOR48 (SCALEFACTOR4 + SCALEFACTOR12 + 2) #define SCALEFACTOR60 (SCALEFACTOR4 + SCALEFACTOR15 + 2) #define SCALEFACTOR80 (SCALEFACTOR5 + SCALEFACTOR16 + 2) #define SCALEFACTOR96 (SCALEFACTOR3 + SCALEFACTOR32 + 2) #define SCALEFACTOR120 (SCALEFACTOR8 + SCALEFACTOR15 + 2) #define SCALEFACTOR160 (SCALEFACTOR10 + SCALEFACTOR16 + 2) #define SCALEFACTOR168 (SCALEFACTOR21 + SCALEFACTOR8 + 2) #define SCALEFACTOR192 (SCALEFACTOR12 + SCALEFACTOR16 + 2) #define SCALEFACTOR240 (SCALEFACTOR16 + SCALEFACTOR15 + 2) #define SCALEFACTOR320 (SCALEFACTOR10 + SCALEFACTOR32 + 2) #define SCALEFACTOR336 (SCALEFACTOR21 + SCALEFACTOR16 + 2) #define SCALEFACTOR384 (SCALEFACTOR12 + SCALEFACTOR32 + 2) #define SCALEFACTOR480 (SCALEFACTOR32 + SCALEFACTOR15 + 2) #include "fft.h" #ifndef FUNCTION_fft2 /* Performs the FFT of length 2. Input vector unscaled, output vector scaled * with factor 0.5 */ static FDK_FORCEINLINE void fft2(FIXP_DBL *RESTRICT pDat) { FIXP_DBL r1, i1; FIXP_DBL r2, i2; /* real part */ r1 = pDat[2]; r2 = pDat[0]; /* imaginary part */ i1 = pDat[3]; i2 = pDat[1]; /* real part */ pDat[0] = (r2 + r1) >> 1; pDat[2] = (r2 - r1) >> 1; /* imaginary part */ pDat[1] = (i2 + i1) >> 1; pDat[3] = (i2 - i1) >> 1; } #endif /* FUNCTION_fft2 */ #define C31 (STC(0x91261468)) /* FL2FXCONST_DBL(-0.86602540) = -sqrt(3)/2 */ #ifndef FUNCTION_fft3 /* Performs the FFT of length 3 according to the algorithm after winograd. */ static FDK_FORCEINLINE void fft3(FIXP_DBL *RESTRICT pDat) { FIXP_DBL r1, r2; FIXP_DBL s1, s2; FIXP_DBL pD; /* real part */ r1 = pDat[2] + pDat[4]; r2 = fMultDiv2((pDat[2] - pDat[4]), C31); pD = pDat[0] >> 1; pDat[0] = pD + (r1 >> 1); r1 = pD - (r1 >> 2); /* imaginary part */ s1 = pDat[3] + pDat[5]; s2 = fMultDiv2((pDat[3] - pDat[5]), C31); pD = pDat[1] >> 1; pDat[1] = pD + (s1 >> 1); s1 = pD - (s1 >> 2); /* combination */ pDat[2] = r1 - s2; pDat[4] = r1 + s2; pDat[3] = s1 + r2; pDat[5] = s1 - r2; } #endif /* #ifndef FUNCTION_fft3 */ #define F5C(x) STC(x) #define C51 (F5C(0x79bc3854)) /* FL2FXCONST_DBL( 0.95105652) */ #define C52 (F5C(0x9d839db0)) /* FL2FXCONST_DBL(-1.53884180/2) */ #define C53 (F5C(0xd18053ce)) /* FL2FXCONST_DBL(-0.36327126) */ #define C54 (F5C(0x478dde64)) /* FL2FXCONST_DBL( 0.55901699) */ #define C55 (F5C(0xb0000001)) /* FL2FXCONST_DBL(-1.25/2) */ /* performs the FFT of length 5 according to the algorithm after winograd */ /* This version works with a prescale of 2 instead of 3 */ static FDK_FORCEINLINE void fft5(FIXP_DBL *RESTRICT pDat) { FIXP_DBL r1, r2, r3, r4; FIXP_DBL s1, s2, s3, s4; FIXP_DBL t; /* real part */ r1 = (pDat[2] + pDat[8]) >> 1; r4 = (pDat[2] - pDat[8]) >> 1; r3 = (pDat[4] + pDat[6]) >> 1; r2 = (pDat[4] - pDat[6]) >> 1; t = fMult((r1 - r3), C54); r1 = r1 + r3; pDat[0] = (pDat[0] >> 1) + r1; /* Bit shift left because of the constant C55 which was scaled with the factor 0.5 because of the representation of the values as fracts */ r1 = pDat[0] + (fMultDiv2(r1, C55) << (2)); r3 = r1 - t; r1 = r1 + t; t = fMult((r4 + r2), C51); /* Bit shift left because of the constant C55 which was scaled with the factor 0.5 because of the representation of the values as fracts */ r4 = t + (fMultDiv2(r4, C52) << (2)); r2 = t + fMult(r2, C53); /* imaginary part */ s1 = (pDat[3] + pDat[9]) >> 1; s4 = (pDat[3] - pDat[9]) >> 1; s3 = (pDat[5] + pDat[7]) >> 1; s2 = (pDat[5] - pDat[7]) >> 1; t = fMult((s1 - s3), C54); s1 = s1 + s3; pDat[1] = (pDat[1] >> 1) + s1; /* Bit shift left because of the constant C55 which was scaled with the factor 0.5 because of the representation of the values as fracts */ s1 = pDat[1] + (fMultDiv2(s1, C55) << (2)); s3 = s1 - t; s1 = s1 + t; t = fMult((s4 + s2), C51); /* Bit shift left because of the constant C55 which was scaled with the factor 0.5 because of the representation of the values as fracts */ s4 = t + (fMultDiv2(s4, C52) << (2)); s2 = t + fMult(s2, C53); /* combination */ pDat[2] = r1 + s2; pDat[8] = r1 - s2; pDat[4] = r3 - s4; pDat[6] = r3 + s4; pDat[3] = s1 - r2; pDat[9] = s1 + r2; pDat[5] = s3 + r4; pDat[7] = s3 - r4; } #define F5C(x) STC(x) #define C51 (F5C(0x79bc3854)) /* FL2FXCONST_DBL( 0.95105652) */ #define C52 (F5C(0x9d839db0)) /* FL2FXCONST_DBL(-1.53884180/2) */ #define C53 (F5C(0xd18053ce)) /* FL2FXCONST_DBL(-0.36327126) */ #define C54 (F5C(0x478dde64)) /* FL2FXCONST_DBL( 0.55901699) */ #define C55 (F5C(0xb0000001)) /* FL2FXCONST_DBL(-1.25/2) */ /** * \brief Function performs a complex 10-point FFT * The FFT is performed inplace. The result of the FFT * is scaled by SCALEFACTOR10 bits. * * WOPS FLC version: 1093 cycles * WOPS with 32x16 bit multiplications: 196 cycles * * \param [i/o] re real input / output * \param [i/o] im imag input / output * \param [i ] s stride real and imag input / output * * \return void */ static void fft10(FIXP_DBL *x) // FIXP_DBL *re, FIXP_DBL *im, FIXP_SGL s) { FIXP_DBL t; FIXP_DBL x0, x1, x2, x3, x4; FIXP_DBL r1, r2, r3, r4; FIXP_DBL s1, s2, s3, s4; FIXP_DBL y00, y01, y02, y03, y04, y05, y06, y07, y08, y09; FIXP_DBL y10, y11, y12, y13, y14, y15, y16, y17, y18, y19; const int s = 1; // stride factor /* 2 fft5 stages */ /* real part */ x0 = (x[s * 0] >> SCALEFACTOR10); x1 = (x[s * 4] >> SCALEFACTOR10); x2 = (x[s * 8] >> SCALEFACTOR10); x3 = (x[s * 12] >> SCALEFACTOR10); x4 = (x[s * 16] >> SCALEFACTOR10); r1 = (x3 + x2); r4 = (x3 - x2); r3 = (x1 + x4); r2 = (x1 - x4); t = fMult((r1 - r3), C54); r1 = (r1 + r3); y00 = (x0 + r1); r1 = (y00 + ((fMult(r1, C55) << 1))); r3 = (r1 - t); r1 = (r1 + t); t = fMult((r4 + r2), C51); r4 = (t + (fMult(r4, C52) << 1)); r2 = (t + fMult(r2, C53)); /* imaginary part */ x0 = (x[s * 0 + 1] >> SCALEFACTOR10); x1 = (x[s * 4 + 1] >> SCALEFACTOR10); x2 = (x[s * 8 + 1] >> SCALEFACTOR10); x3 = (x[s * 12 + 1] >> SCALEFACTOR10); x4 = (x[s * 16 + 1] >> SCALEFACTOR10); s1 = (x3 + x2); s4 = (x3 - x2); s3 = (x1 + x4); s2 = (x1 - x4); t = fMult((s1 - s3), C54); s1 = (s1 + s3); y01 = (x0 + s1); s1 = (y01 + (fMult(s1, C55) << 1)); s3 = (s1 - t); s1 = (s1 + t); t = fMult((s4 + s2), C51); s4 = (t + (fMult(s4, C52) << 1)); s2 = (t + fMult(s2, C53)); /* combination */ y04 = (r1 + s2); y16 = (r1 - s2); y08 = (r3 - s4); y12 = (r3 + s4); y05 = (s1 - r2); y17 = (s1 + r2); y09 = (s3 + r4); y13 = (s3 - r4); /* real part */ x0 = (x[s * 10] >> SCALEFACTOR10); x1 = (x[s * 2] >> SCALEFACTOR10); x2 = (x[s * 6] >> SCALEFACTOR10); x3 = (x[s * 14] >> SCALEFACTOR10); x4 = (x[s * 18] >> SCALEFACTOR10); r1 = (x1 + x4); r4 = (x1 - x4); r3 = (x3 + x2); r2 = (x3 - x2); t = fMult((r1 - r3), C54); r1 = (r1 + r3); y02 = (x0 + r1); r1 = (y02 + ((fMult(r1, C55) << 1))); r3 = (r1 - t); r1 = (r1 + t); t = fMult(((r4 + r2)), C51); r4 = (t + (fMult(r4, C52) << 1)); r2 = (t + fMult(r2, C53)); /* imaginary part */ x0 = (x[s * 10 + 1] >> SCALEFACTOR10); x1 = (x[s * 2 + 1] >> SCALEFACTOR10); x2 = (x[s * 6 + 1] >> SCALEFACTOR10); x3 = (x[s * 14 + 1] >> SCALEFACTOR10); x4 = (x[s * 18 + 1] >> SCALEFACTOR10); s1 = (x1 + x4); s4 = (x1 - x4); s3 = (x3 + x2); s2 = (x3 - x2); t = fMult((s1 - s3), C54); s1 = (s1 + s3); y03 = (x0 + s1); s1 = (y03 + (fMult(s1, C55) << 1)); s3 = (s1 - t); s1 = (s1 + t); t = fMult((s4 + s2), C51); s4 = (t + (fMult(s4, C52) << 1)); s2 = (t + fMult(s2, C53)); /* combination */ y06 = (r1 + s2); y18 = (r1 - s2); y10 = (r3 - s4); y14 = (r3 + s4); y07 = (s1 - r2); y19 = (s1 + r2); y11 = (s3 + r4); y15 = (s3 - r4); /* 5 fft2 stages */ x[s * 0] = (y00 + y02); x[s * 0 + 1] = (y01 + y03); x[s * 10] = (y00 - y02); x[s * 10 + 1] = (y01 - y03); x[s * 4] = (y04 + y06); x[s * 4 + 1] = (y05 + y07); x[s * 14] = (y04 - y06); x[s * 14 + 1] = (y05 - y07); x[s * 8] = (y08 + y10); x[s * 8 + 1] = (y09 + y11); x[s * 18] = (y08 - y10); x[s * 18 + 1] = (y09 - y11); x[s * 12] = (y12 + y14); x[s * 12 + 1] = (y13 + y15); x[s * 2] = (y12 - y14); x[s * 2 + 1] = (y13 - y15); x[s * 16] = (y16 + y18); x[s * 16 + 1] = (y17 + y19); x[s * 6] = (y16 - y18); x[s * 6 + 1] = (y17 - y19); } #ifndef FUNCTION_fft12 #define FUNCTION_fft12 #undef C31 #define C31 (STC(0x91261468)) /* FL2FXCONST_DBL(-0.86602540) = -sqrt(3)/2 */ static inline void fft12(FIXP_DBL *pInput) { FIXP_DBL aDst[24]; FIXP_DBL *pSrc, *pDst; int i; pSrc = pInput; pDst = aDst; FIXP_DBL r1, r2, s1, s2, pD; /* First 3*2 samples are shifted right by 2 before output */ r1 = pSrc[8] + pSrc[16]; r2 = fMultDiv2((pSrc[8] - pSrc[16]), C31); pD = pSrc[0] >> 1; pDst[0] = (pD + (r1 >> 1)) >> 1; r1 = pD - (r1 >> 2); /* imaginary part */ s1 = pSrc[9] + pSrc[17]; s2 = fMultDiv2((pSrc[9] - pSrc[17]), C31); pD = pSrc[1] >> 1; pDst[1] = (pD + (s1 >> 1)) >> 1; s1 = pD - (s1 >> 2); /* combination */ pDst[2] = (r1 - s2) >> 1; pDst[3] = (s1 + r2) >> 1; pDst[4] = (r1 + s2) >> 1; pDst[5] = (s1 - r2) >> 1; pSrc += 2; pDst += 6; const FIXP_STB *pVecRe = RotVectorReal12; const FIXP_STB *pVecIm = RotVectorImag12; FIXP_DBL re, im; FIXP_STB vre, vim; for (i = 0; i < 2; i++) { /* sample 0,1 are shifted right by 2 before output */ /* sample 2,3 4,5 are shifted right by 1 and complex multiplied before * output */ r1 = pSrc[8] + pSrc[16]; r2 = fMultDiv2((pSrc[8] - pSrc[16]), C31); pD = pSrc[0] >> 1; pDst[0] = (pD + (r1 >> 1)) >> 1; r1 = pD - (r1 >> 2); /* imaginary part */ s1 = pSrc[9] + pSrc[17]; s2 = fMultDiv2((pSrc[9] - pSrc[17]), C31); pD = pSrc[1] >> 1; pDst[1] = (pD + (s1 >> 1)) >> 1; s1 = pD - (s1 >> 2); /* combination */ re = (r1 - s2) >> 0; im = (s1 + r2) >> 0; vre = *pVecRe++; vim = *pVecIm++; cplxMultDiv2(&pDst[3], &pDst[2], im, re, vre, vim); re = (r1 + s2) >> 0; im = (s1 - r2) >> 0; vre = *pVecRe++; vim = *pVecIm++; cplxMultDiv2(&pDst[5], &pDst[4], im, re, vre, vim); pDst += 6; pSrc += 2; } /* sample 0,1 are shifted right by 2 before output */ /* sample 2,3 is shifted right by 1 and complex multiplied with (0.0,+1.0) */ /* sample 4,5 is shifted right by 1 and complex multiplied with (-1.0,0.0) */ r1 = pSrc[8] + pSrc[16]; r2 = fMultDiv2((pSrc[8] - pSrc[16]), C31); pD = pSrc[0] >> 1; pDst[0] = (pD + (r1 >> 1)) >> 1; r1 = pD - (r1 >> 2); /* imaginary part */ s1 = pSrc[9] + pSrc[17]; s2 = fMultDiv2((pSrc[9] - pSrc[17]), C31); pD = pSrc[1] >> 1; pDst[1] = (pD + (s1 >> 1)) >> 1; s1 = pD - (s1 >> 2); /* combination */ pDst[2] = (s1 + r2) >> 1; pDst[3] = (s2 - r1) >> 1; pDst[4] = -((r1 + s2) >> 1); pDst[5] = (r2 - s1) >> 1; /* Perform 3 times the fft of length 4. The input samples are at the address of aDst and the output samples are at the address of pInput. The input vector for the fft of length 4 is built of the interleaved samples in aDst, the output samples are stored consecutively at the address of pInput. */ pSrc = aDst; pDst = pInput; for (i = 0; i < 3; i++) { /* inline FFT4 merged with incoming resorting loop */ FIXP_DBL a00, a10, a20, a30, tmp0, tmp1; a00 = (pSrc[0] + pSrc[12]) >> 1; /* Re A + Re B */ a10 = (pSrc[6] + pSrc[18]) >> 1; /* Re C + Re D */ a20 = (pSrc[1] + pSrc[13]) >> 1; /* Im A + Im B */ a30 = (pSrc[7] + pSrc[19]) >> 1; /* Im C + Im D */ pDst[0] = a00 + a10; /* Re A' = Re A + Re B + Re C + Re D */ pDst[1] = a20 + a30; /* Im A' = Im A + Im B + Im C + Im D */ tmp0 = a00 - pSrc[12]; /* Re A - Re B */ tmp1 = a20 - pSrc[13]; /* Im A - Im B */ pDst[12] = a00 - a10; /* Re C' = Re A + Re B - Re C - Re D */ pDst[13] = a20 - a30; /* Im C' = Im A + Im B - Im C - Im D */ a10 = a10 - pSrc[18]; /* Re C - Re D */ a30 = a30 - pSrc[19]; /* Im C - Im D */ pDst[6] = tmp0 + a30; /* Re B' = Re A - Re B + Im C - Im D */ pDst[18] = tmp0 - a30; /* Re D' = Re A - Re B - Im C + Im D */ pDst[7] = tmp1 - a10; /* Im B' = Im A - Im B - Re C + Re D */ pDst[19] = tmp1 + a10; /* Im D' = Im A - Im B + Re C - Re D */ pSrc += 2; pDst += 2; } } #endif /* FUNCTION_fft12 */ #ifndef FUNCTION_fft15 #define N3 3 #define N5 5 #define N6 6 #define N15 15 /* Performs the FFT of length 15. It is split into FFTs of length 3 and * length 5. */ static inline void fft15(FIXP_DBL *pInput) { FIXP_DBL aDst[2 * N15]; FIXP_DBL aDst1[2 * N15]; int i, k, l; /* Sort input vector for fft's of length 3 input3(0:2) = [input(0) input(5) input(10)]; input3(3:5) = [input(3) input(8) input(13)]; input3(6:8) = [input(6) input(11) input(1)]; input3(9:11) = [input(9) input(14) input(4)]; input3(12:14) = [input(12) input(2) input(7)]; */ { const FIXP_DBL *pSrc = pInput; FIXP_DBL *RESTRICT pDst = aDst; /* Merge 3 loops into one, skip call of fft3 */ for (i = 0, l = 0, k = 0; i < N5; i++, k += 6) { pDst[k + 0] = pSrc[l]; pDst[k + 1] = pSrc[l + 1]; l += 2 * N5; if (l >= (2 * N15)) l -= (2 * N15); pDst[k + 2] = pSrc[l]; pDst[k + 3] = pSrc[l + 1]; l += 2 * N5; if (l >= (2 * N15)) l -= (2 * N15); pDst[k + 4] = pSrc[l]; pDst[k + 5] = pSrc[l + 1]; l += (2 * N5) + (2 * N3); if (l >= (2 * N15)) l -= (2 * N15); /* fft3 merged with shift right by 2 loop */ FIXP_DBL r1, r2, r3; FIXP_DBL s1, s2; /* real part */ r1 = pDst[k + 2] + pDst[k + 4]; r2 = fMult((pDst[k + 2] - pDst[k + 4]), C31); s1 = pDst[k + 0]; pDst[k + 0] = (s1 + r1) >> 2; r1 = s1 - (r1 >> 1); /* imaginary part */ s1 = pDst[k + 3] + pDst[k + 5]; s2 = fMult((pDst[k + 3] - pDst[k + 5]), C31); r3 = pDst[k + 1]; pDst[k + 1] = (r3 + s1) >> 2; s1 = r3 - (s1 >> 1); /* combination */ pDst[k + 2] = (r1 - s2) >> 2; pDst[k + 4] = (r1 + s2) >> 2; pDst[k + 3] = (s1 + r2) >> 2; pDst[k + 5] = (s1 - r2) >> 2; } } /* Sort input vector for fft's of length 5 input5(0:4) = [output3(0) output3(3) output3(6) output3(9) output3(12)]; input5(5:9) = [output3(1) output3(4) output3(7) output3(10) output3(13)]; input5(10:14) = [output3(2) output3(5) output3(8) output3(11) output3(14)]; */ /* Merge 2 loops into one, brings about 10% */ { const FIXP_DBL *pSrc = aDst; FIXP_DBL *RESTRICT pDst = aDst1; for (i = 0, l = 0, k = 0; i < N3; i++, k += 10) { l = 2 * i; pDst[k + 0] = pSrc[l + 0]; pDst[k + 1] = pSrc[l + 1]; pDst[k + 2] = pSrc[l + 0 + (2 * N3)]; pDst[k + 3] = pSrc[l + 1 + (2 * N3)]; pDst[k + 4] = pSrc[l + 0 + (4 * N3)]; pDst[k + 5] = pSrc[l + 1 + (4 * N3)]; pDst[k + 6] = pSrc[l + 0 + (6 * N3)]; pDst[k + 7] = pSrc[l + 1 + (6 * N3)]; pDst[k + 8] = pSrc[l + 0 + (8 * N3)]; pDst[k + 9] = pSrc[l + 1 + (8 * N3)]; fft5(&pDst[k]); } } /* Sort output vector of length 15 output = [out5(0) out5(6) out5(12) out5(3) out5(9) out5(10) out5(1) out5(7) out5(13) out5(4) out5(5) out5(11) out5(2) out5(8) out5(14)]; */ /* optimize clumsy loop, brings about 5% */ { const FIXP_DBL *pSrc = aDst1; FIXP_DBL *RESTRICT pDst = pInput; for (i = 0, l = 0, k = 0; i < N3; i++, k += 10) { pDst[k + 0] = pSrc[l]; pDst[k + 1] = pSrc[l + 1]; l += (2 * N6); if (l >= (2 * N15)) l -= (2 * N15); pDst[k + 2] = pSrc[l]; pDst[k + 3] = pSrc[l + 1]; l += (2 * N6); if (l >= (2 * N15)) l -= (2 * N15); pDst[k + 4] = pSrc[l]; pDst[k + 5] = pSrc[l + 1]; l += (2 * N6); if (l >= (2 * N15)) l -= (2 * N15); pDst[k + 6] = pSrc[l]; pDst[k + 7] = pSrc[l + 1]; l += (2 * N6); if (l >= (2 * N15)) l -= (2 * N15); pDst[k + 8] = pSrc[l]; pDst[k + 9] = pSrc[l + 1]; l += 2; /* no modulo check needed, it cannot occur */ } } } #endif /* FUNCTION_fft15 */ /* Select shift placement. Some processors like ARM may shift "for free" in combination with an addition or substraction, but others don't so either combining shift with +/- or reduce the total amount or shift operations is optimal */ #if !defined(__arm__) #define SHIFT_A >> 1 #define SHIFT_B #else #define SHIFT_A #define SHIFT_B >> 1 #endif #ifndef FUNCTION_fft_16 /* we check, if fft_16 (FIXP_DBL *) is not yet defined \ */ /* This defines prevents this array to be declared twice, if 16-bit fft is * enabled too */ #define FUNCTION_DATA_fft_16_w16 static const FIXP_STP fft16_w16[2] = {STCP(0x7641af3d, 0x30fbc54d), STCP(0x30fbc54d, 0x7641af3d)}; LNK_SECTION_CODE_L1 inline void fft_16(FIXP_DBL *RESTRICT x) { FIXP_DBL vr, ur; FIXP_DBL vr2, ur2; FIXP_DBL vr3, ur3; FIXP_DBL vr4, ur4; FIXP_DBL vi, ui; FIXP_DBL vi2, ui2; FIXP_DBL vi3, ui3; vr = (x[0] >> 1) + (x[16] >> 1); /* Re A + Re B */ ur = (x[1] >> 1) + (x[17] >> 1); /* Im A + Im B */ vi = (x[8] SHIFT_A) + (x[24] SHIFT_A); /* Re C + Re D */ ui = (x[9] SHIFT_A) + (x[25] SHIFT_A); /* Im C + Im D */ x[0] = vr + (vi SHIFT_B); /* Re A' = ReA + ReB +ReC + ReD */ x[1] = ur + (ui SHIFT_B); /* Im A' = sum of imag values */ vr2 = (x[4] >> 1) + (x[20] >> 1); /* Re A + Re B */ ur2 = (x[5] >> 1) + (x[21] >> 1); /* Im A + Im B */ x[4] = vr - (vi SHIFT_B); /* Re C' = -(ReC+ReD) + (ReA+ReB) */ x[5] = ur - (ui SHIFT_B); /* Im C' = -Im C -Im D +Im A +Im B */ vr -= x[16]; /* Re A - Re B */ vi = (vi SHIFT_B)-x[24]; /* Re C - Re D */ ur -= x[17]; /* Im A - Im B */ ui = (ui SHIFT_B)-x[25]; /* Im C - Im D */ vr3 = (x[2] >> 1) + (x[18] >> 1); /* Re A + Re B */ ur3 = (x[3] >> 1) + (x[19] >> 1); /* Im A + Im B */ x[2] = ui + vr; /* Re B' = Im C - Im D + Re A - Re B */ x[3] = ur - vi; /* Im B'= -Re C + Re D + Im A - Im B */ vr4 = (x[6] >> 1) + (x[22] >> 1); /* Re A + Re B */ ur4 = (x[7] >> 1) + (x[23] >> 1); /* Im A + Im B */ x[6] = vr - ui; /* Re D' = -Im C + Im D + Re A - Re B */ x[7] = vi + ur; /* Im D'= Re C - Re D + Im A - Im B */ vi2 = (x[12] SHIFT_A) + (x[28] SHIFT_A); /* Re C + Re D */ ui2 = (x[13] SHIFT_A) + (x[29] SHIFT_A); /* Im C + Im D */ x[8] = vr2 + (vi2 SHIFT_B); /* Re A' = ReA + ReB +ReC + ReD */ x[9] = ur2 + (ui2 SHIFT_B); /* Im A' = sum of imag values */ x[12] = vr2 - (vi2 SHIFT_B); /* Re C' = -(ReC+ReD) + (ReA+ReB) */ x[13] = ur2 - (ui2 SHIFT_B); /* Im C' = -Im C -Im D +Im A +Im B */ vr2 -= x[20]; /* Re A - Re B */ ur2 -= x[21]; /* Im A - Im B */ vi2 = (vi2 SHIFT_B)-x[28]; /* Re C - Re D */ ui2 = (ui2 SHIFT_B)-x[29]; /* Im C - Im D */ vi = (x[10] SHIFT_A) + (x[26] SHIFT_A); /* Re C + Re D */ ui = (x[11] SHIFT_A) + (x[27] SHIFT_A); /* Im C + Im D */ x[10] = ui2 + vr2; /* Re B' = Im C - Im D + Re A - Re B */ x[11] = ur2 - vi2; /* Im B'= -Re C + Re D + Im A - Im B */ vi3 = (x[14] SHIFT_A) + (x[30] SHIFT_A); /* Re C + Re D */ ui3 = (x[15] SHIFT_A) + (x[31] SHIFT_A); /* Im C + Im D */ x[14] = vr2 - ui2; /* Re D' = -Im C + Im D + Re A - Re B */ x[15] = vi2 + ur2; /* Im D'= Re C - Re D + Im A - Im B */ x[16] = vr3 + (vi SHIFT_B); /* Re A' = ReA + ReB +ReC + ReD */ x[17] = ur3 + (ui SHIFT_B); /* Im A' = sum of imag values */ x[20] = vr3 - (vi SHIFT_B); /* Re C' = -(ReC+ReD) + (ReA+ReB) */ x[21] = ur3 - (ui SHIFT_B); /* Im C' = -Im C -Im D +Im A +Im B */ vr3 -= x[18]; /* Re A - Re B */ ur3 -= x[19]; /* Im A - Im B */ vi = (vi SHIFT_B)-x[26]; /* Re C - Re D */ ui = (ui SHIFT_B)-x[27]; /* Im C - Im D */ x[18] = ui + vr3; /* Re B' = Im C - Im D + Re A - Re B */ x[19] = ur3 - vi; /* Im B'= -Re C + Re D + Im A - Im B */ x[24] = vr4 + (vi3 SHIFT_B); /* Re A' = ReA + ReB +ReC + ReD */ x[28] = vr4 - (vi3 SHIFT_B); /* Re C' = -(ReC+ReD) + (ReA+ReB) */ x[25] = ur4 + (ui3 SHIFT_B); /* Im A' = sum of imag values */ x[29] = ur4 - (ui3 SHIFT_B); /* Im C' = -Im C -Im D +Im A +Im B */ vr4 -= x[22]; /* Re A - Re B */ ur4 -= x[23]; /* Im A - Im B */ x[22] = vr3 - ui; /* Re D' = -Im C + Im D + Re A - Re B */ x[23] = vi + ur3; /* Im D'= Re C - Re D + Im A - Im B */ vi3 = (vi3 SHIFT_B)-x[30]; /* Re C - Re D */ ui3 = (ui3 SHIFT_B)-x[31]; /* Im C - Im D */ x[26] = ui3 + vr4; /* Re B' = Im C - Im D + Re A - Re B */ x[30] = vr4 - ui3; /* Re D' = -Im C + Im D + Re A - Re B */ x[27] = ur4 - vi3; /* Im B'= -Re C + Re D + Im A - Im B */ x[31] = vi3 + ur4; /* Im D'= Re C - Re D + Im A - Im B */ // xt1 = 0 // xt2 = 8 vr = x[8]; vi = x[9]; ur = x[0] >> 1; ui = x[1] >> 1; x[0] = ur + (vr >> 1); x[1] = ui + (vi >> 1); x[8] = ur - (vr >> 1); x[9] = ui - (vi >> 1); // xt1 = 4 // xt2 = 12 vr = x[13]; vi = x[12]; ur = x[4] >> 1; ui = x[5] >> 1; x[4] = ur + (vr >> 1); x[5] = ui - (vi >> 1); x[12] = ur - (vr >> 1); x[13] = ui + (vi >> 1); // xt1 = 16 // xt2 = 24 vr = x[24]; vi = x[25]; ur = x[16] >> 1; ui = x[17] >> 1; x[16] = ur + (vr >> 1); x[17] = ui + (vi >> 1); x[24] = ur - (vr >> 1); x[25] = ui - (vi >> 1); // xt1 = 20 // xt2 = 28 vr = x[29]; vi = x[28]; ur = x[20] >> 1; ui = x[21] >> 1; x[20] = ur + (vr >> 1); x[21] = ui - (vi >> 1); x[28] = ur - (vr >> 1); x[29] = ui + (vi >> 1); // xt1 = 2 // xt2 = 10 SUMDIFF_PIFOURTH(vi, vr, x[10], x[11]) // vr = fMultDiv2((x[11] + x[10]),W_PiFOURTH); // vi = fMultDiv2((x[11] - x[10]),W_PiFOURTH); ur = x[2]; ui = x[3]; x[2] = (ur >> 1) + vr; x[3] = (ui >> 1) + vi; x[10] = (ur >> 1) - vr; x[11] = (ui >> 1) - vi; // xt1 = 6 // xt2 = 14 SUMDIFF_PIFOURTH(vr, vi, x[14], x[15]) ur = x[6]; ui = x[7]; x[6] = (ur >> 1) + vr; x[7] = (ui >> 1) - vi; x[14] = (ur >> 1) - vr; x[15] = (ui >> 1) + vi; // xt1 = 18 // xt2 = 26 SUMDIFF_PIFOURTH(vi, vr, x[26], x[27]) ur = x[18]; ui = x[19]; x[18] = (ur >> 1) + vr; x[19] = (ui >> 1) + vi; x[26] = (ur >> 1) - vr; x[27] = (ui >> 1) - vi; // xt1 = 22 // xt2 = 30 SUMDIFF_PIFOURTH(vr, vi, x[30], x[31]) ur = x[22]; ui = x[23]; x[22] = (ur >> 1) + vr; x[23] = (ui >> 1) - vi; x[30] = (ur >> 1) - vr; x[31] = (ui >> 1) + vi; // xt1 = 0 // xt2 = 16 vr = x[16]; vi = x[17]; ur = x[0] >> 1; ui = x[1] >> 1; x[0] = ur + (vr >> 1); x[1] = ui + (vi >> 1); x[16] = ur - (vr >> 1); x[17] = ui - (vi >> 1); // xt1 = 8 // xt2 = 24 vi = x[24]; vr = x[25]; ur = x[8] >> 1; ui = x[9] >> 1; x[8] = ur + (vr >> 1); x[9] = ui - (vi >> 1); x[24] = ur - (vr >> 1); x[25] = ui + (vi >> 1); // xt1 = 2 // xt2 = 18 cplxMultDiv2(&vi, &vr, x[19], x[18], fft16_w16[0]); ur = x[2]; ui = x[3]; x[2] = (ur >> 1) + vr; x[3] = (ui >> 1) + vi; x[18] = (ur >> 1) - vr; x[19] = (ui >> 1) - vi; // xt1 = 10 // xt2 = 26 cplxMultDiv2(&vr, &vi, x[27], x[26], fft16_w16[0]); ur = x[10]; ui = x[11]; x[10] = (ur >> 1) + vr; x[11] = (ui >> 1) - vi; x[26] = (ur >> 1) - vr; x[27] = (ui >> 1) + vi; // xt1 = 4 // xt2 = 20 SUMDIFF_PIFOURTH(vi, vr, x[20], x[21]) ur = x[4]; ui = x[5]; x[4] = (ur >> 1) + vr; x[5] = (ui >> 1) + vi; x[20] = (ur >> 1) - vr; x[21] = (ui >> 1) - vi; // xt1 = 12 // xt2 = 28 SUMDIFF_PIFOURTH(vr, vi, x[28], x[29]) ur = x[12]; ui = x[13]; x[12] = (ur >> 1) + vr; x[13] = (ui >> 1) - vi; x[28] = (ur >> 1) - vr; x[29] = (ui >> 1) + vi; // xt1 = 6 // xt2 = 22 cplxMultDiv2(&vi, &vr, x[23], x[22], fft16_w16[1]); ur = x[6]; ui = x[7]; x[6] = (ur >> 1) + vr; x[7] = (ui >> 1) + vi; x[22] = (ur >> 1) - vr; x[23] = (ui >> 1) - vi; // xt1 = 14 // xt2 = 30 cplxMultDiv2(&vr, &vi, x[31], x[30], fft16_w16[1]); ur = x[14]; ui = x[15]; x[14] = (ur >> 1) + vr; x[15] = (ui >> 1) - vi; x[30] = (ur >> 1) - vr; x[31] = (ui >> 1) + vi; } #endif /* FUNCTION_fft_16 */ #ifndef FUNCTION_fft_32 static const FIXP_STP fft32_w32[6] = { STCP(0x7641af3d, 0x30fbc54d), STCP(0x30fbc54d, 0x7641af3d), STCP(0x7d8a5f40, 0x18f8b83c), STCP(0x6a6d98a4, 0x471cece7), STCP(0x471cece7, 0x6a6d98a4), STCP(0x18f8b83c, 0x7d8a5f40)}; #define W_PiFOURTH STC(0x5a82799a) LNK_SECTION_CODE_L1 inline void fft_32(FIXP_DBL *const _x) { /* * 1+2 stage radix 4 */ ///////////////////////////////////////////////////////////////////////////////////////// { FIXP_DBL *const x = _x; FIXP_DBL vi, ui; FIXP_DBL vi2, ui2; FIXP_DBL vi3, ui3; FIXP_DBL vr, ur; FIXP_DBL vr2, ur2; FIXP_DBL vr3, ur3; FIXP_DBL vr4, ur4; // i = 0 vr = (x[0] + x[32]) >> 1; /* Re A + Re B */ ur = (x[1] + x[33]) >> 1; /* Im A + Im B */ vi = (x[16] + x[48]) SHIFT_A; /* Re C + Re D */ ui = (x[17] + x[49]) SHIFT_A; /* Im C + Im D */ x[0] = vr + (vi SHIFT_B); /* Re A' = ReA + ReB +ReC + ReD */ x[1] = ur + (ui SHIFT_B); /* Im A' = sum of imag values */ vr2 = (x[4] + x[36]) >> 1; /* Re A + Re B */ ur2 = (x[5] + x[37]) >> 1; /* Im A + Im B */ x[4] = vr - (vi SHIFT_B); /* Re C' = -(ReC+ReD) + (ReA+ReB) */ x[5] = ur - (ui SHIFT_B); /* Im C' = -Im C -Im D +Im A +Im B */ vr -= x[32]; /* Re A - Re B */ ur -= x[33]; /* Im A - Im B */ vi = (vi SHIFT_B)-x[48]; /* Re C - Re D */ ui = (ui SHIFT_B)-x[49]; /* Im C - Im D */ vr3 = (x[2] + x[34]) >> 1; /* Re A + Re B */ ur3 = (x[3] + x[35]) >> 1; /* Im A + Im B */ x[2] = ui + vr; /* Re B' = Im C - Im D + Re A - Re B */ x[3] = ur - vi; /* Im B'= -Re C + Re D + Im A - Im B */ vr4 = (x[6] + x[38]) >> 1; /* Re A + Re B */ ur4 = (x[7] + x[39]) >> 1; /* Im A + Im B */ x[6] = vr - ui; /* Re D' = -Im C + Im D + Re A - Re B */ x[7] = vi + ur; /* Im D'= Re C - Re D + Im A - Im B */ // i=16 vi = (x[20] + x[52]) SHIFT_A; /* Re C + Re D */ ui = (x[21] + x[53]) SHIFT_A; /* Im C + Im D */ x[16] = vr2 + (vi SHIFT_B); /* Re A' = ReA + ReB +ReC + ReD */ x[17] = ur2 + (ui SHIFT_B); /* Im A' = sum of imag values */ x[20] = vr2 - (vi SHIFT_B); /* Re C' = -(ReC+ReD) + (ReA+ReB) */ x[21] = ur2 - (ui SHIFT_B); /* Im C' = -Im C -Im D +Im A +Im B */ vr2 -= x[36]; /* Re A - Re B */ ur2 -= x[37]; /* Im A - Im B */ vi = (vi SHIFT_B)-x[52]; /* Re C - Re D */ ui = (ui SHIFT_B)-x[53]; /* Im C - Im D */ vi2 = (x[18] + x[50]) SHIFT_A; /* Re C + Re D */ ui2 = (x[19] + x[51]) SHIFT_A; /* Im C + Im D */ x[18] = ui + vr2; /* Re B' = Im C - Im D + Re A - Re B */ x[19] = ur2 - vi; /* Im B'= -Re C + Re D + Im A - Im B */ vi3 = (x[22] + x[54]) SHIFT_A; /* Re C + Re D */ ui3 = (x[23] + x[55]) SHIFT_A; /* Im C + Im D */ x[22] = vr2 - ui; /* Re D' = -Im C + Im D + Re A - Re B */ x[23] = vi + ur2; /* Im D'= Re C - Re D + Im A - Im B */ // i = 32 x[32] = vr3 + (vi2 SHIFT_B); /* Re A' = ReA + ReB +ReC + ReD */ x[33] = ur3 + (ui2 SHIFT_B); /* Im A' = sum of imag values */ x[36] = vr3 - (vi2 SHIFT_B); /* Re C' = -(ReC+ReD) + (ReA+ReB) */ x[37] = ur3 - (ui2 SHIFT_B); /* Im C' = -Im C -Im D +Im A +Im B */ vr3 -= x[34]; /* Re A - Re B */ ur3 -= x[35]; /* Im A - Im B */ vi2 = (vi2 SHIFT_B)-x[50]; /* Re C - Re D */ ui2 = (ui2 SHIFT_B)-x[51]; /* Im C - Im D */ x[34] = ui2 + vr3; /* Re B' = Im C - Im D + Re A - Re B */ x[35] = ur3 - vi2; /* Im B'= -Re C + Re D + Im A - Im B */ // i=48 x[48] = vr4 + (vi3 SHIFT_B); /* Re A' = ReA + ReB +ReC + ReD */ x[52] = vr4 - (vi3 SHIFT_B); /* Re C' = -(ReC+ReD) + (ReA+ReB) */ x[49] = ur4 + (ui3 SHIFT_B); /* Im A' = sum of imag values */ x[53] = ur4 - (ui3 SHIFT_B); /* Im C' = -Im C -Im D +Im A +Im B */ vr4 -= x[38]; /* Re A - Re B */ ur4 -= x[39]; /* Im A - Im B */ x[38] = vr3 - ui2; /* Re D' = -Im C + Im D + Re A - Re B */ x[39] = vi2 + ur3; /* Im D'= Re C - Re D + Im A - Im B */ vi3 = (vi3 SHIFT_B)-x[54]; /* Re C - Re D */ ui3 = (ui3 SHIFT_B)-x[55]; /* Im C - Im D */ x[50] = ui3 + vr4; /* Re B' = Im C - Im D + Re A - Re B */ x[54] = vr4 - ui3; /* Re D' = -Im C + Im D + Re A - Re B */ x[51] = ur4 - vi3; /* Im B'= -Re C + Re D + Im A - Im B */ x[55] = vi3 + ur4; /* Im D'= Re C - Re D + Im A - Im B */ // i=8 vr = (x[8] + x[40]) >> 1; /* Re A + Re B */ ur = (x[9] + x[41]) >> 1; /* Im A + Im B */ vi = (x[24] + x[56]) SHIFT_A; /* Re C + Re D */ ui = (x[25] + x[57]) SHIFT_A; /* Im C + Im D */ x[8] = vr + (vi SHIFT_B); /* Re A' = ReA + ReB +ReC + ReD */ x[9] = ur + (ui SHIFT_B); /* Im A' = sum of imag values */ vr2 = (x[12] + x[44]) >> 1; /* Re A + Re B */ ur2 = (x[13] + x[45]) >> 1; /* Im A + Im B */ x[12] = vr - (vi SHIFT_B); /* Re C' = -(ReC+ReD) + (ReA+ReB) */ x[13] = ur - (ui SHIFT_B); /* Im C' = -Im C -Im D +Im A +Im B */ vr -= x[40]; /* Re A - Re B */ ur -= x[41]; /* Im A - Im B */ vi = (vi SHIFT_B)-x[56]; /* Re C - Re D */ ui = (ui SHIFT_B)-x[57]; /* Im C - Im D */ vr3 = (x[10] + x[42]) >> 1; /* Re A + Re B */ ur3 = (x[11] + x[43]) >> 1; /* Im A + Im B */ x[10] = ui + vr; /* Re B' = Im C - Im D + Re A - Re B */ x[11] = ur - vi; /* Im B'= -Re C + Re D + Im A - Im B */ vr4 = (x[14] + x[46]) >> 1; /* Re A + Re B */ ur4 = (x[15] + x[47]) >> 1; /* Im A + Im B */ x[14] = vr - ui; /* Re D' = -Im C + Im D + Re A - Re B */ x[15] = vi + ur; /* Im D'= Re C - Re D + Im A - Im B */ // i=24 vi = (x[28] + x[60]) SHIFT_A; /* Re C + Re D */ ui = (x[29] + x[61]) SHIFT_A; /* Im C + Im D */ x[24] = vr2 + (vi SHIFT_B); /* Re A' = ReA + ReB +ReC + ReD */ x[28] = vr2 - (vi SHIFT_B); /* Re C' = -(ReC+ReD) + (ReA+ReB) */ x[25] = ur2 + (ui SHIFT_B); /* Im A' = sum of imag values */ x[29] = ur2 - (ui SHIFT_B); /* Im C' = -Im C -Im D +Im A +Im B */ vr2 -= x[44]; /* Re A - Re B */ ur2 -= x[45]; /* Im A - Im B */ vi = (vi SHIFT_B)-x[60]; /* Re C - Re D */ ui = (ui SHIFT_B)-x[61]; /* Im C - Im D */ vi2 = (x[26] + x[58]) SHIFT_A; /* Re C + Re D */ ui2 = (x[27] + x[59]) SHIFT_A; /* Im C + Im D */ x[26] = ui + vr2; /* Re B' = Im C - Im D + Re A - Re B */ x[27] = ur2 - vi; /* Im B'= -Re C + Re D + Im A - Im B */ vi3 = (x[30] + x[62]) SHIFT_A; /* Re C + Re D */ ui3 = (x[31] + x[63]) SHIFT_A; /* Im C + Im D */ x[30] = vr2 - ui; /* Re D' = -Im C + Im D + Re A - Re B */ x[31] = vi + ur2; /* Im D'= Re C - Re D + Im A - Im B */ // i=40 x[40] = vr3 + (vi2 SHIFT_B); /* Re A' = ReA + ReB +ReC + ReD */ x[44] = vr3 - (vi2 SHIFT_B); /* Re C' = -(ReC+ReD) + (ReA+ReB) */ x[41] = ur3 + (ui2 SHIFT_B); /* Im A' = sum of imag values */ x[45] = ur3 - (ui2 SHIFT_B); /* Im C' = -Im C -Im D +Im A +Im B */ vr3 -= x[42]; /* Re A - Re B */ ur3 -= x[43]; /* Im A - Im B */ vi2 = (vi2 SHIFT_B)-x[58]; /* Re C - Re D */ ui2 = (ui2 SHIFT_B)-x[59]; /* Im C - Im D */ x[42] = ui2 + vr3; /* Re B' = Im C - Im D + Re A - Re B */ x[43] = ur3 - vi2; /* Im B'= -Re C + Re D + Im A - Im B */ // i=56 x[56] = vr4 + (vi3 SHIFT_B); /* Re A' = ReA + ReB +ReC + ReD */ x[60] = vr4 - (vi3 SHIFT_B); /* Re C' = -(ReC+ReD) + (ReA+ReB) */ x[57] = ur4 + (ui3 SHIFT_B); /* Im A' = sum of imag values */ x[61] = ur4 - (ui3 SHIFT_B); /* Im C' = -Im C -Im D +Im A +Im B */ vr4 -= x[46]; /* Re A - Re B */ ur4 -= x[47]; /* Im A - Im B */ x[46] = vr3 - ui2; /* Re D' = -Im C + Im D + Re A - Re B */ x[47] = vi2 + ur3; /* Im D'= Re C - Re D + Im A - Im B */ vi3 = (vi3 SHIFT_B)-x[62]; /* Re C - Re D */ ui3 = (ui3 SHIFT_B)-x[63]; /* Im C - Im D */ x[58] = ui3 + vr4; /* Re B' = Im C - Im D + Re A - Re B */ x[62] = vr4 - ui3; /* Re D' = -Im C + Im D + Re A - Re B */ x[59] = ur4 - vi3; /* Im B'= -Re C + Re D + Im A - Im B */ x[63] = vi3 + ur4; /* Im D'= Re C - Re D + Im A - Im B */ } { FIXP_DBL *xt = _x; int j = 4; do { FIXP_DBL vi, ui, vr, ur; vr = xt[8]; vi = xt[9]; ur = xt[0] >> 1; ui = xt[1] >> 1; xt[0] = ur + (vr >> 1); xt[1] = ui + (vi >> 1); xt[8] = ur - (vr >> 1); xt[9] = ui - (vi >> 1); vr = xt[13]; vi = xt[12]; ur = xt[4] >> 1; ui = xt[5] >> 1; xt[4] = ur + (vr >> 1); xt[5] = ui - (vi >> 1); xt[12] = ur - (vr >> 1); xt[13] = ui + (vi >> 1); SUMDIFF_PIFOURTH(vi, vr, xt[10], xt[11]) ur = xt[2]; ui = xt[3]; xt[2] = (ur >> 1) + vr; xt[3] = (ui >> 1) + vi; xt[10] = (ur >> 1) - vr; xt[11] = (ui >> 1) - vi; SUMDIFF_PIFOURTH(vr, vi, xt[14], xt[15]) ur = xt[6]; ui = xt[7]; xt[6] = (ur >> 1) + vr; xt[7] = (ui >> 1) - vi; xt[14] = (ur >> 1) - vr; xt[15] = (ui >> 1) + vi; xt += 16; } while (--j != 0); } { FIXP_DBL *const x = _x; FIXP_DBL vi, ui, vr, ur; vr = x[16]; vi = x[17]; ur = x[0] >> 1; ui = x[1] >> 1; x[0] = ur + (vr >> 1); x[1] = ui + (vi >> 1); x[16] = ur - (vr >> 1); x[17] = ui - (vi >> 1); vi = x[24]; vr = x[25]; ur = x[8] >> 1; ui = x[9] >> 1; x[8] = ur + (vr >> 1); x[9] = ui - (vi >> 1); x[24] = ur - (vr >> 1); x[25] = ui + (vi >> 1); vr = x[48]; vi = x[49]; ur = x[32] >> 1; ui = x[33] >> 1; x[32] = ur + (vr >> 1); x[33] = ui + (vi >> 1); x[48] = ur - (vr >> 1); x[49] = ui - (vi >> 1); vi = x[56]; vr = x[57]; ur = x[40] >> 1; ui = x[41] >> 1; x[40] = ur + (vr >> 1); x[41] = ui - (vi >> 1); x[56] = ur - (vr >> 1); x[57] = ui + (vi >> 1); cplxMultDiv2(&vi, &vr, x[19], x[18], fft32_w32[0]); ur = x[2]; ui = x[3]; x[2] = (ur >> 1) + vr; x[3] = (ui >> 1) + vi; x[18] = (ur >> 1) - vr; x[19] = (ui >> 1) - vi; cplxMultDiv2(&vr, &vi, x[27], x[26], fft32_w32[0]); ur = x[10]; ui = x[11]; x[10] = (ur >> 1) + vr; x[11] = (ui >> 1) - vi; x[26] = (ur >> 1) - vr; x[27] = (ui >> 1) + vi; cplxMultDiv2(&vi, &vr, x[51], x[50], fft32_w32[0]); ur = x[34]; ui = x[35]; x[34] = (ur >> 1) + vr; x[35] = (ui >> 1) + vi; x[50] = (ur >> 1) - vr; x[51] = (ui >> 1) - vi; cplxMultDiv2(&vr, &vi, x[59], x[58], fft32_w32[0]); ur = x[42]; ui = x[43]; x[42] = (ur >> 1) + vr; x[43] = (ui >> 1) - vi; x[58] = (ur >> 1) - vr; x[59] = (ui >> 1) + vi; SUMDIFF_PIFOURTH(vi, vr, x[20], x[21]) ur = x[4]; ui = x[5]; x[4] = (ur >> 1) + vr; x[5] = (ui >> 1) + vi; x[20] = (ur >> 1) - vr; x[21] = (ui >> 1) - vi; SUMDIFF_PIFOURTH(vr, vi, x[28], x[29]) ur = x[12]; ui = x[13]; x[12] = (ur >> 1) + vr; x[13] = (ui >> 1) - vi; x[28] = (ur >> 1) - vr; x[29] = (ui >> 1) + vi; SUMDIFF_PIFOURTH(vi, vr, x[52], x[53]) ur = x[36]; ui = x[37]; x[36] = (ur >> 1) + vr; x[37] = (ui >> 1) + vi; x[52] = (ur >> 1) - vr; x[53] = (ui >> 1) - vi; SUMDIFF_PIFOURTH(vr, vi, x[60], x[61]) ur = x[44]; ui = x[45]; x[44] = (ur >> 1) + vr; x[45] = (ui >> 1) - vi; x[60] = (ur >> 1) - vr; x[61] = (ui >> 1) + vi; cplxMultDiv2(&vi, &vr, x[23], x[22], fft32_w32[1]); ur = x[6]; ui = x[7]; x[6] = (ur >> 1) + vr; x[7] = (ui >> 1) + vi; x[22] = (ur >> 1) - vr; x[23] = (ui >> 1) - vi; cplxMultDiv2(&vr, &vi, x[31], x[30], fft32_w32[1]); ur = x[14]; ui = x[15]; x[14] = (ur >> 1) + vr; x[15] = (ui >> 1) - vi; x[30] = (ur >> 1) - vr; x[31] = (ui >> 1) + vi; cplxMultDiv2(&vi, &vr, x[55], x[54], fft32_w32[1]); ur = x[38]; ui = x[39]; x[38] = (ur >> 1) + vr; x[39] = (ui >> 1) + vi; x[54] = (ur >> 1) - vr; x[55] = (ui >> 1) - vi; cplxMultDiv2(&vr, &vi, x[63], x[62], fft32_w32[1]); ur = x[46]; ui = x[47]; x[46] = (ur >> 1) + vr; x[47] = (ui >> 1) - vi; x[62] = (ur >> 1) - vr; x[63] = (ui >> 1) + vi; vr = x[32]; vi = x[33]; ur = x[0] >> 1; ui = x[1] >> 1; x[0] = ur + (vr >> 1); x[1] = ui + (vi >> 1); x[32] = ur - (vr >> 1); x[33] = ui - (vi >> 1); vi = x[48]; vr = x[49]; ur = x[16] >> 1; ui = x[17] >> 1; x[16] = ur + (vr >> 1); x[17] = ui - (vi >> 1); x[48] = ur - (vr >> 1); x[49] = ui + (vi >> 1); cplxMultDiv2(&vi, &vr, x[35], x[34], fft32_w32[2]); ur = x[2]; ui = x[3]; x[2] = (ur >> 1) + vr; x[3] = (ui >> 1) + vi; x[34] = (ur >> 1) - vr; x[35] = (ui >> 1) - vi; cplxMultDiv2(&vr, &vi, x[51], x[50], fft32_w32[2]); ur = x[18]; ui = x[19]; x[18] = (ur >> 1) + vr; x[19] = (ui >> 1) - vi; x[50] = (ur >> 1) - vr; x[51] = (ui >> 1) + vi; cplxMultDiv2(&vi, &vr, x[37], x[36], fft32_w32[0]); ur = x[4]; ui = x[5]; x[4] = (ur >> 1) + vr; x[5] = (ui >> 1) + vi; x[36] = (ur >> 1) - vr; x[37] = (ui >> 1) - vi; cplxMultDiv2(&vr, &vi, x[53], x[52], fft32_w32[0]); ur = x[20]; ui = x[21]; x[20] = (ur >> 1) + vr; x[21] = (ui >> 1) - vi; x[52] = (ur >> 1) - vr; x[53] = (ui >> 1) + vi; cplxMultDiv2(&vi, &vr, x[39], x[38], fft32_w32[3]); ur = x[6]; ui = x[7]; x[6] = (ur >> 1) + vr; x[7] = (ui >> 1) + vi; x[38] = (ur >> 1) - vr; x[39] = (ui >> 1) - vi; cplxMultDiv2(&vr, &vi, x[55], x[54], fft32_w32[3]); ur = x[22]; ui = x[23]; x[22] = (ur >> 1) + vr; x[23] = (ui >> 1) - vi; x[54] = (ur >> 1) - vr; x[55] = (ui >> 1) + vi; SUMDIFF_PIFOURTH(vi, vr, x[40], x[41]) ur = x[8]; ui = x[9]; x[8] = (ur >> 1) + vr; x[9] = (ui >> 1) + vi; x[40] = (ur >> 1) - vr; x[41] = (ui >> 1) - vi; SUMDIFF_PIFOURTH(vr, vi, x[56], x[57]) ur = x[24]; ui = x[25]; x[24] = (ur >> 1) + vr; x[25] = (ui >> 1) - vi; x[56] = (ur >> 1) - vr; x[57] = (ui >> 1) + vi; cplxMultDiv2(&vi, &vr, x[43], x[42], fft32_w32[4]); ur = x[10]; ui = x[11]; x[10] = (ur >> 1) + vr; x[11] = (ui >> 1) + vi; x[42] = (ur >> 1) - vr; x[43] = (ui >> 1) - vi; cplxMultDiv2(&vr, &vi, x[59], x[58], fft32_w32[4]); ur = x[26]; ui = x[27]; x[26] = (ur >> 1) + vr; x[27] = (ui >> 1) - vi; x[58] = (ur >> 1) - vr; x[59] = (ui >> 1) + vi; cplxMultDiv2(&vi, &vr, x[45], x[44], fft32_w32[1]); ur = x[12]; ui = x[13]; x[12] = (ur >> 1) + vr; x[13] = (ui >> 1) + vi; x[44] = (ur >> 1) - vr; x[45] = (ui >> 1) - vi; cplxMultDiv2(&vr, &vi, x[61], x[60], fft32_w32[1]); ur = x[28]; ui = x[29]; x[28] = (ur >> 1) + vr; x[29] = (ui >> 1) - vi; x[60] = (ur >> 1) - vr; x[61] = (ui >> 1) + vi; cplxMultDiv2(&vi, &vr, x[47], x[46], fft32_w32[5]); ur = x[14]; ui = x[15]; x[14] = (ur >> 1) + vr; x[15] = (ui >> 1) + vi; x[46] = (ur >> 1) - vr; x[47] = (ui >> 1) - vi; cplxMultDiv2(&vr, &vi, x[63], x[62], fft32_w32[5]); ur = x[30]; ui = x[31]; x[30] = (ur >> 1) + vr; x[31] = (ui >> 1) - vi; x[62] = (ur >> 1) - vr; x[63] = (ui >> 1) + vi; } } #endif /* #ifndef FUNCTION_fft_32 */ /** * \brief Apply rotation vectors to a data buffer. * \param cl length of each row of input data. * \param l total length of input data. * \param pVecRe real part of rotation coefficient vector. * \param pVecIm imaginary part of rotation coefficient vector. */ /* This defines patches each inaccurate 0x7FFF i.e. 0.9999 and uses 0x8000 (-1.0) instead. At the end, the sign of the result is inverted */ #define noFFT_APPLY_ROT_VECTOR_HQ #ifndef FUNCTION_fft_apply_rot_vector__FIXP_DBL static inline void fft_apply_rot_vector(FIXP_DBL *RESTRICT pData, const int cl, const int l, const FIXP_STB *pVecRe, const FIXP_STB *pVecIm) { FIXP_DBL re, im; FIXP_STB vre, vim; int i, c; for (i = 0; i < cl; i++) { re = pData[2 * i]; im = pData[2 * i + 1]; pData[2 * i] = re >> 2; /* * 0.25 */ pData[2 * i + 1] = im >> 2; /* * 0.25 */ } for (; i < l; i += cl) { re = pData[2 * i]; im = pData[2 * i + 1]; pData[2 * i] = re >> 2; /* * 0.25 */ pData[2 * i + 1] = im >> 2; /* * 0.25 */ for (c = i + 1; c < i + cl; c++) { re = pData[2 * c] >> 1; im = pData[2 * c + 1] >> 1; vre = *pVecRe++; vim = *pVecIm++; cplxMultDiv2(&pData[2 * c + 1], &pData[2 * c], im, re, vre, vim); } } } #endif /* FUNCTION_fft_apply_rot_vector__FIXP_DBL */ /* select either switch case of function pointer. */ //#define FFT_TWO_STAGE_SWITCH_CASE #ifndef FUNCTION_fftN2_func static inline void fftN2_func(FIXP_DBL *pInput, const int length, const int dim1, const int dim2, void (*const fft1)(FIXP_DBL *), void (*const fft2)(FIXP_DBL *), const FIXP_STB *RotVectorReal, const FIXP_STB *RotVectorImag, FIXP_DBL *aDst, FIXP_DBL *aDst2) { /* The real part of the input samples are at the addresses with even indices and the imaginary part of the input samples are at the addresses with odd indices. The output samples are stored at the address of pInput */ FIXP_DBL *pSrc, *pDst, *pDstOut; int i; FDK_ASSERT(length == dim1 * dim2); /* Perform dim2 times the fft of length dim1. The input samples are at the address of pSrc and the output samples are at the address of pDst. The input vector for the fft of length dim1 is built of the interleaved samples in pSrc, the output samples are stored consecutively. */ pSrc = pInput; pDst = aDst; for (i = 0; i < dim2; i++) { for (int j = 0; j < dim1; j++) { pDst[2 * j] = pSrc[2 * j * dim2]; pDst[2 * j + 1] = pSrc[2 * j * dim2 + 1]; } /* fft of size dim1 */ #ifndef FFT_TWO_STAGE_SWITCH_CASE fft1(pDst); #else switch (dim1) { case 2: fft2(pDst); break; case 3: fft3(pDst); break; case 4: fft_4(pDst); break; /* case 5: fft5(pDst); break; */ /* case 8: fft_8(pDst); break; */ case 12: fft12(pDst); break; /* case 15: fft15(pDst); break; */ case 16: fft_16(pDst); break; case 32: fft_32(pDst); break; /*case 64: fft_64(pDst); break;*/ /* case 128: fft_128(pDst); break; */ } #endif pSrc += 2; pDst = pDst + 2 * dim1; } /* Perform the modulation of the output of the fft of length dim1 */ pSrc = aDst; fft_apply_rot_vector(pSrc, dim1, length, RotVectorReal, RotVectorImag); /* Perform dim1 times the fft of length dim2. The input samples are at the address of aDst and the output samples are at the address of pInput. The input vector for the fft of length dim2 is built of the interleaved samples in aDst, the output samples are stored consecutively at the address of pInput. */ pSrc = aDst; pDst = aDst2; pDstOut = pInput; for (i = 0; i < dim1; i++) { for (int j = 0; j < dim2; j++) { pDst[2 * j] = pSrc[2 * j * dim1]; pDst[2 * j + 1] = pSrc[2 * j * dim1 + 1]; } #ifndef FFT_TWO_STAGE_SWITCH_CASE fft2(pDst); #else switch (dim2) { case 4: fft_4(pDst); break; case 9: fft9(pDst); break; case 12: fft12(pDst); break; case 15: fft15(pDst); break; case 16: fft_16(pDst); break; case 32: fft_32(pDst); break; } #endif for (int j = 0; j < dim2; j++) { pDstOut[2 * j * dim1] = pDst[2 * j]; pDstOut[2 * j * dim1 + 1] = pDst[2 * j + 1]; } pSrc += 2; pDstOut += 2; } } #endif /* FUNCTION_fftN2_function */ #define fftN2(DATA_TYPE, pInput, length, dim1, dim2, fft_func1, fft_func2, \ RotVectorReal, RotVectorImag) \ { \ C_AALLOC_SCRATCH_START(aDst, DATA_TYPE, 2 * length) \ C_AALLOC_SCRATCH_START(aDst2, DATA_TYPE, 2 * dim2) \ fftN2_func(pInput, length, dim1, dim2, fft_func1, fft_func2, \ RotVectorReal, RotVectorImag, aDst, aDst2); \ C_AALLOC_SCRATCH_END(aDst2, DATA_TYPE, 2 * dim2) \ C_AALLOC_SCRATCH_END(aDst, DATA_TYPE, 2 * length) \ } /*! * * \brief complex FFT of length 12,18,24,30,48,60,96, 192, 240, 384, 480 * \param pInput contains the input signal prescaled right by 2 * pInput contains the output signal scaled by SCALEFACTOR<#length> * The output signal does not have any fixed headroom * \return void * */ #ifndef FUNCTION_fft6 static inline void fft6(FIXP_DBL *pInput) { fftN2(FIXP_DBL, pInput, 6, 2, 3, fft2, fft3, RotVectorReal6, RotVectorImag6); } #endif /* #ifndef FUNCTION_fft6 */ #ifndef FUNCTION_fft12 static inline void fft12(FIXP_DBL *pInput) { fftN2(FIXP_DBL, pInput, 12, 3, 4, fft3, fft_4, RotVectorReal12, RotVectorImag12); /* 16,58 */ } #endif /* #ifndef FUNCTION_fft12 */ #ifndef FUNCTION_fft20 static inline void fft20(FIXP_DBL *pInput) { fftN2(FIXP_DBL, pInput, 20, 4, 5, fft_4, fft5, RotVectorReal20, RotVectorImag20); } #endif /* FUNCTION_fft20 */ static inline void fft24(FIXP_DBL *pInput) { fftN2(FIXP_DBL, pInput, 24, 2, 12, fft2, fft12, RotVectorReal24, RotVectorImag24); /* 16,73 */ } static inline void fft48(FIXP_DBL *pInput) { fftN2(FIXP_DBL, pInput, 48, 4, 12, fft_4, fft12, RotVectorReal48, RotVectorImag48); /* 16,32 */ } #ifndef FUNCTION_fft60 static inline void fft60(FIXP_DBL *pInput) { fftN2(FIXP_DBL, pInput, 60, 4, 15, fft_4, fft15, RotVectorReal60, RotVectorImag60); /* 15,51 */ } #endif /* FUNCTION_fft60 */ #ifndef FUNCTION_fft80 static inline void fft80(FIXP_DBL *pInput) { fftN2(FIXP_DBL, pInput, 80, 5, 16, fft5, fft_16, RotVectorReal80, RotVectorImag80); /* */ } #endif #ifndef FUNCTION_fft96 static inline void fft96(FIXP_DBL *pInput) { fftN2(FIXP_DBL, pInput, 96, 3, 32, fft3, fft_32, RotVectorReal96, RotVectorImag96); /* 15,47 */ } #endif /* FUNCTION_fft96*/ #ifndef FUNCTION_fft120 static inline void fft120(FIXP_DBL *pInput) { fftN2(FIXP_DBL, pInput, 120, 8, 15, fft_8, fft15, RotVectorReal120, RotVectorImag120); } #endif /* FUNCTION_fft120 */ #ifndef FUNCTION_fft192 static inline void fft192(FIXP_DBL *pInput) { fftN2(FIXP_DBL, pInput, 192, 16, 12, fft_16, fft12, RotVectorReal192, RotVectorImag192); /* 15,50 */ } #endif #ifndef FUNCTION_fft240 static inline void fft240(FIXP_DBL *pInput) { fftN2(FIXP_DBL, pInput, 240, 16, 15, fft_16, fft15, RotVectorReal240, RotVectorImag240); /* 15.44 */ } #endif #ifndef FUNCTION_fft384 static inline void fft384(FIXP_DBL *pInput) { fftN2(FIXP_DBL, pInput, 384, 12, 32, fft12, fft_32, RotVectorReal384, RotVectorImag384); /* 16.02 */ } #endif /* FUNCTION_fft384 */ #ifndef FUNCTION_fft480 static inline void fft480(FIXP_DBL *pInput) { fftN2(FIXP_DBL, pInput, 480, 32, 15, fft_32, fft15, RotVectorReal480, RotVectorImag480); /* 15.84 */ } #endif /* FUNCTION_fft480 */ void fft(int length, FIXP_DBL *pInput, INT *pScalefactor) { /* Ensure, that the io-ptr is always (at least 8-byte) aligned */ C_ALLOC_ALIGNED_CHECK(pInput); if (length == 32) { fft_32(pInput); *pScalefactor += SCALEFACTOR32; } else { switch (length) { case 16: fft_16(pInput); *pScalefactor += SCALEFACTOR16; break; case 8: fft_8(pInput); *pScalefactor += SCALEFACTOR8; break; case 2: fft2(pInput); *pScalefactor += SCALEFACTOR2; break; case 3: fft3(pInput); *pScalefactor += SCALEFACTOR3; break; case 4: fft_4(pInput); *pScalefactor += SCALEFACTOR4; break; case 5: fft5(pInput); *pScalefactor += SCALEFACTOR5; break; case 6: fft6(pInput); *pScalefactor += SCALEFACTOR6; break; case 10: fft10(pInput); *pScalefactor += SCALEFACTOR10; break; case 12: fft12(pInput); *pScalefactor += SCALEFACTOR12; break; case 15: fft15(pInput); *pScalefactor += SCALEFACTOR15; break; case 20: fft20(pInput); *pScalefactor += SCALEFACTOR20; break; case 24: fft24(pInput); *pScalefactor += SCALEFACTOR24; break; case 48: fft48(pInput); *pScalefactor += SCALEFACTOR48; break; case 60: fft60(pInput); *pScalefactor += SCALEFACTOR60; break; case 64: dit_fft(pInput, 6, SineTable512, 512); *pScalefactor += SCALEFACTOR64; break; case 80: fft80(pInput); *pScalefactor += SCALEFACTOR80; break; case 96: fft96(pInput); *pScalefactor += SCALEFACTOR96; break; case 120: fft120(pInput); *pScalefactor += SCALEFACTOR120; break; case 128: dit_fft(pInput, 7, SineTable512, 512); *pScalefactor += SCALEFACTOR128; break; case 192: fft192(pInput); *pScalefactor += SCALEFACTOR192; break; case 240: fft240(pInput); *pScalefactor += SCALEFACTOR240; break; case 256: dit_fft(pInput, 8, SineTable512, 512); *pScalefactor += SCALEFACTOR256; break; case 384: fft384(pInput); *pScalefactor += SCALEFACTOR384; break; case 480: fft480(pInput); *pScalefactor += SCALEFACTOR480; break; case 512: dit_fft(pInput, 9, SineTable512, 512); *pScalefactor += SCALEFACTOR512; break; default: FDK_ASSERT(0); /* FFT length not supported! */ break; } } } void ifft(int length, FIXP_DBL *pInput, INT *scalefactor) { switch (length) { default: FDK_ASSERT(0); /* IFFT length not supported! */ break; } } fdk-aac-2.0.2/libFDK/src/fft_rad2.cpp000066400000000000000000000235621404225310700170640ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): M. Lohwasser, M. Gayer Description: *******************************************************************************/ #include "fft_rad2.h" #include "scramble.h" #define __FFT_RAD2_CPP__ #if defined(__arm__) #include "arm/fft_rad2_arm.cpp" #elif defined(__GNUC__) && defined(__mips__) && defined(__mips_dsp) && !defined(__mips16) #include "mips/fft_rad2_mips.cpp" #endif /***************************************************************************** functionname: dit_fft (analysis) description: dit-tukey-algorithm scrambles data at entry i.e. loop is made with scrambled data returns: input: output: *****************************************************************************/ #ifndef FUNCTION_dit_fft void dit_fft(FIXP_DBL *x, const INT ldn, const FIXP_STP *trigdata, const INT trigDataSize) { const INT n = 1 << ldn; INT trigstep, i, ldm; C_ALLOC_ALIGNED_CHECK(x); scramble(x, n); /* * 1+2 stage radix 4 */ for (i = 0; i < n * 2; i += 8) { FIXP_DBL a00, a10, a20, a30; a00 = (x[i + 0] + x[i + 2]) >> 1; /* Re A + Re B */ a10 = (x[i + 4] + x[i + 6]) >> 1; /* Re C + Re D */ a20 = (x[i + 1] + x[i + 3]) >> 1; /* Im A + Im B */ a30 = (x[i + 5] + x[i + 7]) >> 1; /* Im C + Im D */ x[i + 0] = a00 + a10; /* Re A' = Re A + Re B + Re C + Re D */ x[i + 4] = a00 - a10; /* Re C' = Re A + Re B - Re C - Re D */ x[i + 1] = a20 + a30; /* Im A' = Im A + Im B + Im C + Im D */ x[i + 5] = a20 - a30; /* Im C' = Im A + Im B - Im C - Im D */ a00 = a00 - x[i + 2]; /* Re A - Re B */ a10 = a10 - x[i + 6]; /* Re C - Re D */ a20 = a20 - x[i + 3]; /* Im A - Im B */ a30 = a30 - x[i + 7]; /* Im C - Im D */ x[i + 2] = a00 + a30; /* Re B' = Re A - Re B + Im C - Im D */ x[i + 6] = a00 - a30; /* Re D' = Re A - Re B - Im C + Im D */ x[i + 3] = a20 - a10; /* Im B' = Im A - Im B - Re C + Re D */ x[i + 7] = a20 + a10; /* Im D' = Im A - Im B + Re C - Re D */ } for (ldm = 3; ldm <= ldn; ++ldm) { INT m = (1 << ldm); INT mh = (m >> 1); INT j, r; trigstep = ((trigDataSize << 2) >> ldm); FDK_ASSERT(trigstep > 0); /* Do first iteration with c=1.0 and s=0.0 separately to avoid loosing to much precision. Beware: The impact on the overal FFT precision is rather large. */ { /* block 1 */ j = 0; for (r = 0; r < n; r += m) { INT t1 = (r + j) << 1; INT t2 = t1 + (mh << 1); FIXP_DBL vr, vi, ur, ui; // cplxMultDiv2(&vi, &vr, x[t2+1], x[t2], (FIXP_SGL)1.0, (FIXP_SGL)0.0); vi = x[t2 + 1] >> 1; vr = x[t2] >> 1; ur = x[t1] >> 1; ui = x[t1 + 1] >> 1; x[t1] = ur + vr; x[t1 + 1] = ui + vi; x[t2] = ur - vr; x[t2 + 1] = ui - vi; t1 += mh; t2 = t1 + (mh << 1); // cplxMultDiv2(&vr, &vi, x[t2+1], x[t2], (FIXP_SGL)1.0, (FIXP_SGL)0.0); vr = x[t2 + 1] >> 1; vi = x[t2] >> 1; ur = x[t1] >> 1; ui = x[t1 + 1] >> 1; x[t1] = ur + vr; x[t1 + 1] = ui - vi; x[t2] = ur - vr; x[t2 + 1] = ui + vi; } } /* end of block 1 */ for (j = 1; j < mh / 4; ++j) { FIXP_STP cs; cs = trigdata[j * trigstep]; for (r = 0; r < n; r += m) { INT t1 = (r + j) << 1; INT t2 = t1 + (mh << 1); FIXP_DBL vr, vi, ur, ui; cplxMultDiv2(&vi, &vr, x[t2 + 1], x[t2], cs); ur = x[t1] >> 1; ui = x[t1 + 1] >> 1; x[t1] = ur + vr; x[t1 + 1] = ui + vi; x[t2] = ur - vr; x[t2 + 1] = ui - vi; t1 += mh; t2 = t1 + (mh << 1); cplxMultDiv2(&vr, &vi, x[t2 + 1], x[t2], cs); ur = x[t1] >> 1; ui = x[t1 + 1] >> 1; x[t1] = ur + vr; x[t1 + 1] = ui - vi; x[t2] = ur - vr; x[t2 + 1] = ui + vi; /* Same as above but for t1,t2 with j>mh/4 and thus cs swapped */ t1 = (r + mh / 2 - j) << 1; t2 = t1 + (mh << 1); cplxMultDiv2(&vi, &vr, x[t2], x[t2 + 1], cs); ur = x[t1] >> 1; ui = x[t1 + 1] >> 1; x[t1] = ur + vr; x[t1 + 1] = ui - vi; x[t2] = ur - vr; x[t2 + 1] = ui + vi; t1 += mh; t2 = t1 + (mh << 1); cplxMultDiv2(&vr, &vi, x[t2], x[t2 + 1], cs); ur = x[t1] >> 1; ui = x[t1 + 1] >> 1; x[t1] = ur - vr; x[t1 + 1] = ui - vi; x[t2] = ur + vr; x[t2 + 1] = ui + vi; } } { /* block 2 */ j = mh / 4; for (r = 0; r < n; r += m) { INT t1 = (r + j) << 1; INT t2 = t1 + (mh << 1); FIXP_DBL vr, vi, ur, ui; cplxMultDiv2(&vi, &vr, x[t2 + 1], x[t2], STC(0x5a82799a), STC(0x5a82799a)); ur = x[t1] >> 1; ui = x[t1 + 1] >> 1; x[t1] = ur + vr; x[t1 + 1] = ui + vi; x[t2] = ur - vr; x[t2 + 1] = ui - vi; t1 += mh; t2 = t1 + (mh << 1); cplxMultDiv2(&vr, &vi, x[t2 + 1], x[t2], STC(0x5a82799a), STC(0x5a82799a)); ur = x[t1] >> 1; ui = x[t1 + 1] >> 1; x[t1] = ur + vr; x[t1 + 1] = ui - vi; x[t2] = ur - vr; x[t2 + 1] = ui + vi; } } /* end of block 2 */ } } #endif fdk-aac-2.0.2/libFDK/src/fixpoint_math.cpp000066400000000000000000000774211404225310700202510ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): M. Gayer Description: Fixed point specific mathematical functions *******************************************************************************/ #include "fixpoint_math.h" /* * Hardware specific implementations */ /* * Fallback implementations */ /***************************************************************************** functionname: LdDataVector *****************************************************************************/ LNK_SECTION_CODE_L1 void LdDataVector(FIXP_DBL *srcVector, FIXP_DBL *destVector, INT n) { INT i; for (i = 0; i < n; i++) { destVector[i] = fLog2(srcVector[i], 0); } } #define MAX_POW2_PRECISION 8 #ifndef SINETABLE_16BIT #define POW2_PRECISION MAX_POW2_PRECISION #else #define POW2_PRECISION 5 #endif /* Taylor series coefficients of the function x^2. The first coefficient is ommited (equal to 1.0). pow2Coeff[i-1] = (1/i!) d^i(2^x)/dx^i, i=1..MAX_POW2_PRECISION To evaluate the taylor series around x = 0, the coefficients are: 1/!i * ln(2)^i */ #ifndef POW2COEFF_16BIT RAM_ALIGN LNK_SECTION_CONSTDATA_L1 static const FIXP_DBL pow2Coeff[MAX_POW2_PRECISION] = { FL2FXCONST_DBL(0.693147180559945309417232121458177), /* ln(2)^1 /1! */ FL2FXCONST_DBL(0.240226506959100712333551263163332), /* ln(2)^2 /2! */ FL2FXCONST_DBL(0.0555041086648215799531422637686218), /* ln(2)^3 /3! */ FL2FXCONST_DBL(0.00961812910762847716197907157365887), /* ln(2)^4 /4! */ FL2FXCONST_DBL(0.00133335581464284434234122219879962), /* ln(2)^5 /5! */ FL2FXCONST_DBL(1.54035303933816099544370973327423e-4), /* ln(2)^6 /6! */ FL2FXCONST_DBL(1.52527338040598402800254390120096e-5), /* ln(2)^7 /7! */ FL2FXCONST_DBL(1.32154867901443094884037582282884e-6) /* ln(2)^8 /8! */ }; #else RAM_ALIGN LNK_SECTION_CONSTDATA_L1 static const FIXP_SGL pow2Coeff[MAX_POW2_PRECISION] = { FL2FXCONST_SGL(0.693147180559945309417232121458177), /* ln(2)^1 /1! */ FL2FXCONST_SGL(0.240226506959100712333551263163332), /* ln(2)^2 /2! */ FL2FXCONST_SGL(0.0555041086648215799531422637686218), /* ln(2)^3 /3! */ FL2FXCONST_SGL(0.00961812910762847716197907157365887), /* ln(2)^4 /4! */ FL2FXCONST_SGL(0.00133335581464284434234122219879962), /* ln(2)^5 /5! */ FL2FXCONST_SGL(1.54035303933816099544370973327423e-4), /* ln(2)^6 /6! */ FL2FXCONST_SGL(1.52527338040598402800254390120096e-5), /* ln(2)^7 /7! */ FL2FXCONST_SGL(1.32154867901443094884037582282884e-6) /* ln(2)^8 /8! */ }; #endif /***************************************************************************** functionname: CalcInvLdData description: Delivers the inverse of function CalcLdData(). Delivers 2^(op*LD_DATA_SCALING) input: Input op is assumed to be fractional -1.0 < op < 1.0 output: For op == 0, the result is MAXVAL_DBL (almost 1.0). For negative input values the output should be treated as a positive fractional value. For positive input values the output should be treated as a positive integer value. This function does not output negative values. *****************************************************************************/ /* Date: 06-JULY-2012 Arthur Tritthart, IIS Fraunhofer Erlangen */ /* Version with 3 table lookup and 1 linear interpolations */ /* Algorithm: compute power of 2, argument x is in Q7.25 format */ /* result = 2^(x/64) */ /* We split exponent (x/64) into 5 components: */ /* integer part: represented by b31..b25 (exp) */ /* fractional part 1: represented by b24..b20 (lookup1) */ /* fractional part 2: represented by b19..b15 (lookup2) */ /* fractional part 3: represented by b14..b10 (lookup3) */ /* fractional part 4: represented by b09..b00 (frac) */ /* => result = (lookup1*lookup2*(lookup3+C1*frac)<<3)>>exp */ /* Due to the fact, that all lookup values contain a factor 0.5 */ /* the result has to be shifted by 3 to the right also. */ /* Table exp2_tab_long contains the log2 for 0 to 1.0 in steps */ /* of 1/32, table exp2w_tab_long the log2 for 0 to 1/32 in steps*/ /* of 1/1024, table exp2x_tab_long the log2 for 0 to 1/1024 in */ /* steps of 1/32768. Since the 2-logarithm of very very small */ /* negative value is rather linear, we can use interpolation. */ /* Limitations: */ /* For x <= 0, the result is fractional positive */ /* For x > 0, the result is integer in range 1...7FFF.FFFF */ /* For x < -31/64, we have to clear the result */ /* For x = 0, the result is ~1.0 (0x7FFF.FFFF) */ /* For x >= 31/64, the result is 0x7FFF.FFFF */ /* This table is used for lookup 2^x with */ /* x in range [0...1.0[ in steps of 1/32 */ LNK_SECTION_DATA_L1 const UINT exp2_tab_long[32] = { 0x40000000, 0x4166C34C, 0x42D561B4, 0x444C0740, 0x45CAE0F2, 0x47521CC6, 0x48E1E9BA, 0x4A7A77D4, 0x4C1BF829, 0x4DC69CDD, 0x4F7A9930, 0x51382182, 0x52FF6B55, 0x54D0AD5A, 0x56AC1F75, 0x5891FAC1, 0x5A82799A, 0x5C7DD7A4, 0x5E8451D0, 0x60962665, 0x62B39509, 0x64DCDEC3, 0x6712460B, 0x69540EC9, 0x6BA27E65, 0x6DFDDBCC, 0x70666F76, 0x72DC8374, 0x75606374, 0x77F25CCE, 0x7A92BE8B, 0x7D41D96E // 0x80000000 }; /* This table is used for lookup 2^x with */ /* x in range [0...1/32[ in steps of 1/1024 */ LNK_SECTION_DATA_L1 const UINT exp2w_tab_long[32] = { 0x40000000, 0x400B1818, 0x4016321B, 0x40214E0C, 0x402C6BE9, 0x40378BB4, 0x4042AD6D, 0x404DD113, 0x4058F6A8, 0x40641E2B, 0x406F479E, 0x407A7300, 0x4085A051, 0x4090CF92, 0x409C00C4, 0x40A733E6, 0x40B268FA, 0x40BD9FFF, 0x40C8D8F5, 0x40D413DD, 0x40DF50B8, 0x40EA8F86, 0x40F5D046, 0x410112FA, 0x410C57A2, 0x41179E3D, 0x4122E6CD, 0x412E3152, 0x41397DCC, 0x4144CC3B, 0x41501CA0, 0x415B6EFB, // 0x4166C34C, }; /* This table is used for lookup 2^x with */ /* x in range [0...1/1024[ in steps of 1/32768 */ LNK_SECTION_DATA_L1 const UINT exp2x_tab_long[32] = { 0x40000000, 0x400058B9, 0x4000B173, 0x40010A2D, 0x400162E8, 0x4001BBA3, 0x4002145F, 0x40026D1B, 0x4002C5D8, 0x40031E95, 0x40037752, 0x4003D011, 0x400428CF, 0x4004818E, 0x4004DA4E, 0x4005330E, 0x40058BCE, 0x4005E48F, 0x40063D51, 0x40069613, 0x4006EED5, 0x40074798, 0x4007A05B, 0x4007F91F, 0x400851E4, 0x4008AAA8, 0x4009036E, 0x40095C33, 0x4009B4FA, 0x400A0DC0, 0x400A6688, 0x400ABF4F, // 0x400B1818 }; /***************************************************************************** functionname: InitLdInt and CalcLdInt description: Create and access table with integer LdData (0 to LD_INT_TAB_LEN) *****************************************************************************/ #ifndef LD_INT_TAB_LEN #define LD_INT_TAB_LEN \ 193 /* Default tab length. Lower value should be set in fix.h */ #endif #if (LD_INT_TAB_LEN <= 120) LNK_SECTION_CONSTDATA_L1 static const FIXP_DBL ldIntCoeff[] = { (FIXP_DBL)0x80000001, (FIXP_DBL)0x00000000, (FIXP_DBL)0x02000000, (FIXP_DBL)0x032b8034, (FIXP_DBL)0x04000000, (FIXP_DBL)0x04a4d3c2, (FIXP_DBL)0x052b8034, (FIXP_DBL)0x059d5da0, (FIXP_DBL)0x06000000, (FIXP_DBL)0x06570069, (FIXP_DBL)0x06a4d3c2, (FIXP_DBL)0x06eb3a9f, (FIXP_DBL)0x072b8034, (FIXP_DBL)0x0766a009, (FIXP_DBL)0x079d5da0, (FIXP_DBL)0x07d053f7, (FIXP_DBL)0x08000000, (FIXP_DBL)0x082cc7ee, (FIXP_DBL)0x08570069, (FIXP_DBL)0x087ef05b, (FIXP_DBL)0x08a4d3c2, (FIXP_DBL)0x08c8ddd4, (FIXP_DBL)0x08eb3a9f, (FIXP_DBL)0x090c1050, (FIXP_DBL)0x092b8034, (FIXP_DBL)0x0949a785, (FIXP_DBL)0x0966a009, (FIXP_DBL)0x0982809d, (FIXP_DBL)0x099d5da0, (FIXP_DBL)0x09b74949, (FIXP_DBL)0x09d053f7, (FIXP_DBL)0x09e88c6b, (FIXP_DBL)0x0a000000, (FIXP_DBL)0x0a16bad3, (FIXP_DBL)0x0a2cc7ee, (FIXP_DBL)0x0a423162, (FIXP_DBL)0x0a570069, (FIXP_DBL)0x0a6b3d79, (FIXP_DBL)0x0a7ef05b, (FIXP_DBL)0x0a92203d, (FIXP_DBL)0x0aa4d3c2, (FIXP_DBL)0x0ab7110e, (FIXP_DBL)0x0ac8ddd4, (FIXP_DBL)0x0ada3f60, (FIXP_DBL)0x0aeb3a9f, (FIXP_DBL)0x0afbd42b, (FIXP_DBL)0x0b0c1050, (FIXP_DBL)0x0b1bf312, (FIXP_DBL)0x0b2b8034, (FIXP_DBL)0x0b3abb40, (FIXP_DBL)0x0b49a785, (FIXP_DBL)0x0b584822, (FIXP_DBL)0x0b66a009, (FIXP_DBL)0x0b74b1fd, (FIXP_DBL)0x0b82809d, (FIXP_DBL)0x0b900e61, (FIXP_DBL)0x0b9d5da0, (FIXP_DBL)0x0baa708f, (FIXP_DBL)0x0bb74949, (FIXP_DBL)0x0bc3e9ca, (FIXP_DBL)0x0bd053f7, (FIXP_DBL)0x0bdc899b, (FIXP_DBL)0x0be88c6b, (FIXP_DBL)0x0bf45e09, (FIXP_DBL)0x0c000000, (FIXP_DBL)0x0c0b73cb, (FIXP_DBL)0x0c16bad3, (FIXP_DBL)0x0c21d671, (FIXP_DBL)0x0c2cc7ee, (FIXP_DBL)0x0c379085, (FIXP_DBL)0x0c423162, (FIXP_DBL)0x0c4caba8, (FIXP_DBL)0x0c570069, (FIXP_DBL)0x0c6130af, (FIXP_DBL)0x0c6b3d79, (FIXP_DBL)0x0c7527b9, (FIXP_DBL)0x0c7ef05b, (FIXP_DBL)0x0c88983f, (FIXP_DBL)0x0c92203d, (FIXP_DBL)0x0c9b8926, (FIXP_DBL)0x0ca4d3c2, (FIXP_DBL)0x0cae00d2, (FIXP_DBL)0x0cb7110e, (FIXP_DBL)0x0cc0052b, (FIXP_DBL)0x0cc8ddd4, (FIXP_DBL)0x0cd19bb0, (FIXP_DBL)0x0cda3f60, (FIXP_DBL)0x0ce2c97d, (FIXP_DBL)0x0ceb3a9f, (FIXP_DBL)0x0cf39355, (FIXP_DBL)0x0cfbd42b, (FIXP_DBL)0x0d03fda9, (FIXP_DBL)0x0d0c1050, (FIXP_DBL)0x0d140ca0, (FIXP_DBL)0x0d1bf312, (FIXP_DBL)0x0d23c41d, (FIXP_DBL)0x0d2b8034, (FIXP_DBL)0x0d3327c7, (FIXP_DBL)0x0d3abb40, (FIXP_DBL)0x0d423b08, (FIXP_DBL)0x0d49a785, (FIXP_DBL)0x0d510118, (FIXP_DBL)0x0d584822, (FIXP_DBL)0x0d5f7cff, (FIXP_DBL)0x0d66a009, (FIXP_DBL)0x0d6db197, (FIXP_DBL)0x0d74b1fd, (FIXP_DBL)0x0d7ba190, (FIXP_DBL)0x0d82809d, (FIXP_DBL)0x0d894f75, (FIXP_DBL)0x0d900e61, (FIXP_DBL)0x0d96bdad, (FIXP_DBL)0x0d9d5da0, (FIXP_DBL)0x0da3ee7f, (FIXP_DBL)0x0daa708f, (FIXP_DBL)0x0db0e412, (FIXP_DBL)0x0db74949, (FIXP_DBL)0x0dbda072, (FIXP_DBL)0x0dc3e9ca, (FIXP_DBL)0x0dca258e}; #elif (LD_INT_TAB_LEN <= 193) LNK_SECTION_CONSTDATA_L1 static const FIXP_DBL ldIntCoeff[] = { (FIXP_DBL)0x80000001, (FIXP_DBL)0x00000000, (FIXP_DBL)0x02000000, (FIXP_DBL)0x032b8034, (FIXP_DBL)0x04000000, (FIXP_DBL)0x04a4d3c2, (FIXP_DBL)0x052b8034, (FIXP_DBL)0x059d5da0, (FIXP_DBL)0x06000000, (FIXP_DBL)0x06570069, (FIXP_DBL)0x06a4d3c2, (FIXP_DBL)0x06eb3a9f, (FIXP_DBL)0x072b8034, (FIXP_DBL)0x0766a009, (FIXP_DBL)0x079d5da0, (FIXP_DBL)0x07d053f7, (FIXP_DBL)0x08000000, (FIXP_DBL)0x082cc7ee, (FIXP_DBL)0x08570069, (FIXP_DBL)0x087ef05b, (FIXP_DBL)0x08a4d3c2, (FIXP_DBL)0x08c8ddd4, (FIXP_DBL)0x08eb3a9f, (FIXP_DBL)0x090c1050, (FIXP_DBL)0x092b8034, (FIXP_DBL)0x0949a785, (FIXP_DBL)0x0966a009, (FIXP_DBL)0x0982809d, (FIXP_DBL)0x099d5da0, (FIXP_DBL)0x09b74949, (FIXP_DBL)0x09d053f7, (FIXP_DBL)0x09e88c6b, (FIXP_DBL)0x0a000000, (FIXP_DBL)0x0a16bad3, (FIXP_DBL)0x0a2cc7ee, (FIXP_DBL)0x0a423162, (FIXP_DBL)0x0a570069, (FIXP_DBL)0x0a6b3d79, (FIXP_DBL)0x0a7ef05b, (FIXP_DBL)0x0a92203d, (FIXP_DBL)0x0aa4d3c2, (FIXP_DBL)0x0ab7110e, (FIXP_DBL)0x0ac8ddd4, (FIXP_DBL)0x0ada3f60, (FIXP_DBL)0x0aeb3a9f, (FIXP_DBL)0x0afbd42b, (FIXP_DBL)0x0b0c1050, (FIXP_DBL)0x0b1bf312, (FIXP_DBL)0x0b2b8034, (FIXP_DBL)0x0b3abb40, (FIXP_DBL)0x0b49a785, (FIXP_DBL)0x0b584822, (FIXP_DBL)0x0b66a009, (FIXP_DBL)0x0b74b1fd, (FIXP_DBL)0x0b82809d, (FIXP_DBL)0x0b900e61, (FIXP_DBL)0x0b9d5da0, (FIXP_DBL)0x0baa708f, (FIXP_DBL)0x0bb74949, (FIXP_DBL)0x0bc3e9ca, (FIXP_DBL)0x0bd053f7, (FIXP_DBL)0x0bdc899b, (FIXP_DBL)0x0be88c6b, (FIXP_DBL)0x0bf45e09, (FIXP_DBL)0x0c000000, (FIXP_DBL)0x0c0b73cb, (FIXP_DBL)0x0c16bad3, (FIXP_DBL)0x0c21d671, (FIXP_DBL)0x0c2cc7ee, (FIXP_DBL)0x0c379085, (FIXP_DBL)0x0c423162, (FIXP_DBL)0x0c4caba8, (FIXP_DBL)0x0c570069, (FIXP_DBL)0x0c6130af, (FIXP_DBL)0x0c6b3d79, (FIXP_DBL)0x0c7527b9, (FIXP_DBL)0x0c7ef05b, (FIXP_DBL)0x0c88983f, (FIXP_DBL)0x0c92203d, (FIXP_DBL)0x0c9b8926, (FIXP_DBL)0x0ca4d3c2, (FIXP_DBL)0x0cae00d2, (FIXP_DBL)0x0cb7110e, (FIXP_DBL)0x0cc0052b, (FIXP_DBL)0x0cc8ddd4, (FIXP_DBL)0x0cd19bb0, (FIXP_DBL)0x0cda3f60, (FIXP_DBL)0x0ce2c97d, (FIXP_DBL)0x0ceb3a9f, (FIXP_DBL)0x0cf39355, (FIXP_DBL)0x0cfbd42b, (FIXP_DBL)0x0d03fda9, (FIXP_DBL)0x0d0c1050, (FIXP_DBL)0x0d140ca0, (FIXP_DBL)0x0d1bf312, (FIXP_DBL)0x0d23c41d, (FIXP_DBL)0x0d2b8034, (FIXP_DBL)0x0d3327c7, (FIXP_DBL)0x0d3abb40, (FIXP_DBL)0x0d423b08, (FIXP_DBL)0x0d49a785, (FIXP_DBL)0x0d510118, (FIXP_DBL)0x0d584822, (FIXP_DBL)0x0d5f7cff, (FIXP_DBL)0x0d66a009, (FIXP_DBL)0x0d6db197, (FIXP_DBL)0x0d74b1fd, (FIXP_DBL)0x0d7ba190, (FIXP_DBL)0x0d82809d, (FIXP_DBL)0x0d894f75, (FIXP_DBL)0x0d900e61, (FIXP_DBL)0x0d96bdad, (FIXP_DBL)0x0d9d5da0, (FIXP_DBL)0x0da3ee7f, (FIXP_DBL)0x0daa708f, (FIXP_DBL)0x0db0e412, (FIXP_DBL)0x0db74949, (FIXP_DBL)0x0dbda072, (FIXP_DBL)0x0dc3e9ca, (FIXP_DBL)0x0dca258e, (FIXP_DBL)0x0dd053f7, (FIXP_DBL)0x0dd6753e, (FIXP_DBL)0x0ddc899b, (FIXP_DBL)0x0de29143, (FIXP_DBL)0x0de88c6b, (FIXP_DBL)0x0dee7b47, (FIXP_DBL)0x0df45e09, (FIXP_DBL)0x0dfa34e1, (FIXP_DBL)0x0e000000, (FIXP_DBL)0x0e05bf94, (FIXP_DBL)0x0e0b73cb, (FIXP_DBL)0x0e111cd2, (FIXP_DBL)0x0e16bad3, (FIXP_DBL)0x0e1c4dfb, (FIXP_DBL)0x0e21d671, (FIXP_DBL)0x0e275460, (FIXP_DBL)0x0e2cc7ee, (FIXP_DBL)0x0e323143, (FIXP_DBL)0x0e379085, (FIXP_DBL)0x0e3ce5d8, (FIXP_DBL)0x0e423162, (FIXP_DBL)0x0e477346, (FIXP_DBL)0x0e4caba8, (FIXP_DBL)0x0e51daa8, (FIXP_DBL)0x0e570069, (FIXP_DBL)0x0e5c1d0b, (FIXP_DBL)0x0e6130af, (FIXP_DBL)0x0e663b74, (FIXP_DBL)0x0e6b3d79, (FIXP_DBL)0x0e7036db, (FIXP_DBL)0x0e7527b9, (FIXP_DBL)0x0e7a1030, (FIXP_DBL)0x0e7ef05b, (FIXP_DBL)0x0e83c857, (FIXP_DBL)0x0e88983f, (FIXP_DBL)0x0e8d602e, (FIXP_DBL)0x0e92203d, (FIXP_DBL)0x0e96d888, (FIXP_DBL)0x0e9b8926, (FIXP_DBL)0x0ea03232, (FIXP_DBL)0x0ea4d3c2, (FIXP_DBL)0x0ea96df0, (FIXP_DBL)0x0eae00d2, (FIXP_DBL)0x0eb28c7f, (FIXP_DBL)0x0eb7110e, (FIXP_DBL)0x0ebb8e96, (FIXP_DBL)0x0ec0052b, (FIXP_DBL)0x0ec474e4, (FIXP_DBL)0x0ec8ddd4, (FIXP_DBL)0x0ecd4012, (FIXP_DBL)0x0ed19bb0, (FIXP_DBL)0x0ed5f0c4, (FIXP_DBL)0x0eda3f60, (FIXP_DBL)0x0ede8797, (FIXP_DBL)0x0ee2c97d, (FIXP_DBL)0x0ee70525, (FIXP_DBL)0x0eeb3a9f, (FIXP_DBL)0x0eef69ff, (FIXP_DBL)0x0ef39355, (FIXP_DBL)0x0ef7b6b4, (FIXP_DBL)0x0efbd42b, (FIXP_DBL)0x0effebcd, (FIXP_DBL)0x0f03fda9, (FIXP_DBL)0x0f0809cf, (FIXP_DBL)0x0f0c1050, (FIXP_DBL)0x0f10113b, (FIXP_DBL)0x0f140ca0, (FIXP_DBL)0x0f18028d, (FIXP_DBL)0x0f1bf312, (FIXP_DBL)0x0f1fde3d, (FIXP_DBL)0x0f23c41d, (FIXP_DBL)0x0f27a4c0, (FIXP_DBL)0x0f2b8034}; #else #error "ldInt table size too small" #endif LNK_SECTION_INITCODE void InitLdInt() { /* nothing to do! Use preinitialized logarithm table */ } #if (LD_INT_TAB_LEN != 0) LNK_SECTION_CODE_L1 FIXP_DBL CalcLdInt(INT i) { /* calculates ld(op)/LD_DATA_SCALING */ /* op is assumed to be an integer value between 1 and LD_INT_TAB_LEN */ FDK_ASSERT((LD_INT_TAB_LEN > 0) && ((FIXP_DBL)ldIntCoeff[0] == (FIXP_DBL)0x80000001)); /* tab has to be initialized */ if ((i > 0) && (i < LD_INT_TAB_LEN)) return ldIntCoeff[i]; else { return (0); } } #endif /* (LD_INT_TAB_LEN!=0) */ #if !defined(FUNCTION_schur_div) /***************************************************************************** functionname: schur_div description: delivers op1/op2 with op3-bit accuracy *****************************************************************************/ FIXP_DBL schur_div(FIXP_DBL num, FIXP_DBL denum, INT count) { INT L_num = (LONG)num >> 1; INT L_denum = (LONG)denum >> 1; INT div = 0; INT k = count; FDK_ASSERT(num >= (FIXP_DBL)0); FDK_ASSERT(denum > (FIXP_DBL)0); FDK_ASSERT(num <= denum); if (L_num != 0) while (--k) { div <<= 1; L_num <<= 1; if (L_num >= L_denum) { L_num -= L_denum; div++; } } return (FIXP_DBL)(div << (DFRACT_BITS - count)); } #endif /* !defined(FUNCTION_schur_div) */ #ifndef FUNCTION_fMultNorm FIXP_DBL fMultNorm(FIXP_DBL f1, FIXP_DBL f2, INT *result_e) { INT product = 0; INT norm_f1, norm_f2; if ((f1 == (FIXP_DBL)0) || (f2 == (FIXP_DBL)0)) { *result_e = 0; return (FIXP_DBL)0; } norm_f1 = CountLeadingBits(f1); f1 = f1 << norm_f1; norm_f2 = CountLeadingBits(f2); f2 = f2 << norm_f2; if ((f1 == (FIXP_DBL)MINVAL_DBL) && (f2 == (FIXP_DBL)MINVAL_DBL)) { product = -((FIXP_DBL)MINVAL_DBL >> 1); *result_e = -(norm_f1 + norm_f2 - 1); } else { product = fMult(f1, f2); *result_e = -(norm_f1 + norm_f2); } return (FIXP_DBL)product; } #endif #ifndef FUNCTION_fDivNorm FIXP_DBL fDivNorm(FIXP_DBL L_num, FIXP_DBL L_denum, INT *result_e) { FIXP_DBL div; INT norm_num, norm_den; FDK_ASSERT(L_num >= (FIXP_DBL)0); FDK_ASSERT(L_denum > (FIXP_DBL)0); if (L_num == (FIXP_DBL)0) { *result_e = 0; return ((FIXP_DBL)0); } norm_num = CountLeadingBits(L_num); L_num = L_num << norm_num; L_num = L_num >> 1; *result_e = -norm_num + 1; norm_den = CountLeadingBits(L_denum); L_denum = L_denum << norm_den; *result_e -= -norm_den; div = schur_div(L_num, L_denum, FRACT_BITS); return div; } #endif /* !FUNCTION_fDivNorm */ #ifndef FUNCTION_fDivNorm FIXP_DBL fDivNorm(FIXP_DBL num, FIXP_DBL denom) { INT e; FIXP_DBL res; FDK_ASSERT(denom >= num); res = fDivNorm(num, denom, &e); /* Avoid overflow since we must output a value with exponent 0 there is no other choice than saturating to almost 1.0f */ if (res == (FIXP_DBL)(1 << (DFRACT_BITS - 2)) && e == 1) { res = (FIXP_DBL)MAXVAL_DBL; } else { res = scaleValue(res, e); } return res; } #endif /* !FUNCTION_fDivNorm */ #ifndef FUNCTION_fDivNormSigned FIXP_DBL fDivNormSigned(FIXP_DBL num, FIXP_DBL denom) { INT e; FIXP_DBL res; int sign; if (denom == (FIXP_DBL)0) { return (FIXP_DBL)MAXVAL_DBL; } sign = ((num >= (FIXP_DBL)0) != (denom >= (FIXP_DBL)0)); res = fDivNormSigned(num, denom, &e); /* Saturate since we must output a value with exponent 0 */ if ((e > 0) && (fAbs(res) >= FL2FXCONST_DBL(0.5))) { if (sign) { res = (FIXP_DBL)MINVAL_DBL; } else { res = (FIXP_DBL)MAXVAL_DBL; } } else { res = scaleValue(res, e); } return res; } FIXP_DBL fDivNormSigned(FIXP_DBL L_num, FIXP_DBL L_denum, INT *result_e) { FIXP_DBL div; INT norm_num, norm_den; int sign; sign = ((L_num >= (FIXP_DBL)0) != (L_denum >= (FIXP_DBL)0)); if (L_num == (FIXP_DBL)0) { *result_e = 0; return ((FIXP_DBL)0); } if (L_denum == (FIXP_DBL)0) { *result_e = 14; return ((FIXP_DBL)MAXVAL_DBL); } norm_num = CountLeadingBits(L_num); L_num = L_num << norm_num; L_num = L_num >> 2; L_num = fAbs(L_num); *result_e = -norm_num + 1; norm_den = CountLeadingBits(L_denum); L_denum = L_denum << norm_den; L_denum = L_denum >> 1; L_denum = fAbs(L_denum); *result_e -= -norm_den; div = schur_div(L_num, L_denum, FRACT_BITS); if (sign) { div = -div; } return div; } #endif /* FUNCTION_fDivNormSigned */ #ifndef FUNCTION_fDivNormHighPrec FIXP_DBL fDivNormHighPrec(FIXP_DBL num, FIXP_DBL denom, INT *result_e) { FIXP_DBL div; INT norm_num, norm_den; FDK_ASSERT(num >= (FIXP_DBL)0); FDK_ASSERT(denom > (FIXP_DBL)0); if (num == (FIXP_DBL)0) { *result_e = 0; return ((FIXP_DBL)0); } norm_num = CountLeadingBits(num); num = num << norm_num; num = num >> 1; *result_e = -norm_num + 1; norm_den = CountLeadingBits(denom); denom = denom << norm_den; *result_e -= -norm_den; div = schur_div(num, denom, 31); return div; } #endif /* !FUNCTION_fDivNormHighPrec */ #ifndef FUNCTION_fPow FIXP_DBL f2Pow(const FIXP_DBL exp_m, const INT exp_e, INT *result_e) { FIXP_DBL frac_part, result_m; INT int_part; if (exp_e > 0) { INT exp_bits = DFRACT_BITS - 1 - exp_e; int_part = exp_m >> exp_bits; frac_part = exp_m - (FIXP_DBL)(int_part << exp_bits); frac_part = frac_part << exp_e; } else { int_part = 0; frac_part = exp_m >> -exp_e; } /* Best accuracy is around 0, so try to get there with the fractional part. */ if (frac_part > FL2FXCONST_DBL(0.5f)) { int_part = int_part + 1; frac_part = frac_part + FL2FXCONST_DBL(-1.0f); } if (frac_part < FL2FXCONST_DBL(-0.5f)) { int_part = int_part - 1; frac_part = -(FL2FXCONST_DBL(-1.0f) - frac_part); } /* "+ 1" compensates fMultAddDiv2() of the polynomial evaluation below. */ *result_e = int_part + 1; /* Evaluate taylor polynomial which approximates 2^x */ { FIXP_DBL p; /* result_m ~= 2^frac_part */ p = frac_part; /* First taylor series coefficient a_0 = 1.0, scaled by 0.5 due to * fMultDiv2(). */ result_m = FL2FXCONST_DBL(1.0f / 2.0f); for (INT i = 0; i < POW2_PRECISION; i++) { /* next taylor series term: a_i * x^i, x=0 */ result_m = fMultAddDiv2(result_m, pow2Coeff[i], p); p = fMult(p, frac_part); } } return result_m; } FIXP_DBL f2Pow(const FIXP_DBL exp_m, const INT exp_e) { FIXP_DBL result_m; INT result_e; result_m = f2Pow(exp_m, exp_e, &result_e); result_e = fixMin(DFRACT_BITS - 1, fixMax(-(DFRACT_BITS - 1), result_e)); return scaleValue(result_m, result_e); } FIXP_DBL fPow(FIXP_DBL base_m, INT base_e, FIXP_DBL exp_m, INT exp_e, INT *result_e) { INT ans_lg2_e, baselg2_e; FIXP_DBL base_lg2, ans_lg2, result; if (base_m <= (FIXP_DBL)0) { result = (FIXP_DBL)0; *result_e = 0; return result; } /* Calc log2 of base */ base_lg2 = fLog2(base_m, base_e, &baselg2_e); /* Prepare exp */ { INT leadingBits; leadingBits = CountLeadingBits(fAbs(exp_m)); exp_m = exp_m << leadingBits; exp_e -= leadingBits; } /* Calc base pow exp */ ans_lg2 = fMult(base_lg2, exp_m); ans_lg2_e = exp_e + baselg2_e; /* Calc antilog */ result = f2Pow(ans_lg2, ans_lg2_e, result_e); return result; } FIXP_DBL fLdPow(FIXP_DBL baseLd_m, INT baseLd_e, FIXP_DBL exp_m, INT exp_e, INT *result_e) { INT ans_lg2_e; FIXP_DBL ans_lg2, result; /* Prepare exp */ { INT leadingBits; leadingBits = CountLeadingBits(fAbs(exp_m)); exp_m = exp_m << leadingBits; exp_e -= leadingBits; } /* Calc base pow exp */ ans_lg2 = fMult(baseLd_m, exp_m); ans_lg2_e = exp_e + baseLd_e; /* Calc antilog */ result = f2Pow(ans_lg2, ans_lg2_e, result_e); return result; } FIXP_DBL fLdPow(FIXP_DBL baseLd_m, INT baseLd_e, FIXP_DBL exp_m, INT exp_e) { FIXP_DBL result_m; int result_e; result_m = fLdPow(baseLd_m, baseLd_e, exp_m, exp_e, &result_e); return SATURATE_SHIFT(result_m, -result_e, DFRACT_BITS); } FIXP_DBL fPowInt(FIXP_DBL base_m, INT base_e, INT exp, INT *pResult_e) { FIXP_DBL result; if (exp != 0) { INT result_e = 0; if (base_m != (FIXP_DBL)0) { { INT leadingBits; leadingBits = CountLeadingBits(base_m); base_m <<= leadingBits; base_e -= leadingBits; } result = base_m; { int i; for (i = 1; i < fAbs(exp); i++) { result = fMult(result, base_m); } } if (exp < 0) { /* 1.0 / ans */ result = fDivNorm(FL2FXCONST_DBL(0.5f), result, &result_e); result_e++; } else { int ansScale = CountLeadingBits(result); result <<= ansScale; result_e -= ansScale; } result_e += exp * base_e; } else { result = (FIXP_DBL)0; } *pResult_e = result_e; } else { result = FL2FXCONST_DBL(0.5f); *pResult_e = 1; } return result; } #endif /* FUNCTION_fPow */ #ifndef FUNCTION_fLog2 FIXP_DBL CalcLog2(FIXP_DBL base_m, INT base_e, INT *result_e) { return fLog2(base_m, base_e, result_e); } #endif /* FUNCTION_fLog2 */ INT fixp_floorToInt(FIXP_DBL f_inp, INT sf) { FDK_ASSERT(sf >= 0); INT floorInt = (INT)(f_inp >> ((DFRACT_BITS - 1) - sf)); return floorInt; } FIXP_DBL fixp_floor(FIXP_DBL f_inp, INT sf) { FDK_ASSERT(sf >= 0); INT floorInt = fixp_floorToInt(f_inp, sf); FIXP_DBL f_floor = (FIXP_DBL)(floorInt << ((DFRACT_BITS - 1) - sf)); return f_floor; } INT fixp_ceilToInt(FIXP_DBL f_inp, INT sf) // sf mantissaBits left of dot { FDK_ASSERT(sf >= 0); INT sx = (DFRACT_BITS - 1) - sf; // sx mantissaBits right of dot INT inpINT = (INT)f_inp; INT mask = (0x1 << sx) - 1; INT ceilInt = (INT)(f_inp >> sx); if (inpINT & mask) { ceilInt++; // increment only, if there is at least one set mantissaBit // right of dot [in inpINT] } return ceilInt; } FIXP_DBL fixp_ceil(FIXP_DBL f_inp, INT sf) { FDK_ASSERT(sf >= 0); INT sx = (DFRACT_BITS - 1) - sf; INT ceilInt = fixp_ceilToInt(f_inp, sf); ULONG mask = (ULONG)0x1 << (DFRACT_BITS - 1); // 0x80000000 ceilInt = ceilInt << sx; // no fract warn bec. shift into saturation done on int side if ((f_inp > FL2FXCONST_DBL(0.0f)) && (ceilInt & mask)) { --ceilInt; } FIXP_DBL f_ceil = (FIXP_DBL)ceilInt; return f_ceil; } /***************************************************************************** fixp_truncateToInt() Just remove the fractional part which is located right of decimal point Same as which is done when a float is casted to (INT) : result_INTtype = (INT)b_floatTypeInput; returns INT *****************************************************************************/ INT fixp_truncateToInt(FIXP_DBL f_inp, INT sf) // sf mantissaBits left of dot // (without sign) e.g. at width // 32 this would be [sign]7. // supposed sf equals 8. { FDK_ASSERT(sf >= 0); INT sx = (DFRACT_BITS - 1) - sf; // sx mantissaBits right of dot // at width 32 this would be .24 // supposed sf equals 8. INT fbaccu = (INT)f_inp; INT mask = (0x1 << sx); if ((fbaccu < 0) && (fbaccu & (mask - 1))) { fbaccu = fbaccu + mask; } fbaccu = fbaccu >> sx; return fbaccu; } /***************************************************************************** fixp_truncate() Just remove the fractional part which is located right of decimal point returns FIXP_DBL *****************************************************************************/ FIXP_DBL fixp_truncate(FIXP_DBL f_inp, INT sf) { FDK_ASSERT(sf >= 0); INT truncateInt = fixp_truncateToInt(f_inp, sf); FIXP_DBL f_truncate = (FIXP_DBL)(truncateInt << ((DFRACT_BITS - 1) - sf)); return f_truncate; } /***************************************************************************** fixp_roundToInt() round [typical rounding] See fct roundRef() [which is the reference] returns INT *****************************************************************************/ INT fixp_roundToInt(FIXP_DBL f_inp, INT sf) { FDK_ASSERT(sf >= 0); INT sx = DFRACT_BITS - 1 - sf; INT inp = (INT)f_inp; INT mask1 = (0x1 << (sx - 1)); INT mask2 = (0x1 << (sx)) - 1; INT mask3 = 0x7FFFFFFF; INT iam = inp & mask2; INT rnd; if ((inp < 0) && !(iam == mask1)) rnd = inp + mask1; else if ((inp > 0) && !(inp == mask3)) rnd = inp + mask1; else rnd = inp; rnd = rnd >> sx; if (inp == mask3) rnd++; return rnd; } /***************************************************************************** fixp_round() round [typical rounding] See fct roundRef() [which is the reference] returns FIXP_DBL *****************************************************************************/ FIXP_DBL fixp_round(FIXP_DBL f_inp, INT sf) { FDK_ASSERT(sf >= 0); INT sx = DFRACT_BITS - 1 - sf; INT r = fixp_roundToInt(f_inp, sf); ULONG mask = (ULONG)0x1 << (DFRACT_BITS - 1); // 0x80000000 r = r << sx; if ((f_inp > FL2FXCONST_DBL(0.0f)) && (r & mask)) { --r; } FIXP_DBL f_round = (FIXP_DBL)r; return f_round; } fdk-aac-2.0.2/libFDK/src/huff_nodes.cpp000066400000000000000000001575171404225310700175250ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Omer Osman Description: MPEG-D SAC/USAC/SAOC Huffman Part0 Tables *******************************************************************************/ #include "huff_nodes.h" const HUFF_PT0_NODES FDK_huffPart0Nodes = { {{2, 1}, {4, 3}, {6, 5}, {8, 7}, {10, 9}, {12, 11}, {14, 13}, {-8, 15}, {-9, 16}, {-10, 17}, {-18, 18}, {-17, -19}, {-16, 19}, {-11, -20}, {-15, -21}, {-7, 20}, {-22, 21}, {-12, -14}, {-13, -23}, {23, 22}, {-24, -31}, {-6, 24}, {-25, -26}, {26, 25}, {-5, -27}, {-28, 27}, {-4, 28}, {-29, 29}, {-1, -30}, {-2, -3}}, {{2, 1}, {-5, 3}, {-4, -6}, {-3, 4}, {-2, 5}, {-1, 6}, {-7, -8}}, {{-1, 1}, {-8, 2}, {-2, 3}, {5, 4}, {-7, 6}, {-3, -5}, {-4, -6}}, {{-1, 1}, {3, 2}, {-8, 4}, {6, 5}, {-16, 7}, {9, 8}, {11, 10}, {-2, -7}, {-6, 12}, {-4, -5}, {-3, 13}, {-10, 14}, {-11, -12}, {-14, -15}, {-9, -13}}, {{2, 1}, {4, 3}, {6, 5}, {8, 7}, {10, 9}, {12, 11}, {14, 13}, {16, 15}, {18, 17}, {20, 19}, {22, 21}, {24, 23}, {26, 25}, {28, 27}, {30, 29}, {32, 31}, {-47, 33}, {-54, 34}, {-46, 35}, {-48, 36}, {-23, -27}, {-45, 37}, {-55, 38}, {-22, -49}, {-24, -53}, {-44, 39}, {-57, 40}, {-28, 41}, {-52, -56}, {-43, 42}, {-50, 43}, {-25, -26}, {-29, -64}, {-62, 44}, {-21, -51}, {-58, 45}, {-32, 46}, {-31, -42}, {-60, 47}, {-30, 48}, {-20, -61}, {-41, -63}, {-19, -59}, {-40, 49}, {-18, -38}, {-39, 50}, {-36, -37}, {-35, 51}, {-17, 52}, {-16, -34}, {-33, 53}, {-15, 54}, {-14, 55}, {-13, 56}, {-12, 57}, {-11, 58}, {-10, 59}, {-9, 60}, {-7, 61}, {-1, -4}, {-6, 62}, {-5, -8}, {-2, -3}}}; const HUFF_LAV_NODES FDK_huffLavIdxNodes = {{{-1, 1}, {-2, 2}, {-3, -4}}}; static const HUFF_ICC_NOD_1D FDK_huffICCNodes_h1D_0 = { {{-1, 1}, {-2, 2}, {-3, 3}, {-4, 4}, {-5, 5}, {-6, 6}, {-7, -8}}}; static const HUFF_ICC_NOD_2D FDK_huffICCNodes_h2D_0_0 = { {{-1, 1}, {-18, 2}, {-2, -17}}, {{2, 1}, {-1, -52}, {-2, 3}, {5, 4}, {-51, 6}, {-18, 7}, {-17, 8}, {-3, 9}, {-36, 10}, {-19, -50}, {-35, 11}, {-4, 12}, {-34, 13}, {-33, 14}, {-20, -49}}, {{2, 1}, {-86, 3}, {-1, 4}, {6, 5}, {-2, 7}, {-85, 8}, {-18, 9}, {11, 10}, {-17, 12}, {14, 13}, {-70, 15}, {-3, -19}, {-69, 16}, {-84, 17}, {-68, 18}, {-20, -35}, {-34, -83}, {20, 19}, {-4, 21}, {-33, 22}, {-5, 23}, {-53, 24}, {-36, -52}, {-67, 25}, {-21, -82}, {-54, 26}, {-6, 27}, {-51, 28}, {-50, 29}, {-49, 30}, {-37, 31}, {-38, 0}, {0, 0}, {0, 0}, {0, 0}}, {{2, 1}, {4, 3}, {-1, -120}, {6, 5}, {8, 7}, {-18, 9}, {-2, 10}, {12, 11}, {14, 13}, {-17, -119}, {16, 15}, {-103, 17}, {-104, 18}, {-52, 19}, {21, 20}, {-69, 22}, {24, 23}, {-3, -35}, {-19, 25}, {-34, -85}, {27, 26}, {-86, 28}, {-118, 29}, {-37, 30}, {32, 31}, {-102, 33}, {-20, -22}, {-4, -117}, {-87, 34}, {-100, 35}, {-33, -36}, {37, 36}, {-70, -88}, {-101, 38}, {-5, 39}, {-51, -53}, {-50, 40}, {-115, 41}, {-21, 42}, {-116, 43}, {-38, 44}, {-23, -84}, {-49, -99}, {46, 45}, {-6, -114}, {-7, -72}, {-71, 47}, {-8, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}}}; static const HUFF_ICC_NOD_2D FDK_huffICCNodes_h2D_0_1 = { {{-1, 1}, {-18, 2}, {-2, -17}}, {{2, 1}, {-1, -52}, {-17, 3}, {5, 4}, {-36, 6}, {-2, 7}, {-18, -33}, {9, 8}, {-20, 10}, {-34, -51}, {-49, 11}, {-35, 12}, {-19, 13}, {-3, 14}, {-4, -50}}, {{2, 1}, {-86, 3}, {-1, 4}, {-17, 5}, {7, 6}, {-70, 8}, {-33, 9}, {-18, 10}, {-2, 11}, {-54, 12}, {-49, 13}, {-38, 14}, {-34, -65}, {-85, 15}, {-50, 16}, {-69, 17}, {-22, 18}, {-53, 19}, {21, 20}, {-19, -81}, {-66, 22}, {-3, -35}, {24, 23}, {-37, 25}, {-68, -84}, {-51, 26}, {28, 27}, {-20, -52}, {30, 29}, {-4, -36}, {-83, 31}, {-67, 32}, {-82, 33}, {-21, 34}, {-5, -6}}, {{2, 1}, {-1, 3}, {-120, 4}, {-17, 5}, {7, 6}, {-104, 8}, {-33, 9}, {11, 10}, {13, 12}, {-49, 14}, {-88, 15}, {-18, -97}, {-65, 16}, {-40, 17}, {-2, -72}, {19, 18}, {-113, 20}, {-34, 21}, {-56, -81}, {23, 22}, {-50, 24}, {-82, -119}, {-24, -103}, {26, 25}, {28, 27}, {30, 29}, {-55, -87}, {-66, 31}, {33, 32}, {-98, 34}, {-35, -67}, {-19, 35}, {-70, 36}, {-71, 37}, {-51, -52}, {-3, 38}, {40, 39}, {-86, -118}, {42, 41}, {-39, -69}, {-54, -83}, {44, 43}, {-102, 45}, {-101, 46}, {-68, -85}, {-36, -53}, {-5, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}}}; static const HUFF_ICC_NOD_2D FDK_huffICCNodes_h2D_1_0 = { {{-1, 1}, {-18, 2}, {-2, -17}}, {{-52, 1}, {-1, 2}, {4, 3}, {-2, -17}, {-18, 5}, {-36, 6}, {-51, 7}, {9, 8}, {-33, 10}, {-34, 11}, {-35, 12}, {-19, -20}, {-3, 13}, {-49, 14}, {-4, -50}}, {{-1, 1}, {-86, 2}, {4, 3}, {-17, 5}, {-2, 6}, {-18, 7}, {-70, 8}, {-85, 9}, {11, 10}, {13, 12}, {-33, 14}, {16, 15}, {-34, -54}, {-69, 17}, {-38, 18}, {-50, 19}, {-35, -53}, {-49, 20}, {-19, 21}, {-3, 22}, {-65, 23}, {-68, 24}, {-22, 25}, {-81, -84}, {-66, 26}, {-37, 27}, {-20, -51}, {29, 28}, {-52, 30}, {-4, -83}, {-36, 31}, {-67, 32}, {-5, 33}, {-82, 34}, {-21, 0}}, {{-1, 1}, {-120, 2}, {4, 3}, {-17, 5}, {-2, 6}, {8, 7}, {-18, 9}, {-104, 10}, {12, 11}, {14, 13}, {16, 15}, {-119, 17}, {-81, 18}, {20, 19}, {-33, 21}, {-88, 22}, {-103, 23}, {-34, 24}, {-56, 25}, {-72, 26}, {-49, 27}, {-82, 28}, {-50, 29}, {-65, 30}, {-55, -87}, {-19, 31}, {-67, 32}, {-35, -40}, {34, 33}, {-52, -71}, {-66, 35}, {-70, 36}, {38, 37}, {-51, -97}, {-86, -102}, {-3, 39}, {-118, 40}, {42, 41}, {-24, -85}, {-54, 43}, {-39, 44}, {-98, -113}, {-36, -37}, {-20, -69}, {-4, 45}, {-5, 46}, {-21, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}}}; static const HUFF_ICC_NOD_2D FDK_huffICCNodes_h2D_1_1 = { {{-1, 1}, {-18, 2}, {-2, -17}}, {{-52, 1}, {-1, 2}, {4, 3}, {-2, 5}, {-17, -18}, {-51, 6}, {-36, 7}, {9, 8}, {-35, 10}, {-3, 11}, {-19, -34}, {-33, 12}, {-50, 13}, {-20, 14}, {-4, -49}}, {{2, 1}, {-86, 3}, {-1, 4}, {6, 5}, {-18, 7}, {-2, -17}, {9, 8}, {-70, 10}, {-69, -85}, {-35, 11}, {13, 12}, {-34, 14}, {-19, 15}, {-53, 16}, {-68, 17}, {-33, 18}, {-3, -52}, {20, 19}, {-54, 21}, {-84, 22}, {-50, 23}, {-20, -51}, {-36, 24}, {26, 25}, {-83, 27}, {-4, -38}, {-49, 28}, {-37, 29}, {-67, 30}, {-5, 31}, {-21, 32}, {-65, -66}, {-82, 33}, {-22, 34}, {-6, -81}}, {{2, 1}, {-1, -120}, {4, 3}, {6, 5}, {-18, 7}, {9, 8}, {-17, 10}, {-2, 11}, {-103, 12}, {-52, 13}, {-35, -104}, {-119, 14}, {16, 15}, {-69, -86}, {18, 17}, {-34, 19}, {-19, 20}, {22, 21}, {-70, 23}, {-87, 24}, {-102, 25}, {-85, 26}, {-33, 27}, {-36, 28}, {-3, 29}, {-88, 30}, {-51, 31}, {-118, 32}, {34, 33}, {-68, 35}, {-53, 36}, {-67, 37}, {-20, 38}, {-101, 39}, {-50, 40}, {42, 41}, {-37, 43}, {-116, 44}, {-117, 45}, {-49, 46}, {-21, -100}, {48, 47}, {-55, -71}, {-4, 49}, {-22, -84}, {-115, 50}, {-66, -82}, {-72, 51}, {-5, -6}, {-54, 52}, {-38, 53}, {-83, 54}, {-40, 55}, {-39, 56}, {-99, 57}, {-23, -56}, {-7, 58}, {-65, -97}, {-8, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}}}; const HUFF_ICC_NODES FDK_huffICCNodes = { {&FDK_huffICCNodes_h1D_0, &FDK_huffICCNodes_h1D_0, &FDK_huffICCNodes_h1D_0}, {{&FDK_huffICCNodes_h2D_0_0, &FDK_huffICCNodes_h2D_0_1}, {&FDK_huffICCNodes_h2D_1_0, &FDK_huffICCNodes_h2D_1_1}, {&FDK_huffICCNodes_h2D_0_1, &FDK_huffICCNodes_h2D_0_1}}}; static const HUFF_CLD_NOD_1D FDK_huffCLDNodes_h1D_0 = { {{-1, 1}, {-2, 2}, {-3, 3}, {-4, 4}, {-5, 5}, {-6, 6}, {-7, 7}, {-8, 8}, {-9, 9}, {-10, 10}, {-11, 11}, {-12, 12}, {-13, 13}, {15, 14}, {-14, 16}, {-15, 17}, {-16, 18}, {-17, 19}, {-18, 20}, {-19, 21}, {-20, -21}, {-23, 22}, {-22, 23}, {-24, 24}, {-25, 25}, {27, 26}, {29, 28}, {-30, -31}, {-28, -29}, {-26, -27}}}; static const HUFF_CLD_NOD_1D FDK_huffCLDNodes_h1D_1 = { {{-1, 1}, {-2, 2}, {-3, 3}, {-4, 4}, {-5, 5}, {-6, 6}, {-7, 7}, {9, 8}, {-8, 10}, {-9, 11}, {-10, 12}, {-11, 13}, {-12, 14}, {-13, 15}, {-14, 16}, {-15, 17}, {-16, 18}, {-17, 19}, {-18, 20}, {-19, -20}, {-21, 21}, {-22, 22}, {-23, 23}, {25, 24}, {-24, 26}, {-25, 27}, {29, 28}, {-26, -31}, {-29, -30}, {-27, -28}}}; static const HUFF_CLD_NOD_2D FDK_huffCLDNodes_h2_0_0 = { {{2, 1}, {-1, -52}, {4, 3}, {-2, 5}, {-51, 6}, {-17, -18}, {8, 7}, {10, 9}, {-3, -36}, {-19, 11}, {-35, -50}, {-34, 12}, {-4, 13}, {-33, 14}, {-20, -49}}, {{2, 1}, {4, 3}, {-86, 5}, {7, 6}, {9, 8}, {-1, -2}, {-85, 10}, {-18, 11}, {-17, 12}, {14, 13}, {-70, 15}, {17, 16}, {-19, -69}, {-84, 18}, {-3, 19}, {21, 20}, {-34, -68}, {-20, 22}, {-35, 23}, {-83, 24}, {-33, 25}, {-4, 26}, {-53, 27}, {-54, -67}, {-36, 28}, {-21, -52}, {-82, 29}, {-5, -50}, {-51, 30}, {-38, 31}, {-37, -49}, {-6, 32}, {-66, 33}, {-65, 34}, {-22, -81}}, {{2, 1}, {4, 3}, {-120, 5}, {7, 6}, {9, 8}, {11, 10}, {-1, 12}, {-18, -119}, {-2, 13}, {15, 14}, {-17, 16}, {-104, 17}, {19, 18}, {-19, 20}, {-103, 21}, {-118, 22}, {24, 23}, {-3, 25}, {27, 26}, {-34, 28}, {-102, 29}, {-20, 30}, {-35, 31}, {33, 32}, {-117, 34}, {-33, 35}, {-88, 36}, {-4, 37}, {-87, 38}, {40, 39}, {-36, -101}, {-86, 41}, {-21, -37}, {-85, -100}, {-52, 42}, {-22, 43}, {-116, 44}, {-50, 45}, {47, 46}, {-5, -51}, {-115, 48}, {-70, 49}, {-84, 50}, {-38, -49}, {-72, -99}, {-53, 51}, {-69, -71}, {-23, 52}, {-6, -67}, {-114, 53}, {-7, 54}, {-66, -68}, {-55, 55}, {57, 56}, {-54, -65}, {-8, -56}, {-82, -83}, {59, 58}, {-39, -40}, {-81, 60}, {-98, 61}, {-97, 62}, {-24, -113}}, {{2, 1}, {4, 3}, {6, 5}, {-154, 7}, {9, 8}, {11, 10}, {13, 12}, {15, 14}, {-18, 16}, {-153, 17}, {-1, -2}, {19, 18}, {-138, 20}, {-17, 21}, {23, 22}, {25, 24}, {-19, -137}, {27, 26}, {-152, 28}, {30, 29}, {-3, -34}, {32, 31}, {34, 33}, {36, 35}, {-136, 37}, {-35, 38}, {-20, 39}, {-122, 40}, {-151, 41}, {-33, 42}, {-121, 43}, {45, 44}, {47, 46}, {-4, 48}, {-36, -120}, {-135, 49}, {51, 50}, {-21, 52}, {54, 53}, {56, 55}, {-50, -150}, {58, 57}, {-51, 59}, {61, 60}, {-119, 62}, {-52, 63}, {-5, 64}, {-37, 65}, {-117, -134}, {-39, -54}, {-22, 66}, {-106, 67}, {-69, -102}, {-132, 68}, {-105, 69}, {-49, 70}, {-149, 71}, {-24, -104}, {73, 72}, {-53, 74}, {-38, -118}, {-103, 75}, {-6, 76}, {-66, -87}, {-133, -147}, {-23, 77}, {-67, 78}, {-68, -86}, {-70, -101}, {-40, -148}, {-116, 79}, {-55, 80}, {-84, -131}, {82, 81}, {-89, -90}, {-7, -25}, {-85, -88}, {-65, 83}, {-72, -146}, {85, 84}, {-9, -71}, {-83, 86}, {-82, 87}, {-8, 88}, {-100, 89}, {-74, -99}, {-73, 90}, {-10, -81}, {-56, 91}, {-57, -98}, {93, 92}, {-58, -114}, {-97, -115}, {95, 94}, {-41, 96}, {-42, 97}, {-26, -129}, {-113, 98}, {-130, -145}}}; static const HUFF_CLD_NOD_2D FDK_huffCLDNodes_h2_0_1 = { {{-1, 1}, {-52, 2}, {-17, 3}, {5, 4}, {-36, 6}, {-33, 7}, {-2, -18}, {-20, 8}, {10, 9}, {-34, -49}, {-51, 11}, {-35, 12}, {-19, 13}, {-3, 14}, {-4, -50}}, {{2, 1}, {4, 3}, {-86, 5}, {-1, 6}, {-17, 7}, {-70, 8}, {10, 9}, {-18, 11}, {-33, 12}, {-54, 13}, {-2, 14}, {-34, 15}, {-38, 16}, {-49, 17}, {-85, 18}, {-50, 19}, {-69, 20}, {-53, -65}, {-22, 21}, {-66, 22}, {-19, 23}, {-37, 24}, {-35, -81}, {-3, 25}, {-51, 26}, {-68, -84}, {-52, 27}, {29, 28}, {-20, 30}, {-4, -36}, {-83, 31}, {-67, 32}, {-21, 33}, {-5, 34}, {-6, -82}}, {{2, 1}, {4, 3}, {6, 5}, {-120, 7}, {-17, 8}, {-1, -104}, {10, 9}, {12, 11}, {-18, 13}, {-33, -88}, {15, 14}, {17, 16}, {-2, 18}, {-34, 19}, {-72, 20}, {-49, 21}, {-119, 22}, {-50, 23}, {-103, 24}, {-56, 25}, {-65, 26}, {28, 27}, {-40, -87}, {-66, 29}, {-82, 30}, {32, 31}, {-19, -81}, {-71, 33}, {-97, 34}, {-35, -55}, {-24, 35}, {37, 36}, {-3, -98}, {-51, 38}, {-67, 39}, {-39, -118}, {-113, 40}, {-102, 41}, {-86, 42}, {-70, -83}, {44, 43}, {-20, -54}, {-52, 45}, {-36, 46}, {-4, 47}, {-68, 48}, {-85, 49}, {-101, -117}, {-69, 50}, {52, 51}, {-21, -37}, {-53, 53}, {55, 54}, {-5, -100}, {-116, 56}, {-84, 57}, {-38, 58}, {-22, -99}, {-115, 59}, {-6, 60}, {-23, 61}, {-7, 62}, {-114, 0}}, {{2, 1}, {4, 3}, {6, 5}, {-154, 7}, {9, 8}, {-17, 10}, {-138, 11}, {-1, 12}, {14, 13}, {16, 15}, {-33, -122}, {-18, 17}, {19, 18}, {-34, 20}, {-2, 21}, {-106, 22}, {-49, 23}, {25, 24}, {-50, 26}, {-153, 27}, {-90, 28}, {-137, 29}, {-65, 30}, {32, 31}, {-66, 33}, {-121, 34}, {-74, 35}, {-81, 36}, {38, 37}, {-42, 39}, {-82, 40}, {-105, 41}, {-19, -114}, {-58, 42}, {-35, 43}, {-97, 44}, {46, 45}, {-129, 47}, {-26, -89}, {-57, -98}, {-51, 48}, {-3, 49}, {-113, 50}, {-130, 51}, {-152, 52}, {-67, -73}, {-99, -136}, {-145, 53}, {-120, 54}, {-41, 55}, {-83, 56}, {-72, 57}, {-104, 58}, {-115, 59}, {-20, 60}, {62, 61}, {-36, -88}, {-84, 63}, {-52, -56}, {65, 64}, {-4, -87}, {-68, 66}, {-151, 67}, {-100, -135}, {69, 68}, {-69, -119}, {-103, 70}, {-71, 71}, {73, 72}, {-21, 74}, {-85, 75}, {-37, -53}, {-86, 76}, {78, 77}, {-102, -150}, {-5, 79}, {-134, 80}, {-118, 81}, {-54, -117}, {83, 82}, {-38, -70}, {-22, 84}, {-6, 85}, {87, 86}, {-55, 88}, {-101, 89}, {-133, -149}, {-24, -39}, {91, 90}, {-132, 92}, {-23, 93}, {-7, 94}, {-147, -148}, {-116, -131}, {-25, 95}, {-40, 0}, {0, 0}, {0, 0}, {0, 0}}}; static const HUFF_CLD_NOD_2D FDK_huffCLDNodes_h2_1_0 = { {{-1, 1}, {-52, 2}, {-17, 3}, {5, 4}, {-2, -36}, {-18, 6}, {8, 7}, {-51, 9}, {-33, 10}, {-34, 11}, {-20, -35}, {-19, 12}, {-3, 13}, {-49, 14}, {-4, -50}}, {{2, 1}, {-86, 3}, {-1, 4}, {-17, 5}, {7, 6}, {-70, 8}, {-2, -18}, {10, 9}, {12, 11}, {-85, 13}, {-33, 14}, {-34, -54}, {16, 15}, {-69, 17}, {19, 18}, {-50, -53}, {-19, 20}, {-38, 21}, {-35, -49}, {-3, 22}, {24, 23}, {-68, 25}, {-84, 26}, {-65, 27}, {-51, -66}, {-22, -37}, {-52, 28}, {-20, 29}, {-36, 30}, {-81, 31}, {-4, -83}, {-67, 32}, {-21, 33}, {-5, 34}, {-6, -82}}, {{2, 1}, {-120, 3}, {-1, 4}, {6, 5}, {-17, 7}, {-104, 8}, {-18, 9}, {-2, 10}, {12, 11}, {14, 13}, {-119, 15}, {-33, 16}, {-34, -88}, {-103, 17}, {19, 18}, {21, 20}, {23, 22}, {25, 24}, {-19, -72}, {-50, 26}, {-49, 27}, {-87, 28}, {30, 29}, {32, 31}, {-3, -35}, {34, 33}, {-56, 35}, {-65, -66}, {-40, 36}, {-82, -118}, {-71, 37}, {-55, 38}, {-67, -102}, {-51, 39}, {-70, 40}, {42, 41}, {-81, 43}, {-86, 44}, {-52, -97}, {-98, 45}, {-24, -39}, {-20, 46}, {-54, -83}, {-36, 47}, {-85, 48}, {-68, 49}, {-4, 50}, {-69, -113}, {-117, 51}, {-37, -101}, {-53, 52}, {-21, 53}, {55, 54}, {-84, -100}, {-5, 56}, {-116, 57}, {-22, 58}, {-38, -115}, {60, 59}, {-6, -99}, {-23, 61}, {-114, 62}, {-7, -8}}, {{2, 1}, {-154, 3}, {5, 4}, {-1, 6}, {8, 7}, {-17, 9}, {-138, 10}, {-18, 11}, {-2, 12}, {14, 13}, {16, 15}, {-153, 17}, {-34, 18}, {-33, -122}, {20, 19}, {22, 21}, {-137, 23}, {25, 24}, {27, 26}, {-106, 28}, {30, 29}, {-50, 31}, {-19, 32}, {-49, -121}, {34, 33}, {36, 35}, {-35, 37}, {-90, 38}, {-66, 39}, {-3, 40}, {42, 41}, {-65, 43}, {-105, 44}, {46, 45}, {-74, 47}, {-51, 48}, {-82, -152}, {-136, 49}, {-81, 50}, {-42, -89}, {-114, 51}, {53, 52}, {-57, -58}, {-120, 54}, {-98, 55}, {-67, 56}, {-97, 57}, {59, 58}, {-99, 60}, {-73, -104}, {-72, 61}, {-113, 62}, {-20, -83}, {-84, -130}, {-36, 63}, {-26, 64}, {-41, 65}, {-52, -129}, {-87, -88}, {67, 66}, {-115, 68}, {-68, 69}, {-56, -69}, {-4, -100}, {-151, 70}, {-135, 71}, {-103, -119}, {73, 72}, {-71, -145}, {-102, 74}, {76, 75}, {-53, -85}, {-37, 77}, {-21, -86}, {79, 78}, {-5, 80}, {-54, -134}, {-150, 81}, {-118, 82}, {-70, 83}, {-117, 84}, {-22, -38}, {-101, 85}, {-55, 86}, {-149, 87}, {-39, 88}, {-133, 89}, {-6, 90}, {-116, 91}, {-24, 92}, {-7, -132}, {-23, 93}, {-40, 94}, {-131, -148}, {-25, 95}, {-147, 96}, {-146, 97}, {-8, 0}, {0, 0}}}; static const HUFF_CLD_NOD_2D FDK_huffCLDNodes_h2_1_1 = { {{-1, 1}, {-52, 2}, {4, 3}, {-2, 5}, {-17, 6}, {-18, 7}, {-36, -51}, {9, 8}, {-35, 10}, {-34, 11}, {-19, -33}, {-3, 12}, {-20, 13}, {-50, 14}, {-4, -49}}, {{2, 1}, {-86, 3}, {5, 4}, {-1, 6}, {8, 7}, {-17, -18}, {-2, 9}, {-70, 10}, {-85, 11}, {13, 12}, {-69, 14}, {-34, 15}, {17, 16}, {-19, 18}, {-33, -35}, {-54, 19}, {-53, 20}, {-3, 21}, {-68, 22}, {-84, 23}, {-50, 24}, {-52, 25}, {-51, 26}, {-20, -36}, {-49, 27}, {-38, 28}, {-37, 29}, {-4, -83}, {-67, 30}, {-66, 31}, {-21, 32}, {-22, -65}, {-5, 33}, {-82, 34}, {-6, -81}}, {{2, 1}, {4, 3}, {-120, 5}, {7, 6}, {9, 8}, {-1, 10}, {-18, 11}, {-17, 12}, {-2, -104}, {-119, 13}, {15, 14}, {-103, 16}, {18, 17}, {-34, 19}, {-19, 20}, {22, 21}, {-35, 23}, {-33, 24}, {-88, 25}, {-87, 26}, {28, 27}, {-3, -102}, {-86, 29}, {-52, -118}, {31, 30}, {-50, 32}, {-51, 33}, {-70, 34}, {-36, 35}, {-85, 36}, {-20, 37}, {39, 38}, {-69, -71}, {-72, 40}, {-49, -67}, {42, 41}, {-68, 43}, {-4, -101}, {-53, -117}, {-37, 44}, {-66, 45}, {-55, 46}, {48, 47}, {-54, 49}, {-21, 50}, {-84, -100}, {-56, -65}, {52, 51}, {-82, -83}, {54, 53}, {-5, -116}, {-22, 55}, {-38, 56}, {-39, -40}, {58, 57}, {-81, -115}, {-98, -99}, {-6, 59}, {-23, 60}, {-24, 61}, {-7, -97}, {-114, 62}, {-8, -113}}, {{2, 1}, {4, 3}, {-154, 5}, {7, 6}, {9, 8}, {11, 10}, {-1, 12}, {-18, 13}, {-17, 14}, {-2, -138}, {16, 15}, {-153, 17}, {-137, 18}, {20, 19}, {22, 21}, {-34, 23}, {-19, 24}, {-35, 25}, {27, 26}, {29, 28}, {-121, 30}, {-120, 31}, {-136, 32}, {-33, -122}, {34, 33}, {-152, 35}, {-3, 36}, {-51, 37}, {-52, 38}, {-69, 39}, {-36, 40}, {-50, 41}, {43, 42}, {-20, 44}, {-104, 45}, {-103, 46}, {-87, 47}, {-119, 48}, {-105, 49}, {-86, 50}, {-102, 51}, {-106, 52}, {-49, -135}, {-68, 53}, {55, 54}, {-53, 56}, {-67, -151}, {-4, 57}, {-84, 58}, {-85, 59}, {-66, 60}, {-37, 61}, {-70, 62}, {-54, -88}, {-21, 63}, {65, 64}, {-89, 66}, {-118, 67}, {-72, 68}, {-90, 69}, {-71, 70}, {-65, -134}, {-150, 71}, {-83, 72}, {-5, 73}, {-101, -117}, {-82, 74}, {76, 75}, {-99, 77}, {-38, 78}, {-100, 79}, {-22, 80}, {-73, 81}, {-39, -74}, {83, 82}, {-55, -81}, {-57, 84}, {-133, -149}, {-56, 85}, {-6, 86}, {-98, 87}, {-132, 88}, {-23, 89}, {-114, 90}, {-116, 91}, {-58, -115}, {-24, 92}, {-97, -148}, {-40, -41}, {-7, -42}, {-147, 93}, {95, 94}, {-131, 96}, {-8, -130}, {-25, -113}, {-9, 97}, {-26, -129}, {-146, 98}, {-10, -145}}}; const HUFF_CLD_NODES FDK_huffCLDNodes = { {&FDK_huffCLDNodes_h1D_0, &FDK_huffCLDNodes_h1D_1, &FDK_huffCLDNodes_h1D_1}, {{&FDK_huffCLDNodes_h2_0_0, &FDK_huffCLDNodes_h2_0_1}, {&FDK_huffCLDNodes_h2_1_0, &FDK_huffCLDNodes_h2_1_1}, {&FDK_huffCLDNodes_h2_0_1, &FDK_huffCLDNodes_h2_0_1}}}; const HUFF_RES_NODES FDK_huffReshapeNodes = { {{2, 1}, {4, 3}, {6, 5}, {-33, 7}, {-17, 8}, {-49, 9}, {-34, 10}, {12, 11}, {-18, -35}, {-50, 13}, {15, 14}, {-40, 16}, {-36, 17}, {-19, 18}, {-1, -37}, {-51, 19}, {21, 20}, {-38, -65}, {-2, -39}, {-20, 22}, {-52, 23}, {25, 24}, {-21, 26}, {-66, 27}, {-53, 28}, {-3, 29}, {31, 30}, {-22, 32}, {-54, 33}, {-4, 34}, {-56, 35}, {-24, -67}, {-23, -55}, {-8, -72}, {-5, 36}, {-68, 37}, {-6, 38}, {-7, -69}, {-70, -71}}}; const HUFF_IPD_NODES FDK_huffIPDNodes = { {{{{-1, 1}, {-8, 2}, {-2, 3}, {5, 4}, {-3, -7}, {-6, 6}, {-4, -5}}}, {{{-1, 1}, {-2, 2}, {-8, 3}, {-3, 4}, {-7, 5}, {-4, 6}, {-5, -6}}}, {{{-1, 1}, {-8, 2}, {-2, 3}, {5, 4}, {-3, -7}, {-6, 6}, {-4, -5}}}}, {{{{{-1, 1}, {-18, 2}, {-17, 0}}, {{-1, 1}, {-36, 2}, {-18, 3}, {-35, 4}, {-52, 5}, {7, 6}, {-34, 8}, {-33, -49}, {-20, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}}, {{-1, 1}, {3, 2}, {5, 4}, {-86, 6}, {-66, 7}, {9, 8}, {11, 10}, {-18, 12}, {-51, 13}, {-37, -52}, {-69, 14}, {-38, 15}, {-53, 16}, {-35, 17}, {-50, -70}, {-22, -49}, {-33, 18}, {-17, 19}, {-34, -65}, {-81, 20}, {-54, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}}, {{2, 1}, {4, 3}, {-1, 5}, {-69, 6}, {-120, 7}, {-68, 8}, {10, 9}, {12, 11}, {14, 13}, {-52, -54}, {-18, 15}, {-70, 16}, {-67, 17}, {19, 18}, {-17, 20}, {-113, 21}, {23, 22}, {-83, 24}, {-24, 25}, {-103, -104}, {-51, -55}, {27, 26}, {-71, 28}, {-86, 29}, {-35, 30}, {-66, 31}, {-39, -50}, {-82, -98}, {-72, 32}, {-56, -87}, {-34, 33}, {-33, -88}, {-40, -97}, {-65, 34}, {-49, 35}, {-81, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}}}, {{{-1, 1}, {-18, 2}, {-17, 0}}, {{-1, 1}, {-36, 2}, {-18, 3}, {-35, 4}, {-52, 5}, {7, 6}, {-34, 8}, {-33, -49}, {-20, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}}, {{-1, 1}, {3, 2}, {5, 4}, {-86, 6}, {-66, 7}, {9, 8}, {11, 10}, {-18, 12}, {-51, 13}, {-37, -52}, {-69, 14}, {-38, 15}, {-53, 16}, {-35, 17}, {-50, -70}, {-22, -49}, {-33, 18}, {-17, 19}, {-34, -65}, {-81, 20}, {-54, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}}, {{2, 1}, {4, 3}, {-1, 5}, {-69, 6}, {-120, 7}, {-68, 8}, {10, 9}, {12, 11}, {14, 13}, {-52, -54}, {-18, 15}, {-70, 16}, {-67, 17}, {19, 18}, {-17, 20}, {-113, 21}, {23, 22}, {-83, 24}, {-24, 25}, {-103, -104}, {-51, -55}, {27, 26}, {-71, 28}, {-86, 29}, {-35, 30}, {-66, 31}, {-39, -50}, {-82, -98}, {-72, 32}, {-56, -87}, {-34, 33}, {-33, -88}, {-40, -97}, {-65, 34}, {-49, 35}, {-81, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}}}}, {{{{-1, 1}, {-18, 2}, {-17, 0}}, {{-1, 1}, {3, 2}, {-18, 4}, {-52, 5}, {-34, -36}, {-35, 6}, {-17, 7}, {-33, 8}, {-20, 9}, {-49, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}}, {{-1, 1}, {3, 2}, {-52, 4}, {-86, 5}, {-35, 6}, {-53, 7}, {-70, 8}, {-17, 9}, {-37, 10}, {12, 11}, {-38, -66}, {-18, 13}, {-51, 14}, {16, 15}, {-34, -69}, {18, 17}, {-54, -65}, {-50, 19}, {-33, -49}, {-22, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}}, {{-1, 1}, {-69, 2}, {4, 3}, {-120, 5}, {7, 6}, {-113, 8}, {-68, 9}, {11, 10}, {-17, 12}, {-52, 13}, {-24, 14}, {-18, 15}, {17, 16}, {-104, 18}, {20, 19}, {-54, -70}, {22, 21}, {24, 23}, {-86, -97}, {-103, 25}, {-83, 26}, {-35, 27}, {-34, -98}, {-40, 28}, {-39, -67}, {30, 29}, {-33, -51}, {-87, 31}, {-88, 32}, {-82, 33}, {-55, -81}, {-56, -71}, {-72, 34}, {-50, -66}, {-65, 35}, {-49, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}}}, {{{-1, 1}, {-18, 2}, {-17, 0}}, {{-1, 1}, {3, 2}, {-18, 4}, {-52, 5}, {-34, -36}, {-35, 6}, {-17, 7}, {-33, 8}, {-20, 9}, {-49, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}}, {{-1, 1}, {3, 2}, {-52, 4}, {-86, 5}, {-35, 6}, {-53, 7}, {-70, 8}, {-17, 9}, {-37, 10}, {12, 11}, {-38, -66}, {-18, 13}, {-51, 14}, {16, 15}, {-34, -69}, {18, 17}, {-54, -65}, {-50, 19}, {-33, -49}, {-22, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}}, {{-1, 1}, {-69, 2}, {4, 3}, {-120, 5}, {7, 6}, {-113, 8}, {-68, 9}, {11, 10}, {-17, 12}, {-52, 13}, {-24, 14}, {-18, 15}, {17, 16}, {-104, 18}, {20, 19}, {-54, -70}, {22, 21}, {24, 23}, {-86, -97}, {-103, 25}, {-83, 26}, {-35, 27}, {-34, -98}, {-40, 28}, {-39, -67}, {30, 29}, {-33, -51}, {-87, 31}, {-88, 32}, {-82, 33}, {-55, -81}, {-56, -71}, {-72, 34}, {-50, -66}, {-65, 35}, {-49, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}}}}, {{{{-1, 1}, {-18, 2}, {-17, 0}}, {{-1, 1}, {-36, 2}, {-18, 3}, {-35, 4}, {-52, 5}, {7, 6}, {-34, 8}, {-33, -49}, {-20, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}}, {{-1, 1}, {3, 2}, {5, 4}, {-86, 6}, {-66, 7}, {9, 8}, {11, 10}, {-18, 12}, {-51, 13}, {-37, -52}, {-69, 14}, {-38, 15}, {-53, 16}, {-35, 17}, {-50, -70}, {-22, -49}, {-33, 18}, {-17, 19}, {-34, -65}, {-81, 20}, {-54, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}}, {{2, 1}, {4, 3}, {-1, 5}, {-69, 6}, {-120, 7}, {-68, 8}, {10, 9}, {12, 11}, {14, 13}, {-52, -54}, {-18, 15}, {-70, 16}, {-67, 17}, {19, 18}, {-17, 20}, {-113, 21}, {23, 22}, {-83, 24}, {-24, 25}, {-103, -104}, {-51, -55}, {27, 26}, {-71, 28}, {-86, 29}, {-35, 30}, {-66, 31}, {-39, -50}, {-82, -98}, {-72, 32}, {-56, -87}, {-34, 33}, {-33, -88}, {-40, -97}, {-65, 34}, {-49, 35}, {-81, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}}}, {{{-1, 1}, {-18, 2}, {-17, 0}}, {{-1, 1}, {-36, 2}, {-18, 3}, {-35, 4}, {-52, 5}, {7, 6}, {-34, 8}, {-33, -49}, {-20, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}}, {{-1, 1}, {3, 2}, {5, 4}, {-86, 6}, {-66, 7}, {9, 8}, {11, 10}, {-18, 12}, {-51, 13}, {-37, -52}, {-69, 14}, {-38, 15}, {-53, 16}, {-35, 17}, {-50, -70}, {-22, -49}, {-33, 18}, {-17, 19}, {-34, -65}, {-81, 20}, {-54, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}}, {{2, 1}, {4, 3}, {-1, 5}, {-69, 6}, {-120, 7}, {-68, 8}, {10, 9}, {12, 11}, {14, 13}, {-52, -54}, {-18, 15}, {-70, 16}, {-67, 17}, {19, 18}, {-17, 20}, {-113, 21}, {23, 22}, {-83, 24}, {-24, 25}, {-103, -104}, {-51, -55}, {27, 26}, {-71, 28}, {-86, 29}, {-35, 30}, {-66, 31}, {-39, -50}, {-82, -98}, {-72, 32}, {-56, -87}, {-34, 33}, {-33, -88}, {-40, -97}, {-65, 34}, {-49, 35}, {-81, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}}}}}}; static const HUFF_OLD_NOD_1D huffOLDNodes_h1D_0 = {{{-1, 1}, {3, 2}, {-2, 4}, {-3, 5}, {-4, 6}, {-5, 7}, {-6, -8}, {-7, 8}, {10, 9}, {12, 11}, {-9, -11}, {-10, 13}, {-12, 14}, {-13, -16}, {-14, -15}}}; static const HUFF_OLD_NOD_1D huffOLDNodes_h1D_1 = {{{-1, 1}, {-2, 2}, {4, 3}, {-3, 5}, {-4, 6}, {-5, 7}, {-6, -8}, {-7, 8}, {10, 9}, {12, 11}, {-9, 13}, {-16, 14}, {-10, -15}, {-11, -12}, {-13, -14}}}; static const HUFF_OLD_NOD_2D huffOLDNodes_h2D_0_0 = { {{2, 1}, {-1, 3}, {5, 4}, {-2, 6}, {-3, -4}, {-17, 7}, {-18, 8}, {-19, 9}, {-20, 10}, {-52, 11}, {-33, 12}, {-34, -35}, {-36, 13}, {-51, 14}, {-49, -50}}, {{-1, 1}, {3, 2}, {5, 4}, {7, 6}, {-103, 8}, {10, 9}, {12, 11}, {-18, 13}, {15, 14}, {-2, 16}, {-86, 17}, {-35, 18}, {20, 19}, {-102, 21}, {23, 22}, {-69, 24}, {-87, 25}, {-3, 26}, {-17, 27}, {-19, 28}, {-52, 29}, {-34, -101}, {31, 30}, {-85, 32}, {34, 33}, {-20, -70}, {-4, 35}, {-71, -100}, {-5, -33}, {-50, 36}, {-36, -55}, {-54, -84}, {38, 37}, {-51, -53}, {-21, 39}, {-6, -99}, {-37, -68}, {-83, 40}, {-7, -49}, {-22, -98}, {42, 41}, {44, 43}, {-66, 45}, {-67, 46}, {-38, -39}, {-65, -82}, {-23, 47}, {-81, -97}}, {{-1, 1}, {3, 2}, {5, 4}, {7, 6}, {9, 8}, {11, 10}, {13, 12}, {15, 14}, {-154, 16}, {-103, 17}, {19, 18}, {21, 20}, {-18, 22}, {24, 23}, {26, 25}, {28, 27}, {-137, 29}, {31, 30}, {-2, -51}, {33, 32}, {-35, 34}, {-26, 35}, {37, 36}, {-8, 38}, {-70, -153}, {40, 39}, {-120, 41}, {-52, 42}, {44, 43}, {-3, -138}, {46, 45}, {48, 47}, {-34, 49}, {-7, 50}, {-19, 51}, {-17, 52}, {-152, 53}, {-4, -151}, {-33, 54}, {-106, 55}, {-53, -122}, {-105, -136}, {-121, 56}, {-104, 57}, {-50, -118}, {-20, 58}, {-5, 59}, {-38, 60}, {-133, 61}, {-148, 62}, {-23, -135}, {-36, 63}, {-6, 64}, {66, 65}, {-21, -150}, {68, 67}, {-49, 69}, {-134, 70}, {-119, 71}, {-37, 72}, {-149, 73}, {-9, 74}, {-69, 75}, {-86, 76}, {-22, 77}, {-68, 78}, {80, 79}, {82, 81}, {84, 83}, {-88, 85}, {-132, 86}, {-90, 87}, {-10, -117}, {-67, 88}, {-71, 89}, {-87, 90}, {-54, -66}, {-25, 91}, {-89, 92}, {-72, 93}, {-131, 94}, {-113, -115}, {-99, 95}, {-73, -116}, {-24, -85}, {-84, -102}, {-39, 96}, {-55, -98}, {-81, -97}, {-82, -83}, {-114, 97}, {-146, -147}, {-42, -101}, {-57, -100}, {-65, -130}, {-74, 98}, {-56, -58}, {-40, -129}, {-41, -145}}, {{2, 1}, {4, 3}, {6, 5}, {8, 7}, {10, 9}, {12, 11}, {-4, 13}, {-11, -28}, {-21, 14}, {-1, 15}, {17, 16}, {19, 18}, {-38, 20}, {22, 21}, {24, 23}, {26, 25}, {28, 27}, {-54, 29}, {31, 30}, {-44, 32}, {-45, 33}, {-37, 34}, {-5, 35}, {-27, 36}, {38, 37}, {40, 39}, {-53, 41}, {-12, 42}, {-22, 43}, {-20, 44}, {-36, 45}, {-43, 46}, {-6, 47}, {-205, 48}, {-51, -52}, {-35, 49}, {-34, 50}, {-13, 51}, {-42, 52}, {-29, 53}, {-18, -41}, {55, 54}, {-17, -26}, {-19, 56}, {-7, 57}, {-23, -188}, {59, 58}, {-10, 60}, {62, 61}, {-39, 63}, {-33, 64}, {-2, 65}, {-204, 66}, {68, 67}, {-189, 69}, {-171, 70}, {72, 71}, {74, 73}, {-203, 75}, {-3, -25}, {-24, 76}, {78, 77}, {80, 79}, {82, 81}, {-173, 83}, {-172, -187}, {85, 84}, {-86, 86}, {-50, 87}, {-202, 88}, {90, 89}, {-154, 91}, {93, 92}, {-120, 94}, {96, 95}, {-186, 97}, {99, 98}, {-69, 100}, {-156, -157}, {102, 101}, {104, 103}, {-170, -201}, {-103, 105}, {107, 106}, {-155, 108}, {-137, 109}, {-185, 110}, {-49, 111}, {-8, 112}, {-66, 113}, {-67, 114}, {116, 115}, {-169, 117}, {-141, 118}, {120, 119}, {122, 121}, {-200, 123}, {-68, -121}, {125, 124}, {-136, 126}, {-140, 127}, {-71, 128}, {-139, 129}, {-151, -184}, {-82, 130}, {-56, -101}, {132, 131}, {-9, -153}, {-40, 133}, {-138, 134}, {-83, -199}, {-84, 135}, {-90, -168}, {-65, -91}, {-102, 136}, {-135, -166}, {-72, -183}, {-87, -150}, {-181, 137}, {-125, 138}, {-55, -70}, {-85, -152}, {-106, -124}, {-89, -123}, {-198, 139}, {-57, 140}, {-105, 141}, {-167, -196}, {-81, -122}, {-182, 142}, {-99, -180}, {-100, -104}, {-116, -165}, {-98, 143}, {-117, -119}, {-88, -134}, {-197, 144}, {-73, -195}, {-92, -149}, {-118, -164}, {-58, -108}, {-107, -179}, {-109, 145}, {-93, -97}, {-115, -194}, {-114, 146}, {-113, 147}, {149, 148}, {151, 150}, {153, 152}, {155, 154}, {157, 156}, {159, 158}, {161, 160}, {163, 162}, {165, 164}, {167, 166}, {-178, -193}, {-163, -177}, {-161, -162}, {-147, -148}, {-145, -146}, {-132, -133}, {-130, -131}, {-77, -129}, {-75, -76}, {-61, -74}, {-59, -60}}}; static const HUFF_OLD_NOD_2D huffOLDNodes_h2D_0_1 = { {{-1, 1}, {3, 2}, {5, 4}, {-52, 6}, {-49, 7}, {9, 8}, {-17, 10}, {-36, 11}, {-18, 12}, {-2, -3}, {-35, 13}, {-34, -50}, {-4, -33}, {-20, 14}, {-19, -51}}, {{-1, 1}, {3, 2}, {-103, 4}, {6, 5}, {8, 7}, {-18, 9}, {11, 10}, {-87, 12}, {-17, 13}, {15, 14}, {-86, 16}, {18, 17}, {-71, 19}, {21, 20}, {-33, -35}, {-34, 22}, {-55, 23}, {-2, 24}, {-50, -102}, {26, 25}, {-49, 27}, {-69, -70}, {-39, 28}, {-65, 29}, {-66, 30}, {-54, 31}, {-19, 32}, {-23, -52}, {-51, 33}, {-81, 34}, {-82, 35}, {-3, -38}, {-85, -101}, {-67, -97}, {37, 36}, {-20, -53}, {-36, 38}, {40, 39}, {-100, 41}, {-4, -84}, {-68, 42}, {-21, 43}, {-37, 44}, {-99, 45}, {-5, -83}, {-22, 46}, {-98, 47}, {-6, -7}}, {{-1, 1}, {3, 2}, {5, 4}, {7, 6}, {9, 8}, {-154, 10}, {12, 11}, {14, 13}, {-18, 15}, {17, 16}, {19, 18}, {21, 20}, {-17, 22}, {-137, 23}, {-35, 24}, {-138, 25}, {27, 26}, {-113, 28}, {-34, 29}, {31, 30}, {33, 32}, {-122, 34}, {-33, 35}, {-73, 36}, {38, 37}, {40, 39}, {-106, 41}, {-52, 42}, {-58, -120}, {-50, 43}, {45, 44}, {-49, 46}, {-10, -103}, {-36, 47}, {-54, -90}, {-53, 48}, {-2, 49}, {-98, -153}, {-121, 50}, {-66, 51}, {-65, -72}, {-51, 52}, {-74, 53}, {-9, 54}, {-105, 55}, {-71, -82}, {-19, -55}, {-81, 56}, {58, 57}, {-83, 59}, {-68, -88}, {-89, -97}, {-70, 60}, {-3, 61}, {-67, 62}, {64, 63}, {-69, 65}, {-104, 66}, {-136, -152}, {68, 67}, {-8, -26}, {-37, 69}, {-4, 70}, {72, 71}, {-22, 73}, {-42, 74}, {-7, -20}, {76, 75}, {78, 77}, {-6, 79}, {-114, 80}, {-25, -135}, {-119, -151}, {-24, 81}, {-57, 82}, {-5, 83}, {-99, 84}, {-23, -130}, {-129, 85}, {-118, 86}, {-21, -41}, {-86, 87}, {-115, -145}, {-84, 88}, {-87, -150}, {-38, -56}, {-134, 89}, {-100, 90}, {-85, -133}, {-149, 91}, {-102, 92}, {-117, -148}, {94, 93}, {-39, 95}, {-101, 96}, {-116, 97}, {-131, -132}, {-40, 98}, {-146, -147}}, {{2, 1}, {-1, 3}, {5, 4}, {7, 6}, {9, 8}, {-205, 10}, {12, 11}, {14, 13}, {16, 15}, {-18, 17}, {19, 18}, {21, 20}, {23, 22}, {-189, 24}, {-188, 25}, {27, 26}, {-17, 28}, {-173, 29}, {31, 30}, {33, 32}, {-34, -157}, {-35, 34}, {-33, 35}, {37, 36}, {39, 38}, {41, 40}, {-50, 42}, {-49, 43}, {-141, 44}, {-204, 45}, {-2, -171}, {-172, 46}, {-66, 47}, {49, 48}, {51, 50}, {-65, 52}, {-125, 53}, {-156, 54}, {-82, 55}, {57, 56}, {59, 58}, {-19, -52}, {61, 60}, {-81, 62}, {64, 63}, {-109, -140}, {-51, 65}, {67, 66}, {-98, 68}, {70, 69}, {72, 71}, {-67, -93}, {74, 73}, {-203, 75}, {-154, 76}, {-124, 77}, {-97, -187}, {-114, 78}, {-61, 79}, {-155, 80}, {82, 81}, {-113, 83}, {-3, -146}, {-83, 84}, {-108, 85}, {-20, 86}, {-76, 87}, {-45, -77}, {-139, 88}, {90, 89}, {-69, -130}, {-129, 91}, {-36, 92}, {-99, -161}, {94, 93}, {-92, -162}, {-68, 95}, {-29, 96}, {-86, 97}, {-60, 98}, {-123, -177}, {-145, 99}, {-91, -131}, {101, 100}, {-137, -178}, {-115, 102}, {-84, -116}, {-147, 103}, {-4, 104}, {-106, -202}, {106, 105}, {-132, -186}, {-107, 107}, {-193, 108}, {-100, -120}, {-75, -170}, {-44, 109}, {-122, -163}, {-138, 110}, {-90, 111}, {-37, 112}, {-101, 113}, {-121, 114}, {116, 115}, {-103, 117}, {-74, -201}, {-21, -85}, {-53, -59}, {-117, 118}, {-148, 119}, {-5, 120}, {-169, 121}, {-105, -185}, {123, 122}, {-102, -133}, {-136, 124}, {-153, 125}, {127, 126}, {-54, 128}, {130, 129}, {-22, -104}, {-38, 131}, {-89, -118}, {-184, 132}, {-71, 133}, {-87, 134}, {-70, 135}, {-200, 136}, {-168, 137}, {-152, 138}, {-6, -23}, {-39, 139}, {-119, -199}, {141, 140}, {-55, 142}, {-7, -151}, {-183, 143}, {145, 144}, {-135, 146}, {-56, 147}, {-150, 148}, {-40, 149}, {-72, -198}, {-88, 150}, {-57, -134}, {-41, 151}, {-166, -167}, {-25, -165}, {-9, 152}, {-8, -24}, {-73, -181}, {-182, 153}, {155, 154}, {-197, 156}, {-42, -180}, {158, 157}, {-43, -149}, {-196, 159}, {-58, -164}, {-26, 160}, {162, 161}, {164, 163}, {166, 165}, {-195, 167}, {-179, -194}, {-27, -28}, {-12, -13}, {-10, -11}}}; static const HUFF_OLD_NOD_2D huffOLDNodes_h2D_1_0 = { {{-1, 1}, {-52, 2}, {4, 3}, {-18, 5}, {7, 6}, {-17, 8}, {-36, 9}, {-35, 10}, {-2, 11}, {-19, 12}, {-33, -51}, {-20, -34}, {14, 13}, {-3, -49}, {-4, -50}}, {{-1, 1}, {3, 2}, {5, 4}, {-103, 6}, {8, 7}, {-18, 9}, {11, 10}, {13, 12}, {-86, 14}, {-87, 15}, {17, 16}, {-35, 18}, {-17, 19}, {21, 20}, {-34, -71}, {23, 22}, {-50, -55}, {-33, 24}, {-69, 25}, {-2, -70}, {27, 26}, {-102, 28}, {-49, 29}, {-66, 30}, {-39, -54}, {-52, 31}, {-51, 32}, {-65, 33}, {-19, 34}, {-38, -82}, {-23, -85}, {-67, 35}, {-81, 36}, {-3, 37}, {-53, -101}, {-20, -97}, {39, 38}, {-36, 40}, {-84, 41}, {-100, 42}, {-4, -68}, {-21, 43}, {-37, 44}, {-83, 45}, {-5, -99}, {-22, 46}, {-98, 47}, {-6, -7}}, {{-1, 1}, {3, 2}, {5, 4}, {7, 6}, {9, 8}, {-154, 10}, {12, 11}, {14, 13}, {-18, 15}, {17, 16}, {-113, 18}, {20, 19}, {-137, 21}, {23, 22}, {25, 24}, {27, 26}, {-35, 28}, {-138, 29}, {-58, 30}, {-103, 31}, {-98, 32}, {34, 33}, {-122, 35}, {-120, 36}, {-17, -73}, {-34, 37}, {-106, 38}, {-50, 39}, {-83, -90}, {-74, 40}, {-52, 41}, {-66, -121}, {-33, -88}, {43, 42}, {-82, -105}, {-49, 44}, {-68, -153}, {-2, -89}, {-51, -65}, {-67, 45}, {-81, -97}, {47, 46}, {-104, 48}, {-19, 49}, {51, 50}, {53, 52}, {55, 54}, {-136, 56}, {-152, 57}, {-3, 58}, {60, 59}, {62, 61}, {64, 63}, {-36, 65}, {-20, 66}, {-53, 67}, {-114, 68}, {-57, -99}, {-72, 69}, {-69, 70}, {-42, 71}, {-151, 72}, {-119, 73}, {-84, -118}, {-135, 74}, {-4, -130}, {-115, 75}, {-26, -41}, {-87, 76}, {-56, -86}, {-100, 77}, {-37, -129}, {-21, 78}, {-38, 79}, {-71, -145}, {-134, 80}, {-85, 81}, {-150, 82}, {-5, 83}, {-133, 84}, {-102, 85}, {-22, 86}, {-23, 87}, {-54, 88}, {-149, 89}, {-117, -148}, {-70, 90}, {-6, -101}, {92, 91}, {-8, -55}, {-7, 93}, {-132, 94}, {-39, -116}, {-24, 95}, {-147, 96}, {-40, 97}, {-10, -131}, {-146, 98}, {-9, -25}}, {{2, 1}, {-1, 3}, {5, 4}, {7, 6}, {9, 8}, {11, 10}, {13, 12}, {-205, 14}, {16, 15}, {18, 17}, {20, 19}, {-18, 21}, {23, 22}, {25, 24}, {27, 26}, {29, 28}, {-188, 30}, {32, 31}, {34, 33}, {36, 35}, {-189, 37}, {39, 38}, {-35, 40}, {42, 41}, {44, 43}, {46, 45}, {-173, 47}, {49, 48}, {-34, 50}, {-17, 51}, {53, 52}, {-157, 54}, {56, 55}, {58, 57}, {-171, 59}, {-50, 60}, {62, 61}, {-66, -141}, {-172, 63}, {-125, 64}, {66, 65}, {-33, 67}, {-52, 68}, {-204, 69}, {-82, 70}, {-156, 71}, {-2, 72}, {74, 73}, {-109, 75}, {-51, -98}, {77, 76}, {-49, -140}, {79, 78}, {-146, 80}, {-124, 81}, {-61, -93}, {-19, -76}, {-81, -154}, {-65, -114}, {83, 82}, {-83, -108}, {-67, 84}, {-77, 85}, {-130, 86}, {-99, -155}, {88, 87}, {-97, 89}, {-69, -91}, {-92, 90}, {-131, 91}, {93, 92}, {-116, -187}, {-123, 94}, {-60, 95}, {-86, -139}, {97, 96}, {-68, -162}, {99, 98}, {-45, -113}, {-147, -203}, {-115, 100}, {-75, 101}, {-84, -106}, {-129, 102}, {-3, 103}, {-137, 104}, {-132, 105}, {-44, -120}, {-107, 106}, {-20, -100}, {-36, 107}, {-90, -163}, {-161, 108}, {-59, -145}, {-101, 109}, {-29, -138}, {-121, 110}, {-177, -178}, {-186, 111}, {-122, -148}, {-117, 112}, {-85, -170}, {-202, 113}, {-4, 114}, {-37, -105}, {-74, 115}, {-133, 116}, {-102, 117}, {119, 118}, {-89, -193}, {-103, 120}, {-21, -53}, {-153, 121}, {123, 122}, {125, 124}, {-185, 126}, {-104, -169}, {-201, 127}, {-136, 128}, {-118, 129}, {-87, 130}, {-5, 131}, {-38, 132}, {-54, 133}, {-70, -184}, {-71, -168}, {-22, 134}, {136, 135}, {-151, -152}, {-55, 137}, {-6, 138}, {-39, -72}, {-200, 139}, {-167, 140}, {142, 141}, {-119, -166}, {-88, 143}, {-23, -135}, {-199, 144}, {-165, 145}, {-56, -150}, {-57, -183}, {-7, 146}, {-41, 147}, {-181, 148}, {-134, 149}, {-24, -25}, {-40, 150}, {-73, 151}, {-9, 152}, {-43, 153}, {-182, -197}, {-8, -195}, {-198, 154}, {-149, 155}, {157, 156}, {159, 158}, {161, 160}, {163, 162}, {165, 164}, {167, 166}, {-194, -196}, {-179, -180}, {-58, -164}, {-28, -42}, {-26, -27}, {-12, -13}, {-10, -11}}}; static const HUFF_OLD_NOD_2D huffOLDNodes_h2D_1_1 = { {{-1, 1}, {-52, 2}, {4, 3}, {6, 5}, {-18, 7}, {-2, 8}, {-17, 9}, {-35, 10}, {-36, -51}, {-34, 11}, {-33, 12}, {-19, 13}, {-3, -20}, {-50, 14}, {-4, -49}}, {{-1, 1}, {3, 2}, {5, 4}, {-103, 6}, {8, 7}, {-18, 9}, {11, 10}, {13, 12}, {-86, 14}, {16, 15}, {-2, -35}, {-17, 17}, {-87, 18}, {-102, 19}, {21, 20}, {-69, 22}, {-34, 23}, {-19, 24}, {26, 25}, {-3, 27}, {-52, -70}, {-33, -71}, {-85, 28}, {-101, 29}, {31, 30}, {-50, 32}, {-51, 33}, {-20, 34}, {-36, 35}, {-4, -55}, {-54, 36}, {-49, -100}, {-53, 37}, {-84, 38}, {-68, 39}, {41, 40}, {-5, 42}, {-21, 43}, {-65, -66}, {-67, 44}, {-37, -99}, {-39, 45}, {-6, 46}, {-38, -83}, {-22, 47}, {-81, -82}, {-7, -98}, {-23, -97}}, {{-1, 1}, {3, 2}, {5, 4}, {7, 6}, {9, 8}, {-154, 10}, {-103, 11}, {13, 12}, {-18, 14}, {16, 15}, {-137, 17}, {19, 18}, {-35, 20}, {22, 21}, {-120, 23}, {25, 24}, {-52, 26}, {-2, 27}, {-138, 28}, {-153, 29}, {-17, 30}, {32, 31}, {34, 33}, {-34, 35}, {-19, 36}, {38, 37}, {40, 39}, {-3, 41}, {-121, 42}, {-122, 43}, {-136, -152}, {-33, 44}, {-104, 45}, {-105, 46}, {-51, -106}, {-50, 47}, {-36, 48}, {-20, 49}, {-53, -119}, {-4, 50}, {-135, -151}, {-68, 51}, {53, 52}, {-49, 54}, {56, 55}, {-118, 57}, {-88, 58}, {60, 59}, {-5, -8}, {-38, 61}, {63, 62}, {-21, 64}, {-37, -83}, {-67, 65}, {-66, -133}, {-6, 66}, {-150, 67}, {-134, 68}, {-23, -65}, {-73, -90}, {-69, -89}, {-148, 69}, {-7, -22}, {-98, -113}, {71, 70}, {-82, 72}, {-86, -149}, {-58, -81}, {-74, 73}, {75, 74}, {77, 76}, {-87, -97}, {-102, 78}, {80, 79}, {-84, 81}, {-85, 82}, {-54, 83}, {-70, 84}, {-72, 85}, {-117, 86}, {-71, 87}, {-99, 88}, {-101, 89}, {-39, -100}, {-55, 90}, {-57, 91}, {-132, 92}, {-56, 93}, {-24, -114}, {-115, 94}, {-40, -116}, {-42, -147}, {-9, -41}, {-131, 95}, {97, 96}, {-129, 98}, {-25, -130}, {-26, -146}, {-10, -145}}, {{2, 1}, {-1, 3}, {5, 4}, {7, 6}, {9, 8}, {11, 10}, {13, 12}, {-205, 14}, {16, 15}, {18, 17}, {-18, 19}, {21, 20}, {23, 22}, {-188, 24}, {26, 25}, {28, 27}, {30, 29}, {-35, 31}, {33, 32}, {35, 34}, {-171, 36}, {-189, 37}, {-204, 38}, {40, 39}, {-2, 41}, {43, 42}, {-17, 44}, {-52, 45}, {-34, 46}, {-19, 47}, {49, 48}, {-154, 50}, {52, 51}, {54, 53}, {-172, 55}, {-173, 56}, {-69, -187}, {-203, 57}, {59, 58}, {-86, 60}, {-3, 61}, {63, 62}, {-33, -50}, {-51, 64}, {-36, 65}, {-137, 66}, {-20, 67}, {69, 68}, {-120, 70}, {72, 71}, {-156, -157}, {-155, 73}, {-170, 74}, {76, 75}, {-186, -202}, {78, 77}, {80, 79}, {82, 81}, {-4, -67}, {-49, -103}, {-66, 83}, {-68, 84}, {-53, 85}, {-21, 86}, {-37, 87}, {89, 88}, {91, 90}, {93, 92}, {-138, 94}, {-140, 95}, {-141, -153}, {-139, 96}, {-201, 97}, {-185, 98}, {-121, 99}, {-169, 100}, {-5, 101}, {-136, 102}, {-65, -84}, {-83, -85}, {-82, 103}, {-70, 104}, {-54, 105}, {-38, 106}, {108, 107}, {-101, 109}, {-22, -102}, {-122, -123}, {111, 110}, {113, 112}, {-125, 114}, {-87, -124}, {-71, 115}, {-168, 116}, {-6, -200}, {-184, 117}, {-152, 118}, {-81, 119}, {121, 120}, {-105, 122}, {-106, 123}, {-99, 124}, {-98, -100}, {-23, 125}, {-104, 126}, {-39, 127}, {-135, 128}, {-55, -151}, {130, 129}, {-91, -119}, {-7, -199}, {-183, 131}, {-107, -108}, {-116, 132}, {-109, -117}, {-56, -167}, {-97, 133}, {-90, 134}, {-72, 135}, {-115, -118}, {-92, 136}, {-93, -166}, {-24, -114}, {-89, 137}, {-88, -150}, {139, 138}, {-8, 140}, {-40, 141}, {-198, 142}, {-134, 143}, {-113, 144}, {-182, 145}, {147, 146}, {-41, 148}, {-57, -181}, {-131, 149}, {151, 150}, {-25, 152}, {-132, 153}, {155, 154}, {-9, -76}, {-42, -165}, {-73, -133}, {-77, 156}, {-130, 157}, {-75, -149}, {-10, -146}, {-26, 158}, {-197, 159}, {-180, 160}, {-147, -196}, {-58, -74}, {-27, 161}, {-129, -148}, {-11, -61}, {-60, 162}, {-59, 163}, {-43, -145}, {-12, -164}, {-161, 164}, {-163, 165}, {-162, -195}, {-179, 166}, {-177, 167}, {-28, -178}, {-45, -194}, {-29, -44}, {-13, -193}}}; const HUFF_OLD_NODES huffOLDNodes = { {&huffOLDNodes_h1D_0, &huffOLDNodes_h1D_1, &huffOLDNodes_h1D_1}, {{&huffOLDNodes_h2D_0_0, &huffOLDNodes_h2D_0_1}, {&huffOLDNodes_h2D_1_0, &huffOLDNodes_h2D_1_1}, {&huffOLDNodes_h2D_0_1, &huffOLDNodes_h2D_0_1}}}; fdk-aac-2.0.2/libFDK/src/mdct.cpp000066400000000000000000000645731404225310700163330ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Josef Hoepfl, Manuel Jander, Youliy Ninov, Daniel Hagel Description: MDCT/MDST routines *******************************************************************************/ #include "mdct.h" #include "FDK_tools_rom.h" #include "dct.h" #include "fixpoint_math.h" void mdct_init(H_MDCT hMdct, FIXP_DBL *overlap, INT overlapBufferSize) { hMdct->overlap.freq = overlap; // FDKmemclear(overlap, overlapBufferSize*sizeof(FIXP_DBL)); hMdct->prev_fr = 0; hMdct->prev_nr = 0; hMdct->prev_tl = 0; hMdct->ov_size = overlapBufferSize; hMdct->prevAliasSymmetry = 0; hMdct->prevPrevAliasSymmetry = 0; hMdct->pFacZir = NULL; hMdct->pAsymOvlp = NULL; } /* This program implements the forward MDCT transform on an input block of data. The input block is in a form (A,B,C,D) where A,B,C and D are the respective 1/4th segments of the block. The program takes the input block and folds it in the form: (-D-Cr,A-Br). This block is twice shorter and here the 'r' suffix denotes flipping of the sequence (reversing the order of the samples). While folding the input block in the above mentioned shorter block the program windows the data. Because the two operations (windowing and folding) are not implemented sequentially, but together the program's structure is not easy to understand. Once the output (already windowed) block (-D-Cr,A-Br) is ready it is passed to the DCT IV for processing. */ INT mdct_block(H_MDCT hMdct, const INT_PCM *RESTRICT timeData, const INT noInSamples, FIXP_DBL *RESTRICT mdctData, const INT nSpec, const INT tl, const FIXP_WTP *pRightWindowPart, const INT fr, SHORT *pMdctData_e) { int i, n; /* tl: transform length fl: left window slope length nl: left window slope offset fr: right window slope length nr: right window slope offset See FDK_tools/doc/intern/mdct.tex for more detail. */ int fl, nl, nr; const FIXP_WTP *wls, *wrs; wrs = pRightWindowPart; /* Detect FRprevious / FL mismatches and override parameters accordingly */ if (hMdct->prev_fr == 0) { /* At start just initialize and pass parameters as they are */ hMdct->prev_fr = fr; hMdct->prev_wrs = wrs; hMdct->prev_tl = tl; } /* Derive NR */ nr = (tl - fr) >> 1; /* Skip input samples if tl is smaller than block size */ timeData += (noInSamples - tl) >> 1; /* windowing */ for (n = 0; n < nSpec; n++) { /* * MDCT scale: * + 1: fMultDiv2() in windowing. * + 1: Because of factor 1/2 in Princen-Bradley compliant windowed TDAC. */ INT mdctData_e = 1 + 1; /* Derive left parameters */ wls = hMdct->prev_wrs; fl = hMdct->prev_fr; nl = (tl - fl) >> 1; /* Here we implement a simplified version of what happens after the this piece of code (see the comments below). We implement the folding of A and B segments to (A-Br) but A is zero, because in this part of the MDCT sequence the window coefficients with which A must be multiplied are zero. */ for (i = 0; i < nl; i++) { #if SAMPLE_BITS == DFRACT_BITS /* SPC_BITS and DFRACT_BITS should be equal. */ mdctData[(tl / 2) + i] = -((FIXP_DBL)timeData[tl - i - 1] >> (1)); #else mdctData[(tl / 2) + i] = -(FIXP_DBL)timeData[tl - i - 1] << (DFRACT_BITS - SAMPLE_BITS - 1); /* 0(A)-Br */ #endif } /* Implements the folding and windowing of the left part of the sequence, that is segments A and B. The A segment is multiplied by the respective left window coefficient and placed in a temporary variable. tmp0 = fMultDiv2((FIXP_PCM)timeData[i+nl], pLeftWindowPart[i].v.im); After this the B segment taken in reverse order is multiplied by the left window and subtracted from the previously derived temporary variable, so that finally we implement the A-Br operation. This output is written to the right part of the MDCT output : (-D-Cr,A-Br). mdctData[(tl/2)+i+nl] = fMultSubDiv2(tmp0, (FIXP_PCM)timeData[tl-nl-i-1], pLeftWindowPart[i].v.re);//A*window-Br*window The (A-Br) data is written to the output buffer (mdctData) without being flipped. */ for (i = 0; i < fl / 2; i++) { FIXP_DBL tmp0; tmp0 = fMultDiv2((FIXP_PCM)timeData[i + nl], wls[i].v.im); /* a*window */ mdctData[(tl / 2) + i + nl] = fMultSubDiv2(tmp0, (FIXP_PCM)timeData[tl - nl - i - 1], wls[i].v.re); /* A*window-Br*window */ } /* Right window slope offset */ /* Here we implement a simplified version of what happens after the this piece of code (see the comments below). We implement the folding of C and D segments to (-D-Cr) but D is zero, because in this part of the MDCT sequence the window coefficients with which D must be multiplied are zero. */ for (i = 0; i < nr; i++) { #if SAMPLE_BITS == \ DFRACT_BITS /* This should be SPC_BITS instead of DFRACT_BITS. */ mdctData[(tl / 2) - 1 - i] = -((FIXP_DBL)timeData[tl + i] >> (1)); #else mdctData[(tl / 2) - 1 - i] = -(FIXP_DBL)timeData[tl + i] << (DFRACT_BITS - SAMPLE_BITS - 1); /* -C flipped at placing */ #endif } /* Implements the folding and windowing of the right part of the sequence, that is, segments C and D. The C segment is multiplied by the respective right window coefficient and placed in a temporary variable. tmp1 = fMultDiv2((FIXP_PCM)timeData[tl+nr+i], pRightWindowPart[i].v.re); After this the D segment taken in reverse order is multiplied by the right window and added from the previously derived temporary variable, so that we get (C+Dr) operation. This output is negated to get (-C-Dr) and written to the left part of the MDCT output while being reversed (flipped) at the same time, so that from (-C-Dr) we get (-D-Cr)=> (-D-Cr,A-Br). mdctData[(tl/2)-nr-i-1] = -fMultAddDiv2(tmp1, (FIXP_PCM)timeData[(tl*2)-nr-i-1], pRightWindowPart[i].v.im);*/ for (i = 0; i < fr / 2; i++) { FIXP_DBL tmp1; tmp1 = fMultDiv2((FIXP_PCM)timeData[tl + nr + i], wrs[i].v.re); /* C*window */ mdctData[(tl / 2) - nr - i - 1] = -fMultAddDiv2(tmp1, (FIXP_PCM)timeData[(tl * 2) - nr - i - 1], wrs[i].v.im); /* -(C*window+Dr*window) and flip before placing -> -Cr - D */ } /* We pass the shortened folded data (-D-Cr,A-Br) to the MDCT function */ dct_IV(mdctData, tl, &mdctData_e); pMdctData_e[n] = (SHORT)mdctData_e; timeData += tl; mdctData += tl; hMdct->prev_wrs = wrs; hMdct->prev_fr = fr; hMdct->prev_tl = tl; } return nSpec * tl; } void imdct_gain(FIXP_DBL *pGain_m, int *pGain_e, int tl) { FIXP_DBL gain_m = *pGain_m; int gain_e = *pGain_e; int log2_tl; gain_e += -MDCT_OUTPUT_GAIN - MDCT_OUT_HEADROOM + 1; if (tl == 0) { /* Dont regard the 2/N factor from the IDCT. It is compensated for somewhere * else. */ *pGain_e = gain_e; return; } log2_tl = DFRACT_BITS - 1 - fNormz((FIXP_DBL)tl); gain_e += -log2_tl; /* Detect non-radix 2 transform length and add amplitude compensation factor which cannot be included into the exponent above */ switch ((tl) >> (log2_tl - 2)) { case 0x7: /* 10 ms, 1/tl = 1.0/(FDKpow(2.0, -log2_tl) * 0.53333333333333333333) */ if (gain_m == (FIXP_DBL)0) { gain_m = FL2FXCONST_DBL(0.53333333333333333333f); } else { gain_m = fMult(gain_m, FL2FXCONST_DBL(0.53333333333333333333f)); } break; case 0x6: /* 3/4 of radix 2, 1/tl = 1.0/(FDKpow(2.0, -log2_tl) * 2.0/3.0) */ if (gain_m == (FIXP_DBL)0) { gain_m = FL2FXCONST_DBL(2.0 / 3.0f); } else { gain_m = fMult(gain_m, FL2FXCONST_DBL(2.0 / 3.0f)); } break; case 0x5: /* 0.8 of radix 2 (e.g. tl 160), 1/tl = 1.0/(FDKpow(2.0, -log2_tl) * 0.8/1.5) */ if (gain_m == (FIXP_DBL)0) { gain_m = FL2FXCONST_DBL(0.53333333333333333333f); } else { gain_m = fMult(gain_m, FL2FXCONST_DBL(0.53333333333333333333f)); } break; case 0x4: /* radix 2, nothing to do. */ break; default: /* unsupported */ FDK_ASSERT(0); break; } *pGain_m = gain_m; *pGain_e = gain_e; } INT imdct_drain(H_MDCT hMdct, FIXP_DBL *output, INT nrSamplesRoom) { int buffered_samples = 0; if (nrSamplesRoom > 0) { buffered_samples = hMdct->ov_offset; FDK_ASSERT(buffered_samples <= nrSamplesRoom); if (buffered_samples > 0) { FDKmemcpy(output, hMdct->overlap.time, buffered_samples * sizeof(FIXP_DBL)); hMdct->ov_offset = 0; } } return buffered_samples; } INT imdct_copy_ov_and_nr(H_MDCT hMdct, FIXP_DBL *pTimeData, INT nrSamples) { FIXP_DBL *pOvl; int nt, nf, i; nt = fMin(hMdct->ov_offset, nrSamples); nrSamples -= nt; nf = fMin(hMdct->prev_nr, nrSamples); FDKmemcpy(pTimeData, hMdct->overlap.time, nt * sizeof(FIXP_DBL)); pTimeData += nt; pOvl = hMdct->overlap.freq + hMdct->ov_size - 1; if (hMdct->prevPrevAliasSymmetry == 0) { for (i = 0; i < nf; i++) { FIXP_DBL x = -(*pOvl--); *pTimeData = IMDCT_SCALE_DBL(x); pTimeData++; } } else { for (i = 0; i < nf; i++) { FIXP_DBL x = (*pOvl--); *pTimeData = IMDCT_SCALE_DBL(x); pTimeData++; } } return (nt + nf); } void imdct_adapt_parameters(H_MDCT hMdct, int *pfl, int *pnl, int tl, const FIXP_WTP *wls, int noOutSamples) { int fl = *pfl, nl = *pnl; int window_diff, use_current = 0, use_previous = 0; if (hMdct->prev_tl == 0) { hMdct->prev_wrs = wls; hMdct->prev_fr = fl; hMdct->prev_nr = (noOutSamples - fl) >> 1; hMdct->prev_tl = noOutSamples; hMdct->ov_offset = 0; use_current = 1; } window_diff = (hMdct->prev_fr - fl) >> 1; /* check if the previous window slope can be adjusted to match the current * window slope */ if (hMdct->prev_nr + window_diff > 0) { use_current = 1; } /* check if the current window slope can be adjusted to match the previous * window slope */ if (nl - window_diff > 0) { use_previous = 1; } /* if both is possible choose the larger of both window slope lengths */ if (use_current && use_previous) { if (fl < hMdct->prev_fr) { use_current = 0; } } /* * If the previous transform block is big enough, enlarge previous window * overlap, if not, then shrink current window overlap. */ if (use_current) { hMdct->prev_nr += window_diff; hMdct->prev_fr = fl; hMdct->prev_wrs = wls; } else { nl -= window_diff; fl = hMdct->prev_fr; } *pfl = fl; *pnl = nl; } /* This program implements the inverse modulated lapped transform, a generalized version of the inverse MDCT transform. Setting none of the MLT_*_ALIAS_FLAG flags computes the IMDCT, setting all of them computes the IMDST. Other combinations of these flags compute type III transforms used by the RSVD60 multichannel tool for transitions between MDCT/MDST. The following description relates to the IMDCT only. If we pass the data block (A,B,C,D,E,F) to the FORWARD MDCT it will produce two outputs. The first one will be over the (A,B,C,D) part =>(-D-Cr,A-Br) and the second one will be over the (C,D,E,F) part => (-F-Er,C-Dr), since there is a overlap between consequtive passes of the algorithm. This overlap is over the (C,D) segments. The two outputs will be given sequentially to the DCT IV algorithm. At the INVERSE MDCT side we get two consecutive outputs from the IDCT IV algorithm, namely the same blocks: (-D-Cr,A-Br) and (-F-Er,C-Dr). The first of them lands in the Overlap buffer and the second is in the working one, which, one algorithm pass later will substitute the one residing in the overlap register. The IMDCT algorithm has to produce the C and D segments from the two buffers. In order to do this we take the left part of the overlap buffer(-D-Cr,A-Br), namely (-D-Cr) and add it appropriately to the right part of the working buffer (-F-Er,C-Dr), namely (C-Dr), so that we get first the C segment and later the D segment. We do this in the following way: From the right part of the working buffer(C-Dr) we subtract the flipped left part of the overlap buffer(-D-Cr): Result = (C-Dr) - flipped(-D-Cr) = C -Dr + Dr + C = 2C We divide by two and get the C segment. What we did is adding the right part of the first frame to the left part of the second one. While applying these operation we multiply the respective segments with the appropriate window functions. In order to get the D segment we do the following: From the negated second part of the working buffer(C-Dr) we subtract the flipped first part of the overlap buffer (-D-Cr): Result= - (C -Dr) - flipped(-D-Cr)= -C +Dr +Dr +C = 2Dr. After dividing by two and flipping we get the D segment.What we did is adding the right part of the first frame to the left part of the second one. While applying these operation we multiply the respective segments with the appropriate window functions. Once we have obtained the C and D segments the overlap buffer is emptied and the current buffer is sent in it, so that the E and F segments are available for decoding in the next algorithm pass.*/ INT imlt_block(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *spectrum, const SHORT scalefactor[], const INT nSpec, const INT noOutSamples, const INT tl, const FIXP_WTP *wls, INT fl, const FIXP_WTP *wrs, const INT fr, FIXP_DBL gain, int flags) { FIXP_DBL *pOvl; FIXP_DBL *pOut0 = output, *pOut1; INT nl, nr; int w, i, nrSamples = 0, specShiftScale, transform_gain_e = 0; int currAliasSymmetry = (flags & MLT_FLAG_CURR_ALIAS_SYMMETRY); /* Derive NR and NL */ nr = (tl - fr) >> 1; nl = (tl - fl) >> 1; /* Include 2/N IMDCT gain into gain factor and exponent. */ imdct_gain(&gain, &transform_gain_e, tl); /* Detect FRprevious / FL mismatches and override parameters accordingly */ if (hMdct->prev_fr != fl) { imdct_adapt_parameters(hMdct, &fl, &nl, tl, wls, noOutSamples); } pOvl = hMdct->overlap.freq + hMdct->ov_size - 1; if (noOutSamples > nrSamples) { /* Purge buffered output. */ for (i = 0; i < hMdct->ov_offset; i++) { *pOut0 = hMdct->overlap.time[i]; pOut0++; } nrSamples = hMdct->ov_offset; hMdct->ov_offset = 0; } for (w = 0; w < nSpec; w++) { FIXP_DBL *pSpec, *pCurr; const FIXP_WTP *pWindow; /* Detect FRprevious / FL mismatches and override parameters accordingly */ if (hMdct->prev_fr != fl) { imdct_adapt_parameters(hMdct, &fl, &nl, tl, wls, noOutSamples); } specShiftScale = transform_gain_e; /* Setup window pointers */ pWindow = hMdct->prev_wrs; /* Current spectrum */ pSpec = spectrum + w * tl; /* DCT IV of current spectrum. */ if (currAliasSymmetry == 0) { if (hMdct->prevAliasSymmetry == 0) { dct_IV(pSpec, tl, &specShiftScale); } else { FIXP_DBL _tmp[1024 + ALIGNMENT_DEFAULT / sizeof(FIXP_DBL)]; FIXP_DBL *tmp = (FIXP_DBL *)ALIGN_PTR(_tmp); C_ALLOC_ALIGNED_REGISTER(tmp, sizeof(_tmp)); dct_III(pSpec, tmp, tl, &specShiftScale); C_ALLOC_ALIGNED_UNREGISTER(tmp); } } else { if (hMdct->prevAliasSymmetry == 0) { FIXP_DBL _tmp[1024 + ALIGNMENT_DEFAULT / sizeof(FIXP_DBL)]; FIXP_DBL *tmp = (FIXP_DBL *)ALIGN_PTR(_tmp); C_ALLOC_ALIGNED_REGISTER(tmp, sizeof(_tmp)); dst_III(pSpec, tmp, tl, &specShiftScale); C_ALLOC_ALIGNED_UNREGISTER(tmp); } else { dst_IV(pSpec, tl, &specShiftScale); } } /* Optional scaling of time domain - no yet windowed - of current spectrum */ /* and de-scale current spectrum signal (time domain, no yet windowed) */ if (gain != (FIXP_DBL)0) { for (i = 0; i < tl; i++) { pSpec[i] = fMult(pSpec[i], gain); } } { int loc_scale = fixmin_I(scalefactor[w] + specShiftScale, (INT)DFRACT_BITS - 1); DWORD_ALIGNED(pSpec); scaleValuesSaturate(pSpec, tl, loc_scale); } if (noOutSamples <= nrSamples) { /* Divert output first half to overlap buffer if we already got enough * output samples. */ pOut0 = hMdct->overlap.time + hMdct->ov_offset; hMdct->ov_offset += hMdct->prev_nr + fl / 2; } else { /* Account output samples */ nrSamples += hMdct->prev_nr + fl / 2; } /* NR output samples 0 .. NR. -overlap[TL/2..TL/2-NR] */ if ((hMdct->pFacZir != 0) && (hMdct->prev_nr == fl / 2)) { /* In the case of ACELP -> TCX20 -> FD short add FAC ZIR on nr signal part */ for (i = 0; i < hMdct->prev_nr; i++) { FIXP_DBL x = -(*pOvl--); *pOut0 = fAddSaturate(x, IMDCT_SCALE_DBL(hMdct->pFacZir[i])); pOut0++; } hMdct->pFacZir = NULL; } else { /* Here we implement a simplified version of what happens after the this piece of code (see the comments below). We implement the folding of C and D segments from (-D-Cr) but D is zero, because in this part of the MDCT sequence the window coefficients with which D must be multiplied are zero. "pOut0" writes sequentially the C block from left to right. */ if (hMdct->prevPrevAliasSymmetry == 0) { for (i = 0; i < hMdct->prev_nr; i++) { FIXP_DBL x = -(*pOvl--); *pOut0 = IMDCT_SCALE_DBL(x); pOut0++; } } else { for (i = 0; i < hMdct->prev_nr; i++) { FIXP_DBL x = *pOvl--; *pOut0 = IMDCT_SCALE_DBL(x); pOut0++; } } } if (noOutSamples <= nrSamples) { /* Divert output second half to overlap buffer if we already got enough * output samples. */ pOut1 = hMdct->overlap.time + hMdct->ov_offset + fl / 2 - 1; hMdct->ov_offset += fl / 2 + nl; } else { pOut1 = pOut0 + (fl - 1); nrSamples += fl / 2 + nl; } /* output samples before window crossing point NR .. TL/2. * -overlap[TL/2-NR..TL/2-NR-FL/2] + current[NR..TL/2] */ /* output samples after window crossing point TL/2 .. TL/2+FL/2. * -overlap[0..FL/2] - current[TL/2..FL/2] */ pCurr = pSpec + tl - fl / 2; DWORD_ALIGNED(pCurr); C_ALLOC_ALIGNED_REGISTER(pWindow, fl); DWORD_ALIGNED(pWindow); C_ALLOC_ALIGNED_UNREGISTER(pWindow); if (hMdct->prevPrevAliasSymmetry == 0) { if (hMdct->prevAliasSymmetry == 0) { if (!hMdct->pAsymOvlp) { for (i = 0; i < fl / 2; i++) { FIXP_DBL x0, x1; cplxMultDiv2(&x1, &x0, *pCurr++, -*pOvl--, pWindow[i]); *pOut0 = IMDCT_SCALE_DBL_LSH1(x0); *pOut1 = IMDCT_SCALE_DBL_LSH1(-x1); pOut0++; pOut1--; } } else { FIXP_DBL *pAsymOvl = hMdct->pAsymOvlp + fl / 2 - 1; for (i = 0; i < fl / 2; i++) { FIXP_DBL x0, x1; x1 = -fMultDiv2(*pCurr, pWindow[i].v.re) + fMultDiv2(*pAsymOvl, pWindow[i].v.im); x0 = fMultDiv2(*pCurr, pWindow[i].v.im) - fMultDiv2(*pOvl, pWindow[i].v.re); pCurr++; pOvl--; pAsymOvl--; *pOut0++ = IMDCT_SCALE_DBL_LSH1(x0); *pOut1-- = IMDCT_SCALE_DBL_LSH1(x1); } hMdct->pAsymOvlp = NULL; } } else { /* prevAliasingSymmetry == 1 */ for (i = 0; i < fl / 2; i++) { FIXP_DBL x0, x1; cplxMultDiv2(&x1, &x0, *pCurr++, -*pOvl--, pWindow[i]); *pOut0 = IMDCT_SCALE_DBL_LSH1(x0); *pOut1 = IMDCT_SCALE_DBL_LSH1(x1); pOut0++; pOut1--; } } } else { /* prevPrevAliasingSymmetry == 1 */ if (hMdct->prevAliasSymmetry == 0) { for (i = 0; i < fl / 2; i++) { FIXP_DBL x0, x1; cplxMultDiv2(&x1, &x0, *pCurr++, *pOvl--, pWindow[i]); *pOut0 = IMDCT_SCALE_DBL_LSH1(x0); *pOut1 = IMDCT_SCALE_DBL_LSH1(-x1); pOut0++; pOut1--; } } else { /* prevAliasingSymmetry == 1 */ for (i = 0; i < fl / 2; i++) { FIXP_DBL x0, x1; cplxMultDiv2(&x1, &x0, *pCurr++, *pOvl--, pWindow[i]); *pOut0 = IMDCT_SCALE_DBL_LSH1(x0); *pOut1 = IMDCT_SCALE_DBL_LSH1(x1); pOut0++; pOut1--; } } } if (hMdct->pFacZir != 0) { /* add FAC ZIR of previous ACELP -> mdct transition */ FIXP_DBL *pOut = pOut0 - fl / 2; FDK_ASSERT(fl / 2 <= 128); for (i = 0; i < fl / 2; i++) { pOut[i] = fAddSaturate(pOut[i], IMDCT_SCALE_DBL(hMdct->pFacZir[i])); } hMdct->pFacZir = NULL; } pOut0 += (fl / 2) + nl; /* NL output samples TL/2+FL/2..TL. - current[FL/2..0] */ pOut1 += (fl / 2) + 1; pCurr = pSpec + tl - fl / 2 - 1; /* Here we implement a simplified version of what happens above the this piece of code (see the comments above). We implement the folding of C and D segments from (C-Dr) but C is zero, because in this part of the MDCT sequence the window coefficients with which C must be multiplied are zero. "pOut1" writes sequentially the D block from left to right. */ if (hMdct->prevAliasSymmetry == 0) { for (i = 0; i < nl; i++) { FIXP_DBL x = -(*pCurr--); *pOut1++ = IMDCT_SCALE_DBL(x); } } else { for (i = 0; i < nl; i++) { FIXP_DBL x = *pCurr--; *pOut1++ = IMDCT_SCALE_DBL(x); } } /* Set overlap source pointer for next window pOvl = pSpec + tl/2 - 1; */ pOvl = pSpec + tl / 2 - 1; /* Previous window values. */ hMdct->prev_nr = nr; hMdct->prev_fr = fr; hMdct->prev_tl = tl; hMdct->prev_wrs = wrs; /* Previous aliasing symmetry */ hMdct->prevPrevAliasSymmetry = hMdct->prevAliasSymmetry; hMdct->prevAliasSymmetry = currAliasSymmetry; } /* Save overlap */ pOvl = hMdct->overlap.freq + hMdct->ov_size - tl / 2; FDKmemcpy(pOvl, &spectrum[(nSpec - 1) * tl], (tl / 2) * sizeof(FIXP_DBL)); return nrSamples; } fdk-aac-2.0.2/libFDK/src/mips/000077500000000000000000000000001404225310700156315ustar00rootroot00000000000000fdk-aac-2.0.2/libFDK/src/mips/fft_rad2_mips.cpp000066400000000000000000000140421404225310700210550ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Description: dit_fft MIPS assembler replacements. *******************************************************************************/ #ifndef __FFT_RAD2_CPP__ #error \ "Do not compile this file separately. It is included on demand from fft_rad2.cpp" #endif #if defined(MIPS_DSP_LIB) #include "dsplib_util.h" #include "dsplib_dsp.h" #define FUNCTION_dit_fft #ifdef FUNCTION_dit_fft #include "mips_fft_twiddles.cpp" void dit_fft(FIXP_DBL *x, const INT ldn, const FIXP_STP *trigdata, const INT trigDataSize) { int i; int32c *din = (int32c *)x; int32c *dout = (int32c *)x; int32c scratch[1024]; int32c *twiddles; switch (ldn) { case 4: twiddles = (int32c *)__twiddles_mips_fft32_16; break; case 5: twiddles = (int32c *)__twiddles_mips_fft32_32; break; case 6: twiddles = (int32c *)__twiddles_mips_fft32_64; break; case 7: twiddles = (int32c *)__twiddles_mips_fft32_128; break; case 8: twiddles = (int32c *)__twiddles_mips_fft32_256; break; case 9: twiddles = (int32c *)__twiddles_mips_fft32_512; break; case 10: twiddles = (int32c *)__twiddles_mips_fft32_1024; break; default: FDK_ASSERT(0); break; } mips_fft32(dout, din, twiddles, scratch, ldn); for (i = 0; i < (1 << ldn); i++) { x[2 * i] = dout[i].re << 1; x[2 * i + 1] = dout[i].im << 1; } } #endif #endif /* defined(MIPS_DSP_LIB) */ fdk-aac-2.0.2/libFDK/src/mips/mips_fft_twiddles.cpp000066400000000000000000001774471404225310700220670ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Description: *******************************************************************************/ static const INT __twiddles_mips_fft32_16[] = { (0x7FFFFFFF), (0x00000000), (0x7641AF32), (0xCF043A9E), (0x5A827978), (0xA57D8646), (0x30FBC547), (0x89BE50C2), (0xFFFFFFA3), (0x80000002), (0xCF043AF8), (0x89BE50A8), (0xA57D865E), (0xA57D8670), (0x89BE5100), (0xCF043A24), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000)}; static const INT __twiddles_mips_fft32_32[] = { (0x7FFFFFFF), (0x00000000), (0x7D8A5F3C), (0xE70747B9), (0x7641AF32), (0xCF043A9E), (0x6A6D98A1), (0xB8E31318), (0x5A827978), (0xA57D8646), (0x471CED05), (0x95926772), (0x30FBC547), (0x89BE50C2), (0x18F8B888), (0x8275A0D1), (0xFFFFFFA3), (0x80000002), (0xE70747BB), (0x8275A0C3), (0xCF043AF8), (0x89BE50A8), (0xB8E3139E), (0x95926705), (0xA57D865E), (0xA57D8670), (0x959266F7), (0xB8E313B4), (0x89BE5100), (0xCF043A24), (0x8275A0BE), (0xE70747D4), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x37000000), (0x000080A3), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000)}; static const INT __twiddles_mips_fft32_64[] = { (0x7FFFFFFF), (0x00000000), (0x7F62368D), (0xF3742C9D), (0x7D8A5F3C), (0xE70747B9), (0x7A7D0559), (0xDAD7F3A2), (0x7641AF32), (0xCF043A9E), (0x70E2CBCD), (0xC3A945A1), (0x6A6D98A1), (0xB8E31318), (0x62F201C4), (0xAECC338B), (0x5A827978), (0xA57D8646), (0x5133CC8F), (0x9D0DFE52), (0x471CED05), (0x95926772), (0x3C56BAB5), (0x8F1D3461), (0x30FBC547), (0x89BE50C2), (0x25280C05), (0x8582FA8C), (0x18F8B888), (0x8275A0D1), (0x0C8BD356), (0x809DC971), (0xFFFFFFA3), (0x80000002), (0xF3742CEE), (0x809DC96B), (0xE70747BB), (0x8275A0C3), (0xDAD7F348), (0x8582FAC2), (0xCF043AF8), (0x89BE50A8), (0xC3A94669), (0x8F1D33C8), (0xB8E3139E), (0x95926705), (0xAECC33A5), (0x9D0DFE27), (0xA57D865E), (0xA57D8670), (0x9D0DFE16), (0xAECC33B9), (0x959266F7), (0xB8E313B4), (0x8F1D34AD), (0xC3A944BC), (0x89BE5100), (0xCF043A24), (0x8582FABB), (0xDAD7F360), (0x8275A0BE), (0xE70747D4), (0x809DC968), (0xF3742D08), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0xFF7B0000), (0x2E8050F1), (0x10214482), (0x1BA00005), (0x0000C0FF), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000)}; static const INT __twiddles_mips_fft32_128[] = { (0x7FFFFFFF), (0x00000000), (0x7FD8878C), (0xF9B82681), (0x7F62368D), (0xF3742C9D), (0x7E9D55FB), (0xED37EF91), (0x7D8A5F3C), (0xE70747B9), (0x7C29FBEF), (0xE0E6068F), (0x7A7D0559), (0xDAD7F3A2), (0x78848419), (0xD4E0CB28), (0x7641AF32), (0xCF043A9E), (0x73B5EBCF), (0xC945DFEB), (0x70E2CBCD), (0xC3A945A1), (0x6DCA0D27), (0xBE31E1BF), (0x6A6D98A1), (0xB8E31318), (0x66CF8103), (0xB3C01FE9), (0x62F201C4), (0xAECC338B), (0x5ED77C86), (0xAA0A5B2C), (0x5A827978), (0xA57D8646), (0x55F5A4ED), (0xA1288391), (0x5133CC8F), (0x9D0DFE52), (0x4C3FDFCC), (0x99307EC5), (0x471CED05), (0x95926772), (0x41CE1ECD), (0x9235F32C), (0x3C56BAB5), (0x8F1D3461), (0x36BA2034), (0x8C4A1440), (0x30FBC547), (0x89BE50C2), (0x2B1F34BC), (0x877B7BDD), (0x25280C05), (0x8582FA8C), (0x1F19F9F0), (0x83D60431), (0x18F8B888), (0x8275A0D1), (0x12C81090), (0x8162AA0A), (0x0C8BD356), (0x809DC971), (0x0647D949), (0x80277871), (0xFFFFFFA3), (0x80000002), (0xF9B826FB), (0x8027786E), (0xF3742CEE), (0x809DC96B), (0xED37EFB3), (0x8162AA00), (0xE70747BB), (0x8275A0C3), (0xE0E60653), (0x83D60420), (0xDAD7F348), (0x8582FAC2), (0xD4E0CB84), (0x877B7BC6), (0xCF043AF8), (0x89BE50A8), (0xC945E00A), (0x8C4A1423), (0xC3A94669), (0x8F1D33C8), (0xBE31E16E), (0x9235F309), (0xB8E3139E), (0x95926705), (0xB3C01F9D), (0x99307F35), (0xAECC33A5), (0x9D0DFE27), (0xAA0A5A88), (0xA128840F), (0xA57D865E), (0xA57D8670), (0xA12883FE), (0xAA0A5A9B), (0x9D0DFE16), (0xAECC33B9), (0x99307F26), (0xB3C01FB2), (0x959266F7), (0xB8E313B4), (0x9235F2FC), (0xBE31E184), (0x8F1D34AD), (0xC3A944BC), (0x8C4A1418), (0xC945E021), (0x89BE5100), (0xCF043A24), (0x877B7BBD), (0xD4E0CB9C), (0x8582FABB), (0xDAD7F360), (0x83D603DC), (0xE0E60764), (0x8275A0BE), (0xE70747D4), (0x8162AA22), (0xED37EECF), (0x809DC968), (0xF3742D08), (0x80277879), (0xF9B82615), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0xA4370000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0xFF770000), (0x2E8050F1), (0x1A214462), (0x04D6D583), (0x40031282), (0x6469A735), (0x0B4C99A3), (0x0AF88594), (0xF90969D7), (0x96131C92), (0x025EEA10), (0x3A1FE421), (0x614FF390), (0x1CFDC327), (0xC177E04F), (0xF4D87E82), (0x78253F39), (0xBB839F94), (0x998B3EB1), (0x0CBCC021), (0x41BEC843), (0xAC0EE121), (0x52719643), (0x909A2B1D), (0xF38931E1), (0xF41327FF), (0xF6099847), (0xA70D219C), (0x7BD52135), (0x78060F71), (0xEA3C87D8), (0x3FAF3A24), (0xE4B2C421), (0xB99D1453), (0x2741E264), (0x2239813A), (0x2944DA20), (0x441EF7C4), (0x0BD0B720), (0xED84EA26), (0x73E0C0D2), (0x678E3039), (0x21420109), (0x8607492E), (0x28CEF440), (0x022768C8), (0xF68E3611), (0x84E84D55), (0x73004C04), (0xF9B6630E), (0x677FFA8F), (0x530CB18F), (0x2C4EE310), (0xABC7537C), (0x82CFDBCB), (0x100C63A2), (0x876D75BA), (0xC683F888), (0x0CDC1E1E), (0xA600E833), (0x33036066), (0x4305049C), (0x40890713), (0x9D12532E), (0x7B6E2FE2), (0xE244CC09), (0x9FFB2082), (0xAB3735D3), (0x00000080), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0x00000000)}; static const INT __twiddles_mips_fft32_256[] = { (0x7FFFFFFF), (0x00000000), (0x7FF62181), (0xFCDBD540), (0x7FD8878C), (0xF9B82681), (0x7FA736B2), (0xF6956FB7), (0x7F62368D), (0xF3742C9D), (0x7F0991C3), (0xF054D8DA), (0x7E9D55FB), (0xED37EF91), (0x7E1D93EA), (0xEA1DEBC6), (0x7D8A5F3C), (0xE70747B9), (0x7CE3CEB0), (0xE3F47D95), (0x7C29FBEF), (0xE0E6068F), (0x7B5D03A1), (0xDDDC5B4F), (0x7A7D0559), (0xDAD7F3A2), (0x798A23A8), (0xD7D946C3), (0x78848419), (0xD4E0CB28), (0x776C4ED9), (0xD1EEF59D), (0x7641AF32), (0xCF043A9E), (0x7504D34C), (0xCC210D8B), (0x73B5EBCF), (0xC945DFEB), (0x72552C79), (0xC67322B9), (0x70E2CBCD), (0xC3A945A1), (0x6F5F02CF), (0xC0E8B67E), (0x6DCA0D27), (0xBE31E1BF), (0x6C242969), (0xBB8532C0), (0x6A6D98A1), (0xB8E31318), (0x68A69E72), (0xB64BEAB9), (0x66CF8103), (0xB3C01FE9), (0x64E8894A), (0xB140178C), (0x62F201C4), (0xAECC338B), (0x60EC383A), (0xAC64D51F), (0x5ED77C86), (0xAA0A5B2C), (0x5CB420CD), (0xA7BD229A), (0x5A827978), (0xA57D8646), (0x5842DD7E), (0xA34BDF4B), (0x55F5A4ED), (0xA1288391), (0x539B2AFB), (0x9F13C7DC), (0x5133CC8F), (0x9D0DFE52), (0x4EBFE88F), (0x9B1776CB), (0x4C3FDFCC), (0x99307EC5), (0x49B41562), (0x975961A2), (0x471CED05), (0x95926772), (0x447ACD5D), (0x93DBD6AA), (0x41CE1ECD), (0x9235F32C), (0x3F17499F), (0x90A0FD42), (0x3C56BAB5), (0x8F1D3461), (0x398CDCF3), (0x8DAAD35D), (0x36BA2034), (0x8C4A1440), (0x33DEF21F), (0x8AFB2C8E), (0x30FBC547), (0x89BE50C2), (0x2E110ABF), (0x8893B14A), (0x2B1F34BC), (0x877B7BDD), (0x2826B95E), (0x8675DC62), (0x25280C05), (0x8582FA8C), (0x2223A4D2), (0x84A2FC68), (0x1F19F9F0), (0x83D60431), (0x1C0B824E), (0x831C314A), (0x18F8B888), (0x8275A0D1), (0x15E213FD), (0x81E26C0B), (0x12C81090), (0x8162AA0A), (0x0FAB26B9), (0x80F66E30), (0x0C8BD356), (0x809DC971), (0x096A90AB), (0x8058C955), (0x0647D949), (0x80277871), (0x03242AF6), (0x8009DE81), (0xFFFFFFA3), (0x80000002), (0xFCDBD54E), (0x8009DE7F), (0xF9B826FB), (0x8027786E), (0xF6956F99), (0x8058C950), (0xF3742CEE), (0x809DC96B), (0xF054D88D), (0x80F66E47), (0xED37EFB3), (0x8162AA00), (0xEA1DEB4A), (0x81E26C2C), (0xE70747BB), (0x8275A0C3), (0xE3F47DF5), (0x831C313B), (0xE0E60653), (0x83D60420), (0xDDDC5B6F), (0x84A2FC55), (0xDAD7F348), (0x8582FAC2), (0xD7D946E3), (0x8675DC4D), (0xD4E0CB84), (0x877B7BC6), (0xD1EEF581), (0x8893B132), (0xCF043AF8), (0x89BE50A8), (0xCC210D35), (0x8AFB2CDB), (0xC945E00A), (0x8C4A1423), (0xC6732265), (0x8DAAD3B2), (0xC3A94669), (0x8F1D33C8), (0xC0E8B69C), (0x90A0FD21), (0xBE31E16E), (0x9235F309), (0xBB853205), (0x93DBD70E), (0xB8E3139E), (0x95926705), (0xB64BEAD5), (0x9759617B), (0xB3C01F9D), (0x99307F35), (0xB140180C), (0x9B177652), (0xAECC33A5), (0x9D0DFE27), (0xAC64D4D8), (0x9F13C803), (0xAA0A5A88), (0xA128840F), (0xA7BD2310), (0xA34BDEC3), (0xA57D865E), (0xA57D8670), (0xA34BDEB2), (0xA7BD2322), (0xA12883FE), (0xAA0A5A9B), (0x9F13C7F2), (0xAC64D4EB), (0x9D0DFE16), (0xAECC33B9), (0x9B177642), (0xB1401820), (0x99307F26), (0xB3C01FB2), (0x9759616C), (0xB64BEAEA), (0x959266F7), (0xB8E313B4), (0x93DBD700), (0xBB85321A), (0x9235F2FC), (0xBE31E184), (0x90A0FD14), (0xC0E8B6B2), (0x8F1D34AD), (0xC3A944BC), (0x8DAAD3A6), (0xC673227C), (0x8C4A1418), (0xC945E021), (0x8AFB2C68), (0xCC210E37), (0x89BE5100), (0xCF043A24), (0x8893B128), (0xD1EEF599), (0x877B7BBD), (0xD4E0CB9C), (0x8675DC95), (0xD7D94608), (0x8582FABB), (0xDAD7F360), (0x84A2FC4F), (0xDDDC5B88), (0x83D603DC), (0xE0E60764), (0x831C316D), (0xE3F47D14), (0x8275A0BE), (0xE70747D4), (0x81E26BFB), (0xEA1DEC5F), (0x8162AA22), (0xED37EECF), (0x80F66E44), (0xF054D8A6), (0x809DC968), (0xF3742D08), (0x8058C93B), (0xF69570B2), (0x80277879), (0xF9B82615), (0x8009DE7E), (0xFCDBD568), (0x00000000), (0x00000000), (0x00000000), (0x00000000), (0xF1FF7A00), (0xE23A8050), (0x841A2114), (0xC588DA75), (0x02400143), (0xAED6AAD7), (0x94C2545B), (0x3E46156C), (0x05BFDF27), (0xB0822075), (0xD24C7BC4), (0x62122054), (0x8CC41C49), (0x2C414972), (0x718D7C5E), (0xDEE2A3D8), (0x4544C135), (0x64292573), (0x76B51F8A), (0x4B7D6557), (0x87164907), (0xF59DBFEC), (0x04CEE9DE), (0xB4BC4C90), (0x9AFCCD30), (0x8C7A316F), (0x90FDACA7), (0x4E0D6383), (0xAEF16C0B), (0x0622020A), (0x37B6A738), (0xFFBB7D71), (0x5FEDD12F), (0xEE47FDC8), (0x3DC7913F), (0xEAEEF7FD), (0x219DADEC), (0xF9E53F5F), (0xB3D82D9E), (0xBBB79BCB), (0x43AD2AAB), (0x0E7ADBC0), (0xABBD0952), (0x1A6EF43A), (0xCD9B2A9D), (0x7222B366), (0x979E89E9), (0x02C9113F), (0x1D511466), (0x57A92206), (0x4A412075), (0xFD31D783), (0xA7542EFC), (0x24B76975), (0xB99962C7), (0x19F1D356), (0x9D12C3CB), (0x6D37D542), (0x29E7214D), (0x8900F645), (0xB154AD4A), (0xF047D988), (0x3F2E5E77), (0xAFB68C49), (0xC9F19E51), (0x6192F53A), (0x5AE3D080), (0x1DA6C677), (0xD485D50C), (0x9AA7E8B2), (0xF55EE1D5), (0xC00B6932), (0x85ACE912), (0x8208B1B9), (0x80E14170), (0xA1D67C3F), (0x058035B3), (0xA4FE36CA), (0xBEE547E4), (0xEF276052), (0x42AA4388), (0x9C11A2ED), (0x10202541), (0x2D480910), (0xB09B1AC4), (0x19E7D17D), (0x52082450), (0x3EED1705), (0xF20FF8A1), (0x89F6E17F), (0x17D7DC00), (0x09B9F2C3), (0x6378968C), (0xAF0607F6), (0xFDFDE0C9), (0x3CE3DFEE), (0x5168229E), (0x7CF79734), (0xF5FFF56F), (0x30700093), (0x135F5C75), (0xE6D73EEE), (0x6E400DF9), (0x58252ACB), (0x557311CA), (0x5539B303), (0x56557355), (0xB6BDACEF), (0x59F9428D), (0x9AE63020), (0x558FF7E7), (0x7955806D), (0x09D549F3), (0x603BF5B7), (0x3134413B), (0xBCA7C1AA), (0xA98D1339), (0x57B29C97), (0x50F1FF07), (0xE4A13980), (0xD0881A21), (0xC1FFE30A), (0xB0A66C19), (0x18416490), (0x160E0214), (0xA584CBB5), (0x02AAA82A), (0x73981340), (0x964A24DE), (0x6E829FC4), (0xDFD62360), (0x4C3BF152), (0x38721574), (0xD46BC912), (0x1EB7FDD8), (0x1CA6CBC0), (0xBBCBC111), (0xA6FD1403), (0x76C688D9), (0xB47186CB), (0x5E98936D), (0x9BBF7E2E), (0x8B7FBCEC), (0x9DAF7EF5), (0x7E3A79F1), (0x9C896ACD), (0x628F249A), (0xDD6E51AF), (0xAF73994B), (0x5966995D), (0xB7620597), (0x369B1E54), (0x7CEDC787), (0xE7B15B1F), (0xB36F5F7D), (0xB5B73EF6), (0xD76EEEE2), (0x14E86186), (0xCC159291), (0x4EC85CA3), (0xEC3C704E), (0x0B6633B3), (0x4D992D4B), (0xEB1B5774), (0xDFB9D508), (0x11783719), (0x77BB10B0), (0xFCDF8350), (0xD34A7C65), (0xEF6E57BB), (0xECD56C68), (0x8EB7CA36), (0x2F26B136), (0x55631269), (0xEF1213E1), (0xA56AB844), (0xAA4B1D64), (0x64341EA9), (0x0CE9DB48), (0x460C2144), (0xADC50205), (0x515C472F), (0x304C29BD), (0x76AF8711), (0x557CADB7), (0x2A9BBC8E), (0xE50E145B), (0x9C2031FD), (0xD9121A8C), (0xF51A96FD), (0x45CCBEBF), (0x5F884871), (0x66DAE291), (0x95ADC9E5), (0x33CF304D), (0x23C87DCE), (0xA7014FB3), (0xFB351E71), (0x8D60F66F), (0xBFD6DDD5), (0x151FE6F1), (0xDCC7935A), (0x0CACFBF0), (0xEE55A371), (0xBE46B755), (0x0F79DCD2), (0x23DFF76D), (0x83DE76D7), (0x8B0E3AEE), (0xB01BFB44), (0x4E3619FA), (0xA19F9E64), (0xBD492AD9), (0x0899E45F), (0x8156E41D), (0x5DA539BA), (0x0AF35B2C), (0xDB89BAAE), (0xA128C966), (0xA02CEAEE), (0xC5A1B291), (0x08CBA93E), (0x90FD887A), (0x60304386), (0x6A59F1EA), (0xA461E67B), (0xC1622B76), (0x6FA2F906), (0xAA3A401A), (0xB68EA47B), (0xC617A889), (0x6024A57B), (0x2DD53555), (0xF1FFEA30), (0x813A8050), (0x8D1A21B4), (0xFFE72CF7), (0xA37415E1), (0x30A2C830), (0xA3A1CA28), (0x4C5E3586), (0x309861B9), (0x1F49850D), (0x6164C748), (0x9D7048E5), (0xC124569E), (0xCCB0E111), (0x15B2BB86), (0x18371D9C), (0xB2ABD94D), (0x8EC656B3), (0xBF010995)}; static const INT __twiddles_mips_fft32_512[] = { (0x7FFFFFFF), (0x00000000), (0x7FFD8859), (0xFE6DE2E0), (0x7FF62181), (0xFCDBD540), (0x7FE9CBBE), (0xFB49E6A3), (0x7FD8878C), (0xF9B82681), (0x7FC25595), (0xF826A464), (0x7FA736B2), (0xF6956FB7), (0x7F872BF2), (0xF5049800), (0x7F62368D), (0xF3742C9D), (0x7F3857F4), (0xF1E43D1C), (0x7F0991C3), (0xF054D8DA), (0x7ED5E5C6), (0xEEC60F3C), (0x7E9D55FB), (0xED37EF91), (0x7E5FE490), (0xEBAA8944), (0x7E1D93EA), (0xEA1DEBC6), (0x7DD6668D), (0xE8922621), (0x7D8A5F3C), (0xE70747B9), (0x7D3980ED), (0xE57D5FE4), (0x7CE3CEB0), (0xE3F47D95), (0x7C894BDA), (0xE26CB010), (0x7C29FBEF), (0xE0E6068F), (0x7BC5E296), (0xDF609002), (0x7B5D03A1), (0xDDDC5B4F), (0x7AEF6325), (0xDC59778B), (0x7A7D0559), (0xDAD7F3A2), (0x7A05EEA8), (0xD957DE6F), (0x798A23A8), (0xD7D946C3), (0x7909A935), (0xD65C3B98), (0x78848419), (0xD4E0CB28), (0x77FAB98A), (0xD367044F), (0x776C4ED9), (0xD1EEF59D), (0x76D94983), (0xD078AD93), (0x7641AF32), (0xCF043A9E), (0x75A585D9), (0xCD91AB55), (0x7504D34C), (0xCC210D8B), (0x745F9DD3), (0xCAB26FB2), (0x73B5EBCF), (0xC945DFEB), (0x7307C3C9), (0xC7DB6C45), (0x72552C79), (0xC67322B9), (0x719E2CDE), (0xC50D1164), (0x70E2CBCD), (0xC3A945A1), (0x7023109C), (0xC247CD62), (0x6F5F02CF), (0xC0E8B67E), (0x6E96A995), (0xBF8C0DD8), (0x6DCA0D27), (0xBE31E1BF), (0x6CF934E8), (0xBCDA3EAE), (0x6C242969), (0xBB8532C0), (0x6B4AF258), (0xBA32CA42), (0x6A6D98A1), (0xB8E31318), (0x698C2487), (0xB79619C6), (0x68A69E72), (0xB64BEAB9), (0x67BD0FCC), (0xB5049380), (0x66CF8103), (0xB3C01FE9), (0x65DDFBD6), (0xB27E9D43), (0x64E8894A), (0xB140178C), (0x63EF3286), (0xB0049AA9), (0x62F201C4), (0xAECC338B), (0x61F10027), (0xAD96ED77), (0x60EC383A), (0xAC64D51F), (0x5FE3B366), (0xAB35F58C), (0x5ED77C86), (0xAA0A5B2C), (0x5DC79D9C), (0xA8E2112C), (0x5CB420CD), (0xA7BD229A), (0x5B9D1166), (0xA69B9B7D), (0x5A827978), (0xA57D8646), (0x5964649B), (0xA462EEB2), (0x5842DD7E), (0xA34BDF4B), (0x571DEEED), (0xA238627B), (0x55F5A4ED), (0xA1288391), (0x54CA0A2E), (0xA01C4C5C), (0x539B2AFB), (0x9F13C7DC), (0x52691241), (0x9E0EFF9D), (0x5133CC8F), (0x9D0DFE52), (0x4FFB6572), (0x9C10CD90), (0x4EBFE88F), (0x9B1776CB), (0x4D8162D8), (0x9A22043F), (0x4C3FDFCC), (0x99307EC5), (0x4AFB6C9B), (0x9842F048), (0x49B41562), (0x975961A2), (0x4869E657), (0x9673DB8C), (0x471CED05), (0x95926772), (0x45CD356F), (0x94B50D74), (0x447ACD5D), (0x93DBD6AA), (0x4325C102), (0x9306CAE7), (0x41CE1ECD), (0x9235F32C), (0x4073F245), (0x9169567D), (0x3F17499F), (0x90A0FD42), (0x3DB8324C), (0x8FDCEF37), (0x3C56BAB5), (0x8F1D3461), (0x3AF2EEBA), (0x8E61D331), (0x398CDCF3), (0x8DAAD35D), (0x38249413), (0x8CF83C62), (0x36BA2034), (0x8C4A1440), (0x354D9033), (0x8BA06221), (0x33DEF21F), (0x8AFB2C8E), (0x326E5505), (0x8A5A7A4D), (0x30FBC547), (0x89BE50C2), (0x2F875216), (0x8926B65A), (0x2E110ABF), (0x8893B14A), (0x2C98FBD1), (0x88054682), (0x2B1F34BC), (0x877B7BDD), (0x29A3C40F), (0x86F656AC), (0x2826B95E), (0x8675DC62), (0x26A82174), (0x85FA114F), (0x25280C05), (0x8582FA8C), (0x23A688D3), (0x85109CF7), (0x2223A4D2), (0x84A2FC68), (0x209F6FE1), (0x843A1D62), (0x1F19F9F0), (0x83D60431), (0x1D935011), (0x8376B42E), (0x1C0B824E), (0x831C314A), (0x1A829FC0), (0x82C67F00), (0x18F8B888), (0x8275A0D1), (0x176DD9E1), (0x82299974), (0x15E213FD), (0x81E26C0B), (0x1455771D), (0x81A01B80), (0x12C81090), (0x8162AA0A), (0x1139F0A7), (0x812A1A36), (0x0FAB26B9), (0x80F66E30), (0x0E1BC326), (0x80C7A813), (0x0C8BD356), (0x809DC971), (0x0AFB67B2), (0x8078D407), (0x096A90AB), (0x8058C955), (0x07D95BB6), (0x803DAA6D), (0x0647D949), (0x80277871), (0x04B618DF), (0x8016343D), (0x03242AF6), (0x8009DE81), (0x01921D0C), (0x800277A7), (0xFFFFFFA3), (0x80000002), (0xFE6DE338), (0x800277A6), (0xFCDBD54E), (0x8009DE7F), (0xFB49E665), (0x80163444), (0xF9B826FB), (0x8027786E), (0xF826A48E), (0x803DAA69), (0xF6956F99), (0x8058C950), (0xF5049793), (0x8078D418), (0xF3742CEE), (0x809DC96B), (0xF1E43D1E), (0x80C7A80C), (0xF054D88D), (0x80F66E47), (0xEEC60F9D), (0x812A1A2D), (0xED37EFB3), (0x8162AA00), (0xEBAA8927), (0x81A01B75), (0xEA1DEB4A), (0x81E26C2C), (0xE8922662), (0x82299967), (0xE70747BB), (0x8275A0C3), (0xE57D5F89), (0x82C67F27), (0xE3F47DF5), (0x831C313B), (0xE26CB031), (0x8376B41E), (0xE0E60653), (0x83D60420), (0xDF608F69), (0x843A1D92), (0xDDDC5B6F), (0x84A2FC55), (0xDC59776E), (0x85109CE4), (0xDAD7F348), (0x8582FAC2), (0xD957DECD), (0x85FA113A), (0xD7D946E3), (0x8675DC4D), (0xD65C3B40), (0x86F656E9), (0xD4E0CB84), (0x877B7BC6), (0xD367046F), (0x8805466A), (0xD1EEF581), (0x8893B132), (0xD078AD3C), (0x8926B6A0), (0xCF043AF8), (0x89BE50A8), (0xCD91AB39), (0x8A5A7A32), (0xCC210D35), (0x8AFB2CDB), (0xCAB2700B), (0x8BA06204), (0xC945E00A), (0x8C4A1423), (0xC7DB6C2A), (0x8CF83C44), (0xC6732265), (0x8DAAD3B2), (0xC50D109F), (0x8E61D388), (0xC3A94669), (0x8F1D33C8), (0xC247CDF0), (0x8FDCEF16), (0xC0E8B69C), (0x90A0FD21), (0xBF8C0DF6), (0x9169565A), (0xBE31E16E), (0x9235F309), (0xBCDA3E5E), (0x9306CB49), (0xBB853205), (0x93DBD70E), (0xBA32CB35), (0x94B50D09), (0xB8E3139E), (0x95926705), (0xB79619E2), (0x9673DB66), (0xB64BEAD5), (0x9759617B), (0xB5049334), (0x9842F06B), (0xB3C01F9D), (0x99307F35), (0xB27E9C92), (0x9A2204B0), (0xB140180C), (0x9B177652), (0xB0049B27), (0x9C10CD15), (0xAECC33A5), (0x9D0DFE27), (0xAD96ED91), (0x9E0EFFC3), (0xAC64D4D8), (0x9F13C803), (0xAB35F545), (0xA01C4CD8), (0xAA0A5A88), (0xA128840F), (0xA8E211A2), (0xA23861F5), (0xA7BD2310), (0xA34BDEC3), (0xA69B9B96), (0xA462EE82), (0xA57D865E), (0xA57D8670), (0xA462EE70), (0xA69B9BA8), (0xA34BDEB2), (0xA7BD2322), (0xA23861E4), (0xA8E211B5), (0xA12883FE), (0xAA0A5A9B), (0xA01C4CC7), (0xAB35F559), (0x9F13C7F2), (0xAC64D4EB), (0x9E0EFFB3), (0xAD96EDA5), (0x9D0DFE16), (0xAECC33B9), (0x9C10CD05), (0xB0049B3B), (0x9B177642), (0xB1401820), (0x9A2204A1), (0xB27E9CA6), (0x99307F26), (0xB3C01FB2), (0x9842F05C), (0xB5049349), (0x9759616C), (0xB64BEAEA), (0x9673DB57), (0xB79619F7), (0x959266F7), (0xB8E313B4), (0x94B50E13), (0xBA32C99E), (0x93DBD700), (0xBB85321A), (0x9306CB3C), (0xBCDA3E74), (0x9235F2FC), (0xBE31E184), (0x9169564D), (0xBF8C0E0C), (0x90A0FD14), (0xC0E8B6B2), (0x8FDCEF09), (0xC247CE07), (0x8F1D34AD), (0xC3A944BC), (0x8E61D37C), (0xC50D10B6), (0x8DAAD3A6), (0xC673227C), (0x8CF83C39), (0xC7DB6C41), (0x8C4A1418), (0xC945E021), (0x8BA061F9), (0xCAB27022), (0x8AFB2C68), (0xCC210E37), (0x8A5A7A8D), (0xCD91AA66), (0x89BE5100), (0xCF043A24), (0x8926B697), (0xD078AD53), (0x8893B128), (0xD1EEF599), (0x88054661), (0xD3670487), (0x877B7BBD), (0xD4E0CB9C), (0x86F6568E), (0xD65C3C4A), (0x8675DC95), (0xD7D94608), (0x85FA1180), (0xD957DDF1), (0x8582FABB), (0xDAD7F360), (0x85109CDD), (0xDC597787), (0x84A2FC4F), (0xDDDC5B88), (0x843A1D4B), (0xDF60907A), (0x83D603DC), (0xE0E60764), (0x8376B454), (0xE26CAF51), (0x831C316D), (0xE3F47D14), (0x82C67F21), (0xE57D5FA2), (0x8275A0BE), (0xE70747D4), (0x82299962), (0xE892267C), (0x81E26BFB), (0xEA1DEC5F), (0x81A01B48), (0xEBAA8A3D), (0x8162AA22), (0xED37EECF), (0x812A1A4C), (0xEEC60EB9), (0x80F66E44), (0xF054D8A6), (0x80C7A809), (0xF1E43D38), (0x809DC968), (0xF3742D08), (0x8078D3FF), (0xF50498AB), (0x8058C93B), (0xF69570B2), (0x803DAA77), (0xF826A3A8), (0x80277879), (0xF9B82615), (0x80163443), (0xFB49E67F), (0x8009DE7E), (0xFCDBD568), (0x800277A6), (0xFE6DE352), (0x465220BB), (0xC66E1FFC), (0x2C61D356), (0xAFFB176D), (0x88067E23), (0x3796F008), (0xC8D128F3), (0xB661A5FA), (0xF19CE4F0), (0x3F7D0345), (0xE97E6030), (0x4ABA91CF), (0xDEEDAB0C), (0x35FD1D77), (0xDD636439), (0x2F8B794C), (0x764E1D32), (0x7A8D0B1B), (0x8EACB27F), (0x577AD67B), (0x2B3FE2C5), (0x8ACD024A), (0xEDF5C42C), (0xF1CA951C), (0xB98B6D04), (0x6D12B864), (0x50E74D08), (0x73D1503B), (0xC7D3F6A9), (0x1F365E97), (0xE1340E59), (0x89D21B9D), (0x9B756733), (0x9024CD50), (0xE2C7FF52), (0xD9F061EF), (0x77151392), (0xA0E4A004), (0x60BE820F), (0xC7603BAF), (0xA62E3DA9), (0xA400E11F), (0x155111D4), (0x9030A374), (0x14185124), (0x8D043A00), (0x50945258), (0x20A6653E), (0xA19BA8B9), (0x3CFBE692), (0x91471D98), (0x0B9D03FF), (0xA5BA8E29), (0xFFC8EB22), (0x142CB1B0), (0x55317319), (0x33ADC77C), (0xF69BADCE), (0x78423D3C), (0xBE67BBF0), (0x7D88BD15), (0x1E9EF673), (0x512098B3), (0x326A3D3D), (0x161EF3B6), (0x0E4C4ED5), (0x4E9F9982), (0xD0AB6A36), (0xF17DD377), (0x5E53F31C), (0xBE9112EA), (0x3DE1B0FF), (0xD7F39AE7), (0xA34B9635), (0x49CFEE6C), (0x8A063AC6), (0xBA79BC0D), (0x6A588782), (0x439BE593), (0x0A5F7E8D), (0xD2FC2561), (0x4698AC8F), (0x247C3BC2), (0x5826523D), (0x4D2BD753), (0x4B8BCC17), (0x9B743503), (0x9F84E434), (0x8B97B41A), (0x3CFCE75F), (0x32D2D487), (0x3E80B4AF), (0x42F8B0EA), (0xEC0C81EE), (0x17279309), (0x681FA0F3), (0x8BAD208E), (0xD11792C6), (0xF1FF07B3), (0x21338050), (0x8A1A21A4), (0xFF0358D8), (0x416C13C1), (0x46348858), (0x83820519), (0xABC08822), (0x6A1A5200), (0x0C83122A), (0xA67FDF06), (0x0629F970), (0xEFAA2743), (0x0C00AD95), (0xFF09F5CF), (0xC784DB8B), (0xCB00BA7D), (0x6220909F), (0x7149B7DF), (0x5BD3EBEC), (0x0485EF71), (0x07FE23B1), (0x80737668), (0x42A32645), (0xD6EDBB18), (0x407E72C5), (0x901402E0), (0x0C1B7905), (0x01CE11FC), (0x00C64914), (0x3037978A), (0xFABC9190), (0x43A0AC89), (0xB9998370), (0xD6D69209), (0xF132455E), (0x9DAFF482), (0xA2D635C2), (0xBD976B5E), (0x79ED93ED), (0x7D42CA22), (0x2E23F9C1), (0xCF0E7521), (0x96D88AC1), (0x420C2845), (0x1E19573D), (0xF8FE9279), (0xC1495100), (0x201249C2), (0x8B183723), (0x450615A5), (0x004C0112), (0x85EB5235), (0x1BD8BD48), (0xA2914F03), (0x51420EAC), (0xCE788470), (0x0508030B), (0xA4CD2A6C), (0x9626C062), (0xF88D2DE5), (0xA9D6B399), (0xB3C57BFD), (0xCB1E9C4C), (0x004A5094), (0xD0CBC42A), (0x3BE1A694), (0x3C353FA7), (0xA5697C57), (0xFE961665), (0xED6B7C7D), (0x433DCF8D), (0xDA7D6C24), (0x407112C1), (0x4B9D285D), (0x6C9A2F57), (0x1BF9FAEF), (0x4EE3E2E9), (0xE5AE7E92), (0x60A11D3F), (0x4B441B63), (0xC8419BB1), (0x2C4F6DDD), (0xCEC3CF6D), (0x325A65CA), (0x818FC6A4), (0xC55E5E5B), (0x277445A3), (0x6B3289F7), (0x09147D55), (0x4F378D53), (0x588B0DC8), (0xF70BA347), (0xCBC43FAE), (0xD98E6639), (0x07AF9ED0), (0xE2E6305A), (0x5E003571), (0x18523E7A), (0xF70D1E27), (0xD7288950), (0x12C11132), (0xFF1C9D17), (0x318050F1), (0x1A219821), (0xBF21998A), (0x6C13C1FF), (0x8C605861), (0x0A183954), (0x7BE4C1B6), (0xA2CBC410), (0x7CC00A55), (0x5E58A973), (0x1122EF86), (0x80568E39), (0x2310C300), (0x0E152D0E), (0x9E017C1C), (0x851DFC26), (0xD1B1E662), (0x81BA10B0), (0x96C2E977), (0x8688E419), (0x97C94989), (0x250D535E), (0x35FDE732), (0x147AD183), (0xCF9D75C0), (0x6B2C183C), (0x1D326489), (0xC0807D98), (0xC081622E), (0x15C87B87), (0x7404598C), (0xFC3C7FA6), (0x28BA8CF5), (0xE311EA9B), (0xEABFD4A5), (0x8768834A), (0x0F7D7440), (0x40593152), (0xCF1E8BF5), (0xE32017B6), (0x39E66070), (0x560ECE9D), (0xF4132C77), (0x722945C1), (0x96ECB5D5), (0x0C100C06), (0x30E96C02), (0xE0B0159B), (0x163A40CB), (0x1B28B750), (0x11420E11), (0x34DC5981), (0xD2769C3C), (0x569402D0), (0xE4F157C0), (0x120DF24C), (0x40D86153), (0x1942CD11), (0x22484C3B), (0x8B299144), (0x58F1AAF8), (0xB2CD4777), (0x4A53AB02), (0x55A5DAD4), (0xC8D58D79), (0x6CAEC011), (0x1C955C84), (0xDE08A6C6), (0xD402AF7D), (0x695E6FA4), (0x4DA863B9), (0x5118070C), (0x5A97EF1C), (0x8880B388), (0x4D596B2D), (0x37D74DCF), (0x58138B38), (0x5B7BCD4E), (0xAE6C4422), (0x4E5D1AF1), (0xDE7396A5), (0xB442F217), (0x0DF7A006), (0x96EEF8A2), (0xE2847B26), (0x68B9F67B), (0x8FE1FA65), (0xE0A87E56), (0x1BE8FC4C), (0xBBD65F9D), (0xE6F02C9D), (0x867C6EEC), (0x43FAB725), (0xA4144419), (0x69BB6AAF), (0xCE27B744), (0x022241A9), (0xC05DF14E), (0x8050F1FF), (0x218C2131), (0xBE25941A), (0x9041CD06), (0x1EE8A9C0), (0xD530FEDD), (0x7495C808), (0x050918A6), (0x3CE732E9), (0xF87262FC), (0x01B812F1), (0x1FBA4842), (0x3E86C7F2), (0x270DB358), (0xF9D22834), (0x74D4F469), (0x9A6E87D2), (0x8CBB0A50), (0xC0784B92), (0x278DA581), (0x9221A7D5), (0xDE801B54), (0x4B531FDA), (0xC8B91F91), (0x7A90DA40), (0x2967A429), (0x50CC1D8D), (0x9AE9110E), (0x50F250EE), (0xD695B072), (0xFC7A5718), (0x1F1A26E6), (0x7D5ADD4D), (0xC8978597), (0xEB326DC3), (0x384E9824), (0xC3B9D78C), (0xD8D2E214), (0xEFFBB048), (0x00AB7714), (0x9D89CA13), (0xF70688D0), (0xADDA3A37), (0x5FBE5943), (0x4ED9D613), (0x8A9DCC93), (0x19814236), (0x09A68BCC), (0xCEAC33AC), (0x0BA70BB3), (0x326272E1), (0x2A8C6D20), (0x23083D0B), (0xFA809C02), (0x5825C6C2), (0x061B4695), (0x8DB97748), (0x5071E584), (0x08760959), (0x44EA8D04), (0x5CE6E2AA), (0x009AD176), (0xAEC64F19), (0xCADDA632), (0x8D795445), (0x16E3EA24), (0xF12648CA), (0xD5CC350C), (0x615EBAD7), (0x4E769FB6), (0x6C5EB380), (0x5BB46425), (0xEAB6C77C), (0x10945408), (0xC1291B86), (0x913B71A5), (0x003BA329), (0x91812854), (0xE200CB12), (0x7A4696A8), (0x76E55D25), (0xD6D3C577), (0x8B1B79B1), (0x296B04CD), (0xA024CC35), (0x3341A477), (0x8EC12358), (0xF5982B2F), (0x36F120C9), (0x79DBE799), (0xED74F092), (0xA1031AE3), (0xB2D98C6B), (0x358CAB73), (0x41153C19), (0x474D4548), (0xA7E834A9), (0x999738AE), (0x50F1FFE8), (0x88612F80), (0xFD931A21), (0x23839ABD), (0xBE604701), (0x812274B9), (0x4CA9EA32), (0xE5F2D207), (0xC1CF4653), (0x43561B23), (0x53B2E277), (0x4F2B00EC), (0xF2B3BA80), (0xFC708CC5), (0xE7F63F00), (0x26120342), (0x1A49C7A8), (0xA9350A7C), (0xE530A06C), (0x5CB9E550), (0xB1672913), (0x65FA76F9), (0xC9F92DCA), (0x7D4E9195), (0x4DBB8265), (0x2036E26C), (0xEDC026E6), (0xB9A617E0), (0x6469F0EE), (0xE9DAABB2), (0x26063210), (0xAC545DF4), (0x134E035A), (0x216EDE10), (0xEC81158D), (0x122A2CAE), (0xA263D0E1), (0x6CA1FB28), (0xBE287DF0), (0x548104E8), (0x35987D93), (0xBC3E549C), (0x3ACEFD64), (0x3348CD92), (0x914472FD), (0x80F28098), (0xDF3FABCD), (0x24D840AC), (0xBA95321C), (0xDC211D20), (0x15041942), (0x61858ABD), (0x546084A1), (0xCEDC07F4), (0xEAB2DE00), (0x045BC1E8), (0x82B2AE86), (0xE4BF2823), (0x9A01B47A), (0x1769438E), (0x015C7986), (0x5FDFF97D), (0x5F28D62C), (0x83C84C8C), (0xDBAC601A), (0x1AE2844F), (0x79F9DC1C), (0x01C3AB5C), (0xC971D839), (0x44DEA398), (0xC0A64EA1), (0x387E24A5), (0x0600D59F), (0x5169B371), (0xEC782B6B), (0xF7FF769D), (0x317B0AE8), (0xE33846C7), (0x2FE05849), (0x03273853), (0x2ED27A88), (0x24DF4343), (0xFB03C70F), (0x82783231), (0xD31C0D29), (0xADC5BD8B), (0x7D024F1B), (0x91C18D19), (0xA8EE58E2), (0x4C23A59F), (0xC61DCA9F), (0x03BC91EC), (0x51986147), (0x05B8BE24), (0x872AC918), (0x8050F1FF), (0x2184812F), (0xB20D941A), (0x9141CB0C), (0xF600A340), (0x80E82AC2), (0x66028631), (0x120FFA45), (0x889F24CB), (0x81F0BB27), (0x3100D51A), (0x2309A0B3), (0xAE004718), (0xAA584A8D), (0x9432DD2E), (0xDD0E93FA), (0xC12993A6), (0x6E28035D), (0x9BF7772B), (0x1D608C32)}; static const INT __twiddles_mips_fft32_1024[] = { (0x7FFFFFFF), (0x00000000), (0x7FFF6215), (0xFF36F078), (0x7FFD8859), (0xFE6DE2E0), (0x7FFA72D0), (0xFDA4D929), (0x7FF62181), (0xFCDBD540), (0x7FF09476), (0xFC12D91B), (0x7FE9CBBE), (0xFB49E6A3), (0x7FE1C76A), (0xFA80FFCE), (0x7FD8878C), (0xF9B82681), (0x7FCE0C3D), (0xF8EF5CBC), (0x7FC25595), (0xF826A464), (0x7FB563B2), (0xF75DFF6B), (0x7FA736B2), (0xF6956FB7), (0x7F97CEBB), (0xF5CCF73E), (0x7F872BF2), (0xF5049800), (0x7F754E7E), (0xF43C53CB), (0x7F62368D), (0xF3742C9D), (0x7F4DE450), (0xF2AC2473), (0x7F3857F4), (0xF1E43D1C), (0x7F2191B2), (0xF11C7895), (0x7F0991C3), (0xF054D8DA), (0x7EF05860), (0xEF8D5FC8), (0x7ED5E5C6), (0xEEC60F3C), (0x7EBA3A38), (0xEDFEE930), (0x7E9D55FB), (0xED37EF91), (0x7E7F3954), (0xEC71244A), (0x7E5FE490), (0xEBAA8944), (0x7E3F5800), (0xEAE4208A), (0x7E1D93EA), (0xEA1DEBC6), (0x7DFA98A7), (0xE957ED00), (0x7DD6668D), (0xE8922621), (0x7DB0FDF5), (0xE7CC9912), (0x7D8A5F3C), (0xE70747B9), (0x7D628AC7), (0xE642341C), (0x7D3980ED), (0xE57D5FE4), (0x7D0F4217), (0xE4B8CD16), (0x7CE3CEB0), (0xE3F47D95), (0x7CB72721), (0xE3307347), (0x7C894BDA), (0xE26CB010), (0x7C5A3D52), (0xE1A935F1), (0x7C29FBEF), (0xE0E6068F), (0x7BF88830), (0xE02323EA), (0x7BC5E296), (0xDF609002), (0x7B920B86), (0xDE9E4C5B), (0x7B5D03A1), (0xDDDC5B4F), (0x7B26CB49), (0xDD1ABE41), (0x7AEF6325), (0xDC59778B), (0x7AB6CB9A), (0xDB98888E), (0x7A7D0559), (0xDAD7F3A2), (0x7A4210DE), (0xDA17BA63), (0x7A05EEA8), (0xD957DE6F), (0x79C89F71), (0xD898621A), (0x798A23A8), (0xD7D946C3), (0x794A7C11), (0xD71A8EBA), (0x7909A935), (0xD65C3B98), (0x78C7AB9E), (0xD59E4EF9), (0x78848419), (0xD4E0CB28), (0x78403321), (0xD423B181), (0x77FAB98A), (0xD367044F), (0x77B417D4), (0xD2AAC4EB), (0x776C4ED9), (0xD1EEF59D), (0x77235F35), (0xD13397FA), (0x76D94983), (0xD078AD93), (0x768E0EA9), (0xCFBE38AD), (0x7641AF32), (0xCF043A9E), (0x75F42C0B), (0xCE4AB5A6), (0x75A585D9), (0xCD91AB55), (0x7555BD47), (0xCCD91D37), (0x7504D34C), (0xCC210D8B), (0x74B2C87B), (0xCB697DA0), (0x745F9DD3), (0xCAB26FB2), (0x740B53ED), (0xC9FBE50E), (0x73B5EBCF), (0xC945DFEB), (0x735F662F), (0xC89061D1), (0x7307C3C9), (0xC7DB6C45), (0x72AF05AB), (0xC727017A), (0x72552C79), (0xC67322B9), (0x71FA3948), (0xC5BFD232), (0x719E2CDE), (0xC50D1164), (0x71410800), (0xC45AE1D1), (0x70E2CBCD), (0xC3A945A1), (0x708378F4), (0xC2F83E1B), (0x7023109C), (0xC247CD62), (0x6FC19376), (0xC197F4BB), (0x6F5F02CF), (0xC0E8B67E), (0x6EFB5F1D), (0xC03A137E), (0x6E96A995), (0xBF8C0DD8), (0x6E30E32F), (0xBEDEA73A), (0x6DCA0D27), (0xBE31E1BF), (0x6D6227FA), (0xBD85BE33), (0x6CF934E8), (0xBCDA3EAE), (0x6C8F3538), (0xBC2F6544), (0x6C242969), (0xBB8532C0), (0x6BB812C5), (0xBADBA934), (0x6B4AF258), (0xBA32CA42), (0x6ADCC976), (0xB98A97F6), (0x6A6D98A1), (0xB8E31318), (0x69FD6132), (0xB83C3DB1), (0x698C2487), (0xB79619C6), (0x6919E326), (0xB6F0A81D), (0x68A69E72), (0xB64BEAB9), (0x68325786), (0xB5A7E331), (0x67BD0FCC), (0xB5049380), (0x6746C7D1), (0xB461FC6A), (0x66CF8103), (0xB3C01FE9), (0x66573CD5), (0xB31EFFF0), (0x65DDFBD6), (0xB27E9D43), (0x6563BF7E), (0xB1DEF9D2), (0x64E8894A), (0xB140178C), (0x646C59CC), (0xB0A1F730), (0x63EF3286), (0xB0049AA9), (0x637114AB), (0xAF68037B), (0x62F201C4), (0xAECC338B), (0x6271FA69), (0xAE312B94), (0x61F10027), (0xAD96ED77), (0x616F148E), (0xACFD7B13), (0x60EC383A), (0xAC64D51F), (0x60686CC0), (0xABCCFD75), (0x5FE3B366), (0xAB35F58C), (0x5F5E0DC8), (0xAA9FBF38), (0x5ED77C86), (0xAA0A5B2C), (0x5E500140), (0xA975CB39), (0x5DC79D9C), (0xA8E2112C), (0x5D3E523E), (0xA84F2DB4), (0x5CB420CD), (0xA7BD229A), (0x5C290AA0), (0xA72BF148), (0x5B9D1166), (0xA69B9B7D), (0x5B1035C7), (0xA60C21E8), (0x5A827978), (0xA57D8646), (0x59F3DE30), (0xA4EFCA51), (0x5964649B), (0xA462EEB2), (0x58D40E75), (0xA3D6F51F), (0x5842DD7E), (0xA34BDF4B), 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(0xD625F8C0), (0x478D832A), (0xF693D922), (0x3617D816), (0x80000648), (0x00000000), (0x00000003), (0x00000012), (0x9FC49964), (0x46141924), (0x4D834205), (0x16B799A6), (0xA39494A6), (0xCCF23C60), (0x00000001), (0x9FC47468), (0x00000000), (0x00000072), (0x00008001), (0x8FFFFB58), (0x00000003), (0x402C7413), (0x00000016), (0x402C7413), (0x9FC48498), (0x00000000), (0x00000002), (0x00000001), (0x00000072), (0x9FC4783C), (0x8000C690), (0x0358CDD5), (0x00000000), (0x00008001), (0x00000072), (0x80000648), (0x00000000), (0x00008001), (0x8000BD40), (0x1A19F8D8), (0x00008001), (0x800003C0), (0x00000003), (0x9FC428CC), (0xF6192015), (0xFBB77F92), (0x8F2C8BFC), (0x3C391923), (0x4014428C), (0x140045AE), (0xFFFD2BB6), (0x2D8050F1), (0x8000BD40), (0x00000003), (0x00000000), (0x2CC05811), (0x00008001), (0x800003C0), (0x80000648), (0x80000648), (0x9FC4206C), (0x8FFFFD78), (0x8FFFFEEC), (0x8000C668), (0x80000648), (0x8FFFFD78), (0x00000000), (0x00003040), (0x8000FFC0), (0x8000BCD8), (0x80010000), (0x9FC3F7B8), (0x00000000), (0x00002CC0), (0x80013340), (0x8000BCD8), (0x80010FD0), (0x80012548)}; fdk-aac-2.0.2/libFDK/src/mips/scale_mips.cpp000066400000000000000000000130121404225310700204510ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Description: *******************************************************************************/ #if defined(__mips_dsp) && !defined(__mips16) #ifndef FUNCTION_getScalefactor_DBL #define FUNCTION_getScalefactor_DBL /*! * * \brief Calculate max possible scale factor for input vector * * \return Maximum scale factor * * This function can constitute a significant amount of computational * complexity - very much depending on the bitrate. Since it is a rather small * function, effective assembler optimization might be possible. * */ SCALE_INLINE INT getScalefactor(const FIXP_DBL *vector, /*!< Pointer to input vector */ INT len) /*!< Length of input vector */ { INT i; FIXP_DBL maxVal = FL2FX_DBL(0.0f); for (i = len; i != 0; i--) { maxVal |= __builtin_mips_absq_s_w(*vector++); } return fixMax((INT)0, (CntLeadingZeros(maxVal) - 1)); } #endif #endif /*__mips_dsp */ fdk-aac-2.0.2/libFDK/src/nlc_dec.cpp000066400000000000000000000724321404225310700167640ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Omer Osman Description: SAC/SAOC Dec Noiseless Coding *******************************************************************************/ #include "nlc_dec.h" #include "FDK_tools_rom.h" /* MAX_PARAMETER_BANDS defines array length in huffdec */ #ifndef min #define min(a, b) (((a) < (b)) ? (a) : (b)) #endif ERROR_t sym_restoreIPD(HANDLE_FDK_BITSTREAM strm, int lav, SCHAR data[2]) { int sum_val = data[0] + data[1]; int diff_val = data[0] - data[1]; if (sum_val > lav) { data[0] = -sum_val + (2 * lav + 1); data[1] = -diff_val; } else { data[0] = sum_val; data[1] = diff_val; } if (data[0] - data[1] != 0) { ULONG sym_bit; sym_bit = FDKreadBits(strm, 1); if (sym_bit) { int tmp; tmp = data[0]; data[0] = data[1]; data[1] = tmp; } } return HUFFDEC_OK; } static int ilog2(unsigned int i) { int l = 0; if (i) i--; while (i > 0) { i >>= 1; l++; } return l; } static ERROR_t pcm_decode(HANDLE_FDK_BITSTREAM strm, SCHAR* out_data_1, SCHAR* out_data_2, int offset, int num_val, int num_levels) { int i = 0, j = 0, idx = 0; int max_grp_len = 0, next_val = 0; ULONG tmp; int pcm_chunk_size[7] = {0}; switch (num_levels) { case 3: max_grp_len = 5; break; case 7: max_grp_len = 6; break; case 11: max_grp_len = 2; break; case 13: max_grp_len = 4; break; case 19: max_grp_len = 4; break; case 25: max_grp_len = 3; break; case 51: max_grp_len = 4; break; case 4: case 8: case 15: case 16: case 26: case 31: max_grp_len = 1; break; default: return HUFFDEC_NOTOK; } tmp = 1; for (i = 1; i <= max_grp_len; i++) { tmp *= num_levels; pcm_chunk_size[i] = ilog2(tmp); } for (i = 0; i < num_val; i += max_grp_len) { int grp_len, grp_val, data; grp_len = min(max_grp_len, num_val - i); data = FDKreadBits(strm, pcm_chunk_size[grp_len]); grp_val = data; for (j = 0; j < grp_len; j++) { idx = i + (grp_len - j - 1); next_val = grp_val % num_levels; if (out_data_2 == NULL) { out_data_1[idx] = next_val - offset; } else if (out_data_1 == NULL) { out_data_2[idx] = next_val - offset; } else { if (idx % 2) { out_data_2[idx / 2] = next_val - offset; } else { out_data_1[idx / 2] = next_val - offset; } } grp_val = (grp_val - next_val) / num_levels; } } return HUFFDEC_OK; } static ERROR_t huff_read(HANDLE_FDK_BITSTREAM strm, const SHORT (*nodeTab)[MAX_ENTRIES][2], int* out_data) { int node = 0; int len = 0; do { ULONG next_bit; next_bit = FDKreadBits(strm, 1); len++; node = (*nodeTab)[node][next_bit]; } while (node > 0); *out_data = node; return HUFFDEC_OK; } static ERROR_t huff_read_2D(HANDLE_FDK_BITSTREAM strm, const SHORT (*nodeTab)[MAX_ENTRIES][2], SCHAR out_data[2], int* escape) { ERROR_t err = HUFFDEC_OK; int huff_2D_8bit = 0; int node = 0; if ((err = huff_read(strm, nodeTab, &node)) != HUFFDEC_OK) { goto bail; } *escape = (node == 0); if (*escape) { out_data[0] = 0; out_data[1] = 1; } else { huff_2D_8bit = -(node + 1); out_data[0] = huff_2D_8bit >> 4; out_data[1] = huff_2D_8bit & 0xf; } bail: return err; } static ERROR_t sym_restore(HANDLE_FDK_BITSTREAM strm, int lav, SCHAR data[2]) { ULONG sym_bit = 0; int sum_val = data[0] + data[1]; int diff_val = data[0] - data[1]; if (sum_val > lav) { data[0] = -sum_val + (2 * lav + 1); data[1] = -diff_val; } else { data[0] = sum_val; data[1] = diff_val; } if (data[0] + data[1] != 0) { sym_bit = FDKreadBits(strm, 1); if (sym_bit) { data[0] = -data[0]; data[1] = -data[1]; } } if (data[0] - data[1] != 0) { sym_bit = FDKreadBits(strm, 1); if (sym_bit) { int tmp; tmp = data[0]; data[0] = data[1]; data[1] = tmp; } } return HUFFDEC_OK; } static ERROR_t huff_dec_1D(HANDLE_FDK_BITSTREAM strm, const DATA_TYPE data_type, const INT dim1, SCHAR* out_data, const INT num_val, const INT p0_flag) { ERROR_t err = HUFFDEC_OK; int i = 0, node = 0, offset = 0; int od = 0, od_sign = 0; ULONG data = 0; int bitsAvail = 0; const SHORT(*partTab)[MAX_ENTRIES][2] = NULL; const SHORT(*nodeTab)[MAX_ENTRIES][2] = NULL; switch (data_type) { case t_CLD: partTab = (HANDLE_HUFF_NODE)&FDK_huffPart0Nodes.cld[0][0]; nodeTab = (HANDLE_HUFF_NODE)&FDK_huffCLDNodes.h1D[dim1]->nodeTab[0][0]; break; case t_ICC: partTab = (HANDLE_HUFF_NODE)&FDK_huffPart0Nodes.icc[0][0]; nodeTab = (HANDLE_HUFF_NODE)&FDK_huffICCNodes.h1D[dim1]->nodeTab[0][0]; break; case t_OLD: partTab = (HANDLE_HUFF_NODE)&FDK_huffPart0Nodes.old[0][0]; nodeTab = (HANDLE_HUFF_NODE)&huffOLDNodes.h1D[dim1]->nodeTab[0][0]; break; case t_IPD: partTab = (HANDLE_HUFF_NODE)&FDK_huffPart0Nodes.ipd[0][0]; nodeTab = (HANDLE_HUFF_NODE)&FDK_huffIPDNodes.h1D[dim1].nodeTab[0][0]; break; default: FDK_ASSERT(0); err = HUFFDEC_NOTOK; goto bail; } if (p0_flag) { if ((err = huff_read(strm, partTab, &node)) != HUFFDEC_OK) { goto bail; } out_data[0] = -(node + 1); offset = 1; } for (i = offset; i < num_val; i++) { bitsAvail = FDKgetValidBits(strm); if (bitsAvail < 1) { err = HUFFDEC_NOTOK; goto bail; } if ((err = huff_read(strm, nodeTab, &node)) != HUFFDEC_OK) { goto bail; } od = -(node + 1); if (data_type != t_IPD) { if (od != 0) { bitsAvail = FDKgetValidBits(strm); if (bitsAvail < 1) { err = HUFFDEC_NOTOK; goto bail; } data = FDKreadBits(strm, 1); od_sign = data; if (od_sign) od = -od; } } out_data[i] = od; } bail: return err; } static ERROR_t huff_dec_2D(HANDLE_FDK_BITSTREAM strm, const DATA_TYPE data_type, const INT dim1, const INT dim2, SCHAR out_data[][2], const INT num_val, const INT stride, SCHAR* p0_data[2]) { ERROR_t err = HUFFDEC_OK; int i = 0, lav = 0, escape = 0, escCntr = 0; int node = 0; unsigned long data = 0; SCHAR esc_data[2][28] = {{0}}; int escIdx[28] = {0}; const SHORT(*nodeTab)[MAX_ENTRIES][2] = NULL; /* LAV */ if ((err = huff_read(strm, (HANDLE_HUFF_NODE)&FDK_huffLavIdxNodes.nodeTab[0][0], &node)) != HUFFDEC_OK) { goto bail; } data = -(node + 1); switch (data_type) { case t_CLD: lav = 2 * data + 3; /* 3, 5, 7, 9 */ nodeTab = (HANDLE_HUFF_NODE)&FDK_huffPart0Nodes.cld[0][0]; break; case t_ICC: lav = 2 * data + 1; /* 1, 3, 5, 7 */ nodeTab = (HANDLE_HUFF_NODE)&FDK_huffPart0Nodes.icc[0][0]; break; case t_OLD: lav = 3 * data + 3; nodeTab = (HANDLE_HUFF_NODE)&FDK_huffPart0Nodes.old[0][0]; break; case t_IPD: if (data == 0) data = 3; else data--; lav = 2 * data + 1; /* 1, 3, 5, 7 */ nodeTab = (HANDLE_HUFF_NODE)&FDK_huffPart0Nodes.ipd[0][0]; break; default: FDK_ASSERT(0); err = HUFFDEC_NOTOK; goto bail; } /* Partition 0 */ if (p0_data[0] != NULL) { if ((err = huff_read(strm, nodeTab, &node)) != HUFFDEC_OK) { goto bail; } *p0_data[0] = -(node + 1); } if (p0_data[1] != NULL) { if ((err = huff_read(strm, nodeTab, &node)) != HUFFDEC_OK) { goto bail; } *p0_data[1] = -(node + 1); } switch (data_type) { case t_CLD: switch (lav) { case 3: nodeTab = (HANDLE_HUFF_NODE)&FDK_huffCLDNodes.h2D[dim1][dim2]->lav3[0][0]; break; case 5: nodeTab = (HANDLE_HUFF_NODE)&FDK_huffCLDNodes.h2D[dim1][dim2]->lav5[0][0]; break; case 7: nodeTab = (HANDLE_HUFF_NODE)&FDK_huffCLDNodes.h2D[dim1][dim2]->lav7[0][0]; break; case 9: nodeTab = (HANDLE_HUFF_NODE)&FDK_huffCLDNodes.h2D[dim1][dim2]->lav9[0][0]; break; } break; case t_ICC: switch (lav) { case 1: nodeTab = (HANDLE_HUFF_NODE)&FDK_huffICCNodes.h2D[dim1][dim2]->lav1[0][0]; break; case 3: nodeTab = (HANDLE_HUFF_NODE)&FDK_huffICCNodes.h2D[dim1][dim2]->lav3[0][0]; break; case 5: nodeTab = (HANDLE_HUFF_NODE)&FDK_huffICCNodes.h2D[dim1][dim2]->lav5[0][0]; break; case 7: nodeTab = (HANDLE_HUFF_NODE)&FDK_huffICCNodes.h2D[dim1][dim2]->lav7[0][0]; break; } break; case t_OLD: switch (lav) { case 3: nodeTab = (HANDLE_HUFF_NODE)&huffOLDNodes.h2D[dim1][dim2]->lav3[0][0]; break; case 6: nodeTab = (HANDLE_HUFF_NODE)&huffOLDNodes.h2D[dim1][dim2]->lav6[0][0]; break; case 9: nodeTab = (HANDLE_HUFF_NODE)&huffOLDNodes.h2D[dim1][dim2]->lav9[0][0]; break; case 12: nodeTab = (HANDLE_HUFF_NODE)&huffOLDNodes.h2D[dim1][dim2]->lav12[0][0]; break; } break; case t_IPD: switch (lav) { case 1: nodeTab = (HANDLE_HUFF_NODE)&FDK_huffIPDNodes.h2D[dim1][dim2].lav1[0][0]; break; case 3: nodeTab = (HANDLE_HUFF_NODE)&FDK_huffIPDNodes.h2D[dim1][dim2].lav3[0][0]; break; case 5: nodeTab = (HANDLE_HUFF_NODE)&FDK_huffIPDNodes.h2D[dim1][dim2].lav5[0][0]; break; case 7: nodeTab = (HANDLE_HUFF_NODE)&FDK_huffIPDNodes.h2D[dim1][dim2].lav7[0][0]; break; } break; default: break; } for (i = 0; i < num_val; i += stride) { if ((err = huff_read_2D(strm, nodeTab, out_data[i], &escape)) != HUFFDEC_OK) { goto bail; } if (escape) { escIdx[escCntr++] = i; } else { if (data_type == t_IPD) { if ((err = sym_restoreIPD(strm, lav, out_data[i])) != HUFFDEC_OK) { goto bail; } } else { if ((err = sym_restore(strm, lav, out_data[i])) != HUFFDEC_OK) { goto bail; } } } } /* i */ if (escCntr > 0) { if ((err = pcm_decode(strm, esc_data[0], esc_data[1], 0, 2 * escCntr, (2 * lav + 1))) != HUFFDEC_OK) { goto bail; } for (i = 0; i < escCntr; i++) { out_data[escIdx[i]][0] = esc_data[0][i] - lav; out_data[escIdx[i]][1] = esc_data[1][i] - lav; } } bail: return err; } static ERROR_t huff_decode(HANDLE_FDK_BITSTREAM strm, SCHAR* out_data_1, SCHAR* out_data_2, DATA_TYPE data_type, DIFF_TYPE diff_type_1, DIFF_TYPE diff_type_2, int num_val, int* cdg_scheme, int ldMode) { ERROR_t err = HUFFDEC_OK; DIFF_TYPE diff_type; int i = 0; ULONG data = 0; SCHAR pair_vec[28][2]; SCHAR* p0_data_1[2] = {NULL, NULL}; SCHAR* p0_data_2[2] = {NULL, NULL}; int p0_flag[2]; int num_val_1_int = num_val; int num_val_2_int = num_val; SCHAR* out_data_1_int = out_data_1; SCHAR* out_data_2_int = out_data_2; int df_rest_flag_1 = 0; int df_rest_flag_2 = 0; int hufYY1; int hufYY2; int hufYY; /* Coding scheme */ data = FDKreadBits(strm, 1); *cdg_scheme = (data << PAIR_SHIFT); if (*cdg_scheme >> PAIR_SHIFT == HUFF_2D) { if ((out_data_1 != NULL) && (out_data_2 != NULL) && (ldMode == 0)) { data = FDKreadBits(strm, 1); *cdg_scheme = (*cdg_scheme | data); } else { *cdg_scheme = (*cdg_scheme | FREQ_PAIR); } } { hufYY1 = diff_type_1; hufYY2 = diff_type_2; } switch (*cdg_scheme >> PAIR_SHIFT) { case HUFF_1D: p0_flag[0] = (diff_type_1 == DIFF_FREQ); p0_flag[1] = (diff_type_2 == DIFF_FREQ); if (out_data_1 != NULL) { if ((err = huff_dec_1D(strm, data_type, hufYY1, out_data_1, num_val_1_int, p0_flag[0])) != HUFFDEC_OK) { goto bail; } } if (out_data_2 != NULL) { if ((err = huff_dec_1D(strm, data_type, hufYY2, out_data_2, num_val_2_int, p0_flag[1])) != HUFFDEC_OK) { goto bail; } } break; /* HUFF_1D */ case HUFF_2D: switch (*cdg_scheme & PAIR_MASK) { case FREQ_PAIR: if (out_data_1 != NULL) { if (diff_type_1 == DIFF_FREQ) { p0_data_1[0] = &out_data_1[0]; p0_data_1[1] = NULL; num_val_1_int -= 1; out_data_1_int += 1; } df_rest_flag_1 = num_val_1_int % 2; if (df_rest_flag_1) num_val_1_int -= 1; if (num_val_1_int < 0) { err = HUFFDEC_NOTOK; goto bail; } } if (out_data_2 != NULL) { if (diff_type_2 == DIFF_FREQ) { p0_data_2[0] = NULL; p0_data_2[1] = &out_data_2[0]; num_val_2_int -= 1; out_data_2_int += 1; } df_rest_flag_2 = num_val_2_int % 2; if (df_rest_flag_2) num_val_2_int -= 1; if (num_val_2_int < 0) { err = HUFFDEC_NOTOK; goto bail; } } if (out_data_1 != NULL) { if ((err = huff_dec_2D(strm, data_type, hufYY1, FREQ_PAIR, pair_vec, num_val_1_int, 2, p0_data_1)) != HUFFDEC_OK) { goto bail; } if (df_rest_flag_1) { if ((err = huff_dec_1D(strm, data_type, hufYY1, out_data_1_int + num_val_1_int, 1, 0)) != HUFFDEC_OK) { goto bail; } } } if (out_data_2 != NULL) { if ((err = huff_dec_2D(strm, data_type, hufYY2, FREQ_PAIR, pair_vec + 1, num_val_2_int, 2, p0_data_2)) != HUFFDEC_OK) { goto bail; } if (df_rest_flag_2) { if ((err = huff_dec_1D(strm, data_type, hufYY2, out_data_2_int + num_val_2_int, 1, 0)) != HUFFDEC_OK) { goto bail; } } } if (out_data_1 != NULL) { for (i = 0; i < num_val_1_int - 1; i += 2) { out_data_1_int[i] = pair_vec[i][0]; out_data_1_int[i + 1] = pair_vec[i][1]; } } if (out_data_2 != NULL) { for (i = 0; i < num_val_2_int - 1; i += 2) { out_data_2_int[i] = pair_vec[i + 1][0]; out_data_2_int[i + 1] = pair_vec[i + 1][1]; } } break; /* FREQ_PAIR */ case TIME_PAIR: if (((diff_type_1 == DIFF_FREQ) || (diff_type_2 == DIFF_FREQ))) { p0_data_1[0] = &out_data_1[0]; p0_data_1[1] = &out_data_2[0]; out_data_1_int += 1; out_data_2_int += 1; num_val_1_int -= 1; } if ((diff_type_1 == DIFF_TIME) || (diff_type_2 == DIFF_TIME)) { diff_type = DIFF_TIME; } else { diff_type = DIFF_FREQ; } { hufYY = diff_type; } if ((err = huff_dec_2D(strm, data_type, hufYY, TIME_PAIR, pair_vec, num_val_1_int, 1, p0_data_1)) != HUFFDEC_OK) { goto bail; } for (i = 0; i < num_val_1_int; i++) { out_data_1_int[i] = pair_vec[i][0]; out_data_2_int[i] = pair_vec[i][1]; } break; /* TIME_PAIR */ default: break; } break; /* HUFF_2D */ default: break; } bail: return err; } static void diff_freq_decode(const SCHAR* const diff_data, SCHAR* const out_data, const int num_val) { int i = 0; out_data[0] = diff_data[0]; for (i = 1; i < num_val; i++) { out_data[i] = out_data[i - 1] + diff_data[i]; } } static void diff_time_decode_backwards(const SCHAR* const prev_data, const SCHAR* const diff_data, SCHAR* const out_data, const int mixed_diff_type, const int num_val) { int i = 0; /* default start value*/ if (mixed_diff_type) { out_data[0] = diff_data[0]; i = 1; /* new start value */ } for (; i < num_val; i++) { out_data[i] = prev_data[i] + diff_data[i]; } } static void diff_time_decode_forwards(const SCHAR* const prev_data, const SCHAR* const diff_data, SCHAR* const out_data, const int mixed_diff_type, const int num_val) { int i = 0; /* default start value*/ if (mixed_diff_type) { out_data[0] = diff_data[0]; i = 1; /* new start value */ } for (; i < num_val; i++) { out_data[i] = prev_data[i] - diff_data[i]; } } static ERROR_t attach_lsb(HANDLE_FDK_BITSTREAM strm, SCHAR* in_data_msb, int offset, int num_lsb, int num_val, SCHAR* out_data) { int i = 0, lsb = 0; ULONG data = 0; for (i = 0; i < num_val; i++) { int msb; msb = in_data_msb[i]; if (num_lsb > 0) { data = FDKreadBits(strm, num_lsb); lsb = data; out_data[i] = ((msb << num_lsb) | lsb) - offset; } else out_data[i] = msb - offset; } return HUFFDEC_OK; /* dummy */ } ERROR_t EcDataPairDec(DECODER_TYPE DECODER, HANDLE_FDK_BITSTREAM strm, SCHAR* aaOutData1, SCHAR* aaOutData2, SCHAR* aHistory, DATA_TYPE data_type, int startBand, int dataBands, int pair_flag, int coarse_flag, int allowDiffTimeBack_flag) { ERROR_t err = HUFFDEC_OK; // int allowDiffTimeBack_flag = !independency_flag || (setIdx > 0); int attachLsb_flag = 0; int pcmCoding_flag = 0; int mixed_time_pair = 0, numValPcm = 0; int quant_levels = 0, quant_offset = 0; ULONG data = 0; SCHAR aaDataPair[2][28] = {{0}}; SCHAR aaDataDiff[2][28] = {{0}}; SCHAR aHistoryMsb[28] = {0}; SCHAR* pDataVec[2] = {NULL, NULL}; DIFF_TYPE diff_type[2] = {DIFF_FREQ, DIFF_FREQ}; int cdg_scheme = HUFF_1D; DIRECTION direction = BACKWARDS; switch (data_type) { case t_CLD: if (coarse_flag) { attachLsb_flag = 0; quant_levels = 15; quant_offset = 7; } else { attachLsb_flag = 0; quant_levels = 31; quant_offset = 15; } break; case t_ICC: if (coarse_flag) { attachLsb_flag = 0; quant_levels = 4; quant_offset = 0; } else { attachLsb_flag = 0; quant_levels = 8; quant_offset = 0; } break; case t_OLD: if (coarse_flag) { attachLsb_flag = 0; quant_levels = 8; quant_offset = 0; } else { attachLsb_flag = 0; quant_levels = 16; quant_offset = 0; } break; case t_NRG: if (coarse_flag) { attachLsb_flag = 0; quant_levels = 32; quant_offset = 0; } else { attachLsb_flag = 0; quant_levels = 64; quant_offset = 0; } break; case t_IPD: if (!coarse_flag) { attachLsb_flag = 1; quant_levels = 16; quant_offset = 0; } else { attachLsb_flag = 0; quant_levels = 8; quant_offset = 0; } break; default: return HUFFDEC_NOTOK; } data = FDKreadBits(strm, 1); pcmCoding_flag = data; if (pcmCoding_flag) { if (pair_flag) { pDataVec[0] = aaDataPair[0]; pDataVec[1] = aaDataPair[1]; numValPcm = 2 * dataBands; } else { pDataVec[0] = aaDataPair[0]; pDataVec[1] = NULL; numValPcm = dataBands; } err = pcm_decode(strm, pDataVec[0], pDataVec[1], quant_offset, numValPcm, quant_levels); if (err != HUFFDEC_OK) return HUFFDEC_NOTOK; } else { /* Differential/Huffman/LSB Coding */ if (pair_flag) { pDataVec[0] = aaDataDiff[0]; pDataVec[1] = aaDataDiff[1]; } else { pDataVec[0] = aaDataDiff[0]; pDataVec[1] = NULL; } diff_type[0] = DIFF_FREQ; diff_type[1] = DIFF_FREQ; direction = BACKWARDS; { if (pair_flag || allowDiffTimeBack_flag) { data = FDKreadBits(strm, 1); diff_type[0] = (DIFF_TYPE)data; } if (pair_flag && ((diff_type[0] == DIFF_FREQ) || allowDiffTimeBack_flag)) { data = FDKreadBits(strm, 1); diff_type[1] = (DIFF_TYPE)data; } } /* Huffman decoding */ err = huff_decode(strm, pDataVec[0], pDataVec[1], data_type, diff_type[0], diff_type[1], dataBands, &cdg_scheme, (DECODER == SAOC_DECODER)); if (err != HUFFDEC_OK) { return HUFFDEC_NOTOK; } { /* Differential decoding */ if ((diff_type[0] == DIFF_TIME) || (diff_type[1] == DIFF_TIME)) { if (DECODER == SAOC_DECODER) { direction = BACKWARDS; } else { if (pair_flag) { if ((diff_type[0] == DIFF_TIME) && !allowDiffTimeBack_flag) { direction = FORWARDS; } else if (diff_type[1] == DIFF_TIME) { direction = BACKWARDS; } else { data = FDKreadBits(strm, 1); direction = (DIRECTION)data; } } else { direction = BACKWARDS; } } } mixed_time_pair = (diff_type[0] != diff_type[1]) && ((cdg_scheme & PAIR_MASK) == TIME_PAIR); if (direction == BACKWARDS) { if (diff_type[0] == DIFF_FREQ) { diff_freq_decode(aaDataDiff[0], aaDataPair[0], dataBands); } else { int i; for (i = 0; i < dataBands; i++) { aHistoryMsb[i] = aHistory[i + startBand] + quant_offset; if (attachLsb_flag) { aHistoryMsb[i] >>= 1; } } diff_time_decode_backwards(aHistoryMsb, aaDataDiff[0], aaDataPair[0], mixed_time_pair, dataBands); } if (diff_type[1] == DIFF_FREQ) { diff_freq_decode(aaDataDiff[1], aaDataPair[1], dataBands); } else { diff_time_decode_backwards(aaDataPair[0], aaDataDiff[1], aaDataPair[1], mixed_time_pair, dataBands); } } else { /* diff_type[1] MUST BE DIFF_FREQ */ diff_freq_decode(aaDataDiff[1], aaDataPair[1], dataBands); if (diff_type[0] == DIFF_FREQ) { diff_freq_decode(aaDataDiff[0], aaDataPair[0], dataBands); } else { diff_time_decode_forwards(aaDataPair[1], aaDataDiff[0], aaDataPair[0], mixed_time_pair, dataBands); } } } /* LSB decoding */ err = attach_lsb(strm, aaDataPair[0], quant_offset, attachLsb_flag ? 1 : 0, dataBands, aaDataPair[0]); if (err != HUFFDEC_OK) goto bail; if (pair_flag) { err = attach_lsb(strm, aaDataPair[1], quant_offset, attachLsb_flag ? 1 : 0, dataBands, aaDataPair[1]); if (err != HUFFDEC_OK) goto bail; } } /* End: Differential/Huffman/LSB Coding */ /* Copy data to output arrays */ FDKmemcpy(aaOutData1 + startBand, aaDataPair[0], sizeof(SCHAR) * dataBands); if (pair_flag) { FDKmemcpy(aaOutData2 + startBand, aaDataPair[1], sizeof(SCHAR) * dataBands); } bail: return err; } ERROR_t huff_dec_reshape(HANDLE_FDK_BITSTREAM strm, int* out_data, int num_val) { ERROR_t err = HUFFDEC_OK; int val_rcvd = 0, dummy = 0, i = 0, val = 0, len = 0; SCHAR rl_data[2] = {0}; while (val_rcvd < num_val) { err = huff_read_2D(strm, (HANDLE_HUFF_NODE)&FDK_huffReshapeNodes.nodeTab[0][0], rl_data, &dummy); if (err != HUFFDEC_OK) goto bail; val = rl_data[0]; len = rl_data[1] + 1; if (val_rcvd + len > num_val) { err = HUFFDEC_NOTOK; goto bail; } for (i = val_rcvd; i < val_rcvd + len; i++) { out_data[i] = val; } val_rcvd += len; } bail: return err; } fdk-aac-2.0.2/libFDK/src/qmf.cpp000066400000000000000000000665151404225310700161650ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Markus Lohwasser, Josef Hoepfl, Manuel Jander Description: QMF filterbank *******************************************************************************/ /*! \file \brief Complex qmf analysis/synthesis This module contains the qmf filterbank for analysis [ cplxAnalysisQmfFiltering() ] and synthesis [ cplxSynthesisQmfFiltering() ]. It is a polyphase implementation of a complex exponential modulated filter bank. The analysis part usually runs at half the sample rate than the synthesis part. (So called "dual-rate" mode.) The coefficients of the prototype filter are specified in #qmf_pfilt640 (in sbr_rom.cpp). Thus only a 64 channel version (32 on the analysis side) with a 640 tap prototype filter are used. \anchor PolyphaseFiltering

About polyphase filtering

The polyphase implementation of a filterbank requires filtering at the input and output. This is implemented as part of cplxAnalysisQmfFiltering() and cplxSynthesisQmfFiltering(). The implementation requires the filter coefficients in a specific structure as described in #sbr_qmf_64_640_qmf (in sbr_rom.cpp). This module comprises the computationally most expensive functions of the SBR decoder. The accuracy of computations is also important and has a direct impact on the overall sound quality. Therefore a special test program is available which can be used to only test the filterbank: main_audio.cpp This modules also uses scaling of data to provide better SNR on fixed-point processors. See #QMF_SCALE_FACTOR (in sbr_scale.h) for details. An interesting note: The function getScalefactor() can constitute a significant amount of computational complexity - very much depending on the bitrate. Since it is a rather small function, effective assembler optimization might be possible. */ #include "qmf.h" #include "FDK_trigFcts.h" #include "fixpoint_math.h" #include "dct.h" #define QSSCALE (0) #define FX_DBL2FX_QSS(x) (x) #define FX_QSS2FX_DBL(x) (x) /* moved to qmf_pcm.h: -> qmfSynPrototypeFirSlot */ /* moved to qmf_pcm.h: -> qmfSynPrototypeFirSlot_NonSymmetric */ /* moved to qmf_pcm.h: -> qmfSynthesisFilteringSlot */ /*! * * \brief Perform real-valued forward modulation of the time domain * data of timeIn and stores the real part of the subband * samples in rSubband * */ static void qmfForwardModulationLP_even( HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank */ FIXP_DBL *timeIn, /*!< Time Signal */ FIXP_DBL *rSubband) /*!< Real Output */ { int i; int L = anaQmf->no_channels; int M = L >> 1; int scale = 0; FIXP_DBL accu; const FIXP_DBL *timeInTmp1 = (FIXP_DBL *)&timeIn[3 * M]; const FIXP_DBL *timeInTmp2 = timeInTmp1; FIXP_DBL *rSubbandTmp = rSubband; rSubband[0] = timeIn[3 * M] >> 1; for (i = M - 1; i != 0; i--) { accu = ((*--timeInTmp1) >> 1) + ((*++timeInTmp2) >> 1); *++rSubbandTmp = accu; } timeInTmp1 = &timeIn[2 * M]; timeInTmp2 = &timeIn[0]; rSubbandTmp = &rSubband[M]; for (i = L - M; i != 0; i--) { accu = ((*timeInTmp1--) >> 1) - ((*timeInTmp2++) >> 1); *rSubbandTmp++ = accu; } dct_III(rSubband, timeIn, L, &scale); } #if !defined(FUNCTION_qmfForwardModulationLP_odd) static void qmfForwardModulationLP_odd( HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank */ const FIXP_DBL *timeIn, /*!< Time Signal */ FIXP_DBL *rSubband) /*!< Real Output */ { int i; int L = anaQmf->no_channels; int M = L >> 1; int shift = (anaQmf->no_channels >> 6) + 1; for (i = 0; i < M; i++) { rSubband[M + i] = (timeIn[L - 1 - i] >> 1) - (timeIn[i] >> shift); rSubband[M - 1 - i] = (timeIn[L + i] >> 1) + (timeIn[2 * L - 1 - i] >> shift); } dct_IV(rSubband, L, &shift); } #endif /* !defined(FUNCTION_qmfForwardModulationLP_odd) */ /*! * * \brief Perform complex-valued forward modulation of the time domain * data of timeIn and stores the real part of the subband * samples in rSubband, and the imaginary part in iSubband * * */ #if !defined(FUNCTION_qmfForwardModulationHQ) static void qmfForwardModulationHQ( HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank */ const FIXP_DBL *RESTRICT timeIn, /*!< Time Signal */ FIXP_DBL *RESTRICT rSubband, /*!< Real Output */ FIXP_DBL *RESTRICT iSubband /*!< Imaginary Output */ ) { int i; int L = anaQmf->no_channels; int L2 = L << 1; int shift = 0; /* Time advance by one sample, which is equivalent to the complex rotation at the end of the analysis. Works only for STD mode. */ if ((L == 64) && !(anaQmf->flags & (QMF_FLAG_CLDFB | QMF_FLAG_MPSLDFB))) { FIXP_DBL x, y; /*rSubband[0] = u[1] + u[0]*/ /*iSubband[0] = u[1] - u[0]*/ x = timeIn[1] >> 1; y = timeIn[0]; rSubband[0] = x + (y >> 1); iSubband[0] = x - (y >> 1); /*rSubband[n] = u[n+1] - u[2M-n], n=1,...,M-1*/ /*iSubband[n] = u[n+1] + u[2M-n], n=1,...,M-1*/ for (i = 1; i < L; i++) { x = timeIn[i + 1] >> 1; /*u[n+1] */ y = timeIn[L2 - i]; /*u[2M-n] */ rSubband[i] = x - (y >> 1); iSubband[i] = x + (y >> 1); } } else { for (i = 0; i < L; i += 2) { FIXP_DBL x0, x1, y0, y1; x0 = timeIn[i + 0] >> 1; x1 = timeIn[i + 1] >> 1; y0 = timeIn[L2 - 1 - i]; y1 = timeIn[L2 - 2 - i]; rSubband[i + 0] = x0 - (y0 >> 1); rSubband[i + 1] = x1 - (y1 >> 1); iSubband[i + 0] = x0 + (y0 >> 1); iSubband[i + 1] = x1 + (y1 >> 1); } } dct_IV(rSubband, L, &shift); dst_IV(iSubband, L, &shift); /* Do the complex rotation except for the case of 64 bands (in STD mode). */ if ((L != 64) || (anaQmf->flags & (QMF_FLAG_CLDFB | QMF_FLAG_MPSLDFB))) { if (anaQmf->flags & QMF_FLAG_MPSLDFB_OPTIMIZE_MODULATION) { FIXP_DBL iBand; for (i = 0; i < fMin(anaQmf->lsb, L); i += 2) { iBand = rSubband[i]; rSubband[i] = -iSubband[i]; iSubband[i] = iBand; iBand = -rSubband[i + 1]; rSubband[i + 1] = iSubband[i + 1]; iSubband[i + 1] = iBand; } } else { const FIXP_QTW *sbr_t_cos; const FIXP_QTW *sbr_t_sin; const int len = L; /* was len = fMin(anaQmf->lsb, L) but in case of USAC the signal above lsb is actually needed in some cases (HBE?) */ sbr_t_cos = anaQmf->t_cos; sbr_t_sin = anaQmf->t_sin; for (i = 0; i < len; i++) { cplxMult(&iSubband[i], &rSubband[i], iSubband[i], rSubband[i], sbr_t_cos[i], sbr_t_sin[i]); } } } } #endif /* FUNCTION_qmfForwardModulationHQ */ /*! * * \brief Perform low power inverse modulation of the subband * samples stored in rSubband (real part) and iSubband (imaginary * part) and stores the result in pWorkBuffer. * */ inline static void qmfInverseModulationLP_even( HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */ const FIXP_DBL *qmfReal, /*!< Pointer to qmf real subband slot (input) */ const int scaleFactorLowBand, /*!< Scalefactor for Low band */ const int scaleFactorHighBand, /*!< Scalefactor for High band */ FIXP_DBL *pTimeOut /*!< Pointer to qmf subband slot (output)*/ ) { int i; int L = synQmf->no_channels; int M = L >> 1; int scale = 0; FIXP_DBL tmp; FIXP_DBL *RESTRICT tReal = pTimeOut; FIXP_DBL *RESTRICT tImag = pTimeOut + L; /* Move input to output vector with offset */ scaleValuesSaturate(&tReal[0], &qmfReal[0], synQmf->lsb, scaleFactorLowBand); scaleValuesSaturate(&tReal[0 + synQmf->lsb], &qmfReal[0 + synQmf->lsb], synQmf->usb - synQmf->lsb, scaleFactorHighBand); FDKmemclear(&tReal[0 + synQmf->usb], (L - synQmf->usb) * sizeof(FIXP_DBL)); /* Dct type-2 transform */ dct_II(tReal, tImag, L, &scale); /* Expand output and replace inplace the output buffers */ tImag[0] = tReal[M]; tImag[M] = (FIXP_DBL)0; tmp = tReal[0]; tReal[0] = tReal[M]; tReal[M] = tmp; for (i = 1; i < M / 2; i++) { /* Imag */ tmp = tReal[L - i]; tImag[M - i] = tmp; tImag[i + M] = -tmp; tmp = tReal[M + i]; tImag[i] = tmp; tImag[L - i] = -tmp; /* Real */ tReal[M + i] = tReal[i]; tReal[L - i] = tReal[M - i]; tmp = tReal[i]; tReal[i] = tReal[M - i]; tReal[M - i] = tmp; } /* Remaining odd terms */ tmp = tReal[M + M / 2]; tImag[M / 2] = tmp; tImag[M / 2 + M] = -tmp; tReal[M + M / 2] = tReal[M / 2]; } inline static void qmfInverseModulationLP_odd( HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */ const FIXP_DBL *qmfReal, /*!< Pointer to qmf real subband slot (input) */ const int scaleFactorLowBand, /*!< Scalefactor for Low band */ const int scaleFactorHighBand, /*!< Scalefactor for High band */ FIXP_DBL *pTimeOut /*!< Pointer to qmf subband slot (output)*/ ) { int i; int L = synQmf->no_channels; int M = L >> 1; int shift = 0; /* Move input to output vector with offset */ scaleValuesSaturate(pTimeOut + M, qmfReal, synQmf->lsb, scaleFactorLowBand); scaleValuesSaturate(pTimeOut + M + synQmf->lsb, qmfReal + synQmf->lsb, synQmf->usb - synQmf->lsb, scaleFactorHighBand); FDKmemclear(pTimeOut + M + synQmf->usb, (L - synQmf->usb) * sizeof(FIXP_DBL)); dct_IV(pTimeOut + M, L, &shift); for (i = 0; i < M; i++) { pTimeOut[i] = pTimeOut[L - 1 - i]; pTimeOut[2 * L - 1 - i] = -pTimeOut[L + i]; } } #ifndef FUNCTION_qmfInverseModulationHQ /*! * * \brief Perform complex-valued inverse modulation of the subband * samples stored in rSubband (real part) and iSubband (imaginary * part) and stores the result in pWorkBuffer. * */ inline static void qmfInverseModulationHQ( HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */ const FIXP_DBL *qmfReal, /*!< Pointer to qmf real subband slot */ const FIXP_DBL *qmfImag, /*!< Pointer to qmf imag subband slot */ const int scaleFactorLowBand, /*!< Scalefactor for Low band */ const int scaleFactorHighBand, /*!< Scalefactor for High band */ FIXP_DBL *pWorkBuffer /*!< WorkBuffer (output) */ ) { int i; int L = synQmf->no_channels; int M = L >> 1; int shift = 0; FIXP_DBL *RESTRICT tReal = pWorkBuffer; FIXP_DBL *RESTRICT tImag = pWorkBuffer + L; if (synQmf->flags & QMF_FLAG_CLDFB) { for (i = 0; i < synQmf->usb; i++) { cplxMultDiv2(&tImag[i], &tReal[i], qmfImag[i], qmfReal[i], synQmf->t_cos[i], synQmf->t_sin[i]); } scaleValuesSaturate(&tReal[0], synQmf->lsb, scaleFactorLowBand + 1); scaleValuesSaturate(&tReal[0 + synQmf->lsb], synQmf->usb - synQmf->lsb, scaleFactorHighBand + 1); scaleValuesSaturate(&tImag[0], synQmf->lsb, scaleFactorLowBand + 1); scaleValuesSaturate(&tImag[0 + synQmf->lsb], synQmf->usb - synQmf->lsb, scaleFactorHighBand + 1); } if ((synQmf->flags & QMF_FLAG_CLDFB) == 0) { scaleValuesSaturate(&tReal[0], &qmfReal[0], synQmf->lsb, scaleFactorLowBand); scaleValuesSaturate(&tReal[0 + synQmf->lsb], &qmfReal[0 + synQmf->lsb], synQmf->usb - synQmf->lsb, scaleFactorHighBand); scaleValuesSaturate(&tImag[0], &qmfImag[0], synQmf->lsb, scaleFactorLowBand); scaleValuesSaturate(&tImag[0 + synQmf->lsb], &qmfImag[0 + synQmf->lsb], synQmf->usb - synQmf->lsb, scaleFactorHighBand); } FDKmemclear(&tReal[synQmf->usb], (synQmf->no_channels - synQmf->usb) * sizeof(FIXP_DBL)); FDKmemclear(&tImag[synQmf->usb], (synQmf->no_channels - synQmf->usb) * sizeof(FIXP_DBL)); dct_IV(tReal, L, &shift); dst_IV(tImag, L, &shift); if (synQmf->flags & QMF_FLAG_CLDFB) { for (i = 0; i < M; i++) { FIXP_DBL r1, i1, r2, i2; r1 = tReal[i]; i2 = tImag[L - 1 - i]; r2 = tReal[L - i - 1]; i1 = tImag[i]; tReal[i] = (r1 - i1) >> 1; tImag[L - 1 - i] = -(r1 + i1) >> 1; tReal[L - i - 1] = (r2 - i2) >> 1; tImag[i] = -(r2 + i2) >> 1; } } else { /* The array accesses are negative to compensate the missing minus sign in * the low and hi band gain. */ /* 26 cycles on ARM926 */ for (i = 0; i < M; i++) { FIXP_DBL r1, i1, r2, i2; r1 = -tReal[i]; i2 = -tImag[L - 1 - i]; r2 = -tReal[L - i - 1]; i1 = -tImag[i]; tReal[i] = (r1 - i1) >> 1; tImag[L - 1 - i] = -(r1 + i1) >> 1; tReal[L - i - 1] = (r2 - i2) >> 1; tImag[i] = -(r2 + i2) >> 1; } } } #endif /* #ifndef FUNCTION_qmfInverseModulationHQ */ /*! * * \brief Create QMF filter bank instance * * \return 0 if successful * */ static int qmfInitFilterBank( HANDLE_QMF_FILTER_BANK h_Qmf, /*!< Handle to return */ void *pFilterStates, /*!< Handle to filter states */ int noCols, /*!< Number of timeslots per frame */ int lsb, /*!< Lower end of QMF frequency range */ int usb, /*!< Upper end of QMF frequency range */ int no_channels, /*!< Number of channels (bands) */ UINT flags, /*!< flags */ int synflag) /*!< 1: synthesis; 0: analysis */ { FDKmemclear(h_Qmf, sizeof(QMF_FILTER_BANK)); if (flags & QMF_FLAG_MPSLDFB) { flags |= QMF_FLAG_NONSYMMETRIC; flags |= QMF_FLAG_MPSLDFB_OPTIMIZE_MODULATION; h_Qmf->t_cos = NULL; h_Qmf->t_sin = NULL; h_Qmf->filterScale = QMF_MPSLDFB_PFT_SCALE; h_Qmf->p_stride = 1; switch (no_channels) { case 64: h_Qmf->p_filter = qmf_mpsldfb_640; h_Qmf->FilterSize = 640; break; case 32: h_Qmf->p_filter = qmf_mpsldfb_320; h_Qmf->FilterSize = 320; break; default: return -1; } } if (!(flags & QMF_FLAG_MPSLDFB) && (flags & QMF_FLAG_CLDFB)) { flags |= QMF_FLAG_NONSYMMETRIC; h_Qmf->filterScale = QMF_CLDFB_PFT_SCALE; h_Qmf->p_stride = 1; switch (no_channels) { case 64: h_Qmf->t_cos = qmf_phaseshift_cos64_cldfb; h_Qmf->t_sin = qmf_phaseshift_sin64_cldfb; h_Qmf->p_filter = qmf_cldfb_640; h_Qmf->FilterSize = 640; break; case 32: h_Qmf->t_cos = (synflag) ? qmf_phaseshift_cos32_cldfb_syn : qmf_phaseshift_cos32_cldfb_ana; h_Qmf->t_sin = qmf_phaseshift_sin32_cldfb; h_Qmf->p_filter = qmf_cldfb_320; h_Qmf->FilterSize = 320; break; case 16: h_Qmf->t_cos = (synflag) ? qmf_phaseshift_cos16_cldfb_syn : qmf_phaseshift_cos16_cldfb_ana; h_Qmf->t_sin = qmf_phaseshift_sin16_cldfb; h_Qmf->p_filter = qmf_cldfb_160; h_Qmf->FilterSize = 160; break; case 8: h_Qmf->t_cos = (synflag) ? qmf_phaseshift_cos8_cldfb_syn : qmf_phaseshift_cos8_cldfb_ana; h_Qmf->t_sin = qmf_phaseshift_sin8_cldfb; h_Qmf->p_filter = qmf_cldfb_80; h_Qmf->FilterSize = 80; break; default: return -1; } } if (!(flags & QMF_FLAG_MPSLDFB) && ((flags & QMF_FLAG_CLDFB) == 0)) { switch (no_channels) { case 64: h_Qmf->p_filter = qmf_pfilt640; h_Qmf->t_cos = qmf_phaseshift_cos64; h_Qmf->t_sin = qmf_phaseshift_sin64; h_Qmf->p_stride = 1; h_Qmf->FilterSize = 640; h_Qmf->filterScale = 0; break; case 40: if (synflag) { break; } else { h_Qmf->p_filter = qmf_pfilt400; /* Scaling factor 0.8 */ h_Qmf->t_cos = qmf_phaseshift_cos40; h_Qmf->t_sin = qmf_phaseshift_sin40; h_Qmf->filterScale = 1; h_Qmf->p_stride = 1; h_Qmf->FilterSize = no_channels * 10; } break; case 32: h_Qmf->p_filter = qmf_pfilt640; if (flags & QMF_FLAG_DOWNSAMPLED) { h_Qmf->t_cos = qmf_phaseshift_cos_downsamp32; h_Qmf->t_sin = qmf_phaseshift_sin_downsamp32; } else { h_Qmf->t_cos = qmf_phaseshift_cos32; h_Qmf->t_sin = qmf_phaseshift_sin32; } h_Qmf->p_stride = 2; h_Qmf->FilterSize = 640; h_Qmf->filterScale = 0; break; case 20: h_Qmf->p_filter = qmf_pfilt200; h_Qmf->p_stride = 1; h_Qmf->FilterSize = 200; h_Qmf->filterScale = 0; break; case 12: h_Qmf->p_filter = qmf_pfilt120; h_Qmf->p_stride = 1; h_Qmf->FilterSize = 120; h_Qmf->filterScale = 0; break; case 8: h_Qmf->p_filter = qmf_pfilt640; h_Qmf->p_stride = 8; h_Qmf->FilterSize = 640; h_Qmf->filterScale = 0; break; case 16: h_Qmf->p_filter = qmf_pfilt640; h_Qmf->t_cos = qmf_phaseshift_cos16; h_Qmf->t_sin = qmf_phaseshift_sin16; h_Qmf->p_stride = 4; h_Qmf->FilterSize = 640; h_Qmf->filterScale = 0; break; case 24: h_Qmf->p_filter = qmf_pfilt240; h_Qmf->t_cos = qmf_phaseshift_cos24; h_Qmf->t_sin = qmf_phaseshift_sin24; h_Qmf->p_stride = 1; h_Qmf->FilterSize = 240; h_Qmf->filterScale = 1; break; default: return -1; } } h_Qmf->synScalefactor = h_Qmf->filterScale; // DCT|DST dependency switch (no_channels) { case 128: h_Qmf->synScalefactor += ALGORITHMIC_SCALING_IN_SYNTHESIS_FILTERBANK + 1; break; case 40: { h_Qmf->synScalefactor += ALGORITHMIC_SCALING_IN_SYNTHESIS_FILTERBANK - 1; } break; case 64: h_Qmf->synScalefactor += ALGORITHMIC_SCALING_IN_SYNTHESIS_FILTERBANK; break; case 8: h_Qmf->synScalefactor += ALGORITHMIC_SCALING_IN_SYNTHESIS_FILTERBANK - 3; break; case 12: h_Qmf->synScalefactor += ALGORITHMIC_SCALING_IN_SYNTHESIS_FILTERBANK; break; case 20: h_Qmf->synScalefactor += ALGORITHMIC_SCALING_IN_SYNTHESIS_FILTERBANK + 1; break; case 32: h_Qmf->synScalefactor += ALGORITHMIC_SCALING_IN_SYNTHESIS_FILTERBANK - 1; break; case 16: h_Qmf->synScalefactor += ALGORITHMIC_SCALING_IN_SYNTHESIS_FILTERBANK - 2; break; case 24: h_Qmf->synScalefactor += ALGORITHMIC_SCALING_IN_SYNTHESIS_FILTERBANK - 1; break; default: return -1; } h_Qmf->flags = flags; h_Qmf->no_channels = no_channels; h_Qmf->no_col = noCols; h_Qmf->lsb = fMin(lsb, h_Qmf->no_channels); h_Qmf->usb = synflag ? fMin(usb, h_Qmf->no_channels) : usb; /* was: h_Qmf->usb = fMin(usb, h_Qmf->no_channels); */ h_Qmf->FilterStates = (void *)pFilterStates; h_Qmf->outScalefactor = (ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK + h_Qmf->filterScale) + h_Qmf->synScalefactor; h_Qmf->outGain_m = (FIXP_DBL)0x80000000; /* default init value will be not applied */ h_Qmf->outGain_e = 0; return (0); } /*! * * \brief Adjust synthesis qmf filter states * * \return void * */ static inline void qmfAdaptFilterStates( HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Filter Bank */ int scaleFactorDiff) /*!< Scale factor difference to be applied */ { if (synQmf == NULL || synQmf->FilterStates == NULL) { return; } if (scaleFactorDiff > 0) { scaleValuesSaturate((FIXP_QSS *)synQmf->FilterStates, synQmf->no_channels * (QMF_NO_POLY * 2 - 1), scaleFactorDiff); } else { scaleValues((FIXP_QSS *)synQmf->FilterStates, synQmf->no_channels * (QMF_NO_POLY * 2 - 1), scaleFactorDiff); } } /*! * * \brief Create QMF filter bank instance * * * \return 0 if succesful * */ int qmfInitSynthesisFilterBank( HANDLE_QMF_FILTER_BANK h_Qmf, /*!< Returns handle */ FIXP_QSS *pFilterStates, /*!< Handle to filter states */ int noCols, /*!< Number of timeslots per frame */ int lsb, /*!< lower end of QMF */ int usb, /*!< upper end of QMF */ int no_channels, /*!< Number of channels (bands) */ int flags) /*!< Low Power flag */ { int oldOutScale = h_Qmf->outScalefactor; int err = qmfInitFilterBank(h_Qmf, pFilterStates, noCols, lsb, usb, no_channels, flags, 1); if (h_Qmf->FilterStates != NULL) { if (!(flags & QMF_FLAG_KEEP_STATES)) { FDKmemclear( h_Qmf->FilterStates, (2 * QMF_NO_POLY - 1) * h_Qmf->no_channels * sizeof(FIXP_QSS)); } else { qmfAdaptFilterStates(h_Qmf, oldOutScale - h_Qmf->outScalefactor); } } FDK_ASSERT(h_Qmf->no_channels >= h_Qmf->lsb); FDK_ASSERT(h_Qmf->no_channels >= h_Qmf->usb); return err; } /*! * * \brief Change scale factor for output data and adjust qmf filter states * * \return void * */ void qmfChangeOutScalefactor( HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */ int outScalefactor /*!< New scaling factor for output data */ ) { if (synQmf == NULL) { return; } /* Add internal filterbank scale */ outScalefactor += (ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK + synQmf->filterScale) + synQmf->synScalefactor; /* adjust filter states when scale factor has been changed */ if (synQmf->outScalefactor != outScalefactor) { int diff; diff = synQmf->outScalefactor - outScalefactor; qmfAdaptFilterStates(synQmf, diff); /* save new scale factor */ synQmf->outScalefactor = outScalefactor; } } /*! * * \brief Get scale factor change which was set by qmfChangeOutScalefactor() * * \return scaleFactor * */ int qmfGetOutScalefactor( HANDLE_QMF_FILTER_BANK synQmf) /*!< Handle of Qmf Synthesis Bank */ { int scaleFactor = synQmf->outScalefactor ? (synQmf->outScalefactor - (ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK + synQmf->filterScale + synQmf->synScalefactor)) : 0; return scaleFactor; } /*! * * \brief Change gain for output data * * \return void * */ void qmfChangeOutGain( HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */ FIXP_DBL outputGain, /*!< New gain for output data (mantissa) */ int outputGainScale /*!< New gain for output data (exponent) */ ) { synQmf->outGain_m = outputGain; synQmf->outGain_e = outputGainScale; } #define INT_PCM_QMFOUT INT_PCM #define SAMPLE_BITS_QMFOUT SAMPLE_BITS #include "qmf_pcm.h" #if SAMPLE_BITS == 16 /* also create a 32 bit output version */ #undef INT_PCM_QMFOUT #undef SAMPLE_BITS_QMFOUT #undef QMF_PCM_H #undef FIXP_QAS #undef QAS_BITS #undef INT_PCM_QMFIN #define INT_PCM_QMFOUT LONG #define SAMPLE_BITS_QMFOUT 32 #define FIXP_QAS FIXP_DBL #define QAS_BITS 32 #define INT_PCM_QMFIN LONG #include "qmf_pcm.h" #endif fdk-aac-2.0.2/libFDK/src/scale.cpp000066400000000000000000000542541404225310700164660ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* Library for basic calculation routines ******************** Author(s): Description: Scaling operations *******************************************************************************/ #include "common_fix.h" #include "genericStds.h" /************************************************** * Inline definitions **************************************************/ #include "scale.h" #if defined(__mips__) #include "mips/scale_mips.cpp" #elif defined(__arm__) #include "arm/scale_arm.cpp" #endif #ifndef FUNCTION_scaleValues_SGL /*! * * \brief Multiply input vector by \f$ 2^{scalefactor} \f$ * \param len must be larger than 4 * \return void * */ #define FUNCTION_scaleValues_SGL void scaleValues(FIXP_SGL *vector, /*!< Vector */ INT len, /*!< Length */ INT scalefactor /*!< Scalefactor */ ) { INT i; /* Return if scalefactor is Zero */ if (scalefactor == 0) return; if (scalefactor > 0) { scalefactor = fixmin_I(scalefactor, (INT)(FRACT_BITS - 1)); for (i = len & 3; i--;) { *(vector++) <<= scalefactor; } for (i = len >> 2; i--;) { *(vector++) <<= scalefactor; *(vector++) <<= scalefactor; *(vector++) <<= scalefactor; *(vector++) <<= scalefactor; } } else { INT negScalefactor = fixmin_I(-scalefactor, (INT)FRACT_BITS - 1); for (i = len & 3; i--;) { *(vector++) >>= negScalefactor; } for (i = len >> 2; i--;) { *(vector++) >>= negScalefactor; *(vector++) >>= negScalefactor; *(vector++) >>= negScalefactor; *(vector++) >>= negScalefactor; } } } #endif #ifndef FUNCTION_scaleValues_DBL /*! * * \brief Multiply input vector by \f$ 2^{scalefactor} \f$ * \param len must be larger than 4 * \return void * */ #define FUNCTION_scaleValues_DBL SCALE_INLINE void scaleValues(FIXP_DBL *vector, /*!< Vector */ INT len, /*!< Length */ INT scalefactor /*!< Scalefactor */ ) { INT i; /* Return if scalefactor is Zero */ if (scalefactor == 0) return; if (scalefactor > 0) { scalefactor = fixmin_I(scalefactor, (INT)DFRACT_BITS - 1); for (i = len & 3; i--;) { *(vector++) <<= scalefactor; } for (i = len >> 2; i--;) { *(vector++) <<= scalefactor; *(vector++) <<= scalefactor; *(vector++) <<= scalefactor; *(vector++) <<= scalefactor; } } else { INT negScalefactor = fixmin_I(-scalefactor, (INT)DFRACT_BITS - 1); for (i = len & 3; i--;) { *(vector++) >>= negScalefactor; } for (i = len >> 2; i--;) { *(vector++) >>= negScalefactor; *(vector++) >>= negScalefactor; *(vector++) >>= negScalefactor; *(vector++) >>= negScalefactor; } } } #endif #ifndef FUNCTION_scaleValuesSaturate_DBL /*! * * \brief Multiply input vector by \f$ 2^{scalefactor} \f$ * \param vector source/destination buffer * \param len length of vector * \param scalefactor amount of shifts to be applied * \return void * */ #define FUNCTION_scaleValuesSaturate_DBL SCALE_INLINE void scaleValuesSaturate(FIXP_DBL *vector, /*!< Vector */ INT len, /*!< Length */ INT scalefactor /*!< Scalefactor */ ) { INT i; /* Return if scalefactor is Zero */ if (scalefactor == 0) return; scalefactor = fixmax_I(fixmin_I(scalefactor, (INT)DFRACT_BITS - 1), (INT) - (DFRACT_BITS - 1)); for (i = 0; i < len; i++) { vector[i] = scaleValueSaturate(vector[i], scalefactor); } } #endif /* FUNCTION_scaleValuesSaturate_DBL */ #ifndef FUNCTION_scaleValuesSaturate_DBL_DBL /*! * * \brief Multiply input vector by \f$ 2^{scalefactor} \f$ * \param dst destination buffer * \param src source buffer * \param len length of vector * \param scalefactor amount of shifts to be applied * \return void * */ #define FUNCTION_scaleValuesSaturate_DBL_DBL SCALE_INLINE void scaleValuesSaturate(FIXP_DBL *dst, /*!< Output */ const FIXP_DBL *src, /*!< Input */ INT len, /*!< Length */ INT scalefactor /*!< Scalefactor */ ) { INT i; /* Return if scalefactor is Zero */ if (scalefactor == 0) { FDKmemmove(dst, src, len * sizeof(FIXP_DBL)); return; } scalefactor = fixmax_I(fixmin_I(scalefactor, (INT)DFRACT_BITS - 1), (INT) - (DFRACT_BITS - 1)); for (i = 0; i < len; i++) { dst[i] = scaleValueSaturate(src[i], scalefactor); } } #endif /* FUNCTION_scaleValuesSaturate_DBL_DBL */ #ifndef FUNCTION_scaleValuesSaturate_SGL_DBL /*! * * \brief Multiply input vector by \f$ 2^{scalefactor} \f$ * \param dst destination buffer (FIXP_SGL) * \param src source buffer (FIXP_DBL) * \param len length of vector * \param scalefactor amount of shifts to be applied * \return void * */ #define FUNCTION_scaleValuesSaturate_SGL_DBL SCALE_INLINE void scaleValuesSaturate(FIXP_SGL *dst, /*!< Output */ const FIXP_DBL *src, /*!< Input */ INT len, /*!< Length */ INT scalefactor) /*!< Scalefactor */ { INT i; scalefactor = fixmax_I(fixmin_I(scalefactor, (INT)DFRACT_BITS - 1), (INT) - (DFRACT_BITS - 1)); for (i = 0; i < len; i++) { dst[i] = FX_DBL2FX_SGL(fAddSaturate(scaleValueSaturate(src[i], scalefactor), (FIXP_DBL)0x8000)); } } #endif /* FUNCTION_scaleValuesSaturate_SGL_DBL */ #ifndef FUNCTION_scaleValuesSaturate_SGL /*! * * \brief Multiply input vector by \f$ 2^{scalefactor} \f$ * \param vector source/destination buffer * \param len length of vector * \param scalefactor amount of shifts to be applied * \return void * */ #define FUNCTION_scaleValuesSaturate_SGL SCALE_INLINE void scaleValuesSaturate(FIXP_SGL *vector, /*!< Vector */ INT len, /*!< Length */ INT scalefactor /*!< Scalefactor */ ) { INT i; /* Return if scalefactor is Zero */ if (scalefactor == 0) return; scalefactor = fixmax_I(fixmin_I(scalefactor, (INT)DFRACT_BITS - 1), (INT) - (DFRACT_BITS - 1)); for (i = 0; i < len; i++) { vector[i] = FX_DBL2FX_SGL( scaleValueSaturate(FX_SGL2FX_DBL(vector[i]), scalefactor)); } } #endif /* FUNCTION_scaleValuesSaturate_SGL */ #ifndef FUNCTION_scaleValuesSaturate_SGL_SGL /*! * * \brief Multiply input vector by \f$ 2^{scalefactor} \f$ * \param dst destination buffer * \param src source buffer * \param len length of vector * \param scalefactor amount of shifts to be applied * \return void * */ #define FUNCTION_scaleValuesSaturate_SGL_SGL SCALE_INLINE void scaleValuesSaturate(FIXP_SGL *dst, /*!< Output */ const FIXP_SGL *src, /*!< Input */ INT len, /*!< Length */ INT scalefactor /*!< Scalefactor */ ) { INT i; /* Return if scalefactor is Zero */ if (scalefactor == 0) { FDKmemmove(dst, src, len * sizeof(FIXP_SGL)); return; } scalefactor = fixmax_I(fixmin_I(scalefactor, (INT)DFRACT_BITS - 1), (INT) - (DFRACT_BITS - 1)); for (i = 0; i < len; i++) { dst[i] = FX_DBL2FX_SGL(scaleValueSaturate(FX_SGL2FX_DBL(src[i]), scalefactor)); } } #endif /* FUNCTION_scaleValuesSaturate_SGL_SGL */ #ifndef FUNCTION_scaleValues_DBLDBL /*! * * \brief Multiply input vector src by \f$ 2^{scalefactor} \f$ * and place result into dst * \param dst detination buffer * \param src source buffer * \param len must be larger than 4 * \param scalefactor amount of left shifts to be applied * \return void * */ #define FUNCTION_scaleValues_DBLDBL SCALE_INLINE void scaleValues(FIXP_DBL *dst, /*!< dst Vector */ const FIXP_DBL *src, /*!< src Vector */ INT len, /*!< Length */ INT scalefactor /*!< Scalefactor */ ) { INT i; /* Return if scalefactor is Zero */ if (scalefactor == 0) { if (dst != src) FDKmemmove(dst, src, len * sizeof(FIXP_DBL)); } else { if (scalefactor > 0) { scalefactor = fixmin_I(scalefactor, (INT)DFRACT_BITS - 1); for (i = len & 3; i--;) { *(dst++) = *(src++) << scalefactor; } for (i = len >> 2; i--;) { *(dst++) = *(src++) << scalefactor; *(dst++) = *(src++) << scalefactor; *(dst++) = *(src++) << scalefactor; *(dst++) = *(src++) << scalefactor; } } else { INT negScalefactor = fixmin_I(-scalefactor, (INT)DFRACT_BITS - 1); for (i = len & 3; i--;) { *(dst++) = *(src++) >> negScalefactor; } for (i = len >> 2; i--;) { *(dst++) = *(src++) >> negScalefactor; *(dst++) = *(src++) >> negScalefactor; *(dst++) = *(src++) >> negScalefactor; *(dst++) = *(src++) >> negScalefactor; } } } } #endif #if (SAMPLE_BITS == 16) #ifndef FUNCTION_scaleValues_PCMDBL /*! * * \brief Multiply input vector src by \f$ 2^{scalefactor} \f$ * and place result into dst * \param dst detination buffer * \param src source buffer * \param len must be larger than 4 * \param scalefactor amount of left shifts to be applied * \return void * */ #define FUNCTION_scaleValues_PCMDBL SCALE_INLINE void scaleValues(FIXP_PCM *dst, /*!< dst Vector */ const FIXP_DBL *src, /*!< src Vector */ INT len, /*!< Length */ INT scalefactor /*!< Scalefactor */ ) { INT i; scalefactor -= DFRACT_BITS - SAMPLE_BITS; /* Return if scalefactor is Zero */ { if (scalefactor > 0) { scalefactor = fixmin_I(scalefactor, (INT)DFRACT_BITS - 1); for (i = len & 3; i--;) { *(dst++) = (FIXP_PCM)(*(src++) << scalefactor); } for (i = len >> 2; i--;) { *(dst++) = (FIXP_PCM)(*(src++) << scalefactor); *(dst++) = (FIXP_PCM)(*(src++) << scalefactor); *(dst++) = (FIXP_PCM)(*(src++) << scalefactor); *(dst++) = (FIXP_PCM)(*(src++) << scalefactor); } } else { INT negScalefactor = fixmin_I(-scalefactor, (INT)DFRACT_BITS - 1); for (i = len & 3; i--;) { *(dst++) = (FIXP_PCM)(*(src++) >> negScalefactor); } for (i = len >> 2; i--;) { *(dst++) = (FIXP_PCM)(*(src++) >> negScalefactor); *(dst++) = (FIXP_PCM)(*(src++) >> negScalefactor); *(dst++) = (FIXP_PCM)(*(src++) >> negScalefactor); *(dst++) = (FIXP_PCM)(*(src++) >> negScalefactor); } } } } #endif #endif /* (SAMPLE_BITS == 16) */ #ifndef FUNCTION_scaleValues_SGLSGL /*! * * \brief Multiply input vector src by \f$ 2^{scalefactor} \f$ * and place result into dst * \param dst detination buffer * \param src source buffer * \param len must be larger than 4 * \param scalefactor amount of left shifts to be applied * \return void * */ #define FUNCTION_scaleValues_SGLSGL SCALE_INLINE void scaleValues(FIXP_SGL *dst, /*!< dst Vector */ const FIXP_SGL *src, /*!< src Vector */ INT len, /*!< Length */ INT scalefactor /*!< Scalefactor */ ) { INT i; /* Return if scalefactor is Zero */ if (scalefactor == 0) { if (dst != src) FDKmemmove(dst, src, len * sizeof(FIXP_DBL)); } else { if (scalefactor > 0) { scalefactor = fixmin_I(scalefactor, (INT)DFRACT_BITS - 1); for (i = len & 3; i--;) { *(dst++) = *(src++) << scalefactor; } for (i = len >> 2; i--;) { *(dst++) = *(src++) << scalefactor; *(dst++) = *(src++) << scalefactor; *(dst++) = *(src++) << scalefactor; *(dst++) = *(src++) << scalefactor; } } else { INT negScalefactor = fixmin_I(-scalefactor, (INT)DFRACT_BITS - 1); for (i = len & 3; i--;) { *(dst++) = *(src++) >> negScalefactor; } for (i = len >> 2; i--;) { *(dst++) = *(src++) >> negScalefactor; *(dst++) = *(src++) >> negScalefactor; *(dst++) = *(src++) >> negScalefactor; *(dst++) = *(src++) >> negScalefactor; } } } } #endif #ifndef FUNCTION_scaleValuesWithFactor_DBL /*! * * \brief Multiply input vector by \f$ 2^{scalefactor} \f$ * \param len must be larger than 4 * \return void * */ #define FUNCTION_scaleValuesWithFactor_DBL SCALE_INLINE void scaleValuesWithFactor(FIXP_DBL *vector, FIXP_DBL factor, INT len, INT scalefactor) { INT i; /* Compensate fMultDiv2 */ scalefactor++; if (scalefactor > 0) { scalefactor = fixmin_I(scalefactor, (INT)DFRACT_BITS - 1); for (i = len & 3; i--;) { *vector = fMultDiv2(*vector, factor) << scalefactor; vector++; } for (i = len >> 2; i--;) { *vector = fMultDiv2(*vector, factor) << scalefactor; vector++; *vector = fMultDiv2(*vector, factor) << scalefactor; vector++; *vector = fMultDiv2(*vector, factor) << scalefactor; vector++; *vector = fMultDiv2(*vector, factor) << scalefactor; vector++; } } else { INT negScalefactor = fixmin_I(-scalefactor, (INT)DFRACT_BITS - 1); for (i = len & 3; i--;) { *vector = fMultDiv2(*vector, factor) >> negScalefactor; vector++; } for (i = len >> 2; i--;) { *vector = fMultDiv2(*vector, factor) >> negScalefactor; vector++; *vector = fMultDiv2(*vector, factor) >> negScalefactor; vector++; *vector = fMultDiv2(*vector, factor) >> negScalefactor; vector++; *vector = fMultDiv2(*vector, factor) >> negScalefactor; vector++; } } } #endif /* FUNCTION_scaleValuesWithFactor_DBL */ /******************************************* IMPORTANT NOTE for usage of getScalefactor() If the input array contains negative values too, then these functions may sometimes return the actual maximum value minus 1, due to the nature of the applied algorithm. So be careful with possible fractional -1 values that may lead to overflows when being fPow2()'ed. ********************************************/ #ifndef FUNCTION_getScalefactorShort /*! * * \brief Calculate max possible scale factor for input vector of shorts * * \return Maximum scale factor / possible left shift * */ #define FUNCTION_getScalefactorShort SCALE_INLINE INT getScalefactorShort(const SHORT *vector, /*!< Pointer to input vector */ INT len /*!< Length of input vector */ ) { INT i; SHORT temp, maxVal = 0; for (i = len; i != 0; i--) { temp = (SHORT)(*vector++); maxVal |= (temp ^ (temp >> (SHORT_BITS - 1))); } return fixmax_I((INT)0, (INT)(fixnormz_D((INT)maxVal) - (INT)1 - (INT)(DFRACT_BITS - SHORT_BITS))); } #endif #ifndef FUNCTION_getScalefactorPCM /*! * * \brief Calculate max possible scale factor for input vector of shorts * * \return Maximum scale factor * */ #define FUNCTION_getScalefactorPCM SCALE_INLINE INT getScalefactorPCM(const INT_PCM *vector, /*!< Pointer to input vector */ INT len, /*!< Length of input vector */ INT stride) { INT i; INT_PCM temp, maxVal = 0; for (i = len; i != 0; i--) { temp = (INT_PCM)(*vector); vector += stride; maxVal |= (temp ^ (temp >> ((sizeof(INT_PCM) * 8) - 1))); } return fixmax_I((INT)0, (INT)(fixnormz_D((INT)maxVal) - (INT)1 - (INT)(DFRACT_BITS - SAMPLE_BITS))); } #endif #ifndef FUNCTION_getScalefactorShort /*! * * \brief Calculate max possible scale factor for input vector of shorts * \param stride, item increment between vector members. * \return Maximum scale factor * */ #define FUNCTION_getScalefactorShort SCALE_INLINE INT getScalefactorShort(const SHORT *vector, /*!< Pointer to input vector */ INT len, /*!< Length of input vector */ INT stride) { INT i; SHORT temp, maxVal = 0; for (i = len; i != 0; i--) { temp = (SHORT)(*vector); vector += stride; maxVal |= (temp ^ (temp >> (SHORT_BITS - 1))); } return fixmax_I((INT)0, (INT)(fixnormz_D((INT)maxVal) - (INT)1 - (INT)(DFRACT_BITS - SHORT_BITS))); } #endif #ifndef FUNCTION_getScalefactor_DBL /*! * * \brief Calculate max possible scale factor for input vector * * \return Maximum scale factor * * This function can constitute a significant amount of computational * complexity - very much depending on the bitrate. Since it is a rather small * function, effective assembler optimization might be possible. * * If all data is 0xFFFF.FFFF or 0x0000.0000 function returns 31 * Note: You can skip data normalization only if return value is 0 * */ #define FUNCTION_getScalefactor_DBL SCALE_INLINE INT getScalefactor(const FIXP_DBL *vector, /*!< Pointer to input vector */ INT len) /*!< Length of input vector */ { INT i; FIXP_DBL temp, maxVal = (FIXP_DBL)0; for (i = len; i != 0; i--) { temp = (LONG)(*vector++); maxVal |= (FIXP_DBL)((LONG)temp ^ (LONG)(temp >> (DFRACT_BITS - 1))); } return fixmax_I((INT)0, (INT)(fixnormz_D(maxVal) - 1)); } #endif #ifndef FUNCTION_getScalefactor_SGL #define FUNCTION_getScalefactor_SGL SCALE_INLINE INT getScalefactor(const FIXP_SGL *vector, /*!< Pointer to input vector */ INT len) /*!< Length of input vector */ { INT i; SHORT temp, maxVal = (FIXP_SGL)0; for (i = len; i != 0; i--) { temp = (SHORT)(*vector++); maxVal |= (temp ^ (temp >> (FRACT_BITS - 1))); } return fixmax_I((INT)0, (INT)(fixnormz_S((FIXP_SGL)maxVal)) - 1); } #endif fdk-aac-2.0.2/libMpegTPDec/000077500000000000000000000000001404225310700152365ustar00rootroot00000000000000fdk-aac-2.0.2/libMpegTPDec/include/000077500000000000000000000000001404225310700166615ustar00rootroot00000000000000fdk-aac-2.0.2/libMpegTPDec/include/tp_data.h000066400000000000000000000421121404225310700204460ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* MPEG transport format decoder library ********************* Author(s): Manuel Jander Description: MPEG Transport data tables *******************************************************************************/ #ifndef TP_DATA_H #define TP_DATA_H #include "machine_type.h" #include "FDK_audio.h" #include "FDK_bitstream.h" /* * Configuration */ #define TP_USAC_MAX_SPEAKERS (24) #define TP_USAC_MAX_EXT_ELEMENTS ((24)) #define TP_USAC_MAX_ELEMENTS ((24) + TP_USAC_MAX_EXT_ELEMENTS) #define TP_USAC_MAX_CONFIG_LEN \ 512 /* next power of two of maximum of escapedValue(hBs, 4, 4, 8) in \ AudioPreRoll() (285) */ #define TPDEC_USAC_NUM_CONFIG_CHANGE_FRAMES \ (1) /* Number of frames for config change in USAC */ enum { TPDEC_FLUSH_OFF = 0, TPDEC_RSV60_CFG_CHANGE_ATSC_FLUSH_ON = 1, TPDEC_RSV60_DASH_IPF_ATSC_FLUSH_ON = 2, TPDEC_USAC_DASH_IPF_FLUSH_ON = 3 }; enum { TPDEC_BUILD_UP_OFF = 0, TPDEC_RSV60_BUILD_UP_ON = 1, TPDEC_RSV60_BUILD_UP_ON_IN_BAND = 2, TPDEC_USAC_BUILD_UP_ON = 3, TPDEC_RSV60_BUILD_UP_IDLE = 4, TPDEC_RSV60_BUILD_UP_IDLE_IN_BAND = 5 }; /** * ProgramConfig struct. */ /* ISO/IEC 14496-3 4.4.1.1 Table 4.2 Program config element */ #define PC_FSB_CHANNELS_MAX 16 /* Front/Side/Back channels */ #define PC_LFE_CHANNELS_MAX 4 #define PC_ASSOCDATA_MAX 8 #define PC_CCEL_MAX 16 /* CC elements */ #define PC_COMMENTLENGTH 256 #define PC_NUM_HEIGHT_LAYER 3 typedef struct { /* PCE bitstream elements: */ UCHAR ElementInstanceTag; UCHAR Profile; UCHAR SamplingFrequencyIndex; UCHAR NumFrontChannelElements; UCHAR NumSideChannelElements; UCHAR NumBackChannelElements; UCHAR NumLfeChannelElements; UCHAR NumAssocDataElements; UCHAR NumValidCcElements; UCHAR MonoMixdownPresent; UCHAR MonoMixdownElementNumber; UCHAR StereoMixdownPresent; UCHAR StereoMixdownElementNumber; UCHAR MatrixMixdownIndexPresent; UCHAR MatrixMixdownIndex; UCHAR PseudoSurroundEnable; UCHAR FrontElementIsCpe[PC_FSB_CHANNELS_MAX]; UCHAR FrontElementTagSelect[PC_FSB_CHANNELS_MAX]; UCHAR FrontElementHeightInfo[PC_FSB_CHANNELS_MAX]; UCHAR SideElementIsCpe[PC_FSB_CHANNELS_MAX]; UCHAR SideElementTagSelect[PC_FSB_CHANNELS_MAX]; UCHAR SideElementHeightInfo[PC_FSB_CHANNELS_MAX]; UCHAR BackElementIsCpe[PC_FSB_CHANNELS_MAX]; UCHAR BackElementTagSelect[PC_FSB_CHANNELS_MAX]; UCHAR BackElementHeightInfo[PC_FSB_CHANNELS_MAX]; UCHAR LfeElementTagSelect[PC_LFE_CHANNELS_MAX]; UCHAR AssocDataElementTagSelect[PC_ASSOCDATA_MAX]; UCHAR CcElementIsIndSw[PC_CCEL_MAX]; UCHAR ValidCcElementTagSelect[PC_CCEL_MAX]; UCHAR CommentFieldBytes; UCHAR Comment[PC_COMMENTLENGTH]; /* Helper variables for administration: */ UCHAR isValid; /*!< Flag showing if PCE has been read successfully. */ UCHAR NumChannels; /*!< Amount of audio channels summing all channel elements including LFEs */ UCHAR NumEffectiveChannels; /*!< Amount of audio channels summing only SCEs and CPEs */ UCHAR elCounter; } CProgramConfig; typedef enum { ASCEXT_UNKOWN = -1, ASCEXT_SBR = 0x2b7, ASCEXT_PS = 0x548, ASCEXT_MPS = 0x76a, ASCEXT_SAOC = 0x7cb, ASCEXT_LDMPS = 0x7cc } TP_ASC_EXTENSION_ID; /** * GaSpecificConfig struct */ typedef struct { UINT m_frameLengthFlag; UINT m_dependsOnCoreCoder; UINT m_coreCoderDelay; UINT m_extensionFlag; UINT m_extensionFlag3; UINT m_layer; UINT m_numOfSubFrame; UINT m_layerLength; } CSGaSpecificConfig; typedef enum { ELDEXT_TERM = 0x0, /* Termination tag */ ELDEXT_SAOC = 0x1, /* SAOC config */ ELDEXT_LDSAC = 0x2, /* LD MPEG Surround config */ ELDEXT_DOWNSCALEINFO = 0x3 /* ELD sample rate adaptation */ /* reserved */ } ASC_ELD_EXT_TYPE; typedef struct { UCHAR m_frameLengthFlag; UCHAR m_sbrPresentFlag; UCHAR m_useLdQmfTimeAlign; /* Use LD-MPS QMF in SBR to achive time alignment */ UCHAR m_sbrSamplingRate; UCHAR m_sbrCrcFlag; UINT m_downscaledSamplingFrequency; } CSEldSpecificConfig; typedef struct { USAC_EXT_ELEMENT_TYPE usacExtElementType; USHORT usacExtElementConfigLength; USHORT usacExtElementDefaultLength; UCHAR usacExtElementPayloadFrag; UCHAR usacExtElementHasAudioPreRoll; } CSUsacExtElementConfig; typedef struct { MP4_ELEMENT_ID usacElementType; UCHAR m_noiseFilling; UCHAR m_harmonicSBR; UCHAR m_interTes; UCHAR m_pvc; UCHAR m_stereoConfigIndex; CSUsacExtElementConfig extElement; } CSUsacElementConfig; typedef struct { UCHAR m_frameLengthFlag; UCHAR m_coreSbrFrameLengthIndex; UCHAR m_sbrRatioIndex; UCHAR m_nUsacChannels; /* number of audio channels signaled in UsacDecoderConfig() / rsv603daDecoderConfig() via numElements and usacElementType */ UCHAR m_channelConfigurationIndex; UINT m_usacNumElements; CSUsacElementConfig element[TP_USAC_MAX_ELEMENTS]; UCHAR numAudioChannels; UCHAR m_usacConfigExtensionPresent; UCHAR elementLengthPresent; UCHAR UsacConfig[TP_USAC_MAX_CONFIG_LEN]; USHORT UsacConfigBits; } CSUsacConfig; /** * Audio configuration struct, suitable for encoder and decoder configuration. */ typedef struct { /* XYZ Specific Data */ union { CSGaSpecificConfig m_gaSpecificConfig; /**< General audio specific configuration. */ CSEldSpecificConfig m_eldSpecificConfig; /**< ELD specific configuration. */ CSUsacConfig m_usacConfig; /**< USAC specific configuration */ } m_sc; /* Common ASC parameters */ CProgramConfig m_progrConfigElement; /**< Program configuration. */ AUDIO_OBJECT_TYPE m_aot; /**< Audio Object Type. */ UINT m_samplingFrequency; /**< Samplerate. */ UINT m_samplesPerFrame; /**< Amount of samples per frame. */ UINT m_directMapping; /**< Document this please !! */ AUDIO_OBJECT_TYPE m_extensionAudioObjectType; /**< Audio object type */ UINT m_extensionSamplingFrequency; /**< Samplerate */ SCHAR m_channelConfiguration; /**< Channel configuration index */ SCHAR m_epConfig; /**< Error protection index */ SCHAR m_vcb11Flag; /**< aacSectionDataResilienceFlag */ SCHAR m_rvlcFlag; /**< aacScalefactorDataResilienceFlag */ SCHAR m_hcrFlag; /**< aacSpectralDataResilienceFlag */ SCHAR m_sbrPresentFlag; /**< Flag indicating the presence of SBR data in the bitstream */ SCHAR m_psPresentFlag; /**< Flag indicating the presence of parametric stereo data in the bitstream */ UCHAR m_samplingFrequencyIndex; /**< Samplerate index */ UCHAR m_extensionSamplingFrequencyIndex; /**< Samplerate index */ SCHAR m_extensionChannelConfiguration; /**< Channel configuration index */ UCHAR configMode; /**< The flag indicates if the callback shall work in memory allocation mode or in config change detection mode */ UCHAR AacConfigChanged; /**< The flag will be set if at least one aac config parameter has changed that requires a memory reconfiguration, otherwise it will be cleared */ UCHAR SbrConfigChanged; /**< The flag will be set if at least one sbr config parameter has changed that requires a memory reconfiguration, otherwise it will be cleared */ UCHAR SacConfigChanged; /**< The flag will be set if at least one sac config parameter has changed that requires a memory reconfiguration, otherwise it will be cleared */ UCHAR config[TP_USAC_MAX_CONFIG_LEN]; /**< Configuration stored as bitstream */ UINT configBits; /**< Configuration length in bits */ } CSAudioSpecificConfig; typedef struct { SCHAR flushCnt; /**< Flush frame counter */ UCHAR flushStatus; /**< Flag indicates flush mode: on|off */ SCHAR buildUpCnt; /**< Build up frame counter */ UCHAR buildUpStatus; /**< Flag indicates build up mode: on|off */ UCHAR cfgChanged; /**< Flag indicates that the config changed and the decoder needs to be initialized again via callback. Make sure that memory is freed before initialization. */ UCHAR contentChanged; /**< Flag indicates that the content changed i.e. a right truncation occured before */ UCHAR forceCfgChange; /**< Flag indicates if config change has to be forced even if new config is the same */ } CCtrlCFGChange; typedef INT (*cbUpdateConfig_t)(void *, const CSAudioSpecificConfig *, const UCHAR configMode, UCHAR *configChanged); typedef INT (*cbFreeMem_t)(void *, const CSAudioSpecificConfig *); typedef INT (*cbCtrlCFGChange_t)(void *, const CCtrlCFGChange *); typedef INT (*cbSsc_t)(void *, HANDLE_FDK_BITSTREAM, const AUDIO_OBJECT_TYPE coreCodec, const INT samplingRate, const INT frameSize, const INT stereoConfigIndex, const INT coreSbrFrameLengthIndex, const INT configBytes, const UCHAR configMode, UCHAR *configChanged); typedef INT (*cbSbr_t)(void *self, HANDLE_FDK_BITSTREAM hBs, const INT sampleRateIn, const INT sampleRateOut, const INT samplesPerFrame, const AUDIO_OBJECT_TYPE coreCodec, const MP4_ELEMENT_ID elementID, const INT elementIndex, const UCHAR harmonicSbr, const UCHAR stereoConfigIndex, const UCHAR configMode, UCHAR *configChanged, const INT downscaleFactor); typedef INT (*cbUsac_t)(void *self, HANDLE_FDK_BITSTREAM hBs); typedef INT (*cbUniDrc_t)(void *self, HANDLE_FDK_BITSTREAM hBs, const INT fullPayloadLength, const INT payloadType, const INT subStreamIndex, const INT payloadStart, const AUDIO_OBJECT_TYPE); typedef struct { cbUpdateConfig_t cbUpdateConfig; /*!< Function pointer for Config change notify callback. */ void *cbUpdateConfigData; /*!< User data pointer for Config change notify callback. */ cbFreeMem_t cbFreeMem; /*!< Function pointer for free memory callback. */ void *cbFreeMemData; /*!< User data pointer for free memory callback. */ cbCtrlCFGChange_t cbCtrlCFGChange; /*!< Function pointer for config change control callback. */ void *cbCtrlCFGChangeData; /*!< User data pointer for config change control callback. */ cbSsc_t cbSsc; /*!< Function pointer for SSC parser callback. */ void *cbSscData; /*!< User data pointer for SSC parser callback. */ cbSbr_t cbSbr; /*!< Function pointer for SBR header parser callback. */ void *cbSbrData; /*!< User data pointer for SBR header parser callback. */ cbUsac_t cbUsac; void *cbUsacData; cbUniDrc_t cbUniDrc; /*!< Function pointer for uniDrcConfig and loudnessInfoSet parser callback. */ void *cbUniDrcData; /*!< User data pointer for uniDrcConfig and loudnessInfoSet parser callback. */ } CSTpCallBacks; static const UINT SamplingRateTable[] = { 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000, 7350, 0, 0, 57600, 51200, 40000, 38400, 34150, 28800, 25600, 20000, 19200, 17075, 14400, 12800, 9600, 0, 0, 0, 0}; static inline int getSamplingRateIndex(UINT samplingRate, UINT nBits) { UINT sf_index; UINT tableSize = (1 << nBits) - 1; for (sf_index = 0; sf_index < tableSize; sf_index++) { if (SamplingRateTable[sf_index] == samplingRate) break; } if (sf_index > tableSize) { return tableSize - 1; } return sf_index; } /* * Get Channel count from channel configuration */ static inline int getNumberOfTotalChannels(int channelConfig) { switch (channelConfig) { case 1: case 2: case 3: case 4: case 5: case 6: return channelConfig; case 7: case 12: case 14: return 8; case 11: return 7; case 13: return 24; default: return 0; } } static inline int getNumberOfEffectiveChannels( const int channelConfig) { /* index: 0,1,2,3,4,5,6,7,8,9,10,11,12,13,14,15 */ const int n[] = {0, 1, 2, 3, 4, 5, 5, 7, 0, 0, 0, 6, 7, 22, 7, 0}; return n[channelConfig]; } #endif /* TP_DATA_H */ fdk-aac-2.0.2/libMpegTPDec/include/tpdec_lib.h000066400000000000000000000676071404225310700207770ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* MPEG transport format decoder library ********************* Author(s): Manuel Jander Description: MPEG Transport decoder *******************************************************************************/ #ifndef TPDEC_LIB_H #define TPDEC_LIB_H #include "tp_data.h" #include "FDK_bitstream.h" typedef enum { TRANSPORTDEC_OK = 0, /*!< All fine. */ /* Synchronization errors. Wait for new input data and try again. */ tpdec_sync_error_start = 0x100, TRANSPORTDEC_NOT_ENOUGH_BITS, /*!< Out of bits. Provide more bits and try again. */ TRANSPORTDEC_SYNC_ERROR, /*!< No sync was found or sync got lost. Keep trying. */ tpdec_sync_error_end, /* Decode errors. Mostly caused due to bit errors. */ tpdec_decode_error_start = 0x400, TRANSPORTDEC_PARSE_ERROR, /*!< Bitstream data showed inconsistencies (wrong syntax). */ TRANSPORTDEC_UNSUPPORTED_FORMAT, /*!< Unsupported format or feature found in the bitstream data. */ TRANSPORTDEC_CRC_ERROR, /*!< CRC error encountered in bitstream data. */ tpdec_decode_error_end, /* Fatal errors. Stop immediately on one of these errors! */ tpdec_fatal_error_start = 0x200, TRANSPORTDEC_UNKOWN_ERROR, /*!< An unknown error occured. */ TRANSPORTDEC_INVALID_PARAMETER, /*!< An invalid parameter was passed to a function. */ TRANSPORTDEC_NEED_TO_RESTART, /*!< The decoder needs to be restarted, since the requiered configuration change cannot be performed. */ TRANSPORTDEC_TOO_MANY_BITS, /*!< In case of packet based formats: Supplied number of bits exceed the size of the internal bit buffer. */ tpdec_fatal_error_end } TRANSPORTDEC_ERROR; /** Macro to identify decode errors. */ #define TPDEC_IS_DECODE_ERROR(err) \ (((err >= tpdec_decode_error_start) && (err <= tpdec_decode_error_end)) ? 1 \ : 0) /** Macro to identify fatal errors. */ #define TPDEC_IS_FATAL_ERROR(err) \ (((err >= tpdec_fatal_error_start) && (err <= tpdec_fatal_error_end)) ? 1 : 0) /** * \brief Parameter identifiers for transportDec_SetParam() */ typedef enum { TPDEC_PARAM_MINIMIZE_DELAY = 1, /** Delay minimization strategy. 0: none, 1: discard as many frames as possible. */ TPDEC_PARAM_EARLY_CONFIG, /** Enable early config discovery. */ TPDEC_PARAM_IGNORE_BUFFERFULLNESS, /** Ignore buffer fullness. */ TPDEC_PARAM_SET_BITRATE, /** Set average bit rate for bit stream interruption frame misses estimation. */ TPDEC_PARAM_RESET, /** Reset transport decoder instance status. */ TPDEC_PARAM_BURST_PERIOD, /** Set data reception burst period in mili seconds. */ TPDEC_PARAM_TARGETLAYOUT, /** Set CICP target layout */ TPDEC_PARAM_FORCE_CONFIG_CHANGE, /** Force config change for next received config */ TPDEC_PARAM_USE_ELEM_SKIPPING } TPDEC_PARAM; /*! \brief Reset Program Config Element. \param pPce Program Config Element structure. \return void */ void CProgramConfig_Reset(CProgramConfig *pPce); /*! \brief Initialize Program Config Element. \param pPce Program Config Element structure. \return void */ void CProgramConfig_Init(CProgramConfig *pPce); /*! \brief Inquire state of present Program Config Element structure. \param pPce Program Config Element structure. \return 1 if the PCE structure is filled correct, 0 if no valid PCE present. */ int CProgramConfig_IsValid(const CProgramConfig *pPce); /*! \brief Read Program Config Element. \param pPce Program Config Element structure. \param bs Bitstream buffer to read from. \param alignAnchor Align bitstream to alignAnchor bits after all read operations. \return void */ void CProgramConfig_Read(CProgramConfig *pPce, HANDLE_FDK_BITSTREAM bs, UINT alignAnchor); /*! \brief Compare two Program Config Elements. \param pPce1 Pointer to first Program Config Element structure. \param pPce2 Pointer to second Program Config Element structure. \return -1 if PCEs are completely different, 0 if PCEs are completely equal, 1 if PCEs are different but have the same channel config, 2 if PCEs have different channel config but same number of channels. */ int CProgramConfig_Compare(const CProgramConfig *const pPce1, const CProgramConfig *const pPce2); /*! \brief Get a Program Config Element that matches the predefined MPEG-4 channel configurations 1-14. \param pPce Program Config Element structure. \param channelConfig MPEG-4 channel configuration. \return void */ void CProgramConfig_GetDefault(CProgramConfig *pPce, const UINT channelConfig); /** * \brief Lookup and verify a given element. The decoder calls this * method with every new element ID found in the bitstream. * * \param pPce A valid Program config structure. * \param chConfig MPEG-4 channel configuration. * \param tag Tag of the current element to be looked up. * \param channelIdx The current channel count of the decoder parser. * \param chMapping Array to store the canonical channel mapping indexes. * \param chType Array to store the audio channel type. * \param chIndex Array to store the individual audio channel type index. * \param chDescrLen Length of the output channel description array. * \param elMapping Pointer where the canonical element index is stored. * \param elType The element id of the current element to be looked up. * * \return Non-zero if the element belongs to the current program, * zero if it does not. */ int CProgramConfig_LookupElement(CProgramConfig *pPce, UINT chConfig, const UINT tag, const UINT channelIdx, UCHAR chMapping[], AUDIO_CHANNEL_TYPE chType[], UCHAR chIndex[], const UINT chDescrLen, UCHAR *elMapping, MP4_ELEMENT_ID elList[], MP4_ELEMENT_ID elType); /** * \brief Get table of channel indices in the order of their * appearance in by the program config field. * \param pPce A valid program config structure. * \param pceChMap Array to store the channel mapping indices like they * appear in the PCE. * \param pceChMapLen Lenght of the channel mapping index array (pceChMap). * * \return Non-zero if any error occured otherwise zero. */ int CProgramConfig_GetPceChMap(const CProgramConfig *pPce, UCHAR pceChMap[], const UINT pceChMapLen); /** * \brief Get table of elements in canonical order from a * give program config field. * \param pPce A valid program config structure. * \param table An array where the element IDs are stored. * \param elListSize The length of the table array. * \param pChMapIdx Pointer to a field receiving the corresponding * implicit channel configuration index of the given * PCE. If none can be found it receives the value 0. * \return Total element count including all SCE, CPE and LFE. */ int CProgramConfig_GetElementTable(const CProgramConfig *pPce, MP4_ELEMENT_ID table[], const INT elListSize, UCHAR *pChMapIdx); /** * \brief Get channel description (type and index) for implicit configurations (chConfig > 0) in MPEG canonical order. * \param chConfig MPEG-4 channel configuration. * \param chType Array to store the audio channel type. * \param chIndex Array to store the individual audio channel type index. * \return void */ void CProgramConfig_GetChannelDescription(const UINT chConfig, const CProgramConfig *pPce, AUDIO_CHANNEL_TYPE chType[], UCHAR chIndex[]); /** * \brief Initialize a given AudioSpecificConfig structure. * \param pAsc A pointer to an allocated CSAudioSpecificConfig struct. * \return void */ void AudioSpecificConfig_Init(CSAudioSpecificConfig *pAsc); /** * \brief Parse a AudioSpecificConfig from a given bitstream handle. * * \param pAsc A pointer to an allocated * CSAudioSpecificConfig struct. * \param hBs Bitstream handle. * \param fExplicitBackwardCompatible Do explicit backward compatibility * parsing if set (flag). * \param cb pointer to structure holding callback information * \param configMode Config modes: memory allocation mode or config change * detection mode. * \param configChanged Indicates a config change. * \param m_aot in case of unequal AOT_NULL_OBJECT only the specific config is * parsed. * * \return Total element count including all SCE, CPE and LFE. */ TRANSPORTDEC_ERROR AudioSpecificConfig_Parse( CSAudioSpecificConfig *pAsc, HANDLE_FDK_BITSTREAM hBs, int fExplicitBackwardCompatible, CSTpCallBacks *cb, UCHAR configMode, UCHAR configChanged, AUDIO_OBJECT_TYPE m_aot); /* CELP stuff */ enum { MPE = 0, RPE = 1, fs8KHz = 0, fs16KHz = 1 }; /* Defintion of flags that can be passed to transportDecOpen() */ #define TP_FLAG_MPEG4 1 /* Capability flags */ #define CAPF_TPDEC_ADIF \ 0x00001000 /**< Flag indicating support for ADIF transport format. */ #define CAPF_TPDEC_ADTS \ 0x00002000 /**< Flag indicating support for ADTS transport format. */ #define CAPF_TPDEC_LOAS \ 0x00004000 /**< Flag indicating support for LOAS transport format. */ #define CAPF_TPDEC_LATM \ 0x00008000 /**< Flag indicating support for LATM transport format. */ #define CAPF_TPDEC_RAWPACKETS \ 0x00010000 /**< Flag indicating support for raw packets transport format. */ typedef struct TRANSPORTDEC *HANDLE_TRANSPORTDEC; /** * \brief Configure Transport Decoder via a binary coded AudioSpecificConfig or * StreamMuxConfig. The previously requested configuration callback will be * called as well. The buffer conf must containt a SMC in case of * LOAS/LATM transport format, and an ASC elseways. * * \param hTp Handle of a transport decoder. * \param conf UCHAR buffer of the binary coded config (ASC or SMC). * \param length The length in bytes of the conf buffer. * * \return Error code. */ TRANSPORTDEC_ERROR transportDec_OutOfBandConfig(const HANDLE_TRANSPORTDEC hTp, UCHAR *conf, const UINT length, const UINT layer); /** * \brief Configure Transport Decoder via a binary coded USAC/RSV603DA Config. * The buffer newConfig contains a binary coded USAC/RSV603DA config of * length newConfigLength bytes. If the new config and the previous config are * different configChanged is set to 1 otherwise it is set to 0. * * \param hTp Handle of a transport decoder. * \param newConfig buffer of the binary coded config. * \param newConfigLength Length of new config in bytes. * \param buildUpStatus Indicates build up status: off|on|idle. * \param configChanged Indicates if config changed. * \param layer Instance layer. * * \return Error code. */ TRANSPORTDEC_ERROR transportDec_InBandConfig( const HANDLE_TRANSPORTDEC hTp, UCHAR *newConfig, const UINT newConfigLength, const UCHAR buildUpStatus, UCHAR *configChanged, const UINT layer, UCHAR *implicitExplicitCfgDiff); /** * \brief Open Transport medium for reading. * * \param transportDecFmt Format of the transport decoder medium to be accessed. * \param flags Transport decoder flags. Currently only TP_FLAG_MPEG4, * which signals a MPEG4 capable decoder (relevant for ADTS only). * * \return A pointer to a valid and allocated HANDLE_TRANSPORTDEC or a null * pointer on failure. */ HANDLE_TRANSPORTDEC transportDec_Open(TRANSPORT_TYPE transportDecFmt, const UINT flags, const UINT nrOfLayer); /** * \brief Register configuration change callback. * \param hTp Handle of transport decoder. * \param cbUpdateConfig Pointer to a callback function to handle audio config * changes. * \param user_data void pointer for user data passed to the callback as * first parameter. * \return 0 on success. */ int transportDec_RegisterAscCallback(HANDLE_TRANSPORTDEC hTp, const cbUpdateConfig_t cbUpdateConfig, void *user_data); /** * \brief Register free memory callback. * \param hTp Handle of transport decoder. * \param cbFreeMem Pointer to a callback function to free config dependent * memory. * \param user_data void pointer for user data passed to the callback as * first parameter. * \return 0 on success. */ int transportDec_RegisterFreeMemCallback(HANDLE_TRANSPORTDEC hTp, const cbFreeMem_t cbFreeMem, void *user_data); /** * \brief Register config change control callback. * \param hTp Handle of transport decoder. * \param cbCtrlCFGChange Pointer to a callback function for config change * control. * \param user_data void pointer for user data passed to the callback as * first parameter. * \return 0 on success. */ int transportDec_RegisterCtrlCFGChangeCallback( HANDLE_TRANSPORTDEC hTp, const cbCtrlCFGChange_t cbCtrlCFGChange, void *user_data); /** * \brief Register SSC parser callback. * \param hTp Handle of transport decoder. * \param cbUpdateConfig Pointer to a callback function to handle SSC parsing. * \param user_data void pointer for user data passed to the callback as * first parameter. * \return 0 on success. */ int transportDec_RegisterSscCallback(HANDLE_TRANSPORTDEC hTp, const cbSsc_t cbSscParse, void *user_data); /** * \brief Register SBR header parser callback. * \param hTp Handle of transport decoder. * \param cbUpdateConfig Pointer to a callback function to handle SBR header * parsing. * \param user_data void pointer for user data passed to the callback as * first parameter. * \return 0 on success. */ int transportDec_RegisterSbrCallback(HANDLE_TRANSPORTDEC hTpDec, const cbSbr_t cbSbr, void *user_data); /** * \brief Register USAC SC parser callback. * \param hTp Handle of transport decoder. * \param cbUpdateConfig Pointer to a callback function to handle USAC SC * parsing. * \param user_data void pointer for user data passed to the callback as * first parameter. * \return 0 on success. */ int transportDec_RegisterUsacCallback(HANDLE_TRANSPORTDEC hTpDec, const cbUsac_t cbUsac, void *user_data); /** * \brief Register uniDrcConfig and loudnessInfoSet parser * callback. * \param hTp Handle of transport decoder. * \param cbUpdateConfig Pointer to a callback function to handle uniDrcConfig * and loudnessInfoSet parsing. * \param user_data void pointer for user data passed to the callback as * first parameter. * \return 0 on success. */ int transportDec_RegisterUniDrcConfigCallback(HANDLE_TRANSPORTDEC hTpDec, const cbUniDrc_t cbUniDrc, void *user_data, UINT *pLoudnessInfoSetPosition); /** * \brief Fill internal input buffer with bitstream data from the external input * buffer. The function only copies such data as long as the decoder-internal * input buffer is not full. So it grabs whatever it can from pBuffer and * returns information (bytesValid) so that at a subsequent call of * %transportDec_FillData(), the right position in pBuffer can be determined to * grab the next data. * * \param hTp Handle of transportDec. * \param pBuffer Pointer to external input buffer. * \param bufferSize Size of external input buffer. This argument is required * because decoder-internally we need the information to calculate the offset to * pBuffer, where the next available data is, which is then * fed into the decoder-internal buffer (as much as * possible). Our example framework implementation fills the * buffer at pBuffer again, once it contains no available valid bytes anymore * (meaning bytesValid equal 0). * \param bytesValid Number of bitstream bytes in the external bitstream buffer * that have not yet been copied into the decoder's internal bitstream buffer by * calling this function. The value is updated according to * the amount of newly copied bytes. * \param layer The layer the bitstream belongs to. * \return Error code. */ TRANSPORTDEC_ERROR transportDec_FillData(const HANDLE_TRANSPORTDEC hTp, UCHAR *pBuffer, const UINT bufferSize, UINT *pBytesValid, const INT layer); /** * \brief Get transportDec bitstream handle. * \param hTp Pointer to a transport decoder handle. * \return HANDLE_FDK_BITSTREAM bitstream handle. */ HANDLE_FDK_BITSTREAM transportDec_GetBitstream(const HANDLE_TRANSPORTDEC hTp, const UINT layer); /** * \brief Get transport format. * \param hTp Pointer to a transport decoder handle. * \return The transport format. */ TRANSPORT_TYPE transportDec_GetFormat(const HANDLE_TRANSPORTDEC hTp); /** * \brief Get the current buffer fullness value. * * \param hTp Handle of a transport decoder. * * \return Buffer fullness */ INT transportDec_GetBufferFullness(const HANDLE_TRANSPORTDEC hTp); /** * \brief Close and deallocate transportDec. * \param phTp Pointer to a previously allocated transport decoder handle. * \return void */ void transportDec_Close(HANDLE_TRANSPORTDEC *phTp); /** * \brief Read one access unit from the transportDec medium. * \param hTp Handle of transportDec. * \param length On return, this value is overwritten with the actual access * unit length in bits. Set to -1 if length is unknown. * \return Error code. */ TRANSPORTDEC_ERROR transportDec_ReadAccessUnit(const HANDLE_TRANSPORTDEC hTp, const UINT layer); /** * \brief Get AudioSpecificConfig. * \param hTp Handle of transportDec. * \param layer Transport layer. * \param asc Pointer to AudioSpecificConfig. * \return Error code. */ TRANSPORTDEC_ERROR transportDec_GetAsc(const HANDLE_TRANSPORTDEC hTp, const UINT layer, CSAudioSpecificConfig *asc); /** * \brief Get the remaining amount of bits of the current access unit. The * result can be below zero, meaning that too many bits have been read. * \param hTp Handle of transportDec. * \return amount of remaining bits. */ INT transportDec_GetAuBitsRemaining(const HANDLE_TRANSPORTDEC hTp, const UINT layer); /** * \brief Get the total amount of bits of the current access unit. * \param hTp Handle of transportDec. * \return amount of total bits. */ INT transportDec_GetAuBitsTotal(const HANDLE_TRANSPORTDEC hTp, const UINT layer); /** * \brief This function is required to be called when the decoder has * finished parsing one Access Unit for bitstream housekeeping. * \param hTp Transport Handle. * \return Error code. */ TRANSPORTDEC_ERROR transportDec_EndAccessUnit(const HANDLE_TRANSPORTDEC hTp); /** * \brief Obtain the amount of missing access units if applicable in case * of a bit stream synchronization error. Each time * transportDec_ReadAccessUnit() returns TRANSPORTDEC_SYNC_ERROR * this function can be called to retrieve an estimate of the amount * of missing access units. This works only in case of constant * average bit rate (has to be known) and if the parameter * TPDEC_PARAM_SET_BITRATE has been set accordingly. * \param hTp Transport Handle. * \param pNAccessUnits pointer to a memory location where the estimated lost * frame count will be stored into. * \return Error code. */ TRANSPORTDEC_ERROR transportDec_GetMissingAccessUnitCount( INT *pNAccessUnits, HANDLE_TRANSPORTDEC hTp); /** * \brief Set a given setting. * \param hTp Transport Handle. * \param param Identifier of the parameter to be changed. * \param value Value for the parameter to be changed. * \return Error code. */ TRANSPORTDEC_ERROR transportDec_SetParam(const HANDLE_TRANSPORTDEC hTp, const TPDEC_PARAM param, const INT value); /** * \brief Get number of subframes (for LATM or ADTS) * \param hTp Transport Handle. * \return Number of ADTS/LATM subframes (return 1 for all other transport * types). */ UINT transportDec_GetNrOfSubFrames(HANDLE_TRANSPORTDEC hTp); /** * \brief Get info structure of transport decoder library. * \param info A pointer to an allocated LIB_INFO struct. * \return Error code. */ TRANSPORTDEC_ERROR transportDec_GetLibInfo(LIB_INFO *info); /* ADTS CRC support */ /** * \brief Set current bitstream position as start of a new data region. * \param hTp Transport handle. * \param mBits Size in bits of the data region. Set to 0 if it should not be * of a fixed size. * \return Data region ID, which should be used when calling * transportDec_CrcEndReg(). */ int transportDec_CrcStartReg(const HANDLE_TRANSPORTDEC hTp, const INT mBits); /** * \brief Set end of data region. * \param hTp Transport handle. * \param reg Data region ID, opbtained from transportDec_CrcStartReg(). * \return void */ void transportDec_CrcEndReg(const HANDLE_TRANSPORTDEC hTp, const INT reg); /** * \brief Calculate ADTS crc and check if it is correct. The ADTS checksum * is held internally. * \param hTp Transport handle. * \return Return TRANSPORTDEC_OK if the CRC is ok, or error if CRC is not * correct. */ TRANSPORTDEC_ERROR transportDec_CrcCheck(const HANDLE_TRANSPORTDEC hTp); /** * \brief Only check whether a given config seems to be valid without modifying * internal states. * * \param conf UCHAR buffer of the binary coded config (SDC type 9). * \param length The length in bytes of the conf buffer. * * \return Error code. */ TRANSPORTDEC_ERROR transportDec_DrmRawSdcAudioConfig_Check(UCHAR *conf, const UINT length); #endif /* #ifndef TPDEC_LIB_H */ fdk-aac-2.0.2/libMpegTPDec/src/000077500000000000000000000000001404225310700160255ustar00rootroot00000000000000fdk-aac-2.0.2/libMpegTPDec/src/tp_version.h000066400000000000000000000120621404225310700203670ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* MPEG transport format decoder library ********************* Author(s): Description: *******************************************************************************/ #if !defined(TP_VERSION_H) #define TP_VERSION_H /* library info */ #define TP_LIB_VL0 3 #define TP_LIB_VL1 0 #define TP_LIB_VL2 0 #define TP_LIB_TITLE "MPEG Transport" #ifdef __ANDROID__ #define TP_LIB_BUILD_DATE "" #define TP_LIB_BUILD_TIME "" #else #define TP_LIB_BUILD_DATE __DATE__ #define TP_LIB_BUILD_TIME __TIME__ #endif #endif /* !defined(TP_VERSION_H) */ fdk-aac-2.0.2/libMpegTPDec/src/tpdec_adif.cpp000066400000000000000000000144621404225310700206220ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* MPEG transport format decoder library ********************* Author(s): Josef Hoepfl Description: ADIF reader *******************************************************************************/ #include "tpdec_adif.h" #include "FDK_bitstream.h" #include "genericStds.h" TRANSPORTDEC_ERROR adifRead_DecodeHeader(CAdifHeader *pAdifHeader, CProgramConfig *pPce, HANDLE_FDK_BITSTREAM bs) { int i; TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK; UINT startAnchor = FDKgetValidBits(bs); if ((INT)startAnchor < MIN_ADIF_HEADERLENGTH) { return (TRANSPORTDEC_NOT_ENOUGH_BITS); } if (FDKreadBits(bs, 8) != 'A') { return (TRANSPORTDEC_SYNC_ERROR); } if (FDKreadBits(bs, 8) != 'D') { return (TRANSPORTDEC_SYNC_ERROR); } if (FDKreadBits(bs, 8) != 'I') { return (TRANSPORTDEC_SYNC_ERROR); } if (FDKreadBits(bs, 8) != 'F') { return (TRANSPORTDEC_SYNC_ERROR); } if ((pAdifHeader->CopyrightIdPresent = FDKreadBits(bs, 1)) != 0) FDKpushBiDirectional(bs, 72); /* CopyrightId */ pAdifHeader->OriginalCopy = FDKreadBits(bs, 1); pAdifHeader->Home = FDKreadBits(bs, 1); pAdifHeader->BitstreamType = FDKreadBits(bs, 1); /* pAdifHeader->BitRate = FDKreadBits(bs, 23); */ pAdifHeader->BitRate = FDKreadBits(bs, 16); pAdifHeader->BitRate <<= 7; pAdifHeader->BitRate |= FDKreadBits(bs, 7); pAdifHeader->NumProgramConfigElements = FDKreadBits(bs, 4) + 1; if (pAdifHeader->BitstreamType == 0) { FDKpushBiDirectional(bs, 20); /* adif_buffer_fullness */ } /* Parse all PCEs but keep only one */ for (i = 0; i < pAdifHeader->NumProgramConfigElements; i++) { CProgramConfig_Read(pPce, bs, startAnchor); } FDKbyteAlign(bs, startAnchor); return (ErrorStatus); } fdk-aac-2.0.2/libMpegTPDec/src/tpdec_adif.h000066400000000000000000000131201404225310700202550ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* MPEG transport format decoder library ********************* Author(s): Josef Hoepfl Description: ADIF reader *******************************************************************************/ #ifndef TPDEC_ADIF_H #define TPDEC_ADIF_H #include "tpdec_lib.h" #define MIN_ADIF_HEADERLENGTH 63 /* in bits */ typedef struct { INT NumProgramConfigElements; UINT BitRate; UCHAR CopyrightIdPresent; UCHAR OriginalCopy; UCHAR Home; UCHAR BitstreamType; } CAdifHeader; /** * \brief Parse a ADIF header at the given bitstream and store the parsed data * into a given CAdifHeader and CProgramConfig struct * * \param pAdifHeader pointer to a CAdifHeader structure to hold the parsed ADIF * header data. * \param pPce pointer to a CProgramConfig structure where the last PCE will * remain. * * \return TRANSPORTDEC_ERROR error code */ TRANSPORTDEC_ERROR adifRead_DecodeHeader(CAdifHeader *pAdifHeader, CProgramConfig *pPce, HANDLE_FDK_BITSTREAM bs); #endif /* TPDEC_ADIF_H */ fdk-aac-2.0.2/libMpegTPDec/src/tpdec_adts.cpp000066400000000000000000000367561404225310700206640ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* MPEG transport format decoder library ********************* Author(s): Josef Hoepfl Description: ADTS interface *******************************************************************************/ #include "tpdec_adts.h" #include "FDK_bitstream.h" void adtsRead_CrcInit( HANDLE_ADTS pAdts) /*!< pointer to adts crc info stucture */ { FDKcrcInit(&pAdts->crcInfo, 0x8005, 0xFFFF, 16); } int adtsRead_CrcStartReg( HANDLE_ADTS pAdts, /*!< pointer to adts stucture */ HANDLE_FDK_BITSTREAM hBs, /*!< handle to current bit buffer structure */ int mBits /*!< number of bits in crc region */ ) { if (pAdts->bs.protection_absent) { return 0; } return (FDKcrcStartReg(&pAdts->crcInfo, hBs, mBits)); } void adtsRead_CrcEndReg( HANDLE_ADTS pAdts, /*!< pointer to adts crc info stucture */ HANDLE_FDK_BITSTREAM hBs, /*!< handle to current bit buffer structure */ int reg /*!< crc region */ ) { if (pAdts->bs.protection_absent == 0) { FDKcrcEndReg(&pAdts->crcInfo, hBs, reg); } } TRANSPORTDEC_ERROR adtsRead_CrcCheck(HANDLE_ADTS pAdts) { TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK; USHORT crc; if (pAdts->bs.protection_absent) return TRANSPORTDEC_OK; crc = FDKcrcGetCRC(&pAdts->crcInfo); if (crc != pAdts->crcReadValue) { return (TRANSPORTDEC_CRC_ERROR); } return (ErrorStatus); } #define Adts_Length_SyncWord 12 #define Adts_Length_Id 1 #define Adts_Length_Layer 2 #define Adts_Length_ProtectionAbsent 1 #define Adts_Length_Profile 2 #define Adts_Length_SamplingFrequencyIndex 4 #define Adts_Length_PrivateBit 1 #define Adts_Length_ChannelConfiguration 3 #define Adts_Length_OriginalCopy 1 #define Adts_Length_Home 1 #define Adts_Length_CopyrightIdentificationBit 1 #define Adts_Length_CopyrightIdentificationStart 1 #define Adts_Length_FrameLength 13 #define Adts_Length_BufferFullness 11 #define Adts_Length_NumberOfRawDataBlocksInFrame 2 #define Adts_Length_CrcCheck 16 TRANSPORTDEC_ERROR adtsRead_DecodeHeader(HANDLE_ADTS pAdts, CSAudioSpecificConfig *pAsc, HANDLE_FDK_BITSTREAM hBs, const INT ignoreBufferFullness) { INT crcReg; INT valBits; INT cmp_buffer_fullness; int i, adtsHeaderLength; STRUCT_ADTS_BS bs; CProgramConfig oldPce; /* Store the old PCE temporarily. Maybe we'll need it later if we have channelConfig=0 and no PCE in this frame. */ FDKmemcpy(&oldPce, &pAsc->m_progrConfigElement, sizeof(CProgramConfig)); valBits = FDKgetValidBits(hBs) + ADTS_SYNCLENGTH; if (valBits < ADTS_HEADERLENGTH) { return TRANSPORTDEC_NOT_ENOUGH_BITS; } /* adts_fixed_header */ bs.mpeg_id = FDKreadBits(hBs, Adts_Length_Id); bs.layer = FDKreadBits(hBs, Adts_Length_Layer); bs.protection_absent = FDKreadBits(hBs, Adts_Length_ProtectionAbsent); bs.profile = FDKreadBits(hBs, Adts_Length_Profile); bs.sample_freq_index = FDKreadBits(hBs, Adts_Length_SamplingFrequencyIndex); bs.private_bit = FDKreadBits(hBs, Adts_Length_PrivateBit); bs.channel_config = FDKreadBits(hBs, Adts_Length_ChannelConfiguration); bs.original = FDKreadBits(hBs, Adts_Length_OriginalCopy); bs.home = FDKreadBits(hBs, Adts_Length_Home); /* adts_variable_header */ bs.copyright_id = FDKreadBits(hBs, Adts_Length_CopyrightIdentificationBit); bs.copyright_start = FDKreadBits(hBs, Adts_Length_CopyrightIdentificationStart); bs.frame_length = FDKreadBits(hBs, Adts_Length_FrameLength); bs.adts_fullness = FDKreadBits(hBs, Adts_Length_BufferFullness); bs.num_raw_blocks = FDKreadBits(hBs, Adts_Length_NumberOfRawDataBlocksInFrame); bs.num_pce_bits = 0; adtsHeaderLength = ADTS_HEADERLENGTH; if (valBits < bs.frame_length * 8) { goto bail; } FDKcrcReset(&pAdts->crcInfo); if (!bs.protection_absent) { FDKpushBack(hBs, 56); /* complete fixed and variable header! */ crcReg = FDKcrcStartReg(&pAdts->crcInfo, hBs, 0); FDKpushFor(hBs, 56); } if (!bs.protection_absent && bs.num_raw_blocks > 0) { if ((INT)FDKgetValidBits(hBs) < bs.num_raw_blocks * 16) { goto bail; } for (i = 0; i < bs.num_raw_blocks; i++) { pAdts->rawDataBlockDist[i] = (USHORT)FDKreadBits(hBs, 16); adtsHeaderLength += 16; } /* Change raw data blocks to delta values */ pAdts->rawDataBlockDist[bs.num_raw_blocks] = bs.frame_length - 7 - bs.num_raw_blocks * 2 - 2; for (i = bs.num_raw_blocks; i > 0; i--) { pAdts->rawDataBlockDist[i] -= pAdts->rawDataBlockDist[i - 1]; } } /* adts_error_check */ if (!bs.protection_absent) { USHORT crc_check; FDKcrcEndReg(&pAdts->crcInfo, hBs, crcReg); if ((INT)FDKgetValidBits(hBs) < Adts_Length_CrcCheck) { goto bail; } crc_check = FDKreadBits(hBs, Adts_Length_CrcCheck); adtsHeaderLength += Adts_Length_CrcCheck; pAdts->crcReadValue = crc_check; /* Check header CRC in case of multiple raw data blocks */ if (bs.num_raw_blocks > 0) { if (pAdts->crcReadValue != FDKcrcGetCRC(&pAdts->crcInfo)) { return TRANSPORTDEC_CRC_ERROR; } /* Reset CRC for the upcoming raw_data_block() */ FDKcrcReset(&pAdts->crcInfo); } } /* check if valid header */ if ((bs.layer != 0) || // we only support MPEG ADTS (bs.sample_freq_index >= 13) // we only support 96kHz - 7350kHz ) { FDKpushFor(hBs, bs.frame_length * 8); // try again one frame later return TRANSPORTDEC_UNSUPPORTED_FORMAT; } /* special treatment of id-bit */ if ((bs.mpeg_id == 0) && (pAdts->decoderCanDoMpeg4 == 0)) { /* MPEG-2 decoder cannot play MPEG-4 bitstreams */ FDKpushFor(hBs, bs.frame_length * 8); // try again one frame later return TRANSPORTDEC_UNSUPPORTED_FORMAT; } if (!ignoreBufferFullness) { cmp_buffer_fullness = bs.frame_length * 8 + bs.adts_fullness * 32 * getNumberOfEffectiveChannels(bs.channel_config); /* Evaluate buffer fullness */ if (bs.adts_fullness != 0x7FF) { if (pAdts->BufferFullnesStartFlag) { if (valBits < cmp_buffer_fullness) { /* Condition for start of decoding is not fulfilled */ /* The current frame will not be decoded */ FDKpushBack(hBs, adtsHeaderLength); if ((cmp_buffer_fullness + adtsHeaderLength) > (((8192 * 4) << 3) - 7)) { return TRANSPORTDEC_SYNC_ERROR; } else { return TRANSPORTDEC_NOT_ENOUGH_BITS; } } else { pAdts->BufferFullnesStartFlag = 0; } } } } /* Get info from ADTS header */ AudioSpecificConfig_Init(pAsc); pAsc->m_aot = (AUDIO_OBJECT_TYPE)(bs.profile + 1); pAsc->m_samplingFrequencyIndex = bs.sample_freq_index; pAsc->m_samplingFrequency = SamplingRateTable[bs.sample_freq_index]; pAsc->m_channelConfiguration = bs.channel_config; pAsc->m_samplesPerFrame = 1024; if (bs.channel_config == 0) { int pceBits = 0; UINT alignAnchor = FDKgetValidBits(hBs); CProgramConfig tmpPce; if (FDKreadBits(hBs, 3) == ID_PCE) { /* Got luck! Parse the PCE */ crcReg = adtsRead_CrcStartReg(pAdts, hBs, 0); CProgramConfig_Init(&tmpPce); CProgramConfig_Read(&tmpPce, hBs, alignAnchor); if (CProgramConfig_IsValid(&tmpPce)) { if (CProgramConfig_IsValid(&oldPce)) { /* Compare the new and the old PCE (tags ignored) */ switch (CProgramConfig_Compare(&tmpPce, &oldPce)) { case 0: /* Nothing to do because PCE matches the old one exactly. */ case 1: /* Channel configuration not changed. Just new metadata. */ FDKmemcpy(&pAsc->m_progrConfigElement, &tmpPce, sizeof(CProgramConfig)); break; case 2: /* The number of channels are identical but not the config */ case -1: /* The channel configuration is completely different */ default: FDKmemcpy(&pAsc->m_progrConfigElement, &oldPce, sizeof(CProgramConfig)); FDKpushBack(hBs, adtsHeaderLength); return TRANSPORTDEC_PARSE_ERROR; } } else { FDKmemcpy(&pAsc->m_progrConfigElement, &tmpPce, sizeof(CProgramConfig)); } } else { if (CProgramConfig_IsValid(&oldPce)) { FDKmemcpy(&pAsc->m_progrConfigElement, &oldPce, sizeof(CProgramConfig)); } else { FDKpushBack(hBs, adtsHeaderLength); return TRANSPORTDEC_PARSE_ERROR; } } adtsRead_CrcEndReg(pAdts, hBs, crcReg); pceBits = (INT)alignAnchor - (INT)FDKgetValidBits(hBs); adtsHeaderLength += pceBits; if (pceBits > (INT)alignAnchor) { goto bail; } /* store the number of PCE bits */ bs.num_pce_bits = pceBits; } else { /* No PCE in this frame! Push back the ID tag bits. */ FDKpushBack(hBs, 3); /* Encoders do not have to write a PCE in each frame. So if we already have a valid PCE we have to use it. */ if (oldPce.isValid && (bs.sample_freq_index == pAdts->bs.sample_freq_index) /* we could compare the complete fixed header (bytes) here! */ && (bs.channel_config == pAdts->bs.channel_config) /* == 0 */ && (bs.mpeg_id == pAdts->bs.mpeg_id)) { /* Restore previous PCE which is still valid */ FDKmemcpy(&pAsc->m_progrConfigElement, &oldPce, sizeof(CProgramConfig)); } else if (bs.mpeg_id == 0) { /* If not it seems that we have a implicit channel configuration. This mode is not allowed in the context of ISO/IEC 14496-3. Skip this frame and try the next one. */ FDKpushFor(hBs, (bs.frame_length << 3) - adtsHeaderLength - 3); return TRANSPORTDEC_UNSUPPORTED_FORMAT; } /* else { ISO/IEC 13818-7 implicit channel mapping is allowed. So just open the box of chocolates to see what we got. } */ } } /* Copy bit stream data struct to persistent memory now, once we passed all * sanity checks above. */ FDKmemcpy(&pAdts->bs, &bs, sizeof(STRUCT_ADTS_BS)); return TRANSPORTDEC_OK; bail: FDKpushBack(hBs, adtsHeaderLength); return TRANSPORTDEC_NOT_ENOUGH_BITS; } int adtsRead_GetRawDataBlockLength(HANDLE_ADTS pAdts, INT blockNum) { int length; if (pAdts->bs.num_raw_blocks == 0) { length = (pAdts->bs.frame_length - 7) << 3; /* aac_frame_length subtracted by the header size (7 bytes). */ if (pAdts->bs.protection_absent == 0) length -= 16; /* substract 16 bit CRC */ } else { if (pAdts->bs.protection_absent) { length = -1; /* raw data block length is unknown */ } else { if (blockNum < 0 || blockNum > 3) { length = -1; } else { length = (pAdts->rawDataBlockDist[blockNum] << 3) - 16; } } } if (blockNum == 0 && length > 0) { length -= pAdts->bs.num_pce_bits; } return length; } fdk-aac-2.0.2/libMpegTPDec/src/tpdec_adts.h000066400000000000000000000212101404225310700203040ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* MPEG transport format decoder library ********************* Author(s): Josef Hoepfl Description: ADTS interface *******************************************************************************/ #ifndef TPDEC_ADTS_H #define TPDEC_ADTS_H #include "tpdec_lib.h" #define ADTS_SYNCWORD (0xfff) #define ADTS_SYNCLENGTH (12) /* in bits */ #define ADTS_HEADERLENGTH (56) /* minimum header size in bits */ #define ADTS_FIXED_HEADERLENGTH (28) /* in bits */ #define ADTS_VARIABLE_HEADERLENGTH (ADTS_HEADERLENGTH - ADTS_FIXED_HEADERLENGTH) #ifdef CHECK_TWO_SYNCS #define ADTS_MIN_TP_BUF_SIZE (8191 + 2) #else #define ADTS_MIN_TP_BUF_SIZE (8191) #endif #include "FDK_crc.h" typedef struct { /* ADTS header fields */ UCHAR mpeg_id; UCHAR layer; UCHAR protection_absent; UCHAR profile; UCHAR sample_freq_index; UCHAR private_bit; UCHAR channel_config; UCHAR original; UCHAR home; UCHAR copyright_id; UCHAR copyright_start; USHORT frame_length; USHORT adts_fullness; UCHAR num_raw_blocks; UCHAR num_pce_bits; } STRUCT_ADTS_BS; struct STRUCT_ADTS { STRUCT_ADTS_BS bs; UCHAR decoderCanDoMpeg4; UCHAR BufferFullnesStartFlag; FDK_CRCINFO crcInfo; /* CRC state info */ USHORT crcReadValue; /* CRC value read from bitstream data */ USHORT rawDataBlockDist[4]; /* distance between each raw data block. Not the same as found in the bitstream */ }; typedef struct STRUCT_ADTS *HANDLE_ADTS; /*! \brief Initialize ADTS CRC The function initialzes the crc buffer and the crc lookup table. \return none */ void adtsRead_CrcInit(HANDLE_ADTS pAdts); /** * \brief Starts CRC region with a maximum number of bits * If mBits is positive zero padding will be used for CRC calculation, if * there are less than mBits bits available. If mBits is negative no zero * padding is done. If mBits is zero the memory for the buffer is * allocated dynamically, the number of bits is not limited. * * \param pAdts ADTS data handle * \param hBs bitstream handle, on which the CRC region referes to * \param mBits max number of bits in crc region to be considered * * \return ID for the created region, -1 in case of an error */ int adtsRead_CrcStartReg(HANDLE_ADTS pAdts, HANDLE_FDK_BITSTREAM hBs, int mBits); /** * \brief Ends CRC region identified by reg * * \param pAdts ADTS data handle * \param hBs bitstream handle, on which the CRC region referes to * \param reg CRC regions ID returned by adtsRead_CrcStartReg() * * \return none */ void adtsRead_CrcEndReg(HANDLE_ADTS pAdts, HANDLE_FDK_BITSTREAM hBs, int reg); /** * \brief Check CRC * * Checks if the currently calculated CRC matches the CRC field read from the * bitstream Deletes all CRC regions. * * \param pAdts ADTS data handle * * \return Returns 0 if they are identical otherwise 1 */ TRANSPORTDEC_ERROR adtsRead_CrcCheck(HANDLE_ADTS pAdts); /** * \brief Check if we have a valid ADTS frame at the current bitbuffer position * * This function assumes enough bits in buffer for the current frame. * It reads out the header bits to prepare the bitbuffer for the decode loop. * In case the header bits show an invalid bitstream/frame, the whole frame is * skipped. * * \param pAdts ADTS data handle which is filled with parsed ADTS header data * \param bs handle of bitstream from whom the ADTS header is read * * \return error status */ TRANSPORTDEC_ERROR adtsRead_DecodeHeader(HANDLE_ADTS pAdts, CSAudioSpecificConfig *pAsc, HANDLE_FDK_BITSTREAM bs, const INT ignoreBufferFullness); /** * \brief Get the raw data block length of the given block number. * * \param pAdts ADTS data handle * \param blockNum current raw data block index * \param pLength pointer to an INT where the length of the given raw data block * is stored into the returned value might be -1, in which case the raw data * block length is unknown. * * \return error status */ int adtsRead_GetRawDataBlockLength(HANDLE_ADTS pAdts, INT blockNum); #endif /* TPDEC_ADTS_H */ fdk-aac-2.0.2/libMpegTPDec/src/tpdec_asc.cpp000066400000000000000000002703141404225310700204650ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* MPEG transport format decoder library ********************* Author(s): Daniel Homm Description: *******************************************************************************/ #include "tpdec_lib.h" #include "tp_data.h" #include "FDK_crc.h" #include "common_fix.h" /** * The following arrays provide the IDs of the consecutive elements for each * channel configuration. Every channel_configuration has to be finalized with * ID_NONE. */ static const MP4_ELEMENT_ID channel_configuration_0[] = {ID_NONE}; static const MP4_ELEMENT_ID channel_configuration_1[] = {ID_SCE, ID_NONE}; static const MP4_ELEMENT_ID channel_configuration_2[] = {ID_CPE, ID_NONE}; static const MP4_ELEMENT_ID channel_configuration_3[] = {ID_SCE, ID_CPE, ID_NONE}; static const MP4_ELEMENT_ID channel_configuration_4[] = {ID_SCE, ID_CPE, ID_SCE, ID_NONE}; static const MP4_ELEMENT_ID channel_configuration_5[] = {ID_SCE, ID_CPE, ID_CPE, ID_NONE}; static const MP4_ELEMENT_ID channel_configuration_6[] = {ID_SCE, ID_CPE, ID_CPE, ID_LFE, ID_NONE}; static const MP4_ELEMENT_ID channel_configuration_7[] = { ID_SCE, ID_CPE, ID_CPE, ID_CPE, ID_LFE, ID_NONE}; static const MP4_ELEMENT_ID channel_configuration_8[] = { ID_NONE}; /* reserved */ static const MP4_ELEMENT_ID channel_configuration_9[] = { ID_NONE}; /* reserved */ static const MP4_ELEMENT_ID channel_configuration_10[] = { ID_NONE}; /* reserved */ static const MP4_ELEMENT_ID channel_configuration_11[] = { ID_SCE, ID_CPE, ID_CPE, ID_SCE, ID_LFE, ID_NONE}; static const MP4_ELEMENT_ID channel_configuration_12[] = { ID_SCE, ID_CPE, ID_CPE, ID_CPE, ID_LFE, ID_NONE}; static const MP4_ELEMENT_ID channel_configuration_13[] = { ID_SCE, ID_CPE, ID_CPE, ID_CPE, ID_CPE, ID_SCE, ID_LFE, ID_LFE, ID_SCE, ID_CPE, ID_CPE, ID_SCE, ID_CPE, ID_SCE, ID_SCE, ID_CPE, ID_NONE}; static const MP4_ELEMENT_ID channel_configuration_14[] = { ID_SCE, ID_CPE, ID_CPE, ID_LFE, ID_CPE, ID_NONE}; static const MP4_ELEMENT_ID *channel_configuration_array[] = { channel_configuration_0, channel_configuration_1, channel_configuration_2, channel_configuration_3, channel_configuration_4, channel_configuration_5, channel_configuration_6, channel_configuration_7, channel_configuration_8, channel_configuration_9, channel_configuration_10, channel_configuration_11, channel_configuration_12, channel_configuration_13, channel_configuration_14}; #define TP_USAC_MAX_CHANNEL_CONFIGURATION_INDEX (13) #define SC_CHANNEL_CONFIG_TAB_SIZE (TP_USAC_MAX_CHANNEL_CONFIGURATION_INDEX + 1) /* channel config structure used for sanity check */ typedef struct { SCHAR nCh; /* number of channels */ SCHAR nSCE; /* number of SCE's */ SCHAR nCPE; /* number of CPE's */ SCHAR nLFE; /* number of LFE's */ } SC_CHANNEL_CONFIG; static const SC_CHANNEL_CONFIG sc_chan_config_tab[SC_CHANNEL_CONFIG_TAB_SIZE] = { /* nCh, nSCE, nCPE, nLFE, cci */ {0, 0, 0, 0}, /* 0 */ {1, 1, 0, 0}, /* 1 */ {2, 0, 1, 0}, /* 2 */ {3, 1, 1, 0}, /* 3 */ {4, 2, 1, 0}, /* 4 */ {5, 1, 2, 0}, /* 5 */ {6, 1, 2, 1}, /* 6 */ {8, 1, 3, 1}, /* 7 */ {2, 2, 0, 0}, /* 8 */ {3, 1, 1, 0}, /* 9 */ {4, 0, 2, 0}, /* 10 */ {7, 2, 2, 1}, /* 11 */ {8, 1, 3, 1}, /* 12 */ {24, 6, 8, 2} /* 13 */ }; void CProgramConfig_Reset(CProgramConfig *pPce) { pPce->elCounter = 0; } void CProgramConfig_Init(CProgramConfig *pPce) { FDKmemclear(pPce, sizeof(CProgramConfig)); pPce->SamplingFrequencyIndex = 0xf; } int CProgramConfig_IsValid(const CProgramConfig *pPce) { return ((pPce->isValid) ? 1 : 0); } #define PCE_HEIGHT_EXT_SYNC (0xAC) /* * Read the extension for height info. * return 0 if successfull, * -1 if the CRC failed, * -2 if invalid HeightInfo. */ static int CProgramConfig_ReadHeightExt(CProgramConfig *pPce, HANDLE_FDK_BITSTREAM bs, int *const bytesAvailable, const UINT alignmentAnchor) { int err = 0; FDK_CRCINFO crcInfo; /* CRC state info */ INT crcReg; FDKcrcInit(&crcInfo, 0x07, 0xFF, 8); crcReg = FDKcrcStartReg(&crcInfo, bs, 0); UINT startAnchor = FDKgetValidBits(bs); FDK_ASSERT(pPce != NULL); FDK_ASSERT(bs != NULL); FDK_ASSERT(bytesAvailable != NULL); if ((startAnchor >= 24) && (*bytesAvailable >= 3) && (FDKreadBits(bs, 8) == PCE_HEIGHT_EXT_SYNC)) { int i; for (i = 0; i < pPce->NumFrontChannelElements; i++) { if ((pPce->FrontElementHeightInfo[i] = (UCHAR)FDKreadBits(bs, 2)) >= PC_NUM_HEIGHT_LAYER) { err = -2; /* height information is out of the valid range */ } } for (i = 0; i < pPce->NumSideChannelElements; i++) { if ((pPce->SideElementHeightInfo[i] = (UCHAR)FDKreadBits(bs, 2)) >= PC_NUM_HEIGHT_LAYER) { err = -2; /* height information is out of the valid range */ } } for (i = 0; i < pPce->NumBackChannelElements; i++) { if ((pPce->BackElementHeightInfo[i] = (UCHAR)FDKreadBits(bs, 2)) >= PC_NUM_HEIGHT_LAYER) { err = -2; /* height information is out of the valid range */ } } FDKbyteAlign(bs, alignmentAnchor); FDKcrcEndReg(&crcInfo, bs, crcReg); if ((USHORT)FDKreadBits(bs, 8) != FDKcrcGetCRC(&crcInfo)) { /* CRC failed */ err = -1; } if (err != 0) { /* Reset whole height information in case an error occured during parsing. The return value ensures that pPce->isValid is set to 0 and implicit channel mapping is used. */ FDKmemclear(pPce->FrontElementHeightInfo, sizeof(pPce->FrontElementHeightInfo)); FDKmemclear(pPce->SideElementHeightInfo, sizeof(pPce->SideElementHeightInfo)); FDKmemclear(pPce->BackElementHeightInfo, sizeof(pPce->BackElementHeightInfo)); } } else { /* No valid extension data found -> restore the initial bitbuffer state */ FDKpushBack(bs, (INT)startAnchor - (INT)FDKgetValidBits(bs)); } /* Always report the bytes read. */ *bytesAvailable -= ((INT)startAnchor - (INT)FDKgetValidBits(bs)) >> 3; return (err); } void CProgramConfig_Read(CProgramConfig *pPce, HANDLE_FDK_BITSTREAM bs, UINT alignmentAnchor) { int i, err = 0; int commentBytes; pPce->NumEffectiveChannels = 0; pPce->NumChannels = 0; pPce->ElementInstanceTag = (UCHAR)FDKreadBits(bs, 4); pPce->Profile = (UCHAR)FDKreadBits(bs, 2); pPce->SamplingFrequencyIndex = (UCHAR)FDKreadBits(bs, 4); pPce->NumFrontChannelElements = (UCHAR)FDKreadBits(bs, 4); pPce->NumSideChannelElements = (UCHAR)FDKreadBits(bs, 4); pPce->NumBackChannelElements = (UCHAR)FDKreadBits(bs, 4); pPce->NumLfeChannelElements = (UCHAR)FDKreadBits(bs, 2); pPce->NumAssocDataElements = (UCHAR)FDKreadBits(bs, 3); pPce->NumValidCcElements = (UCHAR)FDKreadBits(bs, 4); if ((pPce->MonoMixdownPresent = (UCHAR)FDKreadBits(bs, 1)) != 0) { pPce->MonoMixdownElementNumber = (UCHAR)FDKreadBits(bs, 4); } if ((pPce->StereoMixdownPresent = (UCHAR)FDKreadBits(bs, 1)) != 0) { pPce->StereoMixdownElementNumber = (UCHAR)FDKreadBits(bs, 4); } if ((pPce->MatrixMixdownIndexPresent = (UCHAR)FDKreadBits(bs, 1)) != 0) { pPce->MatrixMixdownIndex = (UCHAR)FDKreadBits(bs, 2); pPce->PseudoSurroundEnable = (UCHAR)FDKreadBits(bs, 1); } for (i = 0; i < pPce->NumFrontChannelElements; i++) { pPce->FrontElementIsCpe[i] = (UCHAR)FDKreadBits(bs, 1); pPce->FrontElementTagSelect[i] = (UCHAR)FDKreadBits(bs, 4); pPce->NumChannels += pPce->FrontElementIsCpe[i] ? 2 : 1; } for (i = 0; i < pPce->NumSideChannelElements; i++) { pPce->SideElementIsCpe[i] = (UCHAR)FDKreadBits(bs, 1); pPce->SideElementTagSelect[i] = (UCHAR)FDKreadBits(bs, 4); pPce->NumChannels += pPce->SideElementIsCpe[i] ? 2 : 1; } for (i = 0; i < pPce->NumBackChannelElements; i++) { pPce->BackElementIsCpe[i] = (UCHAR)FDKreadBits(bs, 1); pPce->BackElementTagSelect[i] = (UCHAR)FDKreadBits(bs, 4); pPce->NumChannels += pPce->BackElementIsCpe[i] ? 2 : 1; } pPce->NumEffectiveChannels = pPce->NumChannels; for (i = 0; i < pPce->NumLfeChannelElements; i++) { pPce->LfeElementTagSelect[i] = (UCHAR)FDKreadBits(bs, 4); pPce->NumChannels += 1; } for (i = 0; i < pPce->NumAssocDataElements; i++) { pPce->AssocDataElementTagSelect[i] = (UCHAR)FDKreadBits(bs, 4); } for (i = 0; i < pPce->NumValidCcElements; i++) { pPce->CcElementIsIndSw[i] = (UCHAR)FDKreadBits(bs, 1); pPce->ValidCcElementTagSelect[i] = (UCHAR)FDKreadBits(bs, 4); } FDKbyteAlign(bs, alignmentAnchor); pPce->CommentFieldBytes = (UCHAR)FDKreadBits(bs, 8); commentBytes = pPce->CommentFieldBytes; /* Search for height info extension and read it if available */ err = CProgramConfig_ReadHeightExt(pPce, bs, &commentBytes, alignmentAnchor); for (i = 0; i < commentBytes; i++) { UCHAR text; text = (UCHAR)FDKreadBits(bs, 8); if (i < PC_COMMENTLENGTH) { pPce->Comment[i] = text; } } pPce->isValid = (err) ? 0 : 1; } /* * Compare two program configurations. * Returns the result of the comparison: * -1 - completely different * 0 - completely equal * 1 - different but same channel configuration * 2 - different channel configuration but same number of channels */ int CProgramConfig_Compare(const CProgramConfig *const pPce1, const CProgramConfig *const pPce2) { int result = 0; /* Innocent until proven false. */ if (FDKmemcmp(pPce1, pPce2, sizeof(CProgramConfig)) != 0) { /* Configurations are not completely equal. So look into details and analyse the channel configurations: */ result = -1; if (pPce1->NumChannels == pPce2->NumChannels) { /* Now the logic changes. We first assume to have the same channel configuration and then prove if this assumption is true. */ result = 1; /* Front channels */ if (pPce1->NumFrontChannelElements != pPce2->NumFrontChannelElements) { result = 2; /* different number of front channel elements */ } else { int el, numCh1 = 0, numCh2 = 0; for (el = 0; el < pPce1->NumFrontChannelElements; el += 1) { if (pPce1->FrontElementHeightInfo[el] != pPce2->FrontElementHeightInfo[el]) { result = 2; /* different height info */ break; } numCh1 += pPce1->FrontElementIsCpe[el] ? 2 : 1; numCh2 += pPce2->FrontElementIsCpe[el] ? 2 : 1; } if (numCh1 != numCh2) { result = 2; /* different number of front channels */ } } /* Side channels */ if (pPce1->NumSideChannelElements != pPce2->NumSideChannelElements) { result = 2; /* different number of side channel elements */ } else { int el, numCh1 = 0, numCh2 = 0; for (el = 0; el < pPce1->NumSideChannelElements; el += 1) { if (pPce1->SideElementHeightInfo[el] != pPce2->SideElementHeightInfo[el]) { result = 2; /* different height info */ break; } numCh1 += pPce1->SideElementIsCpe[el] ? 2 : 1; numCh2 += pPce2->SideElementIsCpe[el] ? 2 : 1; } if (numCh1 != numCh2) { result = 2; /* different number of side channels */ } } /* Back channels */ if (pPce1->NumBackChannelElements != pPce2->NumBackChannelElements) { result = 2; /* different number of back channel elements */ } else { int el, numCh1 = 0, numCh2 = 0; for (el = 0; el < pPce1->NumBackChannelElements; el += 1) { if (pPce1->BackElementHeightInfo[el] != pPce2->BackElementHeightInfo[el]) { result = 2; /* different height info */ break; } numCh1 += pPce1->BackElementIsCpe[el] ? 2 : 1; numCh2 += pPce2->BackElementIsCpe[el] ? 2 : 1; } if (numCh1 != numCh2) { result = 2; /* different number of back channels */ } } /* LFE channels */ if (pPce1->NumLfeChannelElements != pPce2->NumLfeChannelElements) { result = 2; /* different number of lfe channels */ } /* LFEs are always SCEs so we don't need to count the channels. */ } } return result; } void CProgramConfig_GetDefault(CProgramConfig *pPce, const UINT channelConfig) { FDK_ASSERT(pPce != NULL); /* Init PCE */ CProgramConfig_Init(pPce); pPce->Profile = 1; /* Set AAC LC because it is the only supported object type. */ switch (channelConfig) { /* - - - - - - - - - - - - - - - - - - - - - - - - - - - - */ case 32: /* 7.1 side channel configuration as defined in FDK_audio.h */ pPce->NumFrontChannelElements = 2; pPce->FrontElementIsCpe[0] = 0; pPce->FrontElementIsCpe[1] = 1; pPce->NumSideChannelElements = 1; pPce->SideElementIsCpe[0] = 1; pPce->NumBackChannelElements = 1; pPce->BackElementIsCpe[0] = 1; pPce->NumLfeChannelElements = 1; pPce->NumChannels = 8; pPce->NumEffectiveChannels = 7; pPce->isValid = 1; break; /* - - - - - - - - - - - - - - - - - - - - - - - - - - - - */ case 12: /* 3/0/4.1ch surround back */ pPce->BackElementIsCpe[1] = 1; pPce->NumChannels += 1; pPce->NumEffectiveChannels += 1; FDK_FALLTHROUGH; case 11: /* 3/0/3.1ch */ pPce->NumFrontChannelElements += 2; pPce->FrontElementIsCpe[0] = 0; pPce->FrontElementIsCpe[1] = 1; pPce->NumBackChannelElements += 2; pPce->BackElementIsCpe[0] = 1; pPce->BackElementIsCpe[1] += 0; pPce->NumLfeChannelElements += 1; pPce->NumChannels += 7; pPce->NumEffectiveChannels += 6; pPce->isValid = 1; break; /* - - - - - - - - - - - - - - - - - - - - - - - - - - - - */ case 14: /* 2/0/0-3/0/2-0.1ch front height */ pPce->FrontElementHeightInfo[2] = 1; /* Top speaker */ FDK_FALLTHROUGH; case 7: /* 5/0/2.1ch front */ pPce->NumFrontChannelElements += 1; pPce->FrontElementIsCpe[2] = 1; pPce->NumChannels += 2; pPce->NumEffectiveChannels += 2; FDK_FALLTHROUGH; case 6: /* 3/0/2.1ch */ pPce->NumLfeChannelElements += 1; pPce->NumChannels += 1; FDK_FALLTHROUGH; case 5: /* 3/0/2.0ch */ case 4: /* 3/0/1.0ch */ pPce->NumBackChannelElements += 1; pPce->BackElementIsCpe[0] = (channelConfig > 4) ? 1 : 0; pPce->NumChannels += (channelConfig > 4) ? 2 : 1; pPce->NumEffectiveChannels += (channelConfig > 4) ? 2 : 1; FDK_FALLTHROUGH; case 3: /* 3/0/0.0ch */ pPce->NumFrontChannelElements += 1; pPce->FrontElementIsCpe[1] = 1; pPce->NumChannels += 2; pPce->NumEffectiveChannels += 2; FDK_FALLTHROUGH; case 1: /* 1/0/0.0ch */ pPce->NumFrontChannelElements += 1; pPce->FrontElementIsCpe[0] = 0; pPce->NumChannels += 1; pPce->NumEffectiveChannels += 1; pPce->isValid = 1; break; /* - - - - - - - - - - - - - - - - - - - - - - - - - - - - */ case 2: /* 2/0/0.ch */ pPce->NumFrontChannelElements = 1; pPce->FrontElementIsCpe[0] = 1; pPce->NumChannels += 2; pPce->NumEffectiveChannels += 2; pPce->isValid = 1; break; /* - - - - - - - - - - - - - - - - - - - - - - - - - - - - */ default: pPce->isValid = 0; /* To be explicit! */ break; } if (pPce->isValid) { /* Create valid element instance tags */ int el, elTagSce = 0, elTagCpe = 0; for (el = 0; el < pPce->NumFrontChannelElements; el += 1) { pPce->FrontElementTagSelect[el] = (pPce->FrontElementIsCpe[el]) ? elTagCpe++ : elTagSce++; } for (el = 0; el < pPce->NumSideChannelElements; el += 1) { pPce->SideElementTagSelect[el] = (pPce->SideElementIsCpe[el]) ? elTagCpe++ : elTagSce++; } for (el = 0; el < pPce->NumBackChannelElements; el += 1) { pPce->BackElementTagSelect[el] = (pPce->BackElementIsCpe[el]) ? elTagCpe++ : elTagSce++; } elTagSce = 0; for (el = 0; el < pPce->NumLfeChannelElements; el += 1) { pPce->LfeElementTagSelect[el] = elTagSce++; } } } /** * \brief get implicit audio channel type for given channelConfig and MPEG * ordered channel index * \param channelConfig MPEG channelConfiguration from 1 upto 14 * \param index MPEG channel order index * \return audio channel type. */ static void getImplicitAudioChannelTypeAndIndex(AUDIO_CHANNEL_TYPE *chType, UCHAR *chIndex, UINT channelConfig, UINT index) { if (index < 3) { *chType = ACT_FRONT; *chIndex = index; } else { switch (channelConfig) { case 4: /* SCE, CPE, SCE */ case 5: /* SCE, CPE, CPE */ case 6: /* SCE, CPE, CPE, LFE */ switch (index) { case 3: case 4: *chType = ACT_BACK; *chIndex = index - 3; break; case 5: *chType = ACT_LFE; *chIndex = 0; break; } break; case 7: /* SCE,CPE,CPE,CPE,LFE */ switch (index) { case 3: case 4: *chType = ACT_FRONT; *chIndex = index; break; case 5: case 6: *chType = ACT_BACK; *chIndex = index - 5; break; case 7: *chType = ACT_LFE; *chIndex = 0; break; } break; case 11: /* SCE,CPE,CPE,SCE,LFE */ if (index < 6) { *chType = ACT_BACK; *chIndex = index - 3; } else { *chType = ACT_LFE; *chIndex = 0; } break; case 12: /* SCE,CPE,CPE,CPE,LFE */ if (index < 7) { *chType = ACT_BACK; *chIndex = index - 3; } else { *chType = ACT_LFE; *chIndex = 0; } break; case 14: /* SCE,CPE,CPE,LFE,CPE */ switch (index) { case 3: case 4: *chType = ACT_BACK; *chIndex = index - 3; break; case 5: *chType = ACT_LFE; *chIndex = 0; break; case 6: case 7: *chType = ACT_FRONT_TOP; *chIndex = index - 6; /* handle the top layer independently */ break; } break; default: *chType = ACT_NONE; break; } } } int CProgramConfig_LookupElement(CProgramConfig *pPce, UINT channelConfig, const UINT tag, const UINT channelIdx, UCHAR chMapping[], AUDIO_CHANNEL_TYPE chType[], UCHAR chIndex[], const UINT chDescrLen, UCHAR *elMapping, MP4_ELEMENT_ID elList[], MP4_ELEMENT_ID elType) { if (channelConfig > 0) { /* Constant channel mapping must have been set during initialization. */ if (IS_CHANNEL_ELEMENT(elType)) { *elMapping = pPce->elCounter; if (elList[pPce->elCounter] != elType && !IS_USAC_CHANNEL_ELEMENT(elType)) { /* Not in the list */ if ((channelConfig == 2) && (elType == ID_SCE)) { /* This scenario occurs with HE-AAC v2 streams of buggy encoders. In other decoder implementations decoding of this kind of streams is desired. */ channelConfig = 1; } else if ((elList[pPce->elCounter] == ID_LFE) && (elType == ID_SCE)) { /* Decode bitstreams which wrongly use ID_SCE instead of ID_LFE element type. */ ; } else { return 0; } } /* Assume all front channels */ getImplicitAudioChannelTypeAndIndex( &chType[channelIdx], &chIndex[channelIdx], channelConfig, channelIdx); if (elType == ID_CPE || elType == ID_USAC_CPE) { chType[channelIdx + 1] = chType[channelIdx]; chIndex[channelIdx + 1] = chIndex[channelIdx] + 1; } pPce->elCounter++; } /* Accept all non-channel elements, too. */ return 1; } else { if ((!pPce->isValid) || (pPce->NumChannels > chDescrLen)) { /* Implicit channel mapping. */ if (IS_USAC_CHANNEL_ELEMENT(elType)) { *elMapping = pPce->elCounter++; } else if (IS_MP4_CHANNEL_ELEMENT(elType)) { /* Store all channel element IDs */ elList[pPce->elCounter] = elType; *elMapping = pPce->elCounter++; } } else { /* Accept the additional channel(s), only if the tag is in the lists */ int isCpe = 0, i; /* Element counter */ int ec[PC_NUM_HEIGHT_LAYER] = {0}; /* Channel counters */ int cc[PC_NUM_HEIGHT_LAYER] = {0}; int fc[PC_NUM_HEIGHT_LAYER] = {0}; /* front channel counter */ int sc[PC_NUM_HEIGHT_LAYER] = {0}; /* side channel counter */ int bc[PC_NUM_HEIGHT_LAYER] = {0}; /* back channel counter */ int lc = 0; /* lfe channel counter */ /* General MPEG (PCE) composition rules: - Over all: - Within each height layer: - Exception: The LFE channels have no height info and thus they are arranged at the very end of the normal height layer channels. */ switch (elType) { case ID_CPE: isCpe = 1; FDK_FALLTHROUGH; case ID_SCE: /* search in front channels */ for (i = 0; i < pPce->NumFrontChannelElements; i++) { int heightLayer = pPce->FrontElementHeightInfo[i]; if (isCpe == pPce->FrontElementIsCpe[i] && pPce->FrontElementTagSelect[i] == tag) { int h, elIdx = ec[heightLayer], chIdx = cc[heightLayer]; AUDIO_CHANNEL_TYPE aChType = (AUDIO_CHANNEL_TYPE)((heightLayer << 4) | ACT_FRONT); for (h = heightLayer - 1; h >= 0; h -= 1) { int el; /* Count front channels/elements */ for (el = 0; el < pPce->NumFrontChannelElements; el += 1) { if (pPce->FrontElementHeightInfo[el] == h) { elIdx += 1; chIdx += (pPce->FrontElementIsCpe[el]) ? 2 : 1; } } /* Count side channels/elements */ for (el = 0; el < pPce->NumSideChannelElements; el += 1) { if (pPce->SideElementHeightInfo[el] == h) { elIdx += 1; chIdx += (pPce->SideElementIsCpe[el]) ? 2 : 1; } } /* Count back channels/elements */ for (el = 0; el < pPce->NumBackChannelElements; el += 1) { if (pPce->BackElementHeightInfo[el] == h) { elIdx += 1; chIdx += (pPce->BackElementIsCpe[el]) ? 2 : 1; } } if (h == 0) { /* normal height */ elIdx += pPce->NumLfeChannelElements; chIdx += pPce->NumLfeChannelElements; } } chMapping[chIdx] = channelIdx; chType[chIdx] = aChType; chIndex[chIdx] = fc[heightLayer]; if (isCpe) { chMapping[chIdx + 1] = channelIdx + 1; chType[chIdx + 1] = aChType; chIndex[chIdx + 1] = fc[heightLayer] + 1; } *elMapping = elIdx; return 1; } ec[heightLayer] += 1; if (pPce->FrontElementIsCpe[i]) { cc[heightLayer] += 2; fc[heightLayer] += 2; } else { cc[heightLayer] += 1; fc[heightLayer] += 1; } } /* search in side channels */ for (i = 0; i < pPce->NumSideChannelElements; i++) { int heightLayer = pPce->SideElementHeightInfo[i]; if (isCpe == pPce->SideElementIsCpe[i] && pPce->SideElementTagSelect[i] == tag) { int h, elIdx = ec[heightLayer], chIdx = cc[heightLayer]; AUDIO_CHANNEL_TYPE aChType = (AUDIO_CHANNEL_TYPE)((heightLayer << 4) | ACT_SIDE); for (h = heightLayer - 1; h >= 0; h -= 1) { int el; /* Count front channels/elements */ for (el = 0; el < pPce->NumFrontChannelElements; el += 1) { if (pPce->FrontElementHeightInfo[el] == h) { elIdx += 1; chIdx += (pPce->FrontElementIsCpe[el]) ? 2 : 1; } } /* Count side channels/elements */ for (el = 0; el < pPce->NumSideChannelElements; el += 1) { if (pPce->SideElementHeightInfo[el] == h) { elIdx += 1; chIdx += (pPce->SideElementIsCpe[el]) ? 2 : 1; } } /* Count back channels/elements */ for (el = 0; el < pPce->NumBackChannelElements; el += 1) { if (pPce->BackElementHeightInfo[el] == h) { elIdx += 1; chIdx += (pPce->BackElementIsCpe[el]) ? 2 : 1; } } if (h == 0) { /* LFE channels belong to the normal height layer */ elIdx += pPce->NumLfeChannelElements; chIdx += pPce->NumLfeChannelElements; } } chMapping[chIdx] = channelIdx; chType[chIdx] = aChType; chIndex[chIdx] = sc[heightLayer]; if (isCpe) { chMapping[chIdx + 1] = channelIdx + 1; chType[chIdx + 1] = aChType; chIndex[chIdx + 1] = sc[heightLayer] + 1; } *elMapping = elIdx; return 1; } ec[heightLayer] += 1; if (pPce->SideElementIsCpe[i]) { cc[heightLayer] += 2; sc[heightLayer] += 2; } else { cc[heightLayer] += 1; sc[heightLayer] += 1; } } /* search in back channels */ for (i = 0; i < pPce->NumBackChannelElements; i++) { int heightLayer = pPce->BackElementHeightInfo[i]; if (isCpe == pPce->BackElementIsCpe[i] && pPce->BackElementTagSelect[i] == tag) { int h, elIdx = ec[heightLayer], chIdx = cc[heightLayer]; AUDIO_CHANNEL_TYPE aChType = (AUDIO_CHANNEL_TYPE)((heightLayer << 4) | ACT_BACK); for (h = heightLayer - 1; h >= 0; h -= 1) { int el; /* Count front channels/elements */ for (el = 0; el < pPce->NumFrontChannelElements; el += 1) { if (pPce->FrontElementHeightInfo[el] == h) { elIdx += 1; chIdx += (pPce->FrontElementIsCpe[el]) ? 2 : 1; } } /* Count side channels/elements */ for (el = 0; el < pPce->NumSideChannelElements; el += 1) { if (pPce->SideElementHeightInfo[el] == h) { elIdx += 1; chIdx += (pPce->SideElementIsCpe[el]) ? 2 : 1; } } /* Count back channels/elements */ for (el = 0; el < pPce->NumBackChannelElements; el += 1) { if (pPce->BackElementHeightInfo[el] == h) { elIdx += 1; chIdx += (pPce->BackElementIsCpe[el]) ? 2 : 1; } } if (h == 0) { /* normal height */ elIdx += pPce->NumLfeChannelElements; chIdx += pPce->NumLfeChannelElements; } } chMapping[chIdx] = channelIdx; chType[chIdx] = aChType; chIndex[chIdx] = bc[heightLayer]; if (isCpe) { chMapping[chIdx + 1] = channelIdx + 1; chType[chIdx + 1] = aChType; chIndex[chIdx + 1] = bc[heightLayer] + 1; } *elMapping = elIdx; return 1; } ec[heightLayer] += 1; if (pPce->BackElementIsCpe[i]) { cc[heightLayer] += 2; bc[heightLayer] += 2; } else { cc[heightLayer] += 1; bc[heightLayer] += 1; } } break; case ID_LFE: { /* Unfortunately we have to go through all normal height layer elements to get the position of the LFE channels. Start with counting the front channels/elements at normal height */ for (i = 0; i < pPce->NumFrontChannelElements; i += 1) { int heightLayer = pPce->FrontElementHeightInfo[i]; ec[heightLayer] += 1; cc[heightLayer] += (pPce->FrontElementIsCpe[i]) ? 2 : 1; } /* Count side channels/elements at normal height */ for (i = 0; i < pPce->NumSideChannelElements; i += 1) { int heightLayer = pPce->SideElementHeightInfo[i]; ec[heightLayer] += 1; cc[heightLayer] += (pPce->SideElementIsCpe[i]) ? 2 : 1; } /* Count back channels/elements at normal height */ for (i = 0; i < pPce->NumBackChannelElements; i += 1) { int heightLayer = pPce->BackElementHeightInfo[i]; ec[heightLayer] += 1; cc[heightLayer] += (pPce->BackElementIsCpe[i]) ? 2 : 1; } /* search in lfe channels */ for (i = 0; i < pPce->NumLfeChannelElements; i++) { int elIdx = ec[0]; /* LFE channels belong to the normal height layer */ int chIdx = cc[0]; if (pPce->LfeElementTagSelect[i] == tag) { chMapping[chIdx] = channelIdx; *elMapping = elIdx; chType[chIdx] = ACT_LFE; chIndex[chIdx] = lc; return 1; } ec[0] += 1; cc[0] += 1; lc += 1; } } break; /* Non audio elements */ case ID_CCE: /* search in cce channels */ for (i = 0; i < pPce->NumValidCcElements; i++) { if (pPce->ValidCcElementTagSelect[i] == tag) { return 1; } } break; case ID_DSE: /* search associated data elements */ for (i = 0; i < pPce->NumAssocDataElements; i++) { if (pPce->AssocDataElementTagSelect[i] == tag) { return 1; } } break; default: return 0; } return 0; /* not found in any list */ } } return 1; } #define SPEAKER_PLANE_NORMAL 0 #define SPEAKER_PLANE_TOP 1 #define SPEAKER_PLANE_BOTTOM 2 void CProgramConfig_GetChannelDescription(const UINT chConfig, const CProgramConfig *pPce, AUDIO_CHANNEL_TYPE chType[], UCHAR chIndex[]) { FDK_ASSERT(chType != NULL); FDK_ASSERT(chIndex != NULL); if ((chConfig == 0) && (pPce != NULL)) { if (pPce->isValid) { int spkPlane, chIdx = 0; for (spkPlane = SPEAKER_PLANE_NORMAL; spkPlane <= SPEAKER_PLANE_BOTTOM; spkPlane += 1) { int elIdx, grpChIdx = 0; for (elIdx = 0; elIdx < pPce->NumFrontChannelElements; elIdx += 1) { if (pPce->FrontElementHeightInfo[elIdx] == spkPlane) { chType[chIdx] = (AUDIO_CHANNEL_TYPE)((spkPlane << 4) | ACT_FRONT); chIndex[chIdx++] = grpChIdx++; if (pPce->FrontElementIsCpe[elIdx]) { chType[chIdx] = (AUDIO_CHANNEL_TYPE)((spkPlane << 4) | ACT_FRONT); chIndex[chIdx++] = grpChIdx++; } } } grpChIdx = 0; for (elIdx = 0; elIdx < pPce->NumSideChannelElements; elIdx += 1) { if (pPce->SideElementHeightInfo[elIdx] == spkPlane) { chType[chIdx] = (AUDIO_CHANNEL_TYPE)((spkPlane << 4) | ACT_SIDE); chIndex[chIdx++] = grpChIdx++; if (pPce->SideElementIsCpe[elIdx]) { chType[chIdx] = (AUDIO_CHANNEL_TYPE)((spkPlane << 4) | ACT_SIDE); chIndex[chIdx++] = grpChIdx++; } } } grpChIdx = 0; for (elIdx = 0; elIdx < pPce->NumBackChannelElements; elIdx += 1) { if (pPce->BackElementHeightInfo[elIdx] == spkPlane) { chType[chIdx] = (AUDIO_CHANNEL_TYPE)((spkPlane << 4) | ACT_BACK); chIndex[chIdx++] = grpChIdx++; if (pPce->BackElementIsCpe[elIdx]) { chType[chIdx] = (AUDIO_CHANNEL_TYPE)((spkPlane << 4) | ACT_BACK); chIndex[chIdx++] = grpChIdx++; } } } grpChIdx = 0; if (spkPlane == SPEAKER_PLANE_NORMAL) { for (elIdx = 0; elIdx < pPce->NumLfeChannelElements; elIdx += 1) { chType[chIdx] = ACT_LFE; chIndex[chIdx++] = grpChIdx++; } } } } } else { int chIdx; for (chIdx = 0; chIdx < getNumberOfTotalChannels(chConfig); chIdx += 1) { getImplicitAudioChannelTypeAndIndex(&chType[chIdx], &chIndex[chIdx], chConfig, chIdx); } } } int CProgramConfig_GetPceChMap(const CProgramConfig *pPce, UCHAR pceChMap[], const UINT pceChMapLen) { const UCHAR *nElements = &pPce->NumFrontChannelElements; const UCHAR *elHeight[3], *elIsCpe[3]; unsigned chIdx, plane, grp, offset, totCh[3], numCh[3][4]; FDK_ASSERT(pPce != NULL); FDK_ASSERT(pceChMap != NULL); /* Init counter: */ FDKmemclear(totCh, 3 * sizeof(unsigned)); FDKmemclear(numCh, 3 * 4 * sizeof(unsigned)); /* Analyse PCE: */ elHeight[0] = pPce->FrontElementHeightInfo; elIsCpe[0] = pPce->FrontElementIsCpe; elHeight[1] = pPce->SideElementHeightInfo; elIsCpe[1] = pPce->SideElementIsCpe; elHeight[2] = pPce->BackElementHeightInfo; elIsCpe[2] = pPce->BackElementIsCpe; for (plane = 0; plane <= SPEAKER_PLANE_BOTTOM; plane += 1) { for (grp = 0; grp < 3; grp += 1) { /* front, side, back */ unsigned el; for (el = 0; el < nElements[grp]; el += 1) { if (elHeight[grp][el] == plane) { unsigned elCh = elIsCpe[grp][el] ? 2 : 1; numCh[plane][grp] += elCh; totCh[plane] += elCh; } } } if (plane == SPEAKER_PLANE_NORMAL) { unsigned elCh = pPce->NumLfeChannelElements; numCh[plane][grp] += elCh; totCh[plane] += elCh; } } /* Sanity checks: */ chIdx = totCh[SPEAKER_PLANE_NORMAL] + totCh[SPEAKER_PLANE_TOP] + totCh[SPEAKER_PLANE_BOTTOM]; if (chIdx > pceChMapLen) { return -1; } /* Create map: */ offset = grp = 0; unsigned grpThresh = numCh[SPEAKER_PLANE_NORMAL][grp]; for (chIdx = 0; chIdx < totCh[SPEAKER_PLANE_NORMAL]; chIdx += 1) { while ((chIdx >= grpThresh) && (grp < 3)) { offset += numCh[1][grp] + numCh[2][grp]; grp += 1; grpThresh += numCh[SPEAKER_PLANE_NORMAL][grp]; } pceChMap[chIdx] = chIdx + offset; } offset = 0; for (grp = 0; grp < 4; grp += 1) { /* front, side, back and lfe */ offset += numCh[SPEAKER_PLANE_NORMAL][grp]; for (plane = SPEAKER_PLANE_TOP; plane <= SPEAKER_PLANE_BOTTOM; plane += 1) { unsigned mapCh; for (mapCh = 0; mapCh < numCh[plane][grp]; mapCh += 1) { pceChMap[chIdx++] = offset; offset += 1; } } } return 0; } int CProgramConfig_GetElementTable(const CProgramConfig *pPce, MP4_ELEMENT_ID elList[], const INT elListSize, UCHAR *pChMapIdx) { int i, el = 0; FDK_ASSERT(elList != NULL); FDK_ASSERT(pChMapIdx != NULL); FDK_ASSERT(pPce != NULL); *pChMapIdx = 0; if ((elListSize < pPce->NumFrontChannelElements + pPce->NumSideChannelElements + pPce->NumBackChannelElements + pPce->NumLfeChannelElements) || (pPce->NumChannels == 0)) { return 0; } for (i = 0; i < pPce->NumFrontChannelElements; i += 1) { elList[el++] = (pPce->FrontElementIsCpe[i]) ? ID_CPE : ID_SCE; } for (i = 0; i < pPce->NumSideChannelElements; i += 1) { elList[el++] = (pPce->SideElementIsCpe[i]) ? ID_CPE : ID_SCE; } for (i = 0; i < pPce->NumBackChannelElements; i += 1) { elList[el++] = (pPce->BackElementIsCpe[i]) ? ID_CPE : ID_SCE; } for (i = 0; i < pPce->NumLfeChannelElements; i += 1) { elList[el++] = ID_LFE; } /* Find an corresponding channel configuration if possible */ switch (pPce->NumChannels) { case 1: case 2: /* One and two channels have no alternatives. */ *pChMapIdx = pPce->NumChannels; break; case 3: case 4: case 5: case 6: { /* Test if the number of channels can be used as channel config: */ C_ALLOC_SCRATCH_START(tmpPce, CProgramConfig, 1); /* Create a PCE for the config to test ... */ CProgramConfig_GetDefault(tmpPce, pPce->NumChannels); /* ... and compare it with the given one. */ *pChMapIdx = (!(CProgramConfig_Compare(pPce, tmpPce) & 0xE)) ? pPce->NumChannels : 0; /* If compare result is 0 or 1 we can be sure that it is channel * config 11. */ C_ALLOC_SCRATCH_END(tmpPce, CProgramConfig, 1); } break; case 7: { C_ALLOC_SCRATCH_START(tmpPce, CProgramConfig, 1); /* Create a PCE for the config to test ... */ CProgramConfig_GetDefault(tmpPce, 11); /* ... and compare it with the given one. */ *pChMapIdx = (!(CProgramConfig_Compare(pPce, tmpPce) & 0xE)) ? 11 : 0; /* If compare result is 0 or 1 we can be sure that it is channel * config 11. */ C_ALLOC_SCRATCH_END(tmpPce, CProgramConfig, 1); } break; case 8: { /* Try the four possible 7.1ch configurations. One after the other. */ UCHAR testCfg[4] = {32, 14, 12, 7}; C_ALLOC_SCRATCH_START(tmpPce, CProgramConfig, 1); for (i = 0; i < 4; i += 1) { /* Create a PCE for the config to test ... */ CProgramConfig_GetDefault(tmpPce, testCfg[i]); /* ... and compare it with the given one. */ if (!(CProgramConfig_Compare(pPce, tmpPce) & 0xE)) { /* If the compare result is 0 or 1 than the two channel configurations * match. */ /* Explicit mapping of 7.1 side channel configuration to 7.1 rear * channel mapping. */ *pChMapIdx = (testCfg[i] == 32) ? 12 : testCfg[i]; } } C_ALLOC_SCRATCH_END(tmpPce, CProgramConfig, 1); } break; default: /* The PCE does not match any predefined channel configuration. */ *pChMapIdx = 0; break; } return el; } static AUDIO_OBJECT_TYPE getAOT(HANDLE_FDK_BITSTREAM bs) { int tmp = 0; tmp = FDKreadBits(bs, 5); if (tmp == AOT_ESCAPE) { int tmp2 = FDKreadBits(bs, 6); tmp = 32 + tmp2; } return (AUDIO_OBJECT_TYPE)tmp; } static INT getSampleRate(HANDLE_FDK_BITSTREAM bs, UCHAR *index, int nBits) { INT sampleRate; int idx; idx = FDKreadBits(bs, nBits); if (idx == (1 << nBits) - 1) { if (FDKgetValidBits(bs) < 24) { return 0; } sampleRate = FDKreadBits(bs, 24); } else { sampleRate = SamplingRateTable[idx]; } *index = idx; return sampleRate; } static TRANSPORTDEC_ERROR GaSpecificConfig_Parse(CSGaSpecificConfig *self, CSAudioSpecificConfig *asc, HANDLE_FDK_BITSTREAM bs, UINT ascStartAnchor) { TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK; self->m_frameLengthFlag = FDKreadBits(bs, 1); self->m_dependsOnCoreCoder = FDKreadBits(bs, 1); if (self->m_dependsOnCoreCoder) self->m_coreCoderDelay = FDKreadBits(bs, 14); self->m_extensionFlag = FDKreadBits(bs, 1); if (asc->m_channelConfiguration == 0) { CProgramConfig_Read(&asc->m_progrConfigElement, bs, ascStartAnchor); } if ((asc->m_aot == AOT_AAC_SCAL) || (asc->m_aot == AOT_ER_AAC_SCAL)) { self->m_layer = FDKreadBits(bs, 3); } if (self->m_extensionFlag) { if (asc->m_aot == AOT_ER_BSAC) { self->m_numOfSubFrame = FDKreadBits(bs, 5); self->m_layerLength = FDKreadBits(bs, 11); } if ((asc->m_aot == AOT_ER_AAC_LC) || (asc->m_aot == AOT_ER_AAC_LTP) || (asc->m_aot == AOT_ER_AAC_SCAL) || (asc->m_aot == AOT_ER_AAC_LD)) { asc->m_vcb11Flag = FDKreadBits(bs, 1); /* aacSectionDataResilienceFlag */ asc->m_rvlcFlag = FDKreadBits(bs, 1); /* aacScalefactorDataResilienceFlag */ asc->m_hcrFlag = FDKreadBits(bs, 1); /* aacSpectralDataResilienceFlag */ } self->m_extensionFlag3 = FDKreadBits(bs, 1); } return (ErrorStatus); } static INT skipSbrHeader(HANDLE_FDK_BITSTREAM hBs, int isUsac) { /* Dummy parse SbrDfltHeader() */ INT dflt_header_extra1, dflt_header_extra2, bitsToSkip = 0; if (!isUsac) { bitsToSkip = 6; FDKpushFor(hBs, 6); /* amp res 1, xover freq 3, reserved 2 */ } bitsToSkip += 8; FDKpushFor(hBs, 8); /* start / stop freq */ bitsToSkip += 2; dflt_header_extra1 = FDKreadBit(hBs); dflt_header_extra2 = FDKreadBit(hBs); bitsToSkip += 5 * dflt_header_extra1 + 6 * dflt_header_extra2; FDKpushFor(hBs, 5 * dflt_header_extra1 + 6 * dflt_header_extra2); return bitsToSkip; } static INT ld_sbr_header(CSAudioSpecificConfig *asc, const INT dsFactor, HANDLE_FDK_BITSTREAM hBs, CSTpCallBacks *cb) { const int channelConfiguration = asc->m_channelConfiguration; int i = 0, j = 0; INT error = 0; MP4_ELEMENT_ID element = ID_NONE; /* check whether the channelConfiguration is defined in * channel_configuration_array */ if (channelConfiguration < 0 || channelConfiguration > (INT)(sizeof(channel_configuration_array) / sizeof(MP4_ELEMENT_ID **) - 1)) { return TRANSPORTDEC_PARSE_ERROR; } /* read elements of the passed channel_configuration until there is ID_NONE */ while ((element = channel_configuration_array[channelConfiguration][j]) != ID_NONE) { /* Setup LFE element for upsampling too. This is essential especially for * channel configs where the LFE element is not at the last position for * example in channel config 13 or 14. It leads to memory leaks if the setup * of the LFE element would be done later in the core. */ if (element == ID_SCE || element == ID_CPE || element == ID_LFE) { error |= cb->cbSbr( cb->cbSbrData, hBs, asc->m_samplingFrequency / dsFactor, asc->m_extensionSamplingFrequency / dsFactor, asc->m_samplesPerFrame / dsFactor, AOT_ER_AAC_ELD, element, i++, 0, 0, asc->configMode, &asc->SbrConfigChanged, dsFactor); if (error != TRANSPORTDEC_OK) { goto bail; } } j++; } bail: return error; } static TRANSPORTDEC_ERROR EldSpecificConfig_Parse(CSAudioSpecificConfig *asc, HANDLE_FDK_BITSTREAM hBs, CSTpCallBacks *cb) { TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK; CSEldSpecificConfig *esc = &asc->m_sc.m_eldSpecificConfig; UINT eldExtType; int eldExtLen, len, cnt, ldSbrLen = 0, eldExtLenSum, numSbrHeader = 0, sbrIndex, eldExtCnt = 0; unsigned char downscale_fill_nibble; FDKmemclear(esc, sizeof(CSEldSpecificConfig)); esc->m_frameLengthFlag = FDKreadBits(hBs, 1); if (esc->m_frameLengthFlag) { asc->m_samplesPerFrame = 480; } else { asc->m_samplesPerFrame = 512; } asc->m_vcb11Flag = FDKreadBits(hBs, 1); asc->m_rvlcFlag = FDKreadBits(hBs, 1); asc->m_hcrFlag = FDKreadBits(hBs, 1); esc->m_sbrPresentFlag = FDKreadBits(hBs, 1); if (esc->m_sbrPresentFlag == 1) { esc->m_sbrSamplingRate = FDKreadBits(hBs, 1); /* 0: single rate, 1: dual rate */ esc->m_sbrCrcFlag = FDKreadBits(hBs, 1); asc->m_extensionSamplingFrequency = asc->m_samplingFrequency << esc->m_sbrSamplingRate; if (cb->cbSbr != NULL) { /* ELD reduced delay mode: LD-SBR initialization has to know the downscale information. Postpone LD-SBR initialization and read ELD extension information first. */ switch (asc->m_channelConfiguration) { case 1: case 2: numSbrHeader = 1; break; case 3: numSbrHeader = 2; break; case 4: case 5: case 6: numSbrHeader = 3; break; case 7: case 11: case 12: case 14: numSbrHeader = 4; break; default: numSbrHeader = 0; break; } for (sbrIndex = 0; sbrIndex < numSbrHeader; sbrIndex++) { ldSbrLen += skipSbrHeader(hBs, 0); } } else { return TRANSPORTDEC_UNSUPPORTED_FORMAT; } } esc->m_useLdQmfTimeAlign = 0; /* new ELD syntax */ eldExtLenSum = FDKgetValidBits(hBs); esc->m_downscaledSamplingFrequency = asc->m_samplingFrequency; /* parse ExtTypeConfigData */ while (((eldExtType = FDKreadBits(hBs, 4)) != ELDEXT_TERM) && ((INT)FDKgetValidBits(hBs) >= 0) && (eldExtCnt++ < 15)) { eldExtLen = len = FDKreadBits(hBs, 4); if (len == 0xf) { len = FDKreadBits(hBs, 8); eldExtLen += len; if (len == 0xff) { len = FDKreadBits(hBs, 16); eldExtLen += len; } } switch (eldExtType) { case ELDEXT_LDSAC: esc->m_useLdQmfTimeAlign = 1; if (cb->cbSsc != NULL) { ErrorStatus = (TRANSPORTDEC_ERROR)cb->cbSsc( cb->cbSscData, hBs, asc->m_aot, asc->m_samplingFrequency << esc->m_sbrSamplingRate, asc->m_samplesPerFrame << esc->m_sbrSamplingRate, 1, /* stereoConfigIndex */ -1, /* nTimeSlots: read from bitstream */ eldExtLen, asc->configMode, &asc->SacConfigChanged); if (ErrorStatus != TRANSPORTDEC_OK) { return TRANSPORTDEC_PARSE_ERROR; } if (esc->m_downscaledSamplingFrequency != asc->m_samplingFrequency) { return TRANSPORTDEC_UNSUPPORTED_FORMAT; /* ELDv2 w/ ELD downscaled mode not allowed */ } break; } FDK_FALLTHROUGH; default: for (cnt = 0; cnt < eldExtLen; cnt++) { FDKreadBits(hBs, 8); } break; case ELDEXT_DOWNSCALEINFO: UCHAR tmpDownscaleFreqIdx; esc->m_downscaledSamplingFrequency = getSampleRate(hBs, &tmpDownscaleFreqIdx, 4); if (esc->m_downscaledSamplingFrequency == 0 || esc->m_downscaledSamplingFrequency > 96000) { return TRANSPORTDEC_PARSE_ERROR; } downscale_fill_nibble = FDKreadBits(hBs, 4); if (downscale_fill_nibble != 0x0) { return TRANSPORTDEC_PARSE_ERROR; } if (esc->m_useLdQmfTimeAlign == 1) { return TRANSPORTDEC_UNSUPPORTED_FORMAT; /* ELDv2 w/ ELD downscaled mode not allowed */ } break; } } if (eldExtType != ELDEXT_TERM) { return TRANSPORTDEC_PARSE_ERROR; } if ((INT)FDKgetValidBits(hBs) < 0) { return TRANSPORTDEC_PARSE_ERROR; } if (esc->m_sbrPresentFlag == 1 && numSbrHeader != 0) { INT dsFactor = 1; /* Downscale factor must be 1 or even for SBR */ if (esc->m_downscaledSamplingFrequency != 0) { if (asc->m_samplingFrequency % esc->m_downscaledSamplingFrequency != 0) { return TRANSPORTDEC_UNSUPPORTED_FORMAT; } dsFactor = asc->m_samplingFrequency / esc->m_downscaledSamplingFrequency; if (dsFactor != 1 && (dsFactor)&1) { return TRANSPORTDEC_UNSUPPORTED_FORMAT; /* SBR needs an even downscale factor */ } if (dsFactor != 1 && dsFactor != 2 && dsFactor != 4) { dsFactor = 1; /* don't apply dsf for not yet supported even dsfs */ } if ((INT)asc->m_samplesPerFrame % dsFactor != 0) { return TRANSPORTDEC_UNSUPPORTED_FORMAT; /* frameSize/dsf must be an integer number */ } } eldExtLenSum = eldExtLenSum - FDKgetValidBits(hBs); FDKpushBack(hBs, eldExtLenSum + ldSbrLen); if (0 != ld_sbr_header(asc, dsFactor, hBs, cb)) { return TRANSPORTDEC_PARSE_ERROR; } FDKpushFor(hBs, eldExtLenSum); } return (ErrorStatus); } /* Subroutine to store config in UCHAR buffer. Bit stream position does not change. */ static UINT StoreConfigAsBitstream( HANDLE_FDK_BITSTREAM hBs, const INT configSize_bits, /* If < 0 (> 0) config to read is before (after) current bit stream position. */ UCHAR *configTargetBuffer, const USHORT configTargetBufferSize_bytes) { FDK_BITSTREAM usacConf; UINT const nBits = fAbs(configSize_bits); UINT j, tmp; if (nBits > 8 * (UINT)configTargetBufferSize_bytes) { return 1; } FDKmemclear(configTargetBuffer, configTargetBufferSize_bytes); FDKinitBitStream(&usacConf, configTargetBuffer, configTargetBufferSize_bytes, nBits, BS_WRITER); if (configSize_bits < 0) { FDKpushBack(hBs, nBits); } for (j = nBits; j > 31; j -= 32) { tmp = FDKreadBits(hBs, 32); FDKwriteBits(&usacConf, tmp, 32); } if (j > 0) { tmp = FDKreadBits(hBs, j); FDKwriteBits(&usacConf, tmp, j); } FDKsyncCache(&usacConf); if (configSize_bits > 0) { FDKpushBack(hBs, nBits); } return 0; } /* maps coreSbrFrameLengthIndex to coreCoderFrameLength */ static const USHORT usacFrameLength[8] = {768, 1024, 2048, 2048, 4096, 0, 0, 0}; /* maps coreSbrFrameLengthIndex to sbrRatioIndex */ static const UCHAR sbrRatioIndex[8] = {0, 0, 2, 3, 1, 0, 0, 0}; /* subroutine for parsing extension element configuration: UsacExtElementConfig() q.v. ISO/IEC FDIS 23003-3:2011(E) Table 14 rsv603daExtElementConfig() q.v. ISO/IEC DIS 23008-3 Table 13 */ static TRANSPORTDEC_ERROR extElementConfig(CSUsacExtElementConfig *extElement, HANDLE_FDK_BITSTREAM hBs, const CSTpCallBacks *cb, const UCHAR numSignalsInGroup, const UINT coreFrameLength, const int subStreamIndex, const AUDIO_OBJECT_TYPE aot) { TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK; int usacExtElementType = escapedValue(hBs, 4, 8, 16); /* recurve extension elements which are invalid for USAC */ if (aot == AOT_USAC) { switch (usacExtElementType) { case ID_EXT_ELE_FILL: case ID_EXT_ELE_MPEGS: case ID_EXT_ELE_SAOC: case ID_EXT_ELE_AUDIOPREROLL: case ID_EXT_ELE_UNI_DRC: break; default: usacExtElementType = ID_EXT_ELE_UNKNOWN; break; } } extElement->usacExtElementType = (USAC_EXT_ELEMENT_TYPE)usacExtElementType; int usacExtElementConfigLength = escapedValue(hBs, 4, 8, 16); extElement->usacExtElementConfigLength = (USHORT)usacExtElementConfigLength; INT bsAnchor; if (FDKreadBit(hBs)) /* usacExtElementDefaultLengthPresent */ extElement->usacExtElementDefaultLength = escapedValue(hBs, 8, 16, 0) + 1; else extElement->usacExtElementDefaultLength = 0; extElement->usacExtElementPayloadFrag = FDKreadBit(hBs); bsAnchor = (INT)FDKgetValidBits(hBs); switch (usacExtElementType) { case ID_EXT_ELE_UNKNOWN: case ID_EXT_ELE_FILL: break; case ID_EXT_ELE_AUDIOPREROLL: /* No configuration element */ extElement->usacExtElementHasAudioPreRoll = 1; break; case ID_EXT_ELE_UNI_DRC: { if (cb->cbUniDrc != NULL) { ErrorStatus = (TRANSPORTDEC_ERROR)cb->cbUniDrc( cb->cbUniDrcData, hBs, usacExtElementConfigLength, 0, /* uniDrcConfig */ subStreamIndex, 0, aot); if (ErrorStatus != TRANSPORTDEC_OK) { return ErrorStatus; } } } break; default: break; } /* Adjust bit stream position. This is required because of byte alignment and * unhandled extensions. */ { INT left_bits = (usacExtElementConfigLength << 3) - (bsAnchor - (INT)FDKgetValidBits(hBs)); if (left_bits >= 0) { FDKpushFor(hBs, left_bits); } else { /* parsed too many bits */ ErrorStatus = TRANSPORTDEC_PARSE_ERROR; } } return ErrorStatus; } /* subroutine for parsing the USAC / RSVD60 configuration extension: UsacConfigExtension() q.v. ISO/IEC FDIS 23003-3:2011(E) Table 15 rsv603daConfigExtension() q.v. ISO/IEC DIS 23008-3 Table 14 */ static TRANSPORTDEC_ERROR configExtension(CSUsacConfig *usc, HANDLE_FDK_BITSTREAM hBs, const CSTpCallBacks *cb) { TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK; int numConfigExtensions; int usacConfigExtType; int usacConfigExtLength; numConfigExtensions = (int)escapedValue(hBs, 2, 4, 8) + 1; for (int confExtIdx = 0; confExtIdx < numConfigExtensions; confExtIdx++) { INT nbits; int loudnessInfoSetConfigExtensionPosition = FDKgetValidBits(hBs); usacConfigExtType = escapedValue(hBs, 4, 8, 16); usacConfigExtLength = (int)escapedValue(hBs, 4, 8, 16); /* Start bit position of config extension */ nbits = (INT)FDKgetValidBits(hBs); /* Return an error in case the bitbuffer fill level is too low. */ if (nbits < usacConfigExtLength * 8) { return TRANSPORTDEC_PARSE_ERROR; } switch (usacConfigExtType) { case ID_CONFIG_EXT_FILL: for (int i = 0; i < usacConfigExtLength; i++) { if (FDKreadBits(hBs, 8) != 0xa5) { return TRANSPORTDEC_PARSE_ERROR; } } break; case ID_CONFIG_EXT_LOUDNESS_INFO: { if (cb->cbUniDrc != NULL) { ErrorStatus = (TRANSPORTDEC_ERROR)cb->cbUniDrc( cb->cbUniDrcData, hBs, usacConfigExtLength, 1, /* loudnessInfoSet */ 0, loudnessInfoSetConfigExtensionPosition, AOT_USAC); if (ErrorStatus != TRANSPORTDEC_OK) { return ErrorStatus; } } } break; default: break; } /* Skip remaining bits. If too many bits were parsed, assume error. */ usacConfigExtLength = 8 * usacConfigExtLength - (nbits - (INT)FDKgetValidBits(hBs)); if (usacConfigExtLength < 0) { return TRANSPORTDEC_PARSE_ERROR; } FDKpushFor(hBs, usacConfigExtLength); } return ErrorStatus; } /* This function unifies decoder config parsing of USAC and RSV60: rsv603daDecoderConfig() ISO/IEC DIS 23008-3 Table 8 UsacDecoderConfig() ISO/IEC FDIS 23003-3 Table 6 */ static TRANSPORTDEC_ERROR UsacRsv60DecoderConfig_Parse( CSAudioSpecificConfig *asc, HANDLE_FDK_BITSTREAM hBs, const CSTpCallBacks *cb) { TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK; CSUsacConfig *usc = &asc->m_sc.m_usacConfig; int i, numberOfElements; int channelElementIdx = 0; /* index for elements which contain audio channels (sce, cpe, lfe) */ SC_CHANNEL_CONFIG sc_chan_config = {0, 0, 0, 0}; numberOfElements = (int)escapedValue(hBs, 4, 8, 16) + 1; usc->m_usacNumElements = numberOfElements; if (numberOfElements > TP_USAC_MAX_ELEMENTS) { return TRANSPORTDEC_UNSUPPORTED_FORMAT; } usc->m_nUsacChannels = 0; usc->m_channelConfigurationIndex = asc->m_channelConfiguration; if (asc->m_aot == AOT_USAC) { sc_chan_config = sc_chan_config_tab[usc->m_channelConfigurationIndex]; if (sc_chan_config.nCh > (SCHAR)TP_USAC_MAX_SPEAKERS) { return TRANSPORTDEC_PARSE_ERROR; } } for (i = 0; i < numberOfElements; i++) { MP4_ELEMENT_ID usacElementType = (MP4_ELEMENT_ID)( FDKreadBits(hBs, 2) | USAC_ID_BIT); /* set USAC_ID_BIT to map usacElementType to MP4_ELEMENT_ID enum */ usc->element[i].usacElementType = usacElementType; /* sanity check: update element counter */ if (asc->m_aot == AOT_USAC) { switch (usacElementType) { case ID_USAC_SCE: sc_chan_config.nSCE--; break; case ID_USAC_CPE: sc_chan_config.nCPE--; break; case ID_USAC_LFE: sc_chan_config.nLFE--; break; default: break; } if (usc->m_channelConfigurationIndex) { /* sanity check: no element counter may be smaller zero */ if (sc_chan_config.nCPE < 0 || sc_chan_config.nSCE < 0 || sc_chan_config.nLFE < 0) { return TRANSPORTDEC_PARSE_ERROR; } } } switch (usacElementType) { case ID_USAC_SCE: /* UsacCoreConfig() ISO/IEC FDIS 23003-3 Table 10 */ if (FDKreadBit(hBs)) { /* tw_mdct */ return TRANSPORTDEC_UNSUPPORTED_FORMAT; } usc->element[i].m_noiseFilling = FDKreadBits(hBs, 1); /* end of UsacCoreConfig() */ if (usc->m_sbrRatioIndex > 0) { if (cb->cbSbr == NULL) { return TRANSPORTDEC_UNKOWN_ERROR; } /* SbrConfig() ISO/IEC FDIS 23003-3 Table 11 */ usc->element[i].m_harmonicSBR = FDKreadBit(hBs); usc->element[i].m_interTes = FDKreadBit(hBs); usc->element[i].m_pvc = FDKreadBit(hBs); if (cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, asc->m_aot, ID_SCE, channelElementIdx, usc->element[i].m_harmonicSBR, usc->element[i].m_stereoConfigIndex, asc->configMode, &asc->SbrConfigChanged, 1)) { return TRANSPORTDEC_PARSE_ERROR; } /* end of SbrConfig() */ } usc->m_nUsacChannels += 1; channelElementIdx++; break; case ID_USAC_CPE: /* UsacCoreConfig() ISO/IEC FDIS 23003-3 Table 10 */ if (FDKreadBit(hBs)) { /* tw_mdct */ return TRANSPORTDEC_UNSUPPORTED_FORMAT; } usc->element[i].m_noiseFilling = FDKreadBits(hBs, 1); /* end of UsacCoreConfig() */ if (usc->m_sbrRatioIndex > 0) { if (cb->cbSbr == NULL) return TRANSPORTDEC_UNKOWN_ERROR; /* SbrConfig() ISO/IEC FDIS 23003-3 */ usc->element[i].m_harmonicSBR = FDKreadBit(hBs); usc->element[i].m_interTes = FDKreadBit(hBs); usc->element[i].m_pvc = FDKreadBit(hBs); { INT bitsToSkip = skipSbrHeader(hBs, 1); /* read stereoConfigIndex */ usc->element[i].m_stereoConfigIndex = FDKreadBits(hBs, 2); /* rewind */ FDKpushBack(hBs, bitsToSkip + 2); } { MP4_ELEMENT_ID el_type = (usc->element[i].m_stereoConfigIndex == 1 || usc->element[i].m_stereoConfigIndex == 2) ? ID_SCE : ID_CPE; if (cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, asc->m_aot, el_type, channelElementIdx, usc->element[i].m_harmonicSBR, usc->element[i].m_stereoConfigIndex, asc->configMode, &asc->SbrConfigChanged, 1)) { return TRANSPORTDEC_PARSE_ERROR; } } /* end of SbrConfig() */ usc->element[i].m_stereoConfigIndex = FDKreadBits(hBs, 2); /* Needed in RM5 syntax */ if (usc->element[i].m_stereoConfigIndex > 0) { if (cb->cbSsc != NULL) { int samplesPerFrame = asc->m_samplesPerFrame; if (usc->m_sbrRatioIndex == 1) samplesPerFrame <<= 2; if (usc->m_sbrRatioIndex == 2) samplesPerFrame = (samplesPerFrame * 8) / 3; if (usc->m_sbrRatioIndex == 3) samplesPerFrame <<= 1; /* Mps212Config() ISO/IEC FDIS 23003-3 */ if (cb->cbSsc(cb->cbSscData, hBs, asc->m_aot, asc->m_extensionSamplingFrequency, samplesPerFrame, usc->element[i].m_stereoConfigIndex, usc->m_coreSbrFrameLengthIndex, 0, /* don't know the length */ asc->configMode, &asc->SacConfigChanged)) { return TRANSPORTDEC_PARSE_ERROR; } /* end of Mps212Config() */ } else { return TRANSPORTDEC_UNKOWN_ERROR; } } } else { usc->element[i].m_stereoConfigIndex = 0; } usc->m_nUsacChannels += 2; channelElementIdx++; break; case ID_USAC_LFE: usc->element[i].m_noiseFilling = 0; usc->m_nUsacChannels += 1; if (usc->m_sbrRatioIndex > 0) { /* Use SBR for upsampling */ if (cb->cbSbr == NULL) return ErrorStatus = TRANSPORTDEC_UNKOWN_ERROR; usc->element[i].m_harmonicSBR = (UCHAR)0; usc->element[i].m_interTes = (UCHAR)0; usc->element[i].m_pvc = (UCHAR)0; if (cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, asc->m_aot, ID_LFE, channelElementIdx, usc->element[i].m_harmonicSBR, usc->element[i].m_stereoConfigIndex, asc->configMode, &asc->SbrConfigChanged, 1)) { return ErrorStatus = TRANSPORTDEC_PARSE_ERROR; } } channelElementIdx++; break; case ID_USAC_EXT: ErrorStatus = extElementConfig(&usc->element[i].extElement, hBs, cb, 0, asc->m_samplesPerFrame, 0, asc->m_aot); if (ErrorStatus) { return ErrorStatus; } break; default: /* non USAC-element encountered */ return TRANSPORTDEC_PARSE_ERROR; } } if (asc->m_aot == AOT_USAC) { if (usc->m_channelConfigurationIndex) { /* sanity check: all element counter must be zero */ if (sc_chan_config.nCPE | sc_chan_config.nSCE | sc_chan_config.nLFE) { return TRANSPORTDEC_PARSE_ERROR; } } else { /* sanity check: number of audio channels shall be equal to or smaller * than the accumulated sum of all channels */ if ((INT)(-2 * sc_chan_config.nCPE - sc_chan_config.nSCE - sc_chan_config.nLFE) < (INT)usc->numAudioChannels) { return TRANSPORTDEC_PARSE_ERROR; } } } return ErrorStatus; } /* Mapping of coreSbrFrameLengthIndex defined by Table 70 in ISO/IEC 23003-3 */ static TRANSPORTDEC_ERROR UsacConfig_SetCoreSbrFrameLengthIndex( CSAudioSpecificConfig *asc, int coreSbrFrameLengthIndex) { int sbrRatioIndex_val; if (coreSbrFrameLengthIndex > 4) { return TRANSPORTDEC_PARSE_ERROR; /* reserved values */ } asc->m_sc.m_usacConfig.m_coreSbrFrameLengthIndex = coreSbrFrameLengthIndex; asc->m_samplesPerFrame = usacFrameLength[coreSbrFrameLengthIndex]; sbrRatioIndex_val = sbrRatioIndex[coreSbrFrameLengthIndex]; asc->m_sc.m_usacConfig.m_sbrRatioIndex = sbrRatioIndex_val; if (sbrRatioIndex_val > 0) { asc->m_sbrPresentFlag = 1; asc->m_extensionSamplingFrequency = asc->m_samplingFrequency; asc->m_extensionSamplingFrequencyIndex = asc->m_samplingFrequencyIndex; switch (sbrRatioIndex_val) { case 1: /* sbrRatio = 4:1 */ asc->m_samplingFrequency >>= 2; asc->m_samplesPerFrame >>= 2; break; case 2: /* sbrRatio = 8:3 */ asc->m_samplingFrequency = (asc->m_samplingFrequency * 3) / 8; asc->m_samplesPerFrame = (asc->m_samplesPerFrame * 3) / 8; break; case 3: /* sbrRatio = 2:1 */ asc->m_samplingFrequency >>= 1; asc->m_samplesPerFrame >>= 1; break; default: return TRANSPORTDEC_PARSE_ERROR; } asc->m_samplingFrequencyIndex = getSamplingRateIndex(asc->m_samplingFrequency, 4); } return TRANSPORTDEC_OK; } static TRANSPORTDEC_ERROR UsacConfig_Parse(CSAudioSpecificConfig *asc, HANDLE_FDK_BITSTREAM hBs, CSTpCallBacks *cb) { int usacSamplingFrequency, channelConfigurationIndex, coreSbrFrameLengthIndex; TRANSPORTDEC_ERROR err = TRANSPORTDEC_OK; /* Start bit position of usacConfig */ INT nbits = (INT)FDKgetValidBits(hBs); usacSamplingFrequency = getSampleRate(hBs, &asc->m_samplingFrequencyIndex, 5); if (usacSamplingFrequency == 0 || usacSamplingFrequency > 96000) { return TRANSPORTDEC_PARSE_ERROR; } asc->m_samplingFrequency = (UINT)usacSamplingFrequency; coreSbrFrameLengthIndex = FDKreadBits(hBs, 3); if (UsacConfig_SetCoreSbrFrameLengthIndex(asc, coreSbrFrameLengthIndex) != TRANSPORTDEC_OK) { return TRANSPORTDEC_PARSE_ERROR; } channelConfigurationIndex = FDKreadBits(hBs, 5); if (channelConfigurationIndex > 2) { return TRANSPORTDEC_PARSE_ERROR; /* only channelConfigurationIndex = [1,2] are supported */ } if (channelConfigurationIndex == 0) { return TRANSPORTDEC_PARSE_ERROR; /* only channelConfigurationIndex = [1,2] are supported */ } asc->m_channelConfiguration = channelConfigurationIndex; err = UsacRsv60DecoderConfig_Parse(asc, hBs, cb); if (err != TRANSPORTDEC_OK) { return err; } if (FDKreadBits(hBs, 1)) { /* usacConfigExtensionPresent */ err = configExtension(&asc->m_sc.m_usacConfig, hBs, cb); if (err != TRANSPORTDEC_OK) { return err; } } /* sanity check whether number of channels signaled in UsacDecoderConfig() matches the number of channels required by channelConfigurationIndex */ if ((channelConfigurationIndex > 0) && (sc_chan_config_tab[channelConfigurationIndex].nCh != asc->m_sc.m_usacConfig.m_nUsacChannels)) { return TRANSPORTDEC_PARSE_ERROR; } /* Copy UsacConfig() to asc->m_sc.m_usacConfig.UsacConfig[] buffer. */ INT configSize_bits = (INT)FDKgetValidBits(hBs) - nbits; StoreConfigAsBitstream(hBs, configSize_bits, asc->m_sc.m_usacConfig.UsacConfig, TP_USAC_MAX_CONFIG_LEN); asc->m_sc.m_usacConfig.UsacConfigBits = fAbs(configSize_bits); return err; } static TRANSPORTDEC_ERROR AudioSpecificConfig_ExtensionParse( CSAudioSpecificConfig *self, HANDLE_FDK_BITSTREAM bs, CSTpCallBacks *cb) { TP_ASC_EXTENSION_ID lastAscExt, ascExtId = ASCEXT_UNKOWN; INT bitsAvailable = (INT)FDKgetValidBits(bs); while (bitsAvailable >= 11) { lastAscExt = ascExtId; ascExtId = (TP_ASC_EXTENSION_ID)FDKreadBits(bs, 11); bitsAvailable -= 11; switch (ascExtId) { case ASCEXT_SBR: /* 0x2b7 */ if ((self->m_extensionAudioObjectType != AOT_SBR) && (bitsAvailable >= 5)) { self->m_extensionAudioObjectType = getAOT(bs); if ((self->m_extensionAudioObjectType == AOT_SBR) || (self->m_extensionAudioObjectType == AOT_ER_BSAC)) { /* Get SBR extension configuration */ self->m_sbrPresentFlag = FDKreadBits(bs, 1); if (self->m_aot == AOT_USAC && self->m_sbrPresentFlag > 0 && self->m_sc.m_usacConfig.m_sbrRatioIndex == 0) { return TRANSPORTDEC_PARSE_ERROR; } if (self->m_sbrPresentFlag == 1) { self->m_extensionSamplingFrequency = getSampleRate( bs, &self->m_extensionSamplingFrequencyIndex, 4); if (self->m_extensionSamplingFrequency == 0 || self->m_extensionSamplingFrequency > 96000) { return TRANSPORTDEC_PARSE_ERROR; } } if (self->m_extensionAudioObjectType == AOT_ER_BSAC) { self->m_extensionChannelConfiguration = FDKreadBits(bs, 4); } } /* Update counter because of variable length fields (AOT and sampling * rate) */ bitsAvailable = (INT)FDKgetValidBits(bs); } break; case ASCEXT_PS: /* 0x548 */ if ((lastAscExt == ASCEXT_SBR) && (self->m_extensionAudioObjectType == AOT_SBR) && (bitsAvailable > 0)) { /* Get PS extension configuration */ self->m_psPresentFlag = FDKreadBits(bs, 1); bitsAvailable -= 1; } break; case ASCEXT_MPS: /* 0x76a */ if (self->m_extensionAudioObjectType == AOT_MPEGS) break; FDK_FALLTHROUGH; case ASCEXT_LDMPS: /* 0x7cc */ if ((ascExtId == ASCEXT_LDMPS) && (self->m_extensionAudioObjectType == AOT_LD_MPEGS)) break; if (bitsAvailable >= 1) { bitsAvailable -= 1; if (FDKreadBits(bs, 1)) { /* self->m_mpsPresentFlag */ int sscLen = FDKreadBits(bs, 8); bitsAvailable -= 8; if (sscLen == 0xFF) { sscLen += FDKreadBits(bs, 16); bitsAvailable -= 16; } FDKpushFor(bs, sscLen); /* Skip SSC to be able to read the next extension if there is one. */ bitsAvailable -= sscLen * 8; } } break; case ASCEXT_SAOC: if ((ascExtId == ASCEXT_SAOC) && (self->m_extensionAudioObjectType == AOT_SAOC)) break; if (FDKreadBits(bs, 1)) { /* saocPresent */ int saocscLen = FDKreadBits(bs, 8); bitsAvailable -= 8; if (saocscLen == 0xFF) { saocscLen += FDKreadBits(bs, 16); bitsAvailable -= 16; } FDKpushFor(bs, saocscLen); bitsAvailable -= saocscLen * 8; } break; default: /* Just ignore anything. */ return TRANSPORTDEC_OK; } } return TRANSPORTDEC_OK; } /* * API Functions */ void AudioSpecificConfig_Init(CSAudioSpecificConfig *asc) { FDKmemclear(asc, sizeof(CSAudioSpecificConfig)); /* Init all values that should not be zero. */ asc->m_aot = AOT_NONE; asc->m_samplingFrequencyIndex = 0xf; asc->m_epConfig = -1; asc->m_extensionAudioObjectType = AOT_NULL_OBJECT; CProgramConfig_Init(&asc->m_progrConfigElement); } TRANSPORTDEC_ERROR AudioSpecificConfig_Parse( CSAudioSpecificConfig *self, HANDLE_FDK_BITSTREAM bs, int fExplicitBackwardCompatible, CSTpCallBacks *cb, UCHAR configMode, UCHAR configChanged, AUDIO_OBJECT_TYPE m_aot) { TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK; UINT ascStartAnchor = FDKgetValidBits(bs); int frameLengthFlag = -1; AudioSpecificConfig_Init(self); self->configMode = configMode; self->AacConfigChanged = configChanged; self->SbrConfigChanged = configChanged; self->SacConfigChanged = configChanged; if (m_aot != AOT_NULL_OBJECT) { self->m_aot = m_aot; } else { self->m_aot = getAOT(bs); self->m_samplingFrequency = getSampleRate(bs, &self->m_samplingFrequencyIndex, 4); if (self->m_samplingFrequency <= 0 || (self->m_samplingFrequency > 96000 && self->m_aot != 39) || self->m_samplingFrequency > 4 * 96000) { return TRANSPORTDEC_PARSE_ERROR; } self->m_channelConfiguration = FDKreadBits(bs, 4); /* MPEG-04 standard ISO/IEC 14496-3: channelConfiguration == 0 is reserved in er_raw_data_block (table 4.19) and er_raw_data_block_eld (table 4.75) MPEG-04 conformance ISO/IEC 14496-4: channelConfiguration == 0 is not permitted for AOT_ER_AAC_LC, AOT_ER_AAC_LTP, AOT_ER_AAC_LD, AOT_ER_AAC_SCAL (chapter 6.6.4.1.2.1.1) */ if ((self->m_channelConfiguration == 0) && ((self->m_aot == AOT_ER_AAC_LC) || (self->m_aot == AOT_ER_AAC_LTP) || (self->m_aot == AOT_ER_AAC_LD) || (self->m_aot == AOT_ER_AAC_SCAL) || (self->m_aot == AOT_ER_AAC_ELD))) { return TRANSPORTDEC_UNSUPPORTED_FORMAT; } /* MPEG-04 conformance ISO/IEC 14496-4: channelConfiguration > 2 is not * permitted for AOT_AAC_SCAL and AOT_ER_AAC_SCAL (chapter 6.6.4.1.2.1.1) */ if ((self->m_channelConfiguration > 2) && ((self->m_aot == AOT_AAC_SCAL) || (self->m_aot == AOT_ER_AAC_SCAL))) { return TRANSPORTDEC_UNSUPPORTED_FORMAT; } /* SBR extension ( explicit non-backwards compatible mode ) */ self->m_sbrPresentFlag = 0; self->m_psPresentFlag = 0; if (self->m_aot == AOT_SBR || self->m_aot == AOT_PS) { self->m_extensionAudioObjectType = AOT_SBR; self->m_sbrPresentFlag = 1; if (self->m_aot == AOT_PS) { self->m_psPresentFlag = 1; } self->m_extensionSamplingFrequency = getSampleRate(bs, &self->m_extensionSamplingFrequencyIndex, 4); if (self->m_extensionSamplingFrequency == 0 || self->m_extensionSamplingFrequency > 96000) { return TRANSPORTDEC_PARSE_ERROR; } self->m_aot = getAOT(bs); switch (self->m_aot) { case AOT_AAC_LC: break; case AOT_ER_BSAC: break; default: return TRANSPORTDEC_UNSUPPORTED_FORMAT; } if (self->m_aot == AOT_ER_BSAC) { self->m_extensionChannelConfiguration = FDKreadBits(bs, 4); } } else { self->m_extensionAudioObjectType = AOT_NULL_OBJECT; } } /* Parse whatever specific configs */ switch (self->m_aot) { case AOT_AAC_LC: case AOT_AAC_SCAL: case AOT_ER_AAC_LC: case AOT_ER_AAC_LD: case AOT_ER_AAC_SCAL: case AOT_ER_BSAC: if ((ErrorStatus = GaSpecificConfig_Parse(&self->m_sc.m_gaSpecificConfig, self, bs, ascStartAnchor)) != TRANSPORTDEC_OK) { return (ErrorStatus); } frameLengthFlag = self->m_sc.m_gaSpecificConfig.m_frameLengthFlag; break; case AOT_MPEGS: if (cb->cbSsc != NULL) { if (cb->cbSsc(cb->cbSscData, bs, self->m_aot, self->m_samplingFrequency, self->m_samplesPerFrame, 1, -1, /* nTimeSlots: read from bitstream */ 0, /* don't know the length */ self->configMode, &self->SacConfigChanged)) { return TRANSPORTDEC_UNSUPPORTED_FORMAT; } } else { return TRANSPORTDEC_UNSUPPORTED_FORMAT; } break; case AOT_ER_AAC_ELD: if ((ErrorStatus = EldSpecificConfig_Parse(self, bs, cb)) != TRANSPORTDEC_OK) { return (ErrorStatus); } frameLengthFlag = self->m_sc.m_eldSpecificConfig.m_frameLengthFlag; self->m_sbrPresentFlag = self->m_sc.m_eldSpecificConfig.m_sbrPresentFlag; self->m_extensionSamplingFrequency = (self->m_sc.m_eldSpecificConfig.m_sbrSamplingRate + 1) * self->m_samplingFrequency; break; case AOT_USAC: if ((ErrorStatus = UsacConfig_Parse(self, bs, cb)) != TRANSPORTDEC_OK) { return (ErrorStatus); } break; default: return TRANSPORTDEC_UNSUPPORTED_FORMAT; } /* Frame length */ switch (self->m_aot) { case AOT_AAC_LC: case AOT_AAC_SCAL: case AOT_ER_AAC_LC: case AOT_ER_AAC_SCAL: case AOT_ER_BSAC: /*case AOT_USAC:*/ if (!frameLengthFlag) self->m_samplesPerFrame = 1024; else self->m_samplesPerFrame = 960; break; case AOT_ER_AAC_LD: if (!frameLengthFlag) self->m_samplesPerFrame = 512; else self->m_samplesPerFrame = 480; break; default: break; } switch (self->m_aot) { case AOT_ER_AAC_LC: case AOT_ER_AAC_LD: case AOT_ER_AAC_ELD: case AOT_ER_AAC_SCAL: case AOT_ER_CELP: case AOT_ER_HVXC: case AOT_ER_BSAC: self->m_epConfig = FDKreadBits(bs, 2); if (self->m_epConfig > 1) { return TRANSPORTDEC_UNSUPPORTED_FORMAT; // EPCONFIG; } break; default: break; } if (fExplicitBackwardCompatible && (self->m_aot == AOT_AAC_LC || self->m_aot == AOT_ER_AAC_LD || self->m_aot == AOT_ER_BSAC)) { ErrorStatus = AudioSpecificConfig_ExtensionParse(self, bs, cb); } /* Copy config() to asc->config[] buffer. */ if ((ErrorStatus == TRANSPORTDEC_OK) && (self->m_aot == AOT_USAC)) { INT configSize_bits = (INT)FDKgetValidBits(bs) - (INT)ascStartAnchor; StoreConfigAsBitstream(bs, configSize_bits, self->config, TP_USAC_MAX_CONFIG_LEN); self->configBits = fAbs(configSize_bits); } return (ErrorStatus); } static TRANSPORTDEC_ERROR Drm_xHEAACDecoderConfig( CSAudioSpecificConfig *asc, HANDLE_FDK_BITSTREAM hBs, int audioMode, CSTpCallBacks *cb /* use cb == NULL to signal config check only mode */ ) { TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK; CSUsacConfig *usc = &asc->m_sc.m_usacConfig; int elemIdx = 0; usc->element[elemIdx].m_stereoConfigIndex = 0; usc->m_usacNumElements = 1; /* Currently all extension elements are skipped -> only one SCE or CPE. */ switch (audioMode) { case 0: /* mono: ID_USAC_SCE */ usc->element[elemIdx].usacElementType = ID_USAC_SCE; usc->m_nUsacChannels = 1; usc->element[elemIdx].m_noiseFilling = FDKreadBits(hBs, 1); if (usc->m_sbrRatioIndex > 0) { if (cb == NULL) { return ErrorStatus; } if (cb->cbSbr != NULL) { usc->element[elemIdx].m_harmonicSBR = FDKreadBit(hBs); usc->element[elemIdx].m_interTes = FDKreadBit(hBs); usc->element[elemIdx].m_pvc = FDKreadBit(hBs); if (cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, asc->m_aot, ID_SCE, elemIdx, usc->element[elemIdx].m_harmonicSBR, usc->element[elemIdx].m_stereoConfigIndex, asc->configMode, &asc->SbrConfigChanged, 1)) { return ErrorStatus = TRANSPORTDEC_PARSE_ERROR; } } } break; case 2: /* stereo: ID_USAC_CPE */ usc->element[elemIdx].usacElementType = ID_USAC_CPE; usc->m_nUsacChannels = 2; usc->element[elemIdx].m_noiseFilling = FDKreadBits(hBs, 1); if (usc->m_sbrRatioIndex > 0) { usc->element[elemIdx].m_harmonicSBR = FDKreadBit(hBs); usc->element[elemIdx].m_interTes = FDKreadBit(hBs); usc->element[elemIdx].m_pvc = FDKreadBit(hBs); { INT bitsToSkip = skipSbrHeader(hBs, 1); /* read stereoConfigIndex */ usc->element[elemIdx].m_stereoConfigIndex = FDKreadBits(hBs, 2); /* rewind */ FDKpushBack(hBs, bitsToSkip + 2); } /* The application of the following tools is mutually exclusive per audio stream configuration (see clause 5.3.2, xHE-AAC codec configuration): - MPS212 parametric stereo tool with residual coding (stereoConfigIndex>1); and - QMF based Harmonic Transposer (harmonicSBR==1). */ if ((usc->element[elemIdx].m_stereoConfigIndex > 1) && usc->element[elemIdx].m_harmonicSBR) { return ErrorStatus = TRANSPORTDEC_PARSE_ERROR; } /* The 4:1 sbrRatio (sbrRatioIndex==1 in [11]) may only be employed: - in mono operation; or - in stereo operation if parametric stereo (MPS212) without residual coding is applied, i.e. if stereoConfigIndex==1 (see clause 5.3.2, xHE-AAC codec configuration). */ if ((usc->m_sbrRatioIndex == 1) && (usc->element[elemIdx].m_stereoConfigIndex != 1)) { return ErrorStatus = TRANSPORTDEC_PARSE_ERROR; } if (cb == NULL) { return ErrorStatus; } { MP4_ELEMENT_ID el_type = (usc->element[elemIdx].m_stereoConfigIndex == 1 || usc->element[elemIdx].m_stereoConfigIndex == 2) ? ID_SCE : ID_CPE; if (cb->cbSbr == NULL) return ErrorStatus = TRANSPORTDEC_UNKOWN_ERROR; if (cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, asc->m_aot, el_type, elemIdx, usc->element[elemIdx].m_harmonicSBR, usc->element[elemIdx].m_stereoConfigIndex, asc->configMode, &asc->SbrConfigChanged, 1)) { return ErrorStatus = TRANSPORTDEC_PARSE_ERROR; } } /*usc->element[elemIdx].m_stereoConfigIndex =*/FDKreadBits(hBs, 2); if (usc->element[elemIdx].m_stereoConfigIndex > 0) { if (cb->cbSsc != NULL) { int samplesPerFrame = asc->m_samplesPerFrame; if (usc->m_sbrRatioIndex == 1) samplesPerFrame <<= 2; if (usc->m_sbrRatioIndex == 2) samplesPerFrame = (samplesPerFrame * 8) / 3; if (usc->m_sbrRatioIndex == 3) samplesPerFrame <<= 1; ErrorStatus = (TRANSPORTDEC_ERROR)cb->cbSsc( cb->cbSscData, hBs, AOT_DRM_USAC, /* syntax differs from MPEG Mps212Config() */ asc->m_extensionSamplingFrequency, samplesPerFrame, usc->element[elemIdx].m_stereoConfigIndex, usc->m_coreSbrFrameLengthIndex, 0, /* don't know the length */ asc->configMode, &asc->SacConfigChanged); } else { /* ErrorStatus = TRANSPORTDEC_UNSUPPORTED_FORMAT; */ } } } break; default: return TRANSPORTDEC_PARSE_ERROR; } return ErrorStatus; } TRANSPORTDEC_ERROR Drm_xHEAACStaticConfig( CSAudioSpecificConfig *asc, HANDLE_FDK_BITSTREAM bs, int audioMode, CSTpCallBacks *cb /* use cb == NULL to signal config check only mode */ ) { int coreSbrFrameLengthIndexDrm = FDKreadBits(bs, 2); if (UsacConfig_SetCoreSbrFrameLengthIndex( asc, coreSbrFrameLengthIndexDrm + 1) != TRANSPORTDEC_OK) { return TRANSPORTDEC_PARSE_ERROR; } asc->m_channelConfiguration = (audioMode) ? 2 : 1; if (Drm_xHEAACDecoderConfig(asc, bs, audioMode, cb) != TRANSPORTDEC_OK) { return TRANSPORTDEC_PARSE_ERROR; } return TRANSPORTDEC_OK; } /* Mapping of DRM audio sampling rate field to MPEG usacSamplingFrequencyIndex */ const UCHAR mapSr2MPEGIdx[8] = { 0x1b, /* 9.6 kHz */ 0x09, /* 12.0 kHz */ 0x08, /* 16.0 kHz */ 0x17, /* 19.2 kHz */ 0x06, /* 24.0 kHz */ 0x05, /* 32.0 kHz */ 0x12, /* 38.4 kHz */ 0x03 /* 48.0 kHz */ }; TRANSPORTDEC_ERROR DrmRawSdcAudioConfig_Parse( CSAudioSpecificConfig *self, HANDLE_FDK_BITSTREAM bs, CSTpCallBacks *cb, /* use cb == NULL to signal config check only mode */ UCHAR configMode, UCHAR configChanged) { TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK; AudioSpecificConfig_Init(self); if ((INT)FDKgetValidBits(bs) < 16) { ErrorStatus = TRANSPORTDEC_PARSE_ERROR; goto bail; } else { /* DRM - Audio information data entity - type 9 - Short Id 2 bits (not part of the config buffer) - Stream Id 2 bits (not part of the config buffer) - audio coding 2 bits - SBR flag 1 bit - audio mode 2 bits - audio sampling rate 3 bits - text flag 1 bit - enhancement flag 1 bit - coder field 5 bits - rfa 1 bit */ int audioCoding, audioMode, cSamplingFreq, coderField, sfIdx, sbrFlag; self->configMode = configMode; self->AacConfigChanged = configChanged; self->SbrConfigChanged = configChanged; self->SacConfigChanged = configChanged; /* Read the SDC field */ audioCoding = FDKreadBits(bs, 2); sbrFlag = FDKreadBits(bs, 1); audioMode = FDKreadBits(bs, 2); cSamplingFreq = FDKreadBits(bs, 3); /* audio sampling rate */ FDKreadBits(bs, 2); /* Text and enhancement flag */ coderField = FDKreadBits(bs, 5); FDKreadBits(bs, 1); /* rfa */ /* Evaluate configuration and fill the ASC */ if (audioCoding == 3) { sfIdx = (int)mapSr2MPEGIdx[cSamplingFreq]; sbrFlag = 0; /* rfa */ } else { switch (cSamplingFreq) { case 0: /* 8 kHz */ sfIdx = 11; break; case 1: /* 12 kHz */ sfIdx = 9; break; case 2: /* 16 kHz */ sfIdx = 8; break; case 3: /* 24 kHz */ sfIdx = 6; break; case 5: /* 48 kHz */ sfIdx = 3; break; case 4: /* reserved */ case 6: /* reserved */ case 7: /* reserved */ default: ErrorStatus = TRANSPORTDEC_PARSE_ERROR; goto bail; } } self->m_samplingFrequencyIndex = sfIdx; self->m_samplingFrequency = SamplingRateTable[sfIdx]; if (sbrFlag) { UINT i; int tmp = -1; self->m_sbrPresentFlag = 1; self->m_extensionAudioObjectType = AOT_SBR; self->m_extensionSamplingFrequency = self->m_samplingFrequency << 1; for (i = 0; i < (sizeof(SamplingRateTable) / sizeof(SamplingRateTable[0])); i++) { if (SamplingRateTable[i] == self->m_extensionSamplingFrequency) { tmp = i; break; } } self->m_extensionSamplingFrequencyIndex = tmp; } switch (audioCoding) { case 0: /* AAC */ if ((coderField >> 2) && (audioMode != 1)) { self->m_aot = AOT_DRM_SURROUND; /* Set pseudo AOT for Drm Surround */ } else { self->m_aot = AOT_DRM_AAC; /* Set pseudo AOT for Drm AAC */ } switch (audioMode) { case 1: /* parametric stereo */ self->m_psPresentFlag = 1; FDK_FALLTHROUGH; case 0: /* mono */ self->m_channelConfiguration = 1; break; case 2: /* stereo */ self->m_channelConfiguration = 2; break; default: ErrorStatus = TRANSPORTDEC_PARSE_ERROR; goto bail; } self->m_vcb11Flag = 1; self->m_hcrFlag = 1; self->m_samplesPerFrame = 960; self->m_epConfig = 1; break; case 1: /* CELP */ self->m_aot = AOT_ER_CELP; self->m_channelConfiguration = 1; break; case 2: /* HVXC */ self->m_aot = AOT_ER_HVXC; self->m_channelConfiguration = 1; break; case 3: /* xHE-AAC */ { /* payload is MPEG conform -> no pseudo DRM AOT needed */ self->m_aot = AOT_USAC; } switch (audioMode) { case 0: /* mono */ case 2: /* stereo */ /* codec specific config 8n bits */ ErrorStatus = Drm_xHEAACStaticConfig(self, bs, audioMode, cb); break; default: ErrorStatus = TRANSPORTDEC_PARSE_ERROR; goto bail; } break; default: ErrorStatus = TRANSPORTDEC_PARSE_ERROR; self->m_aot = AOT_NONE; break; } if (self->m_psPresentFlag && !self->m_sbrPresentFlag) { ErrorStatus = TRANSPORTDEC_PARSE_ERROR; goto bail; } } bail: return (ErrorStatus); } fdk-aac-2.0.2/libMpegTPDec/src/tpdec_drm.cpp000066400000000000000000000136131404225310700204760ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* MPEG transport format decoder library ********************* Author(s): Christian Griebel Description: DRM transport stuff *******************************************************************************/ #include "tpdec_drm.h" #include "FDK_bitstream.h" void drmRead_CrcInit(HANDLE_DRM pDrm) /*!< pointer to drm crc info stucture */ { FDK_ASSERT(pDrm != NULL); FDKcrcInit(&pDrm->crcInfo, 0x001d, 0xFFFF, 8); } int drmRead_CrcStartReg( HANDLE_DRM pDrm, /*!< pointer to drm stucture */ HANDLE_FDK_BITSTREAM hBs, /*!< handle to current bit buffer structure */ int mBits /*!< number of bits in crc region */ ) { FDK_ASSERT(pDrm != NULL); FDKcrcReset(&pDrm->crcInfo); pDrm->crcReadValue = FDKreadBits(hBs, 8); return (FDKcrcStartReg(&pDrm->crcInfo, hBs, mBits)); } void drmRead_CrcEndReg( HANDLE_DRM pDrm, /*!< pointer to drm crc info stucture */ HANDLE_FDK_BITSTREAM hBs, /*!< handle to current bit buffer structure */ int reg /*!< crc region */ ) { FDK_ASSERT(pDrm != NULL); FDKcrcEndReg(&pDrm->crcInfo, hBs, reg); } TRANSPORTDEC_ERROR drmRead_CrcCheck(HANDLE_DRM pDrm) { TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK; USHORT crc; crc = FDKcrcGetCRC(&pDrm->crcInfo) ^ 0xFF; if (crc != pDrm->crcReadValue) { return (TRANSPORTDEC_CRC_ERROR); } return (ErrorStatus); } fdk-aac-2.0.2/libMpegTPDec/src/tpdec_drm.h000066400000000000000000000200121404225310700201320ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* MPEG transport format decoder library ********************* Author(s): Josef Hoepfl Description: DRM interface *******************************************************************************/ #ifndef TPDEC_DRM_H #define TPDEC_DRM_H #include "tpdec_lib.h" #include "FDK_crc.h" typedef struct { FDK_CRCINFO crcInfo; /* CRC state info */ USHORT crcReadValue; /* CRC value read from bitstream data */ } STRUCT_DRM; typedef STRUCT_DRM *HANDLE_DRM; /*! \brief Initialize DRM CRC The function initialzes the crc buffer and the crc lookup table. \return none */ void drmRead_CrcInit(HANDLE_DRM pDrm); /** * \brief Starts CRC region with a maximum number of bits * If mBits is positive zero padding will be used for CRC calculation, if * there are less than mBits bits available. If mBits is negative no zero * padding is done. If mBits is zero the memory for the buffer is * allocated dynamically, the number of bits is not limited. * * \param pDrm DRM data handle * \param hBs bitstream handle, on which the CRC region referes to * \param mBits max number of bits in crc region to be considered * * \return ID for the created region, -1 in case of an error */ int drmRead_CrcStartReg(HANDLE_DRM pDrm, HANDLE_FDK_BITSTREAM hBs, int mBits); /** * \brief Ends CRC region identified by reg * * \param pDrm DRM data handle * \param hBs bitstream handle, on which the CRC region referes to * \param reg CRC regions ID returned by drmRead_CrcStartReg() * * \return none */ void drmRead_CrcEndReg(HANDLE_DRM pDrm, HANDLE_FDK_BITSTREAM hBs, int reg); /** * \brief Check CRC * * Checks if the currently calculated CRC matches the CRC field read from the * bitstream Deletes all CRC regions. * * \param pDrm DRM data handle * * \return Returns 0 if they are identical otherwise 1 */ TRANSPORTDEC_ERROR drmRead_CrcCheck(HANDLE_DRM pDrm); /** * \brief Check if we have a valid DRM frame at the current bitbuffer position * * This function assumes enough bits in buffer for the current frame. * It reads out the header bits to prepare the bitbuffer for the decode loop. * In case the header bits show an invalid bitstream/frame, the whole frame is * skipped. * * \param pDrm DRM data handle which is filled with parsed DRM header data * \param bs handle of bitstream from whom the DRM header is read * * \return error status */ TRANSPORTDEC_ERROR drmRead_DecodeHeader(HANDLE_DRM pDrm, HANDLE_FDK_BITSTREAM bs); /** * \brief Parse a Drm specific SDC audio config from a given bitstream handle. * * \param pAsc A pointer to an allocated * CSAudioSpecificConfig struct. * \param hBs Bitstream handle. * \param cb A pointer to structure holding callback * information Note: A NULL pointer for cb can be used to signal a "Check Config * only functionality" * \param configMode Config modes: memory allocation mode or * config change detection mode * \param configChanged Indicates a config change * * \return Total element count including all SCE, CPE and LFE. */ TRANSPORTDEC_ERROR DrmRawSdcAudioConfig_Parse(CSAudioSpecificConfig *pAsc, HANDLE_FDK_BITSTREAM hBs, CSTpCallBacks *cb, const UCHAR configMode, const UCHAR configChanged); #endif /* TPDEC_DRM_H */ fdk-aac-2.0.2/libMpegTPDec/src/tpdec_latm.cpp000066400000000000000000000603171404225310700206540ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* MPEG transport format decoder library ********************* Author(s): Daniel Homm Description: *******************************************************************************/ #include "tpdec_latm.h" #include "FDK_bitstream.h" #define TPDEC_TRACKINDEX(p, l) (1 * (p) + (l)) static UINT CLatmDemux_GetValue(HANDLE_FDK_BITSTREAM bs) { UCHAR bytesForValue = 0, tmp = 0; int value = 0; bytesForValue = (UCHAR)FDKreadBits(bs, 2); for (UINT i = 0; i <= bytesForValue; i++) { value <<= 8; tmp = (UCHAR)FDKreadBits(bs, 8); value += tmp; } return value; } static TRANSPORTDEC_ERROR CLatmDemux_ReadAudioMuxElement( HANDLE_FDK_BITSTREAM bs, CLatmDemux *pLatmDemux, int m_muxConfigPresent, CSTpCallBacks *pTpDecCallbacks, CSAudioSpecificConfig *pAsc, int *pfConfigFound) { TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK; if (m_muxConfigPresent) { pLatmDemux->m_useSameStreamMux = FDKreadBits(bs, 1); if (!pLatmDemux->m_useSameStreamMux) { int i; UCHAR configChanged = 0; UCHAR configMode = 0; FDK_BITSTREAM bsAnchor; FDK_BITSTREAM bsAnchorDummyParse; if (!pLatmDemux->applyAsc) { bsAnchorDummyParse = *bs; pLatmDemux->newCfgHasAudioPreRoll = 0; /* do dummy-parsing of ASC to determine if there is an audioPreRoll */ configMode |= AC_CM_DET_CFG_CHANGE; if (TRANSPORTDEC_OK != (ErrorStatus = CLatmDemux_ReadStreamMuxConfig( bs, pLatmDemux, pTpDecCallbacks, pAsc, pfConfigFound, configMode, configChanged))) { goto bail; } /* Allow flushing only when audioPreroll functionality is enabled in * current and new config otherwise the new config can be applied * immediately. */ if (pAsc->m_sc.m_usacConfig.element[0] .extElement.usacExtElementHasAudioPreRoll && pLatmDemux->newCfgHasAudioPreRoll) { pLatmDemux->newCfgHasAudioPreRoll = 0; /* with audioPreRoll we must flush before applying new cfg */ pLatmDemux->applyAsc = 0; } else { *bs = bsAnchorDummyParse; pLatmDemux->applyAsc = 1; /* apply new config immediate */ } } if (pLatmDemux->applyAsc) { for (i = 0; i < 2; i++) { configMode = 0; if (i == 0) { configMode |= AC_CM_DET_CFG_CHANGE; bsAnchor = *bs; } else { configMode |= AC_CM_ALLOC_MEM; *bs = bsAnchor; } if (TRANSPORTDEC_OK != (ErrorStatus = CLatmDemux_ReadStreamMuxConfig( bs, pLatmDemux, pTpDecCallbacks, pAsc, pfConfigFound, configMode, configChanged))) { goto bail; } if (ErrorStatus == TRANSPORTDEC_OK) { if ((i == 0) && (pAsc->AacConfigChanged || pAsc->SbrConfigChanged || pAsc->SacConfigChanged)) { int errC; configChanged = 1; errC = pTpDecCallbacks->cbFreeMem(pTpDecCallbacks->cbFreeMemData, pAsc); if (errC != 0) { ErrorStatus = TRANSPORTDEC_PARSE_ERROR; goto bail; } } } } } } } /* If there was no configuration read, its not possible to parse * PayloadLengthInfo below. */ if (!*pfConfigFound) { ErrorStatus = TRANSPORTDEC_SYNC_ERROR; goto bail; } if (pLatmDemux->m_AudioMuxVersionA == 0) { /* Do only once per call, because parsing and decoding is done in-line. */ if (TRANSPORTDEC_OK != (ErrorStatus = CLatmDemux_ReadPayloadLengthInfo(bs, pLatmDemux))) { *pfConfigFound = 0; goto bail; } } else { /* audioMuxVersionA > 0 is reserved for future extensions */ ErrorStatus = TRANSPORTDEC_UNSUPPORTED_FORMAT; *pfConfigFound = 0; goto bail; } bail: if (ErrorStatus != TRANSPORTDEC_OK) { pLatmDemux->applyAsc = 1; } return (ErrorStatus); } TRANSPORTDEC_ERROR CLatmDemux_Read(HANDLE_FDK_BITSTREAM bs, CLatmDemux *pLatmDemux, TRANSPORT_TYPE tt, CSTpCallBacks *pTpDecCallbacks, CSAudioSpecificConfig *pAsc, int *pfConfigFound, const INT ignoreBufferFullness) { UINT cntBits; UINT cmpBufferFullness; UINT audioMuxLengthBytesLast = 0; TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK; cntBits = FDKgetValidBits(bs); if ((INT)cntBits < MIN_LATM_HEADERLENGTH) { return TRANSPORTDEC_NOT_ENOUGH_BITS; } if (TRANSPORTDEC_OK != (ErrorStatus = CLatmDemux_ReadAudioMuxElement( bs, pLatmDemux, (tt != TT_MP4_LATM_MCP0), pTpDecCallbacks, pAsc, pfConfigFound))) return (ErrorStatus); if (!ignoreBufferFullness) { cmpBufferFullness = 24 + audioMuxLengthBytesLast * 8 + pLatmDemux->m_linfo[0][0].m_bufferFullness * pAsc[TPDEC_TRACKINDEX(0, 0)].m_channelConfiguration * 32; /* evaluate buffer fullness */ if (pLatmDemux->m_linfo[0][0].m_bufferFullness != 0xFF) { if (!pLatmDemux->BufferFullnessAchieved) { if (cntBits < cmpBufferFullness) { /* condition for start of decoding is not fulfilled */ /* the current frame will not be decoded */ return TRANSPORTDEC_NOT_ENOUGH_BITS; } else { pLatmDemux->BufferFullnessAchieved = 1; } } } } return (ErrorStatus); } TRANSPORTDEC_ERROR CLatmDemux_ReadStreamMuxConfig( HANDLE_FDK_BITSTREAM bs, CLatmDemux *pLatmDemux, CSTpCallBacks *pTpDecCallbacks, CSAudioSpecificConfig *pAsc, int *pfConfigFound, UCHAR configMode, UCHAR configChanged) { CSAudioSpecificConfig ascDummy; /* the actual config is needed for flushing, after that new config can be parsed */ CSAudioSpecificConfig *pAscDummy; pAscDummy = &ascDummy; pLatmDemux->usacExplicitCfgChanged = 0; LATM_LAYER_INFO *p_linfo = NULL; TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK; UCHAR updateConfig[1 * 1] = {0}; pLatmDemux->m_AudioMuxVersion = FDKreadBits(bs, 1); if (pLatmDemux->m_AudioMuxVersion == 0) { pLatmDemux->m_AudioMuxVersionA = 0; } else { pLatmDemux->m_AudioMuxVersionA = FDKreadBits(bs, 1); } if (pLatmDemux->m_AudioMuxVersionA == 0) { if (pLatmDemux->m_AudioMuxVersion == 1) { pLatmDemux->m_taraBufferFullness = CLatmDemux_GetValue(bs); } pLatmDemux->m_allStreamsSameTimeFraming = FDKreadBits(bs, 1); pLatmDemux->m_noSubFrames = FDKreadBits(bs, 6) + 1; pLatmDemux->m_numProgram = FDKreadBits(bs, 4) + 1; if (pLatmDemux->m_numProgram > LATM_MAX_PROG) { ErrorStatus = TRANSPORTDEC_UNSUPPORTED_FORMAT; goto bail; } int idCnt = 0; for (UINT prog = 0; prog < pLatmDemux->m_numProgram; prog++) { pLatmDemux->m_numLayer[prog] = FDKreadBits(bs, 3) + 1; if (pLatmDemux->m_numLayer[prog] > LATM_MAX_LAYER) { ErrorStatus = TRANSPORTDEC_UNSUPPORTED_FORMAT; goto bail; } for (UINT lay = 0; lay < pLatmDemux->m_numLayer[prog]; lay++) { int useSameConfig; p_linfo = &pLatmDemux->m_linfo[prog][lay]; p_linfo->m_streamID = idCnt++; p_linfo->m_frameLengthInBits = 0; if ((prog == 0) && (lay == 0)) { useSameConfig = 0; } else { useSameConfig = FDKreadBits(bs, 1); } if (useSameConfig) { if (lay > 0) { FDKmemcpy(&pAsc[TPDEC_TRACKINDEX(prog, lay)], &pAsc[TPDEC_TRACKINDEX(prog, lay - 1)], sizeof(CSAudioSpecificConfig)); } else { ErrorStatus = TRANSPORTDEC_PARSE_ERROR; goto bail; } } else { UINT usacConfigLengthPrev = 0; UCHAR usacConfigPrev[TP_USAC_MAX_CONFIG_LEN]; if (!(pLatmDemux->applyAsc) && (pAsc[TPDEC_TRACKINDEX(prog, lay)].m_aot == AOT_USAC)) { usacConfigLengthPrev = (UINT)(pAsc[TPDEC_TRACKINDEX(prog, lay)] .m_sc.m_usacConfig.UsacConfigBits + 7) >> 3; /* store previous USAC config length */ if (usacConfigLengthPrev > TP_USAC_MAX_CONFIG_LEN) { ErrorStatus = TRANSPORTDEC_PARSE_ERROR; goto bail; } FDKmemclear(usacConfigPrev, TP_USAC_MAX_CONFIG_LEN); FDKmemcpy( usacConfigPrev, &pAsc[TPDEC_TRACKINDEX(prog, lay)].m_sc.m_usacConfig.UsacConfig, usacConfigLengthPrev); /* store previous USAC config */ } if (pLatmDemux->m_AudioMuxVersion == 1) { FDK_BITSTREAM tmpBs; INT ascLen = 0; ascLen = CLatmDemux_GetValue(bs); /* The ascLen could be wrong, so check if validBits<=bufBits*/ if (ascLen < 0 || ascLen > (INT)FDKgetValidBits(bs)) { ErrorStatus = TRANSPORTDEC_PARSE_ERROR; goto bail; } FDKsyncCache(bs); tmpBs = *bs; tmpBs.hBitBuf.ValidBits = ascLen; /* Read ASC */ if (pLatmDemux->applyAsc) { if (TRANSPORTDEC_OK != (ErrorStatus = AudioSpecificConfig_Parse( &pAsc[TPDEC_TRACKINDEX(prog, lay)], &tmpBs, 1, pTpDecCallbacks, configMode, configChanged, AOT_NULL_OBJECT))) goto bail; } else { if (TRANSPORTDEC_OK != (ErrorStatus = AudioSpecificConfig_Parse( pAscDummy, &tmpBs, 1, pTpDecCallbacks, configMode, configChanged, AOT_NULL_OBJECT))) goto bail; } /* The field p_linfo->m_ascLen could be wrong, so check if */ if (0 > (INT)FDKgetValidBits(&tmpBs)) { ErrorStatus = TRANSPORTDEC_PARSE_ERROR; goto bail; } FDKpushFor(bs, ascLen); /* position bitstream after ASC */ } else { /* Read ASC */ if (pLatmDemux->applyAsc) { if (TRANSPORTDEC_OK != (ErrorStatus = AudioSpecificConfig_Parse( &pAsc[TPDEC_TRACKINDEX(prog, lay)], bs, 0, pTpDecCallbacks, configMode, configChanged, AOT_NULL_OBJECT))) goto bail; } else { if (TRANSPORTDEC_OK != (ErrorStatus = AudioSpecificConfig_Parse( pAscDummy, bs, 0, pTpDecCallbacks, configMode, configChanged, AOT_NULL_OBJECT))) goto bail; } } if (!pLatmDemux->applyAsc) { updateConfig[TPDEC_TRACKINDEX(prog, lay)] = 0; } else { updateConfig[TPDEC_TRACKINDEX(prog, lay)] = 1; } if (!pLatmDemux->applyAsc) { if (pAscDummy[TPDEC_TRACKINDEX(prog, lay)].m_aot == AOT_USAC) { /* flush in case SMC has changed */ const UINT usacConfigLength = (UINT)(pAscDummy->m_sc.m_usacConfig.UsacConfigBits + 7) >> 3; if (usacConfigLength > TP_USAC_MAX_CONFIG_LEN) { ErrorStatus = TRANSPORTDEC_PARSE_ERROR; goto bail; } if (usacConfigLength != usacConfigLengthPrev) { FDKmemclear(&pAsc[TPDEC_TRACKINDEX(prog, lay)] .m_sc.m_usacConfig.UsacConfig, TP_USAC_MAX_CONFIG_LEN); FDKmemcpy(&pAsc[TPDEC_TRACKINDEX(prog, lay)] .m_sc.m_usacConfig.UsacConfig, &pAscDummy->m_sc.m_usacConfig.UsacConfig, usacConfigLength); /* store new USAC config */ pAsc[TPDEC_TRACKINDEX(prog, lay)] .m_sc.m_usacConfig.UsacConfigBits = pAscDummy->m_sc.m_usacConfig.UsacConfigBits; pLatmDemux->usacExplicitCfgChanged = 1; } else { if (FDKmemcmp(usacConfigPrev, pAscDummy->m_sc.m_usacConfig.UsacConfig, usacConfigLengthPrev)) { FDKmemclear(&pAsc[TPDEC_TRACKINDEX(prog, lay)] .m_sc.m_usacConfig.UsacConfig, TP_USAC_MAX_CONFIG_LEN); FDKmemcpy(&pAsc[TPDEC_TRACKINDEX(prog, lay)] .m_sc.m_usacConfig.UsacConfig, &pAscDummy->m_sc.m_usacConfig.UsacConfig, usacConfigLength); /* store new USAC config */ pAsc[TPDEC_TRACKINDEX(prog, lay)] .m_sc.m_usacConfig.UsacConfigBits = pAscDummy->m_sc.m_usacConfig.UsacConfigBits; pLatmDemux->usacExplicitCfgChanged = 1; } } if (pAscDummy[TPDEC_TRACKINDEX(prog, lay)] .m_sc.m_usacConfig.m_usacNumElements) { if (pAscDummy[TPDEC_TRACKINDEX(prog, lay)] .m_sc.m_usacConfig.element[0] .extElement.usacExtElementHasAudioPreRoll) { pLatmDemux->newCfgHasAudioPreRoll = 1; /* if dummy parsed cfg has audioPreRoll we first flush before applying new cfg */ } } } } } p_linfo->m_frameLengthType = FDKreadBits(bs, 3); switch (p_linfo->m_frameLengthType) { case 0: p_linfo->m_bufferFullness = FDKreadBits(bs, 8); if (!pLatmDemux->m_allStreamsSameTimeFraming) { if ((lay > 0) && (pAsc[TPDEC_TRACKINDEX(prog, lay)].m_aot == AOT_AAC_SCAL || pAsc[TPDEC_TRACKINDEX(prog, lay)].m_aot == AOT_ER_AAC_SCAL) && (pAsc[TPDEC_TRACKINDEX(prog, lay - 1)].m_aot == AOT_CELP || pAsc[TPDEC_TRACKINDEX(prog, lay - 1)].m_aot == AOT_ER_CELP)) { /* The layer maybe ignored later so read it anyway: */ /* coreFrameOffset = */ FDKreadBits(bs, 6); } } break; case 1: p_linfo->m_frameLengthInBits = FDKreadBits(bs, 9); break; case 3: case 4: case 5: /* CELP */ case 6: case 7: /* HVXC */ default: ErrorStatus = TRANSPORTDEC_PARSE_ERROR; goto bail; } /* switch framelengthtype*/ } /* layer loop */ } /* prog loop */ pLatmDemux->m_otherDataPresent = FDKreadBits(bs, 1); pLatmDemux->m_otherDataLength = 0; if (pLatmDemux->m_otherDataPresent) { if (pLatmDemux->m_AudioMuxVersion == 1) { pLatmDemux->m_otherDataLength = CLatmDemux_GetValue(bs); } else { int otherDataLenEsc = 0; do { pLatmDemux->m_otherDataLength <<= 8; // *= 256 otherDataLenEsc = FDKreadBits(bs, 1); pLatmDemux->m_otherDataLength += FDKreadBits(bs, 8); } while (otherDataLenEsc); } if (pLatmDemux->m_audioMuxLengthBytes < (pLatmDemux->m_otherDataLength >> 3)) { ErrorStatus = TRANSPORTDEC_PARSE_ERROR; goto bail; } } pLatmDemux->m_crcCheckPresent = FDKreadBits(bs, 1); if (pLatmDemux->m_crcCheckPresent) { FDKreadBits(bs, 8); } } else { /* audioMuxVersionA > 0 is reserved for future extensions */ ErrorStatus = TRANSPORTDEC_UNSUPPORTED_FORMAT; } /* Configure source decoder: */ if (ErrorStatus == TRANSPORTDEC_OK) { UINT prog; for (prog = 0; prog < pLatmDemux->m_numProgram; prog++) { UINT lay; for (lay = 0; lay < pLatmDemux->m_numLayer[prog]; lay++) { if (updateConfig[TPDEC_TRACKINDEX(prog, lay)] != 0) { int cbError; cbError = pTpDecCallbacks->cbUpdateConfig( pTpDecCallbacks->cbUpdateConfigData, &pAsc[TPDEC_TRACKINDEX(prog, lay)], pAsc[TPDEC_TRACKINDEX(prog, lay)].configMode, &pAsc[TPDEC_TRACKINDEX(prog, lay)].AacConfigChanged); if (cbError == TRANSPORTDEC_NEED_TO_RESTART) { *pfConfigFound = 0; ErrorStatus = TRANSPORTDEC_NEED_TO_RESTART; goto bail; } if (cbError != 0) { *pfConfigFound = 0; if (lay == 0) { ErrorStatus = TRANSPORTDEC_SYNC_ERROR; goto bail; } } else { *pfConfigFound = 1; } } else { *pfConfigFound = 1; } } } } bail: if (ErrorStatus != TRANSPORTDEC_OK) { UCHAR applyAsc = pLatmDemux->applyAsc; FDKmemclear(pLatmDemux, sizeof(CLatmDemux)); /* reset structure */ pLatmDemux->applyAsc = applyAsc; } else { /* no error and config parsing is finished */ if (configMode == AC_CM_ALLOC_MEM) pLatmDemux->applyAsc = 0; } return (ErrorStatus); } TRANSPORTDEC_ERROR CLatmDemux_ReadPayloadLengthInfo(HANDLE_FDK_BITSTREAM bs, CLatmDemux *pLatmDemux) { TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK; int totalPayloadBits = 0; if (pLatmDemux->m_allStreamsSameTimeFraming == 1) { FDK_ASSERT(pLatmDemux->m_numProgram <= LATM_MAX_PROG); for (UINT prog = 0; prog < pLatmDemux->m_numProgram; prog++) { FDK_ASSERT(pLatmDemux->m_numLayer[prog] <= LATM_MAX_LAYER); for (UINT lay = 0; lay < pLatmDemux->m_numLayer[prog]; lay++) { LATM_LAYER_INFO *p_linfo = &pLatmDemux->m_linfo[prog][lay]; switch (p_linfo->m_frameLengthType) { case 0: p_linfo->m_frameLengthInBits = CLatmDemux_ReadAuChunkLengthInfo(bs); totalPayloadBits += p_linfo->m_frameLengthInBits; break; case 3: case 5: case 7: default: return TRANSPORTDEC_PARSE_ERROR; // AAC_DEC_LATM_INVALIDFRAMELENGTHTYPE; } } } } else { ErrorStatus = TRANSPORTDEC_PARSE_ERROR; // AAC_DEC_LATM_TIMEFRAMING; } if (pLatmDemux->m_audioMuxLengthBytes > (UINT)0 && totalPayloadBits > (int)pLatmDemux->m_audioMuxLengthBytes * 8) { return TRANSPORTDEC_PARSE_ERROR; } return (ErrorStatus); } int CLatmDemux_ReadAuChunkLengthInfo(HANDLE_FDK_BITSTREAM bs) { UCHAR endFlag; int len = 0; do { UCHAR tmp = (UCHAR)FDKreadBits(bs, 8); endFlag = (tmp < 255); len += tmp; } while (endFlag == 0); len <<= 3; /* convert from bytes to bits */ return len; } UINT CLatmDemux_GetFrameLengthInBits(CLatmDemux *pLatmDemux, const UINT prog, const UINT layer) { UINT nFrameLenBits = 0; if (prog < pLatmDemux->m_numProgram) { if (layer < pLatmDemux->m_numLayer[prog]) { nFrameLenBits = pLatmDemux->m_linfo[prog][layer].m_frameLengthInBits; } } return nFrameLenBits; } UINT CLatmDemux_GetOtherDataPresentFlag(CLatmDemux *pLatmDemux) { return pLatmDemux->m_otherDataPresent ? 1 : 0; } UINT CLatmDemux_GetOtherDataLength(CLatmDemux *pLatmDemux) { return pLatmDemux->m_otherDataLength; } UINT CLatmDemux_GetNrOfSubFrames(CLatmDemux *pLatmDemux) { return pLatmDemux->m_noSubFrames; } UINT CLatmDemux_GetNrOfLayers(CLatmDemux *pLatmDemux, const UINT prog) { UINT numLayer = 0; if (prog < pLatmDemux->m_numProgram) { numLayer = pLatmDemux->m_numLayer[prog]; } return numLayer; } fdk-aac-2.0.2/libMpegTPDec/src/tpdec_latm.h000066400000000000000000000174511404225310700203220ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* MPEG transport format decoder library ********************* Author(s): Daniel Homm Description: *******************************************************************************/ #ifndef TPDEC_LATM_H #define TPDEC_LATM_H #include "tpdec_lib.h" #include "FDK_bitstream.h" #define MIN_LATM_HEADERLENGTH 9 #define MIN_LOAS_HEADERLENGTH MIN_LATM_HEADERLENGTH + 24 /* both in bits */ #define MIN_TP_BUF_SIZE_LOAS (8194) enum { LATM_MAX_PROG = 1, LATM_MAX_LAYER = 1, LATM_MAX_VAR_CHUNKS = 16, LATM_MAX_ID = 16 }; typedef struct { UINT m_frameLengthType; UINT m_bufferFullness; UINT m_streamID; UINT m_frameLengthInBits; } LATM_LAYER_INFO; typedef struct { LATM_LAYER_INFO m_linfo[LATM_MAX_PROG][LATM_MAX_LAYER]; UINT m_taraBufferFullness; UINT m_otherDataLength; UINT m_audioMuxLengthBytes; /* Length of LOAS payload */ UCHAR m_useSameStreamMux; UCHAR m_AudioMuxVersion; UCHAR m_AudioMuxVersionA; UCHAR m_allStreamsSameTimeFraming; UCHAR m_noSubFrames; UCHAR m_numProgram; UCHAR m_numLayer[LATM_MAX_PROG]; UCHAR m_otherDataPresent; UCHAR m_crcCheckPresent; SCHAR BufferFullnessAchieved; UCHAR usacExplicitCfgChanged; /* explicit config in case of USAC and LOAS/LATM must be compared to IPF cfg */ UCHAR applyAsc; /* apply ASC immediate without flushing */ UCHAR newCfgHasAudioPreRoll; /* the new (dummy parsed) config has an AudioPreRoll */ } CLatmDemux; int CLatmDemux_ReadAuChunkLengthInfo(HANDLE_FDK_BITSTREAM bs); TRANSPORTDEC_ERROR CLatmDemux_Read(HANDLE_FDK_BITSTREAM bs, CLatmDemux *pLatmDemux, TRANSPORT_TYPE tt, CSTpCallBacks *pTpDecCallbacks, CSAudioSpecificConfig *pAsc, int *pfConfigFound, const INT ignoreBufferFullness); /** * \brief Read StreamMuxConfig * \param bs bit stream handle as data source * \param pLatmDemux pointer to CLatmDemux struct of current LATM context * \param pTpDecCallbacks Call back structure for configuration callbacks * \param pAsc pointer to a ASC for configuration storage * \param pfConfigFound pointer to a flag which is set to 1 if a configuration * was found and processed successfully * \param configMode Config modes: memory allocation mode or config change * detection mode * \param configChanged Indicates a config change * \return error code */ TRANSPORTDEC_ERROR CLatmDemux_ReadStreamMuxConfig( HANDLE_FDK_BITSTREAM bs, CLatmDemux *pLatmDemux, CSTpCallBacks *pTpDecCallbacks, CSAudioSpecificConfig *pAsc, int *pfConfigFound, UCHAR configMode, UCHAR configChanged); TRANSPORTDEC_ERROR CLatmDemux_ReadPayloadLengthInfo(HANDLE_FDK_BITSTREAM bs, CLatmDemux *pLatmDemux); UINT CLatmDemux_GetFrameLengthInBits(CLatmDemux *pLatmDemux, const UINT prog, const UINT layer); UINT CLatmDemux_GetOtherDataPresentFlag(CLatmDemux *pLatmDemux); UINT CLatmDemux_GetOtherDataLength(CLatmDemux *pLatmDemux); UINT CLatmDemux_GetNrOfSubFrames(CLatmDemux *pLatmDemux); UINT CLatmDemux_GetNrOfLayers(CLatmDemux *pLatmDemux, const UINT program); #endif /* TPDEC_LATM_H */ fdk-aac-2.0.2/libMpegTPDec/src/tpdec_lib.cpp000066400000000000000000001701541404225310700204660ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* MPEG transport format decoder library ********************* Author(s): Manuel Jander Description: MPEG Transport decoder *******************************************************************************/ #include "tpdec_lib.h" /* library version */ #include "tp_version.h" #include "tp_data.h" #include "tpdec_adts.h" #include "tpdec_adif.h" #include "tpdec_latm.h" #include "tpdec_drm.h" #include "FDK_crc.h" #define MODULE_NAME "transportDec" typedef union { STRUCT_ADTS adts; CAdifHeader adif; CLatmDemux latm; STRUCT_DRM drm; } transportdec_parser_t; #define MHAS_CONFIG_PRESENT 0x001 #define MHAS_UI_PRESENT 0x002 struct TRANSPORTDEC { TRANSPORT_TYPE transportFmt; /*!< MPEG4 transportDec type. */ CSTpCallBacks callbacks; /*!< Struct holding callback and its data */ FDK_BITSTREAM bitStream[1]; /* Bitstream reader */ UCHAR *bsBuffer; /* Internal bitstreamd data buffer */ transportdec_parser_t parser; /* Format specific parser structs. */ CSAudioSpecificConfig asc[(1 * 1) + 1]; /* Audio specific config from the last config found. One additional CSAudioSpecificConfig is used temporarily for parsing. */ CCtrlCFGChange ctrlCFGChange[(1 * 1)]; /* Controls config change */ UINT globalFramePos; /* Global transport frame reference bit position. */ UINT accessUnitAnchor[1]; /* Current access unit start bit position. */ INT auLength[1]; /* Length of current access unit. */ INT numberOfRawDataBlocks; /* Current number of raw data blocks contained remaining from the current transport frame. */ UINT avgBitRate; /* Average bit rate used for frame loss estimation. */ UINT lastValidBufferFullness; /* Last valid buffer fullness value for frame loss estimation */ INT remainder; /* Reminder in division during lost access unit estimation. */ INT missingAccessUnits; /* Estimated missing access units. */ UINT burstPeriod; /* Data burst period in mili seconds. */ UINT holdOffFrames; /* Amount of frames that were already hold off due to buffer fullness condition not being met. */ UINT flags; /* Flags. */ INT targetLayout; /* CICP target layout. */ UINT *pLoudnessInfoSetPosition; /* Reference and start position (bits) and length (bytes) of loudnessInfoSet within rsv603daConfig. */ }; /* Flag bitmasks for "flags" member of struct TRANSPORTDEC */ #define TPDEC_SYNCOK 1 #define TPDEC_MINIMIZE_DELAY 2 #define TPDEC_IGNORE_BUFFERFULLNESS 4 #define TPDEC_EARLY_CONFIG 8 #define TPDEC_LOST_FRAMES_PENDING 16 #define TPDEC_CONFIG_FOUND 32 #define TPDEC_USE_ELEM_SKIPPING 64 /* force config/content change */ #define TPDEC_FORCE_CONFIG_CHANGE 1 #define TPDEC_FORCE_CONTENT_CHANGE 2 /* skip packet */ #define TPDEC_SKIP_PACKET 1 C_ALLOC_MEM(Ram_TransportDecoder, struct TRANSPORTDEC, 1) C_ALLOC_MEM(Ram_TransportDecoderBuffer, UCHAR, (8192 * 4)) HANDLE_TRANSPORTDEC transportDec_Open(const TRANSPORT_TYPE transportFmt, const UINT flags, const UINT nrOfLayers) { HANDLE_TRANSPORTDEC hInput; hInput = GetRam_TransportDecoder(0); if (hInput == NULL) { return NULL; } /* Init transportDec struct. */ hInput->transportFmt = transportFmt; switch (transportFmt) { case TT_MP4_ADIF: break; case TT_MP4_ADTS: if (flags & TP_FLAG_MPEG4) hInput->parser.adts.decoderCanDoMpeg4 = 1; else hInput->parser.adts.decoderCanDoMpeg4 = 0; adtsRead_CrcInit(&hInput->parser.adts); hInput->parser.adts.BufferFullnesStartFlag = 1; hInput->numberOfRawDataBlocks = 0; break; case TT_DRM: drmRead_CrcInit(&hInput->parser.drm); break; case TT_MP4_LATM_MCP0: case TT_MP4_LATM_MCP1: hInput->parser.latm.usacExplicitCfgChanged = 0; hInput->parser.latm.applyAsc = 1; break; case TT_MP4_LOAS: hInput->parser.latm.usacExplicitCfgChanged = 0; hInput->parser.latm.applyAsc = 1; break; case TT_MP4_RAW: break; default: FreeRam_TransportDecoder(&hInput); hInput = NULL; break; } if (hInput != NULL) { /* Create bitstream */ { hInput->bsBuffer = GetRam_TransportDecoderBuffer(0); if (hInput->bsBuffer == NULL) { transportDec_Close(&hInput); return NULL; } if (nrOfLayers > 1) { transportDec_Close(&hInput); return NULL; } for (UINT i = 0; i < nrOfLayers; i++) { FDKinitBitStream(&hInput->bitStream[i], hInput->bsBuffer, (8192 * 4), 0, BS_READER); } } hInput->burstPeriod = 0; } return hInput; } TRANSPORTDEC_ERROR transportDec_OutOfBandConfig(HANDLE_TRANSPORTDEC hTp, UCHAR *conf, const UINT length, UINT layer) { int i; TRANSPORTDEC_ERROR err = TRANSPORTDEC_OK; FDK_BITSTREAM bs; HANDLE_FDK_BITSTREAM hBs = &bs; int fConfigFound = 0; UCHAR configChanged = 0; UCHAR configMode = AC_CM_DET_CFG_CHANGE; UCHAR tmpConf[1024] = {0}; if (length > 1024) { return TRANSPORTDEC_UNSUPPORTED_FORMAT; } FDKmemcpy(tmpConf, conf, length); FDKinitBitStream(hBs, tmpConf, 1024, length << 3, BS_READER); for (i = 0; i < 2; i++) { if (i > 0) { FDKpushBack(hBs, (INT)length * 8 - (INT)FDKgetValidBits(hBs)); configMode = AC_CM_ALLOC_MEM; } /* config transport decoder */ switch (hTp->transportFmt) { case TT_MP4_LATM_MCP0: case TT_MP4_LATM_MCP1: case TT_MP4_LOAS: { if (layer != 0) { return TRANSPORTDEC_INVALID_PARAMETER; } CLatmDemux *pLatmDemux = &hTp->parser.latm; err = CLatmDemux_ReadStreamMuxConfig(hBs, pLatmDemux, &hTp->callbacks, hTp->asc, &fConfigFound, configMode, configChanged); if (err != TRANSPORTDEC_OK) { return err; } } break; default: fConfigFound = 1; err = AudioSpecificConfig_Parse(&hTp->asc[(1 * 1)], hBs, 1, &hTp->callbacks, configMode, configChanged, AOT_NULL_OBJECT); if (err == TRANSPORTDEC_OK) { int errC; hTp->asc[layer] = hTp->asc[(1 * 1)]; errC = hTp->callbacks.cbUpdateConfig( hTp->callbacks.cbUpdateConfigData, &hTp->asc[layer], hTp->asc[layer].configMode, &hTp->asc[layer].AacConfigChanged); if (errC != 0) { err = TRANSPORTDEC_PARSE_ERROR; } } break; case TT_DRM: fConfigFound = 1; err = DrmRawSdcAudioConfig_Parse(&hTp->asc[layer], hBs, &hTp->callbacks, configMode, configChanged); if (err == TRANSPORTDEC_OK) { int errC; errC = hTp->callbacks.cbUpdateConfig( hTp->callbacks.cbUpdateConfigData, &hTp->asc[layer], hTp->asc[layer].configMode, &hTp->asc[layer].AacConfigChanged); if (errC != 0) { err = TRANSPORTDEC_PARSE_ERROR; } } break; } if (err == TRANSPORTDEC_OK) { if ((i == 0) && (hTp->asc[layer].AacConfigChanged || hTp->asc[layer].SbrConfigChanged || hTp->asc[layer].SacConfigChanged)) { int errC; configChanged = 1; errC = hTp->callbacks.cbFreeMem(hTp->callbacks.cbFreeMemData, &hTp->asc[layer]); if (errC != 0) { err = TRANSPORTDEC_PARSE_ERROR; } } } } if (err == TRANSPORTDEC_OK && fConfigFound) { hTp->flags |= TPDEC_CONFIG_FOUND; } return err; } TRANSPORTDEC_ERROR transportDec_InBandConfig(HANDLE_TRANSPORTDEC hTp, UCHAR *newConfig, const UINT newConfigLength, const UCHAR buildUpStatus, UCHAR *configChanged, UINT layer, UCHAR *implicitExplicitCfgDiff) { int errC; FDK_BITSTREAM bs; HANDLE_FDK_BITSTREAM hBs = &bs; TRANSPORTDEC_ERROR err = TRANSPORTDEC_OK; int fConfigFound = 0; UCHAR configMode = AC_CM_ALLOC_MEM; *implicitExplicitCfgDiff = 0; FDK_ASSERT(hTp->asc->m_aot == AOT_USAC); FDKinitBitStream(hBs, newConfig, TP_USAC_MAX_CONFIG_LEN, newConfigLength << 3, BS_READER); if ((hTp->ctrlCFGChange[layer].flushStatus == TPDEC_FLUSH_OFF) && (hTp->ctrlCFGChange[layer].buildUpStatus != TPDEC_RSV60_BUILD_UP_IDLE_IN_BAND)) { if (hTp->asc->m_aot == AOT_USAC) { if ((UINT)(hTp->asc->m_sc.m_usacConfig.UsacConfigBits + 7) >> 3 == newConfigLength) { if (0 == FDKmemcmp(newConfig, hTp->asc->m_sc.m_usacConfig.UsacConfig, newConfigLength)) { if (hTp->parser.latm.usacExplicitCfgChanged) { /* configChange from LOAS/LATM parser */ hTp->parser.latm.usacExplicitCfgChanged = 0; hTp->ctrlCFGChange[layer].flushCnt = 0; hTp->ctrlCFGChange[layer].flushStatus = TPDEC_USAC_DASH_IPF_FLUSH_ON; hTp->ctrlCFGChange[layer].buildUpCnt = 0; hTp->ctrlCFGChange[layer].buildUpStatus = TPDEC_BUILD_UP_OFF; } else { *configChanged = 0; return err; } } else { *implicitExplicitCfgDiff = 1; } } else { *implicitExplicitCfgDiff = 1; } /* ISO/IEC 23003-3:2012/FDAM 3:2016(E) Annex F.2: explicit and implicit * config shall be identical. */ if (*implicitExplicitCfgDiff) { switch (hTp->transportFmt) { case TT_MP4_LATM_MCP0: case TT_MP4_LATM_MCP1: case TT_MP4_LOAS: /* reset decoder to initial state to achieve definite behavior after * error in config */ hTp->callbacks.cbFreeMem(hTp->callbacks.cbFreeMemData, &hTp->asc[layer]); hTp->parser.latm.usacExplicitCfgChanged = 0; hTp->parser.latm.applyAsc = 1; err = TRANSPORTDEC_PARSE_ERROR; goto bail; default: break; } } } } { if ((hTp->ctrlCFGChange[layer].flushStatus == TPDEC_FLUSH_OFF) && (hTp->ctrlCFGChange[layer].buildUpStatus != TPDEC_RSV60_BUILD_UP_IDLE_IN_BAND)) { hTp->ctrlCFGChange[layer].flushCnt = 0; hTp->ctrlCFGChange[layer].buildUpCnt = 0; hTp->ctrlCFGChange[layer].buildUpStatus = TPDEC_BUILD_UP_OFF; if (hTp->asc->m_aot == AOT_USAC) { hTp->ctrlCFGChange[layer].flushStatus = TPDEC_USAC_DASH_IPF_FLUSH_ON; } } if ((hTp->ctrlCFGChange[layer].flushStatus == TPDEC_RSV60_DASH_IPF_ATSC_FLUSH_ON) || (hTp->ctrlCFGChange[layer].flushStatus == TPDEC_USAC_DASH_IPF_FLUSH_ON)) { SCHAR counter = 0; if (hTp->asc->m_aot == AOT_USAC) { counter = TPDEC_USAC_NUM_CONFIG_CHANGE_FRAMES; } if (hTp->ctrlCFGChange[layer].flushCnt >= counter) { hTp->ctrlCFGChange[layer].flushCnt = 0; hTp->ctrlCFGChange[layer].flushStatus = TPDEC_FLUSH_OFF; hTp->ctrlCFGChange[layer].forceCfgChange = 0; if (hTp->asc->m_aot == AOT_USAC) { hTp->ctrlCFGChange[layer].buildUpCnt = TPDEC_USAC_NUM_CONFIG_CHANGE_FRAMES - 1; hTp->ctrlCFGChange[layer].buildUpStatus = TPDEC_USAC_BUILD_UP_ON; } } /* Activate flush mode. After that continue with build up mode in core */ if (hTp->callbacks.cbCtrlCFGChange(hTp->callbacks.cbCtrlCFGChangeData, &hTp->ctrlCFGChange[layer]) != 0) { err = TRANSPORTDEC_PARSE_ERROR; } if ((hTp->ctrlCFGChange[layer].flushStatus == TPDEC_RSV60_DASH_IPF_ATSC_FLUSH_ON) || (hTp->ctrlCFGChange[layer].flushStatus == TPDEC_USAC_DASH_IPF_FLUSH_ON)) { hTp->ctrlCFGChange[layer].flushCnt++; return err; } } if (hTp->asc->m_aot == AOT_USAC) { fConfigFound = 1; if (err == TRANSPORTDEC_OK) { *configChanged = 0; configMode = AC_CM_DET_CFG_CHANGE; for (int i = 0; i < 2; i++) { if (i > 0) { FDKpushBack(hBs, (INT)newConfigLength * 8 - (INT)FDKgetValidBits(hBs)); configMode = AC_CM_ALLOC_MEM; } /* config transport decoder */ err = AudioSpecificConfig_Parse( &hTp->asc[(1 * 1)], hBs, 0, &hTp->callbacks, configMode, *configChanged, hTp->asc[layer].m_aot); if (err == TRANSPORTDEC_OK) { hTp->asc[layer] = hTp->asc[(1 * 1)]; errC = hTp->callbacks.cbUpdateConfig( hTp->callbacks.cbUpdateConfigData, &hTp->asc[layer], hTp->asc[layer].configMode, &hTp->asc[layer].AacConfigChanged); if (errC != 0) { err = TRANSPORTDEC_PARSE_ERROR; } } if (err == TRANSPORTDEC_OK) { if ((i == 0) && (hTp->asc[layer].AacConfigChanged || hTp->asc[layer].SbrConfigChanged || hTp->asc[layer].SacConfigChanged)) { *configChanged = 1; errC = hTp->callbacks.cbFreeMem(hTp->callbacks.cbFreeMemData, &hTp->asc[layer]); if (errC != 0) { err = TRANSPORTDEC_PARSE_ERROR; } } } /* if an error is detected terminate config parsing to avoid that an * invalid config is accepted in the second pass */ if (err != TRANSPORTDEC_OK) { break; } } } } bail: /* save new config */ if (err == TRANSPORTDEC_OK) { if (hTp->asc->m_aot == AOT_USAC) { hTp->asc->m_sc.m_usacConfig.UsacConfigBits = newConfigLength << 3; FDKmemcpy(hTp->asc->m_sc.m_usacConfig.UsacConfig, newConfig, newConfigLength); /* in case of USAC reset transportDecoder variables here because * otherwise without IPF they are not reset */ hTp->ctrlCFGChange[layer].flushCnt = 0; hTp->ctrlCFGChange[layer].flushStatus = TPDEC_FLUSH_OFF; hTp->ctrlCFGChange[layer].buildUpCnt = 0; hTp->ctrlCFGChange[layer].buildUpStatus = TPDEC_BUILD_UP_OFF; } } else { hTp->numberOfRawDataBlocks = 0; /* If parsing error while config found, clear ctrlCFGChange-struct */ hTp->ctrlCFGChange[layer].flushCnt = 0; hTp->ctrlCFGChange[layer].flushStatus = TPDEC_FLUSH_OFF; hTp->ctrlCFGChange[layer].buildUpCnt = 0; hTp->ctrlCFGChange[layer].buildUpStatus = TPDEC_BUILD_UP_OFF; hTp->ctrlCFGChange[layer].cfgChanged = 0; hTp->ctrlCFGChange[layer].contentChanged = 0; hTp->ctrlCFGChange[layer].forceCfgChange = 0; hTp->callbacks.cbCtrlCFGChange(hTp->callbacks.cbCtrlCFGChangeData, &hTp->ctrlCFGChange[layer]); } } if (err == TRANSPORTDEC_OK && fConfigFound) { hTp->flags |= TPDEC_CONFIG_FOUND; } return err; } int transportDec_RegisterAscCallback(HANDLE_TRANSPORTDEC hTpDec, const cbUpdateConfig_t cbUpdateConfig, void *user_data) { if (hTpDec == NULL) { return -1; } hTpDec->callbacks.cbUpdateConfig = cbUpdateConfig; hTpDec->callbacks.cbUpdateConfigData = user_data; return 0; } int transportDec_RegisterFreeMemCallback(HANDLE_TRANSPORTDEC hTpDec, const cbFreeMem_t cbFreeMem, void *user_data) { if (hTpDec == NULL) { return -1; } hTpDec->callbacks.cbFreeMem = cbFreeMem; hTpDec->callbacks.cbFreeMemData = user_data; return 0; } int transportDec_RegisterCtrlCFGChangeCallback( HANDLE_TRANSPORTDEC hTpDec, const cbCtrlCFGChange_t cbCtrlCFGChange, void *user_data) { if (hTpDec == NULL) { return -1; } hTpDec->callbacks.cbCtrlCFGChange = cbCtrlCFGChange; hTpDec->callbacks.cbCtrlCFGChangeData = user_data; return 0; } int transportDec_RegisterSscCallback(HANDLE_TRANSPORTDEC hTpDec, const cbSsc_t cbSsc, void *user_data) { if (hTpDec == NULL) { return -1; } hTpDec->callbacks.cbSsc = cbSsc; hTpDec->callbacks.cbSscData = user_data; return 0; } int transportDec_RegisterSbrCallback(HANDLE_TRANSPORTDEC hTpDec, const cbSbr_t cbSbr, void *user_data) { if (hTpDec == NULL) { return -1; } hTpDec->callbacks.cbSbr = cbSbr; hTpDec->callbacks.cbSbrData = user_data; return 0; } int transportDec_RegisterUsacCallback(HANDLE_TRANSPORTDEC hTpDec, const cbUsac_t cbUsac, void *user_data) { if (hTpDec == NULL) { return -1; } hTpDec->callbacks.cbUsac = cbUsac; hTpDec->callbacks.cbUsacData = user_data; return 0; } int transportDec_RegisterUniDrcConfigCallback(HANDLE_TRANSPORTDEC hTpDec, const cbUniDrc_t cbUniDrc, void *user_data, UINT *pLoudnessInfoSetPosition) { if (hTpDec == NULL) { return -1; } hTpDec->callbacks.cbUniDrc = cbUniDrc; hTpDec->callbacks.cbUniDrcData = user_data; hTpDec->pLoudnessInfoSetPosition = pLoudnessInfoSetPosition; return 0; } TRANSPORTDEC_ERROR transportDec_FillData(const HANDLE_TRANSPORTDEC hTp, UCHAR *pBuffer, const UINT bufferSize, UINT *pBytesValid, const INT layer) { HANDLE_FDK_BITSTREAM hBs; if ((hTp == NULL) || (layer >= 1)) { return TRANSPORTDEC_INVALID_PARAMETER; } /* set bitbuffer shortcut */ hBs = &hTp->bitStream[layer]; if (TT_IS_PACKET(hTp->transportFmt)) { if (hTp->numberOfRawDataBlocks == 0) { FDKresetBitbuffer(hBs); FDKfeedBuffer(hBs, pBuffer, bufferSize, pBytesValid); if (*pBytesValid != 0) { return TRANSPORTDEC_TOO_MANY_BITS; } } } else { /* ... else feed bitbuffer with new stream data (append). */ if (*pBytesValid == 0) { /* nothing to do */ return TRANSPORTDEC_OK; } else { const int bytesValid = *pBytesValid; FDKfeedBuffer(hBs, pBuffer, bufferSize, pBytesValid); if (hTp->numberOfRawDataBlocks > 0) { hTp->globalFramePos += (bytesValid - *pBytesValid) * 8; hTp->accessUnitAnchor[layer] = FDKgetValidBits(hBs); } } } return TRANSPORTDEC_OK; } HANDLE_FDK_BITSTREAM transportDec_GetBitstream(const HANDLE_TRANSPORTDEC hTp, const UINT layer) { return &hTp->bitStream[layer]; } TRANSPORT_TYPE transportDec_GetFormat(const HANDLE_TRANSPORTDEC hTp) { return hTp->transportFmt; } INT transportDec_GetBufferFullness(const HANDLE_TRANSPORTDEC hTp) { INT bufferFullness = -1; switch (hTp->transportFmt) { case TT_MP4_ADTS: if (hTp->parser.adts.bs.adts_fullness != 0x7ff) { bufferFullness = hTp->parser.adts.bs.frame_length * 8 + hTp->parser.adts.bs.adts_fullness * 32 * getNumberOfEffectiveChannels( hTp->parser.adts.bs.channel_config); } break; case TT_MP4_LOAS: case TT_MP4_LATM_MCP0: case TT_MP4_LATM_MCP1: if (hTp->parser.latm.m_linfo[0][0].m_bufferFullness != 0xff) { bufferFullness = hTp->parser.latm.m_linfo[0][0].m_bufferFullness; } break; default: break; } return bufferFullness; } /** * \brief adjust bit stream position and the end of an access unit. * \param hTp transport decoder handle. * \return error code. */ static TRANSPORTDEC_ERROR transportDec_AdjustEndOfAccessUnit( HANDLE_TRANSPORTDEC hTp) { HANDLE_FDK_BITSTREAM hBs = &hTp->bitStream[0]; TRANSPORTDEC_ERROR err = TRANSPORTDEC_OK; switch (hTp->transportFmt) { case TT_MP4_ADIF: /* Do byte align at the end of raw_data_block() because UsacFrame() is not * byte aligned. */ FDKbyteAlign(hBs, hTp->accessUnitAnchor[0]); break; case TT_MP4_LOAS: case TT_MP4_LATM_MCP0: case TT_MP4_LATM_MCP1: if (hTp->numberOfRawDataBlocks == 0) { /* Do byte align at the end of AudioMuxElement. */ FDKbyteAlign(hBs, hTp->globalFramePos); /* Check global frame length */ if (hTp->transportFmt == TT_MP4_LOAS && hTp->parser.latm.m_audioMuxLengthBytes > 0) { int loasOffset; loasOffset = ((INT)hTp->parser.latm.m_audioMuxLengthBytes * 8 + (INT)FDKgetValidBits(hBs)) - (INT)hTp->globalFramePos; if (loasOffset != 0) { FDKpushBiDirectional(hBs, loasOffset); /* For ELD and other payloads there is an unknown amount of padding, so ignore unread bits, but throw an error only if too many bits where read. */ if (loasOffset < 0) { err = TRANSPORTDEC_PARSE_ERROR; } } } } break; case TT_MP4_ADTS: if (hTp->parser.adts.bs.protection_absent == 0) { int offset; /* Calculate offset to end of AU */ offset = hTp->parser.adts .rawDataBlockDist[hTp->parser.adts.bs.num_raw_blocks - hTp->numberOfRawDataBlocks] << 3; /* CAUTION: The PCE (if available) is declared to be a part of the * header! */ offset -= (INT)hTp->accessUnitAnchor[0] - (INT)FDKgetValidBits(hBs) + 16 + hTp->parser.adts.bs.num_pce_bits; FDKpushBiDirectional(hBs, offset); } if (hTp->parser.adts.bs.num_raw_blocks > 0 && hTp->parser.adts.bs.protection_absent == 0) { /* Note this CRC read currently happens twice because of * transportDec_CrcCheck() */ hTp->parser.adts.crcReadValue = FDKreadBits(hBs, 16); } if (hTp->numberOfRawDataBlocks == 0) { /* Check global frame length */ if (hTp->parser.adts.bs.protection_absent == 0) { int offset; offset = (hTp->parser.adts.bs.frame_length * 8 - ADTS_SYNCLENGTH + (INT)FDKgetValidBits(hBs)) - (INT)hTp->globalFramePos; if (offset != 0) { FDKpushBiDirectional(hBs, offset); } } } break; default: break; } return err; } /** * \brief Determine additional buffer fullness contraint due to burst data * reception. The parameter TPDEC_PARAM_BURSTPERIOD must have been set as a * precondition. * \param hTp transport decoder handle. * \param bufferFullness the buffer fullness value of the first frame to be * decoded. * \param bitsAvail the amount of available bits at the end of the first frame * to be decoded. * \return error code */ static TRANSPORTDEC_ERROR additionalHoldOffNeeded(HANDLE_TRANSPORTDEC hTp, INT bufferFullness, INT bitsAvail) { INT checkLengthBits, avgBitsPerFrame; INT maxAU; /* maximum number of frames per Master Frame */ INT samplesPerFrame = hTp->asc->m_samplesPerFrame; INT samplingFrequency = (INT)hTp->asc->m_samplingFrequency; if ((hTp->avgBitRate == 0) || (hTp->burstPeriod == 0)) { return TRANSPORTDEC_OK; } if ((samplesPerFrame == 0) || (samplingFrequency == 0)) { return TRANSPORTDEC_NOT_ENOUGH_BITS; } /* One Master Frame is sent every hTp->burstPeriod ms */ maxAU = hTp->burstPeriod * samplingFrequency + (samplesPerFrame * 1000 - 1); maxAU = maxAU / (samplesPerFrame * 1000); /* Subtract number of frames which were already held off. */ maxAU -= hTp->holdOffFrames; avgBitsPerFrame = hTp->avgBitRate * samplesPerFrame + (samplingFrequency - 1); avgBitsPerFrame = avgBitsPerFrame / samplingFrequency; /* Consider worst case of bufferFullness quantization. */ switch (hTp->transportFmt) { case TT_MP4_ADIF: case TT_MP4_ADTS: case TT_MP4_LOAS: case TT_MP4_LATM_MCP0: case TT_MP4_LATM_MCP1: bufferFullness += 31; break; default: /* added to avoid compiler warning */ break; /* added to avoid compiler warning */ } checkLengthBits = bufferFullness + (maxAU - 1) * avgBitsPerFrame; /* Check if buffer is big enough to fullfill buffer fullness condition */ if ((checkLengthBits /*+headerBits*/) > (((8192 * 4) << 3) - 7)) { return TRANSPORTDEC_SYNC_ERROR; } if (bitsAvail < checkLengthBits) { return TRANSPORTDEC_NOT_ENOUGH_BITS; } else { return TRANSPORTDEC_OK; } } static TRANSPORTDEC_ERROR transportDec_readHeader( HANDLE_TRANSPORTDEC hTp, HANDLE_FDK_BITSTREAM hBs, int syncLength, int ignoreBufferFullness, int *pRawDataBlockLength, int *pfTraverseMoreFrames, int *pSyncLayerFrameBits, int *pfConfigFound, int *pHeaderBits) { TRANSPORTDEC_ERROR err = TRANSPORTDEC_OK; int rawDataBlockLength = *pRawDataBlockLength; int fTraverseMoreFrames = (pfTraverseMoreFrames != NULL) ? *pfTraverseMoreFrames : 0; int syncLayerFrameBits = (pSyncLayerFrameBits != NULL) ? *pSyncLayerFrameBits : 0; int fConfigFound = (pfConfigFound != NULL) ? *pfConfigFound : 0; int startPos; startPos = (INT)FDKgetValidBits(hBs); switch (hTp->transportFmt) { case TT_MP4_ADTS: if (hTp->numberOfRawDataBlocks <= 0) { int i, errC; hTp->globalFramePos = FDKgetValidBits(hBs); UCHAR configChanged = 0; UCHAR configMode = AC_CM_DET_CFG_CHANGE; for (i = 0; i < 2; i++) { if (i > 0) { FDKpushBack(hBs, (INT)hTp->globalFramePos - (INT)FDKgetValidBits(hBs)); configMode = AC_CM_ALLOC_MEM; } /* Parse ADTS header */ err = adtsRead_DecodeHeader(&hTp->parser.adts, &hTp->asc[0], hBs, ignoreBufferFullness); if (err != TRANSPORTDEC_OK) { if (err != TRANSPORTDEC_NOT_ENOUGH_BITS) { err = TRANSPORTDEC_SYNC_ERROR; } } else { errC = hTp->callbacks.cbUpdateConfig( hTp->callbacks.cbUpdateConfigData, &hTp->asc[0], configMode, &configChanged); if (errC != 0) { if (errC == TRANSPORTDEC_NEED_TO_RESTART) { err = TRANSPORTDEC_NEED_TO_RESTART; goto bail; } else { err = TRANSPORTDEC_SYNC_ERROR; } } else { fConfigFound = 1; hTp->numberOfRawDataBlocks = hTp->parser.adts.bs.num_raw_blocks + 1; } } if (err == TRANSPORTDEC_OK) { if ((i == 0) && configChanged) { errC = hTp->callbacks.cbFreeMem(hTp->callbacks.cbFreeMemData, &hTp->asc[0]); if (errC != 0) { err = TRANSPORTDEC_PARSE_ERROR; } } } /* if an error is detected terminate config parsing to avoid that an * invalid config is accepted in the second pass */ if (err != TRANSPORTDEC_OK) { break; } } } else { /* Reset CRC because the next bits are the beginning of a * raw_data_block() */ FDKcrcReset(&hTp->parser.adts.crcInfo); hTp->parser.adts.bs.num_pce_bits = 0; } if (err == TRANSPORTDEC_OK) { hTp->numberOfRawDataBlocks--; rawDataBlockLength = adtsRead_GetRawDataBlockLength( &hTp->parser.adts, (hTp->parser.adts.bs.num_raw_blocks - hTp->numberOfRawDataBlocks)); if (rawDataBlockLength <= 0) { /* No further frame traversal possible. */ fTraverseMoreFrames = 0; } syncLayerFrameBits = (hTp->parser.adts.bs.frame_length << 3) - (startPos - (INT)FDKgetValidBits(hBs)) - syncLength; if (syncLayerFrameBits <= 0) { err = TRANSPORTDEC_SYNC_ERROR; } } else { hTp->numberOfRawDataBlocks = 0; } break; case TT_MP4_LOAS: if (hTp->numberOfRawDataBlocks <= 0) { syncLayerFrameBits = (INT)FDKreadBits(hBs, 13); hTp->parser.latm.m_audioMuxLengthBytes = syncLayerFrameBits; syncLayerFrameBits <<= 3; } FDK_FALLTHROUGH; case TT_MP4_LATM_MCP1: case TT_MP4_LATM_MCP0: if (hTp->numberOfRawDataBlocks <= 0) { hTp->globalFramePos = FDKgetValidBits(hBs); err = CLatmDemux_Read(hBs, &hTp->parser.latm, hTp->transportFmt, &hTp->callbacks, hTp->asc, &fConfigFound, ignoreBufferFullness); if (err != TRANSPORTDEC_OK) { if ((err != TRANSPORTDEC_NOT_ENOUGH_BITS) && !TPDEC_IS_FATAL_ERROR(err)) { err = TRANSPORTDEC_SYNC_ERROR; } } else { hTp->numberOfRawDataBlocks = CLatmDemux_GetNrOfSubFrames(&hTp->parser.latm); if (hTp->transportFmt == TT_MP4_LOAS) { syncLayerFrameBits -= startPos - (INT)FDKgetValidBits(hBs) - (13); if (syncLayerFrameBits <= 0) { err = TRANSPORTDEC_SYNC_ERROR; } } } } else { err = CLatmDemux_ReadPayloadLengthInfo(hBs, &hTp->parser.latm); if (err != TRANSPORTDEC_OK) { err = TRANSPORTDEC_SYNC_ERROR; } } if (err == TRANSPORTDEC_OK) { int layer; rawDataBlockLength = 0; for (layer = 0; layer < (int)CLatmDemux_GetNrOfLayers(&hTp->parser.latm, 0); layer += 1) { rawDataBlockLength += CLatmDemux_GetFrameLengthInBits(&hTp->parser.latm, 0, layer); } hTp->numberOfRawDataBlocks--; } else { hTp->numberOfRawDataBlocks = 0; } break; default: { syncLayerFrameBits = 0; } break; } bail: *pRawDataBlockLength = rawDataBlockLength; if (pHeaderBits != NULL) { *pHeaderBits += startPos - (INT)FDKgetValidBits(hBs); } for (int i = 0; i < (1 * 1); i++) { /* If parsing error while config found, clear ctrlCFGChange-struct */ if (hTp->ctrlCFGChange[i].cfgChanged && err != TRANSPORTDEC_OK) { hTp->numberOfRawDataBlocks = 0; hTp->ctrlCFGChange[i].flushCnt = 0; hTp->ctrlCFGChange[i].flushStatus = TPDEC_FLUSH_OFF; hTp->ctrlCFGChange[i].buildUpCnt = 0; hTp->ctrlCFGChange[i].buildUpStatus = TPDEC_BUILD_UP_OFF; hTp->ctrlCFGChange[i].cfgChanged = 0; hTp->ctrlCFGChange[i].contentChanged = 0; hTp->ctrlCFGChange[i].forceCfgChange = 0; hTp->callbacks.cbCtrlCFGChange(hTp->callbacks.cbCtrlCFGChangeData, &hTp->ctrlCFGChange[i]); } } if (pfConfigFound != NULL) { *pfConfigFound = fConfigFound; } if (pfTraverseMoreFrames != NULL) { *pfTraverseMoreFrames = fTraverseMoreFrames; } if (pSyncLayerFrameBits != NULL) { *pSyncLayerFrameBits = syncLayerFrameBits; } return err; } /* How many bits to advance for synchronization search. */ #define TPDEC_SYNCSKIP 8 static TRANSPORTDEC_ERROR synchronization(HANDLE_TRANSPORTDEC hTp, INT *pHeaderBits) { TRANSPORTDEC_ERROR err = TRANSPORTDEC_OK, errFirstFrame = TRANSPORTDEC_OK; HANDLE_FDK_BITSTREAM hBs = &hTp->bitStream[0]; INT syncLayerFrameBits = 0; /* Length of sync layer frame (i.e. LOAS) */ INT rawDataBlockLength = 0, rawDataBlockLengthPrevious; INT totalBits; INT headerBits = 0, headerBitsFirstFrame = 0, headerBitsPrevious; INT numFramesTraversed = 0, fTraverseMoreFrames, fConfigFound = (hTp->flags & TPDEC_CONFIG_FOUND), startPosFirstFrame = -1; INT numRawDataBlocksFirstFrame = 0, numRawDataBlocksPrevious, globalFramePosFirstFrame = 0, rawDataBlockLengthFirstFrame = 0; INT ignoreBufferFullness = hTp->flags & (TPDEC_LOST_FRAMES_PENDING | TPDEC_IGNORE_BUFFERFULLNESS | TPDEC_SYNCOK); UINT endTpFrameBitsPrevious = 0; /* Synch parameters */ INT syncLength; /* Length of sync word in bits */ UINT syncWord; /* Sync word to be found */ UINT syncMask; /* Mask for sync word (for adding one bit, so comprising one bit less) */ C_ALLOC_SCRATCH_START(contextFirstFrame, transportdec_parser_t, 1); totalBits = (INT)FDKgetValidBits(hBs); if (totalBits <= 0) { err = TRANSPORTDEC_NOT_ENOUGH_BITS; goto bail; } fTraverseMoreFrames = (hTp->flags & (TPDEC_MINIMIZE_DELAY | TPDEC_EARLY_CONFIG)) && !(hTp->flags & TPDEC_SYNCOK); /* Set transport specific sync parameters */ switch (hTp->transportFmt) { case TT_MP4_ADTS: syncWord = ADTS_SYNCWORD; syncLength = ADTS_SYNCLENGTH; break; case TT_MP4_LOAS: syncWord = 0x2B7; syncLength = 11; break; default: syncWord = 0; syncLength = 0; break; } syncMask = (1 << syncLength) - 1; do { INT bitsAvail = 0; /* Bits available in bitstream buffer */ INT checkLengthBits; /* Helper to check remaining bits and buffer boundaries */ UINT synch; /* Current sync word read from bitstream */ headerBitsPrevious = headerBits; bitsAvail = (INT)FDKgetValidBits(hBs); if (hTp->numberOfRawDataBlocks == 0) { /* search synchword */ FDK_ASSERT((bitsAvail % TPDEC_SYNCSKIP) == 0); if ((bitsAvail - syncLength) < TPDEC_SYNCSKIP) { err = TRANSPORTDEC_NOT_ENOUGH_BITS; headerBits = 0; } else { synch = FDKreadBits(hBs, syncLength); if (!(hTp->flags & TPDEC_SYNCOK)) { for (; (bitsAvail - syncLength) >= TPDEC_SYNCSKIP; bitsAvail -= TPDEC_SYNCSKIP) { if (synch == syncWord) { break; } synch = ((synch << TPDEC_SYNCSKIP) & syncMask) | FDKreadBits(hBs, TPDEC_SYNCSKIP); } } if (synch != syncWord) { /* No correct syncword found. */ err = TRANSPORTDEC_SYNC_ERROR; } else { err = TRANSPORTDEC_OK; } headerBits = syncLength; } } else { headerBits = 0; } /* Save previous raw data block data */ rawDataBlockLengthPrevious = rawDataBlockLength; numRawDataBlocksPrevious = hTp->numberOfRawDataBlocks; /* Parse transport header (raw data block granularity) */ if (err == TRANSPORTDEC_OK) { err = transportDec_readHeader(hTp, hBs, syncLength, ignoreBufferFullness, &rawDataBlockLength, &fTraverseMoreFrames, &syncLayerFrameBits, &fConfigFound, &headerBits); if (headerBits > bitsAvail) { err = (headerBits < (INT)hBs->hBitBuf.bufBits) ? TRANSPORTDEC_NOT_ENOUGH_BITS : TRANSPORTDEC_SYNC_ERROR; } if (TPDEC_IS_FATAL_ERROR(err)) { /* Rewind - TPDEC_SYNCSKIP, in order to look for a synch one bit ahead * next time. Ensure that the bit amount lands at a multiple of * TPDEC_SYNCSKIP. */ FDKpushBiDirectional( hBs, -headerBits + TPDEC_SYNCSKIP + (bitsAvail % TPDEC_SYNCSKIP)); goto bail; } } bitsAvail -= headerBits; checkLengthBits = syncLayerFrameBits; /* Check if the whole frame would fit the bitstream buffer */ if (err == TRANSPORTDEC_OK) { if ((checkLengthBits + headerBits) > (((8192 * 4) << 3) - 7)) { /* We assume that the size of the transport bit buffer has been chosen to meet all system requirements, thus this condition is considered a synchronisation error. */ err = TRANSPORTDEC_SYNC_ERROR; } else { if (bitsAvail < checkLengthBits) { err = TRANSPORTDEC_NOT_ENOUGH_BITS; } } } if (err == TRANSPORTDEC_NOT_ENOUGH_BITS) { break; } if (err == TRANSPORTDEC_SYNC_ERROR) { int bits; /* Enforce re-sync of transport headers. */ hTp->numberOfRawDataBlocks = 0; /* Ensure that the bit amount lands at a multiple of TPDEC_SYNCSKIP */ bits = (bitsAvail + headerBits) % TPDEC_SYNCSKIP; /* Rewind - TPDEC_SYNCSKIP, in order to look for a synch one bit ahead * next time. */ FDKpushBiDirectional(hBs, -(headerBits - TPDEC_SYNCSKIP) + bits); headerBits = 0; } /* Frame traversal */ if (fTraverseMoreFrames) { /* Save parser context for early config discovery "rewind all frames" */ if ((hTp->flags & TPDEC_EARLY_CONFIG) && !(hTp->flags & TPDEC_MINIMIZE_DELAY)) { /* ignore buffer fullness if just traversing additional frames for ECD */ ignoreBufferFullness = 1; /* Save context in order to return later */ if (err == TRANSPORTDEC_OK && startPosFirstFrame == -1) { startPosFirstFrame = FDKgetValidBits(hBs); numRawDataBlocksFirstFrame = hTp->numberOfRawDataBlocks; globalFramePosFirstFrame = hTp->globalFramePos; rawDataBlockLengthFirstFrame = rawDataBlockLength; headerBitsFirstFrame = headerBits; errFirstFrame = err; FDKmemcpy(contextFirstFrame, &hTp->parser, sizeof(transportdec_parser_t)); } /* Break when config was found or it is not possible anymore to find a * config */ if (startPosFirstFrame != -1 && (fConfigFound || err != TRANSPORTDEC_OK)) { /* In case of ECD and sync error, do not rewind anywhere. */ if (err == TRANSPORTDEC_SYNC_ERROR) { startPosFirstFrame = -1; fConfigFound = 0; numFramesTraversed = 0; } break; } } if (err == TRANSPORTDEC_OK) { FDKpushFor(hBs, rawDataBlockLength); numFramesTraversed++; endTpFrameBitsPrevious = (INT)FDKgetValidBits(hBs); /* Ignore error here itentionally. */ transportDec_AdjustEndOfAccessUnit(hTp); endTpFrameBitsPrevious -= FDKgetValidBits(hBs); } } } while (fTraverseMoreFrames || (err == TRANSPORTDEC_SYNC_ERROR && !(hTp->flags & TPDEC_SYNCOK))); /* Restore context in case of ECD frame traversal */ if (startPosFirstFrame != -1 && (fConfigFound || err != TRANSPORTDEC_OK)) { FDKpushBiDirectional(hBs, FDKgetValidBits(hBs) - startPosFirstFrame); FDKmemcpy(&hTp->parser, contextFirstFrame, sizeof(transportdec_parser_t)); hTp->numberOfRawDataBlocks = numRawDataBlocksFirstFrame; hTp->globalFramePos = globalFramePosFirstFrame; rawDataBlockLength = rawDataBlockLengthFirstFrame; headerBits = headerBitsFirstFrame; err = errFirstFrame; numFramesTraversed = 0; } /* Additional burst data mode buffer fullness check. */ if (!(hTp->flags & (TPDEC_LOST_FRAMES_PENDING | TPDEC_IGNORE_BUFFERFULLNESS | TPDEC_SYNCOK)) && err == TRANSPORTDEC_OK) { err = additionalHoldOffNeeded(hTp, transportDec_GetBufferFullness(hTp), (INT)FDKgetValidBits(hBs) - syncLayerFrameBits); if (err == TRANSPORTDEC_NOT_ENOUGH_BITS) { hTp->holdOffFrames++; } } /* Rewind for retry because of not enough bits */ if (err == TRANSPORTDEC_NOT_ENOUGH_BITS) { FDKpushBack(hBs, headerBits); hTp->numberOfRawDataBlocks = numRawDataBlocksPrevious; headerBits = 0; rawDataBlockLength = rawDataBlockLengthPrevious; } else { /* reset hold off frame counter */ hTp->holdOffFrames = 0; } /* Return to last good frame in case of frame traversal but not ECD. */ if (numFramesTraversed > 0) { FDKpushBack(hBs, rawDataBlockLengthPrevious + endTpFrameBitsPrevious); if (err != TRANSPORTDEC_OK) { hTp->numberOfRawDataBlocks = numRawDataBlocksPrevious; headerBits = headerBitsPrevious; rawDataBlockLength = rawDataBlockLengthPrevious; } err = TRANSPORTDEC_OK; } bail: hTp->auLength[0] = rawDataBlockLength; /* Detect pointless TRANSPORTDEC_NOT_ENOUGH_BITS error case, where the bit buffer is already full, or no new burst packet fits. Recover by advancing the bit buffer. */ if ((totalBits > 0) && (TRANSPORTDEC_NOT_ENOUGH_BITS == err) && (FDKgetValidBits(hBs) >= (((8192 * 4) * 8 - ((hTp->avgBitRate * hTp->burstPeriod) / 1000)) - 7))) { FDKpushFor(hBs, TPDEC_SYNCSKIP); err = TRANSPORTDEC_SYNC_ERROR; } if (err == TRANSPORTDEC_OK) { hTp->flags |= TPDEC_SYNCOK; } if (fConfigFound) { hTp->flags |= TPDEC_CONFIG_FOUND; } if (pHeaderBits != NULL) { *pHeaderBits = headerBits; } if (err == TRANSPORTDEC_SYNC_ERROR) { hTp->flags &= ~TPDEC_SYNCOK; } C_ALLOC_SCRATCH_END(contextFirstFrame, transportdec_parser_t, 1); return err; } /** * \brief Synchronize to stream and estimate the amount of missing access units * due to a current synchronization error in case of constant average bit rate. */ static TRANSPORTDEC_ERROR transportDec_readStream(HANDLE_TRANSPORTDEC hTp, const UINT layer) { TRANSPORTDEC_ERROR error = TRANSPORTDEC_OK; HANDLE_FDK_BITSTREAM hBs = &hTp->bitStream[layer]; INT headerBits; INT bitDistance, bfDelta; /* Obtain distance to next synch word */ bitDistance = (INT)FDKgetValidBits(hBs); error = synchronization(hTp, &headerBits); bitDistance -= (INT)FDKgetValidBits(hBs); FDK_ASSERT(bitDistance >= 0); INT nAU = -1; if (error == TRANSPORTDEC_SYNC_ERROR || (hTp->flags & TPDEC_LOST_FRAMES_PENDING)) { /* Check if estimating lost access units is feasible. */ if (hTp->avgBitRate > 0 && hTp->asc[0].m_samplesPerFrame > 0 && hTp->asc[0].m_samplingFrequency > 0) { if (error == TRANSPORTDEC_OK) { int aj; aj = transportDec_GetBufferFullness(hTp); if (aj > 0) { bfDelta = aj; } else { bfDelta = 0; } /* sync was ok: last of a series of bad access units. */ hTp->flags &= ~TPDEC_LOST_FRAMES_PENDING; /* Add up bitDistance until end of the current frame. Later we substract this frame from the grand total, since this current successfully synchronized frame should not be skipped of course; but it must be accounted into the bufferfulness math. */ bitDistance += hTp->auLength[0]; } else { if (!(hTp->flags & TPDEC_LOST_FRAMES_PENDING)) { /* sync not ok: one of many bad access units. */ hTp->flags |= TPDEC_LOST_FRAMES_PENDING; bfDelta = -(INT)hTp->lastValidBufferFullness; } else { bfDelta = 0; } } { int num, denom; /* Obtain estimate of number of lost frames */ num = (INT)hTp->asc[0].m_samplingFrequency * (bfDelta + bitDistance) + hTp->remainder; denom = hTp->avgBitRate * hTp->asc[0].m_samplesPerFrame; if (num > 0) { nAU = num / denom; hTp->remainder = num % denom; } else { hTp->remainder = num; } if (error == TRANSPORTDEC_OK) { /* Final adjustment of remainder, taken -1 into account because current frame should not be skipped, thus substract -1 or do nothing instead of +1-1 accordingly. */ if ((denom - hTp->remainder) >= hTp->remainder) { nAU--; } if (nAU < 0) { /* There was one frame too much concealed, so unfortunately we will * have to skip one good frame. */ transportDec_EndAccessUnit(hTp); error = synchronization(hTp, &headerBits); nAU = -1; } hTp->remainder = 0; /* Enforce last missed frames to be concealed. */ if (nAU > 0) { FDKpushBack(hBs, headerBits); } } } } } /* Be sure that lost frames are handled correctly. This is necessary due to some sync error sequences where later it turns out that there is not enough data, but the bits upto the sync word are discarded, thus causing a value of nAU > 0 */ if (nAU > 0) { error = TRANSPORTDEC_SYNC_ERROR; } hTp->missingAccessUnits = nAU; return error; } /* returns error code */ TRANSPORTDEC_ERROR transportDec_ReadAccessUnit(const HANDLE_TRANSPORTDEC hTp, const UINT layer) { TRANSPORTDEC_ERROR err = TRANSPORTDEC_OK; HANDLE_FDK_BITSTREAM hBs; if (!hTp) { return TRANSPORTDEC_INVALID_PARAMETER; } hBs = &hTp->bitStream[layer]; if ((INT)FDKgetValidBits(hBs) <= 0) { /* This is only relevant for RAW and ADIF cases. * For streaming formats err will get overwritten. */ err = TRANSPORTDEC_NOT_ENOUGH_BITS; hTp->numberOfRawDataBlocks = 0; } switch (hTp->transportFmt) { case TT_MP4_ADIF: /* Read header if not already done */ if (!(hTp->flags & TPDEC_CONFIG_FOUND)) { int i; CProgramConfig *pce; INT bsStart = FDKgetValidBits(hBs); UCHAR configChanged = 0; UCHAR configMode = AC_CM_DET_CFG_CHANGE; for (i = 0; i < 2; i++) { if (i > 0) { FDKpushBack(hBs, bsStart - (INT)FDKgetValidBits(hBs)); configMode = AC_CM_ALLOC_MEM; } AudioSpecificConfig_Init(&hTp->asc[0]); pce = &hTp->asc[0].m_progrConfigElement; err = adifRead_DecodeHeader(&hTp->parser.adif, pce, hBs); if (err) goto bail; /* Map adif header to ASC */ hTp->asc[0].m_aot = (AUDIO_OBJECT_TYPE)(pce->Profile + 1); hTp->asc[0].m_samplingFrequencyIndex = pce->SamplingFrequencyIndex; hTp->asc[0].m_samplingFrequency = SamplingRateTable[pce->SamplingFrequencyIndex]; hTp->asc[0].m_channelConfiguration = 0; hTp->asc[0].m_samplesPerFrame = 1024; hTp->avgBitRate = hTp->parser.adif.BitRate; /* Call callback to decoder. */ { int errC; errC = hTp->callbacks.cbUpdateConfig( hTp->callbacks.cbUpdateConfigData, &hTp->asc[0], configMode, &configChanged); if (errC == 0) { hTp->flags |= TPDEC_CONFIG_FOUND; } else { err = TRANSPORTDEC_PARSE_ERROR; goto bail; } } if (err == TRANSPORTDEC_OK) { if ((i == 0) && configChanged) { int errC; errC = hTp->callbacks.cbFreeMem(hTp->callbacks.cbFreeMemData, &hTp->asc[0]); if (errC != 0) { err = TRANSPORTDEC_PARSE_ERROR; } } } } } hTp->auLength[layer] = -1; /* Access Unit data length is unknown. */ break; case TT_MP4_RAW: case TT_DRM: /* One Access Unit was filled into buffer. So get the length out of the buffer. */ hTp->auLength[layer] = FDKgetValidBits(hBs); hTp->flags |= TPDEC_SYNCOK; break; case TT_MP4_LATM_MCP0: case TT_MP4_LATM_MCP1: if (err == TRANSPORTDEC_OK) { int fConfigFound = hTp->flags & TPDEC_CONFIG_FOUND; err = transportDec_readHeader(hTp, hBs, 0, 1, &hTp->auLength[layer], NULL, NULL, &fConfigFound, NULL); if (fConfigFound) { hTp->flags |= TPDEC_CONFIG_FOUND; } } break; case TT_MP4_ADTS: case TT_MP4_LOAS: err = transportDec_readStream(hTp, layer); break; default: err = TRANSPORTDEC_UNSUPPORTED_FORMAT; break; } if (err == TRANSPORTDEC_OK) { hTp->accessUnitAnchor[layer] = FDKgetValidBits(hBs); } else { hTp->accessUnitAnchor[layer] = 0; } bail: return err; } TRANSPORTDEC_ERROR transportDec_GetAsc(const HANDLE_TRANSPORTDEC hTp, const UINT layer, CSAudioSpecificConfig *asc) { TRANSPORTDEC_ERROR err = TRANSPORTDEC_OK; if (hTp != NULL) { *asc = hTp->asc[layer]; err = TRANSPORTDEC_OK; } else { err = TRANSPORTDEC_INVALID_PARAMETER; } return err; } INT transportDec_GetAuBitsRemaining(const HANDLE_TRANSPORTDEC hTp, const UINT layer) { INT bits; if (hTp->accessUnitAnchor[layer] > 0 && hTp->auLength[layer] > 0) { bits = (INT)FDKgetValidBits(&hTp->bitStream[layer]); if (bits >= 0) { bits = hTp->auLength[layer] - ((INT)hTp->accessUnitAnchor[layer] - bits); } } else { bits = FDKgetValidBits(&hTp->bitStream[layer]); } return bits; } INT transportDec_GetAuBitsTotal(const HANDLE_TRANSPORTDEC hTp, const UINT layer) { return hTp->auLength[layer]; } TRANSPORTDEC_ERROR transportDec_GetMissingAccessUnitCount( INT *pNAccessUnits, HANDLE_TRANSPORTDEC hTp) { *pNAccessUnits = hTp->missingAccessUnits; return TRANSPORTDEC_OK; } /* Inform the transportDec layer that reading of access unit has finished. */ TRANSPORTDEC_ERROR transportDec_EndAccessUnit(HANDLE_TRANSPORTDEC hTp) { TRANSPORTDEC_ERROR err = TRANSPORTDEC_OK; switch (hTp->transportFmt) { case TT_MP4_LOAS: case TT_MP4_LATM_MCP0: case TT_MP4_LATM_MCP1: { HANDLE_FDK_BITSTREAM hBs = &hTp->bitStream[0]; if (hTp->numberOfRawDataBlocks == 0) { /* Read other data if available. */ if (CLatmDemux_GetOtherDataPresentFlag(&hTp->parser.latm)) { int otherDataLen = CLatmDemux_GetOtherDataLength(&hTp->parser.latm); if ((INT)FDKgetValidBits(hBs) >= otherDataLen) { FDKpushFor(hBs, otherDataLen); } else { /* Do byte align at the end of AudioMuxElement. */ if (hTp->numberOfRawDataBlocks == 0) { FDKbyteAlign(hBs, hTp->globalFramePos); } return TRANSPORTDEC_NOT_ENOUGH_BITS; } } } else { /* If bit buffer has not more bits but hTp->numberOfRawDataBlocks > 0 then too many bits were read and obviously no more RawDataBlocks can be read. Set numberOfRawDataBlocks to zero to attempt a new sync attempt. */ if ((INT)FDKgetValidBits(hBs) <= 0) { hTp->numberOfRawDataBlocks = 0; } } } break; default: break; } err = transportDec_AdjustEndOfAccessUnit(hTp); switch (hTp->transportFmt) { default: break; } return err; } TRANSPORTDEC_ERROR transportDec_SetParam(const HANDLE_TRANSPORTDEC hTp, const TPDEC_PARAM param, const INT value) { TRANSPORTDEC_ERROR error = TRANSPORTDEC_OK; if (hTp == NULL) { return TRANSPORTDEC_INVALID_PARAMETER; } switch (param) { case TPDEC_PARAM_MINIMIZE_DELAY: if (value) { hTp->flags |= TPDEC_MINIMIZE_DELAY; } else { hTp->flags &= ~TPDEC_MINIMIZE_DELAY; } break; case TPDEC_PARAM_EARLY_CONFIG: if (value) { hTp->flags |= TPDEC_EARLY_CONFIG; } else { hTp->flags &= ~TPDEC_EARLY_CONFIG; } break; case TPDEC_PARAM_IGNORE_BUFFERFULLNESS: if (value) { hTp->flags |= TPDEC_IGNORE_BUFFERFULLNESS; } else { hTp->flags &= ~TPDEC_IGNORE_BUFFERFULLNESS; } break; case TPDEC_PARAM_SET_BITRATE: hTp->avgBitRate = value; break; case TPDEC_PARAM_BURST_PERIOD: hTp->burstPeriod = value; break; case TPDEC_PARAM_RESET: { int i; for (i = 0; i < (1 * 1); i++) { FDKresetBitbuffer(&hTp->bitStream[i]); hTp->auLength[i] = 0; hTp->accessUnitAnchor[i] = 0; } hTp->flags &= ~(TPDEC_SYNCOK | TPDEC_LOST_FRAMES_PENDING); if (hTp->transportFmt != TT_MP4_ADIF) { hTp->flags &= ~TPDEC_CONFIG_FOUND; } hTp->remainder = 0; hTp->avgBitRate = 0; hTp->missingAccessUnits = 0; hTp->numberOfRawDataBlocks = 0; hTp->globalFramePos = 0; hTp->holdOffFrames = 0; } break; case TPDEC_PARAM_TARGETLAYOUT: hTp->targetLayout = value; break; case TPDEC_PARAM_FORCE_CONFIG_CHANGE: hTp->ctrlCFGChange[value].forceCfgChange = TPDEC_FORCE_CONFIG_CHANGE; break; case TPDEC_PARAM_USE_ELEM_SKIPPING: if (value) { hTp->flags |= TPDEC_USE_ELEM_SKIPPING; } else { hTp->flags &= ~TPDEC_USE_ELEM_SKIPPING; } break; } return error; } UINT transportDec_GetNrOfSubFrames(HANDLE_TRANSPORTDEC hTp) { UINT nSubFrames = 0; if (hTp == NULL) return 0; if (hTp->transportFmt == TT_MP4_LATM_MCP1 || hTp->transportFmt == TT_MP4_LATM_MCP0 || hTp->transportFmt == TT_MP4_LOAS) nSubFrames = CLatmDemux_GetNrOfSubFrames(&hTp->parser.latm); else if (hTp->transportFmt == TT_MP4_ADTS) nSubFrames = hTp->parser.adts.bs.num_raw_blocks; return nSubFrames; } void transportDec_Close(HANDLE_TRANSPORTDEC *phTp) { if (phTp != NULL) { if (*phTp != NULL) { FreeRam_TransportDecoderBuffer(&(*phTp)->bsBuffer); FreeRam_TransportDecoder(phTp); } } } TRANSPORTDEC_ERROR transportDec_GetLibInfo(LIB_INFO *info) { int i; if (info == NULL) { return TRANSPORTDEC_UNKOWN_ERROR; } /* search for next free tab */ for (i = 0; i < FDK_MODULE_LAST; i++) { if (info[i].module_id == FDK_NONE) break; } if (i == FDK_MODULE_LAST) return TRANSPORTDEC_UNKOWN_ERROR; info += i; info->module_id = FDK_TPDEC; #ifdef SUPPRESS_BUILD_DATE_INFO info->build_date = ""; info->build_time = ""; #else info->build_date = __DATE__; info->build_time = __TIME__; #endif info->title = TP_LIB_TITLE; info->version = LIB_VERSION(TP_LIB_VL0, TP_LIB_VL1, TP_LIB_VL2); LIB_VERSION_STRING(info); info->flags = 0 | CAPF_ADIF | CAPF_ADTS | CAPF_LATM | CAPF_LOAS | CAPF_RAWPACKETS | CAPF_DRM; return TRANSPORTDEC_OK; /* FDKERR_NOERROR; */ } int transportDec_CrcStartReg(HANDLE_TRANSPORTDEC pTp, INT mBits) { switch (pTp->transportFmt) { case TT_MP4_ADTS: return adtsRead_CrcStartReg(&pTp->parser.adts, &pTp->bitStream[0], mBits); case TT_DRM: return drmRead_CrcStartReg(&pTp->parser.drm, &pTp->bitStream[0], mBits); default: return -1; } } void transportDec_CrcEndReg(HANDLE_TRANSPORTDEC pTp, INT reg) { switch (pTp->transportFmt) { case TT_MP4_ADTS: adtsRead_CrcEndReg(&pTp->parser.adts, &pTp->bitStream[0], reg); break; case TT_DRM: drmRead_CrcEndReg(&pTp->parser.drm, &pTp->bitStream[0], reg); break; default: break; } } TRANSPORTDEC_ERROR transportDec_CrcCheck(HANDLE_TRANSPORTDEC pTp) { switch (pTp->transportFmt) { case TT_MP4_ADTS: if ((pTp->parser.adts.bs.num_raw_blocks > 0) && (pTp->parser.adts.bs.protection_absent == 0)) { transportDec_AdjustEndOfAccessUnit(pTp); } return adtsRead_CrcCheck(&pTp->parser.adts); case TT_DRM: return drmRead_CrcCheck(&pTp->parser.drm); default: return TRANSPORTDEC_OK; } } TRANSPORTDEC_ERROR transportDec_DrmRawSdcAudioConfig_Check(UCHAR *conf, const UINT length) { CSAudioSpecificConfig asc; FDK_BITSTREAM bs; HANDLE_FDK_BITSTREAM hBs = &bs; FDKinitBitStream(hBs, conf, BUFSIZE_DUMMY_VALUE, length << 3, BS_READER); TRANSPORTDEC_ERROR err = DrmRawSdcAudioConfig_Parse(&asc, hBs, NULL, (UCHAR)AC_CM_ALLOC_MEM, 0); return err; } fdk-aac-2.0.2/libMpegTPEnc/000077500000000000000000000000001404225310700152505ustar00rootroot00000000000000fdk-aac-2.0.2/libMpegTPEnc/include/000077500000000000000000000000001404225310700166735ustar00rootroot00000000000000fdk-aac-2.0.2/libMpegTPEnc/include/tp_data.h000066400000000000000000000421121404225310700204600ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* MPEG transport format encoder library ********************* Author(s): Manuel Jander Description: MPEG Transport data tables *******************************************************************************/ #ifndef TP_DATA_H #define TP_DATA_H #include "machine_type.h" #include "FDK_audio.h" #include "FDK_bitstream.h" /* * Configuration */ #define TP_USAC_MAX_SPEAKERS (24) #define TP_USAC_MAX_EXT_ELEMENTS ((24)) #define TP_USAC_MAX_ELEMENTS ((24) + TP_USAC_MAX_EXT_ELEMENTS) #define TP_USAC_MAX_CONFIG_LEN \ 512 /* next power of two of maximum of escapedValue(hBs, 4, 4, 8) in \ AudioPreRoll() (285) */ #define TPDEC_USAC_NUM_CONFIG_CHANGE_FRAMES \ (1) /* Number of frames for config change in USAC */ enum { TPDEC_FLUSH_OFF = 0, TPDEC_RSV60_CFG_CHANGE_ATSC_FLUSH_ON = 1, TPDEC_RSV60_DASH_IPF_ATSC_FLUSH_ON = 2, TPDEC_USAC_DASH_IPF_FLUSH_ON = 3 }; enum { TPDEC_BUILD_UP_OFF = 0, TPDEC_RSV60_BUILD_UP_ON = 1, TPDEC_RSV60_BUILD_UP_ON_IN_BAND = 2, TPDEC_USAC_BUILD_UP_ON = 3, TPDEC_RSV60_BUILD_UP_IDLE = 4, TPDEC_RSV60_BUILD_UP_IDLE_IN_BAND = 5 }; /** * ProgramConfig struct. */ /* ISO/IEC 14496-3 4.4.1.1 Table 4.2 Program config element */ #define PC_FSB_CHANNELS_MAX 16 /* Front/Side/Back channels */ #define PC_LFE_CHANNELS_MAX 4 #define PC_ASSOCDATA_MAX 8 #define PC_CCEL_MAX 16 /* CC elements */ #define PC_COMMENTLENGTH 256 #define PC_NUM_HEIGHT_LAYER 3 typedef struct { /* PCE bitstream elements: */ UCHAR ElementInstanceTag; UCHAR Profile; UCHAR SamplingFrequencyIndex; UCHAR NumFrontChannelElements; UCHAR NumSideChannelElements; UCHAR NumBackChannelElements; UCHAR NumLfeChannelElements; UCHAR NumAssocDataElements; UCHAR NumValidCcElements; UCHAR MonoMixdownPresent; UCHAR MonoMixdownElementNumber; UCHAR StereoMixdownPresent; UCHAR StereoMixdownElementNumber; UCHAR MatrixMixdownIndexPresent; UCHAR MatrixMixdownIndex; UCHAR PseudoSurroundEnable; UCHAR FrontElementIsCpe[PC_FSB_CHANNELS_MAX]; UCHAR FrontElementTagSelect[PC_FSB_CHANNELS_MAX]; UCHAR FrontElementHeightInfo[PC_FSB_CHANNELS_MAX]; UCHAR SideElementIsCpe[PC_FSB_CHANNELS_MAX]; UCHAR SideElementTagSelect[PC_FSB_CHANNELS_MAX]; UCHAR SideElementHeightInfo[PC_FSB_CHANNELS_MAX]; UCHAR BackElementIsCpe[PC_FSB_CHANNELS_MAX]; UCHAR BackElementTagSelect[PC_FSB_CHANNELS_MAX]; UCHAR BackElementHeightInfo[PC_FSB_CHANNELS_MAX]; UCHAR LfeElementTagSelect[PC_LFE_CHANNELS_MAX]; UCHAR AssocDataElementTagSelect[PC_ASSOCDATA_MAX]; UCHAR CcElementIsIndSw[PC_CCEL_MAX]; UCHAR ValidCcElementTagSelect[PC_CCEL_MAX]; UCHAR CommentFieldBytes; UCHAR Comment[PC_COMMENTLENGTH]; /* Helper variables for administration: */ UCHAR isValid; /*!< Flag showing if PCE has been read successfully. */ UCHAR NumChannels; /*!< Amount of audio channels summing all channel elements including LFEs */ UCHAR NumEffectiveChannels; /*!< Amount of audio channels summing only SCEs and CPEs */ UCHAR elCounter; } CProgramConfig; typedef enum { ASCEXT_UNKOWN = -1, ASCEXT_SBR = 0x2b7, ASCEXT_PS = 0x548, ASCEXT_MPS = 0x76a, ASCEXT_SAOC = 0x7cb, ASCEXT_LDMPS = 0x7cc } TP_ASC_EXTENSION_ID; /** * GaSpecificConfig struct */ typedef struct { UINT m_frameLengthFlag; UINT m_dependsOnCoreCoder; UINT m_coreCoderDelay; UINT m_extensionFlag; UINT m_extensionFlag3; UINT m_layer; UINT m_numOfSubFrame; UINT m_layerLength; } CSGaSpecificConfig; typedef enum { ELDEXT_TERM = 0x0, /* Termination tag */ ELDEXT_SAOC = 0x1, /* SAOC config */ ELDEXT_LDSAC = 0x2, /* LD MPEG Surround config */ ELDEXT_DOWNSCALEINFO = 0x3 /* ELD sample rate adaptation */ /* reserved */ } ASC_ELD_EXT_TYPE; typedef struct { UCHAR m_frameLengthFlag; UCHAR m_sbrPresentFlag; UCHAR m_useLdQmfTimeAlign; /* Use LD-MPS QMF in SBR to achive time alignment */ UCHAR m_sbrSamplingRate; UCHAR m_sbrCrcFlag; UINT m_downscaledSamplingFrequency; } CSEldSpecificConfig; typedef struct { USAC_EXT_ELEMENT_TYPE usacExtElementType; USHORT usacExtElementConfigLength; USHORT usacExtElementDefaultLength; UCHAR usacExtElementPayloadFrag; UCHAR usacExtElementHasAudioPreRoll; } CSUsacExtElementConfig; typedef struct { MP4_ELEMENT_ID usacElementType; UCHAR m_noiseFilling; UCHAR m_harmonicSBR; UCHAR m_interTes; UCHAR m_pvc; UCHAR m_stereoConfigIndex; CSUsacExtElementConfig extElement; } CSUsacElementConfig; typedef struct { UCHAR m_frameLengthFlag; UCHAR m_coreSbrFrameLengthIndex; UCHAR m_sbrRatioIndex; UCHAR m_nUsacChannels; /* number of audio channels signaled in UsacDecoderConfig() / rsv603daDecoderConfig() via numElements and usacElementType */ UCHAR m_channelConfigurationIndex; UINT m_usacNumElements; CSUsacElementConfig element[TP_USAC_MAX_ELEMENTS]; UCHAR numAudioChannels; UCHAR m_usacConfigExtensionPresent; UCHAR elementLengthPresent; UCHAR UsacConfig[TP_USAC_MAX_CONFIG_LEN]; USHORT UsacConfigBits; } CSUsacConfig; /** * Audio configuration struct, suitable for encoder and decoder configuration. */ typedef struct { /* XYZ Specific Data */ union { CSGaSpecificConfig m_gaSpecificConfig; /**< General audio specific configuration. */ CSEldSpecificConfig m_eldSpecificConfig; /**< ELD specific configuration. */ CSUsacConfig m_usacConfig; /**< USAC specific configuration */ } m_sc; /* Common ASC parameters */ CProgramConfig m_progrConfigElement; /**< Program configuration. */ AUDIO_OBJECT_TYPE m_aot; /**< Audio Object Type. */ UINT m_samplingFrequency; /**< Samplerate. */ UINT m_samplesPerFrame; /**< Amount of samples per frame. */ UINT m_directMapping; /**< Document this please !! */ AUDIO_OBJECT_TYPE m_extensionAudioObjectType; /**< Audio object type */ UINT m_extensionSamplingFrequency; /**< Samplerate */ SCHAR m_channelConfiguration; /**< Channel configuration index */ SCHAR m_epConfig; /**< Error protection index */ SCHAR m_vcb11Flag; /**< aacSectionDataResilienceFlag */ SCHAR m_rvlcFlag; /**< aacScalefactorDataResilienceFlag */ SCHAR m_hcrFlag; /**< aacSpectralDataResilienceFlag */ SCHAR m_sbrPresentFlag; /**< Flag indicating the presence of SBR data in the bitstream */ SCHAR m_psPresentFlag; /**< Flag indicating the presence of parametric stereo data in the bitstream */ UCHAR m_samplingFrequencyIndex; /**< Samplerate index */ UCHAR m_extensionSamplingFrequencyIndex; /**< Samplerate index */ SCHAR m_extensionChannelConfiguration; /**< Channel configuration index */ UCHAR configMode; /**< The flag indicates if the callback shall work in memory allocation mode or in config change detection mode */ UCHAR AacConfigChanged; /**< The flag will be set if at least one aac config parameter has changed that requires a memory reconfiguration, otherwise it will be cleared */ UCHAR SbrConfigChanged; /**< The flag will be set if at least one sbr config parameter has changed that requires a memory reconfiguration, otherwise it will be cleared */ UCHAR SacConfigChanged; /**< The flag will be set if at least one sac config parameter has changed that requires a memory reconfiguration, otherwise it will be cleared */ UCHAR config[TP_USAC_MAX_CONFIG_LEN]; /**< Configuration stored as bitstream */ UINT configBits; /**< Configuration length in bits */ } CSAudioSpecificConfig; typedef struct { SCHAR flushCnt; /**< Flush frame counter */ UCHAR flushStatus; /**< Flag indicates flush mode: on|off */ SCHAR buildUpCnt; /**< Build up frame counter */ UCHAR buildUpStatus; /**< Flag indicates build up mode: on|off */ UCHAR cfgChanged; /**< Flag indicates that the config changed and the decoder needs to be initialized again via callback. Make sure that memory is freed before initialization. */ UCHAR contentChanged; /**< Flag indicates that the content changed i.e. a right truncation occured before */ UCHAR forceCfgChange; /**< Flag indicates if config change has to be forced even if new config is the same */ } CCtrlCFGChange; typedef INT (*cbUpdateConfig_t)(void *, const CSAudioSpecificConfig *, const UCHAR configMode, UCHAR *configChanged); typedef INT (*cbFreeMem_t)(void *, const CSAudioSpecificConfig *); typedef INT (*cbCtrlCFGChange_t)(void *, const CCtrlCFGChange *); typedef INT (*cbSsc_t)(void *, HANDLE_FDK_BITSTREAM, const AUDIO_OBJECT_TYPE coreCodec, const INT samplingRate, const INT frameSize, const INT stereoConfigIndex, const INT coreSbrFrameLengthIndex, const INT configBytes, const UCHAR configMode, UCHAR *configChanged); typedef INT (*cbSbr_t)(void *self, HANDLE_FDK_BITSTREAM hBs, const INT sampleRateIn, const INT sampleRateOut, const INT samplesPerFrame, const AUDIO_OBJECT_TYPE coreCodec, const MP4_ELEMENT_ID elementID, const INT elementIndex, const UCHAR harmonicSbr, const UCHAR stereoConfigIndex, const UCHAR configMode, UCHAR *configChanged, const INT downscaleFactor); typedef INT (*cbUsac_t)(void *self, HANDLE_FDK_BITSTREAM hBs); typedef INT (*cbUniDrc_t)(void *self, HANDLE_FDK_BITSTREAM hBs, const INT fullPayloadLength, const INT payloadType, const INT subStreamIndex, const INT payloadStart, const AUDIO_OBJECT_TYPE); typedef struct { cbUpdateConfig_t cbUpdateConfig; /*!< Function pointer for Config change notify callback. */ void *cbUpdateConfigData; /*!< User data pointer for Config change notify callback. */ cbFreeMem_t cbFreeMem; /*!< Function pointer for free memory callback. */ void *cbFreeMemData; /*!< User data pointer for free memory callback. */ cbCtrlCFGChange_t cbCtrlCFGChange; /*!< Function pointer for config change control callback. */ void *cbCtrlCFGChangeData; /*!< User data pointer for config change control callback. */ cbSsc_t cbSsc; /*!< Function pointer for SSC parser callback. */ void *cbSscData; /*!< User data pointer for SSC parser callback. */ cbSbr_t cbSbr; /*!< Function pointer for SBR header parser callback. */ void *cbSbrData; /*!< User data pointer for SBR header parser callback. */ cbUsac_t cbUsac; void *cbUsacData; cbUniDrc_t cbUniDrc; /*!< Function pointer for uniDrcConfig and loudnessInfoSet parser callback. */ void *cbUniDrcData; /*!< User data pointer for uniDrcConfig and loudnessInfoSet parser callback. */ } CSTpCallBacks; static const UINT SamplingRateTable[] = { 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000, 7350, 0, 0, 57600, 51200, 40000, 38400, 34150, 28800, 25600, 20000, 19200, 17075, 14400, 12800, 9600, 0, 0, 0, 0}; static inline int getSamplingRateIndex(UINT samplingRate, UINT nBits) { UINT sf_index; UINT tableSize = (1 << nBits) - 1; for (sf_index = 0; sf_index < tableSize; sf_index++) { if (SamplingRateTable[sf_index] == samplingRate) break; } if (sf_index > tableSize) { return tableSize - 1; } return sf_index; } /* * Get Channel count from channel configuration */ static inline int getNumberOfTotalChannels(int channelConfig) { switch (channelConfig) { case 1: case 2: case 3: case 4: case 5: case 6: return channelConfig; case 7: case 12: case 14: return 8; case 11: return 7; case 13: return 24; default: return 0; } } static inline int getNumberOfEffectiveChannels( const int channelConfig) { /* index: 0,1,2,3,4,5,6,7,8,9,10,11,12,13,14,15 */ const int n[] = {0, 1, 2, 3, 4, 5, 5, 7, 0, 0, 0, 6, 7, 22, 7, 0}; return n[channelConfig]; } #endif /* TP_DATA_H */ fdk-aac-2.0.2/libMpegTPEnc/include/tpenc_lib.h000066400000000000000000000344701404225310700210130ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* MPEG transport format encoder library ********************* Author(s): Manuel Jander Description: MPEG Transport encode *******************************************************************************/ #ifndef TPENC_LIB_H #define TPENC_LIB_H #include "tp_data.h" #include "FDK_bitstream.h" #define TRANSPORTENC_INBUF_SIZE 8192 typedef enum { TRANSPORTENC_OK = 0, /*!< All fine. */ TRANSPORTENC_NO_MEM, /*!< Out of memory. */ TRANSPORTENC_UNKOWN_ERROR = 1, /*!< Unknown error (embarrasing). */ TRANSPORTENC_INVALID_PARAMETER, /*!< An invalid parameter was passed to a function . */ TRANSPORTENC_UNSUPPORTED_FORMAT, /*!< Unsupported transport format. */ TRANSPORTENC_NOT_ENOUGH_BITS, /*!< Out of bits. Provide more bits and try again. */ TRANSPORTENC_INVALID_CONFIG, /*!< Error in configuration. */ TRANSPORTENC_LATM_INVALID_NR_OF_SUBFRAMES, /*!< LATM: number of subframes out of range. */ TRANSPORTENC_LOAS_NOT_AVAILABLE, /*!< LOAS format not supported. */ TRANSPORTENC_INVALID_LATM_ALIGNMENT, /*!< AudioMuxElement length not aligned to 1 byte. */ TRANSPORTENC_INVALID_TRANSMISSION_FRAME_LENGTH, /*!< Invalid transmission frame length (< 0). */ TRANSPORTENC_INVALID_CELP_FRAME_LENGTH, /*!< Invalid CELP frame length found (>= 62). */ TRANSPORTENC_INVALID_FRAME_BITS, /*!< Frame bits is not 40 and not 80. */ TRANSPORTENC_INVALID_AOT, /*!< Unknown AOT found. */ TRANSPORTENC_INVALID_AU_LENGTH /*!< Invalid Access Unit length (not byte-aligned). */ } TRANSPORTENC_ERROR; typedef struct TRANSPORTENC *HANDLE_TRANSPORTENC; /** * \brief Determine a reasonable channel configuration on the basis * of channel_mode. * \param noChannels Number of audio channels. * \return CHANNEL_MODE value that matches the given amount of audio * channels. */ CHANNEL_MODE transportEnc_GetChannelMode(int noChannels); /** * \brief Register SBR heaqder writer callback. * \param hTp Handle of transport decoder. * \param cbUpdateConfig Pointer to a callback function to handle SBR header * writing. * \param user_data void pointer for user data passed to the callback as * first parameter. * \return 0 on success. */ int transportEnc_RegisterSbrCallback(HANDLE_TRANSPORTENC hTpEnc, const cbSbr_t cbSbr, void *user_data); /** * \brief Register USAC SC writer callback. * \param hTp Handle of transport decoder. * \param cbUpdateConfig Pointer to a callback function to handle USAC * SCwriting. * \param user_data void pointer for user data passed to the callback as * first parameter. * \return 0 on success. */ int transportEnc_RegisterUsacCallback(HANDLE_TRANSPORTENC hTpEnc, const cbUsac_t cbUsac, void *user_data); /** * \brief Register SSC writer callback. * \param hTp Handle of transport decoder. * \param cbUpdateConfig Pointer to a callback function to handle SSC writing. * \param user_data void pointer for user data passed to the callback as * first parameter. * \return 0 on success. */ int transportEnc_RegisterSscCallback(HANDLE_TRANSPORTENC hTpEnc, const cbSsc_t cbSsc, void *user_data); /** * \brief Write ASC from given parameters. * \param asc A HANDLE_FDK_BITSTREAM where the ASC is written to. * \param config Structure containing the codec configuration settings. * \param cb callback information structure. * \return 0 on success. */ int transportEnc_writeASC(HANDLE_FDK_BITSTREAM asc, CODER_CONFIG *config, CSTpCallBacks *cb); /* Defintion of flags that can be passed to transportEnc_Open() */ #define TP_FLAG_MPEG4 1 /** MPEG4 (instead of MPEG2) */ #define TP_FLAG_LATM_AMV 2 /** LATM AudioMuxVersion */ #define TP_FLAG_LATM_AMVA 4 /** LATM AudioMuxVersionA */ /** * \brief Allocate transport encoder. * \param phTpEnc Pointer to transport encoder handle. * \return Error code. */ TRANSPORTENC_ERROR transportEnc_Open(HANDLE_TRANSPORTENC *phTpEnc); /** * \brief Init transport encoder. * \param bsBuffer Pointer to transport encoder. * \param bsBuffer Pointer to bitstream buffer. * \param bsBufferSize Size in bytes of bsBuffer. * \param transportFmt Format of the transport to be written. * \param config Pointer to a valid CODER_CONFIG struct. * \param flags Transport encoder flags. * \return Error code. */ TRANSPORTENC_ERROR transportEnc_Init(HANDLE_TRANSPORTENC hTpEnc, UCHAR *bsBuffer, INT bsBufferSize, TRANSPORT_TYPE transportFmt, CODER_CONFIG *config, UINT flags); /** * \brief Write additional bits in transport encoder. * \param config Pointer to a valid CODER_CONFIG struct. * \param nBits Number of additional bits. * \return Error code. */ TRANSPORTENC_ERROR transportEnc_AddOtherDataBits(HANDLE_TRANSPORTENC hTpEnc, const int nBits); /** * \brief Get transport encoder bitstream. * \param hTp Pointer to a transport encoder handle. * \return The handle to the requested FDK bitstream. */ HANDLE_FDK_BITSTREAM transportEnc_GetBitstream(HANDLE_TRANSPORTENC hTp); /** * \brief Get amount of bits required by the transport headers. * \param hTp Handle of transport encoder. * \param auBits Amount of payload bits required for the current subframe. * \return Error code. */ INT transportEnc_GetStaticBits(HANDLE_TRANSPORTENC hTp, int auBits); /** * \brief Close transport encoder. This function assures that all * allocated memory is freed. * \param phTp Pointer to a previously allocated transport encoder handle. */ void transportEnc_Close(HANDLE_TRANSPORTENC *phTp); /** * \brief Write one access unit. * \param hTp Handle of transport encoder. * \param total_bits Amount of total access unit bits. * \param bufferFullness Value of current buffer fullness in bits. * \param noConsideredChannels Number of bitrate wise considered channels (all * minus LFE channels). * \return Error code. */ TRANSPORTENC_ERROR transportEnc_WriteAccessUnit(HANDLE_TRANSPORTENC hTp, INT total_bits, int bufferFullness, int noConsideredChannels); /** * \brief Inform the transportEnc layer that writing of access unit has * finished. This function is required to be called when the encoder has * finished writing one Access one Access Unit for bitstream * housekeeping. * \param hTp Transport handle. * \param pBits Pointer to an int, where the current amount of frame bits is * passed and where the current amount of subframe bits is returned. * * OR: This integer is modified by the amount of extra bit alignment that may * occurr. * * \return Error code. */ TRANSPORTENC_ERROR transportEnc_EndAccessUnit(HANDLE_TRANSPORTENC hTp, int *pBits); /* * \brief Get a payload frame. * \param hTpEnc Transport encoder handle. * \param nBytes Pointer to an int to hold the frame size in bytes. Returns * zero if currently there is no complete frame for output (number of sub frames * > 1). * \return Error code. */ TRANSPORTENC_ERROR transportEnc_GetFrame(HANDLE_TRANSPORTENC hTpEnc, int *nbytes); /* ADTS CRC support */ /** * \brief Set current bitstream position as start of a new data region. * \param hTpEnc Transport encoder handle. * \param mBits Size in bits of the data region. Set to 0 if it should not be * of a fixed size. * \return Data region ID, which should be used when calling * transportEnc_CrcEndReg(). */ int transportEnc_CrcStartReg(HANDLE_TRANSPORTENC hTpEnc, int mBits); /** * \brief Set end of data region. * \param hTpEnc Transport encoder handle. * \param reg Data region ID, opbtained from transportEnc_CrcStartReg(). * \return void */ void transportEnc_CrcEndReg(HANDLE_TRANSPORTENC hTpEnc, int reg); /** * \brief Get AudioSpecificConfig or StreamMuxConfig from transport * encoder handle and write it to dataBuffer. * \param hTpEnc Transport encoder handle. * \param cc Pointer to the current and valid configuration contained * in a CODER_CONFIG struct. * \param dataBuffer Bitbuffer holding binary configuration. * \param confType Pointer to an UINT where the configuration type is * returned (0:ASC, 1:SMC). * \return Error code. */ TRANSPORTENC_ERROR transportEnc_GetConf(HANDLE_TRANSPORTENC hTpEnc, CODER_CONFIG *cc, FDK_BITSTREAM *dataBuffer, UINT *confType); /** * \brief Get information (version among other things) of the transport * encoder library. * \param info Pointer to an allocated LIB_INFO struct. * \return Error code. */ TRANSPORTENC_ERROR transportEnc_GetLibInfo(LIB_INFO *info); #endif /* #ifndef TPENC_LIB_H */ fdk-aac-2.0.2/libMpegTPEnc/src/000077500000000000000000000000001404225310700160375ustar00rootroot00000000000000fdk-aac-2.0.2/libMpegTPEnc/src/tp_version.h000066400000000000000000000120621404225310700204010ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* MPEG transport format encoder library ********************* Author(s): Description: *******************************************************************************/ #if !defined(TP_VERSION_H) #define TP_VERSION_H /* library info */ #define TP_LIB_VL0 3 #define TP_LIB_VL1 0 #define TP_LIB_VL2 0 #define TP_LIB_TITLE "MPEG Transport" #ifdef __ANDROID__ #define TP_LIB_BUILD_DATE "" #define TP_LIB_BUILD_TIME "" #else #define TP_LIB_BUILD_DATE __DATE__ #define TP_LIB_BUILD_TIME __TIME__ #endif #endif /* !defined(TP_VERSION_H) */ fdk-aac-2.0.2/libMpegTPEnc/src/tpenc_adif.cpp000066400000000000000000000154131404225310700206430ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* MPEG transport format encoder library ********************* Author(s): Description: ADIF Transport Headers writing *******************************************************************************/ #include "tpenc_adif.h" #include "tpenc_lib.h" #include "tpenc_asc.h" int adifWrite_EncodeHeader(ADIF_INFO *adif, HANDLE_FDK_BITSTREAM hBs, INT adif_buffer_fullness) { /* ADIF/PCE/ADTS definitions */ const char adifId[5] = "ADIF"; const int copyRightIdPresent = 0; const int originalCopy = 0; const int home = 0; int err = 0; int i; INT totalBitRate = adif->bitRate; if (adif->headerWritten) return 0; /* Align inside PCE with respect to the first bit of the header */ UINT alignAnchor = FDKgetValidBits(hBs); /* Signal variable bitrate if buffer fullnes exceeds 20 bit */ adif->bVariableRate = (adif_buffer_fullness >= (INT)(0x1 << 20)) ? 1 : 0; FDKwriteBits(hBs, adifId[0], 8); FDKwriteBits(hBs, adifId[1], 8); FDKwriteBits(hBs, adifId[2], 8); FDKwriteBits(hBs, adifId[3], 8); FDKwriteBits(hBs, copyRightIdPresent ? 1 : 0, 1); if (copyRightIdPresent) { for (i = 0; i < 72; i++) { FDKwriteBits(hBs, 0, 1); } } FDKwriteBits(hBs, originalCopy ? 1 : 0, 1); FDKwriteBits(hBs, home ? 1 : 0, 1); FDKwriteBits(hBs, adif->bVariableRate ? 1 : 0, 1); FDKwriteBits(hBs, totalBitRate, 23); /* we write only one PCE at the moment */ FDKwriteBits(hBs, 0, 4); if (!adif->bVariableRate) { FDKwriteBits(hBs, adif_buffer_fullness, 20); } /* Write PCE */ transportEnc_writePCE(hBs, adif->cm, adif->samplingRate, adif->instanceTag, adif->profile, adif->matrixMixdownA, (adif->pseudoSurroundEnable) ? 1 : 0, alignAnchor); return err; } int adifWrite_GetHeaderBits(ADIF_INFO *adif) { /* ADIF definitions */ const int copyRightIdPresent = 0; if (adif->headerWritten) return 0; int bits = 0; bits += 8 * 4; /* ADIF ID */ bits += 1; /* Copyright present */ if (copyRightIdPresent) bits += 72; /* Copyright ID */ bits += 26; bits += 4; /* Number of PCE's */ if (!adif->bVariableRate) { bits += 20; } /* write PCE */ bits = transportEnc_GetPCEBits(adif->cm, adif->matrixMixdownA, bits); return bits; } fdk-aac-2.0.2/libMpegTPEnc/src/tpenc_adif.h000066400000000000000000000133551404225310700203130ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* MPEG transport format encoder library ********************* Author(s): Alex Goeschel Description: Transport Headers support *******************************************************************************/ #ifndef TPENC_ADIF_H #define TPENC_ADIF_H #include "machine_type.h" #include "FDK_bitstream.h" #include "tp_data.h" typedef struct { CHANNEL_MODE cm; INT samplingRate; INT bitRate; int profile; int bVariableRate; int instanceTag; int headerWritten; int matrixMixdownA; int pseudoSurroundEnable; } ADIF_INFO; /** * \brief encodes ADIF Header * * \param adif pointer to ADIF_INFO structure * \param hBitStream handle of bitstream, where the ADIF header is written into * \param adif_buffer_fullness buffer fullness value for the ADIF header * * \return 0 on success */ int adifWrite_EncodeHeader(ADIF_INFO *adif, HANDLE_FDK_BITSTREAM hBitStream, INT adif_buffer_fullness); /** * \brief Get bit demand of a ADIF header * * \param adif pointer to ADIF_INFO structure * * \return amount of bits required to write the ADIF header according to the * data contained in the adif parameter */ int adifWrite_GetHeaderBits(ADIF_INFO *adif); #endif /* TPENC_ADIF_H */ fdk-aac-2.0.2/libMpegTPEnc/src/tpenc_adts.cpp000066400000000000000000000266471404225310700207060ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* MPEG transport format encoder library ********************* Author(s): Alex Groeschel Description: ADTS Transport Headers support *******************************************************************************/ #include "tpenc_adts.h" #include "tpenc_lib.h" #include "tpenc_asc.h" int adtsWrite_CrcStartReg( HANDLE_ADTS pAdts, /*!< pointer to adts stucture */ HANDLE_FDK_BITSTREAM hBs, /*!< handle to current bit buffer structure */ int mBits /*!< number of bits in crc region */ ) { if (pAdts->protection_absent) { return 0; } return (FDKcrcStartReg(&pAdts->crcInfo, hBs, mBits)); } void adtsWrite_CrcEndReg( HANDLE_ADTS pAdts, /*!< pointer to adts crc info stucture */ HANDLE_FDK_BITSTREAM hBs, /*!< handle to current bit buffer structure */ int reg /*!< crc region */ ) { if (pAdts->protection_absent == 0) { FDKcrcEndReg(&pAdts->crcInfo, hBs, reg); } } int adtsWrite_GetHeaderBits(HANDLE_ADTS hAdts) { int bits = 0; if (hAdts->currentBlock == 0) { /* Static and variable header bits */ bits = 56; if (!hAdts->protection_absent) { /* Add header/ single raw data block CRC bits */ bits += 16; if (hAdts->num_raw_blocks > 0) { /* Add bits of raw data block position markers */ bits += (hAdts->num_raw_blocks) * 16; } } } if (!hAdts->protection_absent && hAdts->num_raw_blocks > 0) { /* Add raw data block CRC bits. Not really part of the header, put they * cause bit overhead to be accounted. */ bits += 16; } hAdts->headerBits = bits; return bits; } INT adtsWrite_Init(HANDLE_ADTS hAdts, CODER_CONFIG *config) { /* Sanity checks */ if (config->nSubFrames < 1 || config->nSubFrames > 4 || (int)config->aot > 4 || (int)config->aot < 1) { return -1; } /* fixed header */ if (config->flags & CC_MPEG_ID) { hAdts->mpeg_id = 0; /* MPEG 4 */ } else { hAdts->mpeg_id = 1; /* MPEG 2 */ } hAdts->layer = 0; hAdts->protection_absent = !(config->flags & CC_PROTECTION); hAdts->profile = ((int)config->aot) - 1; hAdts->sample_freq_index = getSamplingRateIndex(config->samplingRate, 4); hAdts->sample_freq = config->samplingRate; hAdts->private_bit = 0; hAdts->channel_mode = config->channelMode; hAdts->original = 0; hAdts->home = 0; /* variable header */ hAdts->copyright_id = 0; hAdts->copyright_start = 0; hAdts->num_raw_blocks = config->nSubFrames - 1; /* 0 means 1 raw data block */ hAdts->channel_config_zero = config->channelConfigZero; FDKcrcInit(&hAdts->crcInfo, 0x8005, 0xFFFF, 16); hAdts->currentBlock = 0; return 0; } int adtsWrite_EncodeHeader(HANDLE_ADTS hAdts, HANDLE_FDK_BITSTREAM hBitStream, int buffer_fullness, int frame_length) { INT crcIndex = 0; hAdts->headerBits = adtsWrite_GetHeaderBits(hAdts); FDK_ASSERT(((frame_length + hAdts->headerBits) / 8) < 0x2000); /*13 bit*/ FDK_ASSERT(buffer_fullness < 0x800); /* 11 bit */ if (!hAdts->protection_absent) { FDKcrcReset(&hAdts->crcInfo); } if (hAdts->currentBlock == 0) { FDKresetBitbuffer(hBitStream, BS_WRITER); } hAdts->subFrameStartBit = FDKgetValidBits(hBitStream); /* Skip new header if this is raw data block 1..n */ if (hAdts->currentBlock == 0) { FDKresetBitbuffer(hBitStream, BS_WRITER); if (hAdts->num_raw_blocks == 0) { crcIndex = adtsWrite_CrcStartReg(hAdts, hBitStream, 0); } /* fixed header */ FDKwriteBits(hBitStream, 0xFFF, 12); FDKwriteBits(hBitStream, hAdts->mpeg_id, 1); FDKwriteBits(hBitStream, hAdts->layer, 2); FDKwriteBits(hBitStream, hAdts->protection_absent, 1); FDKwriteBits(hBitStream, hAdts->profile, 2); FDKwriteBits(hBitStream, hAdts->sample_freq_index, 4); FDKwriteBits(hBitStream, hAdts->private_bit, 1); FDKwriteBits( hBitStream, getChannelConfig(hAdts->channel_mode, hAdts->channel_config_zero), 3); FDKwriteBits(hBitStream, hAdts->original, 1); FDKwriteBits(hBitStream, hAdts->home, 1); /* variable header */ FDKwriteBits(hBitStream, hAdts->copyright_id, 1); FDKwriteBits(hBitStream, hAdts->copyright_start, 1); FDKwriteBits(hBitStream, (frame_length + hAdts->headerBits) >> 3, 13); FDKwriteBits(hBitStream, buffer_fullness, 11); FDKwriteBits(hBitStream, hAdts->num_raw_blocks, 2); if (!hAdts->protection_absent) { int i; /* End header CRC portion for single raw data block and write dummy zero * values for unknown fields. */ if (hAdts->num_raw_blocks == 0) { adtsWrite_CrcEndReg(hAdts, hBitStream, crcIndex); } else { for (i = 0; i < hAdts->num_raw_blocks; i++) { FDKwriteBits(hBitStream, 0, 16); } } FDKwriteBits(hBitStream, 0, 16); } } /* End of ADTS header */ return 0; } void adtsWrite_EndRawDataBlock(HANDLE_ADTS hAdts, HANDLE_FDK_BITSTREAM hBs, int *pBits) { if (!hAdts->protection_absent) { FDK_BITSTREAM bsWriter; FDKinitBitStream(&bsWriter, hBs->hBitBuf.Buffer, hBs->hBitBuf.bufSize, 0, BS_WRITER); FDKpushFor(&bsWriter, 56); if (hAdts->num_raw_blocks == 0) { FDKwriteBits(&bsWriter, FDKcrcGetCRC(&hAdts->crcInfo), 16); } else { int distance; /* Write CRC of current raw data block */ FDKwriteBits(hBs, FDKcrcGetCRC(&hAdts->crcInfo), 16); /* Write distance to current data block */ if (hAdts->currentBlock < hAdts->num_raw_blocks) { FDKpushFor(&bsWriter, hAdts->currentBlock * 16); distance = FDKgetValidBits(hBs) - (56 + (hAdts->num_raw_blocks) * 16 + 16); FDKwriteBits(&bsWriter, distance >> 3, 16); } } FDKsyncCache(&bsWriter); } /* Write total frame lenth for multiple raw data blocks and header CRC */ if (hAdts->num_raw_blocks > 0 && hAdts->currentBlock == hAdts->num_raw_blocks) { FDK_BITSTREAM bsWriter; int crcIndex = 0; FDKinitBitStream(&bsWriter, hBs->hBitBuf.Buffer, hBs->hBitBuf.bufSize, 0, BS_WRITER); if (!hAdts->protection_absent) { FDKcrcReset(&hAdts->crcInfo); crcIndex = FDKcrcStartReg(&hAdts->crcInfo, &bsWriter, 0); } /* Write total frame length */ FDKpushFor(&bsWriter, 56 - 28 + 2); FDKwriteBits(&bsWriter, FDKgetValidBits(hBs) >> 3, 13); /* Write header CRC */ if (!hAdts->protection_absent) { FDKpushFor(&bsWriter, 11 + 2 + (hAdts->num_raw_blocks) * 16); FDKcrcEndReg(&hAdts->crcInfo, &bsWriter, crcIndex); FDKwriteBits(&bsWriter, FDKcrcGetCRC(&hAdts->crcInfo), 16); } FDKsyncCache(&bsWriter); } /* Correct *pBits to reflect the amount of bits of the current subframe */ *pBits -= hAdts->subFrameStartBit; if (!hAdts->protection_absent && hAdts->num_raw_blocks > 0) { /* Fixup CRC bits, since they come after each raw data block */ *pBits += 16; } hAdts->currentBlock++; } fdk-aac-2.0.2/libMpegTPEnc/src/tpenc_adts.h000066400000000000000000000200201404225310700203260ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* MPEG transport format encoder library ********************* Author(s): Alex Groeschel Description: ADTS Transport writer *******************************************************************************/ #ifndef TPENC_ADTS_H #define TPENC_ADTS_H #include "tp_data.h" #include "FDK_crc.h" typedef struct { INT sample_freq; CHANNEL_MODE channel_mode; UCHAR decoderCanDoMpeg4; UCHAR mpeg_id; UCHAR layer; UCHAR protection_absent; UCHAR profile; UCHAR sample_freq_index; UCHAR private_bit; UCHAR original; UCHAR home; UCHAR copyright_id; UCHAR copyright_start; USHORT frame_length; UCHAR num_raw_blocks; UCHAR BufferFullnesStartFlag; UCHAR channel_config_zero; int headerBits; /*!< Header bit demand for the current raw data block */ int currentBlock; /*!< Index of current raw data block */ int subFrameStartBit; /*!< Bit position where the current raw data block begins */ FDK_CRCINFO crcInfo; } STRUCT_ADTS; typedef STRUCT_ADTS *HANDLE_ADTS; /** * \brief Initialize ADTS data structure * * \param hAdts ADTS data handle * \param config a valid CODER_CONFIG struct from where the required * information for the ADTS header is extrated from * * \return 0 in case of success. */ INT adtsWrite_Init(HANDLE_ADTS hAdts, CODER_CONFIG *config); /** * \brief Get the total bit overhead caused by ADTS * * \hAdts handle to ADTS data * * \return Amount of additional bits required for the current raw data block */ int adtsWrite_GetHeaderBits(HANDLE_ADTS hAdts); /** * \brief Write an ADTS header into the given bitstream. May not write a header * in case of multiple raw data blocks. * * \param hAdts ADTS data handle * \param hBitStream bitstream handle into which the ADTS may be written into * \param buffer_fullness the buffer fullness value for the ADTS header * \param the current raw data block length * * \return 0 in case of success. */ INT adtsWrite_EncodeHeader(HANDLE_ADTS hAdts, HANDLE_FDK_BITSTREAM hBitStream, int bufferFullness, int frame_length); /** * \brief Finish a ADTS raw data block * * \param hAdts ADTS data handle * \param hBs bitstream handle into which the ADTS may be written into * \param pBits a pointer to a integer holding the current bitstream buffer bit * count, which is corrected to the current raw data block boundary. * */ void adtsWrite_EndRawDataBlock(HANDLE_ADTS hAdts, HANDLE_FDK_BITSTREAM hBs, int *bits); /** * \brief Start CRC region with a maximum number of bits * If mBits is positive zero padding will be used for CRC calculation, if * there are less than mBits bits available. If mBits is negative no zero * padding is done. If mBits is zero the memory for the buffer is * allocated dynamically, the number of bits is not limited. * * \param pAdts ADTS data handle * \param hBs bitstream handle of which the CRC region ends * \param mBits limit of number of bits to be considered for the requested CRC * region * * \return ID for the created region, -1 in case of an error */ int adtsWrite_CrcStartReg(HANDLE_ADTS pAdts, HANDLE_FDK_BITSTREAM hBs, int mBits); /** * \brief Ends CRC region identified by reg * * \param pAdts ADTS data handle * \param hBs bitstream handle of which the CRC region ends * \param reg a CRC region ID returned previously by adtsWrite_CrcStartReg() */ void adtsWrite_CrcEndReg(HANDLE_ADTS pAdts, HANDLE_FDK_BITSTREAM hBs, int reg); #endif /* TPENC_ADTS_H */ fdk-aac-2.0.2/libMpegTPEnc/src/tpenc_asc.cpp000066400000000000000000001061161404225310700205070ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* MPEG transport format encoder library ********************* Author(s): Description: *******************************************************************************/ #include "tp_data.h" #include "tpenc_lib.h" #include "tpenc_asc.h" #include "FDK_bitstream.h" #include "genericStds.h" #include "FDK_crc.h" #define PCE_HEIGHT_EXT_SYNC (0xAC) #define HEIGHT_NORMAL 0 #define HEIGHT_TOP 1 #define HEIGHT_BOTTOM 2 #define MAX_FRONT_ELEMENTS 8 #define MAX_SIDE_ELEMENTS 3 #define MAX_BACK_ELEMENTS 4 /** * Describe additional PCE height information for front, side and back channel * elements. */ typedef struct { UCHAR num_front_height_channel_elements[2]; /*!< Number of front channel elements in top [0] and bottom [1] plane. */ UCHAR num_side_height_channel_elements[2]; /*!< Number of side channel elements in top [0] and bottom [1] plane. */ UCHAR num_back_height_channel_elements[2]; /*!< Number of back channel elements in top [0] and bottom [1] plane. */ } PCE_HEIGHT_NUM; /** * Describe a PCE based on placed channel elements and element type sequence. */ typedef struct { UCHAR num_front_channel_elements; /*!< Number of front channel elements. */ UCHAR num_side_channel_elements; /*!< Number of side channel elements. */ UCHAR num_back_channel_elements; /*!< Number of back channel elements. */ UCHAR num_lfe_channel_elements; /*!< Number of lfe channel elements. */ const MP4_ELEMENT_ID *pEl_type; /*!< List contains sequence describing the elements in present channel mode. (MPEG order) */ const PCE_HEIGHT_NUM *pHeight_num; } PCE_CONFIGURATION; /** * Map an incoming channel mode to a existing PCE configuration entry. */ typedef struct { CHANNEL_MODE channel_mode; /*!< Present channel mode. */ PCE_CONFIGURATION pce_configuration; /*!< Program config element description. */ } CHANNEL_CONFIGURATION; /** * The following arrays provide the IDs of the consecutive elements for each * mode. */ static const MP4_ELEMENT_ID elType_1[] = {ID_SCE}; static const MP4_ELEMENT_ID elType_2[] = {ID_CPE}; static const MP4_ELEMENT_ID elType_1_2[] = {ID_SCE, ID_CPE}; static const MP4_ELEMENT_ID elType_1_2_1[] = {ID_SCE, ID_CPE, ID_SCE}; static const MP4_ELEMENT_ID elType_1_2_2[] = {ID_SCE, ID_CPE, ID_CPE}; static const MP4_ELEMENT_ID elType_1_2_2_1[] = {ID_SCE, ID_CPE, ID_CPE, ID_LFE}; static const MP4_ELEMENT_ID elType_1_2_2_2_1[] = {ID_SCE, ID_CPE, ID_CPE, ID_CPE, ID_LFE}; static const MP4_ELEMENT_ID elType_6_1[] = {ID_SCE, ID_CPE, ID_CPE, ID_SCE, ID_LFE}; static const MP4_ELEMENT_ID elType_7_1_back[] = {ID_SCE, ID_CPE, ID_CPE, ID_CPE, ID_LFE}; static const MP4_ELEMENT_ID elType_7_1_top_front[] = {ID_SCE, ID_CPE, ID_CPE, ID_LFE, ID_CPE}; static const MP4_ELEMENT_ID elType_7_1_rear_surround[] = { ID_SCE, ID_CPE, ID_CPE, ID_CPE, ID_LFE}; static const MP4_ELEMENT_ID elType_7_1_front_center[] = {ID_SCE, ID_CPE, ID_CPE, ID_CPE, ID_LFE}; /** * The following arrays provide information on how many front, side and back * elements are assigned to the top or bottom plane for each mode that comprises * height information. */ static const PCE_HEIGHT_NUM heightNum_7_1_top_front = {{1, 0}, {0, 0}, {0, 0}}; /** * \brief Table contains all supported channel modes and according PCE configuration description. * * The mode identifier is followed by the number of front, side, back, and LFE elements. * These are followed by a pointer to the IDs of the consecutive elements (ID_SCE, ID_CPE, ID_LFE). * * For some modes (MODE_7_1_TOP_FRONT and MODE_22_2) additional height information is transmitted. * In this case the additional pointer provides information on how many front, side and back elements * are assigned to the top or bottom plane.The elements are arranged in the following order: normal height (front, side, back, LFE), top height (front, side, back), bottom height (front, side, back). * * * E.g. MODE_7_1_TOP_FRONT means: * - 3 elements are front channel elements. * - 0 elements are side channel elements. * - 1 element is back channel element. * - 1 element is an LFE channel element. * - the element order is ID_SCE, ID_CPE, ID_CPE, ID_LFE, ID_CPE. * - 1 of the front elements is in the top plane. * * This leads to the following mapping for the cconsecutive elements in the MODE_7_1_TOP_FRONT bitstream: * - ID_SCE -> normal height front, - ID_CPE -> normal height front, - ID_CPE -> normal height back, - ID_LFE -> normal height LFE, - ID_CPE -> top height front. */ static const CHANNEL_CONFIGURATION pceConfigTab[] = { {MODE_1, {1, 0, 0, 0, elType_1, NULL}}, /* don't transmit height information in this mode */ {MODE_2, {1, 0, 0, 0, elType_2, NULL}}, /* don't transmit height information in this mode */ {MODE_1_2, {2, 0, 0, 0, elType_1_2, NULL}}, /* don't transmit height information in this mode */ {MODE_1_2_1, {2, 0, 1, 0, elType_1_2_1, NULL}}, /* don't transmit height information in this mode */ {MODE_1_2_2, {2, 0, 1, 0, elType_1_2_2, NULL}}, /* don't transmit height information in this mode */ {MODE_1_2_2_1, {2, 0, 1, 1, elType_1_2_2_1, NULL}}, /* don't transmit height information in this mode */ {MODE_1_2_2_2_1, {3, 0, 1, 1, elType_1_2_2_2_1, NULL}}, /* don't transmit height information in this mode */ {MODE_6_1, {2, 0, 2, 1, elType_6_1, NULL}}, /* don't transmit height information in this mode */ {MODE_7_1_BACK, {2, 0, 2, 1, elType_7_1_back, NULL}}, /* don't transmit height information in this mode */ {MODE_7_1_TOP_FRONT, {3, 0, 1, 1, elType_7_1_top_front, &heightNum_7_1_top_front}}, {MODE_7_1_REAR_SURROUND, {2, 0, 2, 1, elType_7_1_rear_surround, NULL}}, /* don't transmit height information in this mode */ {MODE_7_1_FRONT_CENTER, {3, 0, 1, 1, elType_7_1_front_center, NULL}} /* don't transmit height information in this mode */ }; /** * \brief Get program config element description for existing channel mode. * * \param channel_mode Current channel mode. * * \return * - Pointer to PCE_CONFIGURATION entry, on success. * - NULL, on failure. */ static const PCE_CONFIGURATION *getPceEntry(const CHANNEL_MODE channel_mode) { UINT i; const PCE_CONFIGURATION *pce_config = NULL; for (i = 0; i < (sizeof(pceConfigTab) / sizeof(CHANNEL_CONFIGURATION)); i++) { if (pceConfigTab[i].channel_mode == channel_mode) { pce_config = &pceConfigTab[i].pce_configuration; break; } } return pce_config; } int getChannelConfig(const CHANNEL_MODE channel_mode, const UCHAR channel_config_zero) { INT chan_config = 0; if (channel_config_zero != 0) { chan_config = 0; } else { switch (channel_mode) { case MODE_1: chan_config = 1; break; case MODE_2: chan_config = 2; break; case MODE_1_2: chan_config = 3; break; case MODE_1_2_1: chan_config = 4; break; case MODE_1_2_2: chan_config = 5; break; case MODE_1_2_2_1: chan_config = 6; break; case MODE_1_2_2_2_1: chan_config = 7; break; case MODE_6_1: chan_config = 11; break; case MODE_7_1_BACK: chan_config = 12; break; case MODE_7_1_TOP_FRONT: chan_config = 14; break; default: chan_config = 0; } } return chan_config; } CHANNEL_MODE transportEnc_GetChannelMode(int noChannels) { CHANNEL_MODE chMode; if (noChannels <= 8 && noChannels > 0) chMode = (CHANNEL_MODE)( (noChannels == 8) ? 7 : noChannels); /* see : iso/mpeg4 v1 audio subpart1*/ else chMode = MODE_UNKNOWN; return chMode; } int transportEnc_writePCE(HANDLE_FDK_BITSTREAM hBs, CHANNEL_MODE channelMode, INT sampleRate, int instanceTagPCE, int profile, int matrixMixdownA, int pseudoSurroundEnable, UINT alignAnchor) { int sampleRateIndex, i; const PCE_CONFIGURATION *config = NULL; const MP4_ELEMENT_ID *pEl_list = NULL; UCHAR cpeCnt = 0, sceCnt = 0, lfeCnt = 0, frntCnt = 0, sdCnt = 0, bckCnt = 0, isCpe = 0, tag = 0, normalFrontEnd = 0, normalSideEnd = 0, normalBackEnd = 0, topFrontEnd = 0, topSideEnd = 0, topBackEnd = 0, bottomFrontEnd = 0, bottomSideEnd = 0; #ifdef FDK_ASSERT_ENABLE UCHAR bottomBackEnd = 0; #endif enum elementDepth { FRONT, SIDE, BACK } elDepth; sampleRateIndex = getSamplingRateIndex(sampleRate, 4); if (sampleRateIndex == 15) { return -1; } if ((config = getPceEntry(channelMode)) == NULL) { return -1; } FDK_ASSERT(config->num_front_channel_elements <= MAX_FRONT_ELEMENTS); FDK_ASSERT(config->num_side_channel_elements <= MAX_SIDE_ELEMENTS); FDK_ASSERT(config->num_back_channel_elements <= MAX_BACK_ELEMENTS); UCHAR frontIsCpe[MAX_FRONT_ELEMENTS] = {0}, frontTag[MAX_FRONT_ELEMENTS] = {0}, sideIsCpe[MAX_SIDE_ELEMENTS] = {0}, sideTag[MAX_SIDE_ELEMENTS] = {0}, backIsCpe[MAX_BACK_ELEMENTS] = {0}, backTag[MAX_BACK_ELEMENTS] = {0}; /* Write general information */ FDKwriteBits(hBs, instanceTagPCE, 4); /* Element instance tag */ FDKwriteBits(hBs, profile, 2); /* Object type */ FDKwriteBits(hBs, sampleRateIndex, 4); /* Sample rate index*/ FDKwriteBits(hBs, config->num_front_channel_elements, 4); /* Front channel Elements */ FDKwriteBits(hBs, config->num_side_channel_elements, 4); /* No Side Channel Elements */ FDKwriteBits(hBs, config->num_back_channel_elements, 4); /* No Back channel Elements */ FDKwriteBits(hBs, config->num_lfe_channel_elements, 2); /* No Lfe channel elements */ FDKwriteBits(hBs, 0, 3); /* No assoc data elements */ FDKwriteBits(hBs, 0, 4); /* No valid cc elements */ FDKwriteBits(hBs, 0, 1); /* Mono mixdown present */ FDKwriteBits(hBs, 0, 1); /* Stereo mixdown present */ if (matrixMixdownA != 0 && ((channelMode == MODE_1_2_2) || (channelMode == MODE_1_2_2_1))) { FDKwriteBits(hBs, 1, 1); /* Matrix mixdown present */ FDKwriteBits(hBs, (matrixMixdownA - 1) & 0x3, 2); /* matrix_mixdown_idx */ FDKwriteBits(hBs, (pseudoSurroundEnable) ? 1 : 0, 1); /* pseudo_surround_enable */ } else { FDKwriteBits(hBs, 0, 1); /* Matrix mixdown not present */ } if (config->pHeight_num != NULL) { /* we have up to three different height levels, and in each height level we * may have front, side and back channels. We need to know where each * section ends to correctly count the tags */ normalFrontEnd = config->num_front_channel_elements - config->pHeight_num->num_front_height_channel_elements[0] - config->pHeight_num->num_front_height_channel_elements[1]; normalSideEnd = normalFrontEnd + config->num_side_channel_elements - config->pHeight_num->num_side_height_channel_elements[0] - config->pHeight_num->num_side_height_channel_elements[1]; normalBackEnd = normalSideEnd + config->num_back_channel_elements - config->pHeight_num->num_back_height_channel_elements[0] - config->pHeight_num->num_back_height_channel_elements[1]; topFrontEnd = normalBackEnd + config->num_lfe_channel_elements + config->pHeight_num->num_front_height_channel_elements[0]; /* only normal height LFEs assumed */ topSideEnd = topFrontEnd + config->pHeight_num->num_side_height_channel_elements[0]; topBackEnd = topSideEnd + config->pHeight_num->num_back_height_channel_elements[0]; bottomFrontEnd = topBackEnd + config->pHeight_num->num_front_height_channel_elements[1]; bottomSideEnd = bottomFrontEnd + config->pHeight_num->num_side_height_channel_elements[1]; #ifdef FDK_ASSERT_ENABLE bottomBackEnd = bottomSideEnd + config->pHeight_num->num_back_height_channel_elements[1]; #endif } else { /* we have only one height level, so we don't care about top or bottom */ normalFrontEnd = config->num_front_channel_elements; normalSideEnd = normalFrontEnd + config->num_side_channel_elements; normalBackEnd = normalSideEnd + config->num_back_channel_elements; } /* assign cpe and tag information to either front, side or back channels */ pEl_list = config->pEl_type; for (i = 0; i < config->num_front_channel_elements + config->num_side_channel_elements + config->num_back_channel_elements + config->num_lfe_channel_elements; i++) { if (*pEl_list == ID_LFE) { pEl_list++; continue; } isCpe = (*pEl_list++ == ID_CPE) ? 1 : 0; tag = (isCpe) ? cpeCnt++ : sceCnt++; if (i < normalFrontEnd) elDepth = FRONT; else if (i < normalSideEnd) elDepth = SIDE; else if (i < normalBackEnd) elDepth = BACK; else if (i < topFrontEnd) elDepth = FRONT; else if (i < topSideEnd) elDepth = SIDE; else if (i < topBackEnd) elDepth = BACK; else if (i < bottomFrontEnd) elDepth = FRONT; else if (i < bottomSideEnd) elDepth = SIDE; else { elDepth = BACK; FDK_ASSERT(i < bottomBackEnd); /* won't fail if implementation of pce configuration table is correct */ } switch (elDepth) { case FRONT: FDK_ASSERT(frntCnt < config->num_front_channel_elements); frontIsCpe[frntCnt] = isCpe; frontTag[frntCnt++] = tag; break; case SIDE: FDK_ASSERT(sdCnt < config->num_side_channel_elements); sideIsCpe[sdCnt] = isCpe; sideTag[sdCnt++] = tag; break; case BACK: FDK_ASSERT(bckCnt < config->num_back_channel_elements); backIsCpe[bckCnt] = isCpe; backTag[bckCnt++] = tag; break; } } /* Write front channel isCpe and tags */ for (i = 0; i < config->num_front_channel_elements; i++) { FDKwriteBits(hBs, frontIsCpe[i], 1); FDKwriteBits(hBs, frontTag[i], 4); } /* Write side channel isCpe and tags */ for (i = 0; i < config->num_side_channel_elements; i++) { FDKwriteBits(hBs, sideIsCpe[i], 1); FDKwriteBits(hBs, sideTag[i], 4); } /* Write back channel isCpe and tags */ for (i = 0; i < config->num_back_channel_elements; i++) { FDKwriteBits(hBs, backIsCpe[i], 1); FDKwriteBits(hBs, backTag[i], 4); } /* Write LFE information */ for (i = 0; i < config->num_lfe_channel_elements; i++) { FDKwriteBits(hBs, lfeCnt++, 4); /* LFE channel Instance Tag. */ } /* - num_valid_cc_elements always 0. - num_assoc_data_elements always 0. */ /* Byte alignment: relative to alignAnchor ADTS: align with respect to the first bit of the raw_data_block() ADIF: align with respect to the first bit of the header LATM: align with respect to the first bit of the ASC */ FDKbyteAlign(hBs, alignAnchor); /* Alignment */ /* Write comment information */ if (config->pHeight_num != NULL) { /* embed height information in comment field */ INT commentBytes = 1 /* PCE_HEIGHT_EXT_SYNC */ + ((((config->num_front_channel_elements + config->num_side_channel_elements + config->num_back_channel_elements) << 1) + 7) >> 3) /* 2 bit height info per element, round up to full bytes */ + 1; /* CRC */ FDKwriteBits(hBs, commentBytes, 8); /* comment size. */ FDK_CRCINFO crcInfo; /* CRC state info */ INT crcReg; FDKcrcInit(&crcInfo, 0x07, 0xFF, 8); crcReg = FDKcrcStartReg(&crcInfo, hBs, 0); FDKwriteBits(hBs, PCE_HEIGHT_EXT_SYNC, 8); /* indicate height extension */ /* front channel height information */ for (i = 0; i < config->num_front_channel_elements - config->pHeight_num->num_front_height_channel_elements[0] - config->pHeight_num->num_front_height_channel_elements[1]; i++) FDKwriteBits(hBs, HEIGHT_NORMAL, 2); for (i = 0; i < config->pHeight_num->num_front_height_channel_elements[0]; i++) FDKwriteBits(hBs, HEIGHT_TOP, 2); for (i = 0; i < config->pHeight_num->num_front_height_channel_elements[1]; i++) FDKwriteBits(hBs, HEIGHT_BOTTOM, 2); /* side channel height information */ for (i = 0; i < config->num_side_channel_elements - config->pHeight_num->num_side_height_channel_elements[0] - config->pHeight_num->num_side_height_channel_elements[1]; i++) FDKwriteBits(hBs, HEIGHT_NORMAL, 2); for (i = 0; i < config->pHeight_num->num_side_height_channel_elements[0]; i++) FDKwriteBits(hBs, HEIGHT_TOP, 2); for (i = 0; i < config->pHeight_num->num_side_height_channel_elements[1]; i++) FDKwriteBits(hBs, HEIGHT_BOTTOM, 2); /* back channel height information */ for (i = 0; i < config->num_back_channel_elements - config->pHeight_num->num_back_height_channel_elements[0] - config->pHeight_num->num_back_height_channel_elements[1]; i++) FDKwriteBits(hBs, HEIGHT_NORMAL, 2); for (i = 0; i < config->pHeight_num->num_back_height_channel_elements[0]; i++) FDKwriteBits(hBs, HEIGHT_TOP, 2); for (i = 0; i < config->pHeight_num->num_back_height_channel_elements[1]; i++) FDKwriteBits(hBs, HEIGHT_BOTTOM, 2); FDKbyteAlign(hBs, alignAnchor); /* Alignment */ FDKcrcEndReg(&crcInfo, hBs, crcReg); FDKwriteBits(hBs, FDKcrcGetCRC(&crcInfo), 8); } else { FDKwriteBits(hBs, 0, 8); /* Do no write any comment or height information. */ } return 0; } int transportEnc_GetPCEBits(CHANNEL_MODE channelMode, int matrixMixdownA, int bits) { const PCE_CONFIGURATION *config = NULL; if ((config = getPceEntry(channelMode)) == NULL) { return -1; /* unsupported channelmapping */ } bits += 4 + 2 + 4; /* Element instance tag + Object type + Sample rate index */ bits += 4 + 4 + 4 + 2; /* No (front + side + back + lfe channel) elements */ bits += 3 + 4; /* No (assoc data + valid cc) elements */ bits += 1 + 1 + 1; /* Mono + Stereo + Matrix mixdown present */ if (matrixMixdownA != 0 && ((channelMode == MODE_1_2_2) || (channelMode == MODE_1_2_2_1))) { bits += 3; /* matrix_mixdown_idx + pseudo_surround_enable */ } bits += (1 + 4) * (INT)config->num_front_channel_elements; bits += (1 + 4) * (INT)config->num_side_channel_elements; bits += (1 + 4) * (INT)config->num_back_channel_elements; bits += (4) * (INT)config->num_lfe_channel_elements; /* - num_valid_cc_elements always 0. - num_assoc_data_elements always 0. */ if ((bits % 8) != 0) { bits += (8 - (bits % 8)); /* Alignment */ } bits += 8; /* Comment field bytes */ if (config->pHeight_num != NULL) { /* Comment field (height extension) */ bits += 8 /* PCE_HEIGHT_EXT_SYNC */ + ((config->num_front_channel_elements + config->num_side_channel_elements + config->num_back_channel_elements) << 1) /* 2 bit height info per element */ + 8; /* CRC */ if ((bits % 8) != 0) { bits += (8 - (bits % 8)); /* Alignment */ } } return bits; } static void writeAot(HANDLE_FDK_BITSTREAM hBitstreamBuffer, AUDIO_OBJECT_TYPE aot) { int tmp = (int)aot; if (tmp > 31) { FDKwriteBits(hBitstreamBuffer, AOT_ESCAPE, 5); FDKwriteBits(hBitstreamBuffer, tmp - 32, 6); /* AudioObjectType */ } else { FDKwriteBits(hBitstreamBuffer, tmp, 5); } } static void writeSampleRate(HANDLE_FDK_BITSTREAM hBs, int sampleRate, int nBits) { int srIdx = getSamplingRateIndex(sampleRate, nBits); FDKwriteBits(hBs, srIdx, nBits); if (srIdx == (1 << nBits) - 1) { FDKwriteBits(hBs, sampleRate, 24); } } static int transportEnc_writeGASpecificConfig(HANDLE_FDK_BITSTREAM asc, CODER_CONFIG *config, int extFlg, UINT alignAnchor) { int aot = config->aot; int samplesPerFrame = config->samplesPerFrame; /* start of GASpecificConfig according to ISO/IEC 14496-3 Subpart 4, 4.4.1 */ FDKwriteBits(asc, ((samplesPerFrame == 960 || samplesPerFrame == 480) ? 1 : 0), 1); /* frameLengthFlag: 1 for a 960/480 (I)MDCT, 0 for a 1024/512 (I)MDCT*/ FDKwriteBits(asc, 0, 1); /* dependsOnCoreCoder: Sampling Rate Coder Specific, see in ISO/IEC 14496-3 Subpart 4, 4.4.1 */ FDKwriteBits(asc, extFlg, 1); /* Extension Flag: Shall be 1 for aot = 17,19,20,21,22,23 */ /* Write PCE if channel config is not 1-7 */ if (getChannelConfig(config->channelMode, config->channelConfigZero) == 0) { transportEnc_writePCE(asc, config->channelMode, config->samplingRate, 0, 1, config->matrixMixdownA, (config->flags & CC_PSEUDO_SURROUND) ? 1 : 0, alignAnchor); } if ((aot == AOT_AAC_SCAL) || (aot == AOT_ER_AAC_SCAL)) { FDKwriteBits(asc, 0, 3); /* layerNr */ } if (extFlg) { if (aot == AOT_ER_BSAC) { FDKwriteBits(asc, config->BSACnumOfSubFrame, 5); /* numOfSubFrame */ FDKwriteBits(asc, config->BSAClayerLength, 11); /* layer_length */ } if ((aot == AOT_ER_AAC_LC) || (aot == AOT_ER_AAC_LTP) || (aot == AOT_ER_AAC_SCAL) || (aot == AOT_ER_AAC_LD)) { FDKwriteBits(asc, (config->flags & CC_VCB11) ? 1 : 0, 1); /* aacSectionDataResillienceFlag */ FDKwriteBits(asc, (config->flags & CC_RVLC) ? 1 : 0, 1); /* aacScaleFactorDataResillienceFlag */ FDKwriteBits(asc, (config->flags & CC_HCR) ? 1 : 0, 1); /* aacSpectralDataResillienceFlag */ } FDKwriteBits(asc, 0, 1); /* extensionFlag3: reserved. Shall be '0' */ } return 0; } static int transportEnc_writeELDSpecificConfig(HANDLE_FDK_BITSTREAM hBs, CODER_CONFIG *config, int epConfig, CSTpCallBacks *cb) { UINT frameLengthFlag = 0; switch (config->samplesPerFrame) { case 512: case 256: case 128: case 64: frameLengthFlag = 0; break; case 480: case 240: case 160: case 120: case 60: frameLengthFlag = 1; break; } FDKwriteBits(hBs, frameLengthFlag, 1); FDKwriteBits(hBs, (config->flags & CC_VCB11) ? 1 : 0, 1); FDKwriteBits(hBs, (config->flags & CC_RVLC) ? 1 : 0, 1); FDKwriteBits(hBs, (config->flags & CC_HCR) ? 1 : 0, 1); FDKwriteBits(hBs, (config->flags & CC_SBR) ? 1 : 0, 1); /* SBR header flag */ if ((config->flags & CC_SBR)) { FDKwriteBits(hBs, (config->samplingRate == config->extSamplingRate) ? 0 : 1, 1); /* Samplerate Flag */ FDKwriteBits(hBs, (config->flags & CC_SBRCRC) ? 1 : 0, 1); /* SBR CRC flag*/ if (cb->cbSbr != NULL) { const PCE_CONFIGURATION *pPce; int e, sbrElementIndex = 0; pPce = getPceEntry(config->channelMode); for (e = 0; e < pPce->num_front_channel_elements + pPce->num_side_channel_elements + pPce->num_back_channel_elements + pPce->num_lfe_channel_elements; e++) { if ((pPce->pEl_type[e] == ID_SCE) || (pPce->pEl_type[e] == ID_CPE)) { cb->cbSbr(cb->cbSbrData, hBs, 0, 0, 0, config->aot, pPce->pEl_type[e], sbrElementIndex, 0, 0, 0, NULL, 1); sbrElementIndex++; } } } } if ((config->flags & CC_SAC) && (cb->cbSsc != NULL)) { FDKwriteBits(hBs, ELDEXT_LDSAC, 4); const INT eldExtLen = (cb->cbSsc(cb->cbSscData, NULL, config->aot, config->extSamplingRate, 0, 0, 0, 0, 0, NULL) + 7) >> 3; INT cnt = eldExtLen; if (cnt < 0xF) { FDKwriteBits(hBs, cnt, 4); } else { FDKwriteBits(hBs, 0xF, 4); cnt -= 0xF; if (cnt < 0xFF) { FDKwriteBits(hBs, cnt, 8); } else { FDKwriteBits(hBs, 0xFF, 8); cnt -= 0xFF; FDK_ASSERT(cnt <= 0xFFFF); FDKwriteBits(hBs, cnt, 16); } } cb->cbSsc(cb->cbSscData, hBs, config->aot, config->extSamplingRate, 0, 0, 0, 0, 0, NULL); } if (config->downscaleSamplingRate != 0 && config->downscaleSamplingRate != config->extSamplingRate) { /* downscale active */ /* eldExtLenDsc: Number of bytes for the ELD downscale extension (srIdx needs 1 byte + downscaleSamplingRate needs additional 3 bytes) */ int eldExtLenDsc = 1; int downscaleSamplingRate = config->downscaleSamplingRate; FDKwriteBits(hBs, ELDEXT_DOWNSCALEINFO, 4); /* ELDEXT_DOWNSCALEINFO */ if ((downscaleSamplingRate != 96000) && (downscaleSamplingRate != 88200) && (downscaleSamplingRate != 64000) && (downscaleSamplingRate != 48000) && (downscaleSamplingRate != 44100) && (downscaleSamplingRate != 32000) && (downscaleSamplingRate != 24000) && (downscaleSamplingRate != 22050) && (downscaleSamplingRate != 16000) && (downscaleSamplingRate != 12000) && (downscaleSamplingRate != 11025) && (downscaleSamplingRate != 8000) && (downscaleSamplingRate != 7350)) { eldExtLenDsc = 4; /* length extends to 4 if downscaleSamplingRate's value is not one of the listed values */ } FDKwriteBits(hBs, eldExtLenDsc, 4); writeSampleRate(hBs, downscaleSamplingRate, 4); FDKwriteBits(hBs, 0x0, 4); /* fill_nibble */ } FDKwriteBits(hBs, ELDEXT_TERM, 4); /* ELDEXT_TERM */ return 0; } static int transportEnc_writeUsacSpecificConfig(HANDLE_FDK_BITSTREAM hBs, int extFlag, CODER_CONFIG *cc, CSTpCallBacks *cb) { FDK_BITSTREAM usacConf; int usacConfigBits = cc->rawConfigBits; if ((usacConfigBits <= 0) || ((usacConfigBits + 7) / 8 > (int)sizeof(cc->rawConfig))) { return TRANSPORTENC_UNSUPPORTED_FORMAT; } FDKinitBitStream(&usacConf, cc->rawConfig, BUFSIZE_DUMMY_VALUE, usacConfigBits, BS_READER); for (; usacConfigBits > 0; usacConfigBits--) { UINT tmp = FDKreadBit(&usacConf); FDKwriteBits(hBs, tmp, 1); } FDKsyncCache(hBs); return TRANSPORTENC_OK; } int transportEnc_writeASC(HANDLE_FDK_BITSTREAM asc, CODER_CONFIG *config, CSTpCallBacks *cb) { UINT extFlag = 0; int err; int epConfig = 0; /* Required for the PCE. */ UINT alignAnchor = FDKgetValidBits(asc); /* Extension Flag: Shall be 1 for aot = 17,19,20,21,22,23,39 */ switch (config->aot) { case AOT_ER_AAC_LC: case AOT_ER_AAC_LTP: case AOT_ER_AAC_SCAL: case AOT_ER_TWIN_VQ: case AOT_ER_BSAC: case AOT_ER_AAC_LD: case AOT_ER_AAC_ELD: case AOT_USAC: extFlag = 1; break; default: break; } if (config->sbrSignaling == SIG_EXPLICIT_HIERARCHICAL && config->sbrPresent) writeAot(asc, config->extAOT); else writeAot(asc, config->aot); /* In case of USAC it is the output not the core sampling rate */ writeSampleRate(asc, config->samplingRate, 4); /* Try to guess a reasonable channel mode if not given */ if (config->channelMode == MODE_INVALID) { config->channelMode = transportEnc_GetChannelMode(config->noChannels); if (config->channelMode == MODE_INVALID) return -1; } FDKwriteBits( asc, getChannelConfig(config->channelMode, config->channelConfigZero), 4); if (config->sbrSignaling == SIG_EXPLICIT_HIERARCHICAL && config->sbrPresent) { writeSampleRate(asc, config->extSamplingRate, 4); writeAot(asc, config->aot); } switch (config->aot) { case AOT_AAC_MAIN: case AOT_AAC_LC: case AOT_AAC_SSR: case AOT_AAC_LTP: case AOT_AAC_SCAL: case AOT_TWIN_VQ: case AOT_ER_AAC_LC: case AOT_ER_AAC_LTP: case AOT_ER_AAC_SCAL: case AOT_ER_TWIN_VQ: case AOT_ER_BSAC: case AOT_ER_AAC_LD: err = transportEnc_writeGASpecificConfig(asc, config, extFlag, alignAnchor); if (err) return err; break; case AOT_ER_AAC_ELD: err = transportEnc_writeELDSpecificConfig(asc, config, epConfig, cb); if (err) return err; break; case AOT_USAC: err = transportEnc_writeUsacSpecificConfig(asc, extFlag, config, cb); if (err) { return err; } break; default: return -1; } switch (config->aot) { case AOT_ER_AAC_LC: case AOT_ER_AAC_LTP: case AOT_ER_AAC_SCAL: case AOT_ER_TWIN_VQ: case AOT_ER_BSAC: case AOT_ER_AAC_LD: case AOT_ER_CELP: case AOT_ER_HVXC: case AOT_ER_HILN: case AOT_ER_PARA: case AOT_ER_AAC_ELD: FDKwriteBits(asc, 0, 2); /* epconfig 0 */ break; default: break; } /* backward compatible explicit signaling of extension AOT */ if (config->sbrSignaling == SIG_EXPLICIT_BW_COMPATIBLE) { TP_ASC_EXTENSION_ID ascExtId = ASCEXT_UNKOWN; if (config->sbrPresent) { ascExtId = ASCEXT_SBR; FDKwriteBits(asc, ascExtId, 11); writeAot(asc, config->extAOT); FDKwriteBits(asc, 1, 1); /* sbrPresentFlag=1 */ writeSampleRate(asc, config->extSamplingRate, 4); if (config->psPresent) { ascExtId = ASCEXT_PS; FDKwriteBits(asc, ascExtId, 11); FDKwriteBits(asc, 1, 1); /* psPresentFlag=1 */ } } } /* Make sure all bits are sync'ed */ FDKsyncCache(asc); return 0; } fdk-aac-2.0.2/libMpegTPEnc/src/tpenc_asc.h000066400000000000000000000143611404225310700201540ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* MPEG transport format encoder library ********************* Author(s): Manuel Jander Description: Audio Specific Config writer *******************************************************************************/ #ifndef TPENC_ASC_H #define TPENC_ASC_H /** * \brief Get channel config from channel mode. * * \param channel_mode channel mode * \param channel_config_zero no standard channel configuration * * \return chanel config */ int getChannelConfig(const CHANNEL_MODE channel_mode, const UCHAR channel_config_zero); /** * \brief Write a Program Config Element. * * \param hBs bitstream handle into which the PCE is appended * \param channelMode the channel mode to be used * \param sampleRate the sample rate * \param instanceTagPCE the instance tag of the Program Config Element * \param profile the MPEG Audio profile to be used * \param matrix mixdown gain * \param pseudo surround indication * \param reference bitstream position for alignment * \return zero on success, non-zero on failure. */ int transportEnc_writePCE(HANDLE_FDK_BITSTREAM hBs, CHANNEL_MODE channelMode, INT sampleRate, int instanceTagPCE, int profile, int matrixMixdownA, int pseudoSurroundEnable, UINT alignAnchor); /** * \brief Get the bit count required by a Program Config Element * * \param channelMode the channel mode to be used * \param matrix mixdown gain * \param bit offset at which the PCE would start * \return the amount of bits required for the PCE including the given bit * offset. */ int transportEnc_GetPCEBits(CHANNEL_MODE channelMode, int matrixMixdownA, int bits); #endif /* TPENC_ASC_H */ fdk-aac-2.0.2/libMpegTPEnc/src/tpenc_latm.cpp000066400000000000000000000661501404225310700207010ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* MPEG transport format encoder library ********************* Author(s): Description: *******************************************************************************/ #include "tpenc_latm.h" #include "genericStds.h" static const short celpFrameLengthTable[64] = { 154, 170, 186, 147, 156, 165, 114, 120, 186, 126, 132, 138, 142, 146, 154, 166, 174, 182, 190, 198, 206, 210, 214, 110, 114, 118, 120, 122, 218, 230, 242, 254, 266, 278, 286, 294, 318, 342, 358, 374, 390, 406, 422, 136, 142, 148, 154, 160, 166, 170, 174, 186, 198, 206, 214, 222, 230, 238, 216, 160, 280, 338, 0, 0}; /******* write value to transport stream first two bits define the size of the value itself then the value itself, with a size of 0-3 bytes *******/ static UINT transportEnc_LatmWriteValue(HANDLE_FDK_BITSTREAM hBs, int value) { UCHAR valueBytes = 4; unsigned int bitsWritten = 0; int i; if (value < (1 << 8)) { valueBytes = 1; } else if (value < (1 << 16)) { valueBytes = 2; } else if (value < (1 << 24)) { valueBytes = 3; } else { valueBytes = 4; } FDKwriteBits(hBs, valueBytes - 1, 2); /* size of value in Bytes */ for (i = 0; i < valueBytes; i++) { /* write most significant Byte first */ FDKwriteBits(hBs, (UCHAR)(value >> ((valueBytes - 1 - i) << 3)), 8); } bitsWritten = (valueBytes << 3) + 2; return bitsWritten; } static UINT transportEnc_LatmCountFixBitDemandHeader(HANDLE_LATM_STREAM hAss) { int bitDemand = 0; int insertSetupData = 0; /* only if start of new latm frame */ if (hAss->subFrameCnt == 0) { /* AudioSyncStream */ if (hAss->tt == TT_MP4_LOAS) { bitDemand += 11; /* syncword */ bitDemand += 13; /* audioMuxLengthBytes */ } /* AudioMuxElement*/ /* AudioMuxElement::Stream Mux Config */ if (hAss->muxConfigPeriod > 0) { insertSetupData = (hAss->latmFrameCounter == 0); } else { insertSetupData = 0; } if (hAss->tt != TT_MP4_LATM_MCP0) { /* AudioMuxElement::useSameStreamMux Flag */ bitDemand += 1; if (insertSetupData) { bitDemand += hAss->streamMuxConfigBits; } } /* AudioMuxElement::otherDataBits */ bitDemand += hAss->otherDataLenBits; /* AudioMuxElement::ByteAlign */ if (bitDemand % 8) { hAss->fillBits = 8 - (bitDemand % 8); bitDemand += hAss->fillBits; } else { hAss->fillBits = 0; } } return bitDemand; } static UINT transportEnc_LatmCountVarBitDemandHeader( HANDLE_LATM_STREAM hAss, unsigned int streamDataLength) { int bitDemand = 0; int prog, layer; /* Payload Length Info*/ if (hAss->allStreamsSameTimeFraming) { for (prog = 0; prog < hAss->noProgram; prog++) { for (layer = 0; layer < LATM_MAX_LAYERS; layer++) { LATM_LAYER_INFO *p_linfo = &(hAss->m_linfo[prog][layer]); if (p_linfo->streamID >= 0) { switch (p_linfo->frameLengthType) { case 0: if (streamDataLength > 0) { streamDataLength -= bitDemand; while (streamDataLength >= (255 << 3)) { bitDemand += 8; streamDataLength -= (255 << 3); } bitDemand += 8; } break; case 1: case 4: case 6: bitDemand += 2; break; default: return 0; } } } } } else { /* there are many possibilities to use this mechanism. */ switch (hAss->varMode) { case LATMVAR_SIMPLE_SEQUENCE: { /* Use the sequence generated by the encoder */ // int streamCntPosition = transportEnc_SetWritePointer( // hAss->hAssemble, 0 ); int streamCntPosition = FDKgetValidBits( // hAss->hAssemble ); bitDemand += 4; hAss->varStreamCnt = 0; for (prog = 0; prog < hAss->noProgram; prog++) { for (layer = 0; layer < LATM_MAX_LAYERS; layer++) { LATM_LAYER_INFO *p_linfo = &(hAss->m_linfo[prog][layer]); if (p_linfo->streamID >= 0) { bitDemand += 4; /* streamID */ switch (p_linfo->frameLengthType) { case 0: streamDataLength -= bitDemand; while (streamDataLength >= (255 << 3)) { bitDemand += 8; streamDataLength -= (255 << 3); } bitDemand += 8; break; /*bitDemand += 1; endFlag break;*/ case 1: case 4: case 6: break; default: return 0; } hAss->varStreamCnt++; } } } bitDemand += 4; // transportEnc_UpdateBitstreamField( hAss->hAssemble, // streamCntPosition, hAss->varStreamCnt-1, 4 ); UINT pos = // streamCntPosition-FDKgetValidBits(hAss->hAssemble); FDKpushBack( // hAss->hAssemble, pos); FDKwriteBits( hAss->hAssemble, // hAss->varStreamCnt-1, 4); FDKpushFor( hAss->hAssemble, pos-4); } break; default: return 0; } } return bitDemand; } TRANSPORTENC_ERROR CreateStreamMuxConfig(HANDLE_LATM_STREAM hAss, HANDLE_FDK_BITSTREAM hBs, int bufferFullness, CSTpCallBacks *cb) { INT streamIDcnt, tmp; int layer, prog; USHORT coreFrameOffset = 0; hAss->taraBufferFullness = 0xFF; hAss->audioMuxVersionA = 0; /* for future extensions */ hAss->streamMuxConfigBits = 0; FDKwriteBits(hBs, hAss->audioMuxVersion, 1); /* audioMuxVersion */ hAss->streamMuxConfigBits += 1; if (hAss->audioMuxVersion == 1) { FDKwriteBits(hBs, hAss->audioMuxVersionA, 1); /* audioMuxVersionA */ hAss->streamMuxConfigBits += 1; } if (hAss->audioMuxVersionA == 0) { if (hAss->audioMuxVersion == 1) { hAss->streamMuxConfigBits += transportEnc_LatmWriteValue( hBs, hAss->taraBufferFullness); /* taraBufferFullness */ } FDKwriteBits(hBs, hAss->allStreamsSameTimeFraming ? 1 : 0, 1); /* allStreamsSameTimeFraming */ FDKwriteBits(hBs, hAss->noSubframes - 1, 6); /* Number of Subframes */ FDKwriteBits(hBs, hAss->noProgram - 1, 4); /* Number of Programs */ hAss->streamMuxConfigBits += 11; streamIDcnt = 0; for (prog = 0; prog < hAss->noProgram; prog++) { int transLayer = 0; FDKwriteBits(hBs, hAss->noLayer[prog] - 1, 3); hAss->streamMuxConfigBits += 3; for (layer = 0; layer < LATM_MAX_LAYERS; layer++) { LATM_LAYER_INFO *p_linfo = &(hAss->m_linfo[prog][layer]); CODER_CONFIG *p_lci = hAss->config[prog][layer]; p_linfo->streamID = -1; if (hAss->config[prog][layer] != NULL) { int useSameConfig = 0; if (transLayer > 0) { FDKwriteBits(hBs, useSameConfig ? 1 : 0, 1); hAss->streamMuxConfigBits += 1; } if ((useSameConfig == 0) || (transLayer == 0)) { const UINT alignAnchor = FDKgetValidBits(hBs); if (0 != (transportEnc_writeASC(hBs, hAss->config[prog][layer], cb))) { return TRANSPORTENC_UNKOWN_ERROR; } if (hAss->audioMuxVersion == 1) { UINT ascLen = transportEnc_LatmWriteValue(hBs, 0); FDKbyteAlign(hBs, alignAnchor); ascLen = FDKgetValidBits(hBs) - alignAnchor - ascLen; FDKpushBack(hBs, FDKgetValidBits(hBs) - alignAnchor); transportEnc_LatmWriteValue(hBs, ascLen); if (0 != (transportEnc_writeASC(hBs, hAss->config[prog][layer], cb))) { return TRANSPORTENC_UNKOWN_ERROR; } FDKbyteAlign(hBs, alignAnchor); /* asc length fillbits */ } hAss->streamMuxConfigBits += FDKgetValidBits(hBs) - alignAnchor; /* add asc length to smc summary */ } transLayer++; if (!hAss->allStreamsSameTimeFraming) { if (streamIDcnt >= LATM_MAX_STREAM_ID) return TRANSPORTENC_INVALID_CONFIG; } p_linfo->streamID = streamIDcnt++; switch (p_lci->aot) { case AOT_AAC_MAIN: case AOT_AAC_LC: case AOT_AAC_SSR: case AOT_AAC_LTP: case AOT_AAC_SCAL: case AOT_ER_AAC_LD: case AOT_ER_AAC_ELD: case AOT_USAC: p_linfo->frameLengthType = 0; FDKwriteBits(hBs, p_linfo->frameLengthType, 3); /* frameLengthType */ FDKwriteBits(hBs, bufferFullness, 8); /* bufferFullness */ hAss->streamMuxConfigBits += 11; if (!hAss->allStreamsSameTimeFraming) { CODER_CONFIG *p_lci_prev = hAss->config[prog][layer - 1]; if (((p_lci->aot == AOT_AAC_SCAL) || (p_lci->aot == AOT_ER_AAC_SCAL)) && ((p_lci_prev->aot == AOT_CELP) || (p_lci_prev->aot == AOT_ER_CELP))) { FDKwriteBits(hBs, coreFrameOffset, 6); /* coreFrameOffset */ hAss->streamMuxConfigBits += 6; } } break; case AOT_TWIN_VQ: p_linfo->frameLengthType = 1; tmp = ((p_lci->bitsFrame + 7) >> 3) - 20; /* transmission frame length in bytes */ if ((tmp < 0)) { return TRANSPORTENC_INVALID_TRANSMISSION_FRAME_LENGTH; } FDKwriteBits(hBs, p_linfo->frameLengthType, 3); /* frameLengthType */ FDKwriteBits(hBs, tmp, 9); hAss->streamMuxConfigBits += 12; p_linfo->frameLengthBits = (tmp + 20) << 3; break; case AOT_CELP: p_linfo->frameLengthType = 4; FDKwriteBits(hBs, p_linfo->frameLengthType, 3); /* frameLengthType */ hAss->streamMuxConfigBits += 3; { int i; for (i = 0; i < 62; i++) { if (celpFrameLengthTable[i] == p_lci->bitsFrame) break; } if (i >= 62) { return TRANSPORTENC_INVALID_CELP_FRAME_LENGTH; } FDKwriteBits(hBs, i, 6); /* CELPframeLengthTabelIndex */ hAss->streamMuxConfigBits += 6; } p_linfo->frameLengthBits = p_lci->bitsFrame; break; case AOT_HVXC: p_linfo->frameLengthType = 6; FDKwriteBits(hBs, p_linfo->frameLengthType, 3); /* frameLengthType */ hAss->streamMuxConfigBits += 3; { int i; if (p_lci->bitsFrame == 40) { i = 0; } else if (p_lci->bitsFrame == 80) { i = 1; } else { return TRANSPORTENC_INVALID_FRAME_BITS; } FDKwriteBits(hBs, i, 1); /* HVXCframeLengthTableIndex */ hAss->streamMuxConfigBits += 1; } p_linfo->frameLengthBits = p_lci->bitsFrame; break; case AOT_NULL_OBJECT: default: return TRANSPORTENC_INVALID_AOT; } } } } FDKwriteBits(hBs, (hAss->otherDataLenBits > 0) ? 1 : 0, 1); /* otherDataPresent */ hAss->streamMuxConfigBits += 1; if (hAss->otherDataLenBits > 0) { FDKwriteBits(hBs, 0, 1); FDKwriteBits(hBs, hAss->otherDataLenBits, 8); hAss->streamMuxConfigBits += 9; } FDKwriteBits(hBs, 0, 1); /* crcCheckPresent=0 */ hAss->streamMuxConfigBits += 1; } else { /* if ( audioMuxVersionA == 0 ) */ /* for future extensions */ } return TRANSPORTENC_OK; } static TRANSPORTENC_ERROR WriteAuPayloadLengthInfo( HANDLE_FDK_BITSTREAM hBitStream, int AuLengthBits) { int restBytes; if (AuLengthBits % 8) return TRANSPORTENC_INVALID_AU_LENGTH; while (AuLengthBits >= 255 * 8) { FDKwriteBits(hBitStream, 255, 8); /* 255 shows incomplete AU */ AuLengthBits -= (255 * 8); } restBytes = (AuLengthBits) >> 3; FDKwriteBits(hBitStream, restBytes, 8); return TRANSPORTENC_OK; } static TRANSPORTENC_ERROR transportEnc_LatmSetNrOfSubframes( HANDLE_LATM_STREAM hAss, INT noSubframes_next) /* nr of access units / payloads within a latm frame */ { /* sanity chk */ if (noSubframes_next < 1 || noSubframes_next > MAX_NR_OF_SUBFRAMES) { return TRANSPORTENC_LATM_INVALID_NR_OF_SUBFRAMES; } hAss->noSubframes_next = noSubframes_next; /* if at start then we can take over the value immediately, otherwise we have * to wait for the next SMC */ if ((hAss->subFrameCnt == 0) && (hAss->latmFrameCounter == 0)) { hAss->noSubframes = noSubframes_next; } return TRANSPORTENC_OK; } static int allStreamsSameTimeFraming(HANDLE_LATM_STREAM hAss, UCHAR noProgram, UCHAR noLayer[] /* return */) { int prog, layer; signed int lastNoSamples = -1; signed int minFrameSamples = FDK_INT_MAX; signed int maxFrameSamples = 0; signed int highestSamplingRate = -1; for (prog = 0; prog < noProgram; prog++) { noLayer[prog] = 0; for (layer = 0; layer < LATM_MAX_LAYERS; layer++) { if (hAss->config[prog][layer] != NULL) { INT hsfSamplesFrame; noLayer[prog]++; if (highestSamplingRate < 0) highestSamplingRate = hAss->config[prog][layer]->samplingRate; hsfSamplesFrame = hAss->config[prog][layer]->samplesPerFrame * highestSamplingRate / hAss->config[prog][layer]->samplingRate; if (hsfSamplesFrame <= minFrameSamples) minFrameSamples = hsfSamplesFrame; if (hsfSamplesFrame >= maxFrameSamples) maxFrameSamples = hsfSamplesFrame; if (lastNoSamples == -1) { lastNoSamples = hsfSamplesFrame; } else { if (hsfSamplesFrame != lastNoSamples) { return 0; } } } } } return 1; } /** * Initialize LATM/LOAS Stream and add layer 0 at program 0. */ static TRANSPORTENC_ERROR transportEnc_InitLatmStream( HANDLE_LATM_STREAM hAss, int fractDelayPresent, signed int muxConfigPeriod, /* insert setup data every muxConfigPeriod frames */ UINT audioMuxVersion, TRANSPORT_TYPE tt) { TRANSPORTENC_ERROR ErrorStatus = TRANSPORTENC_OK; if (hAss == NULL) return TRANSPORTENC_INVALID_PARAMETER; hAss->tt = tt; hAss->noProgram = 1; hAss->audioMuxVersion = audioMuxVersion; /* Fill noLayer array using hAss->config */ hAss->allStreamsSameTimeFraming = allStreamsSameTimeFraming(hAss, hAss->noProgram, hAss->noLayer); /* Only allStreamsSameTimeFraming==1 is supported */ FDK_ASSERT(hAss->allStreamsSameTimeFraming); hAss->fractDelayPresent = fractDelayPresent; hAss->otherDataLenBits = 0; hAss->varMode = LATMVAR_SIMPLE_SEQUENCE; /* initialize counters */ hAss->subFrameCnt = 0; hAss->noSubframes = DEFAULT_LATM_NR_OF_SUBFRAMES; hAss->noSubframes_next = DEFAULT_LATM_NR_OF_SUBFRAMES; /* sync layer related */ hAss->audioMuxLengthBytes = 0; hAss->latmFrameCounter = 0; hAss->muxConfigPeriod = muxConfigPeriod; return ErrorStatus; } /** * */ UINT transportEnc_LatmCountTotalBitDemandHeader(HANDLE_LATM_STREAM hAss, unsigned int streamDataLength) { UINT bitDemand = 0; switch (hAss->tt) { case TT_MP4_LOAS: case TT_MP4_LATM_MCP0: case TT_MP4_LATM_MCP1: if (hAss->subFrameCnt == 0) { bitDemand = transportEnc_LatmCountFixBitDemandHeader(hAss); } bitDemand += transportEnc_LatmCountVarBitDemandHeader( hAss, streamDataLength /*- bitDemand*/); break; default: break; } return bitDemand; } static TRANSPORTENC_ERROR AdvanceAudioMuxElement(HANDLE_LATM_STREAM hAss, HANDLE_FDK_BITSTREAM hBs, int auBits, int bufferFullness, CSTpCallBacks *cb) { TRANSPORTENC_ERROR ErrorStatus = TRANSPORTENC_OK; int insertMuxSetup; /* Insert setup data to assemble Buffer */ if (hAss->subFrameCnt == 0) { if (hAss->muxConfigPeriod > 0) { insertMuxSetup = (hAss->latmFrameCounter == 0); } else { insertMuxSetup = 0; } if (hAss->tt != TT_MP4_LATM_MCP0) { if (insertMuxSetup) { FDKwriteBits(hBs, 0, 1); /* useSameStreamMux useNewStreamMux */ if (TRANSPORTENC_OK != (ErrorStatus = CreateStreamMuxConfig( hAss, hBs, bufferFullness, cb))) { return ErrorStatus; } } else { FDKwriteBits(hBs, 1, 1); /* useSameStreamMux */ } } } /* PayloadLengthInfo */ { int prog, layer; for (prog = 0; prog < hAss->noProgram; prog++) { for (layer = 0; layer < hAss->noLayer[prog]; layer++) { ErrorStatus = WriteAuPayloadLengthInfo(hBs, auBits); if (ErrorStatus != TRANSPORTENC_OK) return ErrorStatus; } } } /* At this point comes the access unit. */ return TRANSPORTENC_OK; } TRANSPORTENC_ERROR transportEnc_LatmWrite(HANDLE_LATM_STREAM hAss, HANDLE_FDK_BITSTREAM hBs, int auBits, int bufferFullness, CSTpCallBacks *cb) { TRANSPORTENC_ERROR ErrorStatus; if (hAss->subFrameCnt == 0) { /* Start new frame */ FDKresetBitbuffer(hBs, BS_WRITER); } hAss->latmSubframeStart = FDKgetValidBits(hBs); /* Insert syncword and syncword distance - only if loas - we must update the syncword distance (=audiomuxlengthbytes) later */ if (hAss->tt == TT_MP4_LOAS && hAss->subFrameCnt == 0) { /* Start new LOAS frame */ FDKwriteBits(hBs, 0x2B7, 11); hAss->audioMuxLengthBytes = 0; hAss->audioMuxLengthBytesPos = FDKgetValidBits(hBs); /* store read pointer position */ FDKwriteBits(hBs, hAss->audioMuxLengthBytes, 13); } ErrorStatus = AdvanceAudioMuxElement(hAss, hBs, auBits, bufferFullness, cb); if (ErrorStatus != TRANSPORTENC_OK) return ErrorStatus; return ErrorStatus; } void transportEnc_LatmAdjustSubframeBits(HANDLE_LATM_STREAM hAss, int *bits) { /* Substract bits from possible previous subframe */ *bits -= hAss->latmSubframeStart; /* Add fill bits */ if (hAss->subFrameCnt == 0) { *bits += hAss->otherDataLenBits; *bits += hAss->fillBits; } } TRANSPORTENC_ERROR transportEnc_LatmGetFrame(HANDLE_LATM_STREAM hAss, HANDLE_FDK_BITSTREAM hBs, int *pBytes) { TRANSPORTENC_ERROR ErrorStatus = TRANSPORTENC_OK; hAss->subFrameCnt++; if (hAss->subFrameCnt >= hAss->noSubframes) { /* Add LOAS frame length if required. */ if (hAss->tt == TT_MP4_LOAS) { FDK_BITSTREAM tmpBuf; /* Determine frame length info */ hAss->audioMuxLengthBytes = ((FDKgetValidBits(hBs) + hAss->otherDataLenBits + 7) >> 3) - 3; /* 3=Syncword + length */ /* Check frame length info */ if (hAss->audioMuxLengthBytes >= (1 << 13)) { ErrorStatus = TRANSPORTENC_INVALID_AU_LENGTH; goto bail; } /* Write length info into assembler buffer */ FDKinitBitStream(&tmpBuf, hBs->hBitBuf.Buffer, hBs->hBitBuf.bufSize, 0, BS_WRITER); FDKpushFor(&tmpBuf, hAss->audioMuxLengthBytesPos); FDKwriteBits(&tmpBuf, hAss->audioMuxLengthBytes, 13); FDKsyncCache(&tmpBuf); } /* Write AudioMuxElement other data bits */ FDKwriteBits(hBs, 0, hAss->otherDataLenBits); /* Write AudioMuxElement byte alignment fill bits */ FDKwriteBits(hBs, 0, hAss->fillBits); FDK_ASSERT((FDKgetValidBits(hBs) % 8) == 0); hAss->subFrameCnt = 0; FDKsyncCache(hBs); *pBytes = (FDKgetValidBits(hBs) + 7) >> 3; if (hAss->muxConfigPeriod > 0) { hAss->latmFrameCounter++; if (hAss->latmFrameCounter >= hAss->muxConfigPeriod) { hAss->latmFrameCounter = 0; hAss->noSubframes = hAss->noSubframes_next; } } } else { /* No data this time */ *pBytes = 0; } bail: return ErrorStatus; } /** * Init LATM/LOAS */ TRANSPORTENC_ERROR transportEnc_Latm_Init(HANDLE_LATM_STREAM hAss, HANDLE_FDK_BITSTREAM hBs, CODER_CONFIG *layerConfig, UINT audioMuxVersion, TRANSPORT_TYPE tt, CSTpCallBacks *cb) { TRANSPORTENC_ERROR ErrorStatus; int fractDelayPresent = 0; int prog, layer; int setupDataDistanceFrames = layerConfig->headerPeriod; FDK_ASSERT(setupDataDistanceFrames >= 0); for (prog = 0; prog < LATM_MAX_PROGRAMS; prog++) { for (layer = 0; layer < LATM_MAX_LAYERS; layer++) { hAss->config[prog][layer] = NULL; hAss->m_linfo[prog][layer].streamID = -1; } } hAss->config[0][0] = layerConfig; hAss->m_linfo[0][0].streamID = 0; ErrorStatus = transportEnc_InitLatmStream(hAss, fractDelayPresent, setupDataDistanceFrames, (audioMuxVersion) ? 1 : 0, tt); if (ErrorStatus != TRANSPORTENC_OK) goto bail; ErrorStatus = transportEnc_LatmSetNrOfSubframes(hAss, layerConfig->nSubFrames); if (ErrorStatus != TRANSPORTENC_OK) goto bail; /* Get the size of the StreamMuxConfig somehow */ if (TRANSPORTENC_OK != (ErrorStatus = AdvanceAudioMuxElement(hAss, hBs, 0, 0, cb))) { goto bail; } // CreateStreamMuxConfig(hAss, hBs, 0); bail: return ErrorStatus; } TRANSPORTENC_ERROR transportEnc_LatmAddOtherDataBits(HANDLE_LATM_STREAM hAss, const int otherDataBits) { TRANSPORTENC_ERROR ErrorStatus = TRANSPORTENC_OK; if ((hAss->otherDataLenBits != 0) || (otherDataBits % 8 != 0)) { /* This implementation allows to add other data bits only once. To keep existing alignment only whole bytes are allowed. */ ErrorStatus = TRANSPORTENC_UNKOWN_ERROR; } else { /* Ensure correct addional bits in payload. */ if (hAss->tt == TT_MP4_LATM_MCP0) { hAss->otherDataLenBits = otherDataBits; } else { hAss->otherDataLenBits = otherDataBits - 9; hAss->streamMuxConfigBits += 9; } } return ErrorStatus; } fdk-aac-2.0.2/libMpegTPEnc/src/tpenc_latm.h000066400000000000000000000243411404225310700203420ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* MPEG transport format encoder library ********************* Author(s): Description: *******************************************************************************/ #ifndef TPENC_LATM_H #define TPENC_LATM_H #include "tpenc_lib.h" #include "FDK_bitstream.h" #define DEFAULT_LATM_NR_OF_SUBFRAMES 1 #define DEFAULT_LATM_SMC_REPEAT 8 #define MAX_AAC_LAYERS 9 #define LATM_MAX_PROGRAMS 1 #define LATM_MAX_STREAM_ID 16 #define LATM_MAX_LAYERS 1 /*MAX_AAC_LAYERS*/ #define MAX_NR_OF_SUBFRAMES \ 2 /* set this carefully to avoid buffer overflows \ */ typedef enum { LATMVAR_SIMPLE_SEQUENCE } LATM_VAR_MODE; typedef struct { signed int frameLengthType; signed int frameLengthBits; signed int varFrameLengthTable[4]; signed int streamID; } LATM_LAYER_INFO; typedef struct { LATM_LAYER_INFO m_linfo[LATM_MAX_PROGRAMS][LATM_MAX_LAYERS]; CODER_CONFIG *config[LATM_MAX_PROGRAMS][LATM_MAX_LAYERS]; LATM_VAR_MODE varMode; TRANSPORT_TYPE tt; int audioMuxLengthBytes; int audioMuxLengthBytesPos; int taraBufferFullness; /* state of the bit reservoir */ int varStreamCnt; UCHAR latmFrameCounter; /* Current frame number. Counts modulo muxConfigPeriod */ UCHAR muxConfigPeriod; /* Distance in frames between MuxConfig */ UCHAR audioMuxVersion; /* AMV1 supports transmission of taraBufferFullness and ASC lengths */ UCHAR audioMuxVersionA; /* for future extensions */ UCHAR noProgram; UCHAR noLayer[LATM_MAX_PROGRAMS]; UCHAR fractDelayPresent; UCHAR allStreamsSameTimeFraming; UCHAR subFrameCnt; /* Current Subframe frame */ UCHAR noSubframes; /* Number of subframes */ UINT latmSubframeStart; /* Position of current subframe start */ UCHAR noSubframes_next; UCHAR otherDataLenBits; /* AudioMuxElement other data bits */ UCHAR fillBits; /* AudioMuxElement fill bits */ UINT streamMuxConfigBits; } LATM_STREAM; typedef LATM_STREAM *HANDLE_LATM_STREAM; /** * \brief Initialize LATM_STREAM Handle. Creates automatically one program with * one layer with the given layerConfig. The layerConfig must be persisten * because references to this pointer are made at any time again. Use * transportEnc_Latm_AddLayer() to add more programs/layers. * * \param hLatmStreamInfo HANDLE_LATM_STREAM handle * \param hBs Bitstream handle * \param layerConfig a valid CODER_CONFIG struct containing the current audio * configuration parameters * \param audioMuxVersion the LATM audioMuxVersion to be used * \param tt the specific TRANSPORT_TYPE to be used, either TT_MP4_LOAS, * TT_MP4_LATM_MCP1 or TT_MP4_LATM_MCP0 LOAS * \param cb callback information structure. * * \return an TRANSPORTENC_ERROR error code */ TRANSPORTENC_ERROR transportEnc_Latm_Init(HANDLE_LATM_STREAM hLatmStreamInfo, HANDLE_FDK_BITSTREAM hBs, CODER_CONFIG *layerConfig, UINT audioMuxVersion, TRANSPORT_TYPE tt, CSTpCallBacks *cb); /** * \brief Write addional other data bits in AudioMuxElement * * \param hAss HANDLE_LATM_STREAM handle * \param otherDataBits number of other data bits to be written * * \return an TRANSPORTENC_ERROR error code */ TRANSPORTENC_ERROR transportEnc_LatmAddOtherDataBits(HANDLE_LATM_STREAM hAss, const int otherDataBits); /** * \brief Get bit demand of next LATM/LOAS header * * \param hAss HANDLE_LATM_STREAM handle * \param streamDataLength the length of the payload * * \return the number of bits required by the LATM/LOAS headers */ unsigned int transportEnc_LatmCountTotalBitDemandHeader( HANDLE_LATM_STREAM hAss, unsigned int streamDataLength); /** * \brief Write LATM/LOAS header into given bitstream handle * * \param hLatmStreamInfo HANDLE_LATM_STREAM handle * \param hBitstream Bitstream handle * \param auBits amount of current payload bits * \param bufferFullness LATM buffer fullness value * \param cb callback information structure. * * \return an TRANSPORTENC_ERROR error code */ TRANSPORTENC_ERROR transportEnc_LatmWrite(HANDLE_LATM_STREAM hAss, HANDLE_FDK_BITSTREAM hBitstream, int auBits, int bufferFullness, CSTpCallBacks *cb); /** * \brief Adjust bit count relative to current subframe * * \param hAss HANDLE_LATM_STREAM handle * \param pBits pointer to an int, where the current frame bit count is * contained, and where the subframe relative bit count will be returned into * * \return void */ void transportEnc_LatmAdjustSubframeBits(HANDLE_LATM_STREAM hAss, int *pBits); /** * \brief Request an LATM frame, which may, or may not be available * * \param hAss HANDLE_LATM_STREAM handle * \param hBs Bitstream handle * \param pBytes pointer to an int, where the current frame byte count stored * into. A return value of zero means that currently no LATM/LOAS frame can be * returned. The latter is expected in case of multiple subframes being * used. * * \return an TRANSPORTENC_ERROR error code */ TRANSPORTENC_ERROR transportEnc_LatmGetFrame(HANDLE_LATM_STREAM hAss, HANDLE_FDK_BITSTREAM hBs, int *pBytes); /** * \brief Write a StreamMuxConfig into the given bitstream handle * * \param hAss HANDLE_LATM_STREAM handle * \param hBs Bitstream handle * \param bufferFullness LATM buffer fullness value * \param cb callback information structure. * * \return void */ TRANSPORTENC_ERROR CreateStreamMuxConfig(HANDLE_LATM_STREAM hAss, HANDLE_FDK_BITSTREAM hBs, int bufferFullness, CSTpCallBacks *cb); #endif /* TPENC_LATM_H */ fdk-aac-2.0.2/libMpegTPEnc/src/tpenc_lib.cpp000066400000000000000000000536411404225310700205130ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /******************* MPEG transport format encoder library ********************* Author(s): Manuel Jander Description: MPEG Transport encode *******************************************************************************/ #include "tpenc_lib.h" /* library info */ #include "tp_version.h" #define MODULE_NAME "transportEnc" #include "tpenc_asc.h" #include "tpenc_adts.h" #include "tpenc_adif.h" #include "tpenc_latm.h" typedef struct { int curSubFrame; int nSubFrames; int prevBits; } RAWPACKETS_INFO; struct TRANSPORTENC { CODER_CONFIG config; TRANSPORT_TYPE transportFmt; /*!< MPEG4 transport type. */ FDK_BITSTREAM bitStream; UCHAR *bsBuffer; INT bsBufferSize; INT pceFrameCounter; /*!< Indicates frame period when PCE must be written in raw_data_block. -1 means not to write a PCE in raw_dat_block. */ union { STRUCT_ADTS adts; ADIF_INFO adif; LATM_STREAM latm; RAWPACKETS_INFO raw; } writer; CSTpCallBacks callbacks; }; typedef struct _TRANSPORTENC_STRUCT TRANSPORTENC_STRUCT; /* * MEMORY Declaration */ C_ALLOC_MEM(Ram_TransportEncoder, struct TRANSPORTENC, 1) TRANSPORTENC_ERROR transportEnc_Open(HANDLE_TRANSPORTENC *phTpEnc) { HANDLE_TRANSPORTENC hTpEnc; if (phTpEnc == NULL) { return TRANSPORTENC_INVALID_PARAMETER; } hTpEnc = GetRam_TransportEncoder(0); if (hTpEnc == NULL) { return TRANSPORTENC_NO_MEM; } *phTpEnc = hTpEnc; return TRANSPORTENC_OK; } /** * \brief Get frame period of PCE in raw_data_block. * * - Write PCE only if necessary. PCE can be part of the ASC if chConfig==0 * whererfore no additonal PCE will be written in raw_data_block. * - A matrixMixdown coefficient can only be written if chConfig is 5.0 or 5.1. * - The PCE repetition rate in raw_data_block can be controlled via * headerPeriod parameter. * * \param channelMode Encoder Channel Mode. * \param channelConfigZero No standard channel configuration. * \param transportFmt Format of the transport to be written. * \param headerPeriod Chosen PCE frame repetition rate. * \param matrixMixdownA Indicates if a valid Matrix Mixdown coefficient * is available. * * \return PCE frame repetition rate. -1 means no PCE present in * raw_data_block. */ static INT getPceRepetitionRate(const CHANNEL_MODE channelMode, const int channelConfigZero, const TRANSPORT_TYPE transportFmt, const int headerPeriod, const int matrixMixdownA) { INT pceFrameCounter = -1; /* variable to be returned */ if (headerPeriod > 0) { switch (getChannelConfig(channelMode, channelConfigZero)) { case 0: switch (transportFmt) { case TT_MP4_ADTS: case TT_MP4_LATM_MCP0: case TT_MP4_RAW: pceFrameCounter = headerPeriod; break; case TT_MP4_ADIF: /* ADIF header comprises PCE */ if ((channelMode == MODE_1_2_2) || (channelMode == MODE_1_2_2_1)) { pceFrameCounter = headerPeriod; /* repeating pce only meaningful for potential matrix mixdown */ break; } FDK_FALLTHROUGH; case TT_MP4_LOAS: /* PCE in ASC if chChonfig==0 */ case TT_MP4_LATM_MCP1: /* PCE in ASC if chChonfig==0 */ default: pceFrameCounter = -1; /* no PCE in raw_data_block */ } break; case 5: /* MODE_1_2_2 */ case 6: /* MODE_1_2_2_1 */ /* matrixMixdownCoefficient can only be written if 5.0 and 5.1 config * present. */ if (matrixMixdownA != 0) { switch (transportFmt) { case TT_MP4_ADIF: /* ADIF header comprises PCE */ case TT_MP4_ADTS: case TT_MP4_LOAS: /* no PCE in ASC because chConfig!=0 */ case TT_MP4_LATM_MCP1: /* no PCE in ASC because chConfig!=0 */ case TT_MP4_LATM_MCP0: case TT_MP4_RAW: pceFrameCounter = headerPeriod; break; default: pceFrameCounter = -1; /* no PCE in raw_data_block */ } /* switch transportFmt */ } /* if matrixMixdownA!=0 */ break; default: pceFrameCounter = -1; /* no PCE in raw_data_block */ } /* switch getChannelConfig() */ } /* if headerPeriod>0 */ else { pceFrameCounter = -1; /* no PCE in raw_data_block */ } return pceFrameCounter; } TRANSPORTENC_ERROR transportEnc_Init(HANDLE_TRANSPORTENC hTpEnc, UCHAR *bsBuffer, INT bsBufferSize, TRANSPORT_TYPE transportFmt, CODER_CONFIG *cconfig, UINT flags) { /* Copy configuration structure */ FDKmemcpy(&hTpEnc->config, cconfig, sizeof(CODER_CONFIG)); /* Init transportEnc struct. */ hTpEnc->transportFmt = transportFmt; hTpEnc->bsBuffer = bsBuffer; hTpEnc->bsBufferSize = bsBufferSize; FDKinitBitStream(&hTpEnc->bitStream, hTpEnc->bsBuffer, hTpEnc->bsBufferSize, 0, BS_WRITER); switch (transportFmt) { case TT_MP4_ADIF: /* Sanity checks */ if ((hTpEnc->config.aot != AOT_AAC_LC) || (hTpEnc->config.samplesPerFrame != 1024)) { return TRANSPORTENC_INVALID_PARAMETER; } hTpEnc->writer.adif.headerWritten = 0; hTpEnc->writer.adif.samplingRate = hTpEnc->config.samplingRate; hTpEnc->writer.adif.bitRate = hTpEnc->config.bitRate; hTpEnc->writer.adif.profile = ((int)hTpEnc->config.aot) - 1; hTpEnc->writer.adif.cm = hTpEnc->config.channelMode; hTpEnc->writer.adif.bVariableRate = 0; hTpEnc->writer.adif.instanceTag = 0; hTpEnc->writer.adif.matrixMixdownA = hTpEnc->config.matrixMixdownA; hTpEnc->writer.adif.pseudoSurroundEnable = (hTpEnc->config.flags & CC_PSEUDO_SURROUND) ? 1 : 0; break; case TT_MP4_ADTS: /* Sanity checks */ if ((hTpEnc->config.aot != AOT_AAC_LC) || (hTpEnc->config.samplesPerFrame != 1024)) { return TRANSPORTENC_INVALID_PARAMETER; } if (adtsWrite_Init(&hTpEnc->writer.adts, &hTpEnc->config) != 0) { return TRANSPORTENC_INVALID_PARAMETER; } break; case TT_MP4_LOAS: case TT_MP4_LATM_MCP0: case TT_MP4_LATM_MCP1: { TRANSPORTENC_ERROR error; error = transportEnc_Latm_Init(&hTpEnc->writer.latm, &hTpEnc->bitStream, &hTpEnc->config, flags & TP_FLAG_LATM_AMV, transportFmt, &hTpEnc->callbacks); if (error != TRANSPORTENC_OK) { return error; } } break; case TT_MP4_RAW: hTpEnc->writer.raw.curSubFrame = 0; hTpEnc->writer.raw.nSubFrames = hTpEnc->config.nSubFrames; break; default: return TRANSPORTENC_INVALID_PARAMETER; } /* pceFrameCounter indicates if PCE must be written in raw_data_block. */ hTpEnc->pceFrameCounter = getPceRepetitionRate( hTpEnc->config.channelMode, hTpEnc->config.channelConfigZero, transportFmt, hTpEnc->config.headerPeriod, hTpEnc->config.matrixMixdownA); return TRANSPORTENC_OK; } TRANSPORTENC_ERROR transportEnc_AddOtherDataBits(HANDLE_TRANSPORTENC hTpEnc, const int nBits) { TRANSPORTENC_ERROR tpErr = TRANSPORTENC_OK; switch (hTpEnc->transportFmt) { case TT_MP4_LATM_MCP0: case TT_MP4_LATM_MCP1: case TT_MP4_LOAS: tpErr = transportEnc_LatmAddOtherDataBits(&hTpEnc->writer.latm, nBits); break; case TT_MP4_ADTS: case TT_MP4_ADIF: case TT_MP4_RAW: default: tpErr = TRANSPORTENC_UNKOWN_ERROR; } return tpErr; } HANDLE_FDK_BITSTREAM transportEnc_GetBitstream(HANDLE_TRANSPORTENC hTp) { return &hTp->bitStream; } int transportEnc_RegisterSbrCallback(HANDLE_TRANSPORTENC hTpEnc, const cbSbr_t cbSbr, void *user_data) { if (hTpEnc == NULL) { return -1; } hTpEnc->callbacks.cbSbr = cbSbr; hTpEnc->callbacks.cbSbrData = user_data; return 0; } int transportEnc_RegisterUsacCallback(HANDLE_TRANSPORTENC hTpEnc, const cbUsac_t cbUsac, void *user_data) { if (hTpEnc == NULL) { return -1; } hTpEnc->callbacks.cbUsac = cbUsac; hTpEnc->callbacks.cbUsacData = user_data; return 0; } int transportEnc_RegisterSscCallback(HANDLE_TRANSPORTENC hTpEnc, const cbSsc_t cbSsc, void *user_data) { if (hTpEnc == NULL) { return -1; } hTpEnc->callbacks.cbSsc = cbSsc; hTpEnc->callbacks.cbSscData = user_data; return 0; } TRANSPORTENC_ERROR transportEnc_WriteAccessUnit(HANDLE_TRANSPORTENC hTp, INT frameUsedBits, int bufferFullness, int ncc) { TRANSPORTENC_ERROR err = TRANSPORTENC_OK; if (!hTp) { return TRANSPORTENC_INVALID_PARAMETER; } HANDLE_FDK_BITSTREAM hBs = &hTp->bitStream; /* In case of writing PCE in raw_data_block frameUsedBits must be adapted. */ if (hTp->pceFrameCounter >= hTp->config.headerPeriod) { frameUsedBits += transportEnc_GetPCEBits( hTp->config.channelMode, hTp->config.matrixMixdownA, 3); /* Consider 3 bits ID signalling in alignment */ } switch (hTp->transportFmt) { case TT_MP4_ADIF: FDKinitBitStream(&hTp->bitStream, hTp->bsBuffer, hTp->bsBufferSize, 0, BS_WRITER); if (0 != adifWrite_EncodeHeader(&hTp->writer.adif, hBs, bufferFullness)) { err = TRANSPORTENC_INVALID_CONFIG; } break; case TT_MP4_ADTS: bufferFullness /= ncc; /* Number of Considered Channels */ bufferFullness /= 32; bufferFullness = FDKmin(0x7FF, bufferFullness); /* Signal variable rate */ adtsWrite_EncodeHeader(&hTp->writer.adts, &hTp->bitStream, bufferFullness, frameUsedBits); break; case TT_MP4_LOAS: case TT_MP4_LATM_MCP0: case TT_MP4_LATM_MCP1: bufferFullness /= ncc; /* Number of Considered Channels */ bufferFullness /= 32; bufferFullness = FDKmin(0xFF, bufferFullness); /* Signal variable rate */ transportEnc_LatmWrite(&hTp->writer.latm, hBs, frameUsedBits, bufferFullness, &hTp->callbacks); break; case TT_MP4_RAW: if (hTp->writer.raw.curSubFrame >= hTp->writer.raw.nSubFrames) { hTp->writer.raw.curSubFrame = 0; FDKinitBitStream(&hTp->bitStream, hTp->bsBuffer, hTp->bsBufferSize, 0, BS_WRITER); } hTp->writer.raw.prevBits = FDKgetValidBits(hBs); break; default: err = TRANSPORTENC_UNSUPPORTED_FORMAT; break; } /* Write PCE in raw_data_block if required */ if (hTp->pceFrameCounter >= hTp->config.headerPeriod) { INT crcIndex = 0; /* Align inside PCE with repsect to the first bit of the raw_data_block() */ UINT alignAnchor = FDKgetValidBits(&hTp->bitStream); /* Write PCE element ID bits */ FDKwriteBits(&hTp->bitStream, ID_PCE, 3); if ((hTp->transportFmt == TT_MP4_ADTS) && !hTp->writer.adts.protection_absent) { crcIndex = adtsWrite_CrcStartReg(&hTp->writer.adts, &hTp->bitStream, 0); } /* Write PCE as first raw_data_block element */ transportEnc_writePCE( &hTp->bitStream, hTp->config.channelMode, hTp->config.samplingRate, 0, 1, hTp->config.matrixMixdownA, (hTp->config.flags & CC_PSEUDO_SURROUND) ? 1 : 0, alignAnchor); if ((hTp->transportFmt == TT_MP4_ADTS) && !hTp->writer.adts.protection_absent) { adtsWrite_CrcEndReg(&hTp->writer.adts, &hTp->bitStream, crcIndex); } hTp->pceFrameCounter = 0; /* reset pce frame counter */ } if (hTp->pceFrameCounter != -1) { hTp->pceFrameCounter++; /* Update pceFrameCounter only if PCE writing is active. */ } return err; } TRANSPORTENC_ERROR transportEnc_EndAccessUnit(HANDLE_TRANSPORTENC hTp, int *bits) { switch (hTp->transportFmt) { case TT_MP4_LATM_MCP0: case TT_MP4_LATM_MCP1: case TT_MP4_LOAS: transportEnc_LatmAdjustSubframeBits(&hTp->writer.latm, bits); break; case TT_MP4_ADTS: adtsWrite_EndRawDataBlock(&hTp->writer.adts, &hTp->bitStream, bits); break; case TT_MP4_ADIF: /* Substract ADIF header from AU bits, not to be considered. */ *bits -= adifWrite_GetHeaderBits(&hTp->writer.adif); hTp->writer.adif.headerWritten = 1; break; case TT_MP4_RAW: *bits -= hTp->writer.raw.prevBits; break; default: break; } return TRANSPORTENC_OK; } TRANSPORTENC_ERROR transportEnc_GetFrame(HANDLE_TRANSPORTENC hTpEnc, int *nbytes) { TRANSPORTENC_ERROR tpErr = TRANSPORTENC_OK; HANDLE_FDK_BITSTREAM hBs = &hTpEnc->bitStream; switch (hTpEnc->transportFmt) { case TT_MP4_LATM_MCP0: case TT_MP4_LATM_MCP1: case TT_MP4_LOAS: *nbytes = hTpEnc->bsBufferSize; tpErr = transportEnc_LatmGetFrame(&hTpEnc->writer.latm, hBs, nbytes); break; case TT_MP4_ADTS: if (hTpEnc->writer.adts.currentBlock >= hTpEnc->writer.adts.num_raw_blocks + 1) { *nbytes = (FDKgetValidBits(hBs) + 7) >> 3; hTpEnc->writer.adts.currentBlock = 0; } else { *nbytes = 0; } break; case TT_MP4_ADIF: FDK_ASSERT((INT)FDKgetValidBits(hBs) >= 0); *nbytes = (FDKgetValidBits(hBs) + 7) >> 3; break; case TT_MP4_RAW: FDKsyncCache(hBs); hTpEnc->writer.raw.curSubFrame++; *nbytes = ((FDKgetValidBits(hBs) - hTpEnc->writer.raw.prevBits) + 7) >> 3; break; default: break; } return tpErr; } INT transportEnc_GetStaticBits(HANDLE_TRANSPORTENC hTp, int auBits) { INT nbits = 0, nPceBits = 0; /* Write PCE within raw_data_block in transport lib. */ if (hTp->pceFrameCounter >= hTp->config.headerPeriod) { nPceBits = transportEnc_GetPCEBits( hTp->config.channelMode, hTp->config.matrixMixdownA, 3); /* Consider 3 bits ID signalling in alignment */ auBits += nPceBits; /* Adapt required raw_data_block bit consumtpion for AU length information e.g. in LATM/LOAS configuration. */ } switch (hTp->transportFmt) { case TT_MP4_ADIF: case TT_MP4_RAW: nbits = 0; /* Do not consider the ADIF header into the total bitrate */ break; case TT_MP4_ADTS: nbits = adtsWrite_GetHeaderBits(&hTp->writer.adts); break; case TT_MP4_LOAS: case TT_MP4_LATM_MCP0: case TT_MP4_LATM_MCP1: nbits = transportEnc_LatmCountTotalBitDemandHeader(&hTp->writer.latm, auBits); break; default: nbits = 0; break; } /* PCE is written in the transport library therefore the bit consumption is * part of the transport static bits. */ nbits += nPceBits; return nbits; } void transportEnc_Close(HANDLE_TRANSPORTENC *phTp) { if (phTp != NULL) { if (*phTp != NULL) { FreeRam_TransportEncoder(phTp); } } } int transportEnc_CrcStartReg(HANDLE_TRANSPORTENC hTpEnc, int mBits) { int crcReg = 0; switch (hTpEnc->transportFmt) { case TT_MP4_ADTS: crcReg = adtsWrite_CrcStartReg(&hTpEnc->writer.adts, &hTpEnc->bitStream, mBits); break; default: break; } return crcReg; } void transportEnc_CrcEndReg(HANDLE_TRANSPORTENC hTpEnc, int reg) { switch (hTpEnc->transportFmt) { case TT_MP4_ADTS: adtsWrite_CrcEndReg(&hTpEnc->writer.adts, &hTpEnc->bitStream, reg); break; default: break; } } TRANSPORTENC_ERROR transportEnc_GetConf(HANDLE_TRANSPORTENC hTpEnc, CODER_CONFIG *cc, FDK_BITSTREAM *dataBuffer, UINT *confType) { TRANSPORTENC_ERROR tpErr = TRANSPORTENC_OK; HANDLE_LATM_STREAM hLatmConfig = &hTpEnc->writer.latm; *confType = 0; /* set confType variable to default */ /* write StreamMuxConfig or AudioSpecificConfig depending on format used */ switch (hTpEnc->transportFmt) { case TT_MP4_LATM_MCP0: case TT_MP4_LATM_MCP1: case TT_MP4_LOAS: tpErr = CreateStreamMuxConfig(hLatmConfig, dataBuffer, 0, &hTpEnc->callbacks); *confType = 1; /* config is SMC */ break; default: if (transportEnc_writeASC(dataBuffer, cc, &hTpEnc->callbacks) != 0) { tpErr = TRANSPORTENC_UNKOWN_ERROR; } } return tpErr; } TRANSPORTENC_ERROR transportEnc_GetLibInfo(LIB_INFO *info) { int i; if (info == NULL) { return TRANSPORTENC_INVALID_PARAMETER; } /* search for next free tab */ for (i = 0; i < FDK_MODULE_LAST; i++) { if (info[i].module_id == FDK_NONE) break; } if (i == FDK_MODULE_LAST) { return TRANSPORTENC_UNKOWN_ERROR; } info += i; info->module_id = FDK_TPENC; info->version = LIB_VERSION(TP_LIB_VL0, TP_LIB_VL1, TP_LIB_VL2); LIB_VERSION_STRING(info); #ifdef SUPPRESS_BUILD_DATE_INFO info->build_date = ""; info->build_time = ""; #else info->build_date = __DATE__; info->build_time = __TIME__; #endif info->title = TP_LIB_TITLE; /* Set flags */ info->flags = 0 | CAPF_ADIF | CAPF_ADTS | CAPF_LATM | CAPF_LOAS | CAPF_RAWPACKETS; return TRANSPORTENC_OK; } fdk-aac-2.0.2/libPCMutils/000077500000000000000000000000001404225310700151665ustar00rootroot00000000000000fdk-aac-2.0.2/libPCMutils/include/000077500000000000000000000000001404225310700166115ustar00rootroot00000000000000fdk-aac-2.0.2/libPCMutils/include/limiter.h000066400000000000000000000317471404225310700204430ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** PCM utility library ****************************** Author(s): Matthias Neusinger Description: Hard limiter for clipping prevention *******************************************************************************/ #ifndef LIMITER_H #define LIMITER_H #include "common_fix.h" #include "FDK_audio.h" #define TDL_ATTACK_DEFAULT_MS (15) /* default attack time in ms */ #define TDL_RELEASE_DEFAULT_MS (50) /* default release time in ms */ #ifdef __cplusplus extern "C" { #endif struct TDLimiter { unsigned int attack; FIXP_DBL attackConst, releaseConst; unsigned int attackMs, releaseMs, maxAttackMs; FIXP_DBL threshold; unsigned int channels, maxChannels; UINT sampleRate, maxSampleRate; FIXP_DBL cor, max; FIXP_DBL* maxBuf; FIXP_DBL* delayBuf; unsigned int maxBufIdx, delayBufIdx; FIXP_DBL smoothState0; FIXP_DBL minGain; INT scaling; }; typedef enum { TDLIMIT_OK = 0, TDLIMIT_UNKNOWN = -1, __error_codes_start = -100, TDLIMIT_INVALID_HANDLE, TDLIMIT_INVALID_PARAMETER, __error_codes_end } TDLIMITER_ERROR; struct TDLimiter; typedef struct TDLimiter* TDLimiterPtr; #define PCM_LIM LONG #define FIXP_DBL2PCM_LIM(x) (x) #define PCM_LIM2FIXP_DBL(x) (x) #define PCM_LIM_BITS 32 #define FIXP_PCM_LIM FIXP_DBL #define SAMPLE_BITS_LIM DFRACT_BITS /****************************************************************************** * pcmLimiter_Reset * * limiter: limiter handle * * returns: error code * ******************************************************************************/ TDLIMITER_ERROR pcmLimiter_Reset(TDLimiterPtr limiter); /****************************************************************************** * pcmLimiter_Destroy * * limiter: limiter handle * * returns: error code * ******************************************************************************/ TDLIMITER_ERROR pcmLimiter_Destroy(TDLimiterPtr limiter); /****************************************************************************** * pcmLimiter_GetDelay * * limiter: limiter handle * * returns: exact delay caused by the limiter in samples per channel * ******************************************************************************/ unsigned int pcmLimiter_GetDelay(TDLimiterPtr limiter); /****************************************************************************** * pcmLimiter_GetMaxGainReduction * * limiter: limiter handle * * returns: maximum gain reduction in last processed block in dB * ******************************************************************************/ INT pcmLimiter_GetMaxGainReduction(TDLimiterPtr limiter); /****************************************************************************** * pcmLimiter_SetNChannels * * limiter: limiter handle * * nChannels: number of channels ( <= maxChannels specified on create) * * returns: error code * ******************************************************************************/ TDLIMITER_ERROR pcmLimiter_SetNChannels(TDLimiterPtr limiter, unsigned int nChannels); /****************************************************************************** * pcmLimiter_SetSampleRate * * limiter: limiter handle * * sampleRate: sampling rate in Hz ( <= maxSampleRate specified on create) * * returns: error code * ******************************************************************************/ TDLIMITER_ERROR pcmLimiter_SetSampleRate(TDLimiterPtr limiter, UINT sampleRate); /****************************************************************************** * pcmLimiter_SetAttack * * limiter: limiter handle * * attackMs: attack time in ms ( <= maxAttackMs specified on create) * * returns: error code * ******************************************************************************/ TDLIMITER_ERROR pcmLimiter_SetAttack(TDLimiterPtr limiter, unsigned int attackMs); /****************************************************************************** * pcmLimiter_SetRelease * * limiter: limiter handle * * releaseMs: release time in ms * * returns: error code * ******************************************************************************/ TDLIMITER_ERROR pcmLimiter_SetRelease(TDLimiterPtr limiter, unsigned int releaseMs); /****************************************************************************** * pcmLimiter_GetLibInfo * * info: pointer to an allocated and initialized LIB_INFO structure * * returns: error code * ******************************************************************************/ TDLIMITER_ERROR pcmLimiter_GetLibInfo(LIB_INFO* info); #ifdef __cplusplus } #endif /****************************************************************************** * pcmLimiter_Create * * maxAttackMs: maximum and initial attack/lookahead time in milliseconds * * releaseMs: release time in milliseconds (90% time constant) * * threshold: limiting threshold * * maxChannels: maximum and initial number of channels * * maxSampleRate: maximum and initial sampling rate in Hz * * returns: limiter handle * ******************************************************************************/ TDLimiterPtr pcmLimiter_Create(unsigned int maxAttackMs, unsigned int releaseMs, FIXP_DBL threshold, unsigned int maxChannels, UINT maxSampleRate); /****************************************************************************** * pcmLimiter_SetThreshold * * limiter: limiter handle * * threshold: limiter threshold * * returns: error code * ******************************************************************************/ TDLIMITER_ERROR pcmLimiter_SetThreshold(TDLimiterPtr limiter, FIXP_DBL threshold); /****************************************************************************** * pcmLimiter_Apply * * limiter: limiter handle * * samplesIn: pointer to input buffer containing interleaved samples * * samplesOut: pointer to output buffer containing interleaved samples * * pGainPerSample: pointer to gains for each sample * * scaling: scaling of output samples * * nSamples: number of samples per channel * * returns: error code * ******************************************************************************/ TDLIMITER_ERROR pcmLimiter_Apply(TDLimiterPtr limiter, PCM_LIM* samplesIn, INT_PCM* samplesOut, FIXP_DBL* pGainPerSample, const INT scaling, const UINT nSamples); #endif /* #ifndef LIMITER_H */ fdk-aac-2.0.2/libPCMutils/include/pcm_utils.h000066400000000000000000000141331404225310700207630ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** PCM utility library ****************************** Author(s): Alfonso Pino Garcia Description: Functions that perform (de)interleaving combined with format change *******************************************************************************/ #if !defined(PCM_UTILS_H) #define PCM_UTILS_H #include "common_fix.h" void FDK_interleave(const FIXP_DBL *RESTRICT pIn, LONG *RESTRICT pOut, const UINT channels, const UINT frameSize, const UINT length); void FDK_interleave(const FIXP_DBL *RESTRICT pIn, SHORT *RESTRICT pOut, const UINT channels, const UINT frameSize, const UINT length); void FDK_interleave(const FIXP_SGL *RESTRICT pIn, SHORT *RESTRICT pOut, const UINT channels, const UINT frameSize, const UINT length); void FDK_deinterleave(const LONG *RESTRICT pIn, SHORT *RESTRICT pOut, const UINT channels, const UINT frameSize, const UINT length); void FDK_deinterleave(const LONG *RESTRICT pIn, LONG *RESTRICT pOut, const UINT channels, const UINT frameSize, const UINT length); void FDK_deinterleave(const SHORT *RESTRICT pIn, SHORT *RESTRICT pOut, const UINT channels, const UINT frameSize, const UINT length); void FDK_deinterleave(const SHORT *RESTRICT pIn, LONG *RESTRICT pOut, const UINT channels, const UINT frameSize, const UINT length); #endif /* !defined(PCM_UTILS_H) */ fdk-aac-2.0.2/libPCMutils/include/pcmdmx_lib.h000066400000000000000000000502751404225310700211110ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** PCM utility library ****************************** Author(s): Christian Griebel Description: *******************************************************************************/ /** * \file pcmdmx_lib.h * \brief FDK PCM audio mixdown library interface header file. \page INTRO Introduction \section SCOPE Scope This document describes the high-level application interface and usage of the FDK PCM audio mixdown module library developed by the Fraunhofer Institute for Integrated Circuits (IIS). Depending on the library configuration, the module can manipulate the number of audio channels of a given PCM signal. It can create for example a two channel stereo audio signal from a given multi-channel configuration (e.g. 5.1 channels). \page ABBREV List of abbreviations \li \b AAC - Advanced Audio Coding\n Is an audio coding standard for lossy digital audio compression standardized by ISO and IEC, as part of the MPEG-2 (ISO/IEC 13818-7:2006) and MPEG-4 (ISO/IEC 14496-3:2009) specifications. \li \b DSE - Data Stream Element\n A syntactical element of the MPEG-2/4 Advanced Audio Coding bitstream standardized in ISO/IEC 14496-3:2009. It can convey any kind of data associated to one program. \li \b PCE - Program Config Element\n A syntactical element of the MPEG-2/4 Advanced Audio Coding bitstream standardized in ISO/IEC 14496-3:2009 that can define the stream configuration for a single program. In addition it can comprise simple downmix meta data. */ #ifndef PCMDMX_LIB_H #define PCMDMX_LIB_H #include "machine_type.h" #include "common_fix.h" #include "FDK_audio.h" #include "FDK_bitstream.h" /** * \enum PCMDMX_ERROR * * Error codes that can be returned by module interface functions. */ typedef enum { PCMDMX_OK = 0x0, /*!< No error happened. */ PCMDMX_UNSUPPORTED = 0x1, /*!< The requested feature/service is unavailable. This can occur if the module was built for a wrong configuration. */ pcm_dmx_fatal_error_start, PCMDMX_OUT_OF_MEMORY, /*!< Not enough memory to set up an instance of the module. */ pcm_dmx_fatal_error_end, PCMDMX_INVALID_HANDLE, /*!< The given instance handle is not valid. */ PCMDMX_INVALID_ARGUMENT, /*!< One of the parameters handed over is invalid. */ PCMDMX_INVALID_CH_CONFIG, /*!< The given channel configuration is not supported and thus no processing was performed. */ PCMDMX_INVALID_MODE, /*!< The set configuration/mode is not applicable. */ PCMDMX_UNKNOWN_PARAM, /*!< The handed parameter is not known/supported. */ PCMDMX_UNABLE_TO_SET_PARAM, /*!< Unable to set the specific parameter. Most probably the value ist out of range. */ PCMDMX_CORRUPT_ANC_DATA, /*!< The read ancillary data was corrupt. */ PCMDMX_OUTPUT_BUFFER_TOO_SMALL /*!< The size of pcm output buffer is too small. */ } PCMDMX_ERROR; /** Macro to identify fatal errors. */ #define PCMDMX_IS_FATAL_ERROR(err) \ ((((err) >= pcm_dmx_fatal_error_start) && \ ((err) <= pcm_dmx_fatal_error_end)) \ ? 1 \ : 0) /** * \enum PCMDMX_PARAM * * Modules dynamic runtime parameters that can be handed to function * pcmDmx_SetParam() and pcmDmx_GetParam(). */ typedef enum { DMX_PROFILE_SETTING = 0x01, /*!< Defines which equations, coefficients and default/ fallback values used for downmixing. See ::DMX_PROFILE_TYPE type for details. */ DMX_BS_DATA_EXPIRY_FRAME = 0x10, /*!< The number of frames without new metadata that have to go by before the bitstream data expires. The value 0 disables expiry. */ DMX_BS_DATA_DELAY = 0x11, /*!< The number of delay frames of the output samples compared to the bitstream data. */ MIN_NUMBER_OF_OUTPUT_CHANNELS = 0x20, /*!< The minimum number of output channels. For all input configurations that have less than the given channels the module will modify the output automatically to obtain the given number of output channels. Mono signals will be duplicated. If more than two output channels are desired the module just adds empty channels. The parameter value must be either -1, 0, 1, 2, 6 or 8. If the value is greater than zero and exceeds the value of parameter ::MAX_NUMBER_OF_OUTPUT_CHANNELS the latter will be set to the same value. Both values -1 and 0 disable the feature. */ MAX_NUMBER_OF_OUTPUT_CHANNELS = 0x21, /*!< The maximum number of output channels. For all input configurations that have more than the given channels the module will apply a mixdown automatically to obtain the given number of output channels. The value must be either -1, 0, 1, 2, 6 or 8. If it's greater than zero and lower or equal than the value of ::MIN_NUMBER_OF_OUTPUT_CHANNELS parameter the latter will be set to the same value. The values -1 and 0 disable the feature. */ DMX_DUAL_CHANNEL_MODE = 0x30, /*!< Downmix mode for two channel audio data. See type ::DUAL_CHANNEL_MODE for details. */ DMX_PSEUDO_SURROUND_MODE = 0x31 /*!< Defines how module handles pseudo surround compatible signals. See ::PSEUDO_SURROUND_MODE type for details. */ } PCMDMX_PARAM; /** * \enum DMX_PROFILE_TYPE * * Valid value list for parameter ::DMX_PROFILE_SETTING. */ typedef enum { DMX_PRFL_STANDARD = 0x0, /*!< The standard profile creates mixdown signals based on the advanced downmix metadata (from a DSE), equations and default values defined in ISO/IEC 14496:3 Ammendment 4. Any other (legacy) downmix metadata will be ignored. */ DMX_PRFL_MATRIX_MIX = 0x1, /*!< This profile behaves just as the standard profile if advanced downmix metadata (from a DSE) is available. If not, the matrix_mixdown information embedded in the program configuration element (PCE) will be applied. If neither is the case the module creates a mixdown using the default coefficients defined in MPEG-4 Ammendment 4. The profile can be used e.g. to support legacy digital TV (e.g. DVB) streams. */ DMX_PRFL_FORCE_MATRIX_MIX = 0x2, /*!< Similar to the ::DMX_PRFL_MATRIX_MIX profile but if both the advanced (DSE) and the legacy (PCE) MPEG downmix metadata are available the latter will be applied. */ DMX_PRFL_ARIB_JAPAN = 0x3 /*!< Downmix creation as described in ABNT NBR 15602-2. But if advanced downmix metadata is available it will be prefered. */ } DMX_PROFILE_TYPE; /** * \enum PSEUDO_SURROUND_MODE * * Valid value list for parameter ::DMX_PSEUDO_SURROUND_MODE. */ typedef enum { NEVER_DO_PS_DMX = -1, /*!< Ignore any metadata and do never create a pseudo surround compatible downmix. (Default) */ AUTO_PS_DMX = 0, /*!< Create a pseudo surround compatible downmix only if signalled in bitstreams meta data. */ FORCE_PS_DMX = 1 /*!< Always create a pseudo surround compatible downmix. CAUTION: This can lead to excessive signal cancellations and signal level differences for non-compatible signals. */ } PSEUDO_SURROUND_MODE; /** * \enum DUAL_CHANNEL_MODE * * Valid value list for parameter ::DMX_DUAL_CHANNEL_MODE. */ typedef enum { STEREO_MODE = 0x0, /*!< Leave stereo signals as they are. */ CH1_MODE = 0x1, /*!< Create a dual mono output signal from channel 1. */ CH2_MODE = 0x2, /*!< Create a dual mono output signal from channel 2. */ MIXED_MODE = 0x3 /*!< Create a dual mono output signal by mixing the two channels. */ } DUAL_CHANNEL_MODE; #define DMX_PCM FIXP_DBL #define DMX_PCMF FIXP_DBL #define DMX_PCM_BITS DFRACT_BITS #define FX_DMX2FX_PCM(x) FX_DBL2FX_PCM((FIXP_DBL)(x)) /* ------------------------ * * MODULES INTERFACE: * * ------------------------ */ typedef struct PCM_DMX_INSTANCE *HANDLE_PCM_DOWNMIX; /*! \addtogroup pcmDmxResetFlags Modules reset flags * Macros that can be used as parameter for function pcmDmx_Reset() to specify * which parts of the module shall be reset. * @{ * * \def PCMDMX_RESET_PARAMS * Only reset the user specific parameters that have been modified with * pcmDmx_SetParam(). * * \def PCMDMX_RESET_BS_DATA * Delete the meta data that has been fed with the appropriate interface * functions. * * \def PCMDMX_RESET_FULL * Reset the complete module instance to the state after pcmDmx_Open() had been * called. */ #define PCMDMX_RESET_PARAMS (1) #define PCMDMX_RESET_BS_DATA (2) #define PCMDMX_RESET_FULL (PCMDMX_RESET_PARAMS | PCMDMX_RESET_BS_DATA) /*! @} */ #ifdef __cplusplus extern "C" { #endif /** Open and initialize an instance of the PCM downmix module * @param[out] pSelf Pointer to a buffer receiving the handle of the new *instance. * @returns Returns an error code of type ::PCMDMX_ERROR. **/ PCMDMX_ERROR pcmDmx_Open(HANDLE_PCM_DOWNMIX *pSelf); /** Set one parameter for a single instance of the PCM downmix module. * @param[in] self Handle of PCM downmix instance. * @param[in] param Parameter to be set. Can be one from the ::PCMDMX_PARAM *list. * @param[in] value Parameter value. * @returns Returns an error code of type ::PCMDMX_ERROR. **/ PCMDMX_ERROR pcmDmx_SetParam(HANDLE_PCM_DOWNMIX self, const PCMDMX_PARAM param, const INT value); /** Get one parameter value of a single PCM downmix module instance. * @param[in] self Handle of PCM downmix module instance. * @param[in] param Parameter to query. Can be one from the ::PCMDMX_PARAM *list. * @param[out] pValue Pointer to buffer receiving the parameter value. * @returns Returns an error code of type ::PCMDMX_ERROR. **/ PCMDMX_ERROR pcmDmx_GetParam(HANDLE_PCM_DOWNMIX self, const PCMDMX_PARAM param, INT *const pValue); /** \cond * Extract relevant downmix meta-data directly from a given bitstream. The *function can handle both data specified in ETSI TS 101 154 or ISO/IEC *14496-3:2009/Amd.4:2013. * @param[in] self Handle of PCM downmix instance. * @param[in] hBitStream Handle of FDK bitstream buffer. * @param[in] ancDataBits Length of ancillary data in bits. * @param[in] isMpeg2 Flag indicating wheter the ancillary data is from a *MPEG-1/2 or a MPEG-4 stream. * @returns Returns an error code of type ::PCMDMX_ERROR. **/ PCMDMX_ERROR pcmDmx_Parse(HANDLE_PCM_DOWNMIX self, HANDLE_FDK_BITSTREAM hBitStream, UINT ancDataBits, int isMpeg2); /** \endcond */ /** Read from a given ancillary data buffer and extract the relevant downmix *meta-data. The function can handle both data specified in ETSI TS 101 154 or *ISO/IEC 14496-3:2009/Amd.4:2013. * @param[in] self Handle of PCM downmix instance. * @param[in] pAncDataBuf Pointer to ancillary buffer holding the data. * @param[in] ancDataBytes Size of ancillary data in bytes. * @param[in] isMpeg2 Flag indicating wheter the ancillary data is from a *MPEG-1/2 or a MPEG-4 stream. * @returns Returns an error code of type ::PCMDMX_ERROR. **/ PCMDMX_ERROR pcmDmx_ReadDvbAncData(HANDLE_PCM_DOWNMIX self, UCHAR *pAncDataBuf, UINT ancDataBytes, int isMpeg2); /** Set the matrix mixdown information extracted from the PCE of an AAC *bitstream. * @param[in] self Handle of PCM downmix instance. * @param[in] matrixMixdownPresent Matrix mixdown index present flag extracted *from PCE. * @param[in] matrixMixdownIdx The 2 bit matrix mixdown index extracted *from PCE. * @param[in] pseudoSurroundEnable The pseudo surround enable flag extracted *from PCE. * @returns Returns an error code of type *::PCMDMX_ERROR. **/ PCMDMX_ERROR pcmDmx_SetMatrixMixdownFromPce(HANDLE_PCM_DOWNMIX self, int matrixMixdownPresent, int matrixMixdownIdx, int pseudoSurroundEnable); /** Reset the module. * @param[in] self Handle of PCM downmix instance. * @param[in] flags Flags telling which parts of the module shall be reset. * See \ref pcmDmxResetFlags for details. * @returns Returns an error code of type ::PCMDMX_ERROR. **/ PCMDMX_ERROR pcmDmx_Reset(HANDLE_PCM_DOWNMIX self, UINT flags); /** Create a mixdown, bypass or extend the output signal depending on the *modules settings and the respective given input configuration. * * \param[in] self Handle of PCM downmix module instance. * \param[in,out] pPcmBuf Pointer to time buffer with PCM samples. * \param[in] pcmBufSize Size of pPcmBuf buffer. * \param[in] frameSize The I/O block size which is the number of samples per channel. * \param[in,out] nChannels Pointer to buffer that holds the number of input channels and * where the amount of output channels is written *to. * \param[in] fInterleaved Input and output samples are processed interleaved. * \param[in,out] channelType Array were the corresponding channel type for each output audio * channel is stored into. * \param[in,out] channelIndices Array were the corresponding channel type index for each output * audio channel is stored into. * \param[in] mapDescr Pointer to a FDK channel mapping descriptor that contains the * channel mapping to be used. * \param[out] pDmxOutScale Pointer on a field receiving the scale factor that has to be * applied on all samples afterwards. If the *handed pointer is NULL the final scaling is done internally. * @returns Returns an error code of type ::PCMDMX_ERROR. **/ PCMDMX_ERROR pcmDmx_ApplyFrame(HANDLE_PCM_DOWNMIX self, DMX_PCM *pPcmBuf, const int pcmBufSize, UINT frameSize, INT *nChannels, INT fInterleaved, AUDIO_CHANNEL_TYPE channelType[], UCHAR channelIndices[], const FDK_channelMapDescr *const mapDescr, INT *pDmxOutScale); /** Close an instance of the PCM downmix module. * @param[in,out] pSelf Pointer to a buffer containing the handle of the *instance. * @returns Returns an error code of type ::PCMDMX_ERROR. **/ PCMDMX_ERROR pcmDmx_Close(HANDLE_PCM_DOWNMIX *pSelf); /** Get library info for this module. * @param[out] info Pointer to an allocated LIB_INFO structure. * @returns Returns an error code of type ::PCMDMX_ERROR. */ PCMDMX_ERROR pcmDmx_GetLibInfo(LIB_INFO *info); #ifdef __cplusplus } #endif #endif /* PCMDMX_LIB_H */ fdk-aac-2.0.2/libPCMutils/src/000077500000000000000000000000001404225310700157555ustar00rootroot00000000000000fdk-aac-2.0.2/libPCMutils/src/limiter.cpp000066400000000000000000000447621404225310700201430ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** PCM utility library ****************************** Author(s): Matthias Neusinger Description: Hard limiter for clipping prevention *******************************************************************************/ #include "limiter.h" #include "FDK_core.h" /* library version */ #include "version.h" /* library title */ #define TDLIMIT_LIB_TITLE "TD Limiter Lib" /* create limiter */ TDLimiterPtr pcmLimiter_Create(unsigned int maxAttackMs, unsigned int releaseMs, FIXP_DBL threshold, unsigned int maxChannels, UINT maxSampleRate) { TDLimiterPtr limiter = NULL; unsigned int attack, release; FIXP_DBL attackConst, releaseConst, exponent; INT e_ans; /* calc attack and release time in samples */ attack = (unsigned int)(maxAttackMs * maxSampleRate / 1000); release = (unsigned int)(releaseMs * maxSampleRate / 1000); /* alloc limiter struct */ limiter = (TDLimiterPtr)FDKcalloc(1, sizeof(struct TDLimiter)); if (!limiter) return NULL; /* alloc max and delay buffers */ limiter->maxBuf = (FIXP_DBL*)FDKcalloc(attack + 1, sizeof(FIXP_DBL)); limiter->delayBuf = (FIXP_DBL*)FDKcalloc(attack * maxChannels, sizeof(FIXP_DBL)); if (!limiter->maxBuf || !limiter->delayBuf) { pcmLimiter_Destroy(limiter); return NULL; } /* attackConst = pow(0.1, 1.0 / (attack + 1)) */ exponent = invFixp(attack + 1); attackConst = fPow(FL2FXCONST_DBL(0.1f), 0, exponent, 0, &e_ans); attackConst = scaleValue(attackConst, e_ans); /* releaseConst = (float)pow(0.1, 1.0 / (release + 1)) */ exponent = invFixp(release + 1); releaseConst = fPow(FL2FXCONST_DBL(0.1f), 0, exponent, 0, &e_ans); releaseConst = scaleValue(releaseConst, e_ans); /* init parameters */ limiter->attackMs = maxAttackMs; limiter->maxAttackMs = maxAttackMs; limiter->releaseMs = releaseMs; limiter->attack = attack; limiter->attackConst = attackConst; limiter->releaseConst = releaseConst; limiter->threshold = threshold; limiter->channels = maxChannels; limiter->maxChannels = maxChannels; limiter->sampleRate = maxSampleRate; limiter->maxSampleRate = maxSampleRate; pcmLimiter_Reset(limiter); return limiter; } /* apply limiter */ TDLIMITER_ERROR pcmLimiter_Apply(TDLimiterPtr limiter, PCM_LIM* samplesIn, INT_PCM* samplesOut, FIXP_DBL* pGainPerSample, const INT scaling, const UINT nSamples) { unsigned int i, j; FIXP_DBL tmp2; FIXP_DBL tmp, old, gain, additionalGain = 0; FIXP_DBL minGain = FL2FXCONST_DBL(1.0f / (1 << 1)); UINT additionalGainAvailable = 1; if (limiter == NULL) return TDLIMIT_INVALID_HANDLE; { unsigned int channels = limiter->channels; unsigned int attack = limiter->attack; FIXP_DBL attackConst = limiter->attackConst; FIXP_DBL releaseConst = limiter->releaseConst; FIXP_DBL threshold = limiter->threshold >> scaling; FIXP_DBL max = limiter->max; FIXP_DBL* maxBuf = limiter->maxBuf; unsigned int maxBufIdx = limiter->maxBufIdx; FIXP_DBL cor = limiter->cor; FIXP_DBL* delayBuf = limiter->delayBuf; unsigned int delayBufIdx = limiter->delayBufIdx; FIXP_DBL smoothState0 = limiter->smoothState0; if (limiter->scaling != scaling) { scaleValuesSaturate(delayBuf, attack * channels, limiter->scaling - scaling); scaleValuesSaturate(maxBuf, attack + 1, limiter->scaling - scaling); max = scaleValueSaturate(max, limiter->scaling - scaling); limiter->scaling = scaling; } if (pGainPerSample == NULL) { additionalGainAvailable = 0; } for (i = 0; i < nSamples; i++) { /* get maximum absolute sample value of all channels, including the * additional gain. */ tmp = (FIXP_DBL)0; for (j = 0; j < channels; j++) { tmp2 = PCM_LIM2FIXP_DBL(samplesIn[j]); tmp2 = (tmp2 == (FIXP_DBL)MINVAL_DBL) ? (FIXP_DBL)MAXVAL_DBL : fAbs(tmp2); tmp = fMax(tmp, tmp2); } if (additionalGainAvailable) { additionalGain = pGainPerSample[i]; tmp = fMult(tmp, additionalGain); } /* set threshold as lower border to save calculations in running maximum * algorithm */ tmp = fMax(tmp, threshold); /* running maximum */ old = maxBuf[maxBufIdx]; maxBuf[maxBufIdx] = tmp; if (tmp >= max) { /* new sample is greater than old maximum, so it is the new maximum */ max = tmp; } else if (old < max) { /* maximum does not change, as the sample, which has left the window was not the maximum */ } else { /* the old maximum has left the window, we have to search the complete buffer for the new max */ max = maxBuf[0]; for (j = 1; j <= attack; j++) { max = fMax(max, maxBuf[j]); } } maxBufIdx++; if (maxBufIdx >= attack + 1) maxBufIdx = 0; /* calc gain */ /* gain is downscaled by one, so that gain = 1.0 can be represented */ if (max > threshold) { gain = fDivNorm(threshold, max) >> 1; } else { gain = FL2FXCONST_DBL(1.0f / (1 << 1)); } /* gain smoothing, method: TDL_EXPONENTIAL */ /* first order IIR filter with attack correction to avoid overshoots */ /* correct the 'aiming' value of the exponential attack to avoid the * remaining overshoot */ if (gain < smoothState0) { cor = fMin(cor, fMultDiv2((gain - fMultDiv2(FL2FXCONST_SGL(0.1f * (1 << 1)), smoothState0)), FL2FXCONST_SGL(1.11111111f / (1 << 1))) << 2); } else { cor = gain; } /* smoothing filter */ if (cor < smoothState0) { smoothState0 = fMult(attackConst, (smoothState0 - cor)) + cor; /* attack */ smoothState0 = fMax(smoothState0, gain); /* avoid overshooting target */ } else { /* sign inversion twice to round towards +infinity, so that gain can converge to 1.0 again, for bit-identical output when limiter is not active */ smoothState0 = -fMult(releaseConst, -(smoothState0 - cor)) + cor; /* release */ } gain = smoothState0; FIXP_DBL* p_delayBuf = &delayBuf[delayBufIdx * channels + 0]; if (gain < FL2FXCONST_DBL(1.0f / (1 << 1))) { gain <<= 1; /* lookahead delay, apply gain */ for (j = 0; j < channels; j++) { tmp = p_delayBuf[j]; if (additionalGainAvailable) { p_delayBuf[j] = fMult((FIXP_PCM_LIM)samplesIn[j], additionalGain); } else { p_delayBuf[j] = PCM_LIM2FIXP_DBL(samplesIn[j]); } /* Apply gain to delayed signal */ tmp = fMultDiv2(tmp, gain); #if (SAMPLE_BITS == DFRACT_BITS) samplesOut[j] = (INT_PCM)FX_DBL2FX_PCM( (FIXP_DBL)SATURATE_LEFT_SHIFT(tmp, scaling + 1, DFRACT_BITS)); #else samplesOut[j] = (INT_PCM)FX_DBL2FX_PCM((FIXP_DBL)SATURATE_LEFT_SHIFT( tmp + ((FIXP_DBL)0x8000 >> (scaling + 1)), scaling + 1, DFRACT_BITS)); #endif } gain >>= 1; } else { /* lookahead delay, apply gain=1.0f */ for (j = 0; j < channels; j++) { tmp = p_delayBuf[j]; if (additionalGainAvailable) { p_delayBuf[j] = fMult((FIXP_PCM_LIM)samplesIn[j], additionalGain); } else { p_delayBuf[j] = PCM_LIM2FIXP_DBL(samplesIn[j]); } #if (SAMPLE_BITS == DFRACT_BITS) samplesOut[j] = (INT_PCM)FX_DBL2FX_PCM( (FIXP_DBL)SATURATE_LEFT_SHIFT(tmp, scaling, DFRACT_BITS)); #else samplesOut[j] = (INT_PCM)FX_DBL2FX_PCM((FIXP_DBL)SATURATE_LEFT_SHIFT( tmp + ((FIXP_DBL)0x8000 >> scaling), scaling, DFRACT_BITS)); #endif } } delayBufIdx++; if (delayBufIdx >= attack) { delayBufIdx = 0; } /* save minimum gain factor */ if (gain < minGain) { minGain = gain; } /* advance sample pointer by samples */ samplesIn += channels; samplesOut += channels; } limiter->max = max; limiter->maxBufIdx = maxBufIdx; limiter->cor = cor; limiter->delayBufIdx = delayBufIdx; limiter->smoothState0 = smoothState0; limiter->minGain = minGain; return TDLIMIT_OK; } } /* set limiter threshold */ TDLIMITER_ERROR pcmLimiter_SetThreshold(TDLimiterPtr limiter, FIXP_DBL threshold) { if (limiter == NULL) return TDLIMIT_INVALID_HANDLE; limiter->threshold = threshold; return TDLIMIT_OK; } /* reset limiter */ TDLIMITER_ERROR pcmLimiter_Reset(TDLimiterPtr limiter) { if (limiter != NULL) { limiter->maxBufIdx = 0; limiter->delayBufIdx = 0; limiter->max = (FIXP_DBL)0; limiter->cor = FL2FXCONST_DBL(1.0f / (1 << 1)); limiter->smoothState0 = FL2FXCONST_DBL(1.0f / (1 << 1)); limiter->minGain = FL2FXCONST_DBL(1.0f / (1 << 1)); limiter->scaling = 0; FDKmemset(limiter->maxBuf, 0, (limiter->attack + 1) * sizeof(FIXP_DBL)); FDKmemset(limiter->delayBuf, 0, limiter->attack * limiter->channels * sizeof(FIXP_DBL)); } else { return TDLIMIT_INVALID_HANDLE; } return TDLIMIT_OK; } /* destroy limiter */ TDLIMITER_ERROR pcmLimiter_Destroy(TDLimiterPtr limiter) { if (limiter != NULL) { FDKfree(limiter->maxBuf); FDKfree(limiter->delayBuf); FDKfree(limiter); } else { return TDLIMIT_INVALID_HANDLE; } return TDLIMIT_OK; } /* get delay in samples */ unsigned int pcmLimiter_GetDelay(TDLimiterPtr limiter) { FDK_ASSERT(limiter != NULL); return limiter->attack; } /* get maximum gain reduction of last processed block */ INT pcmLimiter_GetMaxGainReduction(TDLimiterPtr limiter) { /* maximum gain reduction in dB = -20 * log10(limiter->minGain) = -20 * log2(limiter->minGain)/log2(10) = -6.0206*log2(limiter->minGain) */ int e_ans; FIXP_DBL loggain, maxGainReduction; FDK_ASSERT(limiter != NULL); loggain = fLog2(limiter->minGain, 1, &e_ans); maxGainReduction = fMult(loggain, FL2FXCONST_DBL(-6.0206f / (1 << 3))); return fixp_roundToInt(maxGainReduction, (e_ans + 3)); } /* set number of channels */ TDLIMITER_ERROR pcmLimiter_SetNChannels(TDLimiterPtr limiter, unsigned int nChannels) { if (limiter == NULL) return TDLIMIT_INVALID_HANDLE; if (nChannels > limiter->maxChannels) return TDLIMIT_INVALID_PARAMETER; limiter->channels = nChannels; // pcmLimiter_Reset(limiter); return TDLIMIT_OK; } /* set sampling rate */ TDLIMITER_ERROR pcmLimiter_SetSampleRate(TDLimiterPtr limiter, UINT sampleRate) { unsigned int attack, release; FIXP_DBL attackConst, releaseConst, exponent; INT e_ans; if (limiter == NULL) return TDLIMIT_INVALID_HANDLE; if (sampleRate > limiter->maxSampleRate) return TDLIMIT_INVALID_PARAMETER; /* update attack and release time in samples */ attack = (unsigned int)(limiter->attackMs * sampleRate / 1000); release = (unsigned int)(limiter->releaseMs * sampleRate / 1000); /* attackConst = pow(0.1, 1.0 / (attack + 1)) */ exponent = invFixp(attack + 1); attackConst = fPow(FL2FXCONST_DBL(0.1f), 0, exponent, 0, &e_ans); attackConst = scaleValue(attackConst, e_ans); /* releaseConst = (float)pow(0.1, 1.0 / (release + 1)) */ exponent = invFixp(release + 1); releaseConst = fPow(FL2FXCONST_DBL(0.1f), 0, exponent, 0, &e_ans); releaseConst = scaleValue(releaseConst, e_ans); limiter->attack = attack; limiter->attackConst = attackConst; limiter->releaseConst = releaseConst; limiter->sampleRate = sampleRate; /* reset */ // pcmLimiter_Reset(limiter); return TDLIMIT_OK; } /* set attack time */ TDLIMITER_ERROR pcmLimiter_SetAttack(TDLimiterPtr limiter, unsigned int attackMs) { unsigned int attack; FIXP_DBL attackConst, exponent; INT e_ans; if (limiter == NULL) return TDLIMIT_INVALID_HANDLE; if (attackMs > limiter->maxAttackMs) return TDLIMIT_INVALID_PARAMETER; /* calculate attack time in samples */ attack = (unsigned int)(attackMs * limiter->sampleRate / 1000); /* attackConst = pow(0.1, 1.0 / (attack + 1)) */ exponent = invFixp(attack + 1); attackConst = fPow(FL2FXCONST_DBL(0.1f), 0, exponent, 0, &e_ans); attackConst = scaleValue(attackConst, e_ans); limiter->attack = attack; limiter->attackConst = attackConst; limiter->attackMs = attackMs; return TDLIMIT_OK; } /* set release time */ TDLIMITER_ERROR pcmLimiter_SetRelease(TDLimiterPtr limiter, unsigned int releaseMs) { unsigned int release; FIXP_DBL releaseConst, exponent; INT e_ans; if (limiter == NULL) return TDLIMIT_INVALID_HANDLE; /* calculate release time in samples */ release = (unsigned int)(releaseMs * limiter->sampleRate / 1000); /* releaseConst = (float)pow(0.1, 1.0 / (release + 1)) */ exponent = invFixp(release + 1); releaseConst = fPow(FL2FXCONST_DBL(0.1f), 0, exponent, 0, &e_ans); releaseConst = scaleValue(releaseConst, e_ans); limiter->releaseConst = releaseConst; limiter->releaseMs = releaseMs; return TDLIMIT_OK; } /* Get library info for this module. */ TDLIMITER_ERROR pcmLimiter_GetLibInfo(LIB_INFO* info) { int i; if (info == NULL) { return TDLIMIT_INVALID_PARAMETER; } /* Search for next free tab */ for (i = 0; i < FDK_MODULE_LAST; i++) { if (info[i].module_id == FDK_NONE) break; } if (i == FDK_MODULE_LAST) { return TDLIMIT_UNKNOWN; } /* Add the library info */ info[i].module_id = FDK_TDLIMIT; info[i].version = LIB_VERSION(PCMUTIL_LIB_VL0, PCMUTIL_LIB_VL1, PCMUTIL_LIB_VL2); LIB_VERSION_STRING(info + i); info[i].build_date = PCMUTIL_LIB_BUILD_DATE; info[i].build_time = PCMUTIL_LIB_BUILD_TIME; info[i].title = TDLIMIT_LIB_TITLE; /* Set flags */ info[i].flags = CAPF_LIMITER; /* Add lib info for FDK tools (if not yet done). */ FDK_toolsGetLibInfo(info); return TDLIMIT_OK; } fdk-aac-2.0.2/libPCMutils/src/pcm_utils.cpp000066400000000000000000000171551404225310700204710ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** PCM utility library ****************************** Author(s): Arthur Tritthart, Alfonso Pino Garcia Description: Functions that perform (de)interleaving combined with format change *******************************************************************************/ #include "pcm_utils.h" /* library version */ #include "version.h" void FDK_interleave(const FIXP_DBL *RESTRICT pIn, LONG *RESTRICT pOut, const UINT channels, const UINT frameSize, const UINT length) { for (UINT sample = 0; sample < length; sample++) { const FIXP_DBL *In = &pIn[sample]; for (UINT ch = 0; ch < channels; ch++) { *pOut++ = (LONG)In[0]; In += frameSize; } } } void FDK_interleave(const FIXP_DBL *RESTRICT pIn, SHORT *RESTRICT pOut, const UINT channels, const UINT frameSize, const UINT length) { for (UINT sample = 0; sample < length; sample++) { const FIXP_DBL *In = &pIn[sample]; for (UINT ch = 0; ch < channels; ch++) { *pOut++ = (SHORT)FX_DBL2FX_SGL(In[0]); In += frameSize; } } } void FDK_interleave(const FIXP_SGL *RESTRICT pIn, SHORT *RESTRICT pOut, const UINT channels, const UINT frameSize, const UINT length) { for (UINT sample = 0; sample < length; sample++) { const FIXP_SGL *In = &pIn[sample]; for (UINT ch = 0; ch < channels; ch++) { *pOut++ = (SHORT)In[0]; In += frameSize; } } } void FDK_deinterleave(const LONG *RESTRICT pIn, SHORT *RESTRICT _pOut, const UINT channels, const UINT frameSize, const UINT length) { for (UINT ch = 0; ch < channels; ch++) { SHORT *pOut = _pOut + length * ch; const LONG *In = &pIn[ch]; for (UINT sample = 0; sample < frameSize; sample++) { *pOut++ = (SHORT)(In[0] >> 16); In += channels; } } } void FDK_deinterleave(const LONG *RESTRICT pIn, LONG *RESTRICT _pOut, const UINT channels, const UINT frameSize, const UINT length) { for (UINT ch = 0; ch < channels; ch++) { LONG *pOut = _pOut + length * ch; const LONG *In = &pIn[ch]; for (UINT sample = 0; sample < frameSize; sample++) { *pOut++ = In[0]; In += channels; } } } void FDK_deinterleave(const SHORT *RESTRICT pIn, SHORT *RESTRICT _pOut, const UINT channels, const UINT frameSize, const UINT length) { for (UINT ch = 0; ch < channels; ch++) { SHORT *pOut = _pOut + length * ch; const SHORT *In = &pIn[ch]; for (UINT sample = 0; sample < frameSize; sample++) { *pOut++ = In[0]; In += channels; } } } void FDK_deinterleave(const SHORT *RESTRICT pIn, LONG *RESTRICT _pOut, const UINT channels, const UINT frameSize, const UINT length) { for (UINT ch = 0; ch < channels; ch++) { LONG *pOut = _pOut + length * ch; const SHORT *In = &pIn[ch]; for (UINT sample = 0; sample < frameSize; sample++) { *pOut++ = (LONG)In[0] << 16; In += channels; } } } fdk-aac-2.0.2/libPCMutils/src/pcmdmx_lib.cpp000066400000000000000000003041601404225310700206030ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** PCM utility library ****************************** Author(s): Christian Griebel Description: Defines functions that perform downmixing or a simple channel expansion in the PCM time domain. *******************************************************************************/ #include "pcmdmx_lib.h" #include "genericStds.h" #include "fixpoint_math.h" #include "FDK_core.h" /* library version */ #include "version.h" /* library title */ #define PCMDMX_LIB_TITLE "PCM Downmix Lib" #define FALSE 0 #define TRUE 1 #define IN 0 #define OUT 1 /* Type definitions: */ #define FIXP_DMX FIXP_SGL #define FX_DMX2FX_DBL(x) FX_SGL2FX_DBL((FIXP_SGL)(x)) #define FX_DBL2FX_DMX(x) FX_DBL2FX_SGL(x) #define FL2FXCONST_DMX(x) FL2FXCONST_SGL(x) #define MAXVAL_DMX MAXVAL_SGL #define FX_DMX2SHRT(x) ((SHORT)(x)) #define FX_DMX2FL(x) FX_DBL2FL(FX_DMX2FX_DBL(x)) /* Fixed and unique channel group indices. * The last group index has to be smaller than ( 4 ). */ #define CH_GROUP_FRONT (0) #define CH_GROUP_SIDE (1) #define CH_GROUP_REAR (2) #define CH_GROUP_LFE (3) /* Fixed and unique channel plain indices. */ #define CH_PLAIN_NORMAL (0) #define CH_PLAIN_TOP (1) #define CH_PLAIN_BOTTOM (2) /* The ordering of the following fixed channel labels has to be in MPEG-4 style. * From the center to the back with left and right channel interleaved (starting * with left). The last channel label index has to be smaller than ( 8 ). */ #define CENTER_FRONT_CHANNEL (0) /* C */ #define LEFT_FRONT_CHANNEL (1) /* L */ #define RIGHT_FRONT_CHANNEL (2) /* R */ #define LEFT_REAR_CHANNEL \ (3) /* Lr (aka left back channel) or center back channel */ #define RIGHT_REAR_CHANNEL (4) /* Rr (aka right back channel) */ #define LOW_FREQUENCY_CHANNEL (5) /* Lf */ #define LEFT_MULTIPRPS_CHANNEL (6) /* Left multipurpose channel */ #define RIGHT_MULTIPRPS_CHANNEL (7) /* Right multipurpose channel */ /* 22.2 channel specific fixed channel lables: */ #define LEFT_SIDE_CHANNEL (8) /* Lss */ #define RIGHT_SIDE_CHANNEL (9) /* Rss */ #define CENTER_REAR_CHANNEL (10) /* Cs */ #define CENTER_FRONT_CHANNEL_TOP (11) /* Cv */ #define LEFT_FRONT_CHANNEL_TOP (12) /* Lv */ #define RIGHT_FRONT_CHANNEL_TOP (13) /* Rv */ #define LEFT_SIDE_CHANNEL_TOP (14) /* Lvss */ #define RIGHT_SIDE_CHANNEL_TOP (15) /* Rvss */ #define CENTER_SIDE_CHANNEL_TOP (16) /* Ts */ #define LEFT_REAR_CHANNEL_TOP (17) /* Lvr */ #define RIGHT_REAR_CHANNEL_TOP (18) /* Rvr */ #define CENTER_REAR_CHANNEL_TOP (19) /* Cvr */ #define CENTER_FRONT_CHANNEL_BOTTOM (20) /* Cb */ #define LEFT_FRONT_CHANNEL_BOTTOM (21) /* Lb */ #define RIGHT_FRONT_CHANNEL_BOTTOM (22) /* Rb */ #define LOW_FREQUENCY_CHANNEL_2 (23) /* LFE2 */ /* More constants */ #define ONE_CHANNEL (1) #define TWO_CHANNEL (2) #define SIX_CHANNEL (6) #define EIGHT_CHANNEL (8) #define TWENTY_FOUR_CHANNEL (24) #define PCMDMX_THRESHOLD_MAP_HEAT_1 (0) /* Store only exact matches */ #define PCMDMX_THRESHOLD_MAP_HEAT_2 (20) #define PCMDMX_THRESHOLD_MAP_HEAT_3 \ (256) /* Do not assign normal channels to LFE */ #define SP_Z_NRM (0) #define SP_Z_TOP (2) #define SP_Z_BOT (-2) #define SP_Z_LFE (-18) #define SP_Z_MUL (8) /* Should be smaller than SP_Z_LFE */ typedef struct { SCHAR x; /* horizontal position: center (0), left (-), right (+) */ SCHAR y; /* deepth position: front, side, back, position */ SCHAR z; /* heigth positions: normal, top, bottom, lfe */ } PCM_DMX_SPEAKER_POSITION; /* CAUTION: The maximum x-value should be less or equal to * PCMDMX_SPKR_POS_X_MAX_WIDTH. */ static const PCM_DMX_SPEAKER_POSITION spkrSlotPos[] = { /* x, y, z */ {0, 0, SP_Z_NRM}, /* 0 CENTER_FRONT_CHANNEL */ {-2, 0, SP_Z_NRM}, /* 1 LEFT_FRONT_CHANNEL */ {2, 0, SP_Z_NRM}, /* 2 RIGHT_FRONT_CHANNEL */ {-3, 4, SP_Z_NRM}, /* 3 LEFT_REAR_CHANNEL */ {3, 4, SP_Z_NRM}, /* 4 RIGHT_REAR_CHANNEL */ {0, 0, SP_Z_LFE}, /* 5 LOW_FREQUENCY_CHANNEL */ {-2, 2, SP_Z_MUL}, /* 6 LEFT_MULTIPRPS_CHANNEL */ {2, 2, SP_Z_MUL} /* 7 RIGHT_MULTIPRPS_CHANNEL */ }; /* List of packed channel modes */ typedef enum { /* CH_MODE____ */ CH_MODE_UNDEFINED = 0x0000, /* 1 channel */ CH_MODE_1_0_0_0 = 0x0001, /* chCfg 1 */ /* 2 channels */ CH_MODE_2_0_0_0 = 0x0002 /* chCfg 2 */ /* 3 channels */ , CH_MODE_3_0_0_0 = 0x0003, /* chCfg 3 */ CH_MODE_2_0_1_0 = 0x0102, CH_MODE_2_0_0_1 = 0x1002, /* 4 channels */ CH_MODE_3_0_1_0 = 0x0103, /* chCfg 4 */ CH_MODE_2_0_2_0 = 0x0202, CH_MODE_2_0_1_1 = 0x1102, CH_MODE_4_0_0_0 = 0x0004, /* 5 channels */ CH_MODE_3_0_2_0 = 0x0203, /* chCfg 5 */ CH_MODE_2_0_2_1 = 0x1202, CH_MODE_3_0_1_1 = 0x1103, CH_MODE_3_2_0_0 = 0x0023, CH_MODE_5_0_0_0 = 0x0005, /* 6 channels */ CH_MODE_3_0_2_1 = 0x1203, /* chCfg 6 */ CH_MODE_3_2_0_1 = 0x1023, CH_MODE_3_2_1_0 = 0x0123, CH_MODE_5_0_1_0 = 0x0105, CH_MODE_6_0_0_0 = 0x0006, /* 7 channels */ CH_MODE_2_2_2_1 = 0x1222, CH_MODE_3_0_3_1 = 0x1303, /* chCfg 11 */ CH_MODE_3_2_1_1 = 0x1123, CH_MODE_3_2_2_0 = 0x0223, CH_MODE_3_0_2_2 = 0x2203, CH_MODE_5_0_2_0 = 0x0205, CH_MODE_5_0_1_1 = 0x1105, CH_MODE_7_0_0_0 = 0x0007, /* 8 channels */ CH_MODE_3_2_2_1 = 0x1223, CH_MODE_3_0_4_1 = 0x1403, /* chCfg 12 */ CH_MODE_5_0_2_1 = 0x1205, /* chCfg 7 + 14 */ CH_MODE_5_2_1_0 = 0x0125, CH_MODE_3_2_1_2 = 0x2123, CH_MODE_2_2_2_2 = 0x2222, CH_MODE_3_0_3_2 = 0x2303, CH_MODE_8_0_0_0 = 0x0008 } PCM_DMX_CHANNEL_MODE; /* These are the channel configurations linked to the number of output channels give by the user: */ static const PCM_DMX_CHANNEL_MODE outChModeTable[(8) + 1] = { CH_MODE_UNDEFINED, CH_MODE_1_0_0_0, /* 1 channel */ CH_MODE_2_0_0_0 /* 2 channels */ , CH_MODE_3_0_0_0, /* 3 channels */ CH_MODE_3_0_1_0, /* 4 channels */ CH_MODE_3_0_2_0, /* 5 channels */ CH_MODE_3_0_2_1 /* 6 channels */ , CH_MODE_3_0_3_1, /* 7 channels */ CH_MODE_3_0_4_1 /* 8 channels */ }; static const FIXP_DMX abMixLvlValueTab[8] = { FL2FXCONST_DMX(0.500f), /* scaled by 1 */ FL2FXCONST_DMX(0.841f), FL2FXCONST_DMX(0.707f), FL2FXCONST_DMX(0.596f), FL2FXCONST_DMX(0.500f), FL2FXCONST_DMX(0.422f), FL2FXCONST_DMX(0.355f), FL2FXCONST_DMX(0.0f)}; static const FIXP_DMX lfeMixLvlValueTab[16] = { /* value, scale */ FL2FXCONST_DMX(0.7905f), /* 2 */ FL2FXCONST_DMX(0.5000f), /* 2 */ FL2FXCONST_DMX(0.8395f), /* 1 */ FL2FXCONST_DMX(0.7065f), /* 1 */ FL2FXCONST_DMX(0.5945f), /* 1 */ FL2FXCONST_DMX(0.500f), /* 1 */ FL2FXCONST_DMX(0.841f), /* 0 */ FL2FXCONST_DMX(0.707f), /* 0 */ FL2FXCONST_DMX(0.596f), /* 0 */ FL2FXCONST_DMX(0.500f), /* 0 */ FL2FXCONST_DMX(0.316f), /* 0 */ FL2FXCONST_DMX(0.178f), /* 0 */ FL2FXCONST_DMX(0.100f), /* 0 */ FL2FXCONST_DMX(0.032f), /* 0 */ FL2FXCONST_DMX(0.010f), /* 0 */ FL2FXCONST_DMX(0.000f) /* 0 */ }; /* MPEG matrix mixdown: Set 1: L' = (1 + 2^-0.5 + A )^-1 * [L + C * 2^-0.5 + A * Ls]; R' = (1 + 2^-0.5 + A )^-1 * [R + C * 2^-0.5 + A * Rs]; Set 2: L' = (1 + 2^-0.5 + 2A )^-1 * [L + C * 2^-0.5 - A * (Ls + Rs)]; R' = (1 + 2^-0.5 + 2A )^-1 * [R + C * 2^-0.5 + A * (Ls + Rs)]; M = (3 + 2A)^-1 * [L + C + R + A*(Ls + Rs)]; */ static const FIXP_DMX mpegMixDownIdx2Coef[4] = { FL2FXCONST_DMX(0.70710678f), FL2FXCONST_DMX(0.5f), FL2FXCONST_DMX(0.35355339f), FL2FXCONST_DMX(0.0f)}; static const FIXP_DMX mpegMixDownIdx2PreFact[3][4] = { {/* Set 1: */ FL2FXCONST_DMX(0.4142135623730950f), FL2FXCONST_DMX(0.4530818393219728f), FL2FXCONST_DMX(0.4852813742385703f), FL2FXCONST_DMX(0.5857864376269050f)}, {/* Set 2: */ FL2FXCONST_DMX(0.3203772410170407f), FL2FXCONST_DMX(0.3693980625181293f), FL2FXCONST_DMX(0.4142135623730950f), FL2FXCONST_DMX(0.5857864376269050f)}, {/* Mono DMX set: */ FL2FXCONST_DMX(0.2265409196609864f), FL2FXCONST_DMX(0.25f), FL2FXCONST_DMX(0.2697521433898179f), FL2FXCONST_DMX(0.3333333333333333f)}}; #define TYPE_NONE (0x00) #define TYPE_PCE_DATA (0x01) #define TYPE_DSE_CLEV_DATA (0x02) #define TYPE_DSE_SLEV_DATA (0x04) #define TYPE_DSE_DMIX_AB_DATA (0x08) #define TYPE_DSE_DMIX_LFE_DATA (0x10) #define TYPE_DSE_DMX_GAIN_DATA (0x20) #define TYPE_DSE_DMX_CGL_DATA (0x40) #define TYPE_DSE_DATA (0x7E) typedef struct { UINT typeFlags; /* From DSE */ UCHAR cLevIdx; UCHAR sLevIdx; UCHAR dmixIdxA; UCHAR dmixIdxB; UCHAR dmixIdxLfe; UCHAR dmxGainIdx2; UCHAR dmxGainIdx5; /* From PCE */ UCHAR matrixMixdownIdx; /* Attributes: */ SCHAR pseudoSurround; /*!< If set to 1 the signal is pseudo surround compatible. The value 0 tells that it is not. If the value is -1 the information is not available. */ UINT expiryCount; /*!< Counter to monitor the life time of a meta data set. */ } DMX_BS_META_DATA; /* Default metadata */ static const DMX_BS_META_DATA dfltMetaData = {0, 2, 2, 2, 2, 15, 0, 0, 0, -1, 0}; /* Dynamic (user) params: See the definition of PCMDMX_PARAM for details on the specific fields. */ typedef struct { DMX_PROFILE_TYPE dmxProfile; /*!< Linked to DMX_PRFL_STANDARD */ UINT expiryFrame; /*!< Linked to DMX_BS_DATA_EXPIRY_FRAME */ DUAL_CHANNEL_MODE dualChannelMode; /*!< Linked to DMX_DUAL_CHANNEL_MODE */ PSEUDO_SURROUND_MODE pseudoSurrMode; /*!< Linked to DMX_PSEUDO_SURROUND_MODE */ SHORT numOutChannelsMin; /*!< Linked to MIN_NUMBER_OF_OUTPUT_CHANNELS */ SHORT numOutChannelsMax; /*!< Linked to MAX_NUMBER_OF_OUTPUT_CHANNELS */ UCHAR frameDelay; /*!< Linked to DMX_BS_DATA_DELAY */ } PCM_DMX_USER_PARAMS; /* Modules main data structure: */ struct PCM_DMX_INSTANCE { /* Metadata */ DMX_BS_META_DATA bsMetaData[(1) + 1]; PCM_DMX_USER_PARAMS userParams; UCHAR applyProcessing; /*!< Flag to en-/disable modules processing. The max channel limiting is done independently. */ }; /* Memory allocation macro */ C_ALLOC_MEM(PcmDmxInstance, struct PCM_DMX_INSTANCE, 1) static UINT getSpeakerDistance(PCM_DMX_SPEAKER_POSITION posA, PCM_DMX_SPEAKER_POSITION posB) { PCM_DMX_SPEAKER_POSITION diff; diff.x = posA.x - posB.x; diff.y = posA.y - posB.y; diff.z = posA.z - posB.z; return ((diff.x * diff.x) + (diff.y * diff.y) + (diff.z * diff.z)); } static PCM_DMX_SPEAKER_POSITION getSpeakerPos(AUDIO_CHANNEL_TYPE chType, UCHAR chIndex, UCHAR numChInGrp) { #define PCMDMX_SPKR_POS_X_MAX_WIDTH (3) #define PCMDMX_SPKR_POS_Y_SPREAD (2) #define PCMDMX_SPKR_POS_Z_SPREAD (2) PCM_DMX_SPEAKER_POSITION spkrPos = {0, 0, 0}; AUDIO_CHANNEL_TYPE chGrp = (AUDIO_CHANNEL_TYPE)(chType & 0x0F); unsigned fHasCenter = numChInGrp & 0x1; unsigned chGrpWidth = numChInGrp >> 1; unsigned fIsCenter = 0; unsigned fIsLfe = (chType == ACT_LFE) ? 1 : 0; int offset = 0; FDK_ASSERT(chIndex < numChInGrp); if ((chGrp == ACT_FRONT) && fHasCenter) { if (chIndex == 0) fIsCenter = 1; chIndex = (UCHAR)fMax(0, chIndex - 1); } else if (fHasCenter && (chIndex == numChInGrp - 1)) { fIsCenter = 1; } /* now all even indices are left (-) */ if (!fIsCenter) { offset = chIndex >> 1; if ((chGrp > ACT_FRONT) && (chType != ACT_SIDE) && !fIsLfe) { /* the higher the index the lower the distance to the center position */ offset = chGrpWidth - fHasCenter - offset; } if ((chIndex & 0x1) == 0) { /* even */ offset = -(offset + 1); } else { offset += 1; } } /* apply the offset */ if (chType == ACT_SIDE) { spkrPos.x = (offset < 0) ? -PCMDMX_SPKR_POS_X_MAX_WIDTH : PCMDMX_SPKR_POS_X_MAX_WIDTH; spkrPos.y = /* 1x */ PCMDMX_SPKR_POS_Y_SPREAD + (SCHAR)fAbs(offset) - 1; spkrPos.z = 0; } else { unsigned spread = ((chGrpWidth == 1) && (!fIsLfe)) ? PCMDMX_SPKR_POS_X_MAX_WIDTH - 1 : 1; spkrPos.x = (SCHAR)offset * (SCHAR)spread; if (fIsLfe) { spkrPos.y = 0; spkrPos.z = SP_Z_LFE; } else { spkrPos.y = (SCHAR)fMax((SCHAR)chGrp - 1, 0) * PCMDMX_SPKR_POS_Y_SPREAD; spkrPos.z = (SCHAR)chType >> 4; if (spkrPos.z == 2) { /* ACT_BOTTOM */ spkrPos.z = -1; } spkrPos.z *= PCMDMX_SPKR_POS_Z_SPREAD; } } return spkrPos; } /** Return the channel mode of a given horizontal channel plain (normal, top, *bottom) for a given channel configuration. NOTE: This function shall get *obsolete once the channel mode has been changed to be nonambiguous. * @param [in] Index of the requested channel plain. * @param [in] The packed channel mode for the complete channel configuration *(all plains). * @param [in] The MPEG-4 channel configuration index which is necessary in *cases where the (packed) channel mode is ambiguous. * @returns Returns the packed channel mode of the requested channel plain. **/ static PCM_DMX_CHANNEL_MODE getChMode4Plain( const int plainIndex, const PCM_DMX_CHANNEL_MODE totChMode, const int chCfg) { PCM_DMX_CHANNEL_MODE plainChMode = totChMode; switch (totChMode) { case CH_MODE_5_0_2_1: if (chCfg == 14) { switch (plainIndex) { case CH_PLAIN_BOTTOM: plainChMode = (PCM_DMX_CHANNEL_MODE)0x0000; break; case CH_PLAIN_TOP: plainChMode = CH_MODE_2_0_0_0; break; case CH_PLAIN_NORMAL: default: plainChMode = CH_MODE_3_0_2_1; break; } } break; default: break; } return plainChMode; } static inline UINT getIdxSum(UCHAR numCh) { UINT result = 0; int i; for (i = 1; i < numCh; i += 1) { result += i; } return result; } /** Evaluate a given channel configuration and extract a packed channel mode. In *addition the function generates a channel offset table for the mapping to the *internal representation. This function is the inverse to the *getChannelDescription() routine. * @param [in] The total number of channels of the given configuration. * @param [in] Array holding the corresponding channel types for each channel. * @param [in] Array holding the corresponding channel type indices for each *channel. * @param [out] Array where the buffer offsets for each channel are stored into. * @param [out] The generated packed channel mode that represents the given *input configuration. * @returns Returns an error code. **/ static PCMDMX_ERROR getChannelMode( const UINT numChannels, /* in */ const AUDIO_CHANNEL_TYPE channelType[], /* in */ UCHAR channelIndices[], /* in */ UCHAR offsetTable[(8)], /* out */ PCM_DMX_CHANNEL_MODE *chMode /* out */ ) { UINT idxSum[(3)][(4)]; UCHAR numCh[(3)][(4)]; UCHAR mapped[(8)]; PCM_DMX_SPEAKER_POSITION spkrPos[(8)]; PCMDMX_ERROR err = PCMDMX_OK; unsigned ch, numMappedInChs = 0; unsigned startSlot; unsigned stopSlot = LOW_FREQUENCY_CHANNEL; FDK_ASSERT(channelType != NULL); FDK_ASSERT(channelIndices != NULL); FDK_ASSERT(offsetTable != NULL); FDK_ASSERT(chMode != NULL); /* For details see ISO/IEC 13818-7:2005(E), 8.5.3 Channel configuration */ FDKmemclear(idxSum, (3) * (4) * sizeof(UINT)); FDKmemclear(numCh, (3) * (4) * sizeof(UCHAR)); FDKmemclear(mapped, (8) * sizeof(UCHAR)); FDKmemclear(spkrPos, (8) * sizeof(PCM_DMX_SPEAKER_POSITION)); /* Init output */ FDKmemset(offsetTable, 255, (8) * sizeof(UCHAR)); *chMode = CH_MODE_UNDEFINED; /* Determine how many channels are assigned to each channels each group: */ for (ch = 0; ch < numChannels; ch += 1) { unsigned chGrp = fMax( (channelType[ch] & 0x0F) - 1, 0); /* Assign all undefined channels (ACT_NONE) to front channels. */ numCh[channelType[ch] >> 4][chGrp] += 1; idxSum[channelType[ch] >> 4][chGrp] += channelIndices[ch]; } if (numChannels > TWO_CHANNEL) { int chGrp; /* Sanity check on the indices */ for (chGrp = 0; chGrp < (4); chGrp += 1) { int plane; for (plane = 0; plane < (3); plane += 1) { if (idxSum[plane][chGrp] != getIdxSum(numCh[plane][chGrp])) { unsigned idxCnt = 0; for (ch = 0; ch < numChannels; ch += 1) { if (channelType[ch] == (AUDIO_CHANNEL_TYPE)((plane << 4) | ((chGrp + 1) & 0xF))) { channelIndices[ch] = idxCnt++; } } err = PCMDMX_INVALID_CH_CONFIG; } } } } /* Mapping HEAT 1: * Determine the speaker position of each input channel and map it to a * internal slot if it matches exactly (with zero distance). */ for (ch = 0; ch < numChannels; ch += 1) { UINT mapDist = (unsigned)-1; unsigned mapCh, mapPos = (unsigned)-1; unsigned chGrp = fMax( (channelType[ch] & 0x0F) - 1, 0); /* Assign all undefined channels (ACT_NONE) to front channels. */ if (channelIndices[ch] >= numCh[channelType[ch] >> 4][chGrp]) return PCMDMX_INVALID_CH_CONFIG; spkrPos[ch] = getSpeakerPos(channelType[ch], channelIndices[ch], numCh[channelType[ch] >> 4][chGrp]); for (mapCh = 0; mapCh <= stopSlot; mapCh += 1) { if (offsetTable[mapCh] == 255) { UINT dist = getSpeakerDistance(spkrPos[ch], spkrSlotPos[mapCh]); if (dist < mapDist) { mapPos = mapCh; mapDist = dist; } } } if (mapDist <= PCMDMX_THRESHOLD_MAP_HEAT_1) { offsetTable[mapPos] = (UCHAR)ch; mapped[ch] = 1; numMappedInChs += 1; } } /* Mapping HEAT 2: * Go through the unmapped input channels and assign them to the internal * slots that matches best (least distance). But assign center channels to * center slots only. */ startSlot = ((numCh[CH_PLAIN_NORMAL][CH_GROUP_FRONT] & 0x1) || (numChannels >= (8))) ? 0 : 1; for (ch = 0; ch < (unsigned)numChannels; ch += 1) { if (!mapped[ch]) { UINT mapDist = (unsigned)-1; unsigned mapCh, mapPos = (unsigned)-1; for (mapCh = startSlot; mapCh <= stopSlot; mapCh += 1) { if (offsetTable[mapCh] == 255) { UINT dist = getSpeakerDistance(spkrPos[ch], spkrSlotPos[mapCh]); if (dist < mapDist) { mapPos = mapCh; mapDist = dist; } } } if ((mapPos <= stopSlot) && (mapDist < PCMDMX_THRESHOLD_MAP_HEAT_2) && (((spkrPos[ch].x != 0) && (spkrSlotPos[mapPos].x != 0)) /* XOR */ || ((spkrPos[ch].x == 0) && (spkrSlotPos[mapPos].x == 0)))) { /* Assign center channels to center slots only. */ offsetTable[mapPos] = (UCHAR)ch; mapped[ch] = 1; numMappedInChs += 1; } } } /* Mapping HEAT 3: * Assign the rest by searching for the nearest input channel for each * internal slot. */ for (ch = startSlot; (ch < (8)) && (numMappedInChs < numChannels); ch += 1) { if (offsetTable[ch] == 255) { UINT mapDist = (unsigned)-1; unsigned mapCh, mapPos = (unsigned)-1; for (mapCh = 0; mapCh < (unsigned)numChannels; mapCh += 1) { if (!mapped[mapCh]) { UINT dist = getSpeakerDistance(spkrPos[mapCh], spkrSlotPos[ch]); if (dist < mapDist) { mapPos = mapCh; mapDist = dist; } } } if (mapDist < PCMDMX_THRESHOLD_MAP_HEAT_3) { offsetTable[ch] = (UCHAR)mapPos; mapped[mapPos] = 1; numMappedInChs += 1; if ((spkrPos[mapPos].x == 0) && (spkrSlotPos[ch].x != 0) && (numChannels < (8))) { /* Skip the paired slot if we assigned a center channel. */ ch += 1; } } } } /* Finaly compose the channel mode */ for (ch = 0; ch < (4); ch += 1) { int plane, numChInGrp = 0; for (plane = 0; plane < (3); plane += 1) { numChInGrp += numCh[plane][ch]; } *chMode = (PCM_DMX_CHANNEL_MODE)(*chMode | (numChInGrp << (ch * 4))); } return err; } /** Generate a channel offset table and complete channel description for a given *(packed) channel mode. This function is the inverse to the getChannelMode() *routine but does not support weird channel configurations. * @param [in] The packed channel mode of the configuration to be processed. * @param [in] Array containing the channel mapping to be used (From MPEG PCE *ordering to whatever is required). * @param [out] Array where corresponding channel types for each channels are *stored into. * @param [out] Array where corresponding channel type indices for each output *channel are stored into. * @param [out] Array where the buffer offsets for each channel are stored into. * @returns None. **/ static void getChannelDescription( const PCM_DMX_CHANNEL_MODE chMode, /* in */ const FDK_channelMapDescr *const mapDescr, /* in */ AUDIO_CHANNEL_TYPE channelType[], /* out */ UCHAR channelIndices[], /* out */ UCHAR offsetTable[(8)] /* out */ ) { int grpIdx, plainIdx, numPlains = 1, numTotalChannels = 0; int chCfg, ch = 0; FDK_ASSERT(channelType != NULL); FDK_ASSERT(channelIndices != NULL); FDK_ASSERT(mapDescr != NULL); FDK_ASSERT(offsetTable != NULL); /* Init output arrays */ FDKmemclear(channelType, (8) * sizeof(AUDIO_CHANNEL_TYPE)); FDKmemclear(channelIndices, (8) * sizeof(UCHAR)); FDKmemset(offsetTable, 255, (8) * sizeof(UCHAR)); /* Summerize to get the total number of channels */ for (grpIdx = 0; grpIdx < (4); grpIdx += 1) { numTotalChannels += (chMode >> (grpIdx * 4)) & 0xF; } /* Get the appropriate channel map */ switch (chMode) { case CH_MODE_1_0_0_0: case CH_MODE_2_0_0_0: case CH_MODE_3_0_0_0: case CH_MODE_3_0_1_0: case CH_MODE_3_0_2_0: case CH_MODE_3_0_2_1: chCfg = numTotalChannels; break; case CH_MODE_3_0_3_1: chCfg = 11; break; case CH_MODE_3_0_4_1: chCfg = 12; break; case CH_MODE_5_0_2_1: chCfg = 7; break; default: /* fallback */ chCfg = 0; break; } /* Compose channel offset table */ for (plainIdx = 0; plainIdx < numPlains; plainIdx += 1) { PCM_DMX_CHANNEL_MODE plainChMode; UCHAR numChInGrp[(4)]; plainChMode = getChMode4Plain(plainIdx, chMode, chCfg); /* Extract the number of channels per group */ numChInGrp[CH_GROUP_FRONT] = plainChMode & 0xF; numChInGrp[CH_GROUP_SIDE] = (plainChMode >> 4) & 0xF; numChInGrp[CH_GROUP_REAR] = (plainChMode >> 8) & 0xF; numChInGrp[CH_GROUP_LFE] = (plainChMode >> 12) & 0xF; /* Non-symmetric channels */ if ((numChInGrp[CH_GROUP_FRONT] & 0x1) && (plainIdx == CH_PLAIN_NORMAL)) { /* Odd number of front channels -> we have a center channel. In MPEG-4 the center has the index 0. */ int mappedIdx = FDK_chMapDescr_getMapValue(mapDescr, (UCHAR)ch, chCfg); offsetTable[CENTER_FRONT_CHANNEL] = (UCHAR)mappedIdx; channelType[mappedIdx] = ACT_FRONT; channelIndices[mappedIdx] = 0; ch += 1; } for (grpIdx = 0; grpIdx < (4); grpIdx += 1) { AUDIO_CHANNEL_TYPE type = ACT_NONE; int chMapPos = 0, maxChannels = 0; int chIdx = 0; /* Index of channel within the specific group */ switch (grpIdx) { case CH_GROUP_FRONT: type = (AUDIO_CHANNEL_TYPE)((plainIdx << 4) | ACT_FRONT); switch (plainIdx) { default: chMapPos = LEFT_FRONT_CHANNEL; chIdx = numChInGrp[grpIdx] & 0x1; break; } maxChannels = 3; break; case CH_GROUP_SIDE: /* Always map side channels to the multipurpose group. */ type = (AUDIO_CHANNEL_TYPE)((plainIdx << 4) | ACT_SIDE); if (plainIdx == CH_PLAIN_TOP) { chMapPos = LEFT_SIDE_CHANNEL_TOP; maxChannels = 3; } else { chMapPos = LEFT_MULTIPRPS_CHANNEL; maxChannels = 2; } break; case CH_GROUP_REAR: type = (AUDIO_CHANNEL_TYPE)((plainIdx << 4) | ACT_BACK); if (plainIdx == CH_PLAIN_TOP) { chMapPos = LEFT_REAR_CHANNEL_TOP; maxChannels = 3; } else { chMapPos = LEFT_REAR_CHANNEL; maxChannels = 2; } break; case CH_GROUP_LFE: if (plainIdx == CH_PLAIN_NORMAL) { type = ACT_LFE; chMapPos = LOW_FREQUENCY_CHANNEL; maxChannels = 1; } break; default: break; } /* Map all channels in this group */ for (; chIdx < numChInGrp[grpIdx]; chIdx += 1) { int mappedIdx = FDK_chMapDescr_getMapValue(mapDescr, (UCHAR)ch, chCfg); if ((chIdx == maxChannels) || (offsetTable[chMapPos] < 255)) { /* No space left in this channel group! */ if (offsetTable[LEFT_MULTIPRPS_CHANNEL] == 255) { /* Use the multipurpose group: */ chMapPos = LEFT_MULTIPRPS_CHANNEL; } else { FDK_ASSERT(0); } } offsetTable[chMapPos] = (UCHAR)mappedIdx; channelType[mappedIdx] = type; channelIndices[mappedIdx] = (UCHAR)chIdx; chMapPos += 1; ch += 1; } } } } /** Private helper function for downmix matrix manipulation that initializes * one row in a given downmix matrix (corresponding to one output channel). * @param [inout] Pointer to fixed-point parts of the downmix matrix. * @param [inout] Pointer to scale factor matrix associated to the downmix *factors. * @param [in] Index of channel (row) to be initialized. * @returns Nothing to return. **/ static void dmxInitChannel(FIXP_DMX mixFactors[(8)][(8)], INT mixScales[(8)][(8)], const unsigned int outCh) { unsigned int inCh; for (inCh = 0; inCh < (8); inCh += 1) { if (inCh == outCh) { mixFactors[outCh][inCh] = FL2FXCONST_DMX(0.5f); mixScales[outCh][inCh] = 1; } else { mixFactors[outCh][inCh] = FL2FXCONST_DMX(0.0f); mixScales[outCh][inCh] = 0; } } } /** Private helper function for downmix matrix manipulation that does a reset * of one row in a given downmix matrix (corresponding to one output channel). * @param [inout] Pointer to fixed-point parts of the downmix matrix. * @param [inout] Pointer to scale factor matrix associated to the downmix *factors. * @param [in] Index of channel (row) to be cleared/reset. * @returns Nothing to return. **/ static void dmxClearChannel(FIXP_DMX mixFactors[(8)][(8)], INT mixScales[(8)][(8)], const unsigned int outCh) { FDK_ASSERT((outCh >= 0) && (outCh < (8))); FDKmemclear(&mixFactors[outCh], (8) * sizeof(FIXP_DMX)); FDKmemclear(&mixScales[outCh], (8) * sizeof(INT)); } /** Private helper function for downmix matrix manipulation that applies a *source channel (row) scaled by a given mix factor to a destination channel *(row) in a given downmix matrix. Existing mix factors of the destination *channel (row) will get overwritten. * @param [inout] Pointer to fixed-point parts of the downmix matrix. * @param [inout] Pointer to scale factor matrix associated to the downmix *factors. * @param [in] Index of source channel (row). * @param [in] Index of destination channel (row). * @param [in] Fixed-point part of mix factor to be applied. * @param [in] Scale factor of mix factor to be applied. * @returns Nothing to return. **/ static void dmxSetChannel(FIXP_DMX mixFactors[(8)][(8)], INT mixScales[(8)][(8)], const unsigned int dstCh, const unsigned int srcCh, const FIXP_DMX factor, const INT scale) { int ch; for (ch = 0; ch < (8); ch += 1) { if (mixFactors[srcCh][ch] != (FIXP_DMX)0) { mixFactors[dstCh][ch] = FX_DBL2FX_DMX(fMult(mixFactors[srcCh][ch], factor)); mixScales[dstCh][ch] = mixScales[srcCh][ch] + scale; } } } /** Private helper function for downmix matrix manipulation that adds a source *channel (row) scaled by a given mix factor to a destination channel (row) in a *given downmix matrix. * @param [inout] Pointer to fixed-point parts of the downmix matrix. * @param [inout] Pointer to scale factor matrix associated to the downmix *factors. * @param [in] Index of source channel (row). * @param [in] Index of destination channel (row). * @param [in] Fixed-point part of mix factor to be applied. * @param [in] Scale factor of mix factor to be applied. * @returns Nothing to return. **/ static void dmxAddChannel(FIXP_DMX mixFactors[(8)][(8)], INT mixScales[(8)][(8)], const unsigned int dstCh, const unsigned int srcCh, const FIXP_DMX factor, const INT scale) { int ch; for (ch = 0; ch < (8); ch += 1) { FIXP_DBL addFact = fMult(mixFactors[srcCh][ch], factor); if (addFact != (FIXP_DMX)0) { INT newScale = mixScales[srcCh][ch] + scale; if (mixFactors[dstCh][ch] != (FIXP_DMX)0) { if (newScale > mixScales[dstCh][ch]) { mixFactors[dstCh][ch] >>= newScale - mixScales[dstCh][ch]; } else { addFact >>= mixScales[dstCh][ch] - newScale; newScale = mixScales[dstCh][ch]; } } mixFactors[dstCh][ch] += FX_DBL2FX_DMX(addFact); mixScales[dstCh][ch] = newScale; } } } /** Private function that creates a downmix factor matrix depending on the input and output * configuration, the user parameters as well as the given metadata. This function is the modules * brain and hold all downmix algorithms. * @param [in] Flag that indicates if inChMode holds a real (packed) channel mode or has been converted to a MPEG-4 channel configuration index. * @param [in] Dependent on the inModeIsCfg flag this field hands in a (packed) channel mode or the corresponding MPEG-4 channel configuration index.of the input configuration. * @param [in] The (packed) channel mode of the output configuration. * @param [in] Pointer to structure holding all current user parameter. * @param [in] Pointer to field holding all current meta data. * @param [out] Pointer to fixed-point parts of the downmix matrix. Normalized to one scale factor. * @param [out] The common scale factor of the downmix matrix. * @returns An error code. **/ static PCMDMX_ERROR getMixFactors(const UCHAR inModeIsCfg, PCM_DMX_CHANNEL_MODE inChMode, const PCM_DMX_CHANNEL_MODE outChMode, const PCM_DMX_USER_PARAMS *pParams, const DMX_BS_META_DATA *pMetaData, FIXP_DMX mixFactors[(8)][(8)], INT *pOutScale) { PCMDMX_ERROR err = PCMDMX_OK; INT mixScales[(8)][(8)]; INT maxScale = 0; int numInChannel; int numOutChannel; int dmxMethod; unsigned int outCh, inChCfg = 0; unsigned int valid[(8)] = {0}; FDK_ASSERT(pMetaData != NULL); FDK_ASSERT(mixFactors != NULL); /* Check on a supported output configuration. Add new one only after extensive testing! */ if (!((outChMode == CH_MODE_1_0_0_0) || (outChMode == CH_MODE_2_0_0_0) || (outChMode == CH_MODE_3_0_2_1) || (outChMode == CH_MODE_3_0_4_1) || (outChMode == CH_MODE_5_0_2_1))) { FDK_ASSERT(0); } if (inModeIsCfg) { /* Convert channel config to channel mode: */ inChCfg = (unsigned int)inChMode; switch (inChCfg) { case 1: case 2: case 3: case 4: case 5: case 6: inChMode = outChModeTable[inChCfg]; break; case 11: inChMode = CH_MODE_3_0_3_1; break; case 12: inChMode = CH_MODE_3_0_4_1; break; case 7: case 14: inChMode = CH_MODE_5_0_2_1; break; default: FDK_ASSERT(0); } } /* Extract the total number of input channels */ numInChannel = (inChMode & 0xF) + ((inChMode >> 4) & 0xF) + ((inChMode >> 8) & 0xF) + ((inChMode >> 12) & 0xF); /* Extract the total number of output channels */ numOutChannel = (outChMode & 0xF) + ((outChMode >> 4) & 0xF) + ((outChMode >> 8) & 0xF) + ((outChMode >> 12) & 0xF); /* MPEG ammendment 4 aka ETSI metadata and fallback mode: */ /* Create identity DMX matrix: */ for (outCh = 0; outCh < (8); outCh += 1) { dmxInitChannel(mixFactors, mixScales, outCh); } if (((inChMode >> 12) & 0xF) == 0) { /* Clear empty or wrongly mapped input channel */ dmxClearChannel(mixFactors, mixScales, LOW_FREQUENCY_CHANNEL); } /* FIRST STAGE: */ if (numInChannel > SIX_CHANNEL) { /* Always use MPEG equations either with meta data or with default values. */ FIXP_DMX dMixFactA, dMixFactB; INT dMixScaleA, dMixScaleB; int isValidCfg = TRUE; /* Get factors from meta data */ dMixFactA = abMixLvlValueTab[pMetaData->dmixIdxA]; dMixScaleA = (pMetaData->dmixIdxA == 0) ? 1 : 0; dMixFactB = abMixLvlValueTab[pMetaData->dmixIdxB]; dMixScaleB = (pMetaData->dmixIdxB == 0) ? 1 : 0; /* Check if input is in the list of supported configurations */ switch (inChMode) { case CH_MODE_3_2_1_1: /* chCfg 11 but with side channels */ case CH_MODE_3_2_1_0: isValidCfg = FALSE; err = PCMDMX_INVALID_MODE; FDK_FALLTHROUGH; case CH_MODE_3_0_3_1: /* chCfg 11 */ /* 6.1ch: C' = C; L' = L; R' = R; LFE' = LFE; Ls' = Ls*dmix_a_idx + Cs*dmix_b_idx; Rs' = Rs*dmix_a_idx + Cs*dmix_b_idx; */ dmxClearChannel( mixFactors, mixScales, RIGHT_MULTIPRPS_CHANNEL); /* clear empty input channel */ dmxSetChannel(mixFactors, mixScales, LEFT_REAR_CHANNEL, LEFT_REAR_CHANNEL, dMixFactA, dMixScaleA); dmxSetChannel(mixFactors, mixScales, LEFT_REAR_CHANNEL, LEFT_MULTIPRPS_CHANNEL, dMixFactB, dMixScaleB); dmxSetChannel(mixFactors, mixScales, RIGHT_REAR_CHANNEL, RIGHT_REAR_CHANNEL, dMixFactA, dMixScaleA); dmxSetChannel(mixFactors, mixScales, RIGHT_REAR_CHANNEL, LEFT_MULTIPRPS_CHANNEL, dMixFactB, dMixScaleB); break; case CH_MODE_3_0_4_1: /* chCfg 12 */ /* 7.1ch Surround Back: C' = C; L' = L; R' = R; LFE' = LFE; Ls' = Ls*dmix_a_idx + Lsr*dmix_b_idx; Rs' = Rs*dmix_a_idx + Rsr*dmix_b_idx; */ dmxSetChannel(mixFactors, mixScales, LEFT_REAR_CHANNEL, LEFT_REAR_CHANNEL, dMixFactA, dMixScaleA); dmxSetChannel(mixFactors, mixScales, LEFT_REAR_CHANNEL, LEFT_MULTIPRPS_CHANNEL, dMixFactB, dMixScaleB); dmxSetChannel(mixFactors, mixScales, RIGHT_REAR_CHANNEL, RIGHT_REAR_CHANNEL, dMixFactA, dMixScaleA); dmxSetChannel(mixFactors, mixScales, RIGHT_REAR_CHANNEL, RIGHT_MULTIPRPS_CHANNEL, dMixFactB, dMixScaleB); break; case CH_MODE_5_0_1_0: case CH_MODE_5_0_1_1: dmxClearChannel(mixFactors, mixScales, RIGHT_REAR_CHANNEL); /* clear empty input channel */ dmxSetChannel(mixFactors, mixScales, RIGHT_REAR_CHANNEL, LEFT_REAR_CHANNEL, FL2FXCONST_DMX(0.5f), 1); dmxSetChannel(mixFactors, mixScales, LEFT_REAR_CHANNEL, LEFT_REAR_CHANNEL, FL2FXCONST_DMX(0.5f), 1); FDK_FALLTHROUGH; case CH_MODE_5_2_1_0: isValidCfg = FALSE; err = PCMDMX_INVALID_MODE; FDK_FALLTHROUGH; case CH_MODE_5_0_2_1: /* chCfg 7 || 14 */ if (inChCfg == 14) { /* 7.1ch Front Height: C' = C; Ls' = Ls; Rs' = Rs; LFE' = LFE; L' = L*dmix_a_idx + Lv*dmix_b_idx; R' = R*dmix_a_idx + Rv*dmix_b_idx; */ dmxSetChannel(mixFactors, mixScales, LEFT_FRONT_CHANNEL, LEFT_FRONT_CHANNEL, dMixFactA, dMixScaleA); dmxSetChannel(mixFactors, mixScales, LEFT_FRONT_CHANNEL, LEFT_MULTIPRPS_CHANNEL, dMixFactB, dMixScaleB); dmxSetChannel(mixFactors, mixScales, RIGHT_FRONT_CHANNEL, RIGHT_FRONT_CHANNEL, dMixFactA, dMixScaleA); dmxSetChannel(mixFactors, mixScales, RIGHT_FRONT_CHANNEL, RIGHT_MULTIPRPS_CHANNEL, dMixFactB, dMixScaleB); } else { /* 7.1ch Front: Ls' = Ls; Rs' = Rs; LFE' = LFE; C' = C + (Lc+Rc)*dmix_a_idx; L' = L + Lc*dmix_b_idx; R' = R + Rc*dmix_b_idx; */ dmxSetChannel(mixFactors, mixScales, CENTER_FRONT_CHANNEL, LEFT_MULTIPRPS_CHANNEL, dMixFactA, dMixScaleA); dmxSetChannel(mixFactors, mixScales, CENTER_FRONT_CHANNEL, RIGHT_MULTIPRPS_CHANNEL, dMixFactA, dMixScaleA); dmxSetChannel(mixFactors, mixScales, LEFT_FRONT_CHANNEL, LEFT_MULTIPRPS_CHANNEL, dMixFactB, dMixScaleB); dmxSetChannel(mixFactors, mixScales, LEFT_FRONT_CHANNEL, LEFT_FRONT_CHANNEL, FL2FXCONST_DMX(0.5f), 1); dmxSetChannel(mixFactors, mixScales, RIGHT_FRONT_CHANNEL, RIGHT_MULTIPRPS_CHANNEL, dMixFactB, dMixScaleB); dmxSetChannel(mixFactors, mixScales, RIGHT_FRONT_CHANNEL, RIGHT_FRONT_CHANNEL, FL2FXCONST_DMX(0.5f), 1); } break; default: /* Nothing to do. Just use the identity matrix. */ isValidCfg = FALSE; err = PCMDMX_INVALID_MODE; break; } /* Add additional DMX gain */ if ((isValidCfg == TRUE) && (pMetaData->dmxGainIdx5 != 0)) { /* Apply DMX gain 5 */ FIXP_DMX dmxGain; INT dmxScale; INT sign = (pMetaData->dmxGainIdx5 & 0x40) ? -1 : 1; INT val = pMetaData->dmxGainIdx5 & 0x3F; /* 10^(dmx_gain_5/80) */ dmxGain = FX_DBL2FX_DMX( fLdPow(FL2FXCONST_DBL(0.830482023721841f), 2, /* log2(10) */ (FIXP_DBL)(sign * val * (LONG)FL2FXCONST_DBL(0.0125f)), 0, &dmxScale)); /* Currently only positive scale factors supported! */ if (dmxScale < 0) { dmxGain >>= -dmxScale; dmxScale = 0; } dmxSetChannel(mixFactors, mixScales, CENTER_FRONT_CHANNEL, CENTER_FRONT_CHANNEL, dmxGain, dmxScale); dmxSetChannel(mixFactors, mixScales, LEFT_FRONT_CHANNEL, LEFT_FRONT_CHANNEL, dmxGain, dmxScale); dmxSetChannel(mixFactors, mixScales, RIGHT_FRONT_CHANNEL, RIGHT_FRONT_CHANNEL, dmxGain, dmxScale); dmxSetChannel(mixFactors, mixScales, LEFT_REAR_CHANNEL, LEFT_REAR_CHANNEL, dmxGain, dmxScale); dmxSetChannel(mixFactors, mixScales, RIGHT_REAR_CHANNEL, RIGHT_REAR_CHANNEL, dmxGain, dmxScale); dmxSetChannel(mixFactors, mixScales, LOW_FREQUENCY_CHANNEL, LOW_FREQUENCY_CHANNEL, dmxGain, dmxScale); } /* Mark the output channels */ valid[CENTER_FRONT_CHANNEL] = 1; valid[LEFT_FRONT_CHANNEL] = 1; valid[RIGHT_FRONT_CHANNEL] = 1; valid[LEFT_REAR_CHANNEL] = 1; valid[RIGHT_REAR_CHANNEL] = 1; valid[LOW_FREQUENCY_CHANNEL] = 1; /* Update channel mode for the next stage */ inChMode = CH_MODE_3_0_2_1; } /* For the X (> 6) to 6 channel downmix we had no choice. To mix from 6 to 2 (or 1) channel(s) we have several possibilities (MPEG DSE | MPEG PCE | ITU | ARIB | DLB). Use profile and the metadata available flags to determine which equation to use: */ #define DMX_METHOD_MPEG_AMD4 1 #define DMX_METHOD_MPEG_LEGACY 2 #define DMX_METHOD_ARIB_JAPAN 4 #define DMX_METHOD_ITU_RECOM 8 #define DMX_METHOD_CUSTOM 16 dmxMethod = DMX_METHOD_MPEG_AMD4; /* default */ if ((pParams->dmxProfile == DMX_PRFL_FORCE_MATRIX_MIX) && (pMetaData->typeFlags & TYPE_PCE_DATA)) { dmxMethod = DMX_METHOD_MPEG_LEGACY; } else if (!(pMetaData->typeFlags & (TYPE_DSE_CLEV_DATA | TYPE_DSE_SLEV_DATA))) { switch (pParams->dmxProfile) { default: case DMX_PRFL_STANDARD: /* dmxMethod = DMX_METHOD_MPEG_AMD4; */ break; case DMX_PRFL_MATRIX_MIX: case DMX_PRFL_FORCE_MATRIX_MIX: if (pMetaData->typeFlags & TYPE_PCE_DATA) { dmxMethod = DMX_METHOD_MPEG_LEGACY; } break; case DMX_PRFL_ARIB_JAPAN: dmxMethod = DMX_METHOD_ARIB_JAPAN; break; } } /* SECOND STAGE: */ if (numOutChannel <= TWO_CHANNEL) { /* Create DMX matrix according to input configuration */ switch (inChMode) { case CH_MODE_2_0_0_0: /* chCfg 2 */ /* Apply the dual channel mode. */ switch (pParams->dualChannelMode) { case CH1_MODE: /* L' = 0.707 * Ch1; R' = 0.707 * Ch1; */ dmxSetChannel(mixFactors, mixScales, LEFT_FRONT_CHANNEL, LEFT_FRONT_CHANNEL, FL2FXCONST_DMX(0.707f), 0); dmxSetChannel(mixFactors, mixScales, RIGHT_FRONT_CHANNEL, LEFT_FRONT_CHANNEL, FL2FXCONST_DMX(0.707f), 0); break; case CH2_MODE: /* L' = 0.707 * Ch2; R' = 0.707 * Ch2; */ dmxSetChannel(mixFactors, mixScales, LEFT_FRONT_CHANNEL, RIGHT_FRONT_CHANNEL, FL2FXCONST_DMX(0.707f), 0); dmxSetChannel(mixFactors, mixScales, RIGHT_FRONT_CHANNEL, RIGHT_FRONT_CHANNEL, FL2FXCONST_DMX(0.707f), 0); break; case MIXED_MODE: /* L' = 0.5*Ch1 + 0.5*Ch2; R' = 0.5*Ch1 + 0.5*Ch2; */ dmxSetChannel(mixFactors, mixScales, LEFT_FRONT_CHANNEL, LEFT_FRONT_CHANNEL, FL2FXCONST_DMX(0.5f), 0); dmxAddChannel(mixFactors, mixScales, LEFT_FRONT_CHANNEL, RIGHT_FRONT_CHANNEL, FL2FXCONST_DMX(0.5f), 0); dmxSetChannel(mixFactors, mixScales, RIGHT_FRONT_CHANNEL, LEFT_FRONT_CHANNEL, FL2FXCONST_DMX(0.5f), 0); dmxAddChannel(mixFactors, mixScales, RIGHT_FRONT_CHANNEL, RIGHT_FRONT_CHANNEL, FL2FXCONST_DMX(0.5f), 0); break; default: case STEREO_MODE: /* Nothing to do */ break; } break; /* - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - * - - - - - - - - - - - - - - - - - - - */ case CH_MODE_2_0_1_0: { FIXP_DMX sMixLvl; if (dmxMethod == DMX_METHOD_ARIB_JAPAN) { /* L' = 0.707*L + 0.5*S; R' = 0.707*R + 0.5*S; */ dmxSetChannel(mixFactors, mixScales, LEFT_FRONT_CHANNEL, LEFT_FRONT_CHANNEL, FL2FXCONST_DMX(0.707f), 0); dmxSetChannel(mixFactors, mixScales, RIGHT_FRONT_CHANNEL, RIGHT_FRONT_CHANNEL, FL2FXCONST_DMX(0.707f), 0); sMixLvl = FL2FXCONST_DMX(0.5f); } else { /* L' = L + 0.707*S; R' = R + 0.707*S; */ sMixLvl = FL2FXCONST_DMX(0.707f); } dmxAddChannel(mixFactors, mixScales, LEFT_FRONT_CHANNEL, LEFT_REAR_CHANNEL, sMixLvl, 0); dmxAddChannel(mixFactors, mixScales, RIGHT_FRONT_CHANNEL, LEFT_REAR_CHANNEL, sMixLvl, 0); } break; /* - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - * - - - - - - - - - - - - - - - - - - - */ case CH_MODE_3_0_0_0: /* chCfg 3 */ { FIXP_DMX cMixLvl; if (dmxMethod == DMX_METHOD_ARIB_JAPAN) { /* L' = 0.707*L + 0.5*C; R' = 0.707*R + 0.5*C; */ dmxSetChannel(mixFactors, mixScales, LEFT_FRONT_CHANNEL, LEFT_FRONT_CHANNEL, FL2FXCONST_DMX(0.707f), 0); dmxSetChannel(mixFactors, mixScales, RIGHT_FRONT_CHANNEL, RIGHT_FRONT_CHANNEL, FL2FXCONST_DMX(0.707f), 0); cMixLvl = FL2FXCONST_DMX(0.5f); } else { /* L' = L + 0.707*C; R' = R + 0.707*C; */ cMixLvl = FL2FXCONST_DMX(0.707f); } dmxAddChannel(mixFactors, mixScales, LEFT_FRONT_CHANNEL, CENTER_FRONT_CHANNEL, cMixLvl, 0); dmxAddChannel(mixFactors, mixScales, RIGHT_FRONT_CHANNEL, CENTER_FRONT_CHANNEL, cMixLvl, 0); } break; /* - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - * - - - - - - - - - - - - - - - - - - - */ case CH_MODE_3_0_1_0: /* chCfg 4 */ { FIXP_DMX csMixLvl; if (dmxMethod == DMX_METHOD_ARIB_JAPAN) { /* L' = 0.707*L + 0.5*C + 0.5*S; R' = 0.707*R + 0.5*C + 0.5*S; */ dmxSetChannel(mixFactors, mixScales, LEFT_FRONT_CHANNEL, LEFT_FRONT_CHANNEL, FL2FXCONST_DMX(0.707f), 0); dmxSetChannel(mixFactors, mixScales, RIGHT_FRONT_CHANNEL, RIGHT_FRONT_CHANNEL, FL2FXCONST_DMX(0.707f), 0); csMixLvl = FL2FXCONST_DMX(0.5f); } else { /* L' = L + 0.707*C + 0.707*S; R' = R + 0.707*C + 0.707*S; */ csMixLvl = FL2FXCONST_DMX(0.707f); } dmxAddChannel(mixFactors, mixScales, LEFT_FRONT_CHANNEL, CENTER_FRONT_CHANNEL, csMixLvl, 0); dmxAddChannel(mixFactors, mixScales, LEFT_FRONT_CHANNEL, LEFT_REAR_CHANNEL, csMixLvl, 0); dmxAddChannel(mixFactors, mixScales, RIGHT_FRONT_CHANNEL, CENTER_FRONT_CHANNEL, csMixLvl, 0); dmxAddChannel(mixFactors, mixScales, RIGHT_FRONT_CHANNEL, LEFT_REAR_CHANNEL, csMixLvl, 0); } break; /* - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - * - - - - - - - - - - - - - - - - - - - */ case CH_MODE_3_0_2_0: /* chCfg 5 */ case CH_MODE_3_0_2_1: /* chCfg 6 */ { switch (dmxMethod) { default: case DMX_METHOD_MPEG_AMD4: { FIXP_DMX cMixLvl, sMixLvl, lMixLvl; INT cMixScale, sMixScale, lMixScale; /* Get factors from meta data */ cMixLvl = abMixLvlValueTab[pMetaData->cLevIdx]; cMixScale = (pMetaData->cLevIdx == 0) ? 1 : 0; sMixLvl = abMixLvlValueTab[pMetaData->sLevIdx]; sMixScale = (pMetaData->sLevIdx == 0) ? 1 : 0; lMixLvl = lfeMixLvlValueTab[pMetaData->dmixIdxLfe]; if (pMetaData->dmixIdxLfe <= 1) { lMixScale = 2; } else if (pMetaData->dmixIdxLfe <= 5) { lMixScale = 1; } else { lMixScale = 0; } /* Setup the DMX matrix */ if ((pParams->pseudoSurrMode == FORCE_PS_DMX) || ((pParams->pseudoSurrMode == AUTO_PS_DMX) && (pMetaData->pseudoSurround == 1))) { /* L' = L + C*clev - (Ls+Rs)*slev + LFE*lflev; R' = R + C*clev + (Ls+Rs)*slev + LFE*lflev; */ dmxAddChannel(mixFactors, mixScales, LEFT_FRONT_CHANNEL, CENTER_FRONT_CHANNEL, cMixLvl, cMixScale); dmxAddChannel(mixFactors, mixScales, LEFT_FRONT_CHANNEL, LEFT_REAR_CHANNEL, -sMixLvl, sMixScale); dmxAddChannel(mixFactors, mixScales, LEFT_FRONT_CHANNEL, RIGHT_REAR_CHANNEL, -sMixLvl, sMixScale); dmxAddChannel(mixFactors, mixScales, LEFT_FRONT_CHANNEL, LOW_FREQUENCY_CHANNEL, lMixLvl, lMixScale); dmxAddChannel(mixFactors, mixScales, RIGHT_FRONT_CHANNEL, CENTER_FRONT_CHANNEL, cMixLvl, cMixScale); dmxAddChannel(mixFactors, mixScales, RIGHT_FRONT_CHANNEL, LEFT_REAR_CHANNEL, sMixLvl, sMixScale); dmxAddChannel(mixFactors, mixScales, RIGHT_FRONT_CHANNEL, RIGHT_REAR_CHANNEL, sMixLvl, sMixScale); dmxAddChannel(mixFactors, mixScales, RIGHT_FRONT_CHANNEL, LOW_FREQUENCY_CHANNEL, lMixLvl, lMixScale); } else { /* L' = L + C*clev + Ls*slev + LFE*llev; R' = R + C*clev + Rs*slev + LFE*llev; */ dmxAddChannel(mixFactors, mixScales, LEFT_FRONT_CHANNEL, CENTER_FRONT_CHANNEL, cMixLvl, cMixScale); dmxAddChannel(mixFactors, mixScales, LEFT_FRONT_CHANNEL, LEFT_REAR_CHANNEL, sMixLvl, sMixScale); dmxAddChannel(mixFactors, mixScales, LEFT_FRONT_CHANNEL, LOW_FREQUENCY_CHANNEL, lMixLvl, lMixScale); dmxAddChannel(mixFactors, mixScales, RIGHT_FRONT_CHANNEL, CENTER_FRONT_CHANNEL, cMixLvl, cMixScale); dmxAddChannel(mixFactors, mixScales, RIGHT_FRONT_CHANNEL, RIGHT_REAR_CHANNEL, sMixLvl, sMixScale); dmxAddChannel(mixFactors, mixScales, RIGHT_FRONT_CHANNEL, LOW_FREQUENCY_CHANNEL, lMixLvl, lMixScale); } /* Add additional DMX gain */ if (pMetaData->dmxGainIdx2 != 0) { /* Apply DMX gain 2 */ FIXP_DMX dmxGain; INT dmxScale; INT sign = (pMetaData->dmxGainIdx2 & 0x40) ? -1 : 1; INT val = pMetaData->dmxGainIdx2 & 0x3F; /* 10^(dmx_gain_2/80) */ dmxGain = FX_DBL2FX_DMX( fLdPow(FL2FXCONST_DBL(0.830482023721841f), 2, /* log2(10) */ (FIXP_DBL)(sign * val * (LONG)FL2FXCONST_DBL(0.0125f)), 0, &dmxScale)); /* Currently only positive scale factors supported! */ if (dmxScale < 0) { dmxGain >>= -dmxScale; dmxScale = 0; } dmxSetChannel(mixFactors, mixScales, LEFT_FRONT_CHANNEL, LEFT_FRONT_CHANNEL, dmxGain, dmxScale); dmxSetChannel(mixFactors, mixScales, RIGHT_FRONT_CHANNEL, RIGHT_FRONT_CHANNEL, dmxGain, dmxScale); } } break; case DMX_METHOD_ARIB_JAPAN: case DMX_METHOD_MPEG_LEGACY: { FIXP_DMX flev, clev, slevLL, slevLR, slevRL, slevRR; FIXP_DMX mtrxMixDwnCoef = mpegMixDownIdx2Coef[pMetaData->matrixMixdownIdx]; if ((pParams->pseudoSurrMode == FORCE_PS_DMX) || ((pParams->pseudoSurrMode == AUTO_PS_DMX) && (pMetaData->pseudoSurround == 1))) { if (dmxMethod == DMX_METHOD_ARIB_JAPAN) { /* 3/2 input: L' = 0.707 * [L+0.707*C-k*Ls-k*Rs]; R' = 0.707 * [R+0.707*C+k*Ls+k*Rs]; */ flev = mpegMixDownIdx2Coef[0]; /* a = 0.707 */ } else { /* 3/2 input: L' = (1.707+2*A)^-1 * [L+0.707*C-A*Ls-A*Rs]; R' = (1.707+2*A)^-1 * [R+0.707*C+A*Ls+A*Rs]; */ flev = mpegMixDownIdx2PreFact[1][pMetaData->matrixMixdownIdx]; } slevRR = slevRL = FX_DBL2FX_DMX(fMult(flev, mtrxMixDwnCoef)); slevLL = slevLR = -slevRL; } else { if (dmxMethod == DMX_METHOD_ARIB_JAPAN) { /* 3/2 input: L' = 0.707 * [L+0.707*C+k*Ls]; R' = 0.707 * [R+0.707*C+k*Rs]; */ flev = mpegMixDownIdx2Coef[0]; /* a = 0.707 */ } else { /* 3/2 input: L' = (1.707+A)^-1 * [L+0.707*C+A*Ls]; R' = (1.707+A)^-1 * [R+0.707*C+A*Rs]; */ flev = mpegMixDownIdx2PreFact[0][pMetaData->matrixMixdownIdx]; } slevRR = slevLL = FX_DBL2FX_DMX(fMult(flev, mtrxMixDwnCoef)); slevLR = slevRL = (FIXP_DMX)0; } /* common factor */ clev = FX_DBL2FX_DMX(fMult(flev, mpegMixDownIdx2Coef[0] /* 0.707 */)); dmxSetChannel(mixFactors, mixScales, LEFT_FRONT_CHANNEL, LEFT_FRONT_CHANNEL, flev, 0); dmxAddChannel(mixFactors, mixScales, LEFT_FRONT_CHANNEL, CENTER_FRONT_CHANNEL, clev, 0); dmxAddChannel(mixFactors, mixScales, LEFT_FRONT_CHANNEL, LEFT_REAR_CHANNEL, slevLL, 0); dmxAddChannel(mixFactors, mixScales, LEFT_FRONT_CHANNEL, RIGHT_REAR_CHANNEL, slevLR, 0); dmxSetChannel(mixFactors, mixScales, RIGHT_FRONT_CHANNEL, RIGHT_FRONT_CHANNEL, flev, 0); dmxAddChannel(mixFactors, mixScales, RIGHT_FRONT_CHANNEL, CENTER_FRONT_CHANNEL, clev, 0); dmxAddChannel(mixFactors, mixScales, RIGHT_FRONT_CHANNEL, LEFT_REAR_CHANNEL, slevRL, 0); dmxAddChannel(mixFactors, mixScales, RIGHT_FRONT_CHANNEL, RIGHT_REAR_CHANNEL, slevRR, 0); } break; } /* switch (dmxMethod) */ } break; default: /* This configuration does not fit to any known downmix equation! */ err = PCMDMX_INVALID_MODE; break; } /* switch (inChMode) */ /* Mark the output channels */ FDKmemclear(valid, (8) * sizeof(unsigned int)); valid[LEFT_FRONT_CHANNEL] = 1; valid[RIGHT_FRONT_CHANNEL] = 1; } if (numOutChannel == ONE_CHANNEL) { FIXP_DMX monoMixLevel; INT monoMixScale = 0; dmxClearChannel(mixFactors, mixScales, CENTER_FRONT_CHANNEL); /* C is not in the mix */ if (dmxMethod == DMX_METHOD_MPEG_LEGACY) { /* C' = (3+2*A)^-1 * [C+L+R+A*Ls+A+Rs]; */ monoMixLevel = mpegMixDownIdx2PreFact[2][pMetaData->matrixMixdownIdx]; mixFactors[CENTER_FRONT_CHANNEL][CENTER_FRONT_CHANNEL] = monoMixLevel; mixFactors[CENTER_FRONT_CHANNEL][LEFT_FRONT_CHANNEL] = monoMixLevel; mixFactors[CENTER_FRONT_CHANNEL][RIGHT_FRONT_CHANNEL] = monoMixLevel; monoMixLevel = FX_DBL2FX_DMX(fMult( monoMixLevel, mpegMixDownIdx2Coef[pMetaData->matrixMixdownIdx])); mixFactors[CENTER_FRONT_CHANNEL][LEFT_REAR_CHANNEL] = monoMixLevel; mixFactors[CENTER_FRONT_CHANNEL][RIGHT_REAR_CHANNEL] = monoMixLevel; } else { switch (dmxMethod) { case DMX_METHOD_MPEG_AMD4: /* C' = L + R; */ monoMixLevel = FL2FXCONST_DMX(0.5f); monoMixScale = 1; break; default: /* C' = 0.5*L + 0.5*R; */ monoMixLevel = FL2FXCONST_DMX(0.5f); monoMixScale = 0; break; } dmxSetChannel(mixFactors, mixScales, CENTER_FRONT_CHANNEL, LEFT_FRONT_CHANNEL, monoMixLevel, monoMixScale); dmxAddChannel(mixFactors, mixScales, CENTER_FRONT_CHANNEL, RIGHT_FRONT_CHANNEL, monoMixLevel, monoMixScale); } /* Mark the output channel */ FDKmemclear(valid, (8) * sizeof(unsigned int)); valid[CENTER_FRONT_CHANNEL] = 1; } #define MAX_SEARCH_START_VAL (-7) { LONG chSum[(8)]; INT chSumMax = MAX_SEARCH_START_VAL; /* Determine the current maximum scale factor */ for (outCh = 0; outCh < (8); outCh += 1) { if (valid[outCh] != 0) { unsigned int inCh; for (inCh = 0; inCh < (8); inCh += 1) { if (mixScales[outCh][inCh] > maxScale) { /* Store the new maximum */ maxScale = mixScales[outCh][inCh]; } } } } /* Individualy analyse output chanal levels */ for (outCh = 0; outCh < (8); outCh += 1) { chSum[outCh] = MAX_SEARCH_START_VAL; if (valid[outCh] != 0) { int ovrflwProtScale = 0; unsigned int inCh; /* Accumulate all factors for each output channel */ chSum[outCh] = 0; for (inCh = 0; inCh < (8); inCh += 1) { SHORT addFact = FX_DMX2SHRT(mixFactors[outCh][inCh]); if (mixScales[outCh][inCh] <= maxScale) { addFact >>= maxScale - mixScales[outCh][inCh]; } else { addFact <<= mixScales[outCh][inCh] - maxScale; } chSum[outCh] += addFact; } if (chSum[outCh] > (LONG)MAXVAL_SGL) { while (chSum[outCh] > (LONG)MAXVAL_SGL) { ovrflwProtScale += 1; chSum[outCh] >>= 1; } } else if (chSum[outCh] > 0) { while ((chSum[outCh] << 1) <= (LONG)MAXVAL_SGL) { ovrflwProtScale -= 1; chSum[outCh] <<= 1; } } /* Store the differential scaling in the same array */ chSum[outCh] = ovrflwProtScale; } } for (outCh = 0; outCh < (8); outCh += 1) { if ((valid[outCh] != 0) && (chSum[outCh] > chSumMax)) { /* Store the new maximum */ chSumMax = chSum[outCh]; } } maxScale = fMax(maxScale + chSumMax, 0); /* Normalize all factors */ for (outCh = 0; outCh < (8); outCh += 1) { if (valid[outCh] != 0) { unsigned int inCh; for (inCh = 0; inCh < (8); inCh += 1) { if (mixFactors[outCh][inCh] != (FIXP_DMX)0) { if (mixScales[outCh][inCh] <= maxScale) { mixFactors[outCh][inCh] >>= maxScale - mixScales[outCh][inCh]; } else { mixFactors[outCh][inCh] <<= mixScales[outCh][inCh] - maxScale; } mixScales[outCh][inCh] = maxScale; } } } } } /* return the scale factor */ *pOutScale = maxScale; return (err); } /** Open and initialize an instance of the PCM downmix module * @param [out] Pointer to a buffer receiving the handle of the new instance. * @returns Returns an error code. **/ PCMDMX_ERROR pcmDmx_Open(HANDLE_PCM_DOWNMIX *pSelf) { HANDLE_PCM_DOWNMIX self; if (pSelf == NULL) { return (PCMDMX_INVALID_HANDLE); } *pSelf = NULL; self = (HANDLE_PCM_DOWNMIX)GetPcmDmxInstance(0); if (self == NULL) { return (PCMDMX_OUT_OF_MEMORY); } /* Reset the full instance */ pcmDmx_Reset(self, PCMDMX_RESET_FULL); *pSelf = self; return (PCMDMX_OK); } /** Reset all static values like e.g. mixdown coefficients. * @param [in] Handle of PCM downmix module instance. * @param [in] Flags telling which parts of the module shall be reset. * @returns Returns an error code. **/ PCMDMX_ERROR pcmDmx_Reset(HANDLE_PCM_DOWNMIX self, UINT flags) { if (self == NULL) { return (PCMDMX_INVALID_HANDLE); } if (flags & PCMDMX_RESET_PARAMS) { PCM_DMX_USER_PARAMS *pParams = &self->userParams; pParams->dualChannelMode = STEREO_MODE; pParams->pseudoSurrMode = NEVER_DO_PS_DMX; pParams->numOutChannelsMax = (6); pParams->numOutChannelsMin = (0); pParams->frameDelay = 0; pParams->expiryFrame = (0); self->applyProcessing = 0; } if (flags & PCMDMX_RESET_BS_DATA) { int slot; /* Init all slots with a default set */ for (slot = 0; slot <= (1); slot += 1) { FDKmemcpy(&self->bsMetaData[slot], &dfltMetaData, sizeof(DMX_BS_META_DATA)); } } return (PCMDMX_OK); } /** Set one parameter for one instance of the PCM downmix module. * @param [in] Handle of PCM downmix module instance. * @param [in] Parameter to be set. * @param [in] Parameter value. * @returns Returns an error code. **/ PCMDMX_ERROR pcmDmx_SetParam(HANDLE_PCM_DOWNMIX self, const PCMDMX_PARAM param, const INT value) { switch (param) { case DMX_PROFILE_SETTING: switch ((DMX_PROFILE_TYPE)value) { case DMX_PRFL_STANDARD: case DMX_PRFL_MATRIX_MIX: case DMX_PRFL_FORCE_MATRIX_MIX: case DMX_PRFL_ARIB_JAPAN: break; default: return (PCMDMX_UNABLE_TO_SET_PARAM); } if (self == NULL) return (PCMDMX_INVALID_HANDLE); self->userParams.dmxProfile = (DMX_PROFILE_TYPE)value; break; case DMX_BS_DATA_EXPIRY_FRAME: if (self == NULL) return (PCMDMX_INVALID_HANDLE); self->userParams.expiryFrame = (value > 0) ? (UINT)value : 0; break; case DMX_BS_DATA_DELAY: if ((value > (1)) || (value < 0)) { return (PCMDMX_UNABLE_TO_SET_PARAM); } if (self == NULL) { return (PCMDMX_INVALID_HANDLE); } self->userParams.frameDelay = (UCHAR)value; break; case MIN_NUMBER_OF_OUTPUT_CHANNELS: switch (value) { /* supported output channels */ case -1: case 0: case ONE_CHANNEL: case TWO_CHANNEL: case SIX_CHANNEL: case EIGHT_CHANNEL: break; default: return (PCMDMX_UNABLE_TO_SET_PARAM); } if (self == NULL) return (PCMDMX_INVALID_HANDLE); /* Store the new value */ self->userParams.numOutChannelsMin = (value > 0) ? (SHORT)value : -1; if ((value > 0) && (self->userParams.numOutChannelsMax > 0) && (value > self->userParams .numOutChannelsMax)) { /* MIN > MAX would be an invalid state. Thus set MAX = MIN in this case. */ self->userParams.numOutChannelsMax = self->userParams.numOutChannelsMin; } break; case MAX_NUMBER_OF_OUTPUT_CHANNELS: switch (value) { /* supported output channels */ case -1: case 0: case ONE_CHANNEL: case TWO_CHANNEL: case SIX_CHANNEL: case EIGHT_CHANNEL: break; default: return (PCMDMX_UNABLE_TO_SET_PARAM); } if (self == NULL) return (PCMDMX_INVALID_HANDLE); /* Store the new value */ self->userParams.numOutChannelsMax = (value > 0) ? (SHORT)value : -1; if ((value > 0) && (value < self->userParams .numOutChannelsMin)) { /* MAX < MIN would be an invalid state. Thus set MIN = MAX in this case. */ self->userParams.numOutChannelsMin = self->userParams.numOutChannelsMax; } break; case DMX_DUAL_CHANNEL_MODE: switch ((DUAL_CHANNEL_MODE)value) { case STEREO_MODE: case CH1_MODE: case CH2_MODE: case MIXED_MODE: break; default: return (PCMDMX_UNABLE_TO_SET_PARAM); } if (self == NULL) return (PCMDMX_INVALID_HANDLE); self->userParams.dualChannelMode = (DUAL_CHANNEL_MODE)value; self->applyProcessing = ((DUAL_CHANNEL_MODE)value != STEREO_MODE) ? 1 : 0; /* Force processing if necessary. */ break; case DMX_PSEUDO_SURROUND_MODE: switch ((PSEUDO_SURROUND_MODE)value) { case NEVER_DO_PS_DMX: case AUTO_PS_DMX: case FORCE_PS_DMX: break; default: return (PCMDMX_UNABLE_TO_SET_PARAM); } if (self == NULL) return (PCMDMX_INVALID_HANDLE); self->userParams.pseudoSurrMode = (PSEUDO_SURROUND_MODE)value; break; default: return (PCMDMX_UNKNOWN_PARAM); } return (PCMDMX_OK); } /** Get one parameter value of one PCM downmix module instance. * @param [in] Handle of PCM downmix module instance. * @param [in] Parameter to be set. * @param [out] Pointer to buffer receiving the parameter value. * @returns Returns an error code. **/ PCMDMX_ERROR pcmDmx_GetParam(HANDLE_PCM_DOWNMIX self, const PCMDMX_PARAM param, INT *const pValue) { PCM_DMX_USER_PARAMS *pUsrParams; if ((self == NULL) || (pValue == NULL)) { return (PCMDMX_INVALID_HANDLE); } pUsrParams = &self->userParams; switch (param) { case DMX_PROFILE_SETTING: *pValue = (INT)pUsrParams->dmxProfile; break; case DMX_BS_DATA_EXPIRY_FRAME: *pValue = (INT)pUsrParams->expiryFrame; break; case DMX_BS_DATA_DELAY: *pValue = (INT)pUsrParams->frameDelay; break; case MIN_NUMBER_OF_OUTPUT_CHANNELS: *pValue = (INT)pUsrParams->numOutChannelsMin; break; case MAX_NUMBER_OF_OUTPUT_CHANNELS: *pValue = (INT)pUsrParams->numOutChannelsMax; break; case DMX_DUAL_CHANNEL_MODE: *pValue = (INT)pUsrParams->dualChannelMode; break; case DMX_PSEUDO_SURROUND_MODE: *pValue = (INT)pUsrParams->pseudoSurrMode; break; default: return (PCMDMX_UNKNOWN_PARAM); } return (PCMDMX_OK); } /* * Read DMX meta-data from a data stream element. */ PCMDMX_ERROR pcmDmx_Parse(HANDLE_PCM_DOWNMIX self, HANDLE_FDK_BITSTREAM hBs, UINT ancDataBits, int isMpeg2) { PCMDMX_ERROR errorStatus = PCMDMX_OK; #define MAX_DSE_ANC_BYTES (16) /* 15 bytes */ #define ANC_DATA_SYNC_BYTE (0xBC) /* ancillary data sync byte. */ DMX_BS_META_DATA *pBsMetaData; int skip4Dmx = 0, skip4Ext = 0; int dmxLvlAvail = 0, extDataAvail = 0; UINT foundNewData = 0; UINT minAncBits = ((isMpeg2) ? 5 : 3) * 8; if ((self == NULL) || (hBs == NULL)) { return (PCMDMX_INVALID_HANDLE); } /* sanity checks */ if ((ancDataBits < minAncBits) || (ancDataBits > FDKgetValidBits(hBs))) { return (PCMDMX_CORRUPT_ANC_DATA); } pBsMetaData = &self->bsMetaData[0]; if (isMpeg2) { /* skip DVD ancillary data */ FDKpushFor(hBs, 16); } /* check sync word */ if (FDKreadBits(hBs, 8) != ANC_DATA_SYNC_BYTE) { return (PCMDMX_CORRUPT_ANC_DATA); } /* skip MPEG audio type and Dolby surround mode */ FDKpushFor(hBs, 4); if (isMpeg2) { /* int numAncBytes = */ FDKreadBits(hBs, 4); /* advanced dynamic range control */ if (FDKreadBit(hBs)) skip4Dmx += 24; /* dialog normalization */ if (FDKreadBit(hBs)) skip4Dmx += 8; /* reproduction_level */ if (FDKreadBit(hBs)) skip4Dmx += 8; } else { FDKpushFor(hBs, 2); /* drc presentation mode */ pBsMetaData->pseudoSurround = (SCHAR)FDKreadBit(hBs); FDKpushFor(hBs, 4); /* reserved bits */ } /* downmixing levels MPEGx status */ dmxLvlAvail = FDKreadBit(hBs); if (isMpeg2) { /* scale factor CRC status */ if (FDKreadBit(hBs)) skip4Ext += 16; } else { /* ancillary data extension status */ extDataAvail = FDKreadBit(hBs); } /* audio coding and compression status */ if (FDKreadBit(hBs)) skip4Ext += 16; /* coarse grain timecode status */ if (FDKreadBit(hBs)) skip4Ext += 16; /* fine grain timecode status */ if (FDKreadBit(hBs)) skip4Ext += 16; /* skip the useless data to get to the DMX levels */ FDKpushFor(hBs, skip4Dmx); /* downmix_levels_MPEGX */ if (dmxLvlAvail) { if (FDKreadBit(hBs)) { /* center_mix_level_on */ pBsMetaData->cLevIdx = (UCHAR)FDKreadBits(hBs, 3); foundNewData |= TYPE_DSE_CLEV_DATA; } else { FDKreadBits(hBs, 3); } if (FDKreadBit(hBs)) { /* surround_mix_level_on */ pBsMetaData->sLevIdx = (UCHAR)FDKreadBits(hBs, 3); foundNewData |= TYPE_DSE_SLEV_DATA; } else { FDKreadBits(hBs, 3); } } /* skip the useless data to get to the ancillary data extension */ FDKpushFor(hBs, skip4Ext); /* anc data extension (MPEG-4 only) */ if (extDataAvail) { int extDmxLvlSt, extDmxGainSt, extDmxLfeSt; FDKreadBit(hBs); /* reserved bit */ extDmxLvlSt = FDKreadBit(hBs); extDmxGainSt = FDKreadBit(hBs); extDmxLfeSt = FDKreadBit(hBs); FDKreadBits(hBs, 4); /* reserved bits */ if (extDmxLvlSt) { pBsMetaData->dmixIdxA = (UCHAR)FDKreadBits(hBs, 3); pBsMetaData->dmixIdxB = (UCHAR)FDKreadBits(hBs, 3); FDKreadBits(hBs, 2); /* reserved bits */ foundNewData |= TYPE_DSE_DMIX_AB_DATA; } if (extDmxGainSt) { pBsMetaData->dmxGainIdx5 = (UCHAR)FDKreadBits(hBs, 7); FDKreadBit(hBs); /* reserved bit */ pBsMetaData->dmxGainIdx2 = (UCHAR)FDKreadBits(hBs, 7); FDKreadBit(hBs); /* reserved bit */ foundNewData |= TYPE_DSE_DMX_GAIN_DATA; } if (extDmxLfeSt) { pBsMetaData->dmixIdxLfe = (UCHAR)FDKreadBits(hBs, 4); FDKreadBits(hBs, 4); /* reserved bits */ foundNewData |= TYPE_DSE_DMIX_LFE_DATA; } } /* final sanity check on the amount of read data */ if ((INT)FDKgetValidBits(hBs) < 0) { errorStatus = PCMDMX_CORRUPT_ANC_DATA; } if ((errorStatus == PCMDMX_OK) && (foundNewData != 0)) { /* announce new data */ pBsMetaData->typeFlags |= foundNewData; /* reset expiry counter */ pBsMetaData->expiryCount = 0; } return (errorStatus); } /* * Read DMX meta-data from a data stream element. */ PCMDMX_ERROR pcmDmx_ReadDvbAncData(HANDLE_PCM_DOWNMIX self, UCHAR *pAncDataBuf, UINT ancDataBytes, int isMpeg2) { PCMDMX_ERROR errorStatus = PCMDMX_OK; FDK_BITSTREAM bs; HANDLE_FDK_BITSTREAM hBs = &bs; if (self == NULL) { return (PCMDMX_INVALID_HANDLE); } /* sanity checks */ if ((pAncDataBuf == NULL) || (ancDataBytes == 0)) { return (PCMDMX_CORRUPT_ANC_DATA); } FDKinitBitStream(hBs, pAncDataBuf, MAX_DSE_ANC_BYTES, ancDataBytes * 8, BS_READER); errorStatus = pcmDmx_Parse(self, hBs, ancDataBytes * 8, isMpeg2); return (errorStatus); } /** Set the matrix mixdown information extracted from the PCE of an AAC *bitstream. Note: Call only if matrix_mixdown_idx_present is true. * @param [in] Handle of PCM downmix module instance. * @param [in] The 2 bit matrix mixdown index extracted from PCE. * @param [in] The pseudo surround enable flag extracted from PCE. * @returns Returns an error code. **/ PCMDMX_ERROR pcmDmx_SetMatrixMixdownFromPce(HANDLE_PCM_DOWNMIX self, int matrixMixdownPresent, int matrixMixdownIdx, int pseudoSurroundEnable) { if (self == NULL) { return (PCMDMX_INVALID_HANDLE); } { DMX_BS_META_DATA *pBsMetaData = &self->bsMetaData[0]; if (matrixMixdownPresent) { pBsMetaData->pseudoSurround = (pseudoSurroundEnable) ? 1 : 0; pBsMetaData->matrixMixdownIdx = matrixMixdownIdx & 0x03; pBsMetaData->typeFlags |= TYPE_PCE_DATA; /* Reset expiry counter */ pBsMetaData->expiryCount = 0; } } return (PCMDMX_OK); } /** Apply down or up mixing. * @param [in] Handle of PCM downmix module instance. * @param [inout] Pointer to buffer that hold the time domain signal. * @param [in] Pointer where the amount of output samples is returned into. * @param [in] Size of pPcmBuf. * @param [inout] Pointer where the amount of output channels is returned into. * @param [in] Input and output samples are processed interleaved. * @param [inout] Array where the corresponding channel type for each output *audio channel is stored into. * @param [inout] Array where the corresponding channel type index for each *output audio channel is stored into. * @param [in] Array containing the out channel mapping to be used (From MPEG *PCE ordering to whatever is required). * @param [out] Pointer on a field receiving the scale factor that has to be *applied on all samples afterwards. If the handed pointer is NULL scaling is *done internally. * @returns Returns an error code. **/ PCMDMX_ERROR pcmDmx_ApplyFrame(HANDLE_PCM_DOWNMIX self, DMX_PCM *pPcmBuf, const int pcmBufSize, UINT frameSize, INT *nChannels, INT fInterleaved, AUDIO_CHANNEL_TYPE channelType[], UCHAR channelIndices[], const FDK_channelMapDescr *const mapDescr, INT *pDmxOutScale) { PCM_DMX_USER_PARAMS *pParam = NULL; PCMDMX_ERROR errorStatus = PCMDMX_OK; DUAL_CHANNEL_MODE dualChannelMode; PCM_DMX_CHANNEL_MODE inChMode; PCM_DMX_CHANNEL_MODE outChMode; INT devNull; /* Just a dummy to avoid a lot of branches in the code */ int numOutChannels, numInChannels; int inStride, outStride, offset; int dmxMaxScale, dmxScale; int slot; UCHAR inOffsetTable[(8)]; DMX_BS_META_DATA bsMetaData; if ((self == NULL) || (nChannels == NULL) || (channelType == NULL) || (channelIndices == NULL) || (!FDK_chMapDescr_isValid(mapDescr))) { return (PCMDMX_INVALID_HANDLE); } /* Init the output scaling */ dmxScale = 0; if (pDmxOutScale != NULL) { /* Avoid final scaling internally and hand it to the outside world. */ *pDmxOutScale = 0; dmxMaxScale = (3); } else { /* Apply the scaling internally. */ pDmxOutScale = &devNull; /* redirect to temporal stack memory */ dmxMaxScale = 0; } pParam = &self->userParams; numInChannels = *nChannels; /* Perform some input sanity checks */ if (pPcmBuf == NULL) { return (PCMDMX_INVALID_ARGUMENT); } if (frameSize == 0) { return (PCMDMX_INVALID_ARGUMENT); } if (numInChannels == 0) { return (PCMDMX_INVALID_ARGUMENT); } if (numInChannels > (8)) { return (PCMDMX_INVALID_CH_CONFIG); } /* Check on misconfiguration */ FDK_ASSERT((pParam->numOutChannelsMax <= 0) || (pParam->numOutChannelsMax >= pParam->numOutChannelsMin)); /* Determine if the module has to do processing */ if ((self->applyProcessing == 0) && ((pParam->numOutChannelsMax <= 0) || (pParam->numOutChannelsMax >= numInChannels)) && (pParam->numOutChannelsMin <= numInChannels)) { /* Nothing to do */ return (errorStatus); } /* Determine the number of output channels */ if ((pParam->numOutChannelsMax > 0) && (numInChannels > pParam->numOutChannelsMax)) { numOutChannels = pParam->numOutChannelsMax; } else if (numInChannels < pParam->numOutChannelsMin) { numOutChannels = pParam->numOutChannelsMin; } else { numOutChannels = numInChannels; } /* Check I/O buffer size */ if ((UINT)pcmBufSize < (UINT)numOutChannels * frameSize) { return (PCMDMX_OUTPUT_BUFFER_TOO_SMALL); } dualChannelMode = pParam->dualChannelMode; /* Analyse input channel configuration and get channel offset * table that can be accessed with the fixed channel labels. */ errorStatus = getChannelMode(numInChannels, channelType, channelIndices, inOffsetTable, &inChMode); if (PCMDMX_IS_FATAL_ERROR(errorStatus) || (inChMode == CH_MODE_UNDEFINED)) { /* We don't need to restore because the channel configuration has not been changed. Just exit. */ return (PCMDMX_INVALID_CH_CONFIG); } /* Set input stride and offset */ if (fInterleaved) { inStride = numInChannels; offset = 1; /* Channel specific offset factor */ } else { inStride = 1; offset = frameSize; /* Channel specific offset factor */ } /* Reset downmix meta data if necessary */ if ((pParam->expiryFrame > 0) && (++self->bsMetaData[0].expiryCount > pParam ->expiryFrame)) { /* The metadata read from bitstream is too old. */ #ifdef FDK_ASSERT_ENABLE PCMDMX_ERROR err = pcmDmx_Reset(self, PCMDMX_RESET_BS_DATA); FDK_ASSERT(err == PCMDMX_OK); #else pcmDmx_Reset(self, PCMDMX_RESET_BS_DATA); #endif } FDKmemcpy(&bsMetaData, &self->bsMetaData[pParam->frameDelay], sizeof(DMX_BS_META_DATA)); /* Maintain delay line */ for (slot = pParam->frameDelay; slot > 0; slot -= 1) { FDKmemcpy(&self->bsMetaData[slot], &self->bsMetaData[slot - 1], sizeof(DMX_BS_META_DATA)); } /* - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - * - - - - - - - - - - - - - - - - - - */ if (numInChannels > numOutChannels) { /* Apply downmix */ DMX_PCM *pInPcm[(8)] = {NULL}; DMX_PCM *pOutPcm[(8)] = {NULL}; FIXP_DMX mixFactors[(8)][(8)]; UCHAR outOffsetTable[(8)]; UINT sample; int chCfg = 0; int bypScale = 0; if (numInChannels > SIX_CHANNEL) { AUDIO_CHANNEL_TYPE multiPurposeChType[2]; /* Get the type of the multipurpose channels */ multiPurposeChType[0] = channelType[inOffsetTable[LEFT_MULTIPRPS_CHANNEL]]; multiPurposeChType[1] = channelType[inOffsetTable[RIGHT_MULTIPRPS_CHANNEL]]; /* Check if the input configuration is one defined in the standard. */ switch (inChMode) { case CH_MODE_5_0_2_1: /* chCfg 7 || 14 */ /* Further analyse the input config to distinguish the two * CH_MODE_5_0_2_1 configs. */ if ((multiPurposeChType[0] == ACT_FRONT_TOP) && (multiPurposeChType[1] == ACT_FRONT_TOP)) { chCfg = 14; } else { chCfg = 7; } break; case CH_MODE_3_0_3_1: /* chCfg 11 */ chCfg = 11; break; case CH_MODE_3_0_4_1: /* chCfg 12 */ chCfg = 12; break; default: chCfg = 0; /* Not a known config */ break; } } /* Set this stages output stride and channel mode: */ outStride = (fInterleaved) ? numOutChannels : 1; outChMode = outChModeTable[numOutChannels]; FDK_ASSERT(outChMode != CH_MODE_UNDEFINED); /* Get channel description and channel mapping for the desired output * configuration. */ getChannelDescription(outChMode, mapDescr, channelType, channelIndices, outOffsetTable); /* Now there is no way back because we modified the channel configuration! */ /* Create the DMX matrix */ errorStatus = getMixFactors((chCfg > 0) ? 1 : 0, (chCfg > 0) ? (PCM_DMX_CHANNEL_MODE)chCfg : inChMode, outChMode, pParam, &bsMetaData, mixFactors, &dmxScale); /* No fatal errors can occur here. The function is designed to always return a valid matrix. The error code is used to signal configurations and matrices that are not conform to any standard. */ /* Determine the final scaling */ bypScale = fMin(dmxMaxScale, dmxScale); *pDmxOutScale += bypScale; dmxScale -= bypScale; { /* Set channel pointer for input. Remove empty cols. */ int inCh, outCh, map[(8)]; int ch = 0; for (inCh = 0; inCh < (8); inCh += 1) { if (inOffsetTable[inCh] < (UCHAR)numInChannels) { pInPcm[ch] = &pPcmBuf[inOffsetTable[inCh] * offset]; map[ch++] = inCh; } } for (; ch < (8); ch += 1) { map[ch] = ch; } /* Remove unused cols from factor matrix */ for (inCh = 0; inCh < numInChannels; inCh += 1) { if (inCh != map[inCh]) { for (outCh = 0; outCh < (8); outCh += 1) { mixFactors[outCh][inCh] = mixFactors[outCh][map[inCh]]; } } } /* Set channel pointer for output. Remove empty cols. */ ch = 0; for (outCh = 0; outCh < (8); outCh += 1) { if (outOffsetTable[outCh] < (UCHAR)numOutChannels) { pOutPcm[ch] = &pPcmBuf[outOffsetTable[outCh] * offset]; map[ch++] = outCh; } } for (; ch < (8); ch += 1) { map[ch] = ch; } /* Remove unused rows from factor matrix */ for (outCh = 0; outCh < numOutChannels; outCh += 1) { if (outCh != map[outCh]) { FDKmemcpy(&mixFactors[outCh], &mixFactors[map[outCh]], (8) * sizeof(FIXP_DMX)); } } } /* Sample processing loop */ for (sample = 0; sample < frameSize; sample++) { DMX_PCM tIn[(8)] = {0}; FIXP_DBL tOut[(8)] = {(FIXP_DBL)0}; int inCh, outCh; /* Preload all input samples */ for (inCh = 0; inCh < numInChannels; inCh += 1) { if (pInPcm[inCh] != NULL) { tIn[inCh] = *pInPcm[inCh]; pInPcm[inCh] += inStride; } else { tIn[inCh] = (DMX_PCM)0; } } /* Apply downmix coefficients to input samples and accumulate for output */ for (outCh = 0; outCh < numOutChannels; outCh += 1) { for (inCh = 0; inCh < numInChannels; inCh += 1) { tOut[outCh] += fMult((DMX_PCMF)tIn[inCh], mixFactors[outCh][inCh]); } FDK_ASSERT(pOutPcm[outCh] >= pPcmBuf); FDK_ASSERT(pOutPcm[outCh] < &pPcmBuf[pcmBufSize]); /* Write sample */ *pOutPcm[outCh] = (DMX_PCM)SATURATE_SHIFT( tOut[outCh], DFRACT_BITS - DMX_PCM_BITS - dmxScale, DMX_PCM_BITS); pOutPcm[outCh] += outStride; } } /* Update the number of output channels */ *nChannels = numOutChannels; } /* - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - */ else if (numInChannels < numOutChannels) { /* Apply rudimentary upmix */ /* Set up channel pointer */ UCHAR outOffsetTable[(8)]; /* FIRST STAGE Create a stereo/dual channel signal */ if (numInChannels == ONE_CHANNEL) { DMX_PCM *pInPcm[(8)]; DMX_PCM *pOutLF, *pOutRF; UINT sample; /* Set this stages output stride and channel mode: */ outStride = (fInterleaved) ? TWO_CHANNEL : 1; outChMode = outChModeTable[TWO_CHANNEL]; /* Get channel description and channel mapping for this * stages number of output channels (always STEREO). */ getChannelDescription(outChMode, mapDescr, channelType, channelIndices, outOffsetTable); /* Now there is no way back because we modified the channel configuration! */ /* Set input channel pointer. The first channel is always at index 0. */ pInPcm[CENTER_FRONT_CHANNEL] = &pPcmBuf[(frameSize - 1) * inStride]; /* Considering input mapping could lead to a invalid pointer here if the channel is not declared to be a front channel. */ /* Set output channel pointer (for this stage). */ pOutLF = &pPcmBuf[outOffsetTable[LEFT_FRONT_CHANNEL] * offset + (frameSize - 1) * outStride]; pOutRF = &pPcmBuf[outOffsetTable[RIGHT_FRONT_CHANNEL] * offset + (frameSize - 1) * outStride]; /* 1/0 input: */ for (sample = 0; sample < frameSize; sample++) { /* L' = C; R' = C; */ *pOutLF = *pOutRF = *pInPcm[CENTER_FRONT_CHANNEL]; pInPcm[CENTER_FRONT_CHANNEL] -= inStride; pOutLF -= outStride; pOutRF -= outStride; } /* Prepare for next stage: */ inStride = outStride; inChMode = outChMode; FDKmemcpy(inOffsetTable, outOffsetTable, (8) * sizeof(UCHAR)); } /* SECOND STAGE Extend with zero channels to achieved the desired number of output channels. */ if (numOutChannels > TWO_CHANNEL) { DMX_PCM *pIn[(8)] = {NULL}; DMX_PCM *pOut[(8)] = {NULL}; UINT sample; AUDIO_CHANNEL_TYPE inChTypes[(8)]; UCHAR inChIndices[(8)]; UCHAR numChPerGrp[2][(4)]; int nContentCh = 0; /* Number of channels with content */ int nEmptyCh = 0; /* Number of channels with content */ int ch, chGrp, isCompatible = 1; /* Do not change the signalling which is the channel types and indices. Just reorder and add channels. So first save the input signalling. */ FDKmemcpy(inChTypes, channelType, numInChannels * sizeof(AUDIO_CHANNEL_TYPE)); FDKmemclear(inChTypes + numInChannels, ((8) - numInChannels) * sizeof(AUDIO_CHANNEL_TYPE)); FDKmemcpy(inChIndices, channelIndices, numInChannels * sizeof(UCHAR)); FDKmemclear(inChIndices + numInChannels, ((8) - numInChannels) * sizeof(UCHAR)); /* Set this stages output stride and channel mode: */ outStride = (fInterleaved) ? numOutChannels : 1; outChMode = outChModeTable[numOutChannels]; FDK_ASSERT(outChMode != CH_MODE_UNDEFINED); /* Check if input channel config can be easily mapped to the desired * output config. */ for (chGrp = 0; chGrp < (4); chGrp += 1) { numChPerGrp[IN][chGrp] = (inChMode >> (chGrp * 4)) & 0xF; numChPerGrp[OUT][chGrp] = (outChMode >> (chGrp * 4)) & 0xF; if (numChPerGrp[IN][chGrp] > numChPerGrp[OUT][chGrp]) { isCompatible = 0; break; } } if (isCompatible) { /* Get new channel description and channel * mapping for the desired output channel mode. */ getChannelDescription(outChMode, mapDescr, channelType, channelIndices, outOffsetTable); /* If the input config has a back center channel but the output config has not, copy it to left and right (if available). */ if ((numChPerGrp[IN][CH_GROUP_REAR] % 2) && !(numChPerGrp[OUT][CH_GROUP_REAR] % 2)) { if (numChPerGrp[IN][CH_GROUP_REAR] == 1) { inOffsetTable[RIGHT_REAR_CHANNEL] = inOffsetTable[LEFT_REAR_CHANNEL]; } else if (numChPerGrp[IN][CH_GROUP_REAR] == 3) { inOffsetTable[RIGHT_MULTIPRPS_CHANNEL] = inOffsetTable[LEFT_MULTIPRPS_CHANNEL]; } } } else { /* Just copy and extend the original config */ FDKmemcpy(outOffsetTable, inOffsetTable, (8) * sizeof(UCHAR)); } /* Set I/O channel pointer. Note: The following assignment algorithm clears the channel offset tables. Thus they can not be used afterwards. */ for (ch = 0; ch < (8); ch += 1) { if ((outOffsetTable[ch] < 255) && (inOffsetTable[ch] < 255)) { /* Set I/O pointer: */ pIn[nContentCh] = &pPcmBuf[inOffsetTable[ch] * offset + (frameSize - 1) * inStride]; pOut[nContentCh] = &pPcmBuf[outOffsetTable[ch] * offset + (frameSize - 1) * outStride]; /* Update signalling */ channelType[outOffsetTable[ch]] = inChTypes[inOffsetTable[ch]]; channelIndices[outOffsetTable[ch]] = inChIndices[inOffsetTable[ch]]; inOffsetTable[ch] = 255; outOffsetTable[ch] = 255; nContentCh += 1; } } if (isCompatible) { /* Assign the remaining input channels. This is just a safety appliance. We should never need it. */ for (ch = 0; ch < (8); ch += 1) { if (inOffsetTable[ch] < 255) { int outCh; for (outCh = 0; outCh < (8); outCh += 1) { if (outOffsetTable[outCh] < 255) { break; } } if (outCh >= (8)) { FDK_ASSERT(0); break; } /* Set I/O pointer: */ pIn[nContentCh] = &pPcmBuf[inOffsetTable[ch] * offset + (frameSize - 1) * inStride]; pOut[nContentCh] = &pPcmBuf[outOffsetTable[outCh] * offset + (frameSize - 1) * outStride]; /* Update signalling */ FDK_ASSERT(inOffsetTable[outCh] < numInChannels); FDK_ASSERT(outOffsetTable[outCh] < numOutChannels); channelType[outOffsetTable[outCh]] = inChTypes[inOffsetTable[ch]]; channelIndices[outOffsetTable[outCh]] = inChIndices[inOffsetTable[ch]]; inOffsetTable[ch] = 255; outOffsetTable[outCh] = 255; nContentCh += 1; } } /* Set the remaining output channel pointer */ for (ch = 0; ch < (8); ch += 1) { if (outOffsetTable[ch] < 255) { pOut[nContentCh + nEmptyCh] = &pPcmBuf[outOffsetTable[ch] * offset + (frameSize - 1) * outStride]; /* Expand output signalling */ channelType[outOffsetTable[ch]] = ACT_NONE; channelIndices[outOffsetTable[ch]] = (UCHAR)nEmptyCh; outOffsetTable[ch] = 255; nEmptyCh += 1; } } } else { /* Set the remaining output channel pointer */ for (ch = nContentCh; ch < numOutChannels; ch += 1) { pOut[ch] = &pPcmBuf[ch * offset + (frameSize - 1) * outStride]; /* Expand output signalling */ channelType[ch] = ACT_NONE; channelIndices[ch] = (UCHAR)nEmptyCh; nEmptyCh += 1; } } /* First copy the channels that have signal */ for (sample = 0; sample < frameSize; sample += 1) { DMX_PCM tIn[(8)]; /* Read all channel samples */ for (ch = 0; ch < nContentCh; ch += 1) { tIn[ch] = *pIn[ch]; pIn[ch] -= inStride; } /* Write all channel samples */ for (ch = 0; ch < nContentCh; ch += 1) { *pOut[ch] = tIn[ch]; pOut[ch] -= outStride; } } /* Clear all the other channels */ for (sample = 0; sample < frameSize; sample++) { for (ch = nContentCh; ch < numOutChannels; ch += 1) { *pOut[ch] = (DMX_PCM)0; pOut[ch] -= outStride; } } } /* update the number of output channels */ *nChannels = numOutChannels; } /* - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - */ else if (numInChannels == numOutChannels) { /* Don't need to change the channel description here */ switch (numInChannels) { case 2: { /* Set up channel pointer */ DMX_PCM *pInPcm[(8)]; DMX_PCM *pOutL, *pOutR; FIXP_DMX flev; UINT sample; if (fInterleaved) { inStride = numInChannels; outStride = 2; /* fixed !!! (below stereo is donwmixed to mono if required */ offset = 1; /* Channel specific offset factor */ } else { inStride = 1; outStride = 1; offset = frameSize; /* Channel specific offset factor */ } /* Set input channel pointer */ pInPcm[LEFT_FRONT_CHANNEL] = &pPcmBuf[inOffsetTable[LEFT_FRONT_CHANNEL] * offset]; pInPcm[RIGHT_FRONT_CHANNEL] = &pPcmBuf[inOffsetTable[RIGHT_FRONT_CHANNEL] * offset]; /* Set output channel pointer (same as input) */ pOutL = pInPcm[LEFT_FRONT_CHANNEL]; pOutR = pInPcm[RIGHT_FRONT_CHANNEL]; /* Set downmix levels: */ flev = FL2FXCONST_DMX(0.70710678f); /* 2/0 input: */ switch (dualChannelMode) { case CH1_MODE: /* L' = 0.707 * Ch1; R' = 0.707 * Ch1 */ for (sample = 0; sample < frameSize; sample++) { *pOutL = *pOutR = (DMX_PCM)SATURATE_RIGHT_SHIFT( fMult((DMX_PCMF)*pInPcm[LEFT_FRONT_CHANNEL], flev), DFRACT_BITS - DMX_PCM_BITS, DMX_PCM_BITS); pInPcm[LEFT_FRONT_CHANNEL] += inStride; pOutL += outStride; pOutR += outStride; } break; case CH2_MODE: /* L' = 0.707 * Ch2; R' = 0.707 * Ch2 */ for (sample = 0; sample < frameSize; sample++) { *pOutL = *pOutR = (DMX_PCM)SATURATE_RIGHT_SHIFT( fMult((DMX_PCMF)*pInPcm[RIGHT_FRONT_CHANNEL], flev), DFRACT_BITS - DMX_PCM_BITS, DMX_PCM_BITS); pInPcm[RIGHT_FRONT_CHANNEL] += inStride; pOutL += outStride; pOutR += outStride; } break; case MIXED_MODE: /* L' = 0.5*Ch1 + 0.5*Ch2; R' = 0.5*Ch1 + 0.5*Ch2 */ for (sample = 0; sample < frameSize; sample++) { *pOutL = *pOutR = (*pInPcm[LEFT_FRONT_CHANNEL] >> 1) + (*pInPcm[RIGHT_FRONT_CHANNEL] >> 1); pInPcm[LEFT_FRONT_CHANNEL] += inStride; pInPcm[RIGHT_FRONT_CHANNEL] += inStride; pOutL += outStride; pOutR += outStride; } break; default: case STEREO_MODE: /* nothing to do */ break; } } break; default: /* nothing to do */ break; } } return (errorStatus); } /** Close an instance of the PCM downmix module. * @param [inout] Pointer to a buffer containing the handle of the instance. * @returns Returns an error code. **/ PCMDMX_ERROR pcmDmx_Close(HANDLE_PCM_DOWNMIX *pSelf) { if (pSelf == NULL) { return (PCMDMX_INVALID_HANDLE); } FreePcmDmxInstance(pSelf); *pSelf = NULL; return (PCMDMX_OK); } /** Get library info for this module. * @param [out] Pointer to an allocated LIB_INFO structure. * @returns Returns an error code. */ PCMDMX_ERROR pcmDmx_GetLibInfo(LIB_INFO *info) { int i; if (info == NULL) { return PCMDMX_INVALID_ARGUMENT; } /* Search for next free tab */ for (i = 0; i < FDK_MODULE_LAST; i++) { if (info[i].module_id == FDK_NONE) break; } if (i == FDK_MODULE_LAST) { return PCMDMX_INVALID_ARGUMENT; } /* Add the library info */ info[i].module_id = FDK_PCMDMX; info[i].version = LIB_VERSION(PCMUTIL_LIB_VL0, PCMUTIL_LIB_VL1, PCMUTIL_LIB_VL2); LIB_VERSION_STRING(info + i); info[i].build_date = PCMUTIL_LIB_BUILD_DATE; info[i].build_time = PCMUTIL_LIB_BUILD_TIME; info[i].title = PCMDMX_LIB_TITLE; /* Set flags */ info[i].flags = 0 | CAPF_DMX_BLIND /* At least blind downmixing is possible */ | CAPF_DMX_PCE /* Guided downmix with data from MPEG-2/4 Program Config Elements (PCE). */ | CAPF_DMX_ARIB /* PCE guided downmix with slightly different equations and levels. */ | CAPF_DMX_DVB /* Guided downmix with data from DVB ancillary data fields. */ | CAPF_DMX_CH_EXP /* Simple upmixing by dublicating channels or adding zero channels. */ | CAPF_DMX_6_CH | CAPF_DMX_8_CH; /* Add lib info for FDK tools (if not yet done). */ FDK_toolsGetLibInfo(info); return PCMDMX_OK; } fdk-aac-2.0.2/libPCMutils/src/version.h000066400000000000000000000121401404225310700176110ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** PCM utility library ****************************** Author(s): Description: *******************************************************************************/ #if !defined(VERSION_H) #define VERSION_H /* library info */ #define PCMUTIL_LIB_VL0 3 #define PCMUTIL_LIB_VL1 1 #define PCMUTIL_LIB_VL2 0 #define PCMUTIL_LIB_TITLE "PCM Utility Lib" #ifdef SUPPRESS_BUILD_DATE_INFO #define PCMUTIL_LIB_BUILD_DATE "" #define PCMUTIL_LIB_BUILD_TIME "" #else #define PCMUTIL_LIB_BUILD_DATE __DATE__ #define PCMUTIL_LIB_BUILD_TIME __TIME__ #endif #endif /* !defined(VERSION_H) */ fdk-aac-2.0.2/libSACdec/000077500000000000000000000000001404225310700145505ustar00rootroot00000000000000fdk-aac-2.0.2/libSACdec/include/000077500000000000000000000000001404225310700161735ustar00rootroot00000000000000fdk-aac-2.0.2/libSACdec/include/sac_dec_errorcodes.h000066400000000000000000000146441404225310700221650ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround decoder library ************************* Author(s): Description: error codes for mpeg surround decoder *******************************************************************************/ #ifndef SAC_DEC_ERRORCODES_H #define SAC_DEC_ERRORCODES_H typedef enum { __mps_error_start = -1000, MPS_OK = 0, /* generic/init errors */ MPS_NOTOK = __mps_error_start, MPS_OUTOFMEMORY, MPS_INVALID_HANDLE, MPS_INVALID_PARAMETER, /* SetParam not successfull */ MPS_UNSUPPORTED_HRTFMODEL, /* SetHRTFModel() not successfull */ MPS_UNSUPPORTED_HRTFFREQ, /* SetHRTFModel() not successfull */ MPS_UNSUPPORTED_UPMIX_TYPE, /* CheckLevelTreeUpmixType() */ MPS_UNSUPPORTED_FORMAT, /* various functions; unknown aot or no_channels in filterbank */ MPS_OUTPUT_BUFFER_TOO_SMALL, /* Size of provided output time buffer is too small */ /* ssc errors */ MPS_INVALID_PARAMETERBANDS, /* unsupported numParameterBands in SpatialDecDecodeHeader() */ MPS_INVALID_TREECONFIG, MPS_INVALID_HRTFSET, /* SpatialDecDecodeHeader() */ MPS_INVALID_TTT, /* SpatialDecDecodeHeader() */ MPS_INVALID_BOXIDX, /* ecDataDec() */ MPS_INVALID_SETIDX, /* ecDataDec() */ MPS_INVALID_QUANTMODE, /* SpatialDecParseSpecificConfig() */ MPS_UNEQUAL_SSC, /* FDK_SpatialDecCompareSpatialSpecificConfigHeader() */ MPS_UNSUPPORTED_CONFIG, /* number of core channels; 3DStereoInversion; */ /* parse errors */ MPS_PARSE_ERROR, MPS_INVALID_TEMPSHAPE, /* SpatialDecParseFrameData() */ /* render errors */ MPS_WRONG_PARAMETERSETS, MPS_WRONG_PARAMETERBANDS, /* decodeAndMapFrameSmg() */ MPS_WRONG_TREECONFIG, MPS_WRONG_BLINDCONFIG, MPS_WRONG_OTT, MPS_WRONG_QUANTMODE, MPS_RESDEC_ERROR, MPS_APPLY_M2_ERROR, /* error in applyM2x()selection */ __mps_error_end } SACDEC_ERROR; #endif fdk-aac-2.0.2/libSACdec/include/sac_dec_lib.h000066400000000000000000000515151404225310700205620ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround decoder library ************************* Author(s): Description: Space Decoder *******************************************************************************/ #ifndef SAC_DEC_LIB_H #define SAC_DEC_LIB_H #include "common_fix.h" #include "FDK_audio.h" #include "sac_dec_errorcodes.h" #include "FDK_bitstream.h" #include "FDK_qmf_domain.h" #ifdef __cplusplus extern "C" { #endif /* __cplusplus */ /** * \brief MPEG Surround input data interface mode. **/ typedef enum { SAC_INTERFACE_QMF = 0, /*!< Use QMF domain interface for the input downmix audio. */ SAC_INTERFACE_TIME, /*!< Use time domain interface for the input downmix audio. */ SAC_INTERFACE_AUTO /*!< */ } SAC_INPUT_CONFIG; /** * \brief MPEG Surround output mode. **/ typedef enum { SACDEC_OUT_MODE_NORMAL = 0, /*!< Normal multi channel processing without output restrictions. */ SACDEC_OUT_MODE_BINAURAL, /*!< Two channel output with binaural processsing. */ SACDEC_OUT_MODE_STEREO, /*!< Always two channel output mode. */ SACDEC_OUT_MODE_6CHANNEL /*!< Always process with 5.1 channel output. */ } SAC_DEC_OUTPUT_MODE; /** * \brief MPEG Surround binaural HRTF model. * HRTF will be applied only in combination with upmixtype *SAC_UPMIX_TYPE_BINAURAL. **/ typedef enum { SAC_BINAURAL_HRTF_KEMAR = 0, SAC_BINAURAL_HRTF_VAST, SAC_BINAURAL_HRTF_MPSVT, SAC_BINAURAL_SINGLE_HRTFS } SAC_BINAURAL_HRTF_MODEL; /** * \brief MPEG Surround decoder instance available. **/ typedef enum { SAC_INSTANCE_NOT_FULL_AVAILABLE = 0, /*!< MPEG Surround decoder instance not full available. */ SAC_INSTANCE_FULL_AVAILABLE /*!< MPEG Surround decoder instance full available. */ } SAC_INSTANCE_AVAIL; /** * \brief MPEG Surround decoder dynamic parameters. * * Use mpegSurroundDecoder_SetParam() function to configure internal status of * following parameters. */ typedef enum { SACDEC_OUTPUT_MODE = 0x0001, /*!< Set MPEG Surround decoder output mode. See SAC_DEC_OUTPUT_MODE. */ SACDEC_BLIND_ENABLE = 0x0002, /*!< Multi channel output without MPEG Surround side info. */ SACDEC_PARTIALLY_COMPLEX = 0x0003, /*!< Set partially complex flag for MPEG Surround. 0: Use complex valued QMF data. 1: Use real valued QMF data (low power mode) */ SACDEC_INTERFACE = 0x0004, /*!< Select signal input interface for MPEG Surround. Switch time interface off: 0 Switch time interface on: 1 */ SACDEC_BS_DELAY = 0x0005, /*!< Select bit stream delay for MPEG Surround. Switch bit stream delay off: 0 Switch bit stream delay on: 1 */ SACDEC_BINAURAL_QUALITY = 0x0102, /*!< Set binaural quality for MPEG Surround binaural mode. 0: Low Complexity, 1: High Quality */ SACDEC_BINAURAL_DISTANCE = 0x0103, /*!< Set perceived distance for binaural playback (binaural mode only). The valid values range from 0 to 100. Where 100 corresponds to the farthest perceived distance. */ SACDEC_BINAURAL_DIALOG_CLARITY = 0x0104, /*!< Set dialog clarity (for binaural playback). The valid values range from 0 to 100. */ SACDEC_BINAURAL_FRONT_ANGLE = 0x0105, /*!< Set angle between the virtual front speaker pair (binaural mode only). The valid range is from 0 to 180 angular degrees. */ SACDEC_BINAURAL_BACK_ANGLE = 0x0106, /*!< Set angle between the virtual back speaker pair (binaural mode only). The valid range is from 0 to 180 angular degrees. */ SACDEC_BINAURAL_PRESET = 0x0107, /*!< Set a virtual speaker setup preset for binaural playback (binaural mode only). This meta-parameter implicitly modifies the following parameters: SACDEC_BINAURAL_DISTANCE, SACDEC_BINAURAL_DIALOG_CLARITY, SACDEC_BINAURAL_FRONT_ANGLE and SACDEC_BINAURAL_BACK_ANGLE. The following presets are available: 1: Dry room 2: Living room (default) 3: Cinema */ SACDEC_BS_INTERRUPTION = 0x0200, /*!< If the given value is unequal to 0 hint the MPEG Surround decoder that the next input data is discontinuous, because of frame loss, seeking, etc. Announce the decoder that the bitstream data was interrupted (fSync = 0). This will cause the surround decoder not to parse any new bitstream data until a new header with a valid Spatial Specific Config and a independently decodable frame is found. Specially important when the MPEG Surround data is split accross several frames (for example in the case of AAC-LC downmix with 1024 framelength and 2048 surround frame length) and a discontinuity in the bitstream data occurs. If fSync is 1, assume that MPEG Surround data is in sync (out of band config for example). */ SACDEC_CLEAR_HISTORY = 0x0201, /*!< If the given value is unequal to 0 clear all internal states (delay lines, QMF states, ...) of the MPEG Surround decoder. This will cause a discontinuity in the audio output signal. */ SACDEC_CONCEAL_NUM_KEEP_FRAMES = 0x0301, /*!< Error concealment: The Number of frames the module keeps the last spatial image before fading to the particular spatial scenario starts. The default is 10 frames. */ SACDEC_CONCEAL_FADE_OUT_SLOPE_LENGTH = 0x0302, /*!< Error concealment: Length of the slope (in frames) the module creates to fade from the last spatial scenario to the particular default scenario (downmix) in case of consecutive errors. Default is 5. */ SACDEC_CONCEAL_FADE_IN_SLOPE_LENGTH = 0x0303, /*!< Error concealment: Length of the slope (in frames) the module creates to fade from the default spatial scenario (downmix) to the current scenario after fade-out. Default parameter value is 5. */ SACDEC_CONCEAL_NUM_RELEASE_FRAMES = 0x0304 /*!< Error concealment: The number of error free frames before the module starts fading from default to the current spatial scenario. Default parameter value is 3 frames. */ } SACDEC_PARAM; #define PCM_MPS LONG /** * \brief MPEG Surround decoder handle. */ typedef struct MpegSurroundDecoder CMpegSurroundDecoder; /** * \brief Check if the full MPEG Surround decoder instance is allocated. * * Check if the full MPEG Surround decoder instance is allocated. * * \param pMpegSurroundDecoder A pointer to a decoder stucture. * * \return SACDEC_ERROR error code */ SAC_INSTANCE_AVAIL mpegSurroundDecoder_IsFullMpegSurroundDecoderInstanceAvailable( CMpegSurroundDecoder *pMpegSurroundDecoder); /** * \brief Open one instance of the MPEG Surround decoder. * * Allocate one instance of decoder and input buffers. * - Allocate decoder structure * - Allocate input buffers (QMF/time/MPS data) * * \param pMpegSurroundDecoder A pointer to a decoder handle; filled on * return. * \param splitMemoryAllocation Allocate only outer layer of MPS decoder. Core * part is reallocated later if needed. * \param stereoConfigIndex USAC: Save memory by opening the MPS decoder * for a specific stereoConfigIndex. (Needs optimization macros enabled.) * \param pQmfDomain Pointer to QMF domain data structure. * * \return SACDEC_ERROR error code */ SACDEC_ERROR mpegSurroundDecoder_Open( CMpegSurroundDecoder **pMpegSurroundDecoder, int stereoConfigIndex, HANDLE_FDK_QMF_DOMAIN pQmfDomain); /** * \brief Init one instance of the MPEG Surround decoder. * * Init one instance of the MPEG Surround decoder * * \param pMpegSurroundDecoder A pointer to a decoder handle; * * \return SACDEC_ERROR error code */ SACDEC_ERROR mpegSurroundDecoder_Init( CMpegSurroundDecoder *pMpegSurroundDecoder); /** * \brief Read and parse SpatialSpecificConfig. * * \param pMpegSurroundDecoder A pointer to a decoder handle. * \param hBs bitstream handle config parsing data source. * * \return SACDEC_ERROR error code */ SACDEC_ERROR mpegSurroundDecoder_Config( CMpegSurroundDecoder *pMpegSurroundDecoder, HANDLE_FDK_BITSTREAM hBs, AUDIO_OBJECT_TYPE coreCodec, INT samplingRate, INT frameSize, INT stereoConfigIndex, INT coreSbrFrameLengthIndex, INT configBytes, const UCHAR configMode, UCHAR *configChanged); SACDEC_ERROR mpegSurroundDecoder_ConfigureQmfDomain( CMpegSurroundDecoder *pMpegSurroundDecoder, SAC_INPUT_CONFIG sac_dec_interface, UINT coreSamplingRate, AUDIO_OBJECT_TYPE coreCodec); /** * \brief Parse MPEG Surround data without header * * \param pMpegSurroundDecoder A MPEG Surrround decoder handle. * \param hBs Bit stream handle data input source * \param pMpsDataBits Pointer to number of valid bits in extension * payload. Function updates mpsDataBits while parsing bitstream. * \param fGlobalIndependencyFlag Global independency flag of current frame. * * \return Error code. */ int mpegSurroundDecoder_ParseNoHeader( CMpegSurroundDecoder *pMpegSurroundDecoder, HANDLE_FDK_BITSTREAM hBs, int *pMpsDataBits, int fGlobalIndependencyFlag); /* #ifdef SACDEC_MPS_ENABLE */ /** * \brief Parse MPEG Surround data with header. Header is ancType, ancStart, ancStop (4 bits total). Body is ancDataSegmentByte[i]. * * \param pMpegSurroundDecoder A MPEG Surrround decoder handle. * \param hBs Bit stream handle data input source * \param pMpsDataBits Pointer to number of valid bits in extension payload. Function updates mpsDataBits while parsing bitstream. Needs to be a multiple of 8 + 4 (4 bits header). * \param coreCodec The audio object type of the core codec handling the downmix input signal. * \param sampleRate Samplerate of input downmix data. * \param nChannels Amount of input channels. * \param frameSize Amount of input samples. * \param fGlobalIndependencyFlag Global independency flag of current frame. * * \return Error code. */ int mpegSurroundDecoder_Parse(CMpegSurroundDecoder *pMpegSurroundDecoder, HANDLE_FDK_BITSTREAM hBs, int *pMpsDataBits, AUDIO_OBJECT_TYPE coreCodec, int sampleRate, int frameSize, int fGlobalIndependencyFlag); /* #endif */ /** * \brief Apply MPEG Surround upmix. * * Process one downmix audio frame and decode one surround frame if it applies. * Downmix framing can be different from surround framing, so depending on the * frame size of the downmix audio data and the framing being used by the MPEG * Surround decoder, it could be that only every second call, for example, of * this function actually surround data was decoded. The returned value of * frameSize will be zero, if no surround data was decoded. * * Decoding one MPEG Surround frame. Depending on interface configuration * mpegSurroundDecoder_SetParam(self, SACDEC_INTERFACE, value), the QMF or time * interface will be applied. External access to QMF buffer interface can be * achieved by mpegSurroundDecoder_GetQmfBuffer() call before decode frame. * While using time interface, pTimeData buffer will be shared as input and * output buffer. * * \param pMpegSurroundDecoder A MPEG Surrround decoder handle. * \param pTimeData Pointer to time buffer. Depending on interface * configuration, the content of pTimeData is ignored, and the internal QMF * buffer will be used as input data source. * Otherwise, the MPEG Surround processing is applied to the timesignal * pTimeData. For both variants, the resulting MPEG * Surround signal is written into pTimeData. * \param timeDataSize Size of pTimeData (available buffer size). * \param timeDataFrameSize Frame size of input timedata * \param nChannels Pointer where the amount of input channels is * given and amount of output channels is returned. * \param frameSize Pointer where the amount of output samples is * returned into. * \param channelType Array were the corresponding channel type for * each output audio channel is stored into. * \param channelIndices Array were the corresponding channel type index * for each output audio channel is stored into. * \param mapDescr Channep map descriptor for output channel mapping * to be used (From MPEG PCE ordering to whatever is required). * \param inDataHeadroom Headroom of SAC input time signal to prevent * clipping. * \param outDataHeadroom Pointer to headroom of SAC output time signal to * prevent clipping. * * \return Error code. */ int mpegSurroundDecoder_Apply(CMpegSurroundDecoder *pMpegSurroundDecoder, PCM_MPS *input, PCM_MPS *pTimeData, const int timeDataSize, int timeDataFrameSize, int *nChannels, int *frameSize, int sampleRate, AUDIO_OBJECT_TYPE coreCodec, AUDIO_CHANNEL_TYPE channelType[], UCHAR channelIndices[], const FDK_channelMapDescr *const mapDescr, const INT inDataHeadroom, INT *outDataHeadroom); /** * \brief Deallocate a MPEG Surround decoder instance. * \param pMpegSurroundDecoder A decoder handle. * \return No return value. */ void mpegSurroundDecoder_Close(CMpegSurroundDecoder *pMpegSurroundDecoder); /** * \brief Free config dependent MPEG Surround memory. * \param pMpegSurroundDecoder A decoder handle. * \return error. */ SACDEC_ERROR mpegSurroundDecoder_FreeMem( CMpegSurroundDecoder *pMpegSurroundDecoder); /** * \brief Set one single MPEG Surround decoder parameter. * * \param pMpegSurroundDecoder A MPEG Surrround decoder handle. Must not be * NULL pointer. * \param param Parameter to be set. See SACDEC_PARAM. * \param value Parameter value. See SACDEC_PARAM. * * \return 0 on sucess, and non-zero on failure. */ SACDEC_ERROR mpegSurroundDecoder_SetParam( CMpegSurroundDecoder *pMpegSurroundDecoder, const SACDEC_PARAM param, const INT value); /** * \brief Retrieve MPEG Surround decoder library info and fill info list with all depending library infos. * \param libInfo Pointer to library info list to be filled. * \return 0 on sucess, and non-zero on failure. **/ int mpegSurroundDecoder_GetLibInfo(LIB_INFO *libInfo); /** * \brief Set one single MPEG Surround decoder parameter. * * \param pMpegSurroundDecoder A valid MPEG Surrround decoder handle. * * \return The additional signal delay caused by the module. */ UINT mpegSurroundDecoder_GetDelay(const CMpegSurroundDecoder *self); /** * \brief Get info on whether the USAC pseudo LR feature is active. * * \param pMpegSurroundDecoder A valid MPEG Surrround decoder handle. * \param bsPseudoLr Pointer to return wether pseudo LR USAC feature * is used. * * \return 0 on sucess, and non-zero on failure. */ SACDEC_ERROR mpegSurroundDecoder_IsPseudoLR( CMpegSurroundDecoder *pMpegSurroundDecoder, int *bsPseudoLr); #ifdef __cplusplus } #endif /* __cplusplus */ #endif /* #ifndef SAC_DEC_LIB_H */ fdk-aac-2.0.2/libSACdec/src/000077500000000000000000000000001404225310700153375ustar00rootroot00000000000000fdk-aac-2.0.2/libSACdec/src/sac_bitdec.cpp000066400000000000000000002120521404225310700201250ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround decoder library ************************* Author(s): Description: SAC Dec bitstream decoder *******************************************************************************/ #include "sac_bitdec.h" #include "sac_dec_errorcodes.h" #include "nlc_dec.h" #include "sac_rom.h" #include "FDK_matrixCalloc.h" #include "sac_tsd.h" enum { ottVsTotInactiv = 0, ottVsTotDb1Activ = 1, ottVsTotDb2Activ = 2, ottVsTotDb1Db2Activ = 3 }; static SACDEC_ERROR SpatialDecDecodeHelperInfo( SPATIAL_SPECIFIC_CONFIG *pSpatialSpecificConfig, UPMIXTYPE upmixType) { int i; UINT syntaxFlags; /* Determine bit stream syntax */ syntaxFlags = 0; switch (pSpatialSpecificConfig->coreCodec) { case AOT_ER_AAC_ELD: case AOT_ER_AAC_LD: syntaxFlags |= SACDEC_SYNTAX_LD; break; case AOT_USAC: syntaxFlags |= SACDEC_SYNTAX_USAC; break; case AOT_NONE: default: return MPS_UNSUPPORTED_FORMAT; } pSpatialSpecificConfig->syntaxFlags = syntaxFlags; switch (pSpatialSpecificConfig->treeConfig) { case TREE_212: { pSpatialSpecificConfig->ottCLDdefault[0] = 0; } break; default: return MPS_INVALID_TREECONFIG; } if (syntaxFlags & SACDEC_SYNTAX_USAC) { if (pSpatialSpecificConfig->bsOttBandsPhasePresent) { pSpatialSpecificConfig->numOttBandsIPD = pSpatialSpecificConfig->bsOttBandsPhase; } else { int numParameterBands; numParameterBands = pSpatialSpecificConfig->freqRes; switch (numParameterBands) { case 4: case 5: pSpatialSpecificConfig->numOttBandsIPD = 2; break; case 7: pSpatialSpecificConfig->numOttBandsIPD = 3; break; case 10: pSpatialSpecificConfig->numOttBandsIPD = 5; break; case 14: pSpatialSpecificConfig->numOttBandsIPD = 7; break; case 20: case 28: pSpatialSpecificConfig->numOttBandsIPD = 10; break; default: return MPS_INVALID_PARAMETERBANDS; } } } else { pSpatialSpecificConfig->numOttBandsIPD = 0; } for (i = 0; i < pSpatialSpecificConfig->nOttBoxes; i++) { { pSpatialSpecificConfig->bitstreamOttBands[i] = pSpatialSpecificConfig->freqRes; } { pSpatialSpecificConfig->numOttBands[i] = pSpatialSpecificConfig->bitstreamOttBands[i]; if (syntaxFlags & SACDEC_SYNTAX_USAC && !pSpatialSpecificConfig->bsOttBandsPhasePresent) { if (pSpatialSpecificConfig->bResidualCoding && pSpatialSpecificConfig->ResidualConfig[i].bResidualPresent && (pSpatialSpecificConfig->numOttBandsIPD < pSpatialSpecificConfig->ResidualConfig[i].nResidualBands)) { pSpatialSpecificConfig->numOttBandsIPD = pSpatialSpecificConfig->ResidualConfig[i].nResidualBands; } } } } /* i */ return MPS_OK; } /******************************************************************************* Functionname: SpatialDecParseExtensionConfig ******************************************************************************* Description: Arguments: Return: *******************************************************************************/ static SACDEC_ERROR SpatialDecParseExtensionConfig( HANDLE_FDK_BITSTREAM bitstream, SPATIAL_SPECIFIC_CONFIG *config, int numOttBoxes, int numTttBoxes, int numOutChan, int bitsAvailable) { SACDEC_ERROR err = MPS_OK; INT ba = bitsAvailable; config->sacExtCnt = 0; config->bResidualCoding = 0; ba = fMin((int)FDKgetValidBits(bitstream), ba); while ((ba >= 8) && (config->sacExtCnt < MAX_NUM_EXT_TYPES)) { int bitsRead, nFillBits; INT tmp; UINT sacExtLen; config->sacExtType[config->sacExtCnt] = FDKreadBits(bitstream, 4); ba -= 4; sacExtLen = FDKreadBits(bitstream, 4); ba -= 4; if (sacExtLen == 15) { sacExtLen += FDKreadBits(bitstream, 8); ba -= 8; if (sacExtLen == 15 + 255) { sacExtLen += FDKreadBits(bitstream, 16); ba -= 16; } } tmp = (INT)FDKgetValidBits( bitstream); /* Extension config payload start anchor. */ if ((tmp <= 0) || (tmp < (INT)sacExtLen * 8) || (ba < (INT)sacExtLen * 8)) { err = MPS_PARSE_ERROR; goto bail; } switch (config->sacExtType[config->sacExtCnt]) { default:; /* unknown extension data => do nothing */ } /* skip remaining extension data */ bitsRead = tmp - FDKgetValidBits(bitstream); nFillBits = 8 * sacExtLen - bitsRead; if (nFillBits < 0) { err = MPS_PARSE_ERROR; goto bail; } else { /* Skip fill bits or an unkown extension. */ FDKpushFor(bitstream, nFillBits); } ba -= 8 * sacExtLen; config->sacExtCnt++; } bail: return err; } SACDEC_ERROR SpatialDecParseSpecificConfigHeader( HANDLE_FDK_BITSTREAM bitstream, SPATIAL_SPECIFIC_CONFIG *pSpatialSpecificConfig, AUDIO_OBJECT_TYPE coreCodec, SPATIAL_DEC_UPMIX_TYPE upmixType) { SACDEC_ERROR err = MPS_OK; INT numFillBits; int sacHeaderLen = 0; int sacTimeAlignFlag = 0; sacTimeAlignFlag = FDKreadBits(bitstream, 1); sacHeaderLen = FDKreadBits(bitstream, 7); if (sacHeaderLen == 127) { sacHeaderLen += FDKreadBits(bitstream, 16); } numFillBits = (INT)FDKgetValidBits(bitstream); err = SpatialDecParseSpecificConfig(bitstream, pSpatialSpecificConfig, sacHeaderLen, coreCodec); numFillBits -= (INT)FDKgetValidBits(bitstream); /* the number of read bits (tmpBits) */ numFillBits = (8 * sacHeaderLen) - numFillBits; if (numFillBits < 0) { /* Parsing went wrong */ err = MPS_PARSE_ERROR; } /* Move to the very end of the SSC */ FDKpushBiDirectional(bitstream, numFillBits); if ((err == MPS_OK) && sacTimeAlignFlag) { /* not supported */ FDKreadBits(bitstream, 16); err = MPS_UNSUPPORTED_CONFIG; } /* Derive additional helper variables */ SpatialDecDecodeHelperInfo(pSpatialSpecificConfig, (UPMIXTYPE)upmixType); return err; } SACDEC_ERROR SpatialDecParseMps212Config( HANDLE_FDK_BITSTREAM bitstream, SPATIAL_SPECIFIC_CONFIG *pSpatialSpecificConfig, int samplingRate, AUDIO_OBJECT_TYPE coreCodec, INT stereoConfigIndex, INT coreSbrFrameLengthIndex) { int i; FDKmemclear(pSpatialSpecificConfig, sizeof(SPATIAL_SPECIFIC_CONFIG)); pSpatialSpecificConfig->stereoConfigIndex = stereoConfigIndex; pSpatialSpecificConfig->coreSbrFrameLengthIndex = coreSbrFrameLengthIndex; pSpatialSpecificConfig->freqRes = (SPATIALDEC_FREQ_RES)freqResTable[FDKreadBits(bitstream, 3)]; if (pSpatialSpecificConfig->freqRes == 0) { return MPS_PARSE_ERROR; /* reserved value */ } switch (coreCodec) { case AOT_DRM_USAC: pSpatialSpecificConfig->bsFixedGainDMX = (SPATIALDEC_FIXED_GAINS)FDKreadBits(bitstream, 3); /* tempShapeConfig = (bsTempShapeConfigDrm == 1) ? 3 : 0 */ pSpatialSpecificConfig->tempShapeConfig = (SPATIALDEC_TS_CONF)(FDKreadBits(bitstream, 1) * 3); pSpatialSpecificConfig->decorrConfig = (SPATIALDEC_DECORR_CONF)0; pSpatialSpecificConfig->bsDecorrType = 0; break; case AOT_USAC: pSpatialSpecificConfig->bsFixedGainDMX = (SPATIALDEC_FIXED_GAINS)FDKreadBits(bitstream, 3); pSpatialSpecificConfig->tempShapeConfig = (SPATIALDEC_TS_CONF)FDKreadBits(bitstream, 2); pSpatialSpecificConfig->decorrConfig = (SPATIALDEC_DECORR_CONF)FDKreadBits(bitstream, 2); if (pSpatialSpecificConfig->decorrConfig > 2) { return MPS_PARSE_ERROR; /* reserved value */ } pSpatialSpecificConfig->bsDecorrType = 0; break; default: return MPS_UNSUPPORTED_FORMAT; } pSpatialSpecificConfig->nTimeSlots = (coreSbrFrameLengthIndex == 4) ? 64 : 32; pSpatialSpecificConfig->bsHighRateMode = (UCHAR)FDKreadBits(bitstream, 1); { pSpatialSpecificConfig->bsPhaseCoding = (UCHAR)FDKreadBits(bitstream, 1); pSpatialSpecificConfig->bsOttBandsPhasePresent = (UCHAR)FDKreadBits(bitstream, 1); if (pSpatialSpecificConfig->bsOttBandsPhasePresent) { if (MAX_PARAMETER_BANDS < (pSpatialSpecificConfig->bsOttBandsPhase = FDKreadBits(bitstream, 5))) { return MPS_PARSE_ERROR; } } else { pSpatialSpecificConfig->bsOttBandsPhase = 0; } } if (stereoConfigIndex > 1) { /* do residual coding */ pSpatialSpecificConfig->bResidualCoding = 1; pSpatialSpecificConfig->ResidualConfig->bResidualPresent = 1; if (pSpatialSpecificConfig->freqRes < (pSpatialSpecificConfig->ResidualConfig->nResidualBands = FDKreadBits(bitstream, 5))) { return MPS_PARSE_ERROR; } pSpatialSpecificConfig->bsOttBandsPhase = fMax(pSpatialSpecificConfig->bsOttBandsPhase, pSpatialSpecificConfig->ResidualConfig->nResidualBands); pSpatialSpecificConfig->bsPseudoLr = (UCHAR)FDKreadBits(bitstream, 1); if (pSpatialSpecificConfig->bsPhaseCoding) { pSpatialSpecificConfig->bsPhaseCoding = 3; } } else { pSpatialSpecificConfig->bResidualCoding = 0; pSpatialSpecificConfig->ResidualConfig->bResidualPresent = 0; } if (pSpatialSpecificConfig->tempShapeConfig == 2) { switch (coreCodec) { case AOT_USAC: pSpatialSpecificConfig->envQuantMode = FDKreadBits(bitstream, 1); break; default: /* added to avoid compiler warning */ break; /* added to avoid compiler warning */ } } /* Static parameters */ pSpatialSpecificConfig->samplingFreq = samplingRate; /* wrong for stereoConfigIndex == 3 but value is unused */ pSpatialSpecificConfig->treeConfig = SPATIALDEC_MODE_RSVD7; pSpatialSpecificConfig->nOttBoxes = treePropertyTable[pSpatialSpecificConfig->treeConfig].numOttBoxes; pSpatialSpecificConfig->nInputChannels = treePropertyTable[pSpatialSpecificConfig->treeConfig].numInputChannels; pSpatialSpecificConfig->nOutputChannels = treePropertyTable[pSpatialSpecificConfig->treeConfig].numOutputChannels; pSpatialSpecificConfig->bArbitraryDownmix = 0; for (i = 0; i < pSpatialSpecificConfig->nOttBoxes; i++) { pSpatialSpecificConfig->OttConfig[i].nOttBands = 0; } if (coreCodec == AOT_DRM_USAC) { /* MPS payload is MPEG conform -> no need for pseudo DRM AOT */ coreCodec = AOT_USAC; } pSpatialSpecificConfig->coreCodec = coreCodec; /* Derive additional helper variables */ SpatialDecDecodeHelperInfo(pSpatialSpecificConfig, UPMIXTYPE_NORMAL); return MPS_OK; } SACDEC_ERROR SpatialDecParseSpecificConfig( HANDLE_FDK_BITSTREAM bitstream, SPATIAL_SPECIFIC_CONFIG *pSpatialSpecificConfig, int sacHeaderLen, AUDIO_OBJECT_TYPE coreCodec) { SACDEC_ERROR err = MPS_OK; int i; int bsSamplingFreqIndex; int bsFreqRes, b3DaudioMode = 0; int numHeaderBits; int cfgStartPos, bitsAvailable; int treeConfig; FDKmemclear(pSpatialSpecificConfig, sizeof(SPATIAL_SPECIFIC_CONFIG)); cfgStartPos = FDKgetValidBits(bitstream); /* It might be that we do not know the SSC length beforehand. */ if (sacHeaderLen == 0) { bitsAvailable = cfgStartPos; } else { bitsAvailable = 8 * sacHeaderLen; if (bitsAvailable > cfgStartPos) { err = MPS_PARSE_ERROR; goto bail; } } bsSamplingFreqIndex = FDKreadBits(bitstream, 4); if (bsSamplingFreqIndex == 15) { pSpatialSpecificConfig->samplingFreq = FDKreadBits(bitstream, 24); } else { pSpatialSpecificConfig->samplingFreq = samplingFreqTable[bsSamplingFreqIndex]; if (pSpatialSpecificConfig->samplingFreq == 0) { err = MPS_PARSE_ERROR; goto bail; } } pSpatialSpecificConfig->nTimeSlots = FDKreadBits(bitstream, 5) + 1; if ((pSpatialSpecificConfig->nTimeSlots < 1) || (pSpatialSpecificConfig->nTimeSlots > MAX_TIME_SLOTS)) { err = MPS_PARSE_ERROR; goto bail; } bsFreqRes = FDKreadBits(bitstream, 3); pSpatialSpecificConfig->freqRes = (SPATIALDEC_FREQ_RES)freqResTable_LD[bsFreqRes]; treeConfig = FDKreadBits(bitstream, 4); if (treeConfig != SPATIALDEC_MODE_RSVD7) { err = MPS_UNSUPPORTED_CONFIG; goto bail; } pSpatialSpecificConfig->treeConfig = (SPATIALDEC_TREE_CONFIG) treeConfig; { pSpatialSpecificConfig->nOttBoxes = treePropertyTable[pSpatialSpecificConfig->treeConfig].numOttBoxes; pSpatialSpecificConfig->nTttBoxes = treePropertyTable[pSpatialSpecificConfig->treeConfig].numTttBoxes; pSpatialSpecificConfig->nInputChannels = treePropertyTable[pSpatialSpecificConfig->treeConfig].numInputChannels; pSpatialSpecificConfig->nOutputChannels = treePropertyTable[pSpatialSpecificConfig->treeConfig].numOutputChannels; } pSpatialSpecificConfig->quantMode = (SPATIALDEC_QUANT_MODE)FDKreadBits(bitstream, 2); pSpatialSpecificConfig->bArbitraryDownmix = FDKreadBits(bitstream, 1); pSpatialSpecificConfig->bsFixedGainDMX = (SPATIALDEC_FIXED_GAINS)FDKreadBits(bitstream, 3); pSpatialSpecificConfig->tempShapeConfig = (SPATIALDEC_TS_CONF)FDKreadBits(bitstream, 2); if (pSpatialSpecificConfig->tempShapeConfig > 2) { return MPS_PARSE_ERROR; /* reserved value */ } pSpatialSpecificConfig->decorrConfig = (SPATIALDEC_DECORR_CONF)FDKreadBits(bitstream, 2); if (pSpatialSpecificConfig->decorrConfig > 2) { return MPS_PARSE_ERROR; /* reserved value */ } for (i = 0; i < pSpatialSpecificConfig->nOttBoxes; i++) { pSpatialSpecificConfig->OttConfig[i].nOttBands = 0; } for (i = 0; i < pSpatialSpecificConfig->nTttBoxes; i++) { int bTttDualMode = FDKreadBits(bitstream, 1); FDKreadBits(bitstream, 3); /* not supported */ if (bTttDualMode) { FDKreadBits(bitstream, 8); /* not supported */ } } if (pSpatialSpecificConfig->tempShapeConfig == 2) { pSpatialSpecificConfig->envQuantMode = FDKreadBits(bitstream, 1); } if (b3DaudioMode) { if (FDKreadBits(bitstream, 2) == 0) { /* b3DaudioHRTFset ? */ int hc; int HRTFnumBand; int HRTFfreqRes = FDKreadBits(bitstream, 3); int HRTFnumChan = FDKreadBits(bitstream, 4); int HRTFasymmetric = FDKreadBits(bitstream, 1); HRTFnumBand = freqResTable_LD[HRTFfreqRes]; for (hc = 0; hc < HRTFnumChan; hc++) { FDKpushFor(bitstream, HRTFnumBand * 6); /* HRTFlevelLeft[hc][hb] */ if (HRTFasymmetric) { FDKpushFor(bitstream, HRTFnumBand * 6); /* HRTFlevelRight[hc][hb] */ } if (FDKreadBits(bitstream, 1)) { /* HRTFphase[hc] ? */ FDKpushFor(bitstream, HRTFnumBand * 6); /* HRTFphaseLR[hc][hb] */ } if (FDKreadBits(bitstream, 1)) { /* HRTFicc[hc] ? */ FDKpushFor(bitstream, HRTFnumBand * 6); /* HRTFiccLR[hc][hb] */ } } } } FDKbyteAlign(bitstream, cfgStartPos); /* ISO/IEC FDIS 23003-1: 5.2. ... byte alignment with respect to the beginning of the syntactic element in which ByteAlign() occurs. */ numHeaderBits = cfgStartPos - (INT)FDKgetValidBits(bitstream); bitsAvailable -= numHeaderBits; if (bitsAvailable < 0) { err = MPS_PARSE_ERROR; goto bail; } pSpatialSpecificConfig->sacExtCnt = 0; pSpatialSpecificConfig->bResidualCoding = 0; err = SpatialDecParseExtensionConfig( bitstream, pSpatialSpecificConfig, pSpatialSpecificConfig->nOttBoxes, pSpatialSpecificConfig->nTttBoxes, pSpatialSpecificConfig->nOutputChannels, bitsAvailable); FDKbyteAlign( bitstream, cfgStartPos); /* Same alignment anchor as above because SpatialExtensionConfig() always reads full bytes */ pSpatialSpecificConfig->coreCodec = coreCodec; SpatialDecDecodeHelperInfo(pSpatialSpecificConfig, UPMIXTYPE_NORMAL); bail: if (sacHeaderLen > 0) { /* If the config is of known length then assure that the bitbuffer is exactly at its end when leaving the function. */ FDKpushBiDirectional( bitstream, (sacHeaderLen * 8) - (cfgStartPos - (INT)FDKgetValidBits(bitstream))); } return err; } int SpatialDecDefaultSpecificConfig( SPATIAL_SPECIFIC_CONFIG *pSpatialSpecificConfig, AUDIO_OBJECT_TYPE coreCodec, int samplingFreq, int nTimeSlots, int sacDecoderLevel, int isBlind, int numCoreChannels) { int err = MPS_OK; int i; FDK_ASSERT(coreCodec != AOT_NONE); FDK_ASSERT(nTimeSlots > 0); FDK_ASSERT(samplingFreq > 0); pSpatialSpecificConfig->coreCodec = coreCodec; pSpatialSpecificConfig->samplingFreq = samplingFreq; pSpatialSpecificConfig->nTimeSlots = nTimeSlots; if ((pSpatialSpecificConfig->coreCodec == AOT_ER_AAC_ELD) || (pSpatialSpecificConfig->coreCodec == AOT_ER_AAC_LD)) pSpatialSpecificConfig->freqRes = SPATIALDEC_FREQ_RES_23; else pSpatialSpecificConfig->freqRes = SPATIALDEC_FREQ_RES_28; { pSpatialSpecificConfig->treeConfig = SPATIALDEC_MODE_RSVD7; /* 212 configuration */ } { pSpatialSpecificConfig->nOttBoxes = treePropertyTable[pSpatialSpecificConfig->treeConfig].numOttBoxes; pSpatialSpecificConfig->nInputChannels = treePropertyTable[pSpatialSpecificConfig->treeConfig].numInputChannels; pSpatialSpecificConfig->nOutputChannels = treePropertyTable[pSpatialSpecificConfig->treeConfig].numOutputChannels; } pSpatialSpecificConfig->quantMode = SPATIALDEC_QUANT_FINE_DEF; pSpatialSpecificConfig->bArbitraryDownmix = 0; pSpatialSpecificConfig->bResidualCoding = 0; if ((pSpatialSpecificConfig->coreCodec == AOT_ER_AAC_ELD) || (pSpatialSpecificConfig->coreCodec == AOT_ER_AAC_LD)) pSpatialSpecificConfig->bsFixedGainDMX = SPATIALDEC_GAIN_RSVD2; else pSpatialSpecificConfig->bsFixedGainDMX = SPATIALDEC_GAIN_MODE0; pSpatialSpecificConfig->tempShapeConfig = SPATIALDEC_TS_TPNOWHITE; pSpatialSpecificConfig->decorrConfig = SPATIALDEC_DECORR_MODE0; for (i = 0; i < pSpatialSpecificConfig->nOttBoxes; i++) { pSpatialSpecificConfig->OttConfig[i].nOttBands = 0; } return err; } /******************************************************************************* Functionname: coarse2fine ******************************************************************************* Description: Parameter index mapping from coarse to fine quantization Arguments: Input: Output: *******************************************************************************/ static void coarse2fine(SCHAR *data, DATA_TYPE dataType, int startBand, int numBands) { int i; for (i = startBand; i < startBand + numBands; i++) { data[i] <<= 1; } if (dataType == t_CLD) { for (i = startBand; i < startBand + numBands; i++) { if (data[i] == -14) data[i] = -15; else if (data[i] == 14) data[i] = 15; } } } /******************************************************************************* Functionname: fine2coarse ******************************************************************************* Description: Parameter index mapping from fine to coarse quantization Arguments: Input: Output: *******************************************************************************/ static void fine2coarse(SCHAR *data, DATA_TYPE dataType, int startBand, int numBands) { int i; for (i = startBand; i < startBand + numBands; i++) { /* Note: the if cases below actually make a difference (negative values) */ if (dataType == t_CLD) data[i] /= 2; else data[i] >>= 1; } } /******************************************************************************* Functionname: getStrideMap ******************************************************************************* Description: Index Mapping accroding to pbStrides Arguments: Input: Output: *******************************************************************************/ static int getStrideMap(int freqResStride, int startBand, int stopBand, int *aStrides) { int i, pb, pbStride, dataBands, strOffset; pbStride = pbStrideTable[freqResStride]; dataBands = (stopBand - startBand - 1) / pbStride + 1; aStrides[0] = startBand; for (pb = 1; pb <= dataBands; pb++) { aStrides[pb] = aStrides[pb - 1] + pbStride; } strOffset = 0; while (aStrides[dataBands] > stopBand) { if (strOffset < dataBands) strOffset++; for (i = strOffset; i <= dataBands; i++) { aStrides[i]--; } } return dataBands; } /******************************************************************************* Functionname: ecDataDec ******************************************************************************* Description: Do delta decoding and dequantization Arguments: Input: Output: *******************************************************************************/ static SACDEC_ERROR ecDataDec( const SPATIAL_BS_FRAME *frame, UINT syntaxFlags, HANDLE_FDK_BITSTREAM bitstream, LOSSLESSDATA *const llData, SCHAR (*data)[MAX_PARAMETER_SETS][MAX_PARAMETER_BANDS], SCHAR **lastdata, int datatype, int boxIdx, int startBand, int stopBand, SCHAR defaultValue) { SACDEC_ERROR err = MPS_OK; int i, j, pb, dataSets, setIdx, bsDataPair, dataBands, oldQuantCoarseXXX; INT aStrides[MAX_PARAMETER_BANDS + 1] = {0}; dataSets = 0; for (i = 0; i < frame->numParameterSets; i++) { llData->bsXXXDataMode[i] = (SCHAR)FDKreadBits(bitstream, 2); if ((frame->bsIndependencyFlag == 1) && (i == 0) && (llData->bsXXXDataMode[i] == 1 || llData->bsXXXDataMode[i] == 2)) { /* This check catches bitstreams generated by older encoder that cause trouble */ return MPS_PARSE_ERROR; } if ((i >= frame->numParameterSets - 1) && (llData->bsXXXDataMode[i] == 2)) { /* The interpolation mode must not be active for the last parameter set */ return MPS_PARSE_ERROR; } if (llData->bsXXXDataMode[i] == 3) { dataSets++; } } setIdx = 0; bsDataPair = 0; oldQuantCoarseXXX = llData->state->bsQuantCoarseXXXprevParse; for (i = 0; i < frame->numParameterSets; i++) { if (llData->bsXXXDataMode[i] == 0) { for (pb = startBand; pb < stopBand; pb++) { lastdata[boxIdx][pb] = defaultValue; } oldQuantCoarseXXX = 0; } if (llData->bsXXXDataMode[i] == 3) { if (bsDataPair) { bsDataPair = 0; } else { bsDataPair = FDKreadBits(bitstream, 1); llData->bsQuantCoarseXXX[setIdx] = (UCHAR)FDKreadBits(bitstream, 1); llData->bsFreqResStrideXXX[setIdx] = (UCHAR)FDKreadBits(bitstream, 2); if (llData->bsQuantCoarseXXX[setIdx] != oldQuantCoarseXXX) { if (oldQuantCoarseXXX) { coarse2fine(lastdata[boxIdx], (DATA_TYPE)datatype, startBand, stopBand - startBand); } else { fine2coarse(lastdata[boxIdx], (DATA_TYPE)datatype, startBand, stopBand - startBand); } } dataBands = getStrideMap(llData->bsFreqResStrideXXX[setIdx], startBand, stopBand, aStrides); for (pb = 0; pb < dataBands; pb++) { lastdata[boxIdx][startBand + pb] = lastdata[boxIdx][aStrides[pb]]; } if (boxIdx > MAX_NUM_OTT) return MPS_INVALID_BOXIDX; if ((setIdx + bsDataPair) > MAX_PARAMETER_SETS) return MPS_INVALID_SETIDX; /* DECODER_TYPE defined in FDK_tools */ DECODER_TYPE this_decoder_type = SAC_DECODER; if (syntaxFlags & (SACDEC_SYNTAX_USAC | SACDEC_SYNTAX_RSVD50)) { this_decoder_type = USAC_DECODER; } else if (syntaxFlags & SACDEC_SYNTAX_LD) { this_decoder_type = SAOC_DECODER; } err = (SACDEC_ERROR)EcDataPairDec( this_decoder_type, bitstream, data[boxIdx][setIdx + 0], data[boxIdx][setIdx + 1], lastdata[boxIdx], (DATA_TYPE)datatype, startBand, dataBands, bsDataPair, llData->bsQuantCoarseXXX[setIdx], !(frame->bsIndependencyFlag && (i == 0)) || (setIdx > 0)); if (err != MPS_OK) goto bail; if (datatype == t_IPD) { const SCHAR mask = (llData->bsQuantCoarseXXX[setIdx]) ? 7 : 15; for (pb = 0; pb < dataBands; pb++) { for (j = aStrides[pb]; j < aStrides[pb + 1]; j++) { lastdata[boxIdx][j] = data[boxIdx][setIdx + bsDataPair][startBand + pb] & mask; } } } else { for (pb = 0; pb < dataBands; pb++) { for (j = aStrides[pb]; j < aStrides[pb + 1]; j++) { lastdata[boxIdx][j] = data[boxIdx][setIdx + bsDataPair][startBand + pb]; } } } oldQuantCoarseXXX = llData->bsQuantCoarseXXX[setIdx]; if (bsDataPair) { llData->bsQuantCoarseXXX[setIdx + 1] = llData->bsQuantCoarseXXX[setIdx]; llData->bsFreqResStrideXXX[setIdx + 1] = llData->bsFreqResStrideXXX[setIdx]; } setIdx += bsDataPair + 1; } /* !bsDataPair */ } /* llData->bsXXXDataMode[i] == 3 */ } llData->state->bsQuantCoarseXXXprevParse = oldQuantCoarseXXX; bail: return err; } /******************************************************************************* Functionname: parseArbitraryDownmixData ******************************************************************************* Description: Arguments: Return: *******************************************************************************/ static SACDEC_ERROR parseArbitraryDownmixData( spatialDec *self, const SPATIAL_SPECIFIC_CONFIG *pSSC, const UINT syntaxFlags, const SPATIAL_BS_FRAME *frame, HANDLE_FDK_BITSTREAM bitstream) { SACDEC_ERROR err = MPS_OK; int ch; int offset = pSSC->nOttBoxes; /* CLD (arbitrary down-mix gains) */ for (ch = 0; ch < pSSC->nInputChannels; ch++) { err = ecDataDec(frame, syntaxFlags, bitstream, &frame->CLDLosslessData[offset + ch], frame->cmpArbdmxGainIdx, self->cmpArbdmxGainIdxPrev, t_CLD, ch, 0, pSSC->freqRes, arbdmxGainDefault); if (err != MPS_OK) return err; } return err; } /* parseArbitraryDownmixData */ /******************************************************************************* Functionname: SpatialDecParseFrame ******************************************************************************* Description: Arguments: Input: Output: *******************************************************************************/ static int nBitsParamSlot(int i) { int bitsParamSlot; bitsParamSlot = fMax(0, DFRACT_BITS - 1 - fNormz((FIXP_DBL)i)); if ((1 << bitsParamSlot) < i) { bitsParamSlot++; } FDK_ASSERT((bitsParamSlot >= 0) && (bitsParamSlot <= 32)); return bitsParamSlot; } SACDEC_ERROR SpatialDecParseFrameData( spatialDec_struct *self, SPATIAL_BS_FRAME *frame, HANDLE_FDK_BITSTREAM bitstream, const SPATIAL_SPECIFIC_CONFIG *pSpatialSpecificConfig, UPMIXTYPE upmixType, int fGlobalIndependencyFlag) { SACDEC_ERROR err = MPS_OK; int bsFramingType, dataBands, ps, pg, i; int pb; int numTempShapeChan = 0; int bsNumOutputChannels = treePropertyTable[pSpatialSpecificConfig->treeConfig] .numOutputChannels; /* CAUTION: Maybe different to pSpatialSpecificConfig->treeConfig in some modes! */ int paramSetErr = 0; UINT alignAnchor = FDKgetValidBits( bitstream); /* Anchor for ByteAlign() function. See comment below. */ UINT syntaxFlags; syntaxFlags = pSpatialSpecificConfig->syntaxFlags; if ((syntaxFlags & (SACDEC_SYNTAX_USAC | SACDEC_SYNTAX_RSVD50)) && pSpatialSpecificConfig->bsHighRateMode == 0) { bsFramingType = 0; /* fixed framing */ frame->numParameterSets = 1; } else { bsFramingType = FDKreadBits(bitstream, 1); if (syntaxFlags & SACDEC_SYNTAX_LD) frame->numParameterSets = FDKreadBits(bitstream, 1) + 1; else frame->numParameterSets = FDKreadBits(bitstream, 3) + 1; } /* Any error after this line shall trigger parameter invalidation at bail * label. */ paramSetErr = 1; if (frame->numParameterSets >= MAX_PARAMETER_SETS) { goto bail; } /* Basic config check. */ if (pSpatialSpecificConfig->nInputChannels <= 0 || pSpatialSpecificConfig->nOutputChannels <= 0) { err = MPS_UNSUPPORTED_CONFIG; goto bail; } if (bsFramingType) { int prevParamSlot = -1; int bitsParamSlot; { bitsParamSlot = nBitsParamSlot(pSpatialSpecificConfig->nTimeSlots); for (i = 0; i < frame->numParameterSets; i++) { frame->paramSlot[i] = FDKreadBits(bitstream, bitsParamSlot); /* Sanity check */ if ((frame->paramSlot[i] <= prevParamSlot) || (frame->paramSlot[i] >= pSpatialSpecificConfig->nTimeSlots)) { err = MPS_PARSE_ERROR; goto bail; } prevParamSlot = frame->paramSlot[i]; } } } else { for (i = 0; i < frame->numParameterSets; i++) { frame->paramSlot[i] = ((pSpatialSpecificConfig->nTimeSlots * (i + 1)) / frame->numParameterSets) - 1; } } if ((syntaxFlags & (SACDEC_SYNTAX_USAC | SACDEC_SYNTAX_RSVD50)) && fGlobalIndependencyFlag) { frame->bsIndependencyFlag = 1; } else { frame->bsIndependencyFlag = (UCHAR)FDKreadBits(bitstream, 1); } /* * OttData() */ for (i = 0; i < pSpatialSpecificConfig->nOttBoxes; i++) { err = ecDataDec(frame, syntaxFlags, bitstream, &frame->CLDLosslessData[i], frame->cmpOttCLDidx, self->cmpOttCLDidxPrev, t_CLD, i, 0, pSpatialSpecificConfig->bitstreamOttBands[i], pSpatialSpecificConfig->ottCLDdefault[i]); if (err != MPS_OK) { goto bail; } } /* i < numOttBoxes */ { for (i = 0; i < pSpatialSpecificConfig->nOttBoxes; i++) { err = ecDataDec(frame, syntaxFlags, bitstream, &frame->ICCLosslessData[i], frame->cmpOttICCidx, self->cmpOttICCidxPrev, t_ICC, i, 0, pSpatialSpecificConfig->bitstreamOttBands[i], ICCdefault); if (err != MPS_OK) { goto bail; } } /* i < numOttBoxes */ } /* !oneICC */ if ((pSpatialSpecificConfig->treeConfig == SPATIALDEC_MODE_RSVD7) && (pSpatialSpecificConfig->bsPhaseCoding)) { frame->phaseMode = FDKreadBits(bitstream, 1); if (frame->phaseMode == 0) { for (pb = 0; pb < pSpatialSpecificConfig->numOttBandsIPD; pb++) { self->cmpOttIPDidxPrev[0][pb] = 0; for (i = 0; i < frame->numParameterSets; i++) { frame->cmpOttIPDidx[0][i][pb] = 0; // frame->ottIPDidx[0][i][pb] = 0; } /* self->ottIPDidxPrev[0][pb] = 0; */ } frame->OpdSmoothingMode = 0; } else { frame->OpdSmoothingMode = FDKreadBits(bitstream, 1); err = ecDataDec(frame, syntaxFlags, bitstream, &frame->IPDLosslessData[0], frame->cmpOttIPDidx, self->cmpOttIPDidxPrev, t_IPD, 0, 0, pSpatialSpecificConfig->numOttBandsIPD, IPDdefault); if (err != MPS_OK) { goto bail; } } } /* * SmgData() */ { if (!pSpatialSpecificConfig->bsHighRateMode && (syntaxFlags & SACDEC_SYNTAX_USAC)) { for (ps = 0; ps < frame->numParameterSets; ps++) { frame->bsSmoothMode[ps] = 0; } } else { for (ps = 0; ps < frame->numParameterSets; ps++) { frame->bsSmoothMode[ps] = (UCHAR)FDKreadBits(bitstream, 2); if (frame->bsSmoothMode[ps] >= 2) { frame->bsSmoothTime[ps] = (UCHAR)FDKreadBits(bitstream, 2); } if (frame->bsSmoothMode[ps] == 3) { frame->bsFreqResStrideSmg[ps] = (UCHAR)FDKreadBits(bitstream, 2); dataBands = (pSpatialSpecificConfig->freqRes - 1) / pbStrideTable[frame->bsFreqResStrideSmg[ps]] + 1; for (pg = 0; pg < dataBands; pg++) { frame->bsSmgData[ps][pg] = (UCHAR)FDKreadBits(bitstream, 1); } } } /* ps < numParameterSets */ } } /* * TempShapeData() */ if ((pSpatialSpecificConfig->tempShapeConfig == 3) && (syntaxFlags & SACDEC_SYNTAX_USAC)) { int TsdErr; TsdErr = TsdRead(bitstream, pSpatialSpecificConfig->nTimeSlots, &frame->TsdData[0]); if (TsdErr) { err = MPS_PARSE_ERROR; goto bail; } } else { frame->TsdData[0].bsTsdEnable = 0; } for (i = 0; i < bsNumOutputChannels; i++) { frame->tempShapeEnableChannelSTP[i] = 0; frame->tempShapeEnableChannelGES[i] = 0; } if ((pSpatialSpecificConfig->tempShapeConfig == 1) || (pSpatialSpecificConfig->tempShapeConfig == 2)) { int bsTempShapeEnable = FDKreadBits(bitstream, 1); if (bsTempShapeEnable) { numTempShapeChan = tempShapeChanTable[pSpatialSpecificConfig->tempShapeConfig - 1] [pSpatialSpecificConfig->treeConfig]; switch (pSpatialSpecificConfig->tempShapeConfig) { case 1: /* STP */ for (i = 0; i < numTempShapeChan; i++) { int stpEnable = FDKreadBits(bitstream, 1); frame->tempShapeEnableChannelSTP[i] = stpEnable; } break; case 2: /* GES */ { UCHAR gesChannelEnable[MAX_OUTPUT_CHANNELS]; for (i = 0; i < numTempShapeChan; i++) { gesChannelEnable[i] = (UCHAR)FDKreadBits(bitstream, 1); frame->tempShapeEnableChannelGES[i] = gesChannelEnable[i]; } for (i = 0; i < numTempShapeChan; i++) { if (gesChannelEnable[i]) { int envShapeData_tmp[MAX_TIME_SLOTS]; if (huff_dec_reshape(bitstream, envShapeData_tmp, pSpatialSpecificConfig->nTimeSlots) != 0) { err = MPS_PARSE_ERROR; goto bail; } for (int ts = 0; ts < pSpatialSpecificConfig->nTimeSlots; ts++) { if (!(envShapeData_tmp[ts] >= 0) && (envShapeData_tmp[ts] <= 4)) { err = MPS_PARSE_ERROR; goto bail; } frame->bsEnvShapeData[i][ts] = (UCHAR)envShapeData_tmp[ts]; } } } } break; default: err = MPS_INVALID_TEMPSHAPE; goto bail; } } /* bsTempShapeEnable */ } /* pSpatialSpecificConfig->tempShapeConfig != 0 */ if (pSpatialSpecificConfig->bArbitraryDownmix != 0) { err = parseArbitraryDownmixData(self, pSpatialSpecificConfig, syntaxFlags, frame, bitstream); if (err != MPS_OK) goto bail; } if (1 && (!(syntaxFlags & (SACDEC_SYNTAX_USAC)))) { FDKbyteAlign(bitstream, alignAnchor); /* ISO/IEC FDIS 23003-1: 5.2. ... byte alignment with respect to the beginning of the syntactic element in which ByteAlign() occurs. */ } bail: if (err != MPS_OK && paramSetErr != 0) { /* Since the parameter set data has already been written to the instance we * need to ... */ frame->numParameterSets = 0; /* ... signal that it is corrupt ... */ } return err; } /* SpatialDecParseFrame */ /******************************************************************************* Functionname: createMapping ******************************************************************************* Description: Arguments: Return: *******************************************************************************/ static void createMapping(int aMap[MAX_PARAMETER_BANDS + 1], int startBand, int stopBand, int stride) { int inBands, outBands, bandsAchived, bandsDiff, incr, k, i; int vDk[MAX_PARAMETER_BANDS + 1]; inBands = stopBand - startBand; outBands = (inBands - 1) / stride + 1; if (outBands < 1) { outBands = 1; } bandsAchived = outBands * stride; bandsDiff = inBands - bandsAchived; for (i = 0; i < outBands; i++) { vDk[i] = stride; } if (bandsDiff > 0) { incr = -1; k = outBands - 1; } else { incr = 1; k = 0; } while (bandsDiff != 0) { vDk[k] = vDk[k] - incr; k = k + incr; bandsDiff = bandsDiff + incr; if (k >= outBands) { if (bandsDiff > 0) { k = outBands - 1; } else if (bandsDiff < 0) { k = 0; } } } aMap[0] = startBand; for (i = 0; i < outBands; i++) { aMap[i + 1] = aMap[i] + vDk[i]; } } /* createMapping */ /******************************************************************************* Functionname: mapFrequency ******************************************************************************* Description: Arguments: Return: *******************************************************************************/ static void mapFrequency(const SCHAR *pInput, /* Input */ SCHAR *pOutput, /* Output */ int *pMap, /* Mapping function */ int dataBands) /* Number of data Bands */ { int i, j; int startBand0 = pMap[0]; for (i = 0; i < dataBands; i++) { int startBand, stopBand, value; value = pInput[i + startBand0]; startBand = pMap[i]; stopBand = pMap[i + 1]; for (j = startBand; j < stopBand; j++) { pOutput[j] = value; } } } /******************************************************************************* Functionname: deq ******************************************************************************* Description: Arguments: Return: *******************************************************************************/ static int deqIdx(int value, int paramType) { int idx = -1; switch (paramType) { case t_CLD: if (((value + 15) >= 0) && ((value + 15) < 31)) { idx = (value + 15); } break; case t_ICC: if ((value >= 0) && (value < 8)) { idx = value; } break; case t_IPD: /* (+/-)15 * MAX_PARAMETER_BANDS for differential coding in frequency * domain (according to rbl) */ if ((value >= -420) && (value <= 420)) { idx = (value & 0xf); } break; default: FDK_ASSERT(0); } return idx; } /******************************************************************************* Functionname: factorFunct ******************************************************************************* Description: Arguments: Return: *******************************************************************************/ #define SF_IDX (7) #define SF_FACTOR (3) #define SCALE_FACTOR (1 << SF_FACTOR) #define SCALE_CLD_C1C2 (1 << SF_CLD_C1C2) static FIXP_DBL factorFunct(FIXP_DBL ottVsTotDb, INT quantMode) { FIXP_DBL factor; if (ottVsTotDb > FL2FXCONST_DBL(0.0)) { ottVsTotDb = FL2FXCONST_DBL(0.0); } ottVsTotDb = -ottVsTotDb; switch (quantMode) { case 0: factor = FL2FXCONST_DBL(1.0f / SCALE_FACTOR); break; case 1: if (ottVsTotDb >= FL2FXCONST_DBL(21.0f / SCALE_CLD_C1C2)) factor = FL2FXCONST_DBL(5.0f / SCALE_FACTOR); else if (ottVsTotDb <= FL2FXCONST_DBL(1.0f / SCALE_CLD_C1C2)) factor = FL2FXCONST_DBL(1.0f / SCALE_FACTOR); else factor = (fMult(FL2FXCONST_DBL(0.2f), ottVsTotDb) + FL2FXCONST_DBL(0.8f / SCALE_CLD_C1C2)) << (SF_CLD_C1C2 - SF_FACTOR); break; case 2: if (ottVsTotDb >= FL2FXCONST_DBL(25.0f / SCALE_CLD_C1C2)) { FDK_ASSERT(SF_FACTOR == 3); factor = (FIXP_DBL) MAXVAL_DBL; /* avoid warning: FL2FXCONST_DBL(8.0f/SCALE_FACTOR) */ } else if (ottVsTotDb <= FL2FXCONST_DBL(1.0f / SCALE_CLD_C1C2)) factor = FL2FXCONST_DBL(1.0f / SCALE_FACTOR); else factor = (fMult(FL2FXCONST_DBL(7.0f / 24.0f), ottVsTotDb) + FL2FXCONST_DBL((17.0f / 24.0f) / SCALE_CLD_C1C2)) << (SF_CLD_C1C2 - SF_FACTOR); break; default: factor = FL2FXCONST_DBL(0.0f); } return (factor); } /******************************************************************************* Functionname: factorCLD ******************************************************************************* Description: Arguments: Return: *******************************************************************************/ static void factorCLD(SCHAR *idx, FIXP_DBL ottVsTotDb, FIXP_DBL *ottVsTotDb1, FIXP_DBL *ottVsTotDb2, SCHAR ottVsTotDbMode, INT quantMode) { FIXP_DBL factor; FIXP_DBL cldIdxFract; INT cldIdx; factor = factorFunct(ottVsTotDb, quantMode); cldIdxFract = fMult((FIXP_DBL)((*idx) << ((DFRACT_BITS - 1) - SF_IDX)), factor); cldIdxFract += FL2FXCONST_DBL(15.5f / (1 << (SF_FACTOR + SF_IDX))); cldIdx = fixp_truncateToInt(cldIdxFract, SF_FACTOR + SF_IDX); cldIdx = fMin(cldIdx, 30); cldIdx = fMax(cldIdx, 0); *idx = cldIdx - 15; if (ottVsTotDbMode & ottVsTotDb1Activ) (*ottVsTotDb1) = ottVsTotDb + dequantCLD_c1[cldIdx]; if (ottVsTotDbMode & ottVsTotDb2Activ) (*ottVsTotDb2) = ottVsTotDb + dequantCLD_c1[30 - cldIdx]; } /******************************************************************************* Functionname: mapIndexData ******************************************************************************* Description: Arguments: Return: *******************************************************************************/ static SACDEC_ERROR mapIndexData( LOSSLESSDATA *llData, SCHAR ***outputDataIdx, SCHAR ***outputIdxData, const SCHAR (*cmpIdxData)[MAX_PARAMETER_SETS][MAX_PARAMETER_BANDS], SCHAR ***diffIdxData, SCHAR xttIdx, SCHAR **idxPrev, int paramIdx, int paramType, int startBand, int stopBand, SCHAR defaultValue, int numParameterSets, const int *paramSlot, int extendFrame, int quantMode, SpatialDecConcealmentInfo *concealmentInfo, SCHAR ottVsTotDbMode, FIXP_DBL (*pOttVsTotDbIn)[MAX_PARAMETER_SETS][MAX_PARAMETER_BANDS], FIXP_DBL (*pOttVsTotDb1)[MAX_PARAMETER_SETS][MAX_PARAMETER_BANDS], FIXP_DBL (*pOttVsTotDb2)[MAX_PARAMETER_SETS][MAX_PARAMETER_BANDS]) { int aParamSlots[MAX_PARAMETER_SETS]; int aInterpolate[MAX_PARAMETER_SETS] = {0}; int dataSets; int aMap[MAX_PARAMETER_BANDS + 1]; int setIdx, i, band, parmSlot; int dataBands; int ps, pb; int i1; if (numParameterSets > MAX_PARAMETER_SETS) return MPS_WRONG_PARAMETERSETS; dataSets = 0; for (i = 0; i < numParameterSets; i++) { if (llData->bsXXXDataMode[i] == 3) { aParamSlots[dataSets] = i; dataSets++; } } setIdx = 0; /* Main concealment stage is here: */ SpatialDecConcealment_Apply( concealmentInfo, cmpIdxData[xttIdx], (diffIdxData != NULL) ? diffIdxData[xttIdx] : NULL, idxPrev[xttIdx], llData->bsXXXDataMode, startBand, stopBand, defaultValue, paramType, numParameterSets); /* Prepare data */ for (i = 0; i < numParameterSets; i++) { if (llData->bsXXXDataMode[i] == 0) { llData->nocmpQuantCoarseXXX[i] = 0; for (band = startBand; band < stopBand; band++) { outputIdxData[xttIdx][i][band] = defaultValue; } for (band = startBand; band < stopBand; band++) { idxPrev[xttIdx][band] = outputIdxData[xttIdx][i][band]; } /* Because the idxPrev are also set to the defaultValue -> signalize fine */ llData->state->bsQuantCoarseXXXprev = 0; } if (llData->bsXXXDataMode[i] == 1) { for (band = startBand; band < stopBand; band++) { outputIdxData[xttIdx][i][band] = idxPrev[xttIdx][band]; } llData->nocmpQuantCoarseXXX[i] = llData->state->bsQuantCoarseXXXprev; } if (llData->bsXXXDataMode[i] == 2) { for (band = startBand; band < stopBand; band++) { outputIdxData[xttIdx][i][band] = idxPrev[xttIdx][band]; } llData->nocmpQuantCoarseXXX[i] = llData->state->bsQuantCoarseXXXprev; aInterpolate[i] = 1; } else { aInterpolate[i] = 0; } if (llData->bsXXXDataMode[i] == 3) { int stride; parmSlot = aParamSlots[setIdx]; stride = pbStrideTable[llData->bsFreqResStrideXXX[setIdx]]; dataBands = (stopBand - startBand - 1) / stride + 1; createMapping(aMap, startBand, stopBand, stride); mapFrequency(&cmpIdxData[xttIdx][setIdx][0], &outputIdxData[xttIdx][parmSlot][0], aMap, dataBands); for (band = startBand; band < stopBand; band++) { idxPrev[xttIdx][band] = outputIdxData[xttIdx][parmSlot][band]; } llData->state->bsQuantCoarseXXXprev = llData->bsQuantCoarseXXX[setIdx]; llData->nocmpQuantCoarseXXX[i] = llData->bsQuantCoarseXXX[setIdx]; setIdx++; } if (diffIdxData != NULL) { for (band = startBand; band < stopBand; band++) { outputIdxData[xttIdx][i][band] += diffIdxData[xttIdx][i][band]; } } } /* for( i = 0 ; i < numParameterSets; i++ ) */ /* Map all coarse data to fine */ for (i = 0; i < numParameterSets; i++) { if (llData->nocmpQuantCoarseXXX[i] == 1) { coarse2fine(outputIdxData[xttIdx][i], (DATA_TYPE)paramType, startBand, stopBand - startBand); llData->nocmpQuantCoarseXXX[i] = 0; } } /* Interpolate */ i1 = 0; for (i = 0; i < numParameterSets; i++) { if (aInterpolate[i] != 1) { i1 = i; } else { int xi, i2, x1, x2; for (i2 = i; i2 < numParameterSets; i2++) { if (aInterpolate[i2] != 1) break; } if (i2 >= numParameterSets) return MPS_WRONG_PARAMETERSETS; x1 = paramSlot[i1]; xi = paramSlot[i]; x2 = paramSlot[i2]; for (band = startBand; band < stopBand; band++) { int yi, y1, y2; y1 = outputIdxData[xttIdx][i1][band]; y2 = outputIdxData[xttIdx][i2][band]; if (x1 != x2) { yi = y1 + (xi - x1) * (y2 - y1) / (x2 - x1); } else { yi = y1 /*+ (xi-x1)*(y2-y1)/1e-12*/; } outputIdxData[xttIdx][i][band] = yi; } } } /* for( i = 0 ; i < numParameterSets; i++ ) */ /* Dequantize data and apply factorCLD if necessary */ for (ps = 0; ps < numParameterSets; ps++) { if (quantMode && (paramType == t_CLD)) { if (pOttVsTotDbIn == 0) return MPS_WRONG_OTT; if ((pOttVsTotDb1 == 0) && (ottVsTotDbMode & ottVsTotDb1Activ)) return MPS_WRONG_OTT; if ((pOttVsTotDb2 == 0) && (ottVsTotDbMode & ottVsTotDb2Activ)) return MPS_WRONG_OTT; for (pb = startBand; pb < stopBand; pb++) { factorCLD(&(outputIdxData[xttIdx][ps][pb]), (*pOttVsTotDbIn)[ps][pb], (pOttVsTotDb1 != NULL) ? &((*pOttVsTotDb1)[ps][pb]) : NULL, (pOttVsTotDb2 != NULL) ? &((*pOttVsTotDb2)[ps][pb]) : NULL, ottVsTotDbMode, quantMode); } } /* Dequantize data */ for (band = startBand; band < stopBand; band++) { outputDataIdx[xttIdx][ps][band] = deqIdx(outputIdxData[xttIdx][ps][band], paramType); if (outputDataIdx[xttIdx][ps][band] == -1) { outputDataIdx[xttIdx][ps][band] = defaultValue; } } } /* for( i = 0 ; i < numParameterSets; i++ ) */ if (extendFrame) { if (paramType == t_IPD) { llData->bsQuantCoarseXXX[numParameterSets] = llData->bsQuantCoarseXXX[numParameterSets - 1]; } for (band = startBand; band < stopBand; band++) { outputDataIdx[xttIdx][numParameterSets][band] = outputDataIdx[xttIdx][numParameterSets - 1][band]; } } return MPS_OK; } /******************************************************************************* Functionname: decodeAndMapFrameOtt ******************************************************************************* Description: Do delta decoding and dequantization Arguments: Input: Output: *******************************************************************************/ static SACDEC_ERROR decodeAndMapFrameOtt(HANDLE_SPATIAL_DEC self, SPATIAL_BS_FRAME *pCurBs) { int i, ottIdx; int numOttBoxes; SACDEC_ERROR err = MPS_OK; numOttBoxes = self->numOttBoxes; switch (self->treeConfig) { default: { if (self->quantMode != 0) { goto bail; } } for (i = 0; i < numOttBoxes; i++) { err = mapIndexData( &pCurBs->CLDLosslessData[i], /* LOSSLESSDATA *llData,*/ self->ottCLD__FDK, self->outIdxData, pCurBs ->cmpOttCLDidx, /* int cmpIdxData[MAX_NUM_OTT][MAX_PARAMETER_SETS][MAX_PARAMETER_BANDS], */ NULL, /* no differential data */ i, /* int xttIdx, Which ott/ttt index to use for input and output buffers */ self->ottCLDidxPrev, /* int idxPrev[MAX_NUM_OTT][MAX_PARAMETER_BANDS], */ i, t_CLD, 0, /* int startBand, */ self->pConfigCurrent->bitstreamOttBands[i], /* int stopBand, */ self->pConfigCurrent->ottCLDdefault[i], /* int defaultValue, */ pCurBs->numParameterSets, /* int numParameterSets) */ pCurBs->paramSlot, self->extendFrame, self->quantMode, &(self->concealInfo), ottVsTotInactiv, NULL, NULL, NULL); if (err != MPS_OK) goto bail; } /* for(i = 0; i < numOttBoxes ; i++ ) */ break; } /* case */ for (ottIdx = 0; ottIdx < numOttBoxes; ottIdx++) { /* Read ICC */ err = mapIndexData( &pCurBs->ICCLosslessData[ottIdx], /* LOSSLESSDATA *llData,*/ self->ottICC__FDK, self->outIdxData, pCurBs ->cmpOttICCidx, /* int cmpIdxData[MAX_NUM_OTT][MAX_PARAMETER_SETS][MAX_PARAMETER_BANDS], */ self->ottICCdiffidx, /* differential data */ ottIdx, /* int xttIdx, Which ott/ttt index to use for input and output buffers */ self->ottICCidxPrev, /* int idxPrev[MAX_NUM_OTT][MAX_PARAMETER_BANDS], */ ottIdx, t_ICC, 0, /* int startBand, */ self->pConfigCurrent->bitstreamOttBands[ottIdx], /* int stopBand, */ ICCdefault, /* int defaultValue, */ pCurBs->numParameterSets, /* int numParameterSets) */ pCurBs->paramSlot, self->extendFrame, self->quantMode, &(self->concealInfo), ottVsTotInactiv, NULL, NULL, NULL); if (err != MPS_OK) goto bail; } /* ottIdx */ if ((self->treeConfig == TREE_212) && (self->phaseCoding)) { if (pCurBs->phaseMode == 0) { for (int pb = 0; pb < self->pConfigCurrent->numOttBandsIPD; pb++) { self->ottIPDidxPrev[0][pb] = 0; } } for (ottIdx = 0; ottIdx < numOttBoxes; ottIdx++) { err = mapIndexData( &pCurBs->IPDLosslessData[ottIdx], self->ottIPD__FDK, self->outIdxData, pCurBs->cmpOttIPDidx, NULL, ottIdx, self->ottIPDidxPrev, ottIdx, t_IPD, 0, self->numOttBandsIPD, IPDdefault, pCurBs->numParameterSets, pCurBs->paramSlot, self->extendFrame, self->quantMode, &(self->concealInfo), ottVsTotInactiv, NULL, NULL, NULL); } } bail: return MPS_OK; } /* decodeAndMapFrameOtt */ /******************************************************************************* Functionname: decodeAndMapFrameSmg ******************************************************************************* Description: Decode smoothing flags Arguments: Input: Output: *******************************************************************************/ static SACDEC_ERROR decodeAndMapFrameSmg(HANDLE_SPATIAL_DEC self, const SPATIAL_BS_FRAME *frame) { int ps, pb, pg, pbStride, dataBands, pbStart, pbStop, aGroupToBand[MAX_PARAMETER_BANDS + 1]; if (frame->numParameterSets > MAX_PARAMETER_SETS) return MPS_WRONG_PARAMETERSETS; if (self->bitstreamParameterBands > MAX_PARAMETER_BANDS) return MPS_WRONG_PARAMETERBANDS; for (ps = 0; ps < frame->numParameterSets; ps++) { switch (frame->bsSmoothMode[ps]) { case 0: self->smgTime[ps] = 256; FDKmemclear(self->smgData[ps], self->bitstreamParameterBands * sizeof(UCHAR)); break; case 1: if (ps > 0) { self->smgTime[ps] = self->smgTime[ps - 1]; FDKmemcpy(self->smgData[ps], self->smgData[ps - 1], self->bitstreamParameterBands * sizeof(UCHAR)); } else { self->smgTime[ps] = self->smoothState->prevSmgTime; FDKmemcpy(self->smgData[ps], self->smoothState->prevSmgData, self->bitstreamParameterBands * sizeof(UCHAR)); } break; case 2: self->smgTime[ps] = smgTimeTable[frame->bsSmoothTime[ps]]; for (pb = 0; pb < self->bitstreamParameterBands; pb++) { self->smgData[ps][pb] = 1; } break; case 3: self->smgTime[ps] = smgTimeTable[frame->bsSmoothTime[ps]]; pbStride = pbStrideTable[frame->bsFreqResStrideSmg[ps]]; dataBands = (self->bitstreamParameterBands - 1) / pbStride + 1; createMapping(aGroupToBand, 0, self->bitstreamParameterBands, pbStride); for (pg = 0; pg < dataBands; pg++) { pbStart = aGroupToBand[pg]; pbStop = aGroupToBand[pg + 1]; for (pb = pbStart; pb < pbStop; pb++) { self->smgData[ps][pb] = frame->bsSmgData[ps][pg]; } } break; } } self->smoothState->prevSmgTime = self->smgTime[frame->numParameterSets - 1]; FDKmemcpy(self->smoothState->prevSmgData, self->smgData[frame->numParameterSets - 1], self->bitstreamParameterBands * sizeof(UCHAR)); if (self->extendFrame) { self->smgTime[frame->numParameterSets] = self->smgTime[frame->numParameterSets - 1]; FDKmemcpy(self->smgData[frame->numParameterSets], self->smgData[frame->numParameterSets - 1], self->bitstreamParameterBands * sizeof(UCHAR)); } return MPS_OK; } /******************************************************************************* Functionname: decodeAndMapFrameArbdmx ******************************************************************************* Description: Do delta decoding and dequantization Arguments: Input: Output: *******************************************************************************/ static SACDEC_ERROR decodeAndMapFrameArbdmx(HANDLE_SPATIAL_DEC self, const SPATIAL_BS_FRAME *frame) { SACDEC_ERROR err = MPS_OK; int ch; int offset = self->numOttBoxes; for (ch = 0; ch < self->numInputChannels; ch++) { err = mapIndexData(&frame->CLDLosslessData[offset + ch], self->arbdmxGain__FDK, self->outIdxData, frame->cmpArbdmxGainIdx, NULL, /* no differential data */ ch, self->arbdmxGainIdxPrev, offset + ch, t_CLD, 0, self->bitstreamParameterBands, 0 /*self->arbdmxGainDefault*/, frame->numParameterSets, frame->paramSlot, self->extendFrame, 0, &(self->concealInfo), ottVsTotInactiv, NULL, NULL, NULL); if (err != MPS_OK) goto bail; } bail: return err; } /* decodeAndMapFrameArbdmx */ /******************************************************************************* Functionname: SpatialDecDecodeFrame ******************************************************************************* Description: Arguments: Return: *******************************************************************************/ SACDEC_ERROR SpatialDecDecodeFrame(spatialDec *self, SPATIAL_BS_FRAME *frame) { SACDEC_ERROR err = MPS_OK; self->extendFrame = 0; if (frame->paramSlot[frame->numParameterSets - 1] != self->timeSlots - 1) { self->extendFrame = 1; } self->TsdTs = 0; /****** DTDF and MAP DATA ********/ if ((err = decodeAndMapFrameOtt(self, frame)) != MPS_OK) goto bail; if ((err = decodeAndMapFrameSmg(self, frame)) != MPS_OK) goto bail; if (self->arbitraryDownmix != 0) { if ((err = decodeAndMapFrameArbdmx(self, frame)) != MPS_OK) goto bail; } if (self->extendFrame) { frame->numParameterSets = fixMin(MAX_PARAMETER_SETS, frame->numParameterSets + 1); frame->paramSlot[frame->numParameterSets - 1] = self->timeSlots - 1; for (int p = 0; p < frame->numParameterSets; p++) { if (frame->paramSlot[p] > self->timeSlots - 1) { frame->paramSlot[p] = self->timeSlots - 1; err = MPS_PARSE_ERROR; } } if (err != MPS_OK) { goto bail; } } bail: return err; } /* SpatialDecDecodeFrame() */ /******************************************************************************* Functionname: SpatialDecodeHeader ******************************************************************************* Description: Arguments: Return: *******************************************************************************/ SACDEC_ERROR SpatialDecDecodeHeader( spatialDec *self, SPATIAL_SPECIFIC_CONFIG *pSpatialSpecificConfig) { SACDEC_ERROR err = MPS_OK; int i; self->samplingFreq = pSpatialSpecificConfig->samplingFreq; self->timeSlots = pSpatialSpecificConfig->nTimeSlots; self->frameLength = self->timeSlots * self->qmfBands; self->bitstreamParameterBands = pSpatialSpecificConfig->freqRes; if (self->pConfigCurrent->syntaxFlags & SACDEC_SYNTAX_LD) self->hybridBands = self->qmfBands; else self->hybridBands = SacGetHybridSubbands(self->qmfBands); self->tp_hybBandBorder = 12; self->numParameterBands = self->bitstreamParameterBands; if (self->pConfigCurrent->syntaxFlags & SACDEC_SYNTAX_LD) { switch (self->numParameterBands) { case 4: self->kernels = kernels_4_to_64; break; case 5: self->kernels = kernels_5_to_64; break; case 7: self->kernels = kernels_7_to_64; break; case 9: self->kernels = kernels_9_to_64; break; case 12: self->kernels = kernels_12_to_64; break; case 15: self->kernels = kernels_15_to_64; break; case 23: self->kernels = kernels_23_to_64; break; default: return MPS_INVALID_PARAMETERBANDS; /* unsupported numParameterBands */ } } else { switch (self->numParameterBands) { case 4: self->kernels = kernels_4_to_71; break; case 5: self->kernels = kernels_5_to_71; break; case 7: self->kernels = kernels_7_to_71; break; case 10: self->kernels = kernels_10_to_71; break; case 14: self->kernels = kernels_14_to_71; break; case 20: self->kernels = kernels_20_to_71; break; case 28: self->kernels = kernels_28_to_71; break; default: return MPS_INVALID_PARAMETERBANDS; /* unsupported numParameterBands */ } } /* create param to hyb band table */ FDKmemclear(self->param2hyb, (MAX_PARAMETER_BANDS + 1) * sizeof(int)); for (i = 0; i < self->hybridBands; i++) { self->param2hyb[self->kernels[i] + 1] = i + 1; } { int pb = self->kernels[i - 1] + 2; for (; pb < (MAX_PARAMETER_BANDS + 1); pb++) { self->param2hyb[pb] = i; } for (pb = 0; pb < MAX_PARAMETER_BANDS; pb += 1) { self->kernels_width[pb] = self->param2hyb[pb + 1] - self->param2hyb[pb]; } } self->treeConfig = pSpatialSpecificConfig->treeConfig; self->numOttBoxes = pSpatialSpecificConfig->nOttBoxes; self->numInputChannels = pSpatialSpecificConfig->nInputChannels; self->numOutputChannels = pSpatialSpecificConfig->nOutputChannels; self->quantMode = pSpatialSpecificConfig->quantMode; self->arbitraryDownmix = pSpatialSpecificConfig->bArbitraryDownmix; self->numM2rows = self->numOutputChannels; { self->residualCoding = 0; if (self->arbitraryDownmix == 2) self->arbitraryDownmix = 1; /* no arbitrary downmix residuals */ } if ((self->pConfigCurrent->syntaxFlags & SACDEC_SYNTAX_USAC)) { self->residualCoding = pSpatialSpecificConfig->bResidualCoding; } self->clipProtectGain__FDK = FX_CFG2FX_DBL(clipGainTable__FDK[pSpatialSpecificConfig->bsFixedGainDMX]); self->clipProtectGainSF__FDK = clipGainSFTable__FDK[pSpatialSpecificConfig->bsFixedGainDMX]; self->tempShapeConfig = pSpatialSpecificConfig->tempShapeConfig; self->decorrConfig = pSpatialSpecificConfig->decorrConfig; if (self->upmixType == UPMIXTYPE_BYPASS) { self->numOutputChannels = self->numInputChannels; } self->numOutputChannelsAT = self->numOutputChannels; self->numOttBandsIPD = pSpatialSpecificConfig->numOttBandsIPD; self->phaseCoding = pSpatialSpecificConfig->bsPhaseCoding; for (i = 0; i < self->numOttBoxes; i++) { { self->pConfigCurrent->bitstreamOttBands[i] = self->bitstreamParameterBands; } self->numOttBands[i] = self->pConfigCurrent->bitstreamOttBands[i]; } /* i */ if (self->residualCoding) { int numBoxes = self->numOttBoxes; for (i = 0; i < numBoxes; i++) { self->residualPresent[i] = pSpatialSpecificConfig->ResidualConfig[i].bResidualPresent; if (self->residualPresent[i]) { self->residualBands[i] = pSpatialSpecificConfig->ResidualConfig[i].nResidualBands; /* conversion from hybrid bands to qmf bands */ self->residualQMFBands[i] = fMax(self->param2hyb[self->residualBands[i]] + 3 - 10, 3); /* simplification for the lowest 10 hybrid bands */ } else { self->residualBands[i] = 0; self->residualQMFBands[i] = 0; } } } /* self->residualCoding */ else { int boxes = self->numOttBoxes; for (i = 0; i < boxes; i += 1) { self->residualPresent[i] = 0; self->residualBands[i] = 0; } } switch (self->treeConfig) { case TREE_212: self->numDirektSignals = 1; self->numDecorSignals = 1; self->numXChannels = 1; if (self->arbitraryDownmix == 2) { self->numXChannels += 1; } self->numVChannels = self->numDirektSignals + self->numDecorSignals; break; default: return MPS_INVALID_TREECONFIG; } self->highRateMode = pSpatialSpecificConfig->bsHighRateMode; self->decorrType = pSpatialSpecificConfig->bsDecorrType; SpatialDecDecodeHelperInfo(pSpatialSpecificConfig, UPMIXTYPE_NORMAL); return err; } /******************************************************************************* Functionname: SpatialDecCreateBsFrame ******************************************************************************* Description: Create spatial bitstream structure Arguments: spatialDec* self const SPATIAL_BS_FRAME **bsFrame Return: - *******************************************************************************/ SACDEC_ERROR SpatialDecCreateBsFrame(SPATIAL_BS_FRAME *bsFrame, BS_LL_STATE *llState) { SPATIAL_BS_FRAME *pBs = bsFrame; const int maxNumOtt = MAX_NUM_OTT; const int maxNumInputChannels = MAX_INPUT_CHANNELS; FDK_ALLOCATE_MEMORY_1D_P( pBs->cmpOttIPDidx, maxNumOtt * MAX_PARAMETER_SETS * MAX_PARAMETER_BANDS, SCHAR, SCHAR(*)[MAX_PARAMETER_SETS][MAX_PARAMETER_BANDS]) /* Arbitrary Downmix */ FDK_ALLOCATE_MEMORY_1D_P( pBs->cmpArbdmxGainIdx, maxNumInputChannels * MAX_PARAMETER_SETS * MAX_PARAMETER_BANDS, SCHAR, SCHAR(*)[MAX_PARAMETER_SETS][MAX_PARAMETER_BANDS]) /* Lossless control */ FDK_ALLOCATE_MEMORY_1D(pBs->CLDLosslessData, MAX_NUM_PARAMETERS, LOSSLESSDATA) FDK_ALLOCATE_MEMORY_1D(pBs->ICCLosslessData, MAX_NUM_PARAMETERS, LOSSLESSDATA) FDK_ALLOCATE_MEMORY_1D(pBs->IPDLosslessData, MAX_NUM_PARAMETERS, LOSSLESSDATA) pBs->newBsData = 0; pBs->numParameterSets = 1; /* Link lossless states */ for (int x = 0; x < MAX_NUM_PARAMETERS; x++) { pBs->CLDLosslessData[x].state = &llState->CLDLosslessState[x]; pBs->ICCLosslessData[x].state = &llState->ICCLosslessState[x]; pBs->IPDLosslessData[x].state = &llState->IPDLosslessState[x]; } return MPS_OK; bail: return MPS_OUTOFMEMORY; } /******************************************************************************* Functionname: SpatialDecCloseBsFrame ******************************************************************************* Description: Close spatial bitstream structure Arguments: spatialDec* self Return: - *******************************************************************************/ void SpatialDecCloseBsFrame(SPATIAL_BS_FRAME *pBs) { if (pBs != NULL) { /* These arrays contain the compact indices, only one value per pbstride, * only paramsets actually containing data. */ FDK_FREE_MEMORY_1D(pBs->cmpOttIPDidx); /* Arbitrary Downmix */ FDK_FREE_MEMORY_1D(pBs->cmpArbdmxGainIdx); /* Lossless control */ FDK_FREE_MEMORY_1D(pBs->IPDLosslessData); FDK_FREE_MEMORY_1D(pBs->CLDLosslessData); FDK_FREE_MEMORY_1D(pBs->ICCLosslessData); } } fdk-aac-2.0.2/libSACdec/src/sac_bitdec.h000066400000000000000000000150271404225310700175750ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround decoder library ************************* Author(s): Description: SAC Dec bitstream decoder *******************************************************************************/ /*! \file \brief Spatial Audio bitstream decoder */ #ifndef SAC_BITDEC_H #define SAC_BITDEC_H #include "sac_dec.h" typedef struct { SCHAR numInputChannels; SCHAR numOutputChannels; SCHAR numOttBoxes; SCHAR numTttBoxes; SCHAR ottModeLfe[MAX_NUM_OTT]; } TREEPROPERTIES; enum { TREE_212 = 7, TREE_DUMMY = 255 }; enum { QUANT_FINE = 0, QUANT_EBQ1 = 1, QUANT_EBQ2 = 2 }; SACDEC_ERROR SpatialDecParseSpecificConfigHeader( HANDLE_FDK_BITSTREAM bitstream, SPATIAL_SPECIFIC_CONFIG *pSpatialSpecificConfig, AUDIO_OBJECT_TYPE coreCodec, SPATIAL_DEC_UPMIX_TYPE upmixType); SACDEC_ERROR SpatialDecParseMps212Config( HANDLE_FDK_BITSTREAM bitstream, SPATIAL_SPECIFIC_CONFIG *pSpatialSpecificConfig, int samplingRate, AUDIO_OBJECT_TYPE coreCodec, INT stereoConfigIndex, INT coreSbrFrameLengthIndex); SACDEC_ERROR SpatialDecParseSpecificConfig( HANDLE_FDK_BITSTREAM bitstream, SPATIAL_SPECIFIC_CONFIG *pSpatialSpecificConfig, int sacHeaderLen, AUDIO_OBJECT_TYPE coreCodec); int SpatialDecDefaultSpecificConfig( SPATIAL_SPECIFIC_CONFIG *pSpatialSpecificConfig, AUDIO_OBJECT_TYPE coreCodec, int samplingFreq, int nTimeSlots, int sacDecoderLevel, int isBlind, int coreChannels); SACDEC_ERROR SpatialDecCreateBsFrame(SPATIAL_BS_FRAME *bsFrame, BS_LL_STATE *llState); void SpatialDecCloseBsFrame(SPATIAL_BS_FRAME *bsFrame); SACDEC_ERROR SpatialDecParseFrameData( spatialDec *self, SPATIAL_BS_FRAME *frame, HANDLE_FDK_BITSTREAM bitstream, const SPATIAL_SPECIFIC_CONFIG *pSpatialSpecificConfig, UPMIXTYPE upmixType, int fGlobalIndependencyFlag); SACDEC_ERROR SpatialDecDecodeFrame(spatialDec *self, SPATIAL_BS_FRAME *frame); SACDEC_ERROR SpatialDecDecodeHeader( spatialDec *self, SPATIAL_SPECIFIC_CONFIG *pSpatialSpecificConfig); #endif fdk-aac-2.0.2/libSACdec/src/sac_calcM1andM2.cpp000066400000000000000000000670601404225310700206640ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround decoder library ************************* Author(s): Description: SAC Dec M1 and M2 calculation *******************************************************************************/ #include "sac_calcM1andM2.h" #include "sac_bitdec.h" #include "sac_process.h" #include "sac_rom.h" #include "sac_smoothing.h" #include "FDK_trigFcts.h" /* assorted definitions and constants */ #define ABS_THR2 1.0e-9 #define SQRT2_FDK \ ((FIXP_DBL)FL2FXCONST_DBL(0.70710678118f)) /* FDKsqrt(2.0) scaled by 0.5 */ static void param2UMX_PS__FDK(spatialDec* self, FIXP_DBL H11[MAX_PARAMETER_BANDS], FIXP_DBL H12[MAX_PARAMETER_BANDS], FIXP_DBL H21[MAX_PARAMETER_BANDS], FIXP_DBL H22[MAX_PARAMETER_BANDS], FIXP_DBL c_l[MAX_PARAMETER_BANDS], FIXP_DBL c_r[MAX_PARAMETER_BANDS], int ottBoxIndx, int parameterSetIndx, int resBands); static void param2UMX_PS_Core__FDK( const SCHAR cld[MAX_PARAMETER_BANDS], const SCHAR icc[MAX_PARAMETER_BANDS], const int numOttBands, const int resBands, FIXP_DBL H11[MAX_PARAMETER_BANDS], FIXP_DBL H12[MAX_PARAMETER_BANDS], FIXP_DBL H21[MAX_PARAMETER_BANDS], FIXP_DBL H22[MAX_PARAMETER_BANDS], FIXP_DBL c_l[MAX_PARAMETER_BANDS], FIXP_DBL c_r[MAX_PARAMETER_BANDS]); static void param2UMX_PS_IPD_OPD__FDK( spatialDec* self, const SPATIAL_BS_FRAME* frame, FIXP_DBL H11re[MAX_PARAMETER_BANDS], FIXP_DBL H12re[MAX_PARAMETER_BANDS], FIXP_DBL H21re[MAX_PARAMETER_BANDS], FIXP_DBL H22re[MAX_PARAMETER_BANDS], FIXP_DBL c_l[MAX_PARAMETER_BANDS], FIXP_DBL c_r[MAX_PARAMETER_BANDS], int ottBoxIndx, int parameterSetIndx, int residualBands); static void param2UMX_Prediction__FDK( spatialDec* self, FIXP_DBL H11re[MAX_PARAMETER_BANDS], FIXP_DBL H11im[MAX_PARAMETER_BANDS], FIXP_DBL H12re[MAX_PARAMETER_BANDS], FIXP_DBL H12im[MAX_PARAMETER_BANDS], FIXP_DBL H21re[MAX_PARAMETER_BANDS], FIXP_DBL H21im[MAX_PARAMETER_BANDS], FIXP_DBL H22re[MAX_PARAMETER_BANDS], FIXP_DBL H22im[MAX_PARAMETER_BANDS], int ottBoxIndx, int parameterSetIndx, int resBands); /* static void SpatialDecCalculateM0(spatialDec* self,int ps); */ static SACDEC_ERROR SpatialDecCalculateM1andM2_212( spatialDec* self, int ps, const SPATIAL_BS_FRAME* frame); /******************************************************************************* Functionname: SpatialDecGetResidualIndex ******************************************************************************* Description: Arguments: Input: Output: *******************************************************************************/ int SpatialDecGetResidualIndex(spatialDec* self, int row) { return row2residual[self->treeConfig][row]; } /******************************************************************************* Functionname: UpdateAlpha ******************************************************************************* Description: Arguments: Input: Output: *******************************************************************************/ static void updateAlpha(spatialDec* self) { int nChIn = self->numInputChannels; int ch; for (ch = 0; ch < nChIn; ch++) { FIXP_DBL alpha = /* FL2FXCONST_DBL(1.0f) */ (FIXP_DBL)MAXVAL_DBL; self->arbdmxAlphaPrev__FDK[ch] = self->arbdmxAlpha__FDK[ch]; self->arbdmxAlpha__FDK[ch] = alpha; } } /******************************************************************************* Functionname: SpatialDecCalculateM1andM2 ******************************************************************************* Description: Arguments: *******************************************************************************/ SACDEC_ERROR SpatialDecCalculateM1andM2(spatialDec* self, int ps, const SPATIAL_BS_FRAME* frame) { SACDEC_ERROR err = MPS_OK; if ((self->arbitraryDownmix != 0) && (ps == 0)) { updateAlpha(self); } self->pActivM2ParamBands = NULL; switch (self->upmixType) { case UPMIXTYPE_BYPASS: case UPMIXTYPE_NORMAL: switch (self->treeConfig) { case TREE_212: err = SpatialDecCalculateM1andM2_212(self, ps, frame); break; default: err = MPS_WRONG_TREECONFIG; }; break; default: err = MPS_WRONG_TREECONFIG; } if (err != MPS_OK) { goto bail; } bail: return err; } /******************************************************************************* Functionname: SpatialDecCalculateM1andM2_212 ******************************************************************************* Description: Arguments: Return: *******************************************************************************/ static SACDEC_ERROR SpatialDecCalculateM1andM2_212( spatialDec* self, int ps, const SPATIAL_BS_FRAME* frame) { SACDEC_ERROR err = MPS_OK; int pb; FIXP_DBL H11re[MAX_PARAMETER_BANDS] = {FL2FXCONST_DBL(0.0f)}; FIXP_DBL H12re[MAX_PARAMETER_BANDS] = {FL2FXCONST_DBL(0.0f)}; FIXP_DBL H21re[MAX_PARAMETER_BANDS] = {FL2FXCONST_DBL(0.0f)}; FIXP_DBL H22re[MAX_PARAMETER_BANDS] = {FL2FXCONST_DBL(0.0f)}; FIXP_DBL H11im[MAX_PARAMETER_BANDS] = {FL2FXCONST_DBL(0.0f)}; FIXP_DBL H21im[MAX_PARAMETER_BANDS] = {FL2FXCONST_DBL(0.0f)}; INT phaseCoding = self->phaseCoding; switch (phaseCoding) { case 1: /* phase coding: yes; residuals: no */ param2UMX_PS_IPD_OPD__FDK(self, frame, H11re, H12re, H21re, H22re, NULL, NULL, 0, ps, self->residualBands[0]); break; case 3: /* phase coding: yes; residuals: yes */ param2UMX_Prediction__FDK(self, H11re, H11im, H12re, NULL, H21re, H21im, H22re, NULL, 0, ps, self->residualBands[0]); break; default: if (self->residualCoding) { /* phase coding: no; residuals: yes */ param2UMX_Prediction__FDK(self, H11re, NULL, H12re, NULL, H21re, NULL, H22re, NULL, 0, ps, self->residualBands[0]); } else { /* phase coding: no; residuals: no */ param2UMX_PS__FDK(self, H11re, H12re, H21re, H22re, NULL, NULL, 0, ps, 0); } break; } for (pb = 0; pb < self->numParameterBands; pb++) { self->M2Real__FDK[0][0][pb] = (H11re[pb]); self->M2Real__FDK[0][1][pb] = (H12re[pb]); self->M2Real__FDK[1][0][pb] = (H21re[pb]); self->M2Real__FDK[1][1][pb] = (H22re[pb]); } if (phaseCoding == 3) { for (pb = 0; pb < self->numParameterBands; pb++) { self->M2Imag__FDK[0][0][pb] = (H11im[pb]); self->M2Imag__FDK[1][0][pb] = (H21im[pb]); self->M2Imag__FDK[0][1][pb] = (FIXP_DBL)0; // H12im[pb]; self->M2Imag__FDK[1][1][pb] = (FIXP_DBL)0; // H22im[pb]; } } if (self->phaseCoding == 1) { SpatialDecSmoothOPD( self, frame, ps); /* INPUT: PhaseLeft, PhaseRight, (opdLeftState, opdRightState) */ } return err; } /******************************************************************************* Functionname: param2UMX_PS_Core ******************************************************************************* Description: Arguments: Return: *******************************************************************************/ static void param2UMX_PS_Core__FDK( const SCHAR cld[MAX_PARAMETER_BANDS], const SCHAR icc[MAX_PARAMETER_BANDS], const int numOttBands, const int resBands, FIXP_DBL H11[MAX_PARAMETER_BANDS], FIXP_DBL H12[MAX_PARAMETER_BANDS], FIXP_DBL H21[MAX_PARAMETER_BANDS], FIXP_DBL H22[MAX_PARAMETER_BANDS], FIXP_DBL c_l[MAX_PARAMETER_BANDS], FIXP_DBL c_r[MAX_PARAMETER_BANDS]) { int band; if ((c_l != NULL) && (c_r != NULL)) { for (band = 0; band < numOttBands; band++) { SpatialDequantGetCLDValues(cld[band], &c_l[band], &c_r[band]); } } band = 0; FDK_ASSERT(resBands == 0); for (; band < numOttBands; band++) { /* compute mixing variables: */ const int idx1 = cld[band]; const int idx2 = icc[band]; H11[band] = FX_CFG2FX_DBL(H11_nc[idx1][idx2]); H21[band] = FX_CFG2FX_DBL(H11_nc[30 - idx1][idx2]); H12[band] = FX_CFG2FX_DBL(H12_nc[idx1][idx2]); H22[band] = FX_CFG2FX_DBL(-H12_nc[30 - idx1][idx2]); } } /******************************************************************************* Functionname: param2UMX_PS ******************************************************************************* Description: Arguments: Return: *******************************************************************************/ static void param2UMX_PS__FDK(spatialDec* self, FIXP_DBL H11[MAX_PARAMETER_BANDS], FIXP_DBL H12[MAX_PARAMETER_BANDS], FIXP_DBL H21[MAX_PARAMETER_BANDS], FIXP_DBL H22[MAX_PARAMETER_BANDS], FIXP_DBL c_l[MAX_PARAMETER_BANDS], FIXP_DBL c_r[MAX_PARAMETER_BANDS], int ottBoxIndx, int parameterSetIndx, int residualBands) { int band; param2UMX_PS_Core__FDK(self->ottCLD__FDK[ottBoxIndx][parameterSetIndx], self->ottICC__FDK[ottBoxIndx][parameterSetIndx], self->numOttBands[ottBoxIndx], residualBands, H11, H12, H21, H22, c_l, c_r); for (band = self->numOttBands[ottBoxIndx]; band < self->numParameterBands; band++) { H11[band] = H21[band] = H12[band] = H22[band] = FL2FXCONST_DBL(0.f); } } #define N_CLD (31) #define N_IPD (16) static const FIXP_DBL sinIpd_tab[N_IPD] = { FIXP_DBL(0x00000000), FIXP_DBL(0x30fbc54e), FIXP_DBL(0x5a827999), FIXP_DBL(0x7641af3d), FIXP_DBL(0x7fffffff), FIXP_DBL(0x7641af3d), FIXP_DBL(0x5a82799a), FIXP_DBL(0x30fbc54d), FIXP_DBL(0xffffffff), FIXP_DBL(0xcf043ab3), FIXP_DBL(0xa57d8666), FIXP_DBL(0x89be50c3), FIXP_DBL(0x80000000), FIXP_DBL(0x89be50c3), FIXP_DBL(0xa57d8666), FIXP_DBL(0xcf043ab2), }; /* cosIpd[i] = sinIpd[(i+4)&15] */ #define SIN_IPD(a) (sinIpd_tab[(a)]) #define COS_IPD(a) (sinIpd_tab[((a) + 4) & 15]) //(cosIpd_tab[(a)]) static const FIXP_SGL sqrt_one_minus_ICC2[8] = { FL2FXCONST_SGL(0.0f), FL2FXCONST_SGL(0.349329357483736f), FL2FXCONST_SGL(0.540755219669676f), FL2FXCONST_SGL(0.799309172723546f), FL2FXCONST_SGL(0.929968187843004f), FX_DBL2FXCONST_SGL(MAXVAL_DBL), FL2FXCONST_SGL(0.80813303360276f), FL2FXCONST_SGL(0.141067359796659f), }; /* exponent of sqrt(CLD) */ static const SCHAR sqrt_CLD_e[N_CLD] = { -24, -7, -6, -5, -4, -4, -3, -3, -2, -2, -1, -1, 0, 0, 0, 1, 1, 1, 1, 2, 2, 3, 3, 4, 4, 5, 5, 6, 7, 8, 25}; static const FIXP_DBL sqrt_CLD_m[N_CLD] = { FL2FXCONST_DBL(0.530542153566195f), FL2FXCONST_DBL(0.719796896243647f), FL2FXCONST_DBL(0.64f), FL2FXCONST_DBL(0.569049411212455f), FL2FXCONST_DBL(0.505964425626941f), FL2FXCONST_DBL(0.899746120304559f), FL2FXCONST_DBL(0.635462587779425f), FL2FXCONST_DBL(0.897614763441571f), FL2FXCONST_DBL(0.633957276984445f), FL2FXCONST_DBL(0.895488455427336f), FL2FXCONST_DBL(0.632455532033676f), FL2FXCONST_DBL(0.796214341106995f), FL2FXCONST_DBL(0.501187233627272f), FL2FXCONST_DBL(0.630957344480193f), FL2FXCONST_DBL(0.794328234724281f), FL2FXCONST_DBL(0.5f), FL2FXCONST_DBL(0.629462705897084f), FL2FXCONST_DBL(0.792446596230557f), FL2FXCONST_DBL(0.99763115748444f), FL2FXCONST_DBL(0.627971607877395f), FL2FXCONST_DBL(0.790569415042095f), FL2FXCONST_DBL(0.558354490188704f), FL2FXCONST_DBL(0.788696680600242f), FL2FXCONST_DBL(0.557031836333591f), FL2FXCONST_DBL(0.786828382371355f), FL2FXCONST_DBL(0.555712315637163f), FL2FXCONST_DBL(0.988211768802619f), FL2FXCONST_DBL(0.87865832060992f), FL2FXCONST_DBL(0.78125f), FL2FXCONST_DBL(0.694640394546454f), FL2FXCONST_DBL(0.942432183077448f), }; static const FIXP_DBL CLD_m[N_CLD] = { FL2FXCONST_DBL(0.281474976710656f), FL2FXCONST_DBL(0.518107571841987f), FL2FXCONST_DBL(0.4096f), FL2FXCONST_DBL(0.323817232401242f), FL2FXCONST_DBL(0.256f), FL2FXCONST_DBL(0.809543081003105f), FL2FXCONST_DBL(0.403812700467324f), FL2FXCONST_DBL(0.805712263548267f), FL2FXCONST_DBL(0.401901829041533f), FL2FXCONST_DBL(0.801899573803636f), FL2FXCONST_DBL(0.4f), FL2FXCONST_DBL(0.633957276984445f), FL2FXCONST_DBL(0.251188643150958f), FL2FXCONST_DBL(0.398107170553497f), FL2FXCONST_DBL(0.630957344480193f), FL2FXCONST_DBL(0.25f), FL2FXCONST_DBL(0.396223298115278f), FL2FXCONST_DBL(0.627971607877395f), FL2FXCONST_DBL(0.995267926383743f), FL2FXCONST_DBL(0.394348340300121f), FL2FXCONST_DBL(0.625f), FL2FXCONST_DBL(0.311759736713887f), FL2FXCONST_DBL(0.62204245398984f), FL2FXCONST_DBL(0.310284466689172f), FL2FXCONST_DBL(0.619098903305123f), FL2FXCONST_DBL(0.308816177750818f), FL2FXCONST_DBL(0.9765625f), FL2FXCONST_DBL(0.772040444377046f), FL2FXCONST_DBL(0.6103515625f), FL2FXCONST_DBL(0.482525277735654f), FL2FXCONST_DBL(0.888178419700125), }; static void calculateOpd(spatialDec* self, INT ottBoxIndx, INT parameterSetIndx, FIXP_DBL opd[MAX_PARAMETER_BANDS]) { INT band; for (band = 0; band < self->numOttBandsIPD; band++) { INT idxCld = self->ottCLD__FDK[ottBoxIndx][parameterSetIndx][band]; INT idxIpd = self->ottIPD__FDK[ottBoxIndx][parameterSetIndx][band]; INT idxIcc = self->ottICC__FDK[ottBoxIndx][parameterSetIndx][band]; FIXP_DBL cld, ipd; ipd = FX_CFG2FX_DBL(dequantIPD__FDK[idxIpd]); SpatialDequantGetCLD2Values(idxCld, &cld); /* ipd(idxIpd==8) == PI */ if (((cld == FL2FXCONST_DBL(0.0f)) && (idxIpd == 8)) || (idxIpd == 0)) { opd[2 * band] = FL2FXCONST_DBL(0.0f); } else { FDK_ASSERT(idxIpd > 0); opd[2 * band] = dequantIPD_CLD_ICC_splitAngle__FDK[idxIpd - 1][idxCld][idxIcc]; } opd[2 * band + 1] = opd[2 * band] - ipd; } } /* wrap phase in rad to the range of 0 <= x < 2*pi */ static FIXP_DBL wrapPhase(FIXP_DBL phase) { while (phase < (FIXP_DBL)0) phase += PIx2__IPD; while (phase >= PIx2__IPD) phase -= PIx2__IPD; FDK_ASSERT((phase >= (FIXP_DBL)0) && (phase < PIx2__IPD)); return phase; } /******************************************************************************* Functionname: param2UMX_PS_IPD ******************************************************************************* Description: Arguments: Return: *******************************************************************************/ static void param2UMX_PS_IPD_OPD__FDK( spatialDec* self, const SPATIAL_BS_FRAME* frame, FIXP_DBL H11[MAX_PARAMETER_BANDS], FIXP_DBL H12[MAX_PARAMETER_BANDS], FIXP_DBL H21[MAX_PARAMETER_BANDS], FIXP_DBL H22[MAX_PARAMETER_BANDS], FIXP_DBL c_l[MAX_PARAMETER_BANDS], FIXP_DBL c_r[MAX_PARAMETER_BANDS], int ottBoxIndx, int parameterSetIndx, int residualBands) { INT band; FIXP_DBL opd[2 * MAX_PARAMETER_BANDS]; INT numOttBands = self->numOttBands[ottBoxIndx]; INT numIpdBands; numIpdBands = frame->phaseMode ? self->numOttBandsIPD : 0; FDK_ASSERT(self->residualCoding == 0); param2UMX_PS_Core__FDK(self->ottCLD__FDK[ottBoxIndx][parameterSetIndx], self->ottICC__FDK[ottBoxIndx][parameterSetIndx], self->numOttBands[ottBoxIndx], residualBands, H11, H12, H21, H22, c_l, c_r); for (band = self->numOttBands[ottBoxIndx]; band < self->numParameterBands; band++) { H11[band] = H21[band] = H12[band] = H22[band] = FL2FXCONST_DBL(0.f); } if (frame->phaseMode) { calculateOpd(self, ottBoxIndx, parameterSetIndx, opd); for (band = 0; band < numIpdBands; band++) { self->PhaseLeft__FDK[band] = wrapPhase(opd[2 * band]); self->PhaseRight__FDK[band] = wrapPhase(opd[2 * band + 1]); } } for (band = numIpdBands; band < numOttBands; band++) { self->PhaseLeft__FDK[band] = FL2FXCONST_DBL(0.0f); self->PhaseRight__FDK[band] = FL2FXCONST_DBL(0.0f); } } FDK_INLINE void param2UMX_Prediction_Core__FDK( FIXP_DBL* H11re, FIXP_DBL* H11im, FIXP_DBL* H12re, FIXP_DBL* H12im, FIXP_DBL* H21re, FIXP_DBL* H21im, FIXP_DBL* H22re, FIXP_DBL* H22im, int cldIdx, int iccIdx, int ipdIdx, int band, int numOttBandsIPD, int resBands) { #define MAX_WEIGHT (1.2f) FDK_ASSERT((H12im == NULL) && (H22im == NULL)); /* always == 0 */ if ((band < numOttBandsIPD) && (cldIdx == 15) && (iccIdx == 0) && (ipdIdx == 8)) { const FIXP_DBL gain = FL2FXCONST_DBL(0.5f / MAX_WEIGHT) >> SCALE_PARAM_M2_212_PRED; *H11re = gain; if (band < resBands) { *H21re = gain; *H12re = gain; *H22re = -gain; } else { *H21re = -gain; *H12re = (FIXP_DBL)0; *H22re = (FIXP_DBL)0; } if ((H11im != NULL) && (H21im != NULL) /*&& (H12im!=NULL) && (H22im!=NULL)*/) { *H11im = (FIXP_DBL)0; *H21im = (FIXP_DBL)0; /* *H12im = (FIXP_DBL)0; */ /* *H22im = (FIXP_DBL)0; */ } } else { const FIXP_DBL one_m = (FIXP_DBL)MAXVAL_DBL; const int one_e = 0; /* iidLin = sqrt(cld); */ FIXP_DBL iidLin_m = sqrt_CLD_m[cldIdx]; int iidLin_e = sqrt_CLD_e[cldIdx]; /* iidLin2 = cld; */ FIXP_DBL iidLin2_m = CLD_m[cldIdx]; int iidLin2_e = sqrt_CLD_e[cldIdx] << 1; /* iidLin21 = iidLin2 + 1.0f; */ int iidLin21_e; FIXP_DBL iidLin21_m = fAddNorm(iidLin2_m, iidLin2_e, one_m, one_e, &iidLin21_e); /* iidIcc2 = iidLin * icc * 2.0f; */ FIXP_CFG icc = dequantICC__FDK[iccIdx]; int iidIcc2_e = iidLin_e + 1; FIXP_DBL iidIcc2_m = fMult(iidLin_m, icc); FIXP_DBL temp_m, sqrt_temp_m, inv_temp_m, weight_m; int temp_e, sqrt_temp_e, inv_temp_e, weight_e, scale; FIXP_DBL cosIpd, sinIpd; cosIpd = COS_IPD((band < numOttBandsIPD) ? ipdIdx : 0); sinIpd = SIN_IPD((band < numOttBandsIPD) ? ipdIdx : 0); /* temp = iidLin21 + iidIcc2 * cosIpd; */ temp_m = fAddNorm(iidLin21_m, iidLin21_e, fMult(iidIcc2_m, cosIpd), iidIcc2_e, &temp_e); /* calculate 1/temp needed later */ inv_temp_e = temp_e; inv_temp_m = invFixp(temp_m, &inv_temp_e); /* 1/weight = sqrt(temp) * 1/sqrt(iidLin21) */ if (temp_e & 1) { sqrt_temp_m = temp_m >> 1; sqrt_temp_e = (temp_e + 1) >> 1; } else { sqrt_temp_m = temp_m; sqrt_temp_e = temp_e >> 1; } sqrt_temp_m = sqrtFixp(sqrt_temp_m); if (iidLin21_e & 1) { iidLin21_e += 1; iidLin21_m >>= 1; } /* weight_[m,e] is actually 1/weight in the next few lines */ weight_m = invSqrtNorm2(iidLin21_m, &weight_e); weight_e -= iidLin21_e >> 1; weight_m = fMult(sqrt_temp_m, weight_m); weight_e += sqrt_temp_e; scale = fNorm(weight_m); weight_m = scaleValue(weight_m, scale); weight_e -= scale; /* weight = 0.5 * max(1/weight, 1/maxWeight) */ if ((weight_e < 0) || ((weight_e == 0) && (weight_m < FL2FXCONST_DBL(1.f / MAX_WEIGHT)))) { weight_m = FL2FXCONST_DBL(1.f / MAX_WEIGHT); weight_e = 0; } weight_e -= 1; { FIXP_DBL alphaRe_m, alphaIm_m, accu_m; int alphaRe_e, alphaIm_e, accu_e; /* alphaRe = (1.0f - iidLin2) / temp; */ alphaRe_m = fAddNorm(one_m, one_e, -iidLin2_m, iidLin2_e, &alphaRe_e); alphaRe_m = fMult(alphaRe_m, inv_temp_m); alphaRe_e += inv_temp_e; /* H11re = weight - alphaRe * weight; */ /* H21re = weight + alphaRe * weight; */ accu_m = fMult(alphaRe_m, weight_m); accu_e = alphaRe_e + weight_e; { int accu2_e; FIXP_DBL accu2_m; accu2_m = fAddNorm(weight_m, weight_e, -accu_m, accu_e, &accu2_e); *H11re = scaleValue(accu2_m, accu2_e - SCALE_PARAM_M2_212_PRED); accu2_m = fAddNorm(weight_m, weight_e, accu_m, accu_e, &accu2_e); *H21re = scaleValue(accu2_m, accu2_e - SCALE_PARAM_M2_212_PRED); } if ((H11im != NULL) && (H21im != NULL) /*&& (H12im != NULL) && (H22im != NULL)*/) { /* alphaIm = -iidIcc2 * sinIpd / temp; */ alphaIm_m = fMult(-iidIcc2_m, sinIpd); alphaIm_m = fMult(alphaIm_m, inv_temp_m); alphaIm_e = iidIcc2_e + inv_temp_e; /* H11im = -alphaIm * weight; */ /* H21im = alphaIm * weight; */ accu_m = fMult(alphaIm_m, weight_m); accu_e = alphaIm_e + weight_e; accu_m = scaleValue(accu_m, accu_e - SCALE_PARAM_M2_212_PRED); *H11im = -accu_m; *H21im = accu_m; /* *H12im = (FIXP_DBL)0; */ /* *H22im = (FIXP_DBL)0; */ } } if (band < resBands) { FIXP_DBL weight = scaleValue(weight_m, weight_e - SCALE_PARAM_M2_212_PRED); *H12re = weight; *H22re = -weight; } else { /* beta = 2.0f * iidLin * (float) sqrt(1.0f - icc * icc) * weight / temp; */ FIXP_DBL beta_m; int beta_e; beta_m = FX_SGL2FX_DBL(sqrt_one_minus_ICC2[iccIdx]); beta_e = 1; /* multipication with 2.0f */ beta_m = fMult(beta_m, weight_m); beta_e += weight_e; beta_m = fMult(beta_m, iidLin_m); beta_e += iidLin_e; beta_m = fMult(beta_m, inv_temp_m); beta_e += inv_temp_e; beta_m = scaleValue(beta_m, beta_e - SCALE_PARAM_M2_212_PRED); *H12re = beta_m; *H22re = -beta_m; } } } static void param2UMX_Prediction__FDK(spatialDec* self, FIXP_DBL* H11re, FIXP_DBL* H11im, FIXP_DBL* H12re, FIXP_DBL* H12im, FIXP_DBL* H21re, FIXP_DBL* H21im, FIXP_DBL* H22re, FIXP_DBL* H22im, int ottBoxIndx, int parameterSetIndx, int resBands) { int band; FDK_ASSERT((H12im == NULL) && (H22im == NULL)); /* always == 0 */ for (band = 0; band < self->numParameterBands; band++) { int cldIdx = self->ottCLD__FDK[ottBoxIndx][parameterSetIndx][band]; int iccIdx = self->ottICC__FDK[ottBoxIndx][parameterSetIndx][band]; int ipdIdx = self->ottIPD__FDK[ottBoxIndx][parameterSetIndx][band]; param2UMX_Prediction_Core__FDK( &H11re[band], (H11im ? &H11im[band] : NULL), &H12re[band], NULL, &H21re[band], (H21im ? &H21im[band] : NULL), &H22re[band], NULL, cldIdx, iccIdx, ipdIdx, band, self->numOttBandsIPD, resBands); } } /******************************************************************************* Functionname: initM1andM2 ******************************************************************************* Description: Arguments: Return: *******************************************************************************/ SACDEC_ERROR initM1andM2(spatialDec* self, int initStatesFlag, int configChanged) { SACDEC_ERROR err = MPS_OK; self->bOverwriteM1M2prev = (configChanged && !initStatesFlag) ? 1 : 0; { self->numM2rows = self->numOutputChannels; } if (initStatesFlag) { int i, j, k; for (i = 0; i < self->numM2rows; i++) { for (j = 0; j < self->numVChannels; j++) { for (k = 0; k < MAX_PARAMETER_BANDS; k++) { self->M2Real__FDK[i][j][k] = FL2FXCONST_DBL(0); self->M2RealPrev__FDK[i][j][k] = FL2FXCONST_DBL(0); } } } } return err; } fdk-aac-2.0.2/libSACdec/src/sac_calcM1andM2.h000066400000000000000000000134271404225310700203270ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround decoder library ************************* Author(s): Description: SAC Dec M1 and M2 calculation *******************************************************************************/ /* sa_calcM1andM2.h */ #ifndef SAC_CALCM1ANDM2_H #define SAC_CALCM1ANDM2_H #include "sac_dec.h" #define SCALE_PARAM_M1 3 /* Scaling of M2 matrix, but only for binaural upmix type. */ #define SCALE_PARAM_CALC_M2 (3) #define SCALE_PARAM_M2_515X (3) #define SCALE_PARAM_M2_525 (SCALE_PARAM_M1 + HRG_SF + 1 - SCALE_PARAM_CALC_M2) #define SCALE_PARAM_M2_212_PRED (3) /* Scaling of spectral data after applying M2 matrix, but only for binaural upmix type Scaling is compensated later in synthesis qmf filterbank */ #define SCALE_DATA_APPLY_M2 (1) /* Applying M2 parameter in combination with phase coding needs 2 bits headroom * because up to a maximum of 4 spectral values can be added for USAC */ #define SCALE_DATA_APPLY_M2_PC (2) SACDEC_ERROR initM1andM2(spatialDec* self, int initStatesFlag, int configChanged); int SpatialDecGetResidualIndex(spatialDec* self, int row); SACDEC_ERROR SpatialDecCalculateM1andM2(spatialDec* self, int ps, const SPATIAL_BS_FRAME* frame); #endif /* SAC_CALCM1ANDM2_H */ fdk-aac-2.0.2/libSACdec/src/sac_dec.cpp000066400000000000000000001544741404225310700174430ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround decoder library ************************* Author(s): Description: SAC Decoder Library *******************************************************************************/ #include "sac_dec_errorcodes.h" #include "sac_dec.h" #include "sac_process.h" #include "sac_bitdec.h" #include "sac_smoothing.h" #include "sac_calcM1andM2.h" #include "sac_reshapeBBEnv.h" #include "sac_stp.h" #include "sac_rom.h" #include "FDK_decorrelate.h" #include "FDK_trigFcts.h" #include "FDK_matrixCalloc.h" /* static int pbStrideTable[] = {1, 2, 5, 28}; see sac_rom.cpp */ enum { APPLY_M2_NONE = 0, /* init value */ APPLY_M2 = 1, /* apply m2 fallback implementation */ APPLY_M2_MODE212 = 2, /* apply m2 for 212 mode */ APPLY_M2_MODE212_Res_PhaseCoding = 3 /* apply m2 for 212 mode with residuals and phase coding */ }; /******************************************************************************************/ /* function: FDK_SpatialDecInitDefaultSpatialSpecificConfig */ /* output: struct of type SPATIAL_SPECIFIC_CONFIG */ /* input: core coder audio object type */ /* input: nr of core channels */ /* input: sampling rate */ /* input: nr of time slots */ /* input: decoder level */ /* input: flag indicating upmix type blind */ /* */ /* returns: error code */ /******************************************************************************************/ int FDK_SpatialDecInitDefaultSpatialSpecificConfig( SPATIAL_SPECIFIC_CONFIG *pSpatialSpecificConfig, AUDIO_OBJECT_TYPE coreCodec, int coreChannels, int samplingFreq, int nTimeSlots, int decoderLevel, int isBlind) { return SpatialDecDefaultSpecificConfig(pSpatialSpecificConfig, coreCodec, samplingFreq, nTimeSlots, decoderLevel, isBlind, coreChannels); } /******************************************************************************************/ /* function: FDK_SpatialDecCompareSpatialSpecificConfigHeader */ /* input: 2 pointers to a ssc */ /* */ /* output: - */ /* returns: error code (0 = equal, <>0 unequal) */ /******************************************************************************************/ int FDK_SpatialDecCompareSpatialSpecificConfigHeader( SPATIAL_SPECIFIC_CONFIG *pSsc1, SPATIAL_SPECIFIC_CONFIG *pSsc2) { int result = MPS_OK; /* we assume: every bit must be equal */ if (FDKmemcmp(pSsc1, pSsc2, sizeof(SPATIAL_SPECIFIC_CONFIG)) != 0) { result = MPS_UNEQUAL_SSC; } return result; } /******************************************************************************* Functionname: SpatialDecClearFrameData ******************************************************************************* Description: Clear/Fake frame data to avoid misconfiguration and allow proper error concealment. Arguments: Input: self (frame data) Output: No return value. *******************************************************************************/ static void SpatialDecClearFrameData( spatialDec *self, /* Shall be removed */ SPATIAL_BS_FRAME *bsFrame, const SACDEC_CREATION_PARAMS *const setup) { int i; FDK_ASSERT(self != NULL); FDK_ASSERT(bsFrame != NULL); FDK_ASSERT(setup != NULL); /* do not apply shaping tools (GES or STP) */ for (i = 0; i < setup->maxNumOutputChannels; i += 1) { /* MAX_OUTPUT_CHANNELS */ bsFrame->tempShapeEnableChannelSTP[i] = 0; bsFrame->tempShapeEnableChannelGES[i] = 0; } bsFrame->TsdData->bsTsdEnable = 0; /* use only 1 parameter set at the end of the frame */ bsFrame->numParameterSets = 1; bsFrame->paramSlot[0] = self->timeSlots - 1; /* parameter smoothing tool set to off */ bsFrame->bsSmoothMode[0] = 0; initParameterSmoothing(self); /* reset residual data */ { int resQmfBands, resTimeSlots = (1); resQmfBands = setup->maxNumQmfBands; for (i = 0; i < setup->bProcResidual ? fMin(setup->maxNumResChannels, setup->maxNumOttBoxes + setup->maxNumInputChannels) : 0; i += 1) { for (int j = 0; j < resTimeSlots; j += 1) { for (int k = 0; k < resQmfBands; k += 1) { self->qmfResidualReal__FDK[i][j][k] = FL2FXCONST_DBL(0.0f); self->qmfResidualImag__FDK[i][j][k] = FL2FXCONST_DBL(0.0f); } } } } return; } /******************************************************************************* Functionname: FDK_SpatialDecOpen ******************************************************************************* Description: Arguments: Return: *******************************************************************************/ spatialDec *FDK_SpatialDecOpen(const SPATIAL_DEC_CONFIG *config, int stereoConfigIndex) { int i; int lfSize, hfSize; spatialDec *self = NULL; SACDEC_CREATION_PARAMS setup; switch (config->decoderLevel) { case DECODER_LEVEL_0: /* 212 maxNumOutputChannels== 2 */ setup.maxNumInputChannels = 1; setup.maxNumOutputChannels = 2; setup.maxNumQmfBands = 64; setup.maxNumXChannels = 2; setup.maxNumVChannels = 2; setup.maxNumDecorChannels = 1; setup.bProcResidual = 1; setup.maxNumResidualChannels = 0; setup.maxNumOttBoxes = 1; setup.maxNumParams = setup.maxNumInputChannels + setup.maxNumOttBoxes; break; default: return NULL; } setup.maxNumResChannels = 1; { switch (config->maxNumOutputChannels) { case OUTPUT_CHANNELS_2_0: setup.maxNumOutputChannels = fMin(setup.maxNumOutputChannels, 2); break; case OUTPUT_CHANNELS_DEFAULT: default: break; } } setup.maxNumHybridBands = SacGetHybridSubbands(setup.maxNumQmfBands); switch (config->decoderMode) { case EXT_HQ_ONLY: setup.maxNumCmplxQmfBands = setup.maxNumQmfBands; setup.maxNumCmplxHybBands = setup.maxNumHybridBands; break; default: setup.maxNumCmplxQmfBands = fixMax(PC_NUM_BANDS, setup.maxNumQmfBands); setup.maxNumCmplxHybBands = fixMax(PC_NUM_HYB_BANDS, setup.maxNumHybridBands); break; } /* switch config->decoderMode */ FDK_ALLOCATE_MEMORY_1D_INT(self, 1, spatialDec, SECT_DATA_L2) self->createParams = setup; FDK_ALLOCATE_MEMORY_1D(self->param2hyb, MAX_PARAMETER_BANDS + 1, int) FDK_ALLOCATE_MEMORY_1D(self->numOttBands, setup.maxNumOttBoxes, int) /* allocate arrays */ FDK_ALLOCATE_MEMORY_1D(self->smgTime, MAX_PARAMETER_SETS, int) FDK_ALLOCATE_MEMORY_2D(self->smgData, MAX_PARAMETER_SETS, MAX_PARAMETER_BANDS, UCHAR) FDK_ALLOCATE_MEMORY_3D(self->ottCLD__FDK, setup.maxNumOttBoxes, MAX_PARAMETER_SETS, MAX_PARAMETER_BANDS, SCHAR) FDK_ALLOCATE_MEMORY_3D(self->ottICC__FDK, setup.maxNumOttBoxes, MAX_PARAMETER_SETS, MAX_PARAMETER_BANDS, SCHAR) FDK_ALLOCATE_MEMORY_3D(self->ottIPD__FDK, setup.maxNumOttBoxes, MAX_PARAMETER_SETS, MAX_PARAMETER_BANDS, SCHAR) /* Last parameters from prev frame */ FDK_ALLOCATE_MEMORY_2D(self->ottCLDidxPrev, setup.maxNumOttBoxes, MAX_PARAMETER_BANDS, SCHAR) FDK_ALLOCATE_MEMORY_2D(self->ottICCidxPrev, setup.maxNumOttBoxes, MAX_PARAMETER_BANDS, SCHAR) FDK_ALLOCATE_MEMORY_3D(self->ottICCdiffidx, setup.maxNumOttBoxes, MAX_PARAMETER_SETS, MAX_PARAMETER_BANDS, SCHAR) FDK_ALLOCATE_MEMORY_2D(self->ottIPDidxPrev, setup.maxNumOttBoxes, MAX_PARAMETER_BANDS, SCHAR) FDK_ALLOCATE_MEMORY_2D(self->arbdmxGainIdxPrev, setup.maxNumInputChannels, MAX_PARAMETER_BANDS, SCHAR) FDK_ALLOCATE_MEMORY_2D(self->cmpOttCLDidxPrev, setup.maxNumOttBoxes, MAX_PARAMETER_BANDS, SCHAR) FDK_ALLOCATE_MEMORY_2D(self->cmpOttICCidxPrev, setup.maxNumOttBoxes, MAX_PARAMETER_BANDS, SCHAR) FDK_ALLOCATE_MEMORY_3D(self->outIdxData, setup.maxNumOttBoxes, MAX_PARAMETER_SETS, MAX_PARAMETER_BANDS, SCHAR) FDK_ALLOCATE_MEMORY_3D(self->arbdmxGain__FDK, setup.maxNumInputChannels, MAX_PARAMETER_SETS, MAX_PARAMETER_BANDS, SCHAR) FDK_ALLOCATE_MEMORY_1D(self->arbdmxAlpha__FDK, setup.maxNumInputChannels, FIXP_DBL) FDK_ALLOCATE_MEMORY_1D(self->arbdmxAlphaPrev__FDK, setup.maxNumInputChannels, FIXP_DBL) FDK_ALLOCATE_MEMORY_2D(self->cmpArbdmxGainIdxPrev, setup.maxNumInputChannels, MAX_PARAMETER_BANDS, SCHAR) FDK_ALLOCATE_MEMORY_2D(self->cmpOttIPDidxPrev, setup.maxNumOttBoxes, MAX_PARAMETER_BANDS, SCHAR) FDK_ALLOCATE_MEMORY_3D_INT(self->M2Real__FDK, setup.maxNumOutputChannels, setup.maxNumVChannels, MAX_PARAMETER_BANDS, FIXP_DBL, SECT_DATA_L2) FDK_ALLOCATE_MEMORY_3D(self->M2Imag__FDK, setup.maxNumOutputChannels, setup.maxNumVChannels, MAX_PARAMETER_BANDS, FIXP_DBL) FDK_ALLOCATE_MEMORY_3D_INT(self->M2RealPrev__FDK, setup.maxNumOutputChannels, setup.maxNumVChannels, MAX_PARAMETER_BANDS, FIXP_DBL, SECT_DATA_L2) FDK_ALLOCATE_MEMORY_3D(self->M2ImagPrev__FDK, setup.maxNumOutputChannels, setup.maxNumVChannels, MAX_PARAMETER_BANDS, FIXP_DBL) FDK_ALLOCATE_MEMORY_2D_INT_ALIGNED( self->qmfInputReal__FDK, setup.maxNumInputChannels, setup.maxNumQmfBands, FIXP_DBL, SECT_DATA_L2) FDK_ALLOCATE_MEMORY_2D_INT_ALIGNED( self->qmfInputImag__FDK, setup.maxNumInputChannels, setup.maxNumCmplxQmfBands, FIXP_DBL, SECT_DATA_L2) FDK_ALLOCATE_MEMORY_2D_INT(self->hybInputReal__FDK, setup.maxNumInputChannels, setup.maxNumHybridBands, FIXP_DBL, SECT_DATA_L2) FDK_ALLOCATE_MEMORY_2D_INT(self->hybInputImag__FDK, setup.maxNumInputChannels, setup.maxNumCmplxHybBands, FIXP_DBL, SECT_DATA_L2) if (setup.bProcResidual) { FDK_ALLOCATE_MEMORY_1D(self->qmfResidualReal__FDK, setup.maxNumResChannels, FIXP_DBL **) FDK_ALLOCATE_MEMORY_1D(self->qmfResidualImag__FDK, setup.maxNumResChannels, FIXP_DBL **) FDK_ALLOCATE_MEMORY_1D(self->hybResidualReal__FDK, setup.maxNumResChannels, FIXP_DBL *) FDK_ALLOCATE_MEMORY_1D(self->hybResidualImag__FDK, setup.maxNumResChannels, FIXP_DBL *) for (i = 0; i < setup.maxNumResChannels; i++) { int resQmfBands = (config->decoderMode == EXT_LP_ONLY) ? PC_NUM_BANDS : setup.maxNumQmfBands; int resHybBands = (config->decoderMode == EXT_LP_ONLY) ? PC_NUM_HYB_BANDS : setup.maxNumHybridBands; /* Alignment is needed for USAC residuals because QMF analysis directly * writes to this buffer. */ FDK_ALLOCATE_MEMORY_2D_INT_ALIGNED(self->qmfResidualReal__FDK[i], (1), resQmfBands, FIXP_DBL, SECT_DATA_L1) FDK_ALLOCATE_MEMORY_2D_INT_ALIGNED(self->qmfResidualImag__FDK[i], (1), resQmfBands, FIXP_DBL, SECT_DATA_L1) FDK_ALLOCATE_MEMORY_1D(self->hybResidualReal__FDK[i], setup.maxNumHybridBands, FIXP_DBL) FDK_ALLOCATE_MEMORY_1D(self->hybResidualImag__FDK[i], resHybBands, FIXP_DBL) } } /* if (setup.bProcResidual) */ FDK_ALLOCATE_MEMORY_2D_INT(self->wReal__FDK, setup.maxNumVChannels, setup.maxNumHybridBands, FIXP_DBL, SECT_DATA_L2) FDK_ALLOCATE_MEMORY_2D_INT(self->wImag__FDK, setup.maxNumVChannels, setup.maxNumCmplxHybBands, FIXP_DBL, SECT_DATA_L2) FDK_ALLOCATE_MEMORY_2D_INT(self->hybOutputRealDry__FDK, setup.maxNumOutputChannels, setup.maxNumHybridBands, FIXP_DBL, SECT_DATA_L2) FDK_ALLOCATE_MEMORY_2D_INT(self->hybOutputImagDry__FDK, setup.maxNumOutputChannels, setup.maxNumCmplxHybBands, FIXP_DBL, SECT_DATA_L2) FDK_ALLOCATE_MEMORY_2D_INT(self->hybOutputRealWet__FDK, setup.maxNumOutputChannels, setup.maxNumHybridBands, FIXP_DBL, SECT_DATA_L2) FDK_ALLOCATE_MEMORY_2D_INT(self->hybOutputImagWet__FDK, setup.maxNumOutputChannels, setup.maxNumCmplxHybBands, FIXP_DBL, SECT_DATA_L2) FDK_ALLOCATE_MEMORY_1D(self->hybridSynthesis, setup.maxNumOutputChannels, FDK_SYN_HYB_FILTER) FDK_ALLOCATE_MEMORY_1D( self->hybridAnalysis, setup.bProcResidual ? setup.maxNumInputChannels + setup.maxNumResChannels : setup.maxNumInputChannels, FDK_ANA_HYB_FILTER) lfSize = 2 * BUFFER_LEN_LF * MAX_QMF_BANDS_TO_HYBRID; { hfSize = BUFFER_LEN_HF * ((setup.maxNumQmfBands - MAX_QMF_BANDS_TO_HYBRID) + (setup.maxNumCmplxQmfBands - MAX_QMF_BANDS_TO_HYBRID)); } FDK_ALLOCATE_MEMORY_2D_INT(self->pHybridAnaStatesLFdmx, setup.maxNumInputChannels, lfSize, FIXP_DBL, SECT_DATA_L2) { FDK_ALLOCATE_MEMORY_2D(self->pHybridAnaStatesHFdmx, setup.maxNumInputChannels, hfSize, FIXP_DBL) } for (i = 0; i < setup.maxNumInputChannels; i++) { FIXP_DBL *pHybridAnaStatesHFdmx; pHybridAnaStatesHFdmx = self->pHybridAnaStatesHFdmx[i]; FDKhybridAnalysisOpen(&self->hybridAnalysis[i], self->pHybridAnaStatesLFdmx[i], lfSize * sizeof(FIXP_DBL), pHybridAnaStatesHFdmx, hfSize * sizeof(FIXP_DBL)); } if (setup.bProcResidual) { lfSize = 2 * BUFFER_LEN_LF * MAX_QMF_BANDS_TO_HYBRID; hfSize = BUFFER_LEN_HF * ((((config->decoderMode == EXT_LP_ONLY) ? PC_NUM_BANDS : setup.maxNumQmfBands) - MAX_QMF_BANDS_TO_HYBRID) + (setup.maxNumCmplxQmfBands - MAX_QMF_BANDS_TO_HYBRID)); FDK_ALLOCATE_MEMORY_2D_INT(self->pHybridAnaStatesLFres, setup.maxNumResChannels, lfSize, FIXP_DBL, SECT_DATA_L2) FDK_ALLOCATE_MEMORY_2D(self->pHybridAnaStatesHFres, setup.maxNumResChannels, hfSize, FIXP_DBL) for (i = setup.maxNumInputChannels; i < (setup.maxNumInputChannels + setup.maxNumResChannels); i++) { FDKhybridAnalysisOpen( &self->hybridAnalysis[i], self->pHybridAnaStatesLFres[i - setup.maxNumInputChannels], lfSize * sizeof(FIXP_DBL), self->pHybridAnaStatesHFres[i - setup.maxNumInputChannels], hfSize * sizeof(FIXP_DBL)); } } FDK_ALLOCATE_MEMORY_1D(self->smoothState, 1, SMOOTHING_STATE) FDK_ALLOCATE_MEMORY_1D(self->reshapeBBEnvState, 1, RESHAPE_BBENV_STATE) FDK_ALLOCATE_MEMORY_1D(self->apDecor, setup.maxNumDecorChannels, DECORR_DEC) FDK_ALLOCATE_MEMORY_2D_INT(self->pDecorBufferCplx, setup.maxNumDecorChannels, (2 * ((825) + (373))), FIXP_DBL, SECT_DATA_L2) for (i = 0; i < setup.maxNumDecorChannels; i++) { if (FDKdecorrelateOpen(&self->apDecor[i], self->pDecorBufferCplx[i], (2 * ((825) + (373))))) { goto bail; } } if (subbandTPCreate(&self->hStpDec) != MPS_OK) { goto bail; } /* save general decoder configuration */ self->decoderLevel = config->decoderLevel; self->decoderMode = config->decoderMode; self->binauralMode = config->binauralMode; /* preinitialize configuration */ self->partiallyComplex = (config->decoderMode != EXT_HQ_ONLY) ? 1 : 0; /* Set to default state */ SpatialDecConcealment_Init(&self->concealInfo, MPEGS_CONCEAL_RESET_ALL); /* Everything is fine so return the handle */ return self; bail: /* Collector for all errors. Deallocate all memory and return a invalid handle. */ FDK_SpatialDecClose(self); return NULL; } /******************************************************************************* Functionname: isValidConfig ******************************************************************************* Description: Validate if configuration is supported in present instance Arguments: Return: 1: all okay 0: configuration not supported *******************************************************************************/ static int isValidConfig(spatialDec const *const self, const SPATIAL_DEC_UPMIX_TYPE upmixType, SPATIALDEC_PARAM const *const pUserParams, const AUDIO_OBJECT_TYPE coreAot) { UPMIXTYPE nUpmixType; FDK_ASSERT(self != NULL); FDK_ASSERT(pUserParams != NULL); nUpmixType = (UPMIXTYPE)upmixType; switch (nUpmixType) { case UPMIXTYPE_BYPASS: /* UPMIX_TYPE_BYPASS */ break; case UPMIXTYPE_NORMAL: /* UPMIX_TYPE_NORMAL */ break; default: return 0; /* unsupported upmixType */ } return 1; /* upmixType supported */ } static SACDEC_ERROR CheckLevelTreeUpmixType( const SACDEC_CREATION_PARAMS *const pCreateParams, const SPATIAL_SPECIFIC_CONFIG *const pSsc, const int decoderLevel, const UPMIXTYPE upmixType) { SACDEC_ERROR err = MPS_OK; int nOutputChannels, treeConfig; FDK_ASSERT(pCreateParams != NULL); FDK_ASSERT(pSsc != NULL); treeConfig = pSsc->treeConfig; switch (decoderLevel) { case 0: { if (treeConfig != SPATIALDEC_MODE_RSVD7) { err = MPS_INVALID_TREECONFIG; goto bail; } break; } default: err = MPS_INVALID_PARAMETER /* MPS_UNIMPLEMENTED */; goto bail; } switch (upmixType) { case UPMIXTYPE_BYPASS: nOutputChannels = pSsc->nInputChannels; break; default: nOutputChannels = pSsc->nOutputChannels; break; } /* Is sufficient memory allocated. */ if ((pSsc->nInputChannels > pCreateParams->maxNumInputChannels) || (nOutputChannels > pCreateParams->maxNumOutputChannels) || (pSsc->nOttBoxes > pCreateParams->maxNumOttBoxes)) { err = MPS_INVALID_PARAMETER; } bail: return err; } void SpatialDecInitParserContext(spatialDec *self) { int i, j; for (i = 0; i < self->createParams.maxNumOttBoxes; i += 1) { for (j = 0; j < MAX_PARAMETER_BANDS; j++) { self->ottCLDidxPrev[i][j] = 0; self->ottICCidxPrev[i][j] = 0; self->cmpOttCLDidxPrev[i][j] = 0; self->cmpOttICCidxPrev[i][j] = 0; } } for (i = 0; i < self->createParams.maxNumInputChannels; i++) { for (j = 0; j < MAX_PARAMETER_BANDS; j++) { self->arbdmxGainIdxPrev[i][j] = 0; self->cmpArbdmxGainIdxPrev[i][j] = 0; } } } /******************************************************************************* Functionname: FDK_SpatialDecInit ******************************************************************************* Description: Arguments: Return: *******************************************************************************/ SACDEC_ERROR FDK_SpatialDecInit(spatialDec *self, SPATIAL_BS_FRAME *frame, SPATIAL_SPECIFIC_CONFIG *pSpatialSpecificConfig, int nQmfBands, SPATIAL_DEC_UPMIX_TYPE const upmixType, SPATIALDEC_PARAM *pUserParams, UINT initFlags) { SACDEC_ERROR err = MPS_OK; int nCh, i, j, k; int maxQmfBands; int bypassMode = 0; self->useFDreverb = 0; /* check configuration parameter */ if (!isValidConfig(self, upmixType, pUserParams, pSpatialSpecificConfig->coreCodec)) { return MPS_INVALID_PARAMETER; } /* check tree configuration */ err = CheckLevelTreeUpmixType(&self->createParams, pSpatialSpecificConfig, self->decoderLevel, (UPMIXTYPE)upmixType); if (err != MPS_OK) { goto bail; } /* Store and update instance after all checks passed successfully: */ self->upmixType = (UPMIXTYPE)upmixType; if (initFlags & MPEGS_INIT_PARAMS_ERROR_CONCEALMENT) { /* At least one error concealment parameter changed */ err = SpatialDecConcealment_SetParam( &self->concealInfo, SAC_DEC_CONCEAL_METHOD, pUserParams->concealMethod); if (err != MPS_OK) { goto bail; } err = SpatialDecConcealment_SetParam(&self->concealInfo, SAC_DEC_CONCEAL_NUM_KEEP_FRAMES, pUserParams->concealNumKeepFrames); if (err != MPS_OK) { goto bail; } err = SpatialDecConcealment_SetParam( &self->concealInfo, SAC_DEC_CONCEAL_FADE_OUT_SLOPE_LENGTH, pUserParams->concealFadeOutSlopeLength); if (err != MPS_OK) { goto bail; } err = SpatialDecConcealment_SetParam(&self->concealInfo, SAC_DEC_CONCEAL_FADE_IN_SLOPE_LENGTH, pUserParams->concealFadeInSlopeLength); if (err != MPS_OK) { goto bail; } err = SpatialDecConcealment_SetParam(&self->concealInfo, SAC_DEC_CONCEAL_NUM_RELEASE_FRAMES, pUserParams->concealNumReleaseFrames); if (err != MPS_OK) { goto bail; } } if (initFlags & MPEGS_INIT_STATES_ERROR_CONCEALMENT) { /* Set to default state */ SpatialDecConcealment_Init(&self->concealInfo, MPEGS_CONCEAL_RESET_STATE); } /* determine bypass mode */ bypassMode |= pUserParams->bypassMode; bypassMode |= ((self->upmixType == UPMIXTYPE_BYPASS) ? 1 : 0); /* static decoder scale depends on number of qmf bands */ switch (nQmfBands) { case 16: case 24: case 32: self->staticDecScale = 21; break; case 64: self->staticDecScale = 22; break; default: return MPS_INVALID_PARAMETER; } self->numParameterSetsPrev = 1; self->qmfBands = nQmfBands; /* self->hybridBands will be updated in SpatialDecDecodeHeader() below. */ self->bShareDelayWithSBR = 0; err = SpatialDecDecodeHeader(self, pSpatialSpecificConfig); if (err != MPS_OK) { goto bail; } self->stereoConfigIndex = pSpatialSpecificConfig->stereoConfigIndex; if (initFlags & MPEGS_INIT_STATES_ANA_QMF_FILTER) { self->qmfInputDelayBufPos = 0; self->pc_filterdelay = 1; /* Division by 0 not possible */ } maxQmfBands = self->qmfBands; /* init residual decoder */ /* init tonality smoothing */ if (initFlags & MPEGS_INIT_STATES_PARAM) { initParameterSmoothing(self); } /* init GES */ initBBEnv(self, (initFlags & MPEGS_INIT_STATES_GES) ? 1 : 0); /* Clip protection is applied only for normal processing. */ if (!isTwoChMode(self->upmixType) && !bypassMode) { self->staticDecScale += self->clipProtectGainSF__FDK; } { UINT flags = 0; INT initStatesFlag = (initFlags & MPEGS_INIT_STATES_ANA_QMF_FILTER) ? 1 : 0; INT useLdFilter = (self->pConfigCurrent->syntaxFlags & SACDEC_SYNTAX_LD) ? 1 : 0; flags = self->pQmfDomain->globalConf.flags_requested; flags &= (~(UINT)QMF_FLAG_LP); if (initStatesFlag) flags &= ~QMF_FLAG_KEEP_STATES; else flags |= QMF_FLAG_KEEP_STATES; if (useLdFilter) flags |= QMF_FLAG_MPSLDFB; else flags &= ~QMF_FLAG_MPSLDFB; self->pQmfDomain->globalConf.flags_requested = flags; FDK_QmfDomain_Configure(self->pQmfDomain); /* output scaling */ for (nCh = 0; nCh < self->numOutputChannelsAT; nCh++) { int outputScale = 0, outputGain_e = 0, scale = -(8) + (1); FIXP_DBL outputGain_m = getChGain(self, nCh, &outputGain_e); if (!isTwoChMode(self->upmixType) && !bypassMode) { outputScale += self->clipProtectGainSF__FDK; /* consider clip protection scaling at synthesis qmf */ } scale += outputScale; qmfChangeOutScalefactor(&self->pQmfDomain->QmfDomainOut[nCh].fb, scale); qmfChangeOutGain(&self->pQmfDomain->QmfDomainOut[nCh].fb, outputGain_m, outputGain_e); } } for (nCh = 0; nCh < self->numOutputChannelsAT; nCh++) { FDKhybridSynthesisInit(&self->hybridSynthesis[nCh], THREE_TO_TEN, self->qmfBands, maxQmfBands); } /* for input, residual channels and arbitrary down-mix residual channels */ for (nCh = 0; nCh < self->createParams.maxNumInputChannels; nCh++) { FDKhybridAnalysisInit( &self->hybridAnalysis[nCh], THREE_TO_TEN, self->qmfBands, maxQmfBands, (initFlags & MPEGS_INIT_STATES_ANA_HYB_FILTER) ? 1 : 0); } for (; nCh < (self->createParams.bProcResidual ? (self->createParams.maxNumInputChannels + self->createParams.maxNumResChannels) : self->createParams.maxNumInputChannels); nCh++) { FDKhybridAnalysisInit(&self->hybridAnalysis[nCh], THREE_TO_TEN, maxQmfBands, maxQmfBands, 0); } { for (k = 0; k < self->numDecorSignals; k++) { int errCode, idec; FDK_DECORR_TYPE decorrType = DECORR_PS; decorrType = DECORR_LD; if (self->pConfigCurrent->syntaxFlags & (SACDEC_SYNTAX_USAC | SACDEC_SYNTAX_RSVD50)) { decorrType = ((self->treeConfig == TREE_212) && (self->decorrType == DECORR_PS)) ? DECORR_PS : DECORR_USAC; } { idec = k; if (self->pConfigCurrent->syntaxFlags & SACDEC_SYNTAX_LD) { if (self->treeConfig == TREE_212 && k == 0) { idec = 2; } } } errCode = FDKdecorrelateInit( &self->apDecor[k], self->hybridBands, decorrType, DUCKER_AUTOMATIC, self->decorrConfig, idec, 0, /* self->partiallyComplex */ 0, 0, /* isLegacyPS */ (initFlags & MPEGS_INIT_STATES_DECORRELATOR) ? 1 : 0); if (errCode) return MPS_NOTOK; } } /* !self->partiallyComplex */ err = initM1andM2(self, (initFlags & MPEGS_INIT_STATES_M1M2) ? 1 : 0, (initFlags & MPEGS_INIT_CONFIG) ? 1 : 0); if (err != MPS_OK) return err; /* Initialization of previous frame data */ if (initFlags & MPEGS_INIT_STATES_PARAM) { for (i = 0; i < self->createParams.maxNumOttBoxes; i += 1) { /* reset icc diff data */ for (k = 0; k < MAX_PARAMETER_SETS; k += 1) { for (j = 0; j < MAX_PARAMETER_BANDS; j += 1) { self->ottICCdiffidx[i][k][j] = 0; } } } /* Parameter Smoothing */ /* robustness: init with one of the values of smgTimeTable[] = {64, 128, 256, 512} to avoid division by zero in calcFilterCoeff__FDK() */ self->smoothState->prevSmgTime = smgTimeTable[2]; /* == 256 */ FDKmemclear(self->smoothState->prevSmgData, MAX_PARAMETER_BANDS * sizeof(UCHAR)); FDKmemclear(self->smoothState->opdLeftState__FDK, MAX_PARAMETER_BANDS * sizeof(FIXP_DBL)); FDKmemclear(self->smoothState->opdRightState__FDK, MAX_PARAMETER_BANDS * sizeof(FIXP_DBL)); } self->prevTimeSlot = -1; self->curTimeSlot = MAX_TIME_SLOTS + 1; /* Initialize with a invalid value to trigger concealment if first frame has no valid data. */ self->curPs = 0; subbandTPInit(self->hStpDec); bail: return err; } void SpatialDecChannelProperties(spatialDec *self, AUDIO_CHANNEL_TYPE channelType[], UCHAR channelIndices[], const FDK_channelMapDescr *const mapDescr) { if ((self == NULL) || (channelType == NULL) || (channelIndices == NULL) || (mapDescr == NULL)) { return; /* no extern buffer to be filled */ } if (self->numOutputChannelsAT != treePropertyTable[self->treeConfig].numOutputChannels) { int ch; /* Declare all channels to be front channels: */ for (ch = 0; ch < self->numOutputChannelsAT; ch += 1) { channelType[ch] = ACT_FRONT; channelIndices[ch] = ch; } } else { /* ISO/IEC FDIS 23003-1:2006(E), page 46, Table 40 bsTreeConfig */ switch (self->treeConfig) { case TREE_212: channelType[0] = ACT_FRONT; channelIndices[0] = 0; channelType[1] = ACT_FRONT; channelIndices[1] = 1; break; default:; } } } /******************************************************************************* Functionname: FDK_SpatialDecClose ******************************************************************************* Description: Arguments: Return: *******************************************************************************/ void FDK_SpatialDecClose(spatialDec *self) { if (self) { int k; if (self->apDecor != NULL) { for (k = 0; k < self->createParams.maxNumDecorChannels; k++) { FDKdecorrelateClose(&(self->apDecor[k])); } FDK_FREE_MEMORY_1D(self->apDecor); } if (self->pDecorBufferCplx != NULL) { FDK_FREE_MEMORY_2D(self->pDecorBufferCplx); } subbandTPDestroy(&self->hStpDec); FDK_FREE_MEMORY_1D(self->reshapeBBEnvState); FDK_FREE_MEMORY_1D(self->smoothState); FDK_FREE_MEMORY_2D(self->pHybridAnaStatesLFdmx); FDK_FREE_MEMORY_2D(self->pHybridAnaStatesHFdmx); FDK_FREE_MEMORY_2D(self->pHybridAnaStatesLFres); FDK_FREE_MEMORY_2D(self->pHybridAnaStatesHFres); FDK_FREE_MEMORY_1D(self->hybridAnalysis); FDK_FREE_MEMORY_1D(self->hybridSynthesis); /* The time buffer is passed to the decoder from outside to avoid copying * (zero copy). */ /* FDK_FREE_MEMORY_2D(self->timeOut__FDK); */ FDK_FREE_MEMORY_2D(self->hybOutputImagWet__FDK); FDK_FREE_MEMORY_2D(self->hybOutputRealWet__FDK); FDK_FREE_MEMORY_2D(self->hybOutputImagDry__FDK); FDK_FREE_MEMORY_2D(self->hybOutputRealDry__FDK); FDK_FREE_MEMORY_2D(self->wImag__FDK); FDK_FREE_MEMORY_2D(self->wReal__FDK); if (self->createParams.bProcResidual) { int i; for (i = 0; i < self->createParams.maxNumResChannels; i++) { if (self->hybResidualImag__FDK != NULL) FDK_FREE_MEMORY_1D(self->hybResidualImag__FDK[i]); if (self->hybResidualReal__FDK != NULL) FDK_FREE_MEMORY_1D(self->hybResidualReal__FDK[i]); if (self->qmfResidualImag__FDK != NULL) FDK_FREE_MEMORY_2D_ALIGNED(self->qmfResidualImag__FDK[i]); if (self->qmfResidualReal__FDK != NULL) FDK_FREE_MEMORY_2D_ALIGNED(self->qmfResidualReal__FDK[i]); } FDK_FREE_MEMORY_1D(self->hybResidualImag__FDK); FDK_FREE_MEMORY_1D(self->hybResidualReal__FDK); FDK_FREE_MEMORY_1D(self->qmfResidualImag__FDK); FDK_FREE_MEMORY_1D(self->qmfResidualReal__FDK); } /* self->createParams.bProcResidual */ FDK_FREE_MEMORY_2D(self->hybInputImag__FDK); FDK_FREE_MEMORY_2D(self->hybInputReal__FDK); FDK_FREE_MEMORY_2D_ALIGNED(self->qmfInputImag__FDK); FDK_FREE_MEMORY_2D_ALIGNED(self->qmfInputReal__FDK); FDK_FREE_MEMORY_3D(self->M2ImagPrev__FDK); FDK_FREE_MEMORY_3D(self->M2RealPrev__FDK); FDK_FREE_MEMORY_3D(self->M2Imag__FDK); FDK_FREE_MEMORY_3D(self->M2Real__FDK); FDK_FREE_MEMORY_1D(self->arbdmxAlphaPrev__FDK); FDK_FREE_MEMORY_1D(self->arbdmxAlpha__FDK); FDK_FREE_MEMORY_3D(self->arbdmxGain__FDK); FDK_FREE_MEMORY_3D(self->ottIPD__FDK); FDK_FREE_MEMORY_3D(self->ottICC__FDK); FDK_FREE_MEMORY_3D(self->ottCLD__FDK); /* Last parameters from prev frame */ FDK_FREE_MEMORY_2D(self->ottCLDidxPrev); FDK_FREE_MEMORY_2D(self->ottICCidxPrev); FDK_FREE_MEMORY_3D(self->ottICCdiffidx); FDK_FREE_MEMORY_2D(self->ottIPDidxPrev); FDK_FREE_MEMORY_2D(self->arbdmxGainIdxPrev); FDK_FREE_MEMORY_2D(self->cmpOttCLDidxPrev); FDK_FREE_MEMORY_2D(self->cmpOttICCidxPrev); FDK_FREE_MEMORY_3D(self->outIdxData); FDK_FREE_MEMORY_2D(self->cmpOttIPDidxPrev); FDK_FREE_MEMORY_2D(self->cmpArbdmxGainIdxPrev); FDK_FREE_MEMORY_2D(self->smgData); FDK_FREE_MEMORY_1D(self->smgTime); FDK_FREE_MEMORY_1D(self->numOttBands); FDK_FREE_MEMORY_1D(self->param2hyb); FDK_FREE_MEMORY_1D(self); } return; } /** * \brief Apply Surround bypass buffer copies * \param self spatialDec handle * \param hybInputReal * \param hybInputImag * \param hybOutputReal * \param hybOutputImag * \param numInputChannels amount if input channels available in hybInputReal * and hybInputImag, which may differ from self->numInputChannels. */ static void SpatialDecApplyBypass(spatialDec *self, FIXP_DBL **hybInputReal, FIXP_DBL **hybInputImag, FIXP_DBL **hybOutputReal, FIXP_DBL **hybOutputImag, const int numInputChannels) { int complexHybBands; complexHybBands = self->hybridBands; { int ch; int rf = -1, lf = -1, cf = -1; /* Right Front, Left Front, Center Front */ /* Determine output channel indices according to tree config */ switch (self->treeConfig) { case TREE_212: /* 212 */ lf = 0; rf = 1; break; default:; } /* Note: numInputChannels might not match the tree config ! */ switch (numInputChannels) { case 1: if (cf > 0) { FDKmemcpy(hybOutputReal[cf], hybInputReal[0], self->hybridBands * sizeof(FIXP_DBL)); FDKmemcpy(hybOutputImag[cf], hybInputImag[0], complexHybBands * sizeof(FIXP_DBL)); } else { FDKmemcpy(hybOutputReal[lf], hybInputReal[0], self->hybridBands * sizeof(FIXP_DBL)); FDKmemcpy(hybOutputReal[rf], hybInputReal[0], self->hybridBands * sizeof(FIXP_DBL)); FDKmemcpy(hybOutputImag[lf], hybInputImag[0], complexHybBands * sizeof(FIXP_DBL)); FDKmemcpy(hybOutputImag[rf], hybInputImag[0], complexHybBands * sizeof(FIXP_DBL)); } break; case 2: FDK_ASSERT(lf != -1); FDK_ASSERT(rf != -1); FDKmemcpy(hybOutputReal[lf], hybInputReal[0], self->hybridBands * sizeof(FIXP_DBL)); FDKmemcpy(hybOutputReal[rf], hybInputReal[1], self->hybridBands * sizeof(FIXP_DBL)); FDKmemcpy(hybOutputImag[lf], hybInputImag[0], complexHybBands * sizeof(FIXP_DBL)); FDKmemcpy(hybOutputImag[rf], hybInputImag[1], complexHybBands * sizeof(FIXP_DBL)); break; } for (ch = 0; ch < self->numOutputChannelsAT; ch++) { if (ch == lf || ch == rf || ch == cf) { continue; /* Skip bypassed channels */ } FDKmemclear(hybOutputReal[ch], self->hybridBands * sizeof(FIXP_DBL)); FDKmemclear(hybOutputImag[ch], complexHybBands * sizeof(FIXP_DBL)); } } } /******************************************************************************* Functionname: SpatialDecApplyParameterSets ******************************************************************************* Description: Arguments: Return: *******************************************************************************/ static SACDEC_ERROR SpatialDecApplyParameterSets( spatialDec *self, const SPATIAL_BS_FRAME *frame, SPATIALDEC_INPUT_MODE mode, PCM_MPS *inData, /* Time domain input */ FIXP_DBL **qmfInDataReal, /* QMF domain data l/r */ FIXP_DBL **qmfInDataImag, /* QMF domain data l/r */ UINT nSamples, UINT controlFlags, int numInputChannels, const FDK_channelMapDescr *const mapDescr) { SACDEC_ERROR err = MPS_OK; FIXP_SGL alpha; int ts; int ch; int hyb; int prevSlot = self->prevTimeSlot; int ps = self->curPs; int ts_io = 0; /* i/o dependent slot */ int bypassMode = (controlFlags & MPEGS_BYPASSMODE) ? 1 : 0; /* Bypass can be triggered by the upmixType, too. */ bypassMode |= ((self->upmixType == UPMIXTYPE_BYPASS) ? 1 : 0); /* * Decode available slots */ for (ts = self->curTimeSlot; ts <= fixMin(self->curTimeSlot + (int)nSamples / self->qmfBands - 1, self->timeSlots - 1); ts++, ts_io++) { int currSlot = frame->paramSlot[ps]; /* * Get new parameter set */ if (ts == prevSlot + 1) { err = SpatialDecCalculateM1andM2(self, ps, frame); /* input: ottCLD, ottICC, ... */ /* output: M1param(Real/Imag), M2(Real/Imag) */ if (err != MPS_OK) { bypassMode = 1; if (self->errInt == MPS_OK) { /* store internal error befor it gets overwritten */ self->errInt = err; } err = MPS_OK; } if ((ps == 0) && (self->bOverwriteM1M2prev != 0)) { /* copy matrix entries of M1/M2 of the first parameter set to the previous matrices (of the last frame). This avoids the interpolation of incompatible values. E.g. for residual bands the coefficients are calculated differently compared to non-residual bands. */ SpatialDecBufferMatrices(self); /* input: M(1/2)param(Real/Imag) */ /* output: M(1/2)param(Real/Imag)Prev */ self->bOverwriteM1M2prev = 0; } SpatialDecSmoothM1andM2( self, frame, ps); /* input: M1param(Real/Imag)(Prev), M2(Real/Imag)(Prev) */ /* output: M1param(Real/Imag), M2(Real/Imag) */ } alpha = FX_DBL2FX_SGL(fDivNorm(ts - prevSlot, currSlot - prevSlot)); switch (mode) { case INPUTMODE_QMF_SBR: if (self->pConfigCurrent->syntaxFlags & SACDEC_SYNTAX_LD) self->bShareDelayWithSBR = 0; /* We got no hybrid delay */ else self->bShareDelayWithSBR = 1; SpatialDecFeedQMF(self, qmfInDataReal, qmfInDataImag, ts_io, bypassMode, self->qmfInputReal__FDK, self->qmfInputImag__FDK, self->numInputChannels); break; case INPUTMODE_TIME: self->bShareDelayWithSBR = 0; SpatialDecQMFAnalysis(self, inData, ts_io, bypassMode, self->qmfInputReal__FDK, self->qmfInputImag__FDK, self->numInputChannels); break; default: break; } if ((self->pConfigCurrent->syntaxFlags & SACDEC_SYNTAX_USAC) && self->residualCoding) { int offset; ch = 1; offset = self->pQmfDomain->globalConf.nBandsSynthesis * self->pQmfDomain->globalConf.nQmfTimeSlots; { const PCM_MPS *inSamples = &inData[ts * self->pQmfDomain->globalConf.nBandsAnalysis]; CalculateSpaceAnalysisQmf( &self->pQmfDomain->QmfDomainIn[ch].fb, inSamples + (ch * offset), self->qmfResidualReal__FDK[0][0], self->qmfResidualImag__FDK[0][0]); if (!isTwoChMode(self->upmixType) && !bypassMode) { int i; FIXP_DBL *RESTRICT self_qmfResidualReal__FDK_0_0 = &self->qmfResidualReal__FDK[0][0][0]; FIXP_DBL *RESTRICT self_qmfResidualImag__FDK_0_0 = &self->qmfResidualImag__FDK[0][0][0]; if ((self->pQmfDomain->globalConf.nBandsAnalysis == 24) && !(self->stereoConfigIndex == 3)) { for (i = 0; i < self->qmfBands; i++) { self_qmfResidualReal__FDK_0_0[i] = fMult(scaleValueSaturate(self_qmfResidualReal__FDK_0_0[i], 1 + self->sacInDataHeadroom - (1)), self->clipProtectGain__FDK); self_qmfResidualImag__FDK_0_0[i] = fMult(scaleValueSaturate(self_qmfResidualImag__FDK_0_0[i], 1 + self->sacInDataHeadroom - (1)), self->clipProtectGain__FDK); } } else { for (i = 0; i < self->qmfBands; i++) { self_qmfResidualReal__FDK_0_0[i] = fMult(scaleValueSaturate(self_qmfResidualReal__FDK_0_0[i], self->sacInDataHeadroom - (1)), self->clipProtectGain__FDK); self_qmfResidualImag__FDK_0_0[i] = fMult(scaleValueSaturate(self_qmfResidualImag__FDK_0_0[i], self->sacInDataHeadroom - (1)), self->clipProtectGain__FDK); } } } } } SpatialDecHybridAnalysis( self, /* input: qmfInput(Real/Imag), qmfResidual(Real/Imag) */ self->qmfInputReal__FDK, self->qmfInputImag__FDK, self->hybInputReal__FDK, self->hybInputImag__FDK, ts, numInputChannels); if (bypassMode) { SpatialDecApplyBypass( self, self->hybInputReal__FDK, /* input: hybInput(Real/Imag) */ self->hybInputImag__FDK, self->hybOutputRealDry__FDK, /* output: hybOutput(Real/Imag)Dry */ self->hybOutputImagDry__FDK, numInputChannels); } else /* !bypassMode */ { FIXP_DBL *pxReal[MAX_NUM_XCHANNELS] = {NULL}; FIXP_DBL *pxImag[MAX_NUM_XCHANNELS] = {NULL}; SpatialDecCreateX(self, self->hybInputReal__FDK, /* input: hybInput(Real/Imag), hybResidual(Real/Imag) */ self->hybInputImag__FDK, pxReal, pxImag); { SpatialDecApplyM1_CreateW_Mode212( self, frame, pxReal, pxImag, self->wReal__FDK, /* output: w(Real/Imag) */ self->wImag__FDK); } if (err != MPS_OK) goto bail; int applyM2Config = APPLY_M2_NONE; applyM2Config = APPLY_M2; if ((self->pConfigCurrent->syntaxFlags & (SACDEC_SYNTAX_USAC | SACDEC_SYNTAX_RSVD50)) && (self->tempShapeConfig != 1) && (self->tempShapeConfig != 2)) { if (self->phaseCoding == 3) applyM2Config = APPLY_M2_MODE212_Res_PhaseCoding; else applyM2Config = APPLY_M2_MODE212; } switch (applyM2Config) { case APPLY_M2_MODE212: { err = SpatialDecApplyM2_Mode212( self, ps, alpha, self->wReal__FDK, self->wImag__FDK, self->hybOutputRealDry__FDK, self->hybOutputImagDry__FDK); } break; case APPLY_M2_MODE212_Res_PhaseCoding: err = SpatialDecApplyM2_Mode212_ResidualsPlusPhaseCoding( self, ps, alpha, self->wReal__FDK, self->wImag__FDK, self->hybOutputRealDry__FDK, self->hybOutputImagDry__FDK); break; case APPLY_M2: err = SpatialDecApplyM2( self, ps, alpha, self->wReal__FDK, self->wImag__FDK, self->hybOutputRealDry__FDK, self->hybOutputImagDry__FDK, self->hybOutputRealWet__FDK, self->hybOutputImagWet__FDK); break; default: err = MPS_APPLY_M2_ERROR; goto bail; } if (err != MPS_OK) goto bail; if ((self->tempShapeConfig == 2) && (!isTwoChMode(self->upmixType))) { SpatialDecReshapeBBEnv(self, frame, ts); /* input: reshapeBBEnvState, hybOutput(Real/Imag)(Dry/Wet), hybInput(Real/Imag) */ } /* output: hybOutput(Real/Imag)Dry */ /* Merge parts of the dry and wet QMF buffers. */ if ((self->tempShapeConfig == 1) && (!isTwoChMode(self->upmixType))) { for (ch = 0; ch < self->numOutputChannels; ch++) { for (hyb = 0; hyb < self->tp_hybBandBorder; hyb++) { self->hybOutputRealDry__FDK[ch][hyb] = fAddSaturate(self->hybOutputRealDry__FDK[ch][hyb], self->hybOutputRealWet__FDK[ch][hyb]); self->hybOutputImagDry__FDK[ch][hyb] = fAddSaturate(self->hybOutputImagDry__FDK[ch][hyb], self->hybOutputImagWet__FDK[ch][hyb]); } /* loop hyb */ } /* loop ch */ err = subbandTPApply( self, frame); /* input: hStpDec, hybOutput(Real/Imag)Dry/Wet */ /* output: hStpDec, hybOutput(Real/Imag)Dry */ if (err != MPS_OK) goto bail; } /* (self->tempShapeConfig == 1) */ else { /* The wet signal is added to the dry signal in applyM2 if GES and STP * are disabled */ if ((self->tempShapeConfig == 1) || (self->tempShapeConfig == 2)) { int nHybBands; nHybBands = self->hybridBands; for (ch = 0; ch < self->numOutputChannels; ch++) { FIXP_DBL *RESTRICT pRealDry = self->hybOutputRealDry__FDK[ch]; FIXP_DBL *RESTRICT pImagDry = self->hybOutputImagDry__FDK[ch]; FIXP_DBL *RESTRICT pRealWet = self->hybOutputRealWet__FDK[ch]; FIXP_DBL *RESTRICT pImagWet = self->hybOutputImagWet__FDK[ch]; for (hyb = 0; hyb < nHybBands; hyb++) { pRealDry[hyb] = fAddSaturate(pRealDry[hyb], pRealWet[hyb]); pImagDry[hyb] = fAddSaturate(pImagDry[hyb], pImagWet[hyb]); } /* loop hyb */ for (; hyb < self->hybridBands; hyb++) { pRealDry[hyb] = fAddSaturate(pRealDry[hyb], pRealWet[hyb]); } /* loop hyb */ } /* loop ch */ } /* ( self->tempShapeConfig == 1 ) || ( self->tempShapeConfig == 2 ) */ } /* !self->tempShapeConfig == 1 */ } /* !bypassMode */ if (self->phaseCoding == 1) { /* only if bsPhaseCoding == 1 and bsResidualCoding == 0 */ SpatialDecApplyPhase( self, alpha, (ts == currSlot) /* signal the last slot of the set */ ); } /* * Synthesis Filtering */ err = SpatialDecSynthesis( self, ts_io, self->hybOutputRealDry__FDK, /* input: hybOutput(Real/Imag)Dry */ self->hybOutputImagDry__FDK, self->timeOut__FDK, /* output: timeOut */ numInputChannels, mapDescr); if (err != MPS_OK) goto bail; /* * Update parameter buffer */ if (ts == currSlot) { SpatialDecBufferMatrices(self); /* input: M(1/2)param(Real/Imag) */ /* output: M(1/2)param(Real/Imag)Prev */ prevSlot = currSlot; ps++; } /* if (ts==currSlot) */ } /* ts loop */ /* * Save parameter states */ self->prevTimeSlot = prevSlot; self->curTimeSlot = ts; self->curPs = ps; bail: return err; } SACDEC_ERROR SpatialDecApplyFrame( spatialDec *self, SPATIAL_BS_FRAME *frame, /* parsed frame data to be applied */ SPATIALDEC_INPUT_MODE inputMode, PCM_MPS *inData, /* Time domain input */ FIXP_DBL **qmfInDataReal, /* QMF domain data l/r */ FIXP_DBL **qmfInDataImag, /* QMF domain data l/r */ PCM_MPS *pcmOutBuf, /* MAX_OUTPUT_CHANNELS*MAX_TIME_SLOTS*NUM_QMF_BANDS] */ UINT nSamples, UINT *pControlFlags, int numInputChannels, const FDK_channelMapDescr *const mapDescr) { SACDEC_ERROR err = MPS_OK; int fDecAndMapFrameData; int controlFlags; FDK_ASSERT(self != NULL); FDK_ASSERT(pControlFlags != NULL); FDK_ASSERT(pcmOutBuf != NULL); FDK_ASSERT(self->sacInDataHeadroom >= (1)); self->errInt = err; /* Init internal error */ controlFlags = *pControlFlags; if ((self->pConfigCurrent->syntaxFlags & SACDEC_SYNTAX_USAC) && (self->stereoConfigIndex > 1)) { numInputChannels = 1; /* Do not count residual channel as input channel. It is handled seperately. */ } /* Check if input amount of channels is consistent */ if (numInputChannels != self->numInputChannels) { controlFlags |= MPEGS_CONCEAL; if (numInputChannels > self->createParams.maxNumInputChannels) { return MPS_INVALID_PARAMETER; } } self->timeOut__FDK = pcmOutBuf; /* Determine local function control flags */ fDecAndMapFrameData = frame->newBsData; if (((fDecAndMapFrameData == 0) /* assures that conceal flag will not be set for blind mode */ && (self->curTimeSlot + (int)nSamples / self->qmfBands > self->timeSlots)) || (frame->numParameterSets == 0)) { /* New input samples but missing side info */ fDecAndMapFrameData = 1; controlFlags |= MPEGS_CONCEAL; } if ((fDecAndMapFrameData == 0) && (frame->paramSlot[fMax(0, frame->numParameterSets - 1)] != (self->timeSlots - 1) || self->curTimeSlot > frame->paramSlot[self->curPs])) { /* Detected faulty parameter slot data. */ fDecAndMapFrameData = 1; controlFlags |= MPEGS_CONCEAL; } /* Update concealment state machine */ SpatialDecConcealment_UpdateState( &self->concealInfo, (controlFlags & MPEGS_CONCEAL) ? 0 : 1); /* convert from conceal flag to frame ok flag */ if (fDecAndMapFrameData) { /* Reset spatial framing control vars */ frame->newBsData = 0; self->prevTimeSlot = -1; self->curTimeSlot = 0; self->curPs = 0; if (controlFlags & MPEGS_CONCEAL) { /* Reset frame data to avoid misconfiguration. */ SpatialDecClearFrameData(self, frame, &self->createParams); } { err = SpatialDecDecodeFrame(self, frame); /* input: ... */ /* output: decodeAndMapFrameDATA */ } if (err != MPS_OK) { /* Rescue strategy is to apply bypass mode in order to keep at least the downmix channels continuous. */ controlFlags |= MPEGS_CONCEAL; if (self->errInt == MPS_OK) { /* store internal error befor it gets overwritten */ self->errInt = err; } } } err = SpatialDecApplyParameterSets( self, frame, inputMode, inData, qmfInDataReal, qmfInDataImag, nSamples, controlFlags | ((err == MPS_OK) ? 0 : MPEGS_BYPASSMODE), numInputChannels, mapDescr); if (err != MPS_OK) { goto bail; } bail: *pControlFlags = controlFlags; return err; } fdk-aac-2.0.2/libSACdec/src/sac_dec.h000066400000000000000000000447211404225310700171010ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround decoder library ************************* Author(s): Description: SAC Decoder Library structures *******************************************************************************/ #ifndef SAC_DEC_H #define SAC_DEC_H #include "common_fix.h" #include "sac_dec_interface.h" /* library interface in ../include */ #include "FDK_qmf_domain.h" #include "sac_qmf.h" #include "FDK_bitstream.h" /* mp4 bitbuffer */ #include "sac_calcM1andM2.h" #include "FDK_hybrid.h" #include "FDK_decorrelate.h" #include "sac_reshapeBBEnv.h" #include "sac_dec_conceal.h" #include "sac_tsd.h" #ifndef MAX #define MAX(a, b) ((a) > (b) ? (a) : (b)) #endif #define ICCdefault 0 #define IPDdefault 0 #define arbdmxGainDefault 0 #define CPCdefault 10 #define tttCLD1default 15 #define tttCLD2default 0 #define IS_HQ_ONLY(aot) \ ((aot) == AOT_ER_AAC_LD || (aot) == AOT_ER_AAC_ELD || (aot) == AOT_USAC || \ (aot) == AOT_RSVD50) #define SCONST(x) FL2FXCONST_DBL(x) #define PC_NUM_BANDS (8) #define PC_NUM_HYB_BANDS (PC_NUM_BANDS - 3 + 10) #define ABS_THR (1e-9f * 32768 * 32768) #define MAX_HYBRID_BANDS (MAX_NUM_QMF_BANDS - 3 + 10) #define HYBRID_FILTER_DELAY (6) #define MAX_RESIDUAL_FRAMES (4) #define MAX_RESIDUAL_BISTREAM \ (836) /* 48000 bps * 3 res / (8 * 44100 / 2048 ) */ #define MAX_MDCT_COEFFS (1024) #define SACDEC_RESIDUAL_BS_BUF_SIZE \ (1024) /* used to setup and check residual bitstream buffer */ #define MAX_NUM_PARAMS (MAX_NUM_OTT + 4 * MAX_NUM_TTT + MAX_INPUT_CHANNELS) #define MAX_NUM_PARAMETERS (MAX(MAX_NUM_PARAMS, MAX_NUM_OTT)) #define MAX_PARAMETER_SETS (9) #define MAX_M2_INPUT (MAX_OUTPUT_CHANNELS) /* 3 direct + 5 diffuse */ #define MAX_QMF_BANDS_TO_HYBRID \ (3) /* 3 bands are filtered again in "40 bands" case */ #define PROTO_LEN (13) #define BUFFER_LEN_LF (PROTO_LEN) #define BUFFER_LEN_HF ((PROTO_LEN - 1) / 2) #define MAX_NO_DECORR_CHANNELS (MAX_OUTPUT_CHANNELS) #define HRTF_AZIMUTHS (5) #define MAX_NUM_OTT_AT 0 /* left out */ typedef enum { UPMIXTYPE_BYPASS = -1, /*just bypass the input channels without processing*/ UPMIXTYPE_NORMAL = 0 /*multichannel loudspeaker upmix with spatial data*/ } UPMIXTYPE; static inline int isTwoChMode(UPMIXTYPE upmixType) { int retval = 0; return retval; } /* left out end */ #define MPEGS_BYPASSMODE (0x00000001) #define MPEGS_CONCEAL (0x00000002) typedef struct STP_DEC *HANDLE_STP_DEC; typedef struct { SCHAR bsQuantCoarseXXXprev; SCHAR bsQuantCoarseXXXprevParse; } LOSSLESSSTATE; typedef struct { SCHAR bsXXXDataMode[MAX_PARAMETER_SETS]; SCHAR bsQuantCoarseXXX[MAX_PARAMETER_SETS]; SCHAR bsFreqResStrideXXX[MAX_PARAMETER_SETS]; SCHAR nocmpQuantCoarseXXX[MAX_PARAMETER_SETS]; LOSSLESSSTATE *state; /* Link to persistent state information */ } LOSSLESSDATA; struct SPATIAL_BS_FRAME_struct { UCHAR bsIndependencyFlag; UCHAR newBsData; UCHAR numParameterSets; /* If bsFramingType == 0, then the paramSlot[ps] for 0 <= ps < numParamSets is calculated as follows: paramSlot[ps] = ceil(numSlots*(ps+1)/numParamSets) - 1 Otherwise, it is paramSlot[ps] = bsParamSlot[ps] */ INT paramSlot[MAX_PARAMETER_SETS]; /* These arrays contain the compact indices, only one value per pbstride, only * paramsets actually containing data. */ /* These values are written from the parser in ecDataDec() and read during * decode in mapIndexData() */ SCHAR cmpOttCLDidx[MAX_NUM_OTT + MAX_NUM_OTT_AT][MAX_PARAMETER_SETS] [MAX_PARAMETER_BANDS]; SCHAR cmpOttICCidx[MAX_NUM_OTT][MAX_PARAMETER_SETS][MAX_PARAMETER_BANDS]; /* Smoothing */ UCHAR bsSmoothMode[MAX_PARAMETER_SETS]; UCHAR bsSmoothTime[MAX_PARAMETER_SETS]; UCHAR bsFreqResStrideSmg[MAX_PARAMETER_SETS]; UCHAR bsSmgData[MAX_PARAMETER_SETS] [MAX_PARAMETER_BANDS]; /* smoothing flags, one if band is smoothed, otherwise zero */ /* Arbitrary Downmix */ SCHAR (*cmpArbdmxGainIdx)[MAX_PARAMETER_SETS][MAX_PARAMETER_BANDS]; /* Lossless control */ LOSSLESSDATA *CLDLosslessData; LOSSLESSDATA *ICCLosslessData; /* LOSSLESSDATA *ADGLosslessData; -> is stored in CLDLosslessData[offset] */ LOSSLESSDATA *IPDLosslessData; SCHAR (*cmpOttIPDidx)[MAX_PARAMETER_SETS][MAX_PARAMETER_BANDS]; int phaseMode; int OpdSmoothingMode; UCHAR tempShapeEnableChannelGES[MAX_OUTPUT_CHANNELS]; /*!< GES side info. */ UCHAR bsEnvShapeData[MAX_OUTPUT_CHANNELS] [MAX_TIME_SLOTS]; /*!< GES side info (quantized). */ UCHAR tempShapeEnableChannelSTP[MAX_OUTPUT_CHANNELS]; /*!< STP side info. */ TSD_DATA TsdData[1]; /*!< TSD data structure. */ }; typedef struct { /* Lossless state */ LOSSLESSSTATE CLDLosslessState[MAX_NUM_PARAMETERS]; LOSSLESSSTATE ICCLosslessState[MAX_NUM_PARAMETERS]; LOSSLESSSTATE IPDLosslessState[MAX_NUM_PARAMETERS]; } BS_LL_STATE; typedef struct { int prevParamSlot; int prevSmgTime; UCHAR prevSmgData[MAX_PARAMETER_BANDS]; FIXP_DBL opdLeftState__FDK[MAX_PARAMETER_BANDS]; FIXP_DBL opdRightState__FDK[MAX_PARAMETER_BANDS]; } SMOOTHING_STATE; typedef struct { FIXP_DBL alpha__FDK; FIXP_DBL beta__FDK; FIXP_DBL partNrgPrev__FDK[2 * MAX_OUTPUT_CHANNELS + MAX_INPUT_CHANNELS] [BB_ENV_SIZE]; FIXP_DBL normNrgPrev__FDK[2 * MAX_OUTPUT_CHANNELS + MAX_INPUT_CHANNELS]; FIXP_DBL frameNrgPrev__FDK[2 * MAX_OUTPUT_CHANNELS + MAX_INPUT_CHANNELS]; INT partNrgPrevSF[2 * MAX_OUTPUT_CHANNELS + MAX_INPUT_CHANNELS]; INT partNrgPrev2SF[2 * MAX_OUTPUT_CHANNELS + MAX_INPUT_CHANNELS]; INT normNrgPrevSF[2 * MAX_OUTPUT_CHANNELS + MAX_INPUT_CHANNELS]; INT frameNrgPrevSF[2 * MAX_OUTPUT_CHANNELS + MAX_INPUT_CHANNELS]; } RESHAPE_BBENV_STATE; typedef struct { int maxNumInputChannels; int maxNumOutputChannels; int maxNumQmfBands; int maxNumHybridBands; int maxNumXChannels; int maxNumVChannels; int maxNumDecorChannels; int maxNumCmplxQmfBands; int maxNumCmplxHybBands; int maxNumResChannels; int bProcResidual; /* process residual */ int maxNumResidualChannels; int maxNumOttBoxes; int maxNumParams; } SACDEC_CREATION_PARAMS; struct spatialDec_struct { SACDEC_ERROR errInt; /* Field to store internal errors. Will be clear at the very beginning of each process call. */ int staticDecScale; /* static scale of decoder */ /* GENERAL */ int samplingFreq; /* [Hz] */ CFG_LEVEL decoderLevel; /* 0..5 */ CFG_EXTENT decoderMode; CFG_BINAURAL binauralMode; SACDEC_CREATION_PARAMS createParams; int numComplexProcessingBands; int treeConfig; /* TREE_5151 = 5151, TREE_5152 = 5152, TREE_525 = 525, defined in sac_bitdec.h */ int numInputChannels; /* 1 (M) or 2 (L,R) */ int numOutputChannels; /* 6 for 3/2.1 (FL,FR,FC,LF,BL,BR) */ int numOttBoxes; /* number of ott boxes */ int numM2rows; int numOutputChannelsAT; /* Number of output channels after arbitrary tree processing */ int quantMode; /* QUANT_FINE, QUANT_EBQ1, QUANT_EBQ2, defined in sac_bitdec.h */ int arbitraryDownmix; /* (arbitraryDownmix != 0) 1 arbitrary downmix data present, 2 arbitrary downmix residual data present*/ int residualCoding; /* (residualCoding != 0) => residual coding data present */ UCHAR nrResidualFrame; UCHAR nrArbDownmixResidualFrame; FDK_BITSTREAM **hResidualBitstreams; int tempShapeConfig; /* */ int decorrType; /* Indicates to use PS or none PS decorrelator. */ int decorrConfig; /* chosen decorrelator */ int envQuantMode; /* quantization mode of envelope reshaping data */ FIXP_DBL clipProtectGain__FDK; /* global gain for upmix */ char clipProtectGainSF__FDK; /* global gain for upmix */ /* Currently ignoring center decorr numVChannels = numDirektSignals + numDecorSignals */ int numDirektSignals; /* needed for W, Number of direkt signals 515 -> 1 525 -> 3 */ int wStartResidualIdx; /* Where to start read residuals for W, = 0 for 515, = 1 for 525 since one residual is used in V */ int numDecorSignals; /* needed for W, Number of residual and decorrelated signals, = 2, 3 for center deccorelation*/ int numVChannels; /* direct signals + decorelator signals */ int numXChannels; /* direct input signals + TTT-residuals */ int timeSlots; /* length of spatial frame in QMF samples */ int curTimeSlot; /* pointer to the current time slot used for hyperframing */ int prevTimeSlot; /* */ int curPs; int frameLength; /* number of output waveform samples/channel/frame */ UPMIXTYPE upmixType; int partiallyComplex; int useFDreverb; int bShareDelayWithSBR; int tp_hybBandBorder; /* Hybrid band indicating the HP filter cut-off. */ /* FREQUENCY MAPPING */ int qmfBands; int hybridBands; const SCHAR *kernels; /* Mapping hybrid band to parameter band. */ int TsdTs; /**< TSD QMF slot counter 0<= ts < numSlots */ int *param2hyb; /* Mapping parameter bands to hybrid bands */ int kernels_width[MAX_PARAMETER_BANDS]; /* Mapping parmeter band to hybrid band offsets. */ /* Residual coding */ int residualSamplingFreq; UCHAR residualPresent[MAX_NUM_OTT + MAX_NUM_TTT]; UCHAR residualBands[MAX_NUM_OTT + MAX_NUM_TTT]; /* 0, if no residual data present for this box */ UCHAR residualQMFBands[MAX_NUM_OTT + MAX_NUM_TTT]; /* needed for optimized mdct2qmf calculation */ SPATIAL_SPECIFIC_CONFIG *pConfigCurrent; int arbdmxFramesPerSpatialFrame; int arbdmxUpdQMF; int numParameterBands; /* Number of parameter bands 40, 28, 20, 14, 10, ... .*/ int bitstreamParameterBands; int *numOttBands; /* number of bands for each ott, is != numParameterBands for LFEs */ /* 1 MAPPING */ UCHAR extendFrame; UCHAR numParameterSetsPrev; int *smgTime; UCHAR **smgData; /* PARAMETER DATA decoded and dequantized */ /* Last parameters from prev frame required during decode in mapIndexData() * and not touched during parse */ SCHAR **ottCLDidxPrev; SCHAR **ottICCidxPrev; SCHAR **arbdmxGainIdxPrev; SCHAR **ottIPDidxPrev; SCHAR ***outIdxData; /* is this really persistent memory ? */ /* State mem required during parse in SpatialDecParseFrameData() */ SCHAR **cmpOttCLDidxPrev; SCHAR **cmpOttICCidxPrev; SCHAR ***ottICCdiffidx; SCHAR **cmpOttIPDidxPrev; /* State mem required in parseArbitraryDownmixData */ SCHAR **cmpArbdmxGainIdxPrev; SCHAR ***ottCLD__FDK; SCHAR ***ottICC__FDK; SCHAR ***arbdmxGain__FDK; /* Holds the artistic downmix correction index.*/ FIXP_DBL *arbdmxAlpha__FDK; FIXP_DBL *arbdmxAlphaPrev__FDK; UCHAR stereoConfigIndex; int highRateMode; int phaseCoding; SCHAR ***ottIPD__FDK; FIXP_DBL PhaseLeft__FDK[MAX_PARAMETER_BANDS]; FIXP_DBL PhaseRight__FDK[MAX_PARAMETER_BANDS]; FIXP_DBL PhasePrevLeft__FDK[MAX_PARAMETER_BANDS]; FIXP_DBL PhasePrevRight__FDK[MAX_PARAMETER_BANDS]; int numOttBandsIPD; /* GAIN MATRICIES FOR CURRENT and PREVIOUS PARMATER SET(s)*/ FIXP_DBL ***M2Real__FDK; FIXP_DBL ***M2Imag__FDK; FIXP_DBL ***M2RealPrev__FDK; FIXP_DBL ***M2ImagPrev__FDK; /* INPUT SIGNALS */ FIXP_DBL ***qmfInputRealDelayBuffer__FDK; FIXP_DBL ***qmfInputImagDelayBuffer__FDK; int pc_filterdelay; /* additional delay to align HQ with LP before hybird analysis */ int qmfInputDelayBufPos; FIXP_DBL **qmfInputReal__FDK; FIXP_DBL **qmfInputImag__FDK; FIXP_DBL **hybInputReal__FDK; FIXP_DBL **hybInputImag__FDK; FIXP_DBL **binInputReverb; FIXP_DBL binGain, reverbGain; FIXP_DBL binCenterGain, reverbCenterGain; /* RESIDUAL SIGNALS */ FIXP_DBL ***qmfResidualReal__FDK; FIXP_DBL ***qmfResidualImag__FDK; FIXP_DBL **hybResidualReal__FDK; FIXP_DBL **hybResidualImag__FDK; int qmfOutputRealDryDelayBufPos; FIXP_DBL ***qmfOutputRealDryDelayBuffer__FDK; FIXP_DBL ***qmfOutputImagDryFilterBuffer__FDK; FIXP_DBL *qmfOutputImagDryFilterBufferBase__FDK; /* TEMPORARY SIGNALS */ FIXP_DBL **wReal__FDK; FIXP_DBL **wImag__FDK; /* OUTPUT SIGNALS */ FIXP_DBL **hybOutputRealDry__FDK; FIXP_DBL **hybOutputImagDry__FDK; FIXP_DBL **hybOutputRealWet__FDK; FIXP_DBL **hybOutputImagWet__FDK; PCM_MPS *timeOut__FDK; HANDLE_FDK_QMF_DOMAIN pQmfDomain; FDK_ANA_HYB_FILTER *hybridAnalysis; /*!< pointer Analysis hybrid filterbank array. */ FDK_SYN_HYB_FILTER *hybridSynthesis; /*!< pointer Synthesis hybrid filterbank array. */ FIXP_DBL ** pHybridAnaStatesLFdmx; /*!< pointer to analysis hybrid filter states LF */ FIXP_DBL ** pHybridAnaStatesHFdmx; /*!< pointer to analysis hybrid filter states HF */ FIXP_DBL ** pHybridAnaStatesLFres; /*!< pointer to analysis hybrid filter states LF */ FIXP_DBL ** pHybridAnaStatesHFres; /*!< pointer to analysis hybrid filter states HF */ DECORR_DEC *apDecor; /*!< pointer decorrelator array. */ FIXP_DBL **pDecorBufferCplx; SMOOTHING_STATE *smoothState; /*!< Pointer to smoothing states. */ RESHAPE_BBENV_STATE *reshapeBBEnvState; /*!< GES handle. */ SCHAR row2channelDmxGES[MAX_OUTPUT_CHANNELS]; HANDLE_STP_DEC hStpDec; /*!< STP handle. */ const UCHAR *pActivM2ParamBands; int bOverwriteM1M2prev; /* Overwrite previous M2/M2 params with first set of new frame after SSC change (aka decodeAfterConfigHasChangedFlag). */ SpatialDecConcealmentInfo concealInfo; INT sacInDataHeadroom; /* Headroom of the SAC input time signal to prevent clipping */ }; #define SACDEC_SYNTAX_MPS 1 #define SACDEC_SYNTAX_USAC 2 #define SACDEC_SYNTAX_RSVD50 4 #define SACDEC_SYNTAX_L2 8 #define SACDEC_SYNTAX_L3 16 #define SACDEC_SYNTAX_LD 32 static inline int GetProcBand(spatialDec_struct *self, int qs) { return self->kernels[qs]; } #endif /* SAC_DEC_H */ fdk-aac-2.0.2/libSACdec/src/sac_dec_conceal.cpp000066400000000000000000000322411404225310700211120ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround decoder library ************************* Author(s): Christian Ertel, Christian Griebel Description: SAC Dec error concealment *******************************************************************************/ #include "sac_dec_conceal.h" void SpatialDecConcealment_Init(SpatialDecConcealmentInfo *info, const UINT resetFlags) { FDK_ASSERT(info != NULL); if (resetFlags & MPEGS_CONCEAL_RESET_STATE) { info->concealState = SpatialDecConcealState_Init; /* Frame counters will be initialized implicitely in function * SpatialDecConcealment_UpdateState(). */ } if (resetFlags & MPEGS_CONCEAL_RESET_PARAMETER) { /* Set default params */ info->concealParams.method = MPEGS_CONCEAL_DEFAULT_METHOD; info->concealParams.numKeepFrames = MPEGS_CONCEAL_DEFAULT_NUM_KEEP_FRAMES; info->concealParams.numFadeOutFrames = MPEGS_CONCEAL_DEFAULT_FADE_OUT_SLOPE_LENGTH; info->concealParams.numFadeInFrames = MPEGS_CONCEAL_DEFAULT_FADE_IN_SLOPE_LENGTH; info->concealParams.numReleaseFrames = MPEGS_CONCEAL_DEFAULT_NUM_RELEASE_FRAMES; } return; } int SpatialDecConcealment_Apply( SpatialDecConcealmentInfo *info, const SCHAR (*cmpIdxData)[MAX_PARAMETER_BANDS], SCHAR **diffIdxData, SCHAR * idxPrev, /* char idxPrev[SPATIALDEC_MAX_NUM_OTT][SPATIALDEC_MAX_PARAMETER_BANDS], */ SCHAR *bsXXXDataMode, const int startBand, const int stopBand, const SCHAR defaultValue, const int paramType, const int numParamSets) { int appliedProcessing = 0; int band, dataMode = -1; FDK_ASSERT(info != NULL); FDK_ASSERT(cmpIdxData != NULL); FDK_ASSERT(idxPrev != NULL); FDK_ASSERT(bsXXXDataMode != NULL); /* Processing depends only on the internal state */ switch (info->concealState) { case SpatialDecConcealState_Init: dataMode = 0; /* default */ break; case SpatialDecConcealState_Ok: /* Nothing to do */ break; case SpatialDecConcealState_Keep: dataMode = 1; /* keep */ break; case SpatialDecConcealState_FadeToDefault: { /* Start simple fade out */ FIXP_DBL fac = fDivNorm(info->cntStateFrames + 1, info->concealParams.numFadeOutFrames + 1); for (band = startBand; band < stopBand; band += 1) { /* idxPrev = fac * defaultValue + (1-fac) * idxPrev; */ idxPrev[band] = fMultI(fac, defaultValue - idxPrev[band]) + idxPrev[band]; } dataMode = 1; /* keep */ appliedProcessing = 1; } break; case SpatialDecConcealState_Default: for (band = startBand; band < stopBand; band += 1) { idxPrev[band] = defaultValue; } dataMode = 1; /* keep */ appliedProcessing = 1; break; case SpatialDecConcealState_FadeFromDefault: { FIXP_DBL fac = fDivNorm(info->cntValidFrames + 1, info->concealParams.numFadeInFrames + 1); for (band = startBand; band < stopBand; band += 1) { /* idxPrev = fac * cmpIdxData + (1-fac) * defaultValue; */ idxPrev[band] = fMultI(fac, cmpIdxData[numParamSets - 1][band] - defaultValue) + defaultValue; } dataMode = 1; /* keep */ appliedProcessing = 1; } break; default: FDK_ASSERT(0); /* All valid states shall be handled above. */ break; } if (dataMode >= 0) { int i; for (i = 0; i < numParamSets; i += 1) { bsXXXDataMode[i] = dataMode; if (diffIdxData != NULL) { for (band = startBand; band < stopBand; band += 1) { diffIdxData[i][band] = 0; } } } } return appliedProcessing; } void SpatialDecConcealment_UpdateState(SpatialDecConcealmentInfo *info, const int frameOk) { FDK_ASSERT(info != NULL); if (frameOk) { info->cntValidFrames += 1; } else { info->cntValidFrames = 0; } switch (info->concealState) { case SpatialDecConcealState_Init: if (frameOk) { /* NEXT STATE: Ok */ info->concealState = SpatialDecConcealState_Ok; info->cntStateFrames = 0; } break; case SpatialDecConcealState_Ok: if (!frameOk) { /* NEXT STATE: Keep */ info->concealState = SpatialDecConcealState_Keep; info->cntStateFrames = 0; } break; case SpatialDecConcealState_Keep: info->cntStateFrames += 1; if (frameOk) { /* NEXT STATE: Ok */ info->concealState = SpatialDecConcealState_Ok; } else { if (info->cntStateFrames >= info->concealParams.numKeepFrames) { if (info->concealParams.numFadeOutFrames == 0) { /* NEXT STATE: Default */ info->concealState = SpatialDecConcealState_Default; } else { /* NEXT STATE: Fade to default */ info->concealState = SpatialDecConcealState_FadeToDefault; info->cntStateFrames = 0; } } } break; case SpatialDecConcealState_FadeToDefault: info->cntStateFrames += 1; if (info->cntValidFrames > 0) { /* NEXT STATE: Fade in from default */ info->concealState = SpatialDecConcealState_FadeFromDefault; info->cntStateFrames = 0; } else { if (info->cntStateFrames >= info->concealParams.numFadeOutFrames) { /* NEXT STATE: Default */ info->concealState = SpatialDecConcealState_Default; } } break; case SpatialDecConcealState_Default: if (info->cntValidFrames > 0) { if (info->concealParams.numFadeInFrames == 0) { /* NEXT STATE: Ok */ info->concealState = SpatialDecConcealState_Ok; } else { /* NEXT STATE: Fade in from default */ info->concealState = SpatialDecConcealState_FadeFromDefault; info->cntValidFrames = 0; } } break; case SpatialDecConcealState_FadeFromDefault: info->cntValidFrames += 1; if (frameOk) { if (info->cntValidFrames >= info->concealParams.numFadeInFrames) { /* NEXT STATE: Ok */ info->concealState = SpatialDecConcealState_Ok; } } else { /* NEXT STATE: Fade to default */ info->concealState = SpatialDecConcealState_FadeToDefault; info->cntStateFrames = 0; } break; default: FDK_ASSERT(0); /* All valid states should be handled above! */ break; } } SACDEC_ERROR SpatialDecConcealment_SetParam(SpatialDecConcealmentInfo *self, const SAC_DEC_CONCEAL_PARAM param, const INT value) { SACDEC_ERROR err = MPS_OK; switch (param) { case SAC_DEC_CONCEAL_METHOD: switch ((SpatialDecConcealmentMethod)value) { case SAC_DEC_CONCEAL_WITH_ZERO_VALUED_OUTPUT: case SAC_DEC_CONCEAL_BY_FADING_PARAMETERS: break; default: err = MPS_INVALID_PARAMETER; goto bail; } if (self != NULL) { /* store parameter value */ self->concealParams.method = (SpatialDecConcealmentMethod)value; } else { err = MPS_INVALID_HANDLE; goto bail; } break; case SAC_DEC_CONCEAL_NUM_KEEP_FRAMES: if (value < 0) { err = MPS_INVALID_PARAMETER; goto bail; } if (self != NULL) { /* store parameter value */ self->concealParams.numKeepFrames = (UINT)value; } else { err = MPS_INVALID_HANDLE; goto bail; } break; case SAC_DEC_CONCEAL_FADE_OUT_SLOPE_LENGTH: if (value < 0) { err = MPS_INVALID_PARAMETER; goto bail; } if (self != NULL) { /* store parameter value */ self->concealParams.numFadeOutFrames = (UINT)value; } else { err = MPS_INVALID_HANDLE; goto bail; } break; case SAC_DEC_CONCEAL_FADE_IN_SLOPE_LENGTH: if (value < 0) { err = MPS_INVALID_PARAMETER; goto bail; } if (self != NULL) { /* store parameter value */ self->concealParams.numFadeInFrames = (UINT)value; } else { err = MPS_INVALID_HANDLE; goto bail; } break; case SAC_DEC_CONCEAL_NUM_RELEASE_FRAMES: if (value < 0) { err = MPS_INVALID_PARAMETER; goto bail; } if (self != NULL) { /* store parameter value */ self->concealParams.numReleaseFrames = (UINT)value; } else { err = MPS_INVALID_HANDLE; goto bail; } break; default: err = MPS_INVALID_PARAMETER; goto bail; } bail: return err; } fdk-aac-2.0.2/libSACdec/src/sac_dec_conceal.h000066400000000000000000000172101404225310700205560ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround decoder library ************************* Author(s): Christian Ertel, Christian Griebel Description: SAC Dec error concealment *******************************************************************************/ #ifndef SAC_DEC_CONCEAL_H #define SAC_DEC_CONCEAL_H #include "sac_dec_interface.h" /* Modules dynamic parameters: */ typedef enum { SAC_DEC_CONCEAL_METHOD = 0, SAC_DEC_CONCEAL_NUM_KEEP_FRAMES, SAC_DEC_CONCEAL_FADE_OUT_SLOPE_LENGTH, SAC_DEC_CONCEAL_FADE_IN_SLOPE_LENGTH, SAC_DEC_CONCEAL_NUM_RELEASE_FRAMES } SAC_DEC_CONCEAL_PARAM; /* - - - - - - - - - - - - - - - - - - - - - - - - - - */ /* sac_dec_interface.h */ /* - - - - - - - - - - - - - - - - - - - - - - - - - - */ typedef enum { SAC_DEC_CONCEAL_WITH_ZERO_VALUED_OUTPUT = 0, SAC_DEC_CONCEAL_BY_FADING_PARAMETERS = 1 } SpatialDecConcealmentMethod; /* - - - - - - - - - - - - - - - - - - - - - - - - - - */ /* Default dynamic parameter values: */ #define MPEGS_CONCEAL_DEFAULT_METHOD SAC_DEC_CONCEAL_BY_FADING_PARAMETERS #define MPEGS_CONCEAL_DEFAULT_NUM_KEEP_FRAMES (10) #define MPEGS_CONCEAL_DEFAULT_FADE_OUT_SLOPE_LENGTH (5) #define MPEGS_CONCEAL_DEFAULT_FADE_IN_SLOPE_LENGTH (5) #define MPEGS_CONCEAL_DEFAULT_NUM_RELEASE_FRAMES (3) typedef enum { SpatialDecConcealState_Init = 0, SpatialDecConcealState_Ok, SpatialDecConcealState_Keep, SpatialDecConcealState_FadeToDefault, SpatialDecConcealState_Default, SpatialDecConcealState_FadeFromDefault } SpatialDecConcealmentState; typedef struct { SpatialDecConcealmentMethod method; UINT numKeepFrames; UINT numFadeOutFrames; UINT numFadeInFrames; UINT numReleaseFrames; } SpatialDecConcealmentParams; typedef struct { SpatialDecConcealmentParams concealParams; /* User set params */ SpatialDecConcealmentState concealState; /* State of internal state machine (fade-in/out etc) */ UINT cntStateFrames; /* Counter for fade-in/out handling */ UINT cntValidFrames; /* Counter for the number of consecutive good frames*/ } SpatialDecConcealmentInfo; /* Module reset flags */ #define MPEGS_CONCEAL_RESET_STATE (0x01) #define MPEGS_CONCEAL_RESET_PARAMETER (0x02) #define MPEGS_CONCEAL_RESET_ALL (0xFF) void SpatialDecConcealment_Init(SpatialDecConcealmentInfo *info, const UINT resetFlags); int SpatialDecConcealment_Apply(SpatialDecConcealmentInfo *info, const SCHAR (*cmpIdxData)[MAX_PARAMETER_BANDS], SCHAR **diffIdxData, SCHAR *idxPrev, SCHAR *bsXXXDataMode, const int startBand, const int stopBand, const SCHAR defaultValue, const int paramType, const int numParamSets); void SpatialDecConcealment_UpdateState(SpatialDecConcealmentInfo *info, const int frameOk); SACDEC_ERROR SpatialDecConcealment_SetParam(SpatialDecConcealmentInfo *info, const SAC_DEC_CONCEAL_PARAM param, const INT value); #endif /* SAC_DEC_CONCEAL_H */ fdk-aac-2.0.2/libSACdec/src/sac_dec_interface.h000066400000000000000000000310721404225310700211140ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround decoder library ************************* Author(s): Description: SAC Decoder Library Interface *******************************************************************************/ #ifndef SAC_DEC_INTERFACE_H #define SAC_DEC_INTERFACE_H #include "common_fix.h" #include "FDK_audio.h" #ifdef __cplusplus extern "C" { #endif #include "sac_dec_errorcodes.h" #include "sac_dec_ssc_struct.h" /** * \brief Baseline MPEG-Surround profile Level 1-5. */ typedef enum { DECODER_LEVEL_0 = 0, /*!< Level 0: dummy level; 212 only */ DECODER_LEVEL_6 = 6 /*!< Level 6: no support */ } CFG_LEVEL; /* * \brief Number of output channels restriction. */ typedef enum { OUTPUT_CHANNELS_DEFAULT, /*!< Default configuration depending on Decoder Level */ OUTPUT_CHANNELS_2_0, /*!< Limitation to stereo output */ OUTPUT_CHANNELS_5_1 /*!< Limitation to 5.1 output */ } CFG_RESTRICTION; /* * \brief Supported decoder mode. */ typedef enum { EXT_HQ_ONLY = 0, /*!< High Quality processing only */ EXT_LP_ONLY = 1, /*!< Low Power procesing only */ EXT_HQ_AND_LP = 2 /*!< Support both HQ and LP processing */ } CFG_EXTENT; /* * \brief Supported binaural mode. */ typedef enum { BINAURAL_NONE = -1 /*!< No binaural procesing supported */ } CFG_BINAURAL; /** * \brief Decoder configuration structure. * * These structure contains all parameters necessary for decoder open function. * The configuration specifies the functional range of the decoder instance. */ typedef struct { CFG_LEVEL decoderLevel; CFG_EXTENT decoderMode; CFG_RESTRICTION maxNumOutputChannels; CFG_BINAURAL binauralMode; } SPATIAL_DEC_CONFIG; typedef enum { INPUTMODE_QMF = 1000, INPUTMODE_QMF_SBR = 1001, INPUTMODE_TIME = 1002 } SPATIALDEC_INPUT_MODE; /** * \brief MPEG Surround upmix type mode. **/ typedef enum { UPMIX_TYPE_BYPASS = -1, /*!< Bypass the downmix channels from the core decoder. */ UPMIX_TYPE_NORMAL = 0 /*!< Multi channel output. */ } SPATIAL_DEC_UPMIX_TYPE; /** * \brief Dynamic decoder parameters. */ typedef struct { /* Basics */ UCHAR outputMode; UCHAR blindEnable; UCHAR bypassMode; /* Error concealment */ UCHAR concealMethod; UINT concealNumKeepFrames; UINT concealFadeOutSlopeLength; UINT concealFadeInSlopeLength; UINT concealNumReleaseFrames; } SPATIALDEC_PARAM; /** * \brief Flags which control the initialization **/ typedef enum { MPEGS_INIT_NONE = 0x00000000, /*!< Indicates no initialization */ MPEGS_INIT_CONFIG = 0x00000010, /*!< Indicates a configuration change due to SSC value changes */ MPEGS_INIT_STATES_ANA_QMF_FILTER = 0x00000100, /*!< Controls the initialization of the analysis qmf filter states */ MPEGS_INIT_STATES_SYN_QMF_FILTER = 0x00000200, /*!< Controls the initialization of the synthesis qmf filter states */ MPEGS_INIT_STATES_ANA_HYB_FILTER = 0x00000400, /*!< Controls the initialization of the analysis hybrid filter states */ MPEGS_INIT_STATES_DECORRELATOR = 0x00000800, /*!< Controls the initialization of the decorrelator states */ MPEGS_INIT_STATES_M1M2 = 0x00002000, /*!< Controls the initialization of the history in m1 and m2 parameter calculation */ MPEGS_INIT_STATES_GES = 0x00004000, /*!< Controls the initialization of the history in the ges calculation */ MPEGS_INIT_STATES_REVERB = 0x00008000, /*!< Controls the initialization of the reverb states */ MPEGS_INIT_STATES_PARAM = 0x00020000, /*!< Controls the initialization of the history of all other parameter */ MPEGS_INIT_STATES_ERROR_CONCEALMENT = 0x00080000, /*!< Controls the initialization of the error concealment module state */ MPEGS_INIT_PARAMS_ERROR_CONCEALMENT = 0x00200000 /*!< Controls the initialization of the whole error concealment parameter set */ } MPEGS_INIT_CTRL_FLAGS; #define MASK_MPEGS_INIT_ALL_STATES (0x000FFF00) #define MASK_MPEGS_INIT_ALL_PARAMS (0x00F00000) typedef struct spatialDec_struct spatialDec, *HANDLE_SPATIAL_DEC; typedef struct SPATIAL_BS_FRAME_struct SPATIAL_BS_FRAME; typedef struct { UINT sizePersistent; /* persistent memory */ UINT sizeFastPersistent; /* fast persistent memory */ } MEM_REQUIREMENTS; #define PCM_MPS LONG #define PCM_MPSF FIXP_DBL #define FIXP_DBL2PCM_MPS(x) ((LONG)(x)) /* exposed functions (library interface) */ int FDK_SpatialDecCompareSpatialSpecificConfigHeader( SPATIAL_SPECIFIC_CONFIG *pSsc1, SPATIAL_SPECIFIC_CONFIG *pSsc2); int FDK_SpatialDecInitDefaultSpatialSpecificConfig( SPATIAL_SPECIFIC_CONFIG *pSpatialSpecificConfig, AUDIO_OBJECT_TYPE coreCodec, int coreChannels, int samplingFreq, int nTimeSlots, int decoderLevel, int isBlind); spatialDec *FDK_SpatialDecOpen(const SPATIAL_DEC_CONFIG *config, int stereoConfigIndex); /** * \brief Initialize state variables of the MPS parser */ void SpatialDecInitParserContext(spatialDec *self); /** * \brief Initialize state of MPS decoder. This may happen after the first parse * operation. */ SACDEC_ERROR FDK_SpatialDecInit(spatialDec *self, SPATIAL_BS_FRAME *frame, SPATIAL_SPECIFIC_CONFIG *pSpatialSpecificConfig, int nQmfBands, SPATIAL_DEC_UPMIX_TYPE const upmixType, SPATIALDEC_PARAM *pUserParams, UINT initFlags /* MPEGS_INIT_CTRL_FLAGS */ ); /** * \brief Apply decoded MPEG Surround parameters to time domain or QMF down mix * data. * \param self spatial decoder handle. * \param inData Pointer to time domain input down mix data if any. * \param qmfInDataReal Pointer array of QMF domain down mix input data (real * part). * \param qmfInDataImag Pointer array of QMF domain down mix input data * (imaginary part). * \param pcmOutBuf Pointer to a time domain buffer were the upmixed output data * will be stored into. * \param nSamples Amount of audio samples per channel of down mix input data * (frame length). * \param pControlFlags pointer to control flags field; input/output. * \param numInputChannels amount of down mix input channels. Might not match * the current tree config, useful for internal sanity checks and bypass mode. * \param channelMapping array containing the desired output channel ordering to * transform MPEG PCE style ordering to any other channel ordering. First * dimension is the total channel count. */ SACDEC_ERROR SpatialDecApplyFrame( spatialDec *self, SPATIAL_BS_FRAME *frame, SPATIALDEC_INPUT_MODE inputMode, PCM_MPS *inData, /* Time domain input */ FIXP_DBL **qmfInDataReal, /* interleaved l/r */ FIXP_DBL **qmfInDataImag, /* interleaved l/r */ PCM_MPS *pcmOutBuf, /* MAX_OUTPUT_CHANNELS*MAX_TIME_SLOTS*NUM_QMF_BANDS] */ UINT nSamples, UINT *pControlFlags, int numInputChannels, const FDK_channelMapDescr *const mapDescr); /** * \brief Fill given arrays with audio channel types and indices. * \param self spatial decoder handle. * \param channelType array where corresponding channel types fr each output * channels are stored into. * \param channelIndices array where corresponding channel type indices fr each * output channels are stored into. */ void SpatialDecChannelProperties(spatialDec *self, AUDIO_CHANNEL_TYPE channelType[], UCHAR channelIndices[], const FDK_channelMapDescr *const mapDescr); void FDK_SpatialDecClose(spatialDec *self); #ifdef __cplusplus } #endif #endif /* SAC_DEC_INTERFACE_H */ fdk-aac-2.0.2/libSACdec/src/sac_dec_lib.cpp000066400000000000000000002066221404225310700202620ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround decoder library ************************* Author(s): Description: SAC Decoder Library Interface *******************************************************************************/ #include "sac_dec_lib.h" #include "sac_dec_interface.h" #include "sac_dec.h" #include "sac_bitdec.h" #include "FDK_matrixCalloc.h" #define MPS_DATA_BUFFER_SIZE (2048) /** * \brief MPEG Surround data indication. **/ typedef enum { MPEGS_ANCTYPE_FRAME = 0, /*!< MPEG Surround frame, see ISO/IEC 23003-1 */ MPEGS_ANCTYPE_HEADER_AND_FRAME = 1, /*!< MPEG Surround header and MPEG Surround frame, see ISO/IEC 23003-1 */ MPEGS_ANCTYPE_RESERVED_1 = 2, /*!< reserved, see ISO/IEC 23003-1 */ MPEGS_ANCTYPE_RESERVED_2 = 3 /*!< reserved, see ISO/IEC 23003-1*/ } MPEGS_ANCTYPE; /** * \brief MPEG Surround data segment indication. **/ typedef enum { MPEGS_CONTINUE = 0, /*!< Indicates if data segment continues a data block. */ MPEGS_STOP = 1, /*!< Indicates if data segment ends a data block. */ MPEGS_START = 2, /*!< Indicates if data segment begins a data block. */ MPEGS_START_STOP = 3 /*!< Indicates if data segment begins and ends a data block. */ } MPEGS_ANCSTARTSTOP; /** * \brief MPEG Surround synchronizaiton state. * * CAUTION: Changing the enumeration values can break the sync mechanism *because it is based on comparing the state values. **/ typedef enum { MPEGS_SYNC_LOST = 0, /*!< Indicates lost sync because of current discontinuity. */ MPEGS_SYNC_FOUND = 1, /*!< Parsed a valid header and (re)intialization was successfully completed. */ MPEGS_SYNC_COMPLETE = 2 /*!< In sync and continuous. Found an independent frame in addition to MPEGS_SYNC_FOUND. Precondition: MPEGS_SYNC_FOUND. */ } MPEGS_SYNCSTATE; /** * \brief MPEG Surround operation mode. **/ typedef enum { MPEGS_OPMODE_EMM = 0, /*!< Mode: Enhanced Matrix Mode (Blind) */ MPEGS_OPMODE_MPS_PAYLOAD = 1, /*!< Mode: Normal, Stereo or Binaural */ MPEGS_OPMODE_NO_MPS_PAYLOAD = 2 /*!< Mode: no MPEG Surround payload */ } MPEGS_OPMODE; /** * \brief MPEG Surround init flags. **/ typedef enum { MPEGS_INIT_OK = 0x00000000, /*!< indicate correct initialization */ MPEGS_INIT_ENFORCE_REINIT = 0x00000001, /*!< indicate complete initialization */ MPEGS_INIT_CHANGE_OUTPUT_MODE = 0x00000010, /*!< indicate change of the output mode */ MPEGS_INIT_CHANGE_PARTIALLY_COMPLEX = 0x00000020, /*!< indicate change of low power/high quality */ MPEGS_INIT_CHANGE_TIME_FREQ_INTERFACE = 0x00000040, /*!< indicate change of qmf/time interface */ MPEGS_INIT_CHANGE_HEADER = 0x00000080, /*!< indicate change of header */ MPEGS_INIT_ERROR_PAYLOAD = 0x00000100, /*!< indicate payload/ancType/ancStartStop error */ MPEGS_INIT_BS_INTERRUPTION = 0x00001000, /*!< indicate bitstream interruption */ MPEGS_INIT_CLEAR_HISTORY = 0x00002000, /*!< indicate that all states shall be cleared */ /* Re-initialization of submodules */ MPEGS_INIT_CHANGE_CONCEAL_PARAMS = 0x00100000, /*!< indicate a change of at least one error concealment param */ /* No re-initialization needed, currently not used */ MPEGS_INIT_CHANGE_BYPASS_MODE = 0x01000000, /*!< indicate change of bypass mode */ /* Re-initialization needed, currently not used */ MPEGS_INIT_ERROR_ANC_TYPE = 0x10000000, /*!< indicate ancType error*/ MPEGS_INIT_ERROR_ANC_STARTSTOP = 0x20000000 /*!< indicate ancStartStop error */ } MPEGS_INIT_FLAGS; struct MpegSurroundDecoder { HANDLE_FDK_QMF_DOMAIN pQmfDomain; UCHAR mpsData[MPS_DATA_BUFFER_SIZE]; /* Buffer for MPS payload accross more than one segment */ INT mpsDataBits; /* Amount of bits in mpsData */ /* MPEG Surround decoder */ SPATIAL_SPECIFIC_CONFIG spatialSpecificConfig[1]; /* SSC delay line which is used during decoding */ spatialDec *pSpatialDec; SPATIAL_SPECIFIC_CONFIG spatialSpecificConfigBackup; /* SSC used while parsing */ /* Creation parameter */ UCHAR mpegSurroundDecoderLevel; /* Run-time parameter */ UCHAR mpegSurroundSscIsGlobalCfg; /* Flag telling that the SSC (::spatialSpecificConfig) is a out-of-band configuration. */ UCHAR mpegSurroundUseTimeInterface; SPATIAL_BS_FRAME bsFrames[1]; /* Bitstream Structs that contain data read from the SpatialFrame() bitstream element */ BS_LL_STATE llState; /* Bit stream parser state memory */ UCHAR bsFrameParse; /* Current parse frame context index */ UCHAR bsFrameDecode; /* Current decode/apply frame context index */ UCHAR bsFrameDelay; /* Amount of frames delay between parsing and processing. Required i.e. for interpolation error concealment. */ /* User prameters */ SPATIALDEC_PARAM mpegSurroundUserParams; /* Internal flags */ SPATIAL_DEC_UPMIX_TYPE upmixType; int initFlags[1]; MPEGS_ANCSTARTSTOP ancStartStopPrev; MPEGS_SYNCSTATE fOnSync[1]; /* Inital decoder configuration */ SPATIAL_DEC_CONFIG decConfig; }; SACDEC_ERROR static sscCheckOutOfBand(const SPATIAL_SPECIFIC_CONFIG *pSsc, const INT coreCodec, const INT sampleRate, const INT frameSize); static SACDEC_ERROR sscParseCheck(const SPATIAL_SPECIFIC_CONFIG *pSsc); /** * \brief Get the number of QMF bands from the sampling frequency (in Hz) **/ static int mpegSurroundDecoder_GetNrOfQmfBands( const SPATIAL_SPECIFIC_CONFIG *pSsc, UINT sampleRate) { UINT samplingFrequency = sampleRate; int qmfBands = 64; if (pSsc != NULL) { switch (pSsc->coreCodec) { case AOT_USAC: if ((pSsc->stereoConfigIndex == 3)) { static const UCHAR mapIdx2QmfBands[3] = {24, 32, 16}; FDK_ASSERT((pSsc->coreSbrFrameLengthIndex >= 2) && (pSsc->coreSbrFrameLengthIndex <= 4)); qmfBands = mapIdx2QmfBands[pSsc->coreSbrFrameLengthIndex - 2]; } return qmfBands; default: samplingFrequency = pSsc->samplingFreq; break; } } /* number of QMF bands depend on sampling frequency, see FDIS 23003-1:2006 * Chapter 6.3.3 */ if (samplingFrequency < 27713) { qmfBands = 32; } if (samplingFrequency > 55426) { qmfBands = 128; } return qmfBands; } /** * \brief Analyse init flags **/ static int mpegSurroundDecoder_CalcInitFlags(SPATIAL_SPECIFIC_CONFIG *pSsc1, SPATIAL_SPECIFIC_CONFIG *pSsc2, int upmixTypeFlag, int binauralQualityFlag, int partiallyComplexFlag, int *ctrlFlags) { /* Analyse core coder */ if (pSsc1->coreCodec != pSsc2->coreCodec) { *ctrlFlags |= MASK_MPEGS_INIT_ALL_STATES; *ctrlFlags |= MASK_MPEGS_INIT_ALL_PARAMS; } else { /* Analyse elements for initialization of space analysis qmf filterbank */ if ((partiallyComplexFlag) || (pSsc1->treeConfig != pSsc2->treeConfig) || (pSsc1->samplingFreq != pSsc2->samplingFreq)) { *ctrlFlags |= MPEGS_INIT_STATES_ANA_QMF_FILTER; *ctrlFlags |= MPEGS_INIT_STATES_ANA_HYB_FILTER; } /* Analyse elements for initialization of space synthesis qmf filterbank */ if ((upmixTypeFlag) || (partiallyComplexFlag) || (pSsc1->treeConfig != pSsc2->treeConfig) || (pSsc1->samplingFreq != pSsc2->samplingFreq) || (pSsc1->bsFixedGainDMX != pSsc2->bsFixedGainDMX)) { *ctrlFlags |= MPEGS_INIT_STATES_SYN_QMF_FILTER; } /* Analyse elements for initialization of decorrelator */ if ((upmixTypeFlag) || (partiallyComplexFlag) || (pSsc1->treeConfig != pSsc2->treeConfig) || (pSsc1->samplingFreq != pSsc2->samplingFreq) || (pSsc1->decorrConfig != pSsc2->decorrConfig)) { *ctrlFlags |= MPEGS_INIT_STATES_DECORRELATOR; } /* Analyse elements for initialization of m1 and m2 calculation */ if ((upmixTypeFlag) || (binauralQualityFlag) || (pSsc1->treeConfig != pSsc2->treeConfig) || (pSsc1->samplingFreq != pSsc2->samplingFreq)) { *ctrlFlags |= MPEGS_INIT_STATES_M1M2; } /* Analyse elements for initialization of GES */ if ((upmixTypeFlag) || (pSsc1->treeConfig != pSsc2->treeConfig) || (pSsc1->tempShapeConfig != pSsc2->tempShapeConfig)) { *ctrlFlags |= MPEGS_INIT_STATES_GES; } /* Analyse elements for initialization of FDreverb */ if ((upmixTypeFlag) || (binauralQualityFlag) || (partiallyComplexFlag) || (pSsc1->samplingFreq != pSsc2->samplingFreq) || (pSsc1->nTimeSlots != pSsc2->nTimeSlots)) { *ctrlFlags |= MPEGS_INIT_STATES_REVERB; } /* Reset previous frame data whenever the config changes */ if (*ctrlFlags & MPEGS_INIT_CONFIG) { *ctrlFlags |= MPEGS_INIT_STATES_PARAM; } } return MPS_OK; } /** * \brief Reset MPEG Surround status info **/ static void updateMpegSurroundDecoderStatus( CMpegSurroundDecoder *pMpegSurroundDecoder, int initFlags, MPEGS_SYNCSTATE fOnSync, MPEGS_ANCSTARTSTOP ancStartStopPrev) { pMpegSurroundDecoder->initFlags[pMpegSurroundDecoder->bsFrameDecode] |= initFlags; if ((pMpegSurroundDecoder->mpegSurroundSscIsGlobalCfg != 0) && (pMpegSurroundDecoder->fOnSync[pMpegSurroundDecoder->bsFrameDecode] >= MPEGS_SYNC_FOUND) && (fOnSync < MPEGS_SYNC_FOUND)) { pMpegSurroundDecoder->fOnSync[pMpegSurroundDecoder->bsFrameDecode] = MPEGS_SYNC_FOUND; } else { pMpegSurroundDecoder->fOnSync[pMpegSurroundDecoder->bsFrameDecode] = fOnSync; } pMpegSurroundDecoder->ancStartStopPrev = ancStartStopPrev; } static SACDEC_ERROR mpegSurroundDecoder_Create( CMpegSurroundDecoder **pMpegSurroundDecoder, int stereoConfigIndex, HANDLE_FDK_QMF_DOMAIN pQmfDomain); SAC_INSTANCE_AVAIL mpegSurroundDecoder_IsFullMpegSurroundDecoderInstanceAvailable( CMpegSurroundDecoder *pMpegSurroundDecoder) { SAC_INSTANCE_AVAIL instanceAvailable = SAC_INSTANCE_NOT_FULL_AVAILABLE; if (pMpegSurroundDecoder->pSpatialDec != NULL) { instanceAvailable = SAC_INSTANCE_FULL_AVAILABLE; } return instanceAvailable; } SACDEC_ERROR mpegSurroundDecoder_Open( CMpegSurroundDecoder **pMpegSurroundDecoder, int stereoConfigIndex, HANDLE_FDK_QMF_DOMAIN pQmfDomain) { SACDEC_ERROR error; error = mpegSurroundDecoder_Create(pMpegSurroundDecoder, stereoConfigIndex, pQmfDomain); return error; } /** * \brief Renamed function from getUpmixType to check_UParam_Build_DecConfig. * This function checks if user params, decoder config and SSC are valid * and if the decoder build can handle all this settings. * The upmix type may be modified by this function. * It is called in initMpegSurroundDecoder() after the ssc parse check, * to have all checks in one place and to ensure these checks are always * performed if config changes (inband and out-of-band). * * \param pUserParams User data handle. * \param pDecConfig decoder config handle. * \param pSsc spatial specific config handle. * \param pUpmixType upmix type which is set by this function * * \return MPS_OK on sucess, and else on failure. */ static SACDEC_ERROR check_UParam_Build_DecConfig( SPATIALDEC_PARAM const *pUserParams, SPATIAL_DEC_CONFIG const *pDecConfig, const SPATIAL_SPECIFIC_CONFIG *pSsc, SPATIAL_DEC_UPMIX_TYPE *pUpmixType) { int dmxChannels, outChannels, maxNumOutChannels; FDK_ASSERT(pUserParams != NULL); FDK_ASSERT(pUpmixType != NULL); /* checks if implementation can handle the Ssc */ switch (pSsc->treeConfig) { case SPATIALDEC_MODE_RSVD7: /* 212 */ dmxChannels = 1; outChannels = 2; break; default: return MPS_UNSUPPORTED_CONFIG; } /* ------------------------------------------- */ /* Analyse pDecConfig params */ switch (pDecConfig->binauralMode) { case BINAURAL_NONE: break; default: return MPS_UNSUPPORTED_CONFIG; } switch (pDecConfig->decoderMode) { case EXT_HQ_ONLY: break; default: return MPS_UNSUPPORTED_CONFIG; } switch (pDecConfig->maxNumOutputChannels) { case OUTPUT_CHANNELS_DEFAULT: /* No special restrictions -> Get the level restriction: */ switch (pDecConfig->decoderLevel) { case DECODER_LEVEL_0: maxNumOutChannels = 2; break; default: return MPS_UNSUPPORTED_CONFIG; } break; case OUTPUT_CHANNELS_2_0: maxNumOutChannels = 2; break; default: return MPS_UNSUPPORTED_CONFIG; } /* ------------------------- */ /* check if we can handle user params */ if (pUserParams->blindEnable == 1) { return MPS_UNSUPPORTED_CONFIG; } { switch ((SAC_DEC_OUTPUT_MODE)pUserParams->outputMode) { case SACDEC_OUT_MODE_NORMAL: if (maxNumOutChannels >= outChannels) { *pUpmixType = UPMIX_TYPE_NORMAL; } else { { *pUpmixType = UPMIX_TYPE_BYPASS; } } break; case SACDEC_OUT_MODE_STEREO: if (dmxChannels == 1) { if (outChannels == 2) { *pUpmixType = UPMIX_TYPE_NORMAL; } } else { *pUpmixType = UPMIX_TYPE_BYPASS; } break; case SACDEC_OUT_MODE_6CHANNEL: if (outChannels > 6) { { *pUpmixType = UPMIX_TYPE_BYPASS; } } else { *pUpmixType = UPMIX_TYPE_NORMAL; } break; default: return MPS_UNSUPPORTED_CONFIG; } } return MPS_OK; } /** * \brief Init MPEG Surround decoder. **/ static SACDEC_ERROR initMpegSurroundDecoder( CMpegSurroundDecoder *pMpegSurroundDecoder) { SACDEC_ERROR err; int initFlags = MPEGS_INIT_NONE, initFlagsDec; int upmixTypeCurr = pMpegSurroundDecoder->upmixType; FDK_ASSERT(pMpegSurroundDecoder != NULL); SPATIAL_SPECIFIC_CONFIG *const pSSCinput = &pMpegSurroundDecoder->spatialSpecificConfigBackup; SPATIAL_SPECIFIC_CONFIG *const pSSCtarget = &pMpegSurroundDecoder ->spatialSpecificConfig[pMpegSurroundDecoder->bsFrameDecode]; initFlagsDec = pMpegSurroundDecoder->initFlags[pMpegSurroundDecoder->bsFrameDecode]; if (pSSCinput->coreCodec != AOT_USAC) { /* here we check if we have a valid Ssc */ err = sscParseCheck(pSSCinput); if (err != MPS_OK) goto bail; } /* here we check if Ssc matches build; also check UParams and DecConfig */ /* if desired upmixType is changes */ err = check_UParam_Build_DecConfig( &pMpegSurroundDecoder->mpegSurroundUserParams, &pMpegSurroundDecoder->decConfig, pSSCinput, &pMpegSurroundDecoder->upmixType); if (err != MPS_OK) goto bail; /* init config */ if (initFlagsDec & MPEGS_INIT_CHANGE_HEADER) { initFlags |= MPEGS_INIT_CONFIG; } /* init all states */ if (initFlagsDec & MPEGS_INIT_CLEAR_HISTORY) { initFlags |= MASK_MPEGS_INIT_ALL_STATES; } if (initFlagsDec & MPEGS_INIT_CHANGE_CONCEAL_PARAMS) { initFlags |= MPEGS_INIT_PARAMS_ERROR_CONCEALMENT; } if (initFlagsDec & MPEGS_INIT_ENFORCE_REINIT) { /* init all states */ initFlags |= MASK_MPEGS_INIT_ALL_STATES; initFlags |= MASK_MPEGS_INIT_ALL_PARAMS; } else { /* analyse states which have to be initialized */ mpegSurroundDecoder_CalcInitFlags( pSSCtarget, pSSCinput, (upmixTypeCurr != pMpegSurroundDecoder->upmixType), /* upmixType changed */ 0, (initFlagsDec & MPEGS_INIT_CHANGE_PARTIALLY_COMPLEX) ? 1 : 0, &initFlags); } { int nrOfQmfBands; FDKmemcpy(pSSCtarget, pSSCinput, sizeof(SPATIAL_SPECIFIC_CONFIG)); nrOfQmfBands = mpegSurroundDecoder_GetNrOfQmfBands( pSSCtarget, pSSCtarget->samplingFreq); err = FDK_SpatialDecInit( pMpegSurroundDecoder->pSpatialDec, &pMpegSurroundDecoder->bsFrames[pMpegSurroundDecoder->bsFrameDecode], pSSCtarget, nrOfQmfBands, pMpegSurroundDecoder->upmixType, &pMpegSurroundDecoder->mpegSurroundUserParams, initFlags); if (err != MPS_OK) goto bail; /* Signal that we got a header and can go on decoding */ if (err == MPS_OK) { initFlagsDec = MPEGS_INIT_OK; { pMpegSurroundDecoder->fOnSync[pMpegSurroundDecoder->bsFrameDecode] = MPEGS_SYNC_FOUND; } } } bail: pMpegSurroundDecoder->initFlags[pMpegSurroundDecoder->bsFrameDecode] = initFlagsDec; return err; } /** * \brief Init MPEG Surround decoder. **/ SACDEC_ERROR mpegSurroundDecoder_Init( CMpegSurroundDecoder *pMpegSurroundDecoder) { SACDEC_ERROR err = MPS_OK; if (pMpegSurroundDecoder->initFlags[pMpegSurroundDecoder->bsFrameDecode]) { err = initMpegSurroundDecoder(pMpegSurroundDecoder); } return err; } /** * \brief Open MPEG Surround decoder. **/ static SACDEC_ERROR mpegSurroundDecoder_Create( CMpegSurroundDecoder **pMpegSurroundDecoder, int stereoConfigIndex, HANDLE_FDK_QMF_DOMAIN pQmfDomain) { SACDEC_ERROR err = MPS_OK; CMpegSurroundDecoder *sacDec = NULL; spatialDec *self = NULL; /* decoderLevel decoderMode maxNumOutputChannels binauralMode */ static const SPATIAL_DEC_CONFIG decConfig = { (CFG_LEVEL)(0), EXT_HQ_ONLY, OUTPUT_CHANNELS_DEFAULT, BINAURAL_NONE}; if (*pMpegSurroundDecoder == NULL) { FDK_ALLOCATE_MEMORY_1D(*pMpegSurroundDecoder, 1, CMpegSurroundDecoder) for (int i = 0; i < 1; i++) { err = SpatialDecCreateBsFrame(&(*pMpegSurroundDecoder)->bsFrames[i], &(*pMpegSurroundDecoder)->llState); if (err != MPS_OK) { sacDec = *pMpegSurroundDecoder; goto bail; } } (*pMpegSurroundDecoder)->pQmfDomain = pQmfDomain; (*pMpegSurroundDecoder)->bsFrameDelay = 1; (*pMpegSurroundDecoder)->bsFrameParse = 0; (*pMpegSurroundDecoder)->bsFrameDecode = 0; return err; } else { sacDec = *pMpegSurroundDecoder; } if (sacDec->pSpatialDec == NULL) { if ((self = FDK_SpatialDecOpen(&decConfig, stereoConfigIndex)) == NULL) { err = MPS_OUTOFMEMORY; goto bail; } } else { self = sacDec->pSpatialDec; } self->pQmfDomain = sacDec->pQmfDomain; sacDec->pSpatialDec = self; /* default parameter set */ sacDec->mpegSurroundUserParams.outputMode = SACDEC_OUT_MODE_NORMAL; sacDec->mpegSurroundUserParams.blindEnable = 0; sacDec->mpegSurroundUserParams.bypassMode = 0; sacDec->mpegSurroundUserParams.concealMethod = 1; sacDec->mpegSurroundUserParams.concealNumKeepFrames = 10; sacDec->mpegSurroundUserParams.concealFadeOutSlopeLength = 5; sacDec->mpegSurroundUserParams.concealFadeInSlopeLength = 5; sacDec->mpegSurroundUserParams.concealNumReleaseFrames = 3; sacDec->mpegSurroundSscIsGlobalCfg = 0; sacDec->mpegSurroundUseTimeInterface = 1; sacDec->mpegSurroundDecoderLevel = decConfig.decoderLevel; sacDec->upmixType = UPMIX_TYPE_NORMAL; /* signalize spatial decoder re-initalization */ updateMpegSurroundDecoderStatus(sacDec, MPEGS_INIT_ENFORCE_REINIT, MPEGS_SYNC_LOST, MPEGS_STOP); /* return decoder instance */ *pMpegSurroundDecoder = sacDec; sacDec->decConfig = decConfig; SpatialDecInitParserContext(sacDec->pSpatialDec); return err; bail: if (sacDec != NULL) { mpegSurroundDecoder_Close(sacDec); } *pMpegSurroundDecoder = NULL; if (err == MPS_OK) { return MPS_OUTOFMEMORY; } else { return err; } } /** * \brief Config MPEG Surround decoder. **/ SACDEC_ERROR mpegSurroundDecoder_Config( CMpegSurroundDecoder *pMpegSurroundDecoder, HANDLE_FDK_BITSTREAM hBs, AUDIO_OBJECT_TYPE coreCodec, INT samplingRate, INT frameSize, INT stereoConfigIndex, INT coreSbrFrameLengthIndex, INT configBytes, const UCHAR configMode, UCHAR *configChanged) { SACDEC_ERROR err = MPS_OK; SPATIAL_SPECIFIC_CONFIG spatialSpecificConfig; SPATIAL_SPECIFIC_CONFIG *pSsc = &pMpegSurroundDecoder->spatialSpecificConfigBackup; switch (coreCodec) { case AOT_DRM_USAC: case AOT_USAC: if (configMode == AC_CM_DET_CFG_CHANGE) { /* In config detection mode write spatial specific config parameters * into temporarily allocated structure */ err = SpatialDecParseMps212Config( hBs, &spatialSpecificConfig, samplingRate, coreCodec, stereoConfigIndex, coreSbrFrameLengthIndex); pSsc = &spatialSpecificConfig; } else { err = SpatialDecParseMps212Config( hBs, &pMpegSurroundDecoder->spatialSpecificConfigBackup, samplingRate, coreCodec, stereoConfigIndex, coreSbrFrameLengthIndex); } break; case AOT_ER_AAC_ELD: case AOT_ER_AAC_LD: if (configMode == AC_CM_DET_CFG_CHANGE) { /* In config detection mode write spatial specific config parameters * into temporarily allocated structure */ err = SpatialDecParseSpecificConfig(hBs, &spatialSpecificConfig, configBytes, coreCodec); pSsc = &spatialSpecificConfig; } else { err = SpatialDecParseSpecificConfig( hBs, &pMpegSurroundDecoder->spatialSpecificConfigBackup, configBytes, coreCodec); } break; default: err = MPS_UNSUPPORTED_FORMAT; break; } if (err != MPS_OK) { goto bail; } err = sscCheckOutOfBand(pSsc, coreCodec, samplingRate, frameSize); if (err != MPS_OK) { goto bail; } if (configMode & AC_CM_DET_CFG_CHANGE) { return err; } if (configMode & AC_CM_ALLOC_MEM) { if (*configChanged) { err = mpegSurroundDecoder_Open(&pMpegSurroundDecoder, stereoConfigIndex, NULL); if (err) { return err; } } } { SPATIAL_SPECIFIC_CONFIG *sscParse = &pMpegSurroundDecoder ->spatialSpecificConfig[pMpegSurroundDecoder->bsFrameParse]; if (FDK_SpatialDecCompareSpatialSpecificConfigHeader( &pMpegSurroundDecoder->spatialSpecificConfigBackup, sscParse)) { pMpegSurroundDecoder->initFlags[pMpegSurroundDecoder->bsFrameParse] |= MPEGS_INIT_CHANGE_HEADER; /* Error resilience code */ if (pMpegSurroundDecoder->pSpatialDec == NULL) { err = MPS_NOTOK; goto bail; } SpatialDecInitParserContext(pMpegSurroundDecoder->pSpatialDec); pMpegSurroundDecoder->pSpatialDec->pConfigCurrent = &pMpegSurroundDecoder ->spatialSpecificConfig[pMpegSurroundDecoder->bsFrameDecode]; } } if (err == MPS_OK) { /* We got a valid out-of-band configuration so label it accordingly. */ pMpegSurroundDecoder->mpegSurroundSscIsGlobalCfg = 1; } bail: return err; } /** * \brief Determine MPEG Surround operation mode. **/ static MPEGS_OPMODE mpegSurroundOperationMode( CMpegSurroundDecoder *pMpegSurroundDecoder, int mpsDataBits) { MPEGS_OPMODE mode; { if ((mpsDataBits > 0) && (pMpegSurroundDecoder->mpegSurroundUserParams.blindEnable == 0)) { mode = MPEGS_OPMODE_MPS_PAYLOAD; /* Mode: Normal, Stereo or Binaural */ } else { mode = MPEGS_OPMODE_NO_MPS_PAYLOAD; /* Mode: No MPEG Surround Payload */ updateMpegSurroundDecoderStatus(pMpegSurroundDecoder, MPEGS_INIT_ERROR_PAYLOAD, MPEGS_SYNC_LOST, MPEGS_STOP); } } return (mode); } /** * \brief Check ssc for parse errors. * This one is called in initMpegSurroundDecoder() * to ensure checking of inband and out-of-band mps configs. * Only parse errors checked here! Check for valid config is done * in check_UParam_Build_DecConfig()! * * \param pSsc spatial specific config handle. * * \return MPS_OK on sucess, and else on parse error. */ static SACDEC_ERROR sscParseCheck(const SPATIAL_SPECIFIC_CONFIG *pSsc) { if (pSsc->samplingFreq > 96000) return MPS_PARSE_ERROR; if (pSsc->samplingFreq < 8000) return MPS_PARSE_ERROR; if ((pSsc->treeConfig < 0) || (pSsc->treeConfig > 7)) { return MPS_PARSE_ERROR; } if ((pSsc->quantMode < 0) || (pSsc->quantMode > 2)) { return MPS_PARSE_ERROR; } /* now we are sure there were no parsing errors */ return MPS_OK; } /** * \brief Check number of time slots * * Basically the mps frame length must be a multiple of the core coder frame * length. The below table shows all valid configurations in detail. See ISO/IEC * 23003-1: "Table 4A - Allowed values for bsFrameLength in the Baseline MPEG * Surround Profile" * * Downmix Coder Downmix Code Allowed values for bsFrameLength * Allowed frame sizes for normal, downsampled and upsampled MPS Framelength * (QMF Samples) * * AAC 1024 16 15, 31, 47, 63 1024 2048 3072 4096 * downsampled MPS 32 31, 63 1024 2048 upsampled MPS * 8 7, 15, 23, 31, 39, 47, 55, 63, 71 1024 2048 3072 4096 * 5120 6144 7168 8192 9216 * * AAC 960 15 14, 29, 44, 59 960 1920 2880 3840 * downsampled MPS 30 29, 59 960 1920 upsampled MPS * 7,5 14, 29, 44, 59 1920 3840 5760 7680 * * HE-AAC 1024/2048 32 31, 63 2048 4096 downsampled MPS * 64 63 2048 upsampled MPS * 16 15, 31, 47, 63 2048 4096 6144 8192 * * HE-AAC 960/1920 30 29, 59 1920 3840 downsampled MPS * 60 59 1920 upsampled MPS * 15 14, 29, 44, 59 1920 3840 5760 7680 * * BSAC 16 15, 31, 47, 63 1024 2048 3072 4096 * downsampled MPS 32 31, 63 1024 2048 upsampled MPS * 8 7, 15, 23, 31, 39, 47, 55, 63, 71 1024 2048 3072 4096 * 5120 6144 7168 8192 9216 * * BSAC with SBR 32 31, 63 2048 4096 downsampled MPS * 64 63 2048 upsampled MPS * 16 15, 31, 47, 63 2048 4096 6144 8192 * * AAC LD 512 8 7, 15, 23, 31, 39, 47, 55, 63, 71 * 512 1024 1536 2048 2560 3072 3584 4096 4608 downsampled MPS * 16 15, 31, 47, 63 512 1024 1536 2048 * * AAC ELD 512 8 7, 15, 23, 31, 39, 47, 55, 63, 71 * 512 1024 1536 2048 2560 3072 3584 4096 4608 downsampled MPS * 16 15, 31, 47, 63 512 1024 1536 2048 * * AAC ELD with SBR 512/1024 16 15, 31, 47, 63 1024 2048 3072 4096 * downsampled MPS 32 31, 63 1024 2048 upsampled MPS * 8 7, 15, 23, 31, 39, 47, 55, 63, 71 1024 2048 3072 4096 * 5120 6144 7168 8192 9216 * * MPEG1/2 Layer II 18 17, 35, 53, 71 1152 2304 3456 4608 * downsampled MPS 36 35, 71 1152 2304 * * MPEG1/2 Layer III 18 17, 35, 53, 71 1152 2304 3456 4608 * downsampled MPS 36 35, 71 1152 2304 * * \param frameLength * \param qmfBands * \param timeSlots * * \return error code */ SACDEC_ERROR checkTimeSlots(int frameLength, int qmfBands, int timeSlots) { int len; int maxFrameLength; if (qmfBands == 64) { /* normal MPEG Surround */ switch (frameLength) { case 960: case 1920: maxFrameLength = 3840; break; case 1024: case 2048: maxFrameLength = 4096; break; case 512: case 1152: maxFrameLength = 4608; break; default: return MPS_PARSE_ERROR; } } else if (qmfBands == 32) { /* downsampled MPEG Surround */ switch (frameLength) { case 960: case 1920: maxFrameLength = 1920; break; case 512: case 1024: case 2048: maxFrameLength = 2048; break; case 1152: maxFrameLength = 2304; break; default: return MPS_PARSE_ERROR; } } else if (qmfBands == 128) { /* upsampled MPEG Surround */ switch (frameLength) { case 1920: maxFrameLength = 7680; break; case 1024: maxFrameLength = 9216; break; case 2048: maxFrameLength = 8192; break; case 512: case 960: case 1152: /* no break, no support for upsampled MPEG Surround */ default: return MPS_PARSE_ERROR; } } else { return MPS_PARSE_ERROR; } len = frameLength; while (len <= maxFrameLength) { if (len == timeSlots * qmfBands) { return MPS_OK; } len += frameLength; } return MPS_PARSE_ERROR; } /** * \brief Check ssc for consistency (e.g. bit errors could cause trouble) * First of currently two ssc-checks. * This (old) one is called in mpegSurroundDecoder_Apply() * only if inband mps config is contained in stream. * * New ssc check is split in two functions sscParseCheck() and * check_UParam_Build_DecConfig(). sscParseCheck() checks only for correct * parsing. check_UParam_Build_DecConfig() is used to check if we have a * valid config. Both are called in initMpegSurroundDecoder() to ensure * checking of inband and out-of-band mps configs. * * If this function can be integrated into the new functions. * We can remove this one. * * \param pSsc spatial specific config handle. * \param frameLength * \param sampleRate * * \return MPS_OK on sucess, and else on failure. */ static SACDEC_ERROR sscCheckInBand(SPATIAL_SPECIFIC_CONFIG *pSsc, int frameLength, int sampleRate) { SACDEC_ERROR err = MPS_OK; int qmfBands; FDK_ASSERT(pSsc != NULL); /* check ssc for parse errors */ if (sscParseCheck(pSsc) != MPS_OK) { err = MPS_PARSE_ERROR; } /* core fs and mps fs must match */ if (pSsc->samplingFreq != sampleRate) { err = MPS_PARSE_ERROR /* MPEGSDEC_SSC_PARSE_ERROR */; } qmfBands = mpegSurroundDecoder_GetNrOfQmfBands(pSsc, pSsc->samplingFreq); if (checkTimeSlots(frameLength, qmfBands, pSsc->nTimeSlots) != MPS_OK) { err = MPS_PARSE_ERROR; } return err; } SACDEC_ERROR mpegSurroundDecoder_ConfigureQmfDomain( CMpegSurroundDecoder *pMpegSurroundDecoder, SAC_INPUT_CONFIG sac_dec_interface, UINT coreSamplingRate, AUDIO_OBJECT_TYPE coreCodec) { SACDEC_ERROR err = MPS_OK; FDK_QMF_DOMAIN_GC *pGC = NULL; if (pMpegSurroundDecoder == NULL) { return MPS_INVALID_HANDLE; } FDK_ASSERT(pMpegSurroundDecoder->pSpatialDec); pGC = &pMpegSurroundDecoder->pQmfDomain->globalConf; if (pMpegSurroundDecoder->mpegSurroundSscIsGlobalCfg) { SPATIAL_SPECIFIC_CONFIG *pSSC = &pMpegSurroundDecoder->spatialSpecificConfigBackup; if (sac_dec_interface == SAC_INTERFACE_TIME) { /* For SAC_INTERFACE_QMF these parameters are set by SBR. */ pGC->nBandsAnalysis_requested = mpegSurroundDecoder_GetNrOfQmfBands( pSSC, coreSamplingRate); /* coreSamplingRate == outputSamplingRate for SAC_INTERFACE_TIME */ pGC->nBandsSynthesis_requested = pGC->nBandsAnalysis_requested; pGC->nInputChannels_requested = fMax((UINT)pSSC->nInputChannels, (UINT)pGC->nInputChannels_requested); } pGC->nOutputChannels_requested = fMax((UINT)pSSC->nOutputChannels, (UINT)pGC->nOutputChannels_requested); } else { if (sac_dec_interface == SAC_INTERFACE_TIME) { /* For SAC_INTERFACE_QMF these parameters are set by SBR. */ pGC->nBandsAnalysis_requested = mpegSurroundDecoder_GetNrOfQmfBands( NULL, coreSamplingRate); /* coreSamplingRate == outputSamplingRate for SAC_INTERFACE_TIME */ pGC->nBandsSynthesis_requested = pGC->nBandsAnalysis_requested; pGC->nInputChannels_requested = pMpegSurroundDecoder->pSpatialDec->createParams.maxNumInputChannels; } pGC->nOutputChannels_requested = pMpegSurroundDecoder->pSpatialDec->createParams.maxNumOutputChannels; } pGC->nQmfProcBands_requested = 64; pGC->nQmfProcChannels_requested = fMin((INT)pGC->nInputChannels_requested, pMpegSurroundDecoder->pSpatialDec->createParams.maxNumInputChannels); if (coreCodec == AOT_ER_AAC_ELD) { pGC->flags_requested |= QMF_FLAG_MPSLDFB; pGC->flags_requested &= ~QMF_FLAG_CLDFB; } return err; } /** * \brief Check out-of-band config * * \param pSsc spatial specific config handle. * \param coreCodec core codec. * \param sampleRate sampling frequency. * * \return errorStatus */ SACDEC_ERROR sscCheckOutOfBand(const SPATIAL_SPECIFIC_CONFIG *pSsc, const INT coreCodec, const INT sampleRate, const INT frameSize) { FDK_ASSERT(pSsc != NULL); int qmfBands = 0; /* check ssc for parse errors */ if (sscParseCheck(pSsc) != MPS_OK) { return MPS_PARSE_ERROR; } switch (coreCodec) { case AOT_USAC: case AOT_DRM_USAC: /* ISO/IEC 23003-1:2007(E), Chapter 6.3.3, Support for lower and higher * sampling frequencies */ if (pSsc->samplingFreq >= 55426) { return MPS_PARSE_ERROR; } break; case AOT_ER_AAC_LD: case AOT_ER_AAC_ELD: /* core fs and mps fs must match */ if (pSsc->samplingFreq != sampleRate) { return MPS_PARSE_ERROR; } /* ISO/IEC 14496-3:2009 FDAM 3: Chapter 1.5.2.3, Levels for the Low Delay * AAC v2 profile */ if (pSsc->samplingFreq > 48000) { return MPS_PARSE_ERROR; } qmfBands = mpegSurroundDecoder_GetNrOfQmfBands(pSsc, pSsc->samplingFreq); switch (frameSize) { case 480: if (!((qmfBands == 32) && (pSsc->nTimeSlots == 15))) { return MPS_PARSE_ERROR; } break; case 960: if (!((qmfBands == 64) && (pSsc->nTimeSlots == 15))) { return MPS_PARSE_ERROR; } break; case 512: if (!(((qmfBands == 32) && (pSsc->nTimeSlots == 16)) || ((qmfBands == 64) && (pSsc->nTimeSlots == 8)))) { return MPS_PARSE_ERROR; } break; case 1024: if (!((qmfBands == 64) && (pSsc->nTimeSlots == 16))) { return MPS_PARSE_ERROR; } break; default: return MPS_PARSE_ERROR; } break; default: return MPS_PARSE_ERROR; break; } return MPS_OK; } /** * \brief Decode MPEG Surround frame. **/ int mpegSurroundDecoder_ParseNoHeader( CMpegSurroundDecoder *pMpegSurroundDecoder, HANDLE_FDK_BITSTREAM hBs, int *pMpsDataBits, int fGlobalIndependencyFlag) { SACDEC_ERROR err = MPS_OK; SPATIAL_SPECIFIC_CONFIG *sscParse; int bitsAvail, numSacBits; if (pMpegSurroundDecoder == NULL || hBs == NULL) { return MPS_INVALID_HANDLE; } sscParse = &pMpegSurroundDecoder ->spatialSpecificConfig[pMpegSurroundDecoder->bsFrameParse]; bitsAvail = FDKgetValidBits(hBs); /* First spatial specific config is parsed into spatialSpecificConfigBackup, * second spatialSpecificConfigBackup is copied into * spatialSpecificConfig[bsFrameDecode] */ if (pMpegSurroundDecoder->initFlags[pMpegSurroundDecoder->bsFrameParse]) { FDKmemcpy(sscParse, &pMpegSurroundDecoder->spatialSpecificConfigBackup, sizeof(SPATIAL_SPECIFIC_CONFIG)); pMpegSurroundDecoder->fOnSync[pMpegSurroundDecoder->bsFrameParse] = MPEGS_SYNC_FOUND; } if (bitsAvail <= 0) { err = MPS_PARSE_ERROR; } else { err = SpatialDecParseFrameData( pMpegSurroundDecoder->pSpatialDec, &pMpegSurroundDecoder->bsFrames[pMpegSurroundDecoder->bsFrameParse], hBs, sscParse, (UPMIXTYPE)pMpegSurroundDecoder->upmixType, fGlobalIndependencyFlag); if (err == MPS_OK) { pMpegSurroundDecoder->bsFrames[pMpegSurroundDecoder->bsFrameParse] .newBsData = 1; } } numSacBits = bitsAvail - (INT)FDKgetValidBits(hBs); if (numSacBits > bitsAvail) { pMpegSurroundDecoder->bsFrames[pMpegSurroundDecoder->bsFrameParse] .newBsData = 0; err = MPS_PARSE_ERROR; } *pMpsDataBits -= numSacBits; return err; } /** * \brief Check, if ancType is valid. **/ static int isValidAncType(CMpegSurroundDecoder *pMpegSurroundDecoder, int ancType) { int ret = 1; if ((ancType != MPEGS_ANCTYPE_HEADER_AND_FRAME) && (ancType != MPEGS_ANCTYPE_FRAME)) { ret = 0; } if (ret == 0) { updateMpegSurroundDecoderStatus(pMpegSurroundDecoder, MPEGS_INIT_ERROR_PAYLOAD, MPEGS_SYNC_LOST, MPEGS_STOP); } return (ret); } /** * \brief Check, if ancStartStop is valid. **/ static int isValidAncStartStop(CMpegSurroundDecoder *pMpegSurroundDecoder, int ancStartStop) { int ret = 1; switch (ancStartStop) { case MPEGS_START: /* Sequence start - start and continue - start not allowed */ if ((pMpegSurroundDecoder->ancStartStopPrev == MPEGS_START) || (pMpegSurroundDecoder->ancStartStopPrev == MPEGS_CONTINUE)) { ret = 0; } break; case MPEGS_STOP: /* MPS payload of the previous frame must be valid if current type is stop Sequence startstop - stop and stop - stop not allowed Sequence startstop - continue and stop - continue are allowed */ if ((pMpegSurroundDecoder->ancStartStopPrev == MPEGS_STOP) || (pMpegSurroundDecoder->ancStartStopPrev == MPEGS_START_STOP)) { ret = 0; } break; case MPEGS_CONTINUE: case MPEGS_START_STOP: /* No error detection possible for this states */ break; } if (ret == 0) { updateMpegSurroundDecoderStatus(pMpegSurroundDecoder, MPEGS_INIT_ERROR_PAYLOAD, MPEGS_SYNC_LOST, MPEGS_STOP); } else { pMpegSurroundDecoder->ancStartStopPrev = (MPEGS_ANCSTARTSTOP)ancStartStop; } return (ret); } int mpegSurroundDecoder_Parse(CMpegSurroundDecoder *pMpegSurroundDecoder, HANDLE_FDK_BITSTREAM hBs, int *pMpsDataBits, AUDIO_OBJECT_TYPE coreCodec, int sampleRate, int frameSize, int fGlobalIndependencyFlag) { SACDEC_ERROR err = MPS_OK; SPATIAL_SPECIFIC_CONFIG *sscParse; SPATIAL_BS_FRAME *bsFrame; HANDLE_FDK_BITSTREAM hMpsBsData = NULL; FDK_BITSTREAM mpsBsData; int mpsDataBits = *pMpsDataBits; int mpsBsBits; MPEGS_ANCTYPE ancType; MPEGS_ANCSTARTSTOP ancStartStop; if (pMpegSurroundDecoder == NULL) { return MPS_INVALID_HANDLE; } FDK_ASSERT(pMpegSurroundDecoder->pSpatialDec); mpsBsBits = (INT)FDKgetValidBits(hBs); sscParse = &pMpegSurroundDecoder ->spatialSpecificConfig[pMpegSurroundDecoder->bsFrameParse]; bsFrame = &pMpegSurroundDecoder->bsFrames[pMpegSurroundDecoder->bsFrameParse]; /* Find operation mode of mpeg surround decoder: - MPEGS_OPMODE_EMM: Mode: Enhanced Matrix Mode (Blind) - MPEGS_OPMODE_MPS_PAYLOAD: Mode: Normal, Stereo or Binaural - MPEGS_OPMODE_NO_MPS_PAYLOAD: Mode: No MpegSurround Payload */ { /* Parse ancType and ancStartStop */ ancType = (MPEGS_ANCTYPE)FDKreadBits(hBs, 2); ancStartStop = (MPEGS_ANCSTARTSTOP)FDKreadBits(hBs, 2); mpsDataBits -= 4; /* Set valid anc type flag, if ancType signals a payload with either header * and frame or frame */ if (isValidAncType(pMpegSurroundDecoder, ancType)) { /* Set valid anc startstop flag, if transmitted sequence is not illegal */ if (isValidAncStartStop(pMpegSurroundDecoder, ancStartStop)) { switch (ancStartStop) { case MPEGS_START: /* Assuming that core coder frame size (AAC) is smaller than MPS coder frame size. Save audio data for next frame. */ if (mpsDataBits > MPS_DATA_BUFFER_SIZE * 8) { err = MPS_NOTOK; goto bail; } for (int i = 0; i < mpsDataBits / 8; i++) { pMpegSurroundDecoder->mpsData[i] = FDKreadBits(hBs, 8); } pMpegSurroundDecoder->mpsDataBits = mpsDataBits; break; case MPEGS_CONTINUE: case MPEGS_STOP: /* Assuming that core coder frame size (AAC) is smaller than MPS coder frame size. Save audio data for next frame. */ if ((pMpegSurroundDecoder->mpsDataBits + mpsDataBits) > MPS_DATA_BUFFER_SIZE * 8) { err = MPS_NOTOK; goto bail; } for (int i = 0; i < mpsDataBits / 8; i++) { pMpegSurroundDecoder ->mpsData[(pMpegSurroundDecoder->mpsDataBits / 8) + i] = FDKreadBits(hBs, 8); } pMpegSurroundDecoder->mpsDataBits += mpsDataBits; FDKinitBitStream(&mpsBsData, pMpegSurroundDecoder->mpsData, MAX_BUFSIZE_BYTES, pMpegSurroundDecoder->mpsDataBits, BS_READER); hMpsBsData = &mpsBsData; break; case MPEGS_START_STOP: pMpegSurroundDecoder->mpsDataBits = mpsDataBits; hMpsBsData = hBs; break; default: FDK_ASSERT(0); } if ((ancStartStop == MPEGS_STOP) || (ancStartStop == MPEGS_START_STOP)) { switch (ancType) { case MPEGS_ANCTYPE_HEADER_AND_FRAME: { int parseResult, bitsRead; SPATIAL_SPECIFIC_CONFIG spatialSpecificConfigTmp = pMpegSurroundDecoder->spatialSpecificConfigBackup; /* Parse spatial specific config */ bitsRead = (INT)FDKgetValidBits(hMpsBsData); err = SpatialDecParseSpecificConfigHeader( hMpsBsData, &pMpegSurroundDecoder->spatialSpecificConfigBackup, coreCodec, pMpegSurroundDecoder->upmixType); bitsRead = (bitsRead - (INT)FDKgetValidBits(hMpsBsData)); parseResult = ((err == MPS_OK) ? bitsRead : -bitsRead); if (parseResult < 0) { parseResult = -parseResult; err = MPS_PARSE_ERROR; } else if (err == MPS_OK) { /* Check SSC for consistency (e.g. bit errors could cause * trouble) */ err = sscCheckInBand( &pMpegSurroundDecoder->spatialSpecificConfigBackup, frameSize, sampleRate); } if (err != MPS_OK) { pMpegSurroundDecoder->spatialSpecificConfigBackup = spatialSpecificConfigTmp; break; } pMpegSurroundDecoder->mpsDataBits -= parseResult; /* Initiate re-initialization, if header has changed */ if (FDK_SpatialDecCompareSpatialSpecificConfigHeader( &pMpegSurroundDecoder->spatialSpecificConfigBackup, sscParse) == MPS_UNEQUAL_SSC) { pMpegSurroundDecoder ->initFlags[pMpegSurroundDecoder->bsFrameParse] |= MPEGS_INIT_CHANGE_HEADER; SpatialDecInitParserContext(pMpegSurroundDecoder->pSpatialDec); /* We found a valid in-band configuration. Therefore any * previous config is invalid now. */ pMpegSurroundDecoder->mpegSurroundSscIsGlobalCfg = 0; } } FDK_FALLTHROUGH; case MPEGS_ANCTYPE_FRAME: if (pMpegSurroundDecoder ->initFlags[pMpegSurroundDecoder->bsFrameParse] & MPEGS_INIT_ERROR_PAYLOAD) { err = MPS_PARSE_ERROR; break; } /* First spatial specific config is parsed into * spatialSpecificConfigBackup, second spatialSpecificConfigBackup * is copied into spatialSpecificConfig[bsFrameDecode] */ if (pMpegSurroundDecoder ->initFlags[pMpegSurroundDecoder->bsFrameParse]) { FDKmemcpy(sscParse, &pMpegSurroundDecoder->spatialSpecificConfigBackup, sizeof(SPATIAL_SPECIFIC_CONFIG)); pMpegSurroundDecoder ->fOnSync[pMpegSurroundDecoder->bsFrameParse] = MPEGS_SYNC_FOUND; } if (pMpegSurroundDecoder ->fOnSync[pMpegSurroundDecoder->bsFrameParse] >= MPEGS_SYNC_FOUND) { int nbits = 0, bitsAvail; if (err != MPS_OK) { break; } bitsAvail = FDKgetValidBits(hMpsBsData); if (bitsAvail <= 0) { err = MPS_PARSE_ERROR; } else { err = SpatialDecParseFrameData( pMpegSurroundDecoder->pSpatialDec, bsFrame, hMpsBsData, sscParse, (UPMIXTYPE)pMpegSurroundDecoder->upmixType, fGlobalIndependencyFlag); if (err == MPS_OK) { bsFrame->newBsData = 1; } } nbits = bitsAvail - (INT)FDKgetValidBits(hMpsBsData); if ((nbits > bitsAvail) || (nbits > pMpegSurroundDecoder->mpsDataBits) || (pMpegSurroundDecoder->mpsDataBits > nbits + 7 && !IS_LOWDELAY(coreCodec))) { bsFrame->newBsData = 0; err = MPS_PARSE_ERROR; break; } pMpegSurroundDecoder->mpsDataBits -= nbits; } break; default: /* added to avoid compiler warning */ err = MPS_NOTOK; break; /* added to avoid compiler warning */ } /* switch (ancType) */ if (err == MPS_OK) { pMpegSurroundDecoder->ancStartStopPrev = ancStartStop; } else { updateMpegSurroundDecoderStatus(pMpegSurroundDecoder, MPEGS_INIT_ERROR_PAYLOAD, MPEGS_SYNC_LOST, MPEGS_STOP); pMpegSurroundDecoder->mpsDataBits = 0; } } /* (ancStartStop == MPEGS_STOP) || (ancStartStop == MPEGS_START_STOP) */ } /* validAncStartStop */ } /* validAncType */ } bail: *pMpsDataBits -= (mpsBsBits - (INT)FDKgetValidBits(hBs)); return err; } int mpegSurroundDecoder_Apply(CMpegSurroundDecoder *pMpegSurroundDecoder, PCM_MPS *input, PCM_MPS *pTimeData, const int timeDataSize, int timeDataFrameSize, int *nChannels, int *frameSize, int sampleRate, AUDIO_OBJECT_TYPE coreCodec, AUDIO_CHANNEL_TYPE channelType[], UCHAR channelIndices[], const FDK_channelMapDescr *const mapDescr, const INT inDataHeadroom, INT *outDataHeadroom) { SACDEC_ERROR err = MPS_OK; PCM_MPS *pTimeOut = pTimeData; PCM_MPS *TDinput = NULL; UINT initControlFlags = 0, controlFlags = 0; int timeDataRequiredSize = 0; int newData; if (pMpegSurroundDecoder == NULL) { return MPS_INVALID_HANDLE; } FDK_ASSERT(pMpegSurroundDecoder->pSpatialDec); if (!FDK_chMapDescr_isValid(mapDescr)) { return MPS_INVALID_HANDLE; } if ((*nChannels <= 0) || (*nChannels > 2)) { return MPS_NOTOK; } pMpegSurroundDecoder->pSpatialDec->sacInDataHeadroom = inDataHeadroom; *outDataHeadroom = (INT)(8); pMpegSurroundDecoder->pSpatialDec->pConfigCurrent = &pMpegSurroundDecoder ->spatialSpecificConfig[pMpegSurroundDecoder->bsFrameDecode]; newData = pMpegSurroundDecoder->bsFrames[pMpegSurroundDecoder->bsFrameParse] .newBsData; switch (mpegSurroundOperationMode(pMpegSurroundDecoder, 1000)) { case MPEGS_OPMODE_MPS_PAYLOAD: if (pMpegSurroundDecoder ->initFlags[pMpegSurroundDecoder->bsFrameDecode]) { err = initMpegSurroundDecoder(pMpegSurroundDecoder); } if (err == MPS_OK) { if ((pMpegSurroundDecoder ->fOnSync[pMpegSurroundDecoder->bsFrameDecode] != MPEGS_SYNC_COMPLETE) && (pMpegSurroundDecoder->bsFrames[pMpegSurroundDecoder->bsFrameDecode] .bsIndependencyFlag == 1)) { /* We got a valid header and independently decodeable frame data. -> Go to the next sync level and start processing. */ pMpegSurroundDecoder->fOnSync[pMpegSurroundDecoder->bsFrameDecode] = MPEGS_SYNC_COMPLETE; } } else { /* We got a valid config header but found an error while parsing the bitstream. Wait for the next independent frame and apply error conealment in the meantime. */ pMpegSurroundDecoder->fOnSync[pMpegSurroundDecoder->bsFrameDecode] = MPEGS_SYNC_FOUND; controlFlags |= MPEGS_CONCEAL; err = MPS_OK; } /* Concealment: - Bitstream is available, no sync found during bitstream processing - Bitstream is available, sync lost due to corrupted bitstream - Bitstream is available, sync found but no independent frame */ if (pMpegSurroundDecoder->fOnSync[pMpegSurroundDecoder->bsFrameDecode] != MPEGS_SYNC_COMPLETE) { controlFlags |= MPEGS_CONCEAL; } break; case MPEGS_OPMODE_NO_MPS_PAYLOAD: /* Concealment: No bitstream is available */ controlFlags |= MPEGS_CONCEAL; break; default: err = MPS_NOTOK; } if (err != MPS_OK) { goto bail; } /* * Force BypassMode if choosen by user */ if (pMpegSurroundDecoder->mpegSurroundUserParams.bypassMode) { controlFlags |= MPEGS_BYPASSMODE; } if (pMpegSurroundDecoder->initFlags[pMpegSurroundDecoder->bsFrameDecode]) { int startWithDfltCfg = 0; /* * Init with a default configuration if we came here and are still not * initialized. */ if (pMpegSurroundDecoder->initFlags[pMpegSurroundDecoder->bsFrameDecode] & MPEGS_INIT_ENFORCE_REINIT) { /* Get default spatial specific config */ if (FDK_SpatialDecInitDefaultSpatialSpecificConfig( &pMpegSurroundDecoder->spatialSpecificConfigBackup, coreCodec, *nChannels, sampleRate, *frameSize / mpegSurroundDecoder_GetNrOfQmfBands(NULL, sampleRate), pMpegSurroundDecoder->mpegSurroundDecoderLevel, pMpegSurroundDecoder->mpegSurroundUserParams.blindEnable)) { err = MPS_NOTOK; goto bail; } /* Initiate re-initialization, if header has changed */ if (FDK_SpatialDecCompareSpatialSpecificConfigHeader( &pMpegSurroundDecoder->spatialSpecificConfigBackup, &pMpegSurroundDecoder->spatialSpecificConfig [pMpegSurroundDecoder->bsFrameDecode]) == MPS_UNEQUAL_SSC) { pMpegSurroundDecoder->initFlags[pMpegSurroundDecoder->bsFrameDecode] |= MPEGS_INIT_CHANGE_HEADER; SpatialDecInitParserContext(pMpegSurroundDecoder->pSpatialDec); } startWithDfltCfg = 1; } /* First spatial specific config is parsed into spatialSpecificConfigBackup, * second spatialSpecificConfigBackup is copied into spatialSpecificConfig */ err = initMpegSurroundDecoder(pMpegSurroundDecoder); if (startWithDfltCfg) { /* initialized with default config, but no sync found */ /* maybe use updateMpegSurroundDecoderStatus later on */ pMpegSurroundDecoder->fOnSync[pMpegSurroundDecoder->bsFrameDecode] = MPEGS_SYNC_LOST; } /* Since we do not have state MPEGS_SYNC_COMPLETE apply concealment */ controlFlags |= MPEGS_CONCEAL; if (err != MPS_OK) { goto bail; } } /* * Process MPEG Surround Audio */ initControlFlags = controlFlags; /* Check that provided output buffer is large enough. */ if (pMpegSurroundDecoder->pQmfDomain->globalConf.nBandsAnalysis == 0) { err = MPS_UNSUPPORTED_FORMAT; goto bail; } timeDataRequiredSize = (timeDataFrameSize * pMpegSurroundDecoder->pSpatialDec->numOutputChannelsAT * pMpegSurroundDecoder->pQmfDomain->globalConf.nBandsSynthesis) / pMpegSurroundDecoder->pQmfDomain->globalConf.nBandsAnalysis; if (timeDataSize < timeDataRequiredSize) { err = MPS_OUTPUT_BUFFER_TOO_SMALL; goto bail; } if ((pMpegSurroundDecoder->pSpatialDec->pConfigCurrent->syntaxFlags & SACDEC_SYNTAX_USAC) && (pMpegSurroundDecoder->pSpatialDec->stereoConfigIndex > 1)) { FDK_ASSERT(timeDataRequiredSize >= timeDataFrameSize * *nChannels); /* Place samples comprising QMF time slots spaced at QMF output Band raster * to allow slot wise processing */ int timeDataFrameSizeOut = (timeDataFrameSize * pMpegSurroundDecoder->pQmfDomain->globalConf.nBandsSynthesis) / pMpegSurroundDecoder->pQmfDomain->globalConf.nBandsAnalysis; TDinput = pTimeData + timeDataFrameSizeOut - timeDataFrameSize; for (int i = *nChannels - 1; i >= 0; i--) { FDKmemmove(pTimeData + (i + 1) * timeDataFrameSizeOut - timeDataFrameSize, pTimeData + timeDataFrameSize * i, sizeof(PCM_MPS) * timeDataFrameSize); FDKmemclear(pTimeData + i * timeDataFrameSizeOut, sizeof(PCM_MPS) * (timeDataFrameSizeOut - timeDataFrameSize)); } } else { if (pMpegSurroundDecoder->mpegSurroundUseTimeInterface) { FDKmemcpy(input, pTimeData, sizeof(PCM_MPS) * (*nChannels) * (*frameSize)); TDinput = input; } } /* * Process MPEG Surround Audio */ err = SpatialDecApplyFrame( pMpegSurroundDecoder->pSpatialDec, &pMpegSurroundDecoder->bsFrames[pMpegSurroundDecoder->bsFrameDecode], pMpegSurroundDecoder->mpegSurroundUseTimeInterface ? INPUTMODE_TIME : INPUTMODE_QMF_SBR, TDinput, NULL, NULL, pTimeOut, *frameSize, &controlFlags, *nChannels, mapDescr); *nChannels = pMpegSurroundDecoder->pSpatialDec->numOutputChannelsAT; if (err != MPS_OK) { /* A fatal error occured. Go back to start and try again: */ updateMpegSurroundDecoderStatus(pMpegSurroundDecoder, MPEGS_INIT_ENFORCE_REINIT, MPEGS_SYNC_LOST, MPEGS_STOP); *frameSize = 0; /* Declare that framework can not use the data in pTimeOut. */ } else { if (((controlFlags & MPEGS_CONCEAL) && !(initControlFlags & MPEGS_CONCEAL)) || (pMpegSurroundDecoder->pSpatialDec->errInt != MPS_OK)) { /* Account for errors that occured in SpatialDecApplyFrame(): */ updateMpegSurroundDecoderStatus(pMpegSurroundDecoder, MPEGS_INIT_ERROR_PAYLOAD, MPEGS_SYNC_LOST, MPEGS_STOP); } } if ((err == MPS_OK) && !(controlFlags & MPEGS_BYPASSMODE) && !(pMpegSurroundDecoder->upmixType == UPMIX_TYPE_BYPASS)) { SpatialDecChannelProperties(pMpegSurroundDecoder->pSpatialDec, channelType, channelIndices, mapDescr); } bail: if (newData) { /* numParameterSetsPrev shall only be read in the decode process, because of that we can update this state variable here */ pMpegSurroundDecoder->pSpatialDec->numParameterSetsPrev = pMpegSurroundDecoder->bsFrames[pMpegSurroundDecoder->bsFrameDecode] .numParameterSets; } return (err); } /** * \brief Free config dependent MPEG Surround memory. **/ SACDEC_ERROR mpegSurroundDecoder_FreeMem( CMpegSurroundDecoder *pMpegSurroundDecoder) { SACDEC_ERROR err = MPS_OK; if (pMpegSurroundDecoder != NULL) { FDK_SpatialDecClose(pMpegSurroundDecoder->pSpatialDec); pMpegSurroundDecoder->pSpatialDec = NULL; } return err; } /** * \brief Close MPEG Surround decoder. **/ void mpegSurroundDecoder_Close(CMpegSurroundDecoder *pMpegSurroundDecoder) { if (pMpegSurroundDecoder != NULL) { FDK_SpatialDecClose(pMpegSurroundDecoder->pSpatialDec); pMpegSurroundDecoder->pSpatialDec = NULL; for (int i = 0; i < 1; i++) { SpatialDecCloseBsFrame(&pMpegSurroundDecoder->bsFrames[i]); } FDK_FREE_MEMORY_1D(pMpegSurroundDecoder); } } #define SACDEC_VL0 2 #define SACDEC_VL1 1 #define SACDEC_VL2 0 int mpegSurroundDecoder_GetLibInfo(LIB_INFO *info) { int i; if (info == NULL) { return -1; } /* search for next free tab */ for (i = 0; i < FDK_MODULE_LAST; i++) { if (info[i].module_id == FDK_NONE) break; } if (i == FDK_MODULE_LAST) return -1; info += i; info->module_id = FDK_MPSDEC; #ifdef SUPPRESS_BUILD_DATE_INFO info->build_date = ""; info->build_time = ""; #else info->build_date = __DATE__; info->build_time = __TIME__; #endif info->title = "MPEG Surround Decoder"; info->version = LIB_VERSION(SACDEC_VL0, SACDEC_VL1, SACDEC_VL2); LIB_VERSION_STRING(info); info->flags = 0 | CAPF_MPS_LD | CAPF_MPS_USAC | CAPF_MPS_HQ | CAPF_MPS_1CH_IN | CAPF_MPS_2CH_OUT; /* end flags */ return 0; } SACDEC_ERROR mpegSurroundDecoder_SetParam( CMpegSurroundDecoder *pMpegSurroundDecoder, const SACDEC_PARAM param, const INT value) { SACDEC_ERROR err = MPS_OK; SPATIALDEC_PARAM *pUserParams = NULL; /* check decoder handle */ if (pMpegSurroundDecoder != NULL) { /* init local shortcuts */ pUserParams = &pMpegSurroundDecoder->mpegSurroundUserParams; } else { err = MPS_INVALID_HANDLE; /* check the parameter values before exiting. */ } /* apply param value */ switch (param) { case SACDEC_OUTPUT_MODE: switch ((SAC_DEC_OUTPUT_MODE)value) { case SACDEC_OUT_MODE_NORMAL: case SACDEC_OUT_MODE_STEREO: break; default: err = MPS_INVALID_PARAMETER; } if (err == MPS_OK) { if (0) { err = MPS_INVALID_PARAMETER; } else if (pUserParams->outputMode != (UCHAR)value) { pUserParams->outputMode = (UCHAR)value; pMpegSurroundDecoder ->initFlags[pMpegSurroundDecoder->bsFrameDecode] |= MPEGS_INIT_CHANGE_OUTPUT_MODE; } } break; case SACDEC_INTERFACE: if (value < 0 || value > 1) { err = MPS_INVALID_PARAMETER; } if (err != MPS_OK) { goto bail; } if (pMpegSurroundDecoder->mpegSurroundUseTimeInterface != (UCHAR)value) { pMpegSurroundDecoder->mpegSurroundUseTimeInterface = (UCHAR)value; pMpegSurroundDecoder->initFlags[pMpegSurroundDecoder->bsFrameDecode] |= MPEGS_INIT_CHANGE_TIME_FREQ_INTERFACE; } break; case SACDEC_BS_INTERRUPTION: if ((err == MPS_OK) && (value != 0)) { updateMpegSurroundDecoderStatus(pMpegSurroundDecoder, MPEGS_INIT_BS_INTERRUPTION, MPEGS_SYNC_LOST, MPEGS_STOP); } break; case SACDEC_CLEAR_HISTORY: if ((err == MPS_OK) && (value != 0)) { /* Just reset the states and go on. */ updateMpegSurroundDecoderStatus(pMpegSurroundDecoder, MPEGS_INIT_CLEAR_HISTORY, MPEGS_SYNC_LOST, MPEGS_STOP); } break; case SACDEC_CONCEAL_NUM_KEEP_FRAMES: if (value < 0) { /* Check valid value range */ err = MPS_INVALID_PARAMETER; } if (err != MPS_OK) { goto bail; } if (pUserParams->concealNumKeepFrames != (UINT)value) { pUserParams->concealNumKeepFrames = (UINT)value; pMpegSurroundDecoder->initFlags[pMpegSurroundDecoder->bsFrameDecode] |= MPEGS_INIT_CHANGE_CONCEAL_PARAMS; } break; case SACDEC_CONCEAL_FADE_OUT_SLOPE_LENGTH: if (value < 0) { /* Check valid value range */ err = MPS_INVALID_PARAMETER; } if (err != MPS_OK) { goto bail; } if (pUserParams->concealFadeOutSlopeLength != (UINT)value) { pUserParams->concealFadeOutSlopeLength = (UINT)value; pMpegSurroundDecoder->initFlags[pMpegSurroundDecoder->bsFrameDecode] |= MPEGS_INIT_CHANGE_CONCEAL_PARAMS; } break; case SACDEC_CONCEAL_FADE_IN_SLOPE_LENGTH: if (value < 0) { /* Check valid value range */ err = MPS_INVALID_PARAMETER; } if (err != MPS_OK) { goto bail; } if (pUserParams->concealFadeInSlopeLength != (UINT)value) { pUserParams->concealFadeInSlopeLength = (UINT)value; pMpegSurroundDecoder->initFlags[pMpegSurroundDecoder->bsFrameDecode] |= MPEGS_INIT_CHANGE_CONCEAL_PARAMS; } break; case SACDEC_CONCEAL_NUM_RELEASE_FRAMES: if (value < 0) { /* Check valid value range */ err = MPS_INVALID_PARAMETER; } if (err != MPS_OK) { goto bail; } if (pUserParams->concealNumReleaseFrames != (UINT)value) { pUserParams->concealNumReleaseFrames = (UINT)value; pMpegSurroundDecoder->initFlags[pMpegSurroundDecoder->bsFrameDecode] |= MPEGS_INIT_CHANGE_CONCEAL_PARAMS; } break; default: err = MPS_INVALID_PARAMETER; break; } /* switch(param) */ bail: return err; } SACDEC_ERROR mpegSurroundDecoder_IsPseudoLR( CMpegSurroundDecoder *pMpegSurroundDecoder, int *bsPseudoLr) { if (pMpegSurroundDecoder != NULL) { const SPATIAL_SPECIFIC_CONFIG *sscDecode = &pMpegSurroundDecoder ->spatialSpecificConfig[pMpegSurroundDecoder->bsFrameDecode]; *bsPseudoLr = (int)sscDecode->bsPseudoLr; return MPS_OK; } else return MPS_INVALID_HANDLE; } /** * \brief Get the signal delay caused by the MPEG Surround decoder module. **/ UINT mpegSurroundDecoder_GetDelay(const CMpegSurroundDecoder *self) { INT outputDelay = 0; if (self != NULL) { const SPATIAL_SPECIFIC_CONFIG *sscDecode = &self->spatialSpecificConfig[self->bsFrameDecode]; AUDIO_OBJECT_TYPE coreCodec = sscDecode->coreCodec; /* See chapter 4.5 (delay and synchronization) of ISO/IEC FDIS 23003-1 and chapter 5.4.3 of ISO/IEC FDIS 23003-2 for details on the following figures. */ if (coreCodec > AOT_NULL_OBJECT) { if (IS_LOWDELAY(coreCodec)) { /* All low delay variants (ER-AAC-(E)LD): */ outputDelay += 256; } else if (!IS_USAC(coreCodec)) { /* By the method of elimination this is the GA (AAC-LC, HE-AAC, ...) * branch: */ outputDelay += 320 + 257; /* cos to exp delay + QMF synthesis */ if (self->mpegSurroundUseTimeInterface) { outputDelay += 320 + 384; /* QMF and hybrid analysis */ } } } } return (outputDelay); } fdk-aac-2.0.2/libSACdec/src/sac_dec_ssc_struct.h000066400000000000000000000224111404225310700213450ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround decoder library ************************* Author(s): Description: interface - spatial specific config struct *******************************************************************************/ #ifndef SAC_DEC_SSC_STRUCT_H #define SAC_DEC_SSC_STRUCT_H #include "FDK_audio.h" #define MAX_NUM_QMF_BANDS (128) #define MAX_TIME_SLOTS 64 #define MAX_INPUT_CHANNELS 1 #define MAX_OUTPUT_CHANNELS \ 2 /* CAUTION: This does NOT restrict the number of \ output channels exclusively! In addition it \ affects the max number of bitstream and residual channels! */ #define MAX_NUM_OTT (5) #define MAX_NUM_TTT (0) #define MAX_NUM_EXT_TYPES (8) #define MAX_PARAMETER_BANDS (28) #define MAX_PARAMETER_BANDS_LD (23) #define MAX_NUM_XCHANNELS (6) #define MAX_ARBITRARY_TREE_LEVELS (0) typedef enum { /* CAUTION: Do not change enum values! */ SPATIALDEC_FREQ_RES_40 = 40, SPATIALDEC_FREQ_RES_28 = 28, SPATIALDEC_FREQ_RES_23 = 23, SPATIALDEC_FREQ_RES_20 = 20, SPATIALDEC_FREQ_RES_15 = 15, SPATIALDEC_FREQ_RES_14 = 14, SPATIALDEC_FREQ_RES_12 = 12, SPATIALDEC_FREQ_RES_10 = 10, SPATIALDEC_FREQ_RES_9 = 9, SPATIALDEC_FREQ_RES_7 = 7, SPATIALDEC_FREQ_RES_5 = 5, SPATIALDEC_FREQ_RES_4 = 4 } SPATIALDEC_FREQ_RES; typedef enum { SPATIALDEC_QUANT_FINE_DEF = 0, SPATIALDEC_QUANT_EDQ1 = 1, SPATIALDEC_QUANT_EDQ2 = 2, SPATIALDEC_QUANT_RSVD3 = 3, SPATIALDEC_QUANT_RSVD4 = 4, SPATIALDEC_QUANT_RSVD5 = 5, SPATIALDEC_QUANT_RSVD6 = 6, SPATIALDEC_QUANT_RSVD7 = 7 } SPATIALDEC_QUANT_MODE; typedef enum { SPATIALDEC_MODE_RSVD7 = 7 } SPATIALDEC_TREE_CONFIG; typedef enum { SPATIALDEC_GAIN_MODE0 = 0, SPATIALDEC_GAIN_RSVD1 = 1, SPATIALDEC_GAIN_RSVD2 = 2, SPATIALDEC_GAIN_RSVD3 = 3, SPATIALDEC_GAIN_RSVD4 = 4, SPATIALDEC_GAIN_RSVD5 = 5, SPATIALDEC_GAIN_RSVD6 = 6, SPATIALDEC_GAIN_RSVD7 = 7, SPATIALDEC_GAIN_RSVD8 = 8, SPATIALDEC_GAIN_RSVD9 = 9, SPATIALDEC_GAIN_RSVD10 = 10, SPATIALDEC_GAIN_RSVD11 = 11, SPATIALDEC_GAIN_RSVD12 = 12, SPATIALDEC_GAIN_RSVD13 = 13, SPATIALDEC_GAIN_RSVD14 = 14, SPATIALDEC_GAIN_RSVD15 = 15 } SPATIALDEC_FIXED_GAINS; typedef enum { SPATIALDEC_TS_TPNOWHITE = 0, SPATIALDEC_TS_TPWHITE = 1, SPATIALDEC_TS_TES = 2, SPATIALDEC_TS_NOTS = 3, SPATIALDEC_TS_RSVD4 = 4, SPATIALDEC_TS_RSVD5 = 5, SPATIALDEC_TS_RSVD6 = 6, SPATIALDEC_TS_RSVD7 = 7, SPATIALDEC_TS_RSVD8 = 8, SPATIALDEC_TS_RSVD9 = 9, SPATIALDEC_TS_RSVD10 = 10, SPATIALDEC_TS_RSVD11 = 11, SPATIALDEC_TS_RSVD12 = 12, SPATIALDEC_TS_RSVD13 = 13, SPATIALDEC_TS_RSVD14 = 14, SPATIALDEC_TS_RSVD15 = 15 } SPATIALDEC_TS_CONF; typedef enum { SPATIALDEC_DECORR_MODE0 = 0, SPATIALDEC_DECORR_MODE1 = 1, SPATIALDEC_DECORR_MODE2 = 2, SPATIALDEC_DECORR_RSVD3 = 3, SPATIALDEC_DECORR_RSVD4 = 4, SPATIALDEC_DECORR_RSVD5 = 5, SPATIALDEC_DECORR_RSVD6 = 6, SPATIALDEC_DECORR_RSVD7 = 7, SPATIALDEC_DECORR_RSVD8 = 8, SPATIALDEC_DECORR_RSVD9 = 9, SPATIALDEC_DECORR_RSVD10 = 10, SPATIALDEC_DECORR_RSVD11 = 11, SPATIALDEC_DECORR_RSVD12 = 12, SPATIALDEC_DECORR_RSVD13 = 13, SPATIALDEC_DECORR_RSVD14 = 14, SPATIALDEC_DECORR_RSVD15 = 15 } SPATIALDEC_DECORR_CONF; typedef struct T_SPATIALDEC_OTT_CONF { int nOttBands; } SPATIALDEC_OTT_CONF; typedef struct T_SPATIALDEC_RESIDUAL_CONF { int bResidualPresent; int nResidualBands; } SPATIALDEC_RESIDUAL_CONF; typedef struct T_SPATIAL_SPECIFIC_CONFIG { UINT syntaxFlags; int samplingFreq; int nTimeSlots; SPATIALDEC_FREQ_RES freqRes; SPATIALDEC_TREE_CONFIG treeConfig; SPATIALDEC_QUANT_MODE quantMode; int bArbitraryDownmix; int bResidualCoding; SPATIALDEC_FIXED_GAINS bsFixedGainDMX; SPATIALDEC_TS_CONF tempShapeConfig; SPATIALDEC_DECORR_CONF decorrConfig; int nInputChannels; /* derived from treeConfig */ int nOutputChannels; /* derived from treeConfig */ /* ott config */ int nOttBoxes; /* derived from treeConfig */ SPATIALDEC_OTT_CONF OttConfig[MAX_NUM_OTT]; /* dimension nOttBoxes */ /* ttt config */ int nTttBoxes; /* derived from treeConfig */ /* residual config */ SPATIALDEC_RESIDUAL_CONF ResidualConfig[MAX_NUM_OTT + MAX_NUM_TTT]; /* dimension (nOttBoxes + nTttBoxes) */ int sacExtCnt; int sacExtType[MAX_NUM_EXT_TYPES]; int envQuantMode; AUDIO_OBJECT_TYPE coreCodec; UCHAR stereoConfigIndex; UCHAR coreSbrFrameLengthIndex; /* Table 70 in ISO/IEC FDIS 23003-3:2011 */ UCHAR bsHighRateMode; UCHAR bsDecorrType; UCHAR bsPseudoLr; UCHAR bsPhaseCoding; UCHAR bsOttBandsPhasePresent; int bsOttBandsPhase; SCHAR ottCLDdefault[MAX_NUM_OTT]; UCHAR numOttBandsIPD; UCHAR bitstreamOttBands[MAX_NUM_OTT]; UCHAR numOttBands[MAX_NUM_OTT]; } SPATIAL_SPECIFIC_CONFIG; #endif fdk-aac-2.0.2/libSACdec/src/sac_process.cpp000066400000000000000000001175021404225310700203550ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround decoder library ************************* Author(s): Description: SAC Processing *******************************************************************************/ /* data structures and interfaces for spatial audio reference software */ #include "sac_process.h" #include "sac_bitdec.h" #include "sac_calcM1andM2.h" #include "sac_smoothing.h" #include "sac_rom.h" #include "sac_dec_errorcodes.h" #include "FDK_trigFcts.h" #include "FDK_decorrelate.h" #define SAC_DEC_APPLY_M2_SCALE(spec, s) ((spec) >> (-(s))) /** * \brief Linear interpolation between two parameter values. * a*alpha + b*(1-alpha) * = a*alpha + b - b*alpha * * \param alpha Weighting factor. * \param a Parameter a. * \param b Parameter b. * * \return Interpolated parameter value. */ FDK_INLINE FIXP_DBL interpolateParameter(const FIXP_SGL alpha, const FIXP_DBL a, const FIXP_DBL b) { return (b - fMult(alpha, b) + fMult(alpha, a)); } /** * \brief Map MPEG Surround channel indices to MPEG 4 PCE like channel indices. * \param self Spatial decoder handle. * \param ch MPEG Surround channel index. * \return MPEG 4 PCE style channel index, corresponding to the given MPEG * Surround channel index. */ static UINT mapChannel(spatialDec *self, UINT ch) { static const UCHAR chanelIdx[][8] = { {0, 1, 2, 3, 4, 5, 6, 7}, /* binaural, TREE_212, arbitrary tree */ }; int idx = 0; return (chanelIdx[idx][ch]); } FIXP_DBL getChGain(spatialDec *self, UINT ch, INT *scale) { /* init no gain modifier */ FIXP_DBL gain = 0x80000000; *scale = 0; if ((!isTwoChMode(self->upmixType)) && (self->upmixType != UPMIXTYPE_BYPASS)) { if ((ch == 0) || (ch == 1) || (ch == 2)) { /* no modifier */ } } return gain; } SACDEC_ERROR SpatialDecQMFAnalysis(spatialDec *self, const PCM_MPS *inData, const INT ts, const INT bypassMode, FIXP_DBL **qmfReal, FIXP_DBL **qmfImag, const int numInputChannels) { SACDEC_ERROR err = MPS_OK; int ch, offset; offset = self->pQmfDomain->globalConf.nBandsSynthesis * self->pQmfDomain->globalConf.nQmfTimeSlots; { for (ch = 0; ch < numInputChannels; ch++) { const PCM_MPS *inSamples = &inData[ts * self->pQmfDomain->globalConf.nBandsAnalysis]; FIXP_DBL *pQmfRealAnalysis = qmfReal[ch]; /* no delay in blind mode */ FIXP_DBL *pQmfImagAnalysis = qmfImag[ch]; CalculateSpaceAnalysisQmf(&self->pQmfDomain->QmfDomainIn[ch].fb, inSamples + (ch * offset), pQmfRealAnalysis, pQmfImagAnalysis); if (!isTwoChMode(self->upmixType) && !bypassMode) { int i; for (i = 0; i < self->qmfBands; i++) { qmfReal[ch][i] = fMult( scaleValueSaturate(qmfReal[ch][i], self->sacInDataHeadroom - (1)), self->clipProtectGain__FDK); qmfImag[ch][i] = fMult( scaleValueSaturate(qmfImag[ch][i], self->sacInDataHeadroom - (1)), self->clipProtectGain__FDK); } } } } self->qmfInputDelayBufPos = (self->qmfInputDelayBufPos + 1) % self->pc_filterdelay; return err; } SACDEC_ERROR SpatialDecFeedQMF(spatialDec *self, FIXP_DBL **qmfInDataReal, FIXP_DBL **qmfInDataImag, const INT ts, const INT bypassMode, FIXP_DBL **qmfReal__FDK, FIXP_DBL **qmfImag__FDK, const INT numInputChannels) { SACDEC_ERROR err = MPS_OK; int ch; { for (ch = 0; ch < numInputChannels; ch++) { FIXP_DBL *pQmfRealAnalysis = qmfReal__FDK[ch]; /* no delay in blind mode */ FIXP_DBL *pQmfImagAnalysis = qmfImag__FDK[ch]; /* Write Input data to pQmfRealAnalysis. */ if (self->bShareDelayWithSBR) { FDK_QmfDomain_GetSlot(&self->pQmfDomain->QmfDomainIn[ch], ts + HYBRID_FILTER_DELAY, 0, MAX_QMF_BANDS_TO_HYBRID, pQmfRealAnalysis, pQmfImagAnalysis, 15 + (1)); FDK_QmfDomain_GetSlot(&self->pQmfDomain->QmfDomainIn[ch], ts, MAX_QMF_BANDS_TO_HYBRID, self->qmfBands, pQmfRealAnalysis, pQmfImagAnalysis, 15 + (1)); } else { FDK_QmfDomain_GetSlot(&self->pQmfDomain->QmfDomainIn[ch], ts, 0, self->qmfBands, pQmfRealAnalysis, pQmfImagAnalysis, 15 + (1)); } if (ts == self->pQmfDomain->globalConf.nQmfTimeSlots - 1) { /* Is currently also needed in case we dont have any overlap. We need to * save lb_scale to ov_lb_scale */ FDK_QmfDomain_SaveOverlap(&self->pQmfDomain->QmfDomainIn[ch], 0); } /* Apply clip protection to output. */ if (!isTwoChMode(self->upmixType) && !bypassMode) { int i; for (i = 0; i < self->qmfBands; i++) { qmfReal__FDK[ch][i] = fMult(qmfReal__FDK[ch][i], self->clipProtectGain__FDK); qmfImag__FDK[ch][i] = fMult(qmfImag__FDK[ch][i], self->clipProtectGain__FDK); } } } /* End of loop over numInputChannels */ } self->qmfInputDelayBufPos = (self->qmfInputDelayBufPos + 1) % self->pc_filterdelay; return err; } /******************************************************************************* Functionname: SpatialDecHybridAnalysis ******************************************************************************* Description: Arguments: Input: float** pointers[4] leftReal, leftIm, rightReal, rightIm Output: float self->qmfInputReal[MAX_INPUT_CHANNELS][MAX_TIME_SLOTS][MAX_QMF_BANDS]; float self->qmfInputImag[MAX_INPUT_CHANNELS][MAX_TIME_SLOTS][MAX_QMF_BANDS]; float self->hybInputReal[MAX_INPUT_CHANNELS][MAX_TIME_SLOTS][MAX_HYBRID_BANDS]; float self->hybInputImag[MAX_INPUT_CHANNELS][MAX_TIME_SLOTS][MAX_HYBRID_BANDS]; *******************************************************************************/ SACDEC_ERROR SpatialDecHybridAnalysis(spatialDec *self, FIXP_DBL **qmfInputReal, FIXP_DBL **qmfInputImag, FIXP_DBL **hybOutputReal, FIXP_DBL **hybOutputImag, const INT ts, const INT numInputChannels) { SACDEC_ERROR err = MPS_OK; int ch; for (ch = 0; ch < numInputChannels; ch++) /* hybrid filtering for down-mix signals */ { if (self->pConfigCurrent->syntaxFlags & SACDEC_SYNTAX_LD) { int k; /* No hybrid filtering. Just copy the QMF data. */ for (k = 0; k < self->hybridBands; k += 1) { hybOutputReal[ch][k] = qmfInputReal[ch][k]; hybOutputImag[ch][k] = qmfInputImag[ch][k]; } } else { self->hybridAnalysis[ch].hfMode = self->bShareDelayWithSBR; if (self->stereoConfigIndex == 3) FDK_ASSERT(self->hybridAnalysis[ch].hfMode == 0); FDKhybridAnalysisApply(&self->hybridAnalysis[ch], qmfInputReal[ch], qmfInputImag[ch], hybOutputReal[ch], hybOutputImag[ch]); } } if ((self->pConfigCurrent->syntaxFlags & SACDEC_SYNTAX_USAC) && self->residualCoding) { self->hybridAnalysis[numInputChannels].hfMode = 0; FDKhybridAnalysisApply( &self->hybridAnalysis[numInputChannels], self->qmfResidualReal__FDK[0][0], self->qmfResidualImag__FDK[0][0], self->hybResidualReal__FDK[0], self->hybResidualImag__FDK[0]); } return err; } SACDEC_ERROR SpatialDecCreateX(spatialDec *self, FIXP_DBL **hybInputReal, FIXP_DBL **hybInputImag, FIXP_DBL **pxReal, FIXP_DBL **pxImag) { SACDEC_ERROR err = MPS_OK; int row; /* Creating wDry */ for (row = 0; row < self->numInputChannels; row++) { /* pointer to direct signals */ pxReal[row] = hybInputReal[row]; pxImag[row] = hybInputImag[row]; } return err; } static void M2ParamToKernelMult(FIXP_SGL *RESTRICT pKernel, FIXP_DBL *RESTRICT Mparam, FIXP_DBL *RESTRICT MparamPrev, int *RESTRICT pWidth, FIXP_SGL alpha__FDK, int nBands) { int pb; for (pb = 0; pb < nBands; pb++) { FIXP_SGL tmp = FX_DBL2FX_SGL( interpolateParameter(alpha__FDK, Mparam[pb], MparamPrev[pb])); int i = pWidth[pb]; if (i & 1) *pKernel++ = tmp; if (i & 2) { *pKernel++ = tmp; *pKernel++ = tmp; } for (i >>= 2; i--;) { *pKernel++ = tmp; *pKernel++ = tmp; *pKernel++ = tmp; *pKernel++ = tmp; } } } SACDEC_ERROR SpatialDecApplyM1_CreateW_Mode212( spatialDec *self, const SPATIAL_BS_FRAME *frame, FIXP_DBL **xReal, FIXP_DBL **xImag, FIXP_DBL **vReal, FIXP_DBL **vImag) { SACDEC_ERROR err = MPS_OK; int res; FIXP_DBL *decorrInReal = vReal[0]; FIXP_DBL *decorrInImag = vImag[0]; /* M1 does not do anything in 212 mode, so use simplified processing */ FDK_ASSERT(self->numVChannels == 2); FDK_ASSERT(self->numDirektSignals == 1); FDK_ASSERT(self->numDecorSignals == 1); FDKmemcpy(vReal[0], xReal[0], self->hybridBands * sizeof(FIXP_DBL)); FDKmemcpy(vImag[0], xImag[0], self->hybridBands * sizeof(FIXP_DBL)); if (isTsdActive(frame->TsdData)) { /* Generate v_{x,nonTr} as input for allpass based decorrelator */ TsdGenerateNonTr(self->hybridBands, frame->TsdData, self->TsdTs, vReal[0], vImag[0], vReal[1], vImag[1], &decorrInReal, &decorrInImag); } /* - Decorrelate */ res = SpatialDecGetResidualIndex(self, 1); if (FDKdecorrelateApply(&self->apDecor[0], decorrInReal, decorrInImag, vReal[1], vImag[1], self->param2hyb[self->residualBands[res]])) { return MPS_NOTOK; } if (isTsdActive(frame->TsdData)) { /* Generate v_{x,Tr}, apply transient decorrelator and add to allpass based * decorrelator output */ TsdApply(self->hybridBands, frame->TsdData, &self->TsdTs, vReal[0], /* input: v_x */ vImag[0], vReal[1], /* input: d_{x,nonTr}; output: d_{x,nonTr} + d_{x,Tr} */ vImag[1]); } /* Write residual signal in approriate parameter bands */ if (self->residualBands[res] > 0) { int stopBand = self->param2hyb[self->residualBands[res]]; FDKmemcpy(vReal[1], self->hybResidualReal__FDK[res], fixMin(stopBand, self->hybridBands) * sizeof(FIXP_DBL)); FDKmemcpy(vImag[1], self->hybResidualImag__FDK[res], fixMin(stopBand, self->hybridBands) * sizeof(FIXP_DBL)); } /* (self->residualBands[res]>0) */ return err; } SACDEC_ERROR SpatialDecApplyM2_Mode212(spatialDec *self, INT ps, const FIXP_SGL alpha, FIXP_DBL **wReal, FIXP_DBL **wImag, FIXP_DBL **hybOutputRealDry, FIXP_DBL **hybOutputImagDry) { SACDEC_ERROR err = MPS_OK; INT row; INT *pWidth = self->kernels_width; /* for stereoConfigIndex == 3 case hybridBands is < 71 */ INT pb_max = self->kernels[self->hybridBands - 1] + 1; INT max_row = self->numOutputChannels; INT M2_exp = 0; if (self->residualCoding) M2_exp = 3; for (row = 0; row < max_row; row++) // 2 times { FIXP_DBL *Mparam0 = self->M2Real__FDK[row][0]; FIXP_DBL *Mparam1 = self->M2Real__FDK[row][1]; FIXP_DBL *MparamPrev0 = self->M2RealPrev__FDK[row][0]; FIXP_DBL *MparamPrev1 = self->M2RealPrev__FDK[row][1]; FIXP_DBL *RESTRICT pHybOutRealDry = hybOutputRealDry[row]; FIXP_DBL *RESTRICT pHybOutImagDry = hybOutputImagDry[row]; FIXP_DBL *RESTRICT pWReal0 = wReal[0]; FIXP_DBL *RESTRICT pWReal1 = wReal[1]; FIXP_DBL *RESTRICT pWImag0 = wImag[0]; FIXP_DBL *RESTRICT pWImag1 = wImag[1]; for (INT pb = 0; pb < pb_max; pb++) { FIXP_DBL tmp0, tmp1; tmp0 = interpolateParameter(alpha, Mparam0[pb], MparamPrev0[pb]); tmp1 = interpolateParameter(alpha, Mparam1[pb], MparamPrev1[pb]); INT i = pWidth[pb]; do // about 3-4 times { FIXP_DBL var0, var1, real, imag; var0 = *pWReal0++; var1 = *pWReal1++; real = fMultDiv2(var0, tmp0); var0 = *pWImag0++; real = fMultAddDiv2(real, var1, tmp1); var1 = *pWImag1++; imag = fMultDiv2(var0, tmp0); *pHybOutRealDry++ = real << (1 + M2_exp); imag = fMultAddDiv2(imag, var1, tmp1); *pHybOutImagDry++ = imag << (1 + M2_exp); } while (--i != 0); } } return err; } SACDEC_ERROR SpatialDecApplyM2_Mode212_ResidualsPlusPhaseCoding( spatialDec *self, INT ps, const FIXP_SGL alpha, FIXP_DBL **wReal, FIXP_DBL **wImag, FIXP_DBL **hybOutputRealDry, FIXP_DBL **hybOutputImagDry) { SACDEC_ERROR err = MPS_OK; INT row; INT scale_param_m2; INT *pWidth = self->kernels_width; INT pb_max = self->kernels[self->hybridBands - 1] + 1; scale_param_m2 = SCALE_PARAM_M2_212_PRED + SCALE_DATA_APPLY_M2; for (row = 0; row < self->numM2rows; row++) { INT qs, pb; FIXP_DBL *RESTRICT pWReal0 = wReal[0]; FIXP_DBL *RESTRICT pWImag0 = wImag[0]; FIXP_DBL *RESTRICT pWReal1 = wReal[1]; FIXP_DBL *RESTRICT pWImag1 = wImag[1]; FIXP_DBL *MReal0 = self->M2Real__FDK[row][0]; FIXP_DBL *MImag0 = self->M2Imag__FDK[row][0]; FIXP_DBL *MReal1 = self->M2Real__FDK[row][1]; FIXP_DBL *MRealPrev0 = self->M2RealPrev__FDK[row][0]; FIXP_DBL *MImagPrev0 = self->M2ImagPrev__FDK[row][0]; FIXP_DBL *MRealPrev1 = self->M2RealPrev__FDK[row][1]; FIXP_DBL *RESTRICT pHybOutRealDry = hybOutputRealDry[row]; FIXP_DBL *RESTRICT pHybOutImagDry = hybOutputImagDry[row]; FDK_ASSERT(!(self->pConfigCurrent->syntaxFlags & SACDEC_SYNTAX_LD)); FDK_ASSERT((pWidth[0] + pWidth[1]) >= 3); for (pb = 0, qs = 3; pb < 2; pb++) { INT s; FIXP_DBL maxVal; FIXP_DBL mReal1; FIXP_DBL mReal0, mImag0; FIXP_DBL iReal0, iImag0, iReal1; iReal0 = interpolateParameter(alpha, MReal0[pb], MRealPrev0[pb]); iImag0 = -interpolateParameter(alpha, MImag0[pb], MImagPrev0[pb]); iReal1 = interpolateParameter(alpha, MReal1[pb], MRealPrev1[pb]); maxVal = fAbs(iReal0) | fAbs(iImag0); maxVal |= fAbs(iReal1); s = fMax(CntLeadingZeros(maxVal) - 1, 0); s = fMin(s, scale_param_m2); mReal0 = iReal0 << s; mImag0 = iImag0 << s; mReal1 = iReal1 << s; s = scale_param_m2 - s; INT i = pWidth[pb]; do { FIXP_DBL real, imag, wReal0, wImag0, wReal1, wImag1; wReal0 = *pWReal0++; wImag0 = *pWImag0++; wReal1 = *pWReal1++; wImag1 = *pWImag1++; cplxMultDiv2(&real, &imag, wReal0, wImag0, mReal0, mImag0); *pHybOutRealDry++ = fMultAddDiv2(real, wReal1, mReal1) << s; *pHybOutImagDry++ = fMultAddDiv2(imag, wImag1, mReal1) << s; if (qs > 0) { mImag0 = -mImag0; qs--; } } while (--i != 0); } for (; pb < pb_max; pb++) { INT s; FIXP_DBL maxVal; FIXP_SGL mReal1; FIXP_SGL mReal0, mImag0; FIXP_DBL iReal0, iImag0, iReal1; iReal0 = interpolateParameter(alpha, MReal0[pb], MRealPrev0[pb]); iImag0 = interpolateParameter(alpha, MImag0[pb], MImagPrev0[pb]); iReal1 = interpolateParameter(alpha, MReal1[pb], MRealPrev1[pb]); maxVal = fAbs(iReal0) | fAbs(iImag0); maxVal |= fAbs(iReal1); s = fMax(CntLeadingZeros(maxVal) - 1, 0); s = fMin(s, scale_param_m2); mReal0 = FX_DBL2FX_SGL(iReal0 << s); mImag0 = FX_DBL2FX_SGL(iImag0 << s); mReal1 = FX_DBL2FX_SGL(iReal1 << s); s = scale_param_m2 - s; INT i = pWidth[pb]; do { FIXP_DBL real, imag, wReal0, wImag0, wReal1, wImag1; wReal0 = *pWReal0++; wImag0 = *pWImag0++; wReal1 = *pWReal1++; wImag1 = *pWImag1++; cplxMultDiv2(&real, &imag, wReal0, wImag0, mReal0, mImag0); *pHybOutRealDry++ = fMultAddDiv2(real, wReal1, mReal1) << s; *pHybOutImagDry++ = fMultAddDiv2(imag, wImag1, mReal1) << s; } while (--i != 0); } } return err; } SACDEC_ERROR SpatialDecApplyM2(spatialDec *self, INT ps, const FIXP_SGL alpha, FIXP_DBL **wReal, FIXP_DBL **wImag, FIXP_DBL **hybOutputRealDry, FIXP_DBL **hybOutputImagDry, FIXP_DBL **hybOutputRealWet, FIXP_DBL **hybOutputImagWet) { SACDEC_ERROR err = MPS_OK; { int qs, row, col; int complexHybBands; int complexParBands; int scale_param_m2 = 0; int toolsDisabled; UCHAR activParamBands; FIXP_DBL *RESTRICT pWReal, *RESTRICT pWImag, *RESTRICT pHybOutRealDry, *RESTRICT pHybOutImagDry, *RESTRICT pHybOutRealWet, *RESTRICT pHybOutImagWet; C_ALLOC_SCRATCH_START(pKernel, FIXP_SGL, MAX_HYBRID_BANDS); /* The wet signal is added to the dry signal directly in applyM2 if GES and * STP are disabled */ toolsDisabled = ((self->tempShapeConfig == 1) || (self->tempShapeConfig == 2)) ? 0 : 1; { complexHybBands = self->hybridBands; complexParBands = self->numParameterBands; } FDKmemclear(hybOutputImagDry[0], self->createParams.maxNumOutputChannels * self->createParams.maxNumCmplxHybBands * sizeof(FIXP_DBL)); FDKmemclear(hybOutputRealDry[0], self->createParams.maxNumOutputChannels * self->createParams.maxNumHybridBands * sizeof(FIXP_DBL)); if (!toolsDisabled) { FDKmemclear(hybOutputRealWet[0], self->createParams.maxNumOutputChannels * self->createParams.maxNumHybridBands * sizeof(FIXP_DBL)); FDKmemclear(hybOutputImagWet[0], self->createParams.maxNumOutputChannels * self->createParams.maxNumCmplxHybBands * sizeof(FIXP_DBL)); } if (self->phaseCoding == 3) { scale_param_m2 = -(SCALE_DATA_APPLY_M2_PC - 1); } for (row = 0; row < self->numM2rows; row++) { pHybOutRealDry = hybOutputRealDry[row]; pHybOutImagDry = hybOutputImagDry[row]; if (toolsDisabled) { pHybOutRealWet = hybOutputRealDry[row]; pHybOutImagWet = hybOutputImagDry[row]; } else { pHybOutRealWet = hybOutputRealWet[row]; pHybOutImagWet = hybOutputImagWet[row]; } for (col = 0; col < self->numDirektSignals; col++) { if (self->pActivM2ParamBands == 0) { /* default setting, calculate all rows and columns */ activParamBands = 1; } else { if (self->pActivM2ParamBands[MAX_M2_INPUT * row + col]) /* table with activ and inactiv bands exists for current configuration */ activParamBands = 1; else activParamBands = 0; } if (activParamBands) { pWReal = wReal[col]; pWImag = wImag[col]; M2ParamToKernelMult(pKernel, self->M2Real__FDK[row][col], self->M2RealPrev__FDK[row][col], self->kernels_width, alpha, self->numParameterBands); if (1 && (self->phaseCoding != 3)) { /* direct signals */ { /* only one sample will be assigned to each row, hence * accumulation is not neccessary; that is valid for all * configurations */ for (qs = 0; qs < complexHybBands; qs++) { pHybOutRealDry[qs] = fMult(pWReal[qs], pKernel[qs]); pHybOutImagDry[qs] = fMult(pWImag[qs], pKernel[qs]); } } } else { /* isBinauralMode(self->upmixType) */ for (qs = 0; qs < complexHybBands; qs++) { pHybOutRealDry[qs] += SAC_DEC_APPLY_M2_SCALE( fMultDiv2(pWReal[qs], pKernel[qs]), scale_param_m2); pHybOutImagDry[qs] += SAC_DEC_APPLY_M2_SCALE( fMultDiv2(pWImag[qs], pKernel[qs]), scale_param_m2); } M2ParamToKernelMult(pKernel, self->M2Imag__FDK[row][col], self->M2ImagPrev__FDK[row][col], self->kernels_width, alpha, complexParBands); /* direct signals sign is -1 for qs = 0,2 */ pHybOutRealDry[0] += SAC_DEC_APPLY_M2_SCALE( fMultDiv2(pWImag[0], pKernel[0]), scale_param_m2); pHybOutImagDry[0] -= SAC_DEC_APPLY_M2_SCALE( fMultDiv2(pWReal[0], pKernel[0]), scale_param_m2); pHybOutRealDry[2] += SAC_DEC_APPLY_M2_SCALE( fMultDiv2(pWImag[2], pKernel[2]), scale_param_m2); pHybOutImagDry[2] -= SAC_DEC_APPLY_M2_SCALE( fMultDiv2(pWReal[2], pKernel[2]), scale_param_m2); /* direct signals sign is +1 for qs = 1,3,4,5,...,complexHybBands */ pHybOutRealDry[1] -= SAC_DEC_APPLY_M2_SCALE( fMultDiv2(pWImag[1], pKernel[1]), scale_param_m2); pHybOutImagDry[1] += SAC_DEC_APPLY_M2_SCALE( fMultDiv2(pWReal[1], pKernel[1]), scale_param_m2); for (qs = 3; qs < complexHybBands; qs++) { pHybOutRealDry[qs] -= SAC_DEC_APPLY_M2_SCALE( fMultDiv2(pWImag[qs], pKernel[qs]), scale_param_m2); pHybOutImagDry[qs] += SAC_DEC_APPLY_M2_SCALE( fMultDiv2(pWReal[qs], pKernel[qs]), scale_param_m2); } } /* self->upmixType */ } /* if (activParamBands) */ } /* self->numDirektSignals */ for (; col < self->numVChannels; col++) { if (self->pActivM2ParamBands == 0) { /* default setting, calculate all rows and columns */ activParamBands = 1; } else { if (self->pActivM2ParamBands[MAX_M2_INPUT * row + col]) /* table with activ and inactiv bands exists for current configuration */ activParamBands = 1; else activParamBands = 0; } if (activParamBands) { int resBandIndex; int resHybIndex; resBandIndex = self->residualBands[SpatialDecGetResidualIndex(self, col)]; resHybIndex = self->param2hyb[resBandIndex]; pWReal = wReal[col]; pWImag = wImag[col]; M2ParamToKernelMult(pKernel, self->M2Real__FDK[row][col], self->M2RealPrev__FDK[row][col], self->kernels_width, alpha, self->numParameterBands); if (1 && (self->phaseCoding != 3)) { /* residual signals */ for (qs = 0; qs < resHybIndex; qs++) { pHybOutRealDry[qs] += fMult(pWReal[qs], pKernel[qs]); pHybOutImagDry[qs] += fMult(pWImag[qs], pKernel[qs]); } /* decor signals */ for (; qs < complexHybBands; qs++) { pHybOutRealWet[qs] += fMult(pWReal[qs], pKernel[qs]); pHybOutImagWet[qs] += fMult(pWImag[qs], pKernel[qs]); } } else { /* self->upmixType */ /* residual signals */ FIXP_DBL *RESTRICT pHybOutReal; FIXP_DBL *RESTRICT pHybOutImag; for (qs = 0; qs < resHybIndex; qs++) { pHybOutRealDry[qs] += SAC_DEC_APPLY_M2_SCALE( fMultDiv2(pWReal[qs], pKernel[qs]), scale_param_m2); pHybOutImagDry[qs] += SAC_DEC_APPLY_M2_SCALE( fMultDiv2(pWImag[qs], pKernel[qs]), scale_param_m2); } /* decor signals */ for (; qs < complexHybBands; qs++) { pHybOutRealWet[qs] += SAC_DEC_APPLY_M2_SCALE( fMultDiv2(pWReal[qs], pKernel[qs]), scale_param_m2); pHybOutImagWet[qs] += SAC_DEC_APPLY_M2_SCALE( fMultDiv2(pWImag[qs], pKernel[qs]), scale_param_m2); } M2ParamToKernelMult(pKernel, self->M2Imag__FDK[row][col], self->M2ImagPrev__FDK[row][col], self->kernels_width, alpha, complexParBands); /* direct signals sign is -1 for qs = 0,2 */ /* direct signals sign is +1 for qs = 1,3.. */ if (toolsDisabled) { pHybOutRealDry[0] += SAC_DEC_APPLY_M2_SCALE( fMultDiv2(pWImag[0], pKernel[0]), scale_param_m2); pHybOutImagDry[0] -= SAC_DEC_APPLY_M2_SCALE( fMultDiv2(pWReal[0], pKernel[0]), scale_param_m2); pHybOutRealDry[1] -= SAC_DEC_APPLY_M2_SCALE( fMultDiv2(pWImag[1], pKernel[1]), scale_param_m2); pHybOutImagDry[1] += SAC_DEC_APPLY_M2_SCALE( fMultDiv2(pWReal[1], pKernel[1]), scale_param_m2); pHybOutRealDry[2] += SAC_DEC_APPLY_M2_SCALE( fMultDiv2(pWImag[2], pKernel[2]), scale_param_m2); pHybOutImagDry[2] -= SAC_DEC_APPLY_M2_SCALE( fMultDiv2(pWReal[2], pKernel[2]), scale_param_m2); } else { pHybOutReal = &pHybOutRealDry[0]; pHybOutImag = &pHybOutImagDry[0]; if (0 == resHybIndex) { pHybOutReal = &pHybOutRealWet[0]; pHybOutImag = &pHybOutImagWet[0]; } pHybOutReal[0] += SAC_DEC_APPLY_M2_SCALE( fMultDiv2(pWImag[0], pKernel[0]), scale_param_m2); pHybOutImag[0] -= SAC_DEC_APPLY_M2_SCALE( fMultDiv2(pWReal[0], pKernel[0]), scale_param_m2); if (1 == resHybIndex) { pHybOutReal = &pHybOutRealWet[0]; pHybOutImag = &pHybOutImagWet[0]; } pHybOutReal[1] -= SAC_DEC_APPLY_M2_SCALE( fMultDiv2(pWImag[1], pKernel[1]), scale_param_m2); pHybOutImag[1] += SAC_DEC_APPLY_M2_SCALE( fMultDiv2(pWReal[1], pKernel[1]), scale_param_m2); if (2 == resHybIndex) { pHybOutReal = &pHybOutRealWet[0]; pHybOutImag = &pHybOutImagWet[0]; } pHybOutReal[2] += SAC_DEC_APPLY_M2_SCALE( fMultDiv2(pWImag[2], pKernel[2]), scale_param_m2); pHybOutImag[2] -= SAC_DEC_APPLY_M2_SCALE( fMultDiv2(pWReal[2], pKernel[2]), scale_param_m2); } for (qs = 3; qs < resHybIndex; qs++) { pHybOutRealDry[qs] -= SAC_DEC_APPLY_M2_SCALE( fMultDiv2(pWImag[qs], pKernel[qs]), scale_param_m2); pHybOutImagDry[qs] += SAC_DEC_APPLY_M2_SCALE( fMultDiv2(pWReal[qs], pKernel[qs]), scale_param_m2); } /* decor signals */ for (; qs < complexHybBands; qs++) { pHybOutRealWet[qs] -= SAC_DEC_APPLY_M2_SCALE( fMultDiv2(pWImag[qs], pKernel[qs]), scale_param_m2); pHybOutImagWet[qs] += SAC_DEC_APPLY_M2_SCALE( fMultDiv2(pWReal[qs], pKernel[qs]), scale_param_m2); } } /* self->upmixType */ } /* if (activParamBands) { */ } /* self->numVChannels */ if (self->phaseCoding == 3) { scaleValuesSaturate(pHybOutRealDry, complexHybBands, SCALE_PARAM_M2_212_PRED + SCALE_DATA_APPLY_M2_PC); scaleValuesSaturate(pHybOutImagDry, complexHybBands, SCALE_PARAM_M2_212_PRED + SCALE_DATA_APPLY_M2_PC); if (!toolsDisabled) { scaleValuesSaturate(pHybOutRealWet, complexHybBands, SCALE_PARAM_M2_212_PRED + SCALE_DATA_APPLY_M2_PC); scaleValuesSaturate(pHybOutImagWet, complexHybBands, SCALE_PARAM_M2_212_PRED + SCALE_DATA_APPLY_M2_PC); } } } C_ALLOC_SCRATCH_END(pKernel, FIXP_SGL, MAX_HYBRID_BANDS); } return err; } SACDEC_ERROR SpatialDecSynthesis(spatialDec *self, const INT ts, FIXP_DBL **hybOutputReal, FIXP_DBL **hybOutputImag, PCM_MPS *timeOut, const INT numInputChannels, const FDK_channelMapDescr *const mapDescr) { SACDEC_ERROR err = MPS_OK; int ch; int stride, offset; stride = self->numOutputChannelsAT; offset = 1; PCM_MPS *pTimeOut__FDK = &timeOut[stride * self->pQmfDomain->globalConf.nBandsSynthesis * ts]; C_ALLOC_SCRATCH_START(pQmfReal, FIXP_DBL, QMF_MAX_SYNTHESIS_BANDS); C_ALLOC_SCRATCH_START(pQmfImag, FIXP_DBL, QMF_MAX_SYNTHESIS_BANDS); for (ch = 0; ch < self->numOutputChannelsAT; ch++) { if (self->pConfigCurrent->syntaxFlags & SACDEC_SYNTAX_LD) { int k; /* No hybrid filtering. Just copy the QMF data. */ for (k = 0; k < self->hybridBands; k += 1) { pQmfReal[k] = hybOutputReal[ch][k]; pQmfImag[k] = hybOutputImag[ch][k]; } } else { FDKhybridSynthesisApply(&self->hybridSynthesis[ch], hybOutputReal[ch], hybOutputImag[ch], pQmfReal, pQmfImag); } /* Map channel indices from MPEG Surround -> PCE style -> channelMapping[] */ FDK_ASSERT(self->numOutputChannelsAT <= 6); int outCh = FDK_chMapDescr_getMapValue(mapDescr, mapChannel(self, ch), self->numOutputChannelsAT); { if (self->stereoConfigIndex == 3) { /* MPS -> SBR */ int i; FIXP_DBL *pWorkBufReal, *pWorkBufImag; FDK_ASSERT((self->pQmfDomain->QmfDomainOut[outCh].fb.outGain_m == (FIXP_DBL)0x80000000) && (self->pQmfDomain->QmfDomainOut[outCh].fb.outGain_e == 0)); FDK_QmfDomain_GetWorkBuffer(&self->pQmfDomain->QmfDomainIn[outCh], ts, &pWorkBufReal, &pWorkBufImag); FDK_ASSERT(self->qmfBands <= self->pQmfDomain->QmfDomainIn[outCh].workBuf_nBands); for (i = 0; i < self->qmfBands; i++) { pWorkBufReal[i] = pQmfReal[i]; pWorkBufImag[i] = pQmfImag[i]; } self->pQmfDomain->QmfDomainIn[outCh].scaling.lb_scale = -7; /*-ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK;*/ self->pQmfDomain->QmfDomainIn[outCh].scaling.lb_scale -= self->pQmfDomain->QmfDomainIn[outCh].fb.filterScale; self->pQmfDomain->QmfDomainIn[outCh].scaling.lb_scale -= self->clipProtectGainSF__FDK; self->pQmfDomain->QmfDomainIn[outCh].scaling.lb_scale -= (1); } else { /* Call the QMF synthesis for dry. */ err = CalculateSpaceSynthesisQmf(&self->pQmfDomain->QmfDomainOut[outCh], pQmfReal, pQmfImag, stride, pTimeOut__FDK + (offset * outCh)); } if (err != MPS_OK) goto bail; } } /* ch loop */ bail: C_ALLOC_SCRATCH_END(pQmfImag, FIXP_DBL, QMF_MAX_SYNTHESIS_BANDS); C_ALLOC_SCRATCH_END(pQmfReal, FIXP_DBL, QMF_MAX_SYNTHESIS_BANDS); return err; } void SpatialDecBufferMatrices(spatialDec *self) { int row, col; int complexParBands; complexParBands = self->numParameterBands; /* buffer matrices M2 */ for (row = 0; row < self->numM2rows; row++) { for (col = 0; col < self->numVChannels; col++) { FDKmemcpy(self->M2RealPrev__FDK[row][col], self->M2Real__FDK[row][col], self->numParameterBands * sizeof(FIXP_DBL)); if (0 || (self->phaseCoding == 3)) { FDKmemcpy(self->M2ImagPrev__FDK[row][col], self->M2Imag__FDK[row][col], complexParBands * sizeof(FIXP_DBL)); } } } /* buffer phase */ FDKmemcpy(self->PhasePrevLeft__FDK, self->PhaseLeft__FDK, self->numParameterBands * sizeof(FIXP_DBL)); FDKmemcpy(self->PhasePrevRight__FDK, self->PhaseRight__FDK, self->numParameterBands * sizeof(FIXP_DBL)); } #define PHASE_SCALE 2 #ifndef P_PI #define P_PI 3.1415926535897932 #endif /* For better precision, PI (pi_x2) is already doubled */ static FIXP_DBL interp_angle__FDK(FIXP_DBL angle1, FIXP_DBL angle2, FIXP_SGL alpha, FIXP_DBL pi_x2) { if (angle2 - angle1 > (pi_x2 >> 1)) angle2 -= pi_x2; if (angle1 - angle2 > (pi_x2 >> 1)) angle1 -= pi_x2; return interpolateParameter(alpha, angle2, angle1); } /* * */ void SpatialDecApplyPhase(spatialDec *self, FIXP_SGL alpha__FDK, int lastSlotOfParamSet) { int pb, qs; FIXP_DBL ppb[MAX_PARAMETER_BANDS * 4]; /* left real, imag - right real, imag interleaved */ const FIXP_DBL pi_x2 = PIx2__IPD; for (pb = 0; pb < self->numParameterBands; pb++) { FIXP_DBL pl, pr; pl = interp_angle__FDK(self->PhasePrevLeft__FDK[pb], self->PhaseLeft__FDK[pb], alpha__FDK, pi_x2); pr = interp_angle__FDK(self->PhasePrevRight__FDK[pb], self->PhaseRight__FDK[pb], alpha__FDK, pi_x2); inline_fixp_cos_sin(pl, pr, IPD_SCALE, &ppb[4 * pb]); } /* sign is -1 for qs = 0,2 and +1 for qs = 1 */ const SCHAR *kernels = &self->kernels[0]; FIXP_DBL *Dry_real0 = &self->hybOutputRealDry__FDK[0][0]; FIXP_DBL *Dry_imag0 = &self->hybOutputImagDry__FDK[0][0]; FIXP_DBL *Dry_real1 = &self->hybOutputRealDry__FDK[1][0]; FIXP_DBL *Dry_imag1 = &self->hybOutputImagDry__FDK[1][0]; for (qs = 2; qs >= 0; qs--) { FIXP_DBL out_re, out_im; pb = *kernels++; if (qs == 1) /* sign[qs] >= 0 */ { cplxMultDiv2(&out_re, &out_im, *Dry_real0, *Dry_imag0, ppb[4 * pb + 0], ppb[4 * pb + 1]); out_re <<= PHASE_SCALE - 1; out_im <<= PHASE_SCALE - 1; *Dry_real0++ = out_re; *Dry_imag0++ = out_im; cplxMultDiv2(&out_re, &out_im, *Dry_real1, *Dry_imag1, ppb[4 * pb + 2], ppb[4 * pb + 3]); out_re <<= PHASE_SCALE - 1; out_im <<= PHASE_SCALE - 1; *Dry_real1++ = out_re; *Dry_imag1++ = out_im; } else { cplxMultDiv2(&out_re, &out_im, *Dry_real0, *Dry_imag0, ppb[4 * pb + 0], -ppb[4 * pb + 1]); out_re <<= PHASE_SCALE - 1; out_im <<= PHASE_SCALE - 1; *Dry_real0++ = out_re; *Dry_imag0++ = out_im; cplxMultDiv2(&out_re, &out_im, *Dry_real1, *Dry_imag1, ppb[4 * pb + 2], -ppb[4 * pb + 3]); out_re <<= PHASE_SCALE - 1; out_im <<= PHASE_SCALE - 1; *Dry_real1++ = out_re; *Dry_imag1++ = out_im; } } /* sign is +1 for qs >=3 */ for (qs = self->hybridBands - 3; qs--;) { FIXP_DBL out_re, out_im; pb = *kernels++; cplxMultDiv2(&out_re, &out_im, *Dry_real0, *Dry_imag0, ppb[4 * pb + 0], ppb[4 * pb + 1]); out_re <<= PHASE_SCALE - 1; out_im <<= PHASE_SCALE - 1; *Dry_real0++ = out_re; *Dry_imag0++ = out_im; cplxMultDiv2(&out_re, &out_im, *Dry_real1, *Dry_imag1, ppb[4 * pb + 2], ppb[4 * pb + 3]); out_re <<= PHASE_SCALE - 1; out_im <<= PHASE_SCALE - 1; *Dry_real1++ = out_re; *Dry_imag1++ = out_im; } } fdk-aac-2.0.2/libSACdec/src/sac_process.h000066400000000000000000000307641404225310700200260ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround decoder library ************************* Author(s): Description: SAC Processing *******************************************************************************/ /*! \file \brief Polyphase Filterbank */ #ifndef SAC_PROCESS_H #define SAC_PROCESS_H #include "sac_dec.h" void SpatialDecApplyPhase(spatialDec *self, FIXP_SGL alpha, int lastSlotOfParamSet); /** * \brief Apply QMF Analysis Filterbank. * * Calculates qmf data on downmix input time data. * Delaylines will be applied if necessaray. * * \param self A spatial decoder handle. * \param inData Downmix channel time data as input. * \param ts Signals time slot offset for input buffer. * \param qmfReal Downmix channel qmf output data. * \param qmfImag Downmix channel qmf output data. * * \return Error status. */ SACDEC_ERROR SpatialDecQMFAnalysis(spatialDec *self, const PCM_MPS *inData, const INT ts, const INT bypassMode, FIXP_DBL **qmfReal, FIXP_DBL **qmfImag, const int numInputChannels); /** * \brief Feed spatial decoder with external qmf data. * * \param self A spatial decoder handle. * \param qmfInDataReal External qmf downmix data as input. * \param qmfInDataImag External qmf downmix data as input. * \param ts Signals time slot in input buffer to process. * \param qmfReal Downmix channel qmf output data. * \param qmfImag Downmix channel qmf output data. * \param numInputChannels Number of input channels. Might differ from * self->numInputChannels. * * \return Error status. */ SACDEC_ERROR SpatialDecFeedQMF(spatialDec *self, FIXP_DBL **qmfInDataReal, FIXP_DBL **qmfInDataImag, const INT ts, const INT bypassMode, FIXP_DBL **qmfReal, FIXP_DBL **qmfImag, const INT numInputChannels); /** * \brief Apply Hybrdid Analysis Filterbank. * * Calculates hybrid data on downmix input data. * Residual hybrid signals will also be calculated on current slot if available. * * \param self A spatial decoder handle. * \param qmfInputReal Downmix channel qmf data as input. * \param qmfInputImag Downmix channel qmf data as input. * \param hybOutputReal Downmix channel hybrid output data. * \param hybOutputImag Downmix channel hybrid output data. * \param ts Signals time slot in spatial frame to process. * \param numInputChannels Number of input channels. Might differ from * self->numInputChannels. * * \return Error status. */ SACDEC_ERROR SpatialDecHybridAnalysis(spatialDec *self, FIXP_DBL **qmfInputReal, FIXP_DBL **qmfInputImag, FIXP_DBL **hybOutputReal, FIXP_DBL **hybOutputImag, const INT ts, const INT numInputChannels); /** * \brief Create X data. * * Returns a pointer list over Xchannels pointing to downmix input channels * and to residual channels when provided. * * \param self A spatial decoder handle. * \param hybInputReal Downmix channel hybrid data as input. * \param hybInputImag Downmix channel hybrid data as input. * \param pxReal Pointer to hybrid and residual data as output. * \param pxImag Pointer to hybrid and residual data as output. * * \return Error status. */ SACDEC_ERROR SpatialDecCreateX(spatialDec *self, FIXP_DBL **hybInputReal, FIXP_DBL **hybInputImag, FIXP_DBL **pxReal, FIXP_DBL **pxImag); /** * \brief MPS212 combined version of apply M1 parameters and create wet signal * * \param self A spatial decoder handle. * \param xReal Downmix and residual X data as input. * \param xImag Downmix and residual X data as input. * \param vReal output data: [0] direct signal (V); [1] wet signal * (W). * \param vImag output data: [0] direct signal (V); [1] wet signal * (W). * * \return Error status. */ SACDEC_ERROR SpatialDecApplyM1_CreateW_Mode212( spatialDec *self, const SPATIAL_BS_FRAME *frame, FIXP_DBL **xReal, FIXP_DBL **xImag, FIXP_DBL **vReal, FIXP_DBL **vImag); /** * \brief Apply M2 parameters. * * \param self A spatial decoder handle. * \param ps Signals parameter band from where M2 parameter to * use. * \param alpha Smoothing factor between current and previous * parameter band. Rangeability between 0.f and 1.f. * \param wReal Wet input data. * \param wImag Wet input data. * \param hybOutputRealDry Dry output data. * \param hybOutputImagDry Dry output data. * \param hybOutputRealWet Wet output data. * \param hybOutputImagWet Wet output data. * * \return Error status. */ SACDEC_ERROR SpatialDecApplyM2(spatialDec *self, INT ps, const FIXP_SGL alpha, FIXP_DBL **wReal, FIXP_DBL **wImag, FIXP_DBL **hybOutputRealDry, FIXP_DBL **hybOutputImagDry, FIXP_DBL **hybOutputRealWet, FIXP_DBL **hybOutputImagWet); /** * \brief Apply M2 parameter for 212 mode with residualCoding and phaseCoding. * * \param self [i] A spatial decoder handle. * \param ps [i] Signals parameter band from where M2 parameter * to use. * \param alpha [i] Smoothing factor between current and previous * parameter band. Rangeability between 0.f and 1.f. * \param wReal [i] Wet input data. * \param wImag [i] Wet input data. * \param hybOutputRealDry [o] Dry output data. * \param hybOutputImagDry [o] Dry output data. * * \return error */ SACDEC_ERROR SpatialDecApplyM2_Mode212_ResidualsPlusPhaseCoding( spatialDec *self, INT ps, const FIXP_SGL alpha, FIXP_DBL **wReal, FIXP_DBL **wImag, FIXP_DBL **hybOutputRealDry, FIXP_DBL **hybOutputImagDry); /** * \brief Apply M2 parameter for 212 mode, upmix from mono to stereo. * * \param self [i] A spatial decoder handle. * \param ps [i] Signals parameter band from where M2 parameter * to use. * \param alpha [i] Smoothing factor between current and previous * parameter band. Rangeability between 0.f and 1.f. * \param wReal [i] Wet input data. * \param wImag [i] Wet input data. * \param hybOutputRealDry [o] Dry output data. * \param hybOutputImagDry [o] Dry output data. * * \return error */ SACDEC_ERROR SpatialDecApplyM2_Mode212(spatialDec *self, INT ps, const FIXP_SGL alpha, FIXP_DBL **wReal, FIXP_DBL **wImag, FIXP_DBL **hybOutputRealDry, FIXP_DBL **hybOutputImagDry); /** * \brief Convert Hybrid input to output audio data. * * \param hSpaceSynthesisQmf A spatial decoder handle. * \param ts Signals time slot in spatial frame to process. * \param hybOutputReal Hybrid data as input. * \param hybOutputImag Hybrid data as input. * \param timeOut audio output data. * * \return Error status. */ SACDEC_ERROR SpatialDecSynthesis(spatialDec *self, const INT ts, FIXP_DBL **hybOutputReal, FIXP_DBL **hybOutputImag, PCM_MPS *timeOut, const INT numInputChannels, const FDK_channelMapDescr *const mapDescr); void SpatialDecBufferMatrices(spatialDec *self); FIXP_DBL getChGain(spatialDec *self, UINT ch, INT *scale); #endif fdk-aac-2.0.2/libSACdec/src/sac_qmf.cpp000066400000000000000000000144131404225310700174570ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround decoder library ************************* Author(s): Description: SAC Dec QMF processing *******************************************************************************/ #include "sac_qmf.h" #include "FDK_matrixCalloc.h" #include "sac_dec_interface.h" #include "sac_rom.h" #include "qmf.h" SACDEC_ERROR CalculateSpaceSynthesisQmf( const HANDLE_FDK_QMF_DOMAIN_OUT hQmfDomainOutCh, const FIXP_DBL *Sr, const FIXP_DBL *Si, const INT stride, PCM_MPS *timeSig) { SACDEC_ERROR err = MPS_OK; if (hQmfDomainOutCh == NULL) { err = MPS_INVALID_HANDLE; } else { HANDLE_SPACE_SYNTHESIS_QMF hSpaceSynthesisQmf = &hQmfDomainOutCh->fb; #if (QMF_MAX_SYNTHESIS_BANDS <= 64) C_AALLOC_SCRATCH_START(pWorkBuffer, FIXP_DBL, (QMF_MAX_SYNTHESIS_BANDS << 1)); #else C_AALLOC_STACK_START(pWorkBuffer, FIXP_DBL, (QMF_MAX_SYNTHESIS_BANDS << 1)); #endif qmfSynthesisFilteringSlot(hSpaceSynthesisQmf, Sr, Si, 0, 0, timeSig, stride, pWorkBuffer); #if (QMF_MAX_SYNTHESIS_BANDS <= 64) C_AALLOC_SCRATCH_END(pWorkBuffer, FIXP_DBL, (QMF_MAX_SYNTHESIS_BANDS << 1)); #else C_AALLOC_STACK_END(pWorkBuffer, FIXP_DBL, (QMF_MAX_SYNTHESIS_BANDS << 1)); #endif } return err; } SACDEC_ERROR CalculateSpaceAnalysisQmf( HANDLE_SPACE_ANALYSIS_QMF hSpaceAnalysisQmf, const PCM_MPS *timeSig, FIXP_DBL *Sr, FIXP_DBL *Si) { SACDEC_ERROR err = MPS_OK; if (hSpaceAnalysisQmf == NULL) { err = MPS_INVALID_HANDLE; } else { C_AALLOC_SCRATCH_START(pWorkBuffer, FIXP_DBL, (64 << 1)); qmfAnalysisFilteringSlot(hSpaceAnalysisQmf, Sr, Si, timeSig, 1, pWorkBuffer); C_AALLOC_SCRATCH_END(pWorkBuffer, FIXP_DBL, (64 << 1)); } return err; } fdk-aac-2.0.2/libSACdec/src/sac_qmf.h000066400000000000000000000140371404225310700171260ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround decoder library ************************* Author(s): Description: SAC Dec QMF processing *******************************************************************************/ #ifndef SAC_QMF_H #define SAC_QMF_H #include "common_fix.h" #include "sac_dec_interface.h" #include "FDK_qmf_domain.h" #define HANDLE_SPACE_ANALYSIS_QMF HANDLE_QMF_FILTER_BANK #define HANDLE_SPACE_SYNTHESIS_QMF HANDLE_QMF_FILTER_BANK /** * \brief Convert Qmf input to output audio data. * * \param hSpaceSynthesisQmf A Qmf Synthesis Filterbank handle. * \param Sr Pointer to Qmf input buffer. * \param Si Pointer to Qmf input buffer. * \param stride Stride factor for output data, 1 if none. * \param timeSig (None-)Interleaved audio output data. * * \return Error status. */ SACDEC_ERROR CalculateSpaceSynthesisQmf( const HANDLE_FDK_QMF_DOMAIN_OUT hQmfDomainOutCh, const FIXP_DBL *Sr, const FIXP_DBL *Si, const INT stride, PCM_MPS *timeSig); /** * \brief Convert audio input data to qmf representation. * * \param hSpaceAnalysisQmf A Qmf Analysis Filterbank handle. * \param timeSig (None-)Interleavd audio input data. * \param Sr Pointer to Qmf output buffer. * \param Si Pointer to Qmf output buffer. * * \return Error status. */ SACDEC_ERROR CalculateSpaceAnalysisQmf( HANDLE_SPACE_ANALYSIS_QMF hSpaceAnalysisQmf, const PCM_MPS *timeSig, FIXP_DBL *Sr, FIXP_DBL *Si); #endif /* SAC_QMF_H */ fdk-aac-2.0.2/libSACdec/src/sac_reshapeBBEnv.cpp000066400000000000000000000624201404225310700212010ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround decoder library ************************* Author(s): Description: SAC Dec guided envelope shaping *******************************************************************************/ #include "sac_reshapeBBEnv.h" #include "sac_dec.h" #include "sac_bitdec.h" #include "sac_calcM1andM2.h" #include "sac_reshapeBBEnv.h" #include "sac_rom.h" #define INP_DRY_WET 0 #define INP_DMX 1 #define SF_SHAPE 1 #define SF_DIV32 6 #define SF_FACTOR_SLOT 5 #define START_BB_ENV 0 /* 10 */ #define END_BB_ENV 9 /* 18 */ #define SF_ALPHA1 8 #define SF_BETA1 4 void initBBEnv(spatialDec *self, int initStatesFlag) { INT ch, k; for (ch = 0; ch < self->numOutputChannels; ch++) { k = row2channelGES[self->treeConfig][ch]; self->row2channelDmxGES[ch] = k; if (k == -1) continue; switch (self->treeConfig) { case TREE_212: self->row2channelDmxGES[ch] = 0; break; default:; } } if (initStatesFlag) { for (k = 0; k < 2 * MAX_OUTPUT_CHANNELS + MAX_INPUT_CHANNELS; k++) { self->reshapeBBEnvState->normNrgPrev__FDK[k] = FL2FXCONST_DBL(0.5f); /* 32768.f*32768.f */ self->reshapeBBEnvState->normNrgPrevSF[k] = DFRACT_BITS - 1; self->reshapeBBEnvState->partNrgPrevSF[k] = 0; self->reshapeBBEnvState->partNrgPrev2SF[k] = 0; self->reshapeBBEnvState->frameNrgPrevSF[k] = 0; } } self->reshapeBBEnvState->alpha__FDK = FL2FXCONST_DBL(0.99637845575f); /* FDKexp(-64 / (0.4f * 44100)) */ self->reshapeBBEnvState->beta__FDK = FL2FXCONST_DBL(0.96436909488f); /* FDKexp(-64 / (0.04f * 44100)) */ } static inline void getSlotNrgHQ(FIXP_DBL *RESTRICT pReal, FIXP_DBL *RESTRICT pImag, FIXP_DBL *RESTRICT slotNrg, INT maxValSF, INT hybBands) { INT qs; FIXP_DBL nrg; /* qs = 12, 13, 14 */ slotNrg[0] = ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) + (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1))); slotNrg[1] = ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) + (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1))); slotNrg[2] = ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) + (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1))); /* qs = 15 */ slotNrg[3] = ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) + (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1))); /* qs = 16, 17 */ nrg = ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) + (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1))); slotNrg[4] = nrg + ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) + (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1))); /* qs = 18, 19, 20 */ nrg = ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) + (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1))); nrg += ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) + (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1))); slotNrg[5] = nrg + ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) + (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1))); /* qs = 21, 22 */ nrg = ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) + (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1))); slotNrg[6] = nrg + ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) + (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1))); /* qs = 23, 24 */ if (hybBands > 23) { slotNrg[6] += ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) + (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1))); slotNrg[6] += ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) + (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1))); /* qs = 25, 26, 29, 28, 29 */ nrg = ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) + (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1))); nrg += ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) + (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1))); nrg += ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) + (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1))); nrg += ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) + (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1))); slotNrg[7] = nrg + ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) + (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1))); /* qs = 30 ... min(41,hybBands-1) */ nrg = ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) + (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1))); for (qs = 31; qs < hybBands; qs++) { nrg += ((fPow2Div2((*pReal++) << maxValSF) >> (SF_FACTOR_SLOT - 1)) + (fPow2Div2((*pImag++) << maxValSF) >> (SF_FACTOR_SLOT - 1))); } slotNrg[8] = nrg; } else { slotNrg[7] = (FIXP_DBL)0; slotNrg[8] = (FIXP_DBL)0; } } static inline void combineDryWet(FIXP_DBL *RESTRICT pReal, FIXP_DBL *RESTRICT pImag, FIXP_DBL *RESTRICT pHybOutputRealDry, FIXP_DBL *RESTRICT pHybOutputImagDry, FIXP_DBL *RESTRICT pHybOutputRealWet, FIXP_DBL *RESTRICT pHybOutputImagWet, INT cplxBands, INT hybBands) { INT qs; for (qs = 12; qs < cplxBands; qs++) { pReal[qs] = (pHybOutputRealDry[qs] >> 1) + (pHybOutputRealWet[qs] >> 1); pImag[qs] = (pHybOutputImagDry[qs] >> 1) + (pHybOutputImagWet[qs] >> 1); } for (; qs < hybBands; qs++) { pReal[qs] = (pHybOutputRealDry[qs] >> 1) + (pHybOutputRealWet[qs] >> 1); } } static inline void slotAmp(FIXP_DBL *RESTRICT slotAmp_dry, FIXP_DBL *RESTRICT slotAmp_wet, FIXP_DBL *RESTRICT pHybOutputRealDry, FIXP_DBL *RESTRICT pHybOutputImagDry, FIXP_DBL *RESTRICT pHybOutputRealWet, FIXP_DBL *RESTRICT pHybOutputImagWet, INT cplxBands, INT hybBands) { INT qs; FIXP_DBL dry, wet; dry = wet = FL2FXCONST_DBL(0.0f); for (qs = 0; qs < cplxBands; qs++) { dry = fAddSaturate(dry, fPow2Div2(pHybOutputRealDry[qs] << (1)) + fPow2Div2(pHybOutputImagDry[qs] << (1))); wet = fAddSaturate(wet, fPow2Div2(pHybOutputRealWet[qs] << (1)) + fPow2Div2(pHybOutputImagWet[qs] << (1))); } for (; qs < hybBands; qs++) { dry = fAddSaturate(dry, fPow2Div2(pHybOutputRealDry[qs] << (1))); wet = fAddSaturate(wet, fPow2Div2(pHybOutputRealWet[qs] << (1))); } *slotAmp_dry = dry >> (2 * (1)); *slotAmp_wet = wet >> (2 * (1)); } #if defined(__aarch64__) __attribute__((noinline)) #endif static void shapeBBEnv(FIXP_DBL *pHybOutputRealDry, FIXP_DBL *pHybOutputImagDry, FIXP_DBL dryFac, INT scale, INT cplxBands, INT hybBands) { INT qs; if (scale == 0) { for (qs = 0; qs < cplxBands; qs++) { pHybOutputRealDry[qs] = fMultDiv2(pHybOutputRealDry[qs], dryFac); pHybOutputImagDry[qs] = fMultDiv2(pHybOutputImagDry[qs], dryFac); } for (; qs < hybBands; qs++) { pHybOutputRealDry[qs] = fMultDiv2(pHybOutputRealDry[qs], dryFac); } } else { for (qs = 0; qs < cplxBands; qs++) { pHybOutputRealDry[qs] = SATURATE_LEFT_SHIFT( fMultDiv2(pHybOutputRealDry[qs], dryFac), scale, DFRACT_BITS); pHybOutputImagDry[qs] = SATURATE_LEFT_SHIFT( fMultDiv2(pHybOutputImagDry[qs], dryFac), scale, DFRACT_BITS); } for (; qs < hybBands; qs++) { pHybOutputRealDry[qs] = SATURATE_LEFT_SHIFT( fMultDiv2(pHybOutputRealDry[qs], dryFac), scale, DFRACT_BITS); } } } static void extractBBEnv(spatialDec *self, INT inp, INT start, INT channels, FIXP_DBL *pEnv, const SPATIAL_BS_FRAME *frame) { INT ch, pb, prevChOffs; INT clz, scale, scale_min, envSF; INT scaleCur, scalePrev, commonScale; INT slotNrgSF, partNrgSF, frameNrgSF; INT *pPartNrgPrevSF, *pFrameNrgPrevSF; INT *pNormNrgPrevSF, *pPartNrgPrev2SF; FIXP_DBL maxVal, env, frameNrg, normNrg; FIXP_DBL *pReal, *pImag; FIXP_DBL *partNrg, *partNrgPrev; C_ALLOC_SCRATCH_START(pScratchBuffer, FIXP_DBL, (2 * 42 + MAX_PARAMETER_BANDS)); C_ALLOC_SCRATCH_START(resPb, FIXP_DBL, (END_BB_ENV - START_BB_ENV)); C_ALLOC_SCRATCH_START(resPbSF, INT, (END_BB_ENV - START_BB_ENV)); FIXP_DBL *slotNrg = pScratchBuffer + (2 * 42); RESHAPE_BBENV_STATE *pBBEnvState = self->reshapeBBEnvState; FIXP_DBL alpha = pBBEnvState->alpha__FDK; /*FIXP_DBL alpha1 = (FL2FXCONST_DBL(1.0f) - alpha) << SF_ALPHA1;*/ FIXP_DBL alpha1 = ((FIXP_DBL)MAXVAL_DBL - alpha) << SF_ALPHA1; FIXP_DBL beta = pBBEnvState->beta__FDK; /*FIXP_DBL beta1 = (FL2FXCONST_DBL(1.0f) - beta) << SF_BETA1;*/ FIXP_DBL beta1 = ((FIXP_DBL)MAXVAL_DBL - beta) << SF_BETA1; INT shapeActiv = 1; INT hybBands = fixMin(42, self->hybridBands); INT staticScale = self->staticDecScale + (1); INT cplxBands; cplxBands = fixMin(42, self->hybridBands); for (ch = start; ch < channels; ch++) { if (inp == INP_DRY_WET) { INT ch2 = row2channelGES[self->treeConfig][ch]; if (ch2 == -1) { continue; } else { if (frame->tempShapeEnableChannelGES[ch2]) { shapeActiv = 1; } else { shapeActiv = 0; } } prevChOffs = ch; pReal = pScratchBuffer; pImag = pScratchBuffer + 42; combineDryWet(pReal, pImag, self->hybOutputRealDry__FDK[ch], self->hybOutputImagDry__FDK[ch], self->hybOutputRealWet__FDK[ch], self->hybOutputImagWet__FDK[ch], cplxBands, hybBands); clz = fMin(getScalefactor(&pReal[12], fMax(0, hybBands - 12)), getScalefactor(&pImag[12], fMax(0, cplxBands - 12))); } else { prevChOffs = ch + self->numOutputChannels; pReal = self->hybInputReal__FDK[ch]; pImag = self->hybInputImag__FDK[ch]; clz = fMin(getScalefactor(&pReal[12], fMax(0, hybBands - 12)), getScalefactor(&pImag[12], fMax(0, cplxBands - 12))); } partNrg = partNrgPrev = pBBEnvState->partNrgPrev__FDK[prevChOffs]; pPartNrgPrevSF = &pBBEnvState->partNrgPrevSF[prevChOffs]; pFrameNrgPrevSF = &pBBEnvState->frameNrgPrevSF[prevChOffs]; pNormNrgPrevSF = &pBBEnvState->normNrgPrevSF[prevChOffs]; pPartNrgPrev2SF = &pBBEnvState->partNrgPrev2SF[prevChOffs]; /* calculate slot energy */ { getSlotNrgHQ(&pReal[12], &pImag[12], slotNrg, clz, fixMin(42, self->hybridBands)); /* scale slotNrg: 2*(staticScale-clz) + SF_FACTOR_SLOT */ } slotNrgSF = 2 * (staticScale - clz + ((inp == INP_DRY_WET) ? 1 : 0)) + SF_FACTOR_SLOT; frameNrgSF = 2 * (staticScale - clz + ((inp == INP_DRY_WET) ? 1 : 0)) + SF_FACTOR_SLOT; partNrgSF = fixMax(slotNrgSF - SF_ALPHA1 + 1, pPartNrgPrevSF[0] - pPartNrgPrev2SF[0] + 1); scalePrev = fixMax(fixMin(partNrgSF - pPartNrgPrevSF[0], DFRACT_BITS - 1), -(DFRACT_BITS - 1)); scaleCur = fixMax(fixMin(partNrgSF - slotNrgSF + SF_ALPHA1, DFRACT_BITS - 1), -(DFRACT_BITS - 1)); maxVal = FL2FXCONST_DBL(0.0f); frameNrg = FL2FXCONST_DBL(0.0f); if ((scaleCur < 0) && (scalePrev < 0)) { scaleCur = -scaleCur; scalePrev = -scalePrev; for (pb = START_BB_ENV; pb < END_BB_ENV; pb++) { partNrg[pb] = ((fMultDiv2(alpha1, slotNrg[pb]) << scaleCur) + (fMultDiv2(alpha, partNrgPrev[pb]) << scalePrev)) << 1; maxVal |= partNrg[pb]; frameNrg += slotNrg[pb] >> 3; } } else if ((scaleCur >= 0) && (scalePrev >= 0)) { for (pb = START_BB_ENV; pb < END_BB_ENV; pb++) { partNrg[pb] = ((fMultDiv2(alpha1, slotNrg[pb]) >> scaleCur) + (fMultDiv2(alpha, partNrgPrev[pb]) >> scalePrev)) << 1; maxVal |= partNrg[pb]; frameNrg += slotNrg[pb] >> 3; } } else if ((scaleCur < 0) && (scalePrev >= 0)) { scaleCur = -scaleCur; for (pb = START_BB_ENV; pb < END_BB_ENV; pb++) { partNrg[pb] = ((fMultDiv2(alpha1, slotNrg[pb]) << scaleCur) + (fMultDiv2(alpha, partNrgPrev[pb]) >> scalePrev)) << 1; maxVal |= partNrg[pb]; frameNrg += slotNrg[pb] >> 3; } } else { /* if ( (scaleCur >= 0) && (scalePrev < 0) ) */ scalePrev = -scalePrev; for (pb = START_BB_ENV; pb < END_BB_ENV; pb++) { partNrg[pb] = ((fMultDiv2(alpha1, slotNrg[pb]) >> scaleCur) + (fMultDiv2(alpha, partNrgPrev[pb]) << scalePrev)) << 1; maxVal |= partNrg[pb]; frameNrg += slotNrg[pb] >> 3; } } /* frameNrg /= (END_BB_ENV - START_BB_ENV); 0.88888888888f = * (1/(END_BB_ENV-START_BB_ENV)<<3; shift with 3 is compensated in loop * above */ frameNrg = fMult(frameNrg, FL2FXCONST_DBL(0.88888888888f)); /* store scalefactor and headroom for part nrg prev */ pPartNrgPrevSF[0] = partNrgSF; pPartNrgPrev2SF[0] = fixMax(0, CntLeadingZeros(maxVal) - 1); commonScale = fixMax(frameNrgSF - SF_ALPHA1 + 1, pFrameNrgPrevSF[0] + 1); scalePrev = fixMin(commonScale - pFrameNrgPrevSF[0], DFRACT_BITS - 1); scaleCur = fixMin(commonScale - frameNrgSF + SF_ALPHA1, DFRACT_BITS - 1); frameNrgSF = commonScale; frameNrg = ((fMultDiv2(alpha1, frameNrg) >> scaleCur) + (fMultDiv2(alpha, pBBEnvState->frameNrgPrev__FDK[prevChOffs]) >> scalePrev)) << 1; clz = fixMax(0, CntLeadingZeros(frameNrg) - 1); pBBEnvState->frameNrgPrev__FDK[prevChOffs] = frameNrg << clz; pFrameNrgPrevSF[0] = frameNrgSF - clz; env = FL2FXCONST_DBL(0.0f); scale = clz + partNrgSF - frameNrgSF; scale_min = DFRACT_BITS - 1; for (pb = START_BB_ENV; pb < END_BB_ENV; pb++) { if ((partNrg[pb] | slotNrg[pb]) != FL2FXCONST_DBL(0.0f)) { INT s; INT sc = 0; INT sn = fixMax(0, CntLeadingZeros(slotNrg[pb]) - 1); FIXP_DBL inv_sqrt = invSqrtNorm2(partNrg[pb], &sc); FIXP_DBL res = fMult(slotNrg[pb] << sn, fPow2(inv_sqrt)); s = fixMax(0, CntLeadingZeros(res) - 1); res = res << s; sc = scale - (2 * sc - sn - s); scale_min = fixMin(scale_min, sc); resPb[pb] = res; resPbSF[pb] = sc; } else { resPb[pb] = (FIXP_DBL)0; resPbSF[pb] = 0; } } scale_min = 4 - scale_min; for (pb = START_BB_ENV; pb < END_BB_ENV; pb++) { INT sc = fixMax(fixMin(resPbSF[pb] + scale_min, DFRACT_BITS - 1), -(DFRACT_BITS - 1)); if (sc < 0) { env += resPb[pb] << (-sc); } else { env += resPb[pb] >> (sc); } } env = fMultDiv2(env, pBBEnvState->frameNrgPrev__FDK[prevChOffs]); envSF = slotNrgSF + scale_min + 1; commonScale = fixMax(envSF - SF_BETA1 + 1, pNormNrgPrevSF[0] + 1); scalePrev = fixMin(commonScale - pNormNrgPrevSF[0], DFRACT_BITS - 1); scaleCur = fixMin(commonScale - envSF + SF_BETA1, DFRACT_BITS - 1); normNrg = ((fMultDiv2(beta1, env) >> scaleCur) + (fMultDiv2(beta, pBBEnvState->normNrgPrev__FDK[prevChOffs]) >> scalePrev)) << 1; clz = fixMax(0, CntLeadingZeros(normNrg) - 1); pBBEnvState->normNrgPrev__FDK[prevChOffs] = normNrg << clz; pNormNrgPrevSF[0] = commonScale - clz; if (shapeActiv) { if ((env | normNrg) != FL2FXCONST_DBL(0.0f)) { INT sc, se, sn; se = fixMax(0, CntLeadingZeros(env) - 1); sc = commonScale + SF_DIV32 - envSF + se; env = fMult(sqrtFixp((env << se) >> (sc & 0x1)), invSqrtNorm2(normNrg, &sn)); sc = fixMin((sc >> 1) - sn, DFRACT_BITS - 1); if (sc < 0) { env <<= (-sc); } else { env >>= (sc); } } /* env is scaled by SF_DIV32/2 bits */ } pEnv[ch] = env; } C_ALLOC_SCRATCH_END(resPbSF, INT, (END_BB_ENV - START_BB_ENV)); C_ALLOC_SCRATCH_END(resPb, FIXP_DBL, (END_BB_ENV - START_BB_ENV)); C_ALLOC_SCRATCH_END(pScratchBuffer, FIXP_DBL, (2 * 42 + MAX_PARAMETER_BANDS)); } void SpatialDecReshapeBBEnv(spatialDec *self, const SPATIAL_BS_FRAME *frame, INT ts) { INT ch, scale; INT dryFacSF, slotAmpSF; FIXP_DBL tmp, dryFac, envShape; FIXP_DBL slotAmp_dry, slotAmp_wet, slotAmp_ratio; FIXP_DBL envDry[MAX_OUTPUT_CHANNELS], envDmx[2]; INT cplxBands; INT hybBands = self->hybridBands - 6; cplxBands = self->hybridBands - 6; /* extract downmix envelope(s) */ switch (self->treeConfig) { default: extractBBEnv(self, INP_DMX, 0, fMin(self->numInputChannels, 2), envDmx, frame); } /* extract dry and wet envelopes */ extractBBEnv(self, INP_DRY_WET, 0, self->numOutputChannels, envDry, frame); for (ch = 0; ch < self->numOutputChannels; ch++) { INT ch2; ch2 = row2channelGES[self->treeConfig][ch]; if (ch2 == -1) continue; if (frame->tempShapeEnableChannelGES[ch2]) { INT sc; /* reshape dry and wet signals according to transmitted envelope */ /* De-quantize GES data */ FDK_ASSERT((frame->bsEnvShapeData[ch2][ts] >= 0) && (frame->bsEnvShapeData[ch2][ts] <= 4)); FDK_ASSERT((self->envQuantMode == 0) || (self->envQuantMode == 1)); envShape = FX_CFG2FX_DBL(envShapeDataTable__FDK[frame->bsEnvShapeData[ch2][ts]] [self->envQuantMode]); /* get downmix channel */ ch2 = self->row2channelDmxGES[ch]; /* multiply ratio with dmx envelope; tmp is scaled by SF_DIV32/2+SF_SHAPE * bits */ if (ch2 == 2) { tmp = fMultDiv2(envShape, envDmx[0]) + fMultDiv2(envShape, envDmx[1]); } else { tmp = fMult(envShape, envDmx[ch2]); } /* weighting factors */ dryFacSF = slotAmpSF = 0; dryFac = slotAmp_ratio = FL2FXCONST_DBL(0.0f); /* dryFac will be scaled by dryFacSF bits */ if (envDry[ch] != FL2FXCONST_DBL(0.0f)) { envDry[ch] = invSqrtNorm2(envDry[ch], &dryFacSF); dryFac = fMultDiv2(tmp, fPow2Div2(envDry[ch])) << 2; dryFacSF = SF_SHAPE + 2 * dryFacSF; } /* calculate slotAmp_dry and slotAmp_wet */ slotAmp(&slotAmp_dry, &slotAmp_wet, &self->hybOutputRealDry__FDK[ch][6], &self->hybOutputImagDry__FDK[ch][6], &self->hybOutputRealWet__FDK[ch][6], &self->hybOutputImagWet__FDK[ch][6], cplxBands, hybBands); /* slotAmp_ratio will be scaled by slotAmpSF bits */ if (slotAmp_dry != FL2FXCONST_DBL(0.0f)) { sc = fixMax(0, CntLeadingZeros(slotAmp_wet) - 1); sc = sc - (sc & 1); slotAmp_wet = sqrtFixp(slotAmp_wet << sc); slotAmp_dry = invSqrtNorm2(slotAmp_dry, &slotAmpSF); slotAmp_ratio = fMult(slotAmp_wet, slotAmp_dry); slotAmpSF = slotAmpSF - (sc >> 1); } /* calculate common scale factor */ scale = fixMax(3, fixMax(dryFacSF, slotAmpSF)); /* scale is at least with 3 bits to avoid overflows when calculating dryFac */ dryFac = dryFac >> fixMin(scale - dryFacSF, DFRACT_BITS - 1); slotAmp_ratio = slotAmp_ratio >> fixMin(scale - slotAmpSF, DFRACT_BITS - 1); /* limit dryFac */ dryFac = fixMax( FL2FXCONST_DBL(0.25f) >> (INT)fixMin(2 * scale, DFRACT_BITS - 1), fMult(dryFac, slotAmp_ratio) - (slotAmp_ratio >> fixMin(scale, DFRACT_BITS - 1)) + (dryFac >> fixMin(scale, DFRACT_BITS - 1))); dryFac = fixMin( FL2FXCONST_DBL(0.50f) >> (INT)fixMin(2 * scale - 3, DFRACT_BITS - 1), dryFac); /* reduce shift bits by 3, because upper limit 4.0 is scaled with 3 bits */ scale = 2 * scale + 1; /* improve precision for dryFac */ sc = fixMax(0, CntLeadingZeros(dryFac) - 1); dryFac = dryFac << (INT)fixMin(scale, sc); scale = scale - fixMin(scale, sc); /* shaping */ shapeBBEnv(&self->hybOutputRealDry__FDK[ch][6], &self->hybOutputImagDry__FDK[ch][6], dryFac, fixMin(scale, DFRACT_BITS - 1), cplxBands, hybBands); } } } fdk-aac-2.0.2/libSACdec/src/sac_reshapeBBEnv.h000066400000000000000000000120431404225310700206420ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround decoder library ************************* Author(s): Description: SAC Dec guided envelope shaping *******************************************************************************/ #ifndef SAC_RESHAPEBBENV_H #define SAC_RESHAPEBBENV_H #include "sac_dec_interface.h" #define BB_ENV_SIZE 9 /* END_BB_ENV - START_BB_ENV */ void initBBEnv(spatialDec *self, int initStatesFlag); void SpatialDecReshapeBBEnv(spatialDec *self, const SPATIAL_BS_FRAME *frame, int ts); #endif fdk-aac-2.0.2/libSACdec/src/sac_rom.cpp000066400000000000000000006347341404225310700175070ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround decoder library ************************* Author(s): Description: SAC Dec tables *******************************************************************************/ #include "sac_rom.h" #include "sac_calcM1andM2.h" #define SCALE_CPC(a) (FL2FXCONST_CFG(a / (float)(1 << SCALE_PARAM_M1))) const FIXP_CFG dequantCPC__FDK[] = { SCALE_CPC(-2.0f), SCALE_CPC(-1.9f), SCALE_CPC(-1.8f), SCALE_CPC(-1.7f), SCALE_CPC(-1.6f), SCALE_CPC(-1.5f), SCALE_CPC(-1.4f), SCALE_CPC(-1.3f), SCALE_CPC(-1.2f), SCALE_CPC(-1.1f), SCALE_CPC(-1.0f), SCALE_CPC(-0.9f), SCALE_CPC(-0.8f), SCALE_CPC(-0.7f), SCALE_CPC(-0.6f), SCALE_CPC(-0.5f), SCALE_CPC(-0.4f), SCALE_CPC(-0.3f), SCALE_CPC(-0.2f), SCALE_CPC(-0.1f), SCALE_CPC(0.0f), SCALE_CPC(0.1f), SCALE_CPC(0.2f), SCALE_CPC(0.3f), SCALE_CPC(0.4f), SCALE_CPC(0.5f), SCALE_CPC(0.6f), SCALE_CPC(0.7f), SCALE_CPC(0.8f), SCALE_CPC(0.9f), SCALE_CPC(1.0f), SCALE_CPC(1.1f), SCALE_CPC(1.2f), SCALE_CPC(1.3f), SCALE_CPC(1.4f), SCALE_CPC(1.5f), SCALE_CPC(1.6f), SCALE_CPC(1.7f), SCALE_CPC(1.8f), SCALE_CPC(1.9f), SCALE_CPC(2.0f), SCALE_CPC(2.1f), SCALE_CPC(2.2f), SCALE_CPC(2.3f), SCALE_CPC(2.4f), SCALE_CPC(2.5f), SCALE_CPC(2.6f), SCALE_CPC(2.7f), SCALE_CPC(2.8f), SCALE_CPC(2.9f), SCALE_CPC(3.0f)}; #define SCALE_ICC(a) (FL2FXCONST_CFG(a)) const FIXP_CFG dequantICC__FDK[8] = { /*SCALE_ICC(1.00000f)*/ FX_DBL2FX_CFG(MAXVAL_DBL), SCALE_ICC(0.9370f), SCALE_ICC(0.84118f), SCALE_ICC(0.60092f), SCALE_ICC(0.36764f), SCALE_ICC(0.0000f), SCALE_ICC(-0.58900f), SCALE_ICC(-0.9900f)}; #define SCALE_CLD2(a) (FL2FXCONST_CFG(a / (float)(1 << 8))) const FIXP_CFG dequantCLD__FDK[31] = { SCALE_CLD2(-150.0f), SCALE_CLD2(-45.0f), SCALE_CLD2(-40.0f), SCALE_CLD2(-35.0f), SCALE_CLD2(-30.0f), SCALE_CLD2(-25.0f), SCALE_CLD2(-22.0f), SCALE_CLD2(-19.0f), SCALE_CLD2(-16.0f), SCALE_CLD2(-13.0f), SCALE_CLD2(-10.0f), SCALE_CLD2(-8.0f), SCALE_CLD2(-6.0f), SCALE_CLD2(-4.0f), SCALE_CLD2(-2.0f), SCALE_CLD2(0.0f), SCALE_CLD2(2.0f), SCALE_CLD2(4.0f), SCALE_CLD2(6.0f), SCALE_CLD2(8.0f), SCALE_CLD2(10.0f), SCALE_CLD2(13.0f), SCALE_CLD2(16.0f), SCALE_CLD2(19.0f), SCALE_CLD2(22.0f), SCALE_CLD2(25.0f), SCALE_CLD2(30.0f), SCALE_CLD2(35.0f), SCALE_CLD2(40.0f), SCALE_CLD2(45.0f), SCALE_CLD2(150.0f)}; #define SCALE_IPD(a) (FL2FXCONST_CFG(a / (float)(1 << IPD_SCALE))) const FIXP_CFG dequantIPD__FDK[16] = { /* SCALE_IPD(0.000000000f), SCALE_IPD(0.392699082f), SCALE_IPD(0.785398163f), SCALE_IPD(1.178097245f), SCALE_IPD(1.570796327f), SCALE_IPD(1.963495408f), SCALE_IPD(2.356194490f), SCALE_IPD(2.748893572f), SCALE_IPD(3.141592654f), SCALE_IPD(3.534291735f), SCALE_IPD(3.926990817f), SCALE_IPD(4.319689899f), SCALE_IPD(4.712388980f), SCALE_IPD(5.105088062f), SCALE_IPD(5.497787144f), SCALE_IPD(5.890486225f) */ SCALE_IPD(0.00000000000000f), SCALE_IPD(0.392699082f), SCALE_IPD(0.78539816339745f), SCALE_IPD(1.178097245f), SCALE_IPD(1.57079632679490f), SCALE_IPD(1.963495408f), SCALE_IPD(2.35619449019234f), SCALE_IPD(2.748893572f), SCALE_IPD(3.14159265358979f), SCALE_IPD(3.534291735f), SCALE_IPD(3.92699081698724f), SCALE_IPD(4.319689899f), SCALE_IPD(4.71238898038469f), SCALE_IPD(5.105088062f), SCALE_IPD(5.49778714378214f), SCALE_IPD(5.890486225f)}; #define SCALE_SPLIT_ANGLE(a) (FL2FXCONST_DBL(a / (float)(1 << IPD_SCALE))) /* Generate table dequantIPD_CLD_ICC_splitAngle__FDK[16][31][8]: #define ABS_THR ( 1e-9f * 32768 * 32768 ) float dequantICC[] = {1.0000f,0.9370f,0.84118f,0.60092f,0.36764f,0.0f,-0.5890f,-0.9900f}; float dequantCLD[] = {-150.0,-45.0,-40.0,-35.0,-30.0,-25.0,-22.0,-19.0,-16.0,-13.0,-10.0, -8.0, -6.0, -4.0, -2.0, 0.0, 2.0, 4.0, 6.0, 8.0, 10.0, 13.0, 16.0, 19.0, 22.0, 25.0, 30.0, 35.0, 40.0, 45.0, 150.0 }; float dequantIPD[] = {0.f,0.392699082f,0.785398163f,1.178097245f,1.570796327f,1.963495408f, 2.35619449f,2.748893572f,3.141592654f,3.534291735f,3.926990817f, 4.319689899f,4.71238898f,5.105088062f,5.497787144f,5.890486225f}; for (ipdIdx=0; ipdIdx<16; ipdIdx++) for (cldIdx=0; cldIdx<31; cldIdx++) for (iccIdx=0; iccIdx<8; iccIdx++) { ipd = dequantIPD[ipdIdx]; cld = dequantCLD[cldIdx]; icc = dequantICC[iccIdx]; iidLin = (float) pow(10.0f, cld / 20.0f); iidLin2 = iidLin * iidLin; iidLin21 = iidLin2 + 1.0f; sinIpd = (float) sin(ipd); cosIpd = (float) cos(ipd); temp1 = 2.0f * icc * iidLin; temp1c = temp1 * cosIpd; ratio = (iidLin21 + temp1c) / (iidLin21 + temp1) + ABS_THR; w2 = (float) pow(ratio, 0.25f); w1 = 2.0f - w2; dequantIPD_CLD_ICC_splitAngle__FDK[ipdIdx][cldIdx][iccIdx] = (float) atan2(w2 * sinIpd, w1 * iidLin + w2 * cosIpd); } */ const FIXP_DBL dequantIPD_CLD_ICC_splitAngle__FDK[15][31][8] = { { {SCALE_SPLIT_ANGLE(0.39269909262657), SCALE_SPLIT_ANGLE(0.39269909262657), SCALE_SPLIT_ANGLE(0.39269909262657), SCALE_SPLIT_ANGLE(0.39269909262657), SCALE_SPLIT_ANGLE(0.39269909262657), SCALE_SPLIT_ANGLE(0.39269909262657), SCALE_SPLIT_ANGLE(0.39269909262657), SCALE_SPLIT_ANGLE(0.39269909262657)}, {SCALE_SPLIT_ANGLE(0.39055734872818), SCALE_SPLIT_ANGLE(0.39055740833282), SCALE_SPLIT_ANGLE(0.39055749773979), SCALE_SPLIT_ANGLE(0.39055770635605), SCALE_SPLIT_ANGLE(0.39055791497231), SCALE_SPLIT_ANGLE(0.39055824279785), SCALE_SPLIT_ANGLE(0.39055877923965), SCALE_SPLIT_ANGLE(0.39055916666985)}, {SCALE_SPLIT_ANGLE(0.38890451192856), SCALE_SPLIT_ANGLE(0.38890469074249), SCALE_SPLIT_ANGLE(0.38890495896339), SCALE_SPLIT_ANGLE(0.38890564441681), SCALE_SPLIT_ANGLE(0.38890630006790), SCALE_SPLIT_ANGLE(0.38890734314919), SCALE_SPLIT_ANGLE(0.38890904188156), SCALE_SPLIT_ANGLE(0.38891020417213)}, {SCALE_SPLIT_ANGLE(0.38599541783333), SCALE_SPLIT_ANGLE(0.38599592447281), SCALE_SPLIT_ANGLE(0.38599672913551), SCALE_SPLIT_ANGLE(0.38599878549576), SCALE_SPLIT_ANGLE(0.38600078225136), SCALE_SPLIT_ANGLE(0.38600403070450), SCALE_SPLIT_ANGLE(0.38600939512253), SCALE_SPLIT_ANGLE(0.38601315021515)}, {SCALE_SPLIT_ANGLE(0.38091576099396), SCALE_SPLIT_ANGLE(0.38091731071472), SCALE_SPLIT_ANGLE(0.38091966509819), SCALE_SPLIT_ANGLE(0.38092571496964), SCALE_SPLIT_ANGLE(0.38093179464340), SCALE_SPLIT_ANGLE(0.38094165921211), SCALE_SPLIT_ANGLE(0.38095843791962), SCALE_SPLIT_ANGLE(0.38097056746483)}, {SCALE_SPLIT_ANGLE(0.37216997146606), SCALE_SPLIT_ANGLE(0.37217426300049), SCALE_SPLIT_ANGLE(0.37218090891838), SCALE_SPLIT_ANGLE(0.37219807505608), SCALE_SPLIT_ANGLE(0.37221556901932), SCALE_SPLIT_ANGLE(0.37224492430687), SCALE_SPLIT_ANGLE(0.37229704856873), SCALE_SPLIT_ANGLE(0.37233674526215)}, {SCALE_SPLIT_ANGLE(0.36424967646599), SCALE_SPLIT_ANGLE(0.36425727605820), SCALE_SPLIT_ANGLE(0.36426907777786), SCALE_SPLIT_ANGLE(0.36430004239082), SCALE_SPLIT_ANGLE(0.36433213949203), SCALE_SPLIT_ANGLE(0.36438730359077), SCALE_SPLIT_ANGLE(0.36448970437050), SCALE_SPLIT_ANGLE(0.36457163095474)}, {SCALE_SPLIT_ANGLE(0.35356888175011), SCALE_SPLIT_ANGLE(0.35358175635338), SCALE_SPLIT_ANGLE(0.35360190272331), SCALE_SPLIT_ANGLE(0.35365560650826), SCALE_SPLIT_ANGLE(0.35371258854866), SCALE_SPLIT_ANGLE(0.35381385684013), SCALE_SPLIT_ANGLE(0.35401365160942), SCALE_SPLIT_ANGLE(0.35418578982353)}, {SCALE_SPLIT_ANGLE(0.33942583203316), SCALE_SPLIT_ANGLE(0.33944645524025), SCALE_SPLIT_ANGLE(0.33947896957397), SCALE_SPLIT_ANGLE(0.33956742286682), SCALE_SPLIT_ANGLE(0.33966416120529), SCALE_SPLIT_ANGLE(0.33984372019768), SCALE_SPLIT_ANGLE(0.34023007750511), SCALE_SPLIT_ANGLE(0.34060221910477)}, {SCALE_SPLIT_ANGLE(0.32115450501442), SCALE_SPLIT_ANGLE(0.32118520140648), SCALE_SPLIT_ANGLE(0.32123416662216), SCALE_SPLIT_ANGLE(0.32137048244476), SCALE_SPLIT_ANGLE(0.32152509689331), SCALE_SPLIT_ANGLE(0.32182863354683), SCALE_SPLIT_ANGLE(0.32256561517715), SCALE_SPLIT_ANGLE(0.32340893149376)}, {SCALE_SPLIT_ANGLE(0.29830521345139), SCALE_SPLIT_ANGLE(0.29834723472595), SCALE_SPLIT_ANGLE(0.29841500520706), SCALE_SPLIT_ANGLE(0.29860860109329), SCALE_SPLIT_ANGLE(0.29883742332458), SCALE_SPLIT_ANGLE(0.29931786656380), SCALE_SPLIT_ANGLE(0.30069106817245), SCALE_SPLIT_ANGLE(0.30275771021843)}, {SCALE_SPLIT_ANGLE(0.28050789237022), SCALE_SPLIT_ANGLE(0.28055712580681), SCALE_SPLIT_ANGLE(0.28063708543777), SCALE_SPLIT_ANGLE(0.28086942434311), SCALE_SPLIT_ANGLE(0.28115189075470), SCALE_SPLIT_ANGLE(0.28177404403687), SCALE_SPLIT_ANGLE(0.28380545973778), SCALE_SPLIT_ANGLE(0.28782638907433)}, {SCALE_SPLIT_ANGLE(0.26083287596703), SCALE_SPLIT_ANGLE(0.26088824868202), SCALE_SPLIT_ANGLE(0.26097872853279), SCALE_SPLIT_ANGLE(0.26124578714371), SCALE_SPLIT_ANGLE(0.26157897710800), SCALE_SPLIT_ANGLE(0.26234793663025), SCALE_SPLIT_ANGLE(0.26525345444679), SCALE_SPLIT_ANGLE(0.27373576164246)}, {SCALE_SPLIT_ANGLE(0.23960022628307), SCALE_SPLIT_ANGLE(0.23966011404991), SCALE_SPLIT_ANGLE(0.23975846171379), SCALE_SPLIT_ANGLE(0.24005253612995), SCALE_SPLIT_ANGLE(0.24042735993862), SCALE_SPLIT_ANGLE(0.24132782220840), SCALE_SPLIT_ANGLE(0.24525125324726), SCALE_SPLIT_ANGLE(0.26534587144852)}, {SCALE_SPLIT_ANGLE(0.21727463603020), SCALE_SPLIT_ANGLE(0.21733720600605), SCALE_SPLIT_ANGLE(0.21744030714035), SCALE_SPLIT_ANGLE(0.21775111556053), SCALE_SPLIT_ANGLE(0.21815299987793), SCALE_SPLIT_ANGLE(0.21914559602737), SCALE_SPLIT_ANGLE(0.22397418320179), SCALE_SPLIT_ANGLE(0.27997341752052)}, {SCALE_SPLIT_ANGLE(0.19442924857140), SCALE_SPLIT_ANGLE(0.19449260830879), SCALE_SPLIT_ANGLE(0.19459712505341), SCALE_SPLIT_ANGLE(0.19491320848465), SCALE_SPLIT_ANGLE(0.19532410800457), SCALE_SPLIT_ANGLE(0.19634956121445), SCALE_SPLIT_ANGLE(0.20156545937061), SCALE_SPLIT_ANGLE(0.33650875091553)}, {SCALE_SPLIT_ANGLE(0.17168679833412), SCALE_SPLIT_ANGLE(0.17174908518791), SCALE_SPLIT_ANGLE(0.17185173928738), SCALE_SPLIT_ANGLE(0.17216140031815), SCALE_SPLIT_ANGLE(0.17256212234497), SCALE_SPLIT_ANGLE(0.17355351150036), SCALE_SPLIT_ANGLE(0.17841057479382), SCALE_SPLIT_ANGLE(0.23922468721867)}, {SCALE_SPLIT_ANGLE(0.14965108036995), SCALE_SPLIT_ANGLE(0.14971046149731), SCALE_SPLIT_ANGLE(0.14980803430080), SCALE_SPLIT_ANGLE(0.15010002255440), SCALE_SPLIT_ANGLE(0.15047283470631), SCALE_SPLIT_ANGLE(0.15137128531933), SCALE_SPLIT_ANGLE(0.15533345937729), SCALE_SPLIT_ANGLE(0.17691856622696)}, {SCALE_SPLIT_ANGLE(0.12884500622749), SCALE_SPLIT_ANGLE(0.12889973819256), SCALE_SPLIT_ANGLE(0.12898921966553), SCALE_SPLIT_ANGLE(0.12925371527672), SCALE_SPLIT_ANGLE(0.12958450615406), SCALE_SPLIT_ANGLE(0.13035117089748), SCALE_SPLIT_ANGLE(0.13329000771046), SCALE_SPLIT_ANGLE(0.14226883649826)}, {SCALE_SPLIT_ANGLE(0.10966771841049), SCALE_SPLIT_ANGLE(0.10971628874540), SCALE_SPLIT_ANGLE(0.10979522019625), SCALE_SPLIT_ANGLE(0.11002497375011), SCALE_SPLIT_ANGLE(0.11030506342649), SCALE_SPLIT_ANGLE(0.11092507839203), SCALE_SPLIT_ANGLE(0.11297956109047), SCALE_SPLIT_ANGLE(0.11718676239252)}, {SCALE_SPLIT_ANGLE(0.09237616509199), SCALE_SPLIT_ANGLE(0.09241759032011), SCALE_SPLIT_ANGLE(0.09248441457748), SCALE_SPLIT_ANGLE(0.09267570823431), SCALE_SPLIT_ANGLE(0.09290253371000), SCALE_SPLIT_ANGLE(0.09338124841452), SCALE_SPLIT_ANGLE(0.09476862102747), SCALE_SPLIT_ANGLE(0.09691140055656)}, {SCALE_SPLIT_ANGLE(0.07020132243633), SCALE_SPLIT_ANGLE(0.07023159414530), SCALE_SPLIT_ANGLE(0.07027988880873), SCALE_SPLIT_ANGLE(0.07041462510824), SCALE_SPLIT_ANGLE(0.07056795060635), SCALE_SPLIT_ANGLE(0.07087047398090), SCALE_SPLIT_ANGLE(0.07161350548267), SCALE_SPLIT_ANGLE(0.07247893512249)}, {SCALE_SPLIT_ANGLE(0.05244029685855), SCALE_SPLIT_ANGLE(0.05246063694358), SCALE_SPLIT_ANGLE(0.05249277502298), SCALE_SPLIT_ANGLE(0.05258031934500), SCALE_SPLIT_ANGLE(0.05267634987831), SCALE_SPLIT_ANGLE(0.05285539478064), SCALE_SPLIT_ANGLE(0.05324416980147), SCALE_SPLIT_ANGLE(0.05362326279283)}, {SCALE_SPLIT_ANGLE(0.03864165768027), SCALE_SPLIT_ANGLE(0.03865439072251), SCALE_SPLIT_ANGLE(0.03867433592677), SCALE_SPLIT_ANGLE(0.03872758522630), SCALE_SPLIT_ANGLE(0.03878424316645), SCALE_SPLIT_ANGLE(0.03888526186347), SCALE_SPLIT_ANGLE(0.03908598423004), SCALE_SPLIT_ANGLE(0.03926040604711)}, {SCALE_SPLIT_ANGLE(0.02817477472126), SCALE_SPLIT_ANGLE(0.02818230912089), SCALE_SPLIT_ANGLE(0.02819401398301), SCALE_SPLIT_ANGLE(0.02822477556765), SCALE_SPLIT_ANGLE(0.02825673110783), SCALE_SPLIT_ANGLE(0.02831180393696), SCALE_SPLIT_ANGLE(0.02841453813016), SCALE_SPLIT_ANGLE(0.02849725075066)}, {SCALE_SPLIT_ANGLE(0.02037953026593), SCALE_SPLIT_ANGLE(0.02038379199803), SCALE_SPLIT_ANGLE(0.02039037831128), SCALE_SPLIT_ANGLE(0.02040746994317), SCALE_SPLIT_ANGLE(0.02042490243912), SCALE_SPLIT_ANGLE(0.02045420184731), SCALE_SPLIT_ANGLE(0.02050646767020), SCALE_SPLIT_ANGLE(0.02054644003510)}, {SCALE_SPLIT_ANGLE(0.01173160225153), SCALE_SPLIT_ANGLE(0.01173313986510), SCALE_SPLIT_ANGLE(0.01173550263047), SCALE_SPLIT_ANGLE(0.01174154505134), SCALE_SPLIT_ANGLE(0.01174758374691), SCALE_SPLIT_ANGLE(0.01175745483488), SCALE_SPLIT_ANGLE(0.01177425310016), SCALE_SPLIT_ANGLE(0.01178644131869)}, {SCALE_SPLIT_ANGLE(0.00668647512794), SCALE_SPLIT_ANGLE(0.00668699946254), SCALE_SPLIT_ANGLE(0.00668780272827), SCALE_SPLIT_ANGLE(0.00668983906507), SCALE_SPLIT_ANGLE(0.00669185025617), SCALE_SPLIT_ANGLE(0.00669508520514), SCALE_SPLIT_ANGLE(0.00670044543222), SCALE_SPLIT_ANGLE(0.00670422753319)}, {SCALE_SPLIT_ANGLE(0.00378898042254), SCALE_SPLIT_ANGLE(0.00378915388137), SCALE_SPLIT_ANGLE(0.00378941884264), SCALE_SPLIT_ANGLE(0.00379008660093), SCALE_SPLIT_ANGLE(0.00379074062221), SCALE_SPLIT_ANGLE(0.00379178463481), SCALE_SPLIT_ANGLE(0.00379348872229), SCALE_SPLIT_ANGLE(0.00379467196763)}, {SCALE_SPLIT_ANGLE(0.00213995971717), SCALE_SPLIT_ANGLE(0.00214001606219), SCALE_SPLIT_ANGLE(0.00214010174386), SCALE_SPLIT_ANGLE(0.00214031711221), SCALE_SPLIT_ANGLE(0.00214052735828), 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SCALE_SPLIT_ANGLE(-0.33956736326218), SCALE_SPLIT_ANGLE(-0.33966410160065), SCALE_SPLIT_ANGLE(-0.33984366059303), SCALE_SPLIT_ANGLE(-0.34023004770279), SCALE_SPLIT_ANGLE(-0.34060218930244)}, {SCALE_SPLIT_ANGLE(-0.32115444540977), SCALE_SPLIT_ANGLE(-0.32118517160416), SCALE_SPLIT_ANGLE(-0.32123413681984), SCALE_SPLIT_ANGLE(-0.32137045264244), SCALE_SPLIT_ANGLE(-0.32152506709099), SCALE_SPLIT_ANGLE(-0.32182860374451), SCALE_SPLIT_ANGLE(-0.32256555557251), SCALE_SPLIT_ANGLE(-0.32340887188911)}, {SCALE_SPLIT_ANGLE(-0.29830518364906), SCALE_SPLIT_ANGLE(-0.29834720492363), SCALE_SPLIT_ANGLE(-0.29841497540474), SCALE_SPLIT_ANGLE(-0.29860857129097), SCALE_SPLIT_ANGLE(-0.29883739352226), SCALE_SPLIT_ANGLE(-0.29931783676147), SCALE_SPLIT_ANGLE(-0.30069103837013), SCALE_SPLIT_ANGLE(-0.30275768041611)}, {SCALE_SPLIT_ANGLE(-0.28050783276558), SCALE_SPLIT_ANGLE(-0.28055709600449), SCALE_SPLIT_ANGLE(-0.28063705563545), SCALE_SPLIT_ANGLE(-0.28086939454079), SCALE_SPLIT_ANGLE(-0.28115186095238), SCALE_SPLIT_ANGLE(-0.28177401423454), SCALE_SPLIT_ANGLE(-0.28380542993546), SCALE_SPLIT_ANGLE(-0.28782635927200)}, {SCALE_SPLIT_ANGLE(-0.26083284616470), SCALE_SPLIT_ANGLE(-0.26088821887970), SCALE_SPLIT_ANGLE(-0.26097869873047), SCALE_SPLIT_ANGLE(-0.26124575734138), SCALE_SPLIT_ANGLE(-0.26157894730568), SCALE_SPLIT_ANGLE(-0.26234790682793), SCALE_SPLIT_ANGLE(-0.26525342464447), SCALE_SPLIT_ANGLE(-0.27373573184013)}, {SCALE_SPLIT_ANGLE(-0.23960019648075), SCALE_SPLIT_ANGLE(-0.23966008424759), SCALE_SPLIT_ANGLE(-0.23975844681263), SCALE_SPLIT_ANGLE(-0.24005250632763), SCALE_SPLIT_ANGLE(-0.24042734503746), SCALE_SPLIT_ANGLE(-0.24132779240608), SCALE_SPLIT_ANGLE(-0.24525122344494), SCALE_SPLIT_ANGLE(-0.26534584164619)}, {SCALE_SPLIT_ANGLE(-0.21727462112904), SCALE_SPLIT_ANGLE(-0.21733717620373), SCALE_SPLIT_ANGLE(-0.21744027733803), SCALE_SPLIT_ANGLE(-0.21775110065937), SCALE_SPLIT_ANGLE(-0.21815298497677), SCALE_SPLIT_ANGLE(-0.21914556622505), SCALE_SPLIT_ANGLE(-0.22397416830063), SCALE_SPLIT_ANGLE(-0.27997338771820)}, {SCALE_SPLIT_ANGLE(-0.19442924857140), SCALE_SPLIT_ANGLE(-0.19449259340763), SCALE_SPLIT_ANGLE(-0.19459711015224), SCALE_SPLIT_ANGLE(-0.19491319358349), SCALE_SPLIT_ANGLE(-0.19532407820225), SCALE_SPLIT_ANGLE(-0.19634953141212), SCALE_SPLIT_ANGLE(-0.20156541466713), SCALE_SPLIT_ANGLE(-0.33650863170624)}, {SCALE_SPLIT_ANGLE(-0.17168678343296), SCALE_SPLIT_ANGLE(-0.17174907028675), SCALE_SPLIT_ANGLE(-0.17185172438622), SCALE_SPLIT_ANGLE(-0.17216137051582), SCALE_SPLIT_ANGLE(-0.17256210744381), SCALE_SPLIT_ANGLE(-0.17355349659920), SCALE_SPLIT_ANGLE(-0.17841055989265), SCALE_SPLIT_ANGLE(-0.23922458291054)}, {SCALE_SPLIT_ANGLE(-0.14965106546879), SCALE_SPLIT_ANGLE(-0.14971044659615), SCALE_SPLIT_ANGLE(-0.14980801939964), SCALE_SPLIT_ANGLE(-0.15010000765324), SCALE_SPLIT_ANGLE(-0.15047281980515), SCALE_SPLIT_ANGLE(-0.15137127041817), SCALE_SPLIT_ANGLE(-0.15533344447613), SCALE_SPLIT_ANGLE(-0.17691853642464)}, {SCALE_SPLIT_ANGLE(-0.12884500622749), SCALE_SPLIT_ANGLE(-0.12889972329140), SCALE_SPLIT_ANGLE(-0.12898920476437), SCALE_SPLIT_ANGLE(-0.12925371527672), SCALE_SPLIT_ANGLE(-0.12958449125290), SCALE_SPLIT_ANGLE(-0.13035115599632), SCALE_SPLIT_ANGLE(-0.13329000771046), SCALE_SPLIT_ANGLE(-0.14226882159710)}, {SCALE_SPLIT_ANGLE(-0.10966771095991), SCALE_SPLIT_ANGLE(-0.10971628129482), SCALE_SPLIT_ANGLE(-0.10979521274567), SCALE_SPLIT_ANGLE(-0.11002496629953), SCALE_SPLIT_ANGLE(-0.11030505597591), SCALE_SPLIT_ANGLE(-0.11092507094145), SCALE_SPLIT_ANGLE(-0.11297955363989), SCALE_SPLIT_ANGLE(-0.11718675494194)}, {SCALE_SPLIT_ANGLE(-0.09237615764141), SCALE_SPLIT_ANGLE(-0.09241757541895), SCALE_SPLIT_ANGLE(-0.09248440712690), SCALE_SPLIT_ANGLE(-0.09267570078373), SCALE_SPLIT_ANGLE(-0.09290252625942), SCALE_SPLIT_ANGLE(-0.09338124096394), SCALE_SPLIT_ANGLE(-0.09476861357689), SCALE_SPLIT_ANGLE(-0.09691139310598)}, {SCALE_SPLIT_ANGLE(-0.07020132243633), SCALE_SPLIT_ANGLE(-0.07023159414530), SCALE_SPLIT_ANGLE(-0.07027988135815), SCALE_SPLIT_ANGLE(-0.07041461765766), SCALE_SPLIT_ANGLE(-0.07056794315577), SCALE_SPLIT_ANGLE(-0.07087046653032), SCALE_SPLIT_ANGLE(-0.07161350548267), SCALE_SPLIT_ANGLE(-0.07247892767191)}, {SCALE_SPLIT_ANGLE(-0.05244029313326), SCALE_SPLIT_ANGLE(-0.05246063694358), SCALE_SPLIT_ANGLE(-0.05249277129769), SCALE_SPLIT_ANGLE(-0.05258031189442), SCALE_SPLIT_ANGLE(-0.05267634615302), SCALE_SPLIT_ANGLE(-0.05285539105535), SCALE_SPLIT_ANGLE(-0.05324416607618), SCALE_SPLIT_ANGLE(-0.05362325906754)}, {SCALE_SPLIT_ANGLE(-0.03864165395498), SCALE_SPLIT_ANGLE(-0.03865438699722), SCALE_SPLIT_ANGLE(-0.03867433220148), SCALE_SPLIT_ANGLE(-0.03872758150101), SCALE_SPLIT_ANGLE(-0.03878423944116), SCALE_SPLIT_ANGLE(-0.03888526186347), SCALE_SPLIT_ANGLE(-0.03908598423004), SCALE_SPLIT_ANGLE(-0.03926040232182)}, {SCALE_SPLIT_ANGLE(-0.02817477099597), SCALE_SPLIT_ANGLE(-0.02818230539560), SCALE_SPLIT_ANGLE(-0.02819401212037), SCALE_SPLIT_ANGLE(-0.02822477370501), SCALE_SPLIT_ANGLE(-0.02825672924519), SCALE_SPLIT_ANGLE(-0.02831180207431), SCALE_SPLIT_ANGLE(-0.02841453626752), SCALE_SPLIT_ANGLE(-0.02849724888802)}, {SCALE_SPLIT_ANGLE(-0.02037952840328), SCALE_SPLIT_ANGLE(-0.02038379199803), SCALE_SPLIT_ANGLE(-0.02039037644863), SCALE_SPLIT_ANGLE(-0.02040746808052), SCALE_SPLIT_ANGLE(-0.02042490057647), SCALE_SPLIT_ANGLE(-0.02045419998467), SCALE_SPLIT_ANGLE(-0.02050646580756), SCALE_SPLIT_ANGLE(-0.02054643817246)}, {SCALE_SPLIT_ANGLE(-0.01173160132021), SCALE_SPLIT_ANGLE(-0.01173313893378), SCALE_SPLIT_ANGLE(-0.01173550169915), SCALE_SPLIT_ANGLE(-0.01174154412001), SCALE_SPLIT_ANGLE(-0.01174758281559), SCALE_SPLIT_ANGLE(-0.01175745390356), SCALE_SPLIT_ANGLE(-0.01177425216883), SCALE_SPLIT_ANGLE(-0.01178644038737)}, {SCALE_SPLIT_ANGLE(-0.00668647419661), SCALE_SPLIT_ANGLE(-0.00668699899688), SCALE_SPLIT_ANGLE(-0.00668780226260), SCALE_SPLIT_ANGLE(-0.00668983859941), SCALE_SPLIT_ANGLE(-0.00669184979051), SCALE_SPLIT_ANGLE(-0.00669508427382), SCALE_SPLIT_ANGLE(-0.00670044496655), SCALE_SPLIT_ANGLE(-0.00670422706753)}, {SCALE_SPLIT_ANGLE(-0.00378898018971), SCALE_SPLIT_ANGLE(-0.00378915364854), SCALE_SPLIT_ANGLE(-0.00378941860981), SCALE_SPLIT_ANGLE(-0.00379008613527), SCALE_SPLIT_ANGLE(-0.00379074038938), SCALE_SPLIT_ANGLE(-0.00379178440198), SCALE_SPLIT_ANGLE(-0.00379348848946), SCALE_SPLIT_ANGLE(-0.00379467150196)}, {SCALE_SPLIT_ANGLE(-0.00213995948434), SCALE_SPLIT_ANGLE(-0.00214001582935), SCALE_SPLIT_ANGLE(-0.00214010151103), SCALE_SPLIT_ANGLE(-0.00214031687938), SCALE_SPLIT_ANGLE(-0.00214052735828), SCALE_SPLIT_ANGLE(-0.00214086123742), SCALE_SPLIT_ANGLE(-0.00214140163735), SCALE_SPLIT_ANGLE(-0.00214177393354)}, {SCALE_SPLIT_ANGLE(-0.00000001210151), SCALE_SPLIT_ANGLE(-0.00000001210151), SCALE_SPLIT_ANGLE(-0.00000001210151), SCALE_SPLIT_ANGLE(-0.00000001210151), SCALE_SPLIT_ANGLE(-0.00000001210151), SCALE_SPLIT_ANGLE(-0.00000001210151), SCALE_SPLIT_ANGLE(-0.00000001210151), SCALE_SPLIT_ANGLE(-0.00000001210151)}, }, }; #define SCALE_CLD(a) (FL2FXCONST_CFG(a)) const FIXP_CFG dequantCLD_c_l[31] = { SCALE_CLD(0.0000000316f), SCALE_CLD(0.0056233243f), SCALE_CLD(0.0099994997f), SCALE_CLD(0.0177799836f), SCALE_CLD(0.0316069759f), SCALE_CLD(0.0561454296f), SCALE_CLD(0.0791834071f), SCALE_CLD(0.1115021780f), SCALE_CLD(0.1565355062f), SCALE_CLD(0.2184644639f), SCALE_CLD(0.3015113473f), SCALE_CLD(0.3698741496f), SCALE_CLD(0.4480624795f), SCALE_CLD(0.5336171389f), SCALE_CLD(0.6219832301f), SCALE_CLD(0.7071067691f), SCALE_CLD(0.7830305696f), SCALE_CLD(0.8457261920f), SCALE_CLD(0.8940021992f), SCALE_CLD(0.9290818572f), SCALE_CLD(0.9534626007f), SCALE_CLD(0.9758449197f), SCALE_CLD(0.9876723289f), SCALE_CLD(0.9937641621f), SCALE_CLD(0.9968600869f), SCALE_CLD(0.9984226227f), SCALE_CLD(0.9995003939f), SCALE_CLD(0.9998419285f), SCALE_CLD(0.9999499917f), SCALE_CLD(0.9999842048f), /*SCALE_CLD(1.0000000000f)*/ FX_DBL2FX_CFG(MAXVAL_DBL)}; #define SC_H(a) (FL2FXCONST_CFG(a)) #define DATA_TYPE_H FIXP_CFG /* not correlated tables */ const DATA_TYPE_H H11_nc[31][8] = { {SC_H(0.0000000316f), SC_H(0.0000000296f), SC_H(0.0000000266f), SC_H(0.0000000190f), SC_H(0.0000000116f), SC_H(0.0000000000f), SC_H(-0.0000000186f), SC_H(-0.0000000313f)}, {SC_H(0.0056233243f), SC_H(0.0052728835f), SC_H(0.0047394098f), SC_H(0.0033992692f), SC_H(0.0020946222f), SC_H(0.0000316215f), SC_H(-0.0032913829f), SC_H(-0.0055664564f)}, {SC_H(0.0099994997f), SC_H(0.0093815643f), SC_H(0.0084402543f), SC_H(0.0060722125f), SC_H(0.0037622179f), SC_H(0.0000999898f), SC_H(-0.0058238208f), SC_H(-0.0098974844f)}, {SC_H(0.0177799836f), SC_H(0.0166974831f), SC_H(0.0150465844f), SC_H(0.0108831404f), SC_H(0.0068073822f), SC_H(0.0003161267f), SC_H(-0.0102626514f), SC_H(-0.0175957214f)}, {SC_H(0.0316069759f), SC_H(0.0297324844f), SC_H(0.0268681273f), SC_H(0.0196138974f), SC_H(0.0124691967f), SC_H(0.0009989988f), SC_H(-0.0179452803f), SC_H(-0.0312700421f)}, {SC_H(0.0561454296f), SC_H(0.0529650487f), SC_H(0.0480896905f), SC_H(0.0356564634f), SC_H(0.0232860073f), SC_H(0.0031523081f), SC_H(-0.0309029408f), SC_H(-0.0555154830f)}, {SC_H(0.0791834071f), SC_H(0.0748842582f), SC_H(0.0682762116f), SC_H(0.0513241664f), SC_H(0.0343080349f), SC_H(0.0062700072f), SC_H(-0.0422340371f), SC_H(-0.0782499388f)}, {SC_H(0.1115021780f), SC_H(0.1057924852f), SC_H(0.0969873071f), SC_H(0.0742305145f), SC_H(0.0511277616f), SC_H(0.0124327289f), SC_H(-0.0566596612f), SC_H(-0.1100896299f)}, {SC_H(0.1565355062f), SC_H(0.1491366178f), SC_H(0.1376826316f), SC_H(0.1078186408f), SC_H(0.0770794004f), SC_H(0.0245033558f), SC_H(-0.0735980421f), SC_H(-0.1543303132f)}, {SC_H(0.2184644639f), SC_H(0.2091979682f), SC_H(0.1947948188f), SC_H(0.1568822265f), SC_H(0.1172478944f), SC_H(0.0477267131f), SC_H(-0.0899507254f), SC_H(-0.2148526460f)}, {SC_H(0.3015113473f), SC_H(0.2904391289f), SC_H(0.2731673419f), SC_H(0.2273024023f), SC_H(0.1786239147f), SC_H(0.0909090787f), SC_H(-0.0964255333f), SC_H(-0.2951124907f)}, {SC_H(0.3698741496f), SC_H(0.3578284085f), SC_H(0.3390066922f), SC_H(0.2888108492f), SC_H(0.2351117432f), SC_H(0.1368068755f), SC_H(-0.0850296095f), SC_H(-0.3597966135f)}, {SC_H(0.4480624795f), SC_H(0.4354025424f), SC_H(0.4156077504f), SC_H(0.3627120256f), SC_H(0.3058823943f), SC_H(0.2007599771f), SC_H(-0.0484020934f), SC_H(-0.4304940701f)}, {SC_H(0.5336171389f), SC_H(0.5208471417f), SC_H(0.5008935928f), SC_H(0.4476420581f), SC_H(0.3905044496f), SC_H(0.2847472429f), SC_H(0.0276676007f), SC_H(-0.4966579080f)}, {SC_H(0.6219832301f), SC_H(0.6096963882f), SC_H(0.5905415416f), SC_H(0.5396950245f), SC_H(0.4856070578f), SC_H(0.3868631124f), SC_H(0.1531652957f), SC_H(-0.5045361519f)}, {SC_H(0.7071067691f), SC_H(0.6958807111f), SC_H(0.6784504056f), SC_H(0.6326373219f), SC_H(0.5847306848f), SC_H(0.4999999702f), SC_H(0.3205464482f), SC_H(0.0500000045f)}, {SC_H(0.7830305696f), SC_H(0.7733067870f), SC_H(0.7582961321f), SC_H(0.7194055915f), SC_H(0.6797705293f), SC_H(0.6131368876f), SC_H(0.4997332692f), SC_H(0.6934193969f)}, {SC_H(0.8457261920f), SC_H(0.8377274871f), SC_H(0.8254694939f), SC_H(0.7942851782f), SC_H(0.7635439038f), SC_H(0.7152527571f), SC_H(0.6567122936f), SC_H(0.8229061961f)}, {SC_H(0.8940021992f), SC_H(0.8877248168f), SC_H(0.8781855106f), SC_H(0.8544237614f), SC_H(0.8318918347f), SC_H(0.7992399335f), SC_H(0.7751275301f), SC_H(0.8853276968f)}, {SC_H(0.9290818572f), SC_H(0.9243524075f), SC_H(0.9172304869f), SC_H(0.8998877406f), SC_H(0.8841174841f), SC_H(0.8631930947f), SC_H(0.8565139771f), SC_H(0.9251161218f)}, {SC_H(0.9534626007f), SC_H(0.9500193000f), SC_H(0.9448821545f), SC_H(0.9326565266f), SC_H(0.9220023751f), SC_H(0.9090909362f), SC_H(0.9096591473f), SC_H(0.9514584541f)}, {SC_H(0.9758449197f), SC_H(0.9738122821f), SC_H(0.9708200693f), SC_H(0.9639287591f), SC_H(0.9582763910f), SC_H(0.9522733092f), SC_H(0.9553207159f), SC_H(0.9750427008f)}, {SC_H(0.9876723289f), SC_H(0.9865267277f), SC_H(0.9848603010f), SC_H(0.9811310172f), SC_H(0.9782302976f), SC_H(0.9754966497f), SC_H(0.9779621363f), SC_H(0.9873252511f)}, {SC_H(0.9937641621f), SC_H(0.9931397438f), SC_H(0.9922404289f), SC_H(0.9902750254f), SC_H(0.9888116717f), SC_H(0.9875672460f), SC_H(0.9891131520f), SC_H(0.9936066866f)}, {SC_H(0.9968600869f), SC_H(0.9965277910f), SC_H(0.9960530400f), SC_H(0.9950347543f), SC_H(0.9943022728f), SC_H(0.9937300086f), SC_H(0.9946073294f), SC_H(0.9967863560f)}, {SC_H(0.9984226227f), SC_H(0.9982488155f), SC_H(0.9980020523f), SC_H(0.9974802136f), SC_H(0.9971146584f), SC_H(0.9968476892f), SC_H(0.9973216057f), SC_H(0.9983873963f)}, {SC_H(0.9995003939f), SC_H(0.9994428754f), SC_H(0.9993617535f), SC_H(0.9991930723f), SC_H(0.9990783334f), SC_H(0.9990010262f), SC_H(0.9991616607f), SC_H(0.9994897842f)}, {SC_H(0.9998419285f), SC_H(0.9998232722f), SC_H(0.9997970462f), SC_H(0.9997430444f), SC_H(0.9997069836f), SC_H(0.9996838570f), SC_H(0.9997364879f), SC_H(0.9998386502f)}, {SC_H(0.9999499917f), SC_H(0.9999440312f), SC_H(0.9999356270f), SC_H(0.9999184012f), SC_H(0.9999070764f), SC_H(0.9998999834f), SC_H(0.9999169707f), SC_H(0.9999489784f)}, {SC_H(0.9999842048f), SC_H(0.9999822974f), SC_H(0.9999796152f), SC_H(0.9999741912f), SC_H(0.9999706149f), SC_H(0.9999684095f), SC_H(0.9999738336f), SC_H(0.9999839067f)}, /* { SC_H( 1.0000000000f), SC_H( 1.0000000000f), SC_H( 1.0000000000f), SC_H( 1.0000000000f), SC_H( 1.0000000000f), SC_H( 1.0000000000f), SC_H( 1.0000000000f), SC_H( 1.0000000000f)} */ {FX_DBL2FX_CFG(MAXVAL_DBL), FX_DBL2FX_CFG(MAXVAL_DBL), FX_DBL2FX_CFG(MAXVAL_DBL), FX_DBL2FX_CFG(MAXVAL_DBL), FX_DBL2FX_CFG(MAXVAL_DBL), FX_DBL2FX_CFG(MAXVAL_DBL), FX_DBL2FX_CFG(MAXVAL_DBL), FX_DBL2FX_CFG(MAXVAL_DBL)}}; const DATA_TYPE_H H12_nc[31][8] = { {SC_H(0.0000000000f), SC_H(0.0000000110f), SC_H(0.0000000171f), SC_H(0.0000000253f), SC_H(0.0000000294f), SC_H(0.0000000316f), SC_H(0.0000000256f), SC_H(0.0000000045f)}, {SC_H(0.0000000000f), SC_H(0.0019540924f), SC_H(0.0030265113f), SC_H(0.0044795922f), SC_H(0.0052186525f), SC_H(0.0056232354f), SC_H(0.0045594489f), SC_H(0.0007977085f)}, {SC_H(0.0000000000f), SC_H(0.0034606720f), SC_H(0.0053620986f), SC_H(0.0079446984f), SC_H(0.0092647560f), SC_H(0.0099989995f), SC_H(0.0081285369f), SC_H(0.0014247064f)}, {SC_H(0.0000000000f), SC_H(0.0061091618f), SC_H(0.0094724922f), SC_H(0.0140600521f), SC_H(0.0164252054f), SC_H(0.0177771728f), SC_H(0.0145191532f), SC_H(0.0025531140f)}, {SC_H(0.0000000000f), SC_H(0.0107228858f), SC_H(0.0166464616f), SC_H(0.0247849934f), SC_H(0.0290434174f), SC_H(0.0315911844f), SC_H(0.0260186065f), SC_H(0.0046027615f)}, {SC_H(0.0000000000f), SC_H(0.0186282862f), SC_H(0.0289774220f), SC_H(0.0433696397f), SC_H(0.0510888547f), SC_H(0.0560568646f), SC_H(0.0468755551f), SC_H(0.0083869267f)}, {SC_H(0.0000000000f), SC_H(0.0257363543f), SC_H(0.0401044972f), SC_H(0.0602979437f), SC_H(0.0713650510f), SC_H(0.0789347738f), SC_H(0.0669798329f), SC_H(0.0121226767f)}, {SC_H(0.0000000000f), SC_H(0.0352233723f), SC_H(0.0550108925f), SC_H(0.0832019597f), SC_H(0.0990892947f), SC_H(0.1108068749f), SC_H(0.0960334241f), SC_H(0.0176920593f)}, {SC_H(0.0000000000f), SC_H(0.0475566536f), SC_H(0.0744772255f), SC_H(0.1134835035f), SC_H(0.1362429112f), SC_H(0.1546057910f), SC_H(0.1381545961f), SC_H(0.0261824392f)}, {SC_H(0.0000000000f), SC_H(0.0629518181f), SC_H(0.0989024863f), SC_H(0.1520351619f), SC_H(0.1843357086f), SC_H(0.2131874412f), SC_H(0.1990868896f), SC_H(0.0395608991f)}, {SC_H(0.0000000000f), SC_H(0.0809580907f), SC_H(0.1276271492f), SC_H(0.1980977356f), SC_H(0.2429044843f), SC_H(0.2874797881f), SC_H(0.2856767476f), SC_H(0.0617875643f)}, {SC_H(0.0000000000f), SC_H(0.0936254337f), SC_H(0.1479234397f), SC_H(0.2310739607f), SC_H(0.2855334580f), SC_H(0.3436433673f), SC_H(0.3599678576f), SC_H(0.0857512727f)}, {SC_H(0.0000000000f), SC_H(0.1057573780f), SC_H(0.1674221754f), SC_H(0.2630588412f), SC_H(0.3274079263f), SC_H(0.4005688727f), SC_H(0.4454404712f), SC_H(0.1242370531f)}, {SC_H(0.0000000000f), SC_H(0.1160409302f), SC_H(0.1839915067f), SC_H(0.2904545665f), SC_H(0.3636667728f), SC_H(0.4512939751f), SC_H(0.5328993797f), SC_H(0.1951362640f)}, {SC_H(0.0000000000f), SC_H(0.1230182052f), SC_H(0.1952532977f), SC_H(0.3091802597f), SC_H(0.3886501491f), SC_H(0.4870318770f), SC_H(0.6028295755f), SC_H(0.3637395203f)}, {SC_H(0.0000000000f), SC_H(0.1254990250f), SC_H(0.1992611140f), SC_H(0.3158638775f), SC_H(0.3976053298f), SC_H(0.5000000000f), SC_H(0.6302776933f), SC_H(0.7053368092f)}, {SC_H(0.0000000000f), SC_H(0.1230182052f), SC_H(0.1952533126f), SC_H(0.3091802597f), SC_H(0.3886501491f), SC_H(0.4870319068f), SC_H(0.6028295755f), SC_H(0.3637394905f)}, {SC_H(0.0000000000f), SC_H(0.1160409302f), SC_H(0.1839915216f), SC_H(0.2904545665f), SC_H(0.3636668026f), SC_H(0.4512939751f), SC_H(0.5328993797f), SC_H(0.1951362044f)}, {SC_H(0.0000000000f), SC_H(0.1057573855f), SC_H(0.1674221754f), SC_H(0.2630588710f), SC_H(0.3274079263f), SC_H(0.4005688727f), SC_H(0.4454405010f), SC_H(0.1242370382f)}, {SC_H(0.0000000000f), SC_H(0.0936254337f), SC_H(0.1479234397f), SC_H(0.2310739607f), SC_H(0.2855334580f), SC_H(0.3436433673f), SC_H(0.3599678576f), SC_H(0.0857512653f)}, {SC_H(0.0000000000f), SC_H(0.0809580907f), SC_H(0.1276271492f), SC_H(0.1980977207f), SC_H(0.2429044843f), SC_H(0.2874797881f), SC_H(0.2856767476f), SC_H(0.0617875606f)}, {SC_H(0.0000000000f), SC_H(0.0629518107f), SC_H(0.0989024863f), SC_H(0.1520351619f), SC_H(0.1843357235f), SC_H(0.2131874412f), SC_H(0.1990868896f), SC_H(0.0395609401f)}, {SC_H(0.0000000000f), SC_H(0.0475566462f), SC_H(0.0744772255f), SC_H(0.1134835184f), SC_H(0.1362429112f), SC_H(0.1546057761f), SC_H(0.1381545961f), SC_H(0.0261824802f)}, {SC_H(0.0000000000f), SC_H(0.0352233797f), SC_H(0.0550108962f), SC_H(0.0832019448f), SC_H(0.0990892798f), SC_H(0.1108068526f), SC_H(0.0960334465f), SC_H(0.0176920686f)}, {SC_H(0.0000000000f), SC_H(0.0257363524f), SC_H(0.0401044935f), SC_H(0.0602979474f), SC_H(0.0713650808f), SC_H(0.0789347589f), SC_H(0.0669797957f), SC_H(0.0121226516f)}, {SC_H(0.0000000000f), SC_H(0.0186282881f), SC_H(0.0289774258f), SC_H(0.0433696248f), SC_H(0.0510888547f), SC_H(0.0560568906f), SC_H(0.0468755886f), SC_H(0.0083869714f)}, {SC_H(0.0000000000f), SC_H(0.0107228830f), SC_H(0.0166464727f), SC_H(0.0247849822f), SC_H(0.0290434249f), SC_H(0.0315911621f), SC_H(0.0260186475f), SC_H(0.0046027377f)}, {SC_H(0.0000000000f), SC_H(0.0061091576f), SC_H(0.0094724894f), SC_H(0.0140600465f), SC_H(0.0164251942f), SC_H(0.0177771524f), SC_H(0.0145191504f), SC_H(0.0025530567f)}, {SC_H(0.0000000000f), SC_H(0.0034606743f), SC_H(0.0053620976f), SC_H(0.0079446994f), SC_H(0.0092647672f), SC_H(0.0099990256f), SC_H(0.0081285043f), SC_H(0.0014247177f)}, {SC_H(0.0000000000f), SC_H(0.0019540912f), SC_H(0.0030265225f), SC_H(0.0044795908f), SC_H(0.0052186381f), SC_H(0.0056232223f), SC_H(0.0045594289f), SC_H(0.0007977359f)}, {SC_H(0.0000000000f), SC_H(0.0000000149f), SC_H(0.0000000298f), SC_H(0.0000000298f), SC_H(0.0000000000f), SC_H(0.0000000596f), SC_H(0.0000000000f), SC_H(0.0000000000f)}}; /* for (i=0; i<31; i++) { cld = dequantCLD[i]; val = (float)(FDKexp(cld/dbe)/(1+FDKexp(cld/dbe))); val = (float)(dbe*FDKlog(val)); } */ #define SCALE_CLD_C1C2(a) (FL2FXCONST_DBL(a / (float)(1 << SF_CLD_C1C2))) const FIXP_DBL dequantCLD_c1[31] = {SCALE_CLD_C1C2(-1.5000000000000000e+002f), SCALE_CLD_C1C2(-4.5000137329101563e+001f), SCALE_CLD_C1C2(-4.0000434875488281e+001f), SCALE_CLD_C1C2(-3.5001373291015625e+001f), SCALE_CLD_C1C2(-3.0004341125488281e+001f), SCALE_CLD_C1C2(-2.5013711929321289e+001f), SCALE_CLD_C1C2(-2.2027315139770508e+001f), SCALE_CLD_C1C2(-1.9054332733154297e+001f), SCALE_CLD_C1C2(-1.6107742309570313e+001f), SCALE_CLD_C1C2(-1.3212384223937988e+001f), SCALE_CLD_C1C2(-1.0413927078247070e+001f), SCALE_CLD_C1C2(-8.6389207839965820e+000f), SCALE_CLD_C1C2(-6.9732279777526855e+000f), SCALE_CLD_C1C2(-5.4554042816162109e+000f), SCALE_CLD_C1C2(-4.1244258880615234e+000f), SCALE_CLD_C1C2(-3.0102999210357666e+000f), SCALE_CLD_C1C2(-2.1244258880615234e+000f), SCALE_CLD_C1C2(-1.4554045200347900e+000f), SCALE_CLD_C1C2(-9.7322785854339600e-001f), SCALE_CLD_C1C2(-6.3892036676406860e-001f), SCALE_CLD_C1C2(-4.1392669081687927e-001f), SCALE_CLD_C1C2(-2.1238386631011963e-001f), SCALE_CLD_C1C2(-1.0774217545986176e-001f), SCALE_CLD_C1C2(-5.4333221167325974e-002f), SCALE_CLD_C1C2(-2.7315950021147728e-002f), SCALE_CLD_C1C2(-1.3711934909224510e-002f), SCALE_CLD_C1C2(-4.3406565673649311e-003f), SCALE_CLD_C1C2(-1.3732088264077902e-003f), SCALE_CLD_C1C2(-4.3438826105557382e-004f), SCALE_CLD_C1C2(-1.3745666365139186e-004f), SCALE_CLD_C1C2(0.0000000000000000e+000f)}; /* sac_stp */ /* none scaled */ const FIXP_CFG BP__FDK[] = {FL2FXCONST_CFG(0.73919999599457), FL2FXCONST_CFG(0.97909998893738), FL2FXCONST_CFG(0.99930000305176)}; /* scaled with 26 bits */ const FIXP_CFG BP_GF__FDK[] = { FL2FXCONST_CFG(0.00000000643330), FL2FXCONST_CFG(0.00004396850232), FL2FXCONST_CFG(0.00087456948552), FL2FXCONST_CFG(0.00474648220243), FL2FXCONST_CFG(0.01717987244800), FL2FXCONST_CFG(0.04906742491073), FL2FXCONST_CFG(0.10569518656729), FL2FXCONST_CFG(0.21165767592653), FL2FXCONST_CFG(0.36036762478024), FL2FXCONST_CFG(0.59894182766948), FL2FXCONST_CFG(0.81641678929129), FL2FXCONST_CFG(0.97418481133397), FL2FXCONST_CFG(0.99575411610845), FL2FXCONST_CFG(0.88842666281361), FL2FXCONST_CFG(0.79222317063736), FL2FXCONST_CFG(0.70828604318604), FL2FXCONST_CFG(0.66395054816338), FL2FXCONST_CFG(0.64633739516952), FL2FXCONST_CFG(0.66098278185255)}; /* sac_bitdec */ const INT samplingFreqTable[16] = {96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000, 7350, 0, 0, 0}; const UCHAR freqResTable[] = {0, 28, 20, 14, 10, 7, 5, 4}; const UCHAR freqResTable_LD[] = {0, 23, 15, 12, 9, 7, 5, 4}; const UCHAR tempShapeChanTable[][8] = {{5, 5, 4, 6, 6, 4, 4, 2}, {5, 5, 5, 7, 7, 4, 4, 2}}; const TREEPROPERTIES treePropertyTable[] = { {1, 6, 5, 0, {0, 0, 0, 0, 1}}, {1, 6, 5, 0, {0, 0, 1, 0, 0}}, {2, 6, 3, 1, {1, 0, 0, 0, 0}}, {2, 8, 5, 1, {1, 0, 0, 0, 0}}, {2, 8, 5, 1, {1, 0, 0, 0, 0}}, {6, 8, 2, 0, {0, 0, 0, 0, 0}}, {6, 8, 2, 0, {0, 0, 0, 0, 0}}, {1, 2, 1, 0, {0, 0, 0, 0, 0}}}; const SCHAR kernels_4_to_71[MAX_HYBRID_BANDS] = { 0, 0, 0, 0, 0, 0, 0, 0, 1, 1, 1, 1, 1, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3}; const SCHAR kernels_5_to_71[MAX_HYBRID_BANDS] = { 0, 0, 0, 0, 0, 0, 1, 1, 1, 1, 2, 2, 2, 2, 2, 2, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4}; const SCHAR kernels_7_to_71[MAX_HYBRID_BANDS] = { 0, 0, 0, 0, 0, 0, 1, 1, 2, 2, 2, 3, 3, 3, 3, 4, 4, 4, 4, 4, 4, 5, 5, 5, 5, 5, 5, 5, 5, 5, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6}; const SCHAR kernels_10_to_71[MAX_HYBRID_BANDS] = { 0, 0, 0, 0, 1, 1, 2, 2, 3, 3, 4, 4, 5, 5, 6, 6, 7, 7, 7, 7, 7, 8, 8, 8, 8, 8, 8, 8, 8, 8, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9}; const SCHAR kernels_14_to_71[MAX_HYBRID_BANDS] = { 0, 0, 0, 0, 1, 1, 2, 3, 4, 4, 5, 6, 6, 7, 7, 8, 8, 8, 9, 9, 9, 10, 10, 10, 10, 11, 11, 11, 11, 11, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13}; const SCHAR kernels_20_to_71[MAX_HYBRID_BANDS] = { 0, 0, 1, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 14, 15, 15, 15, 16, 16, 16, 16, 17, 17, 17, 17, 17, 18, 18, 18, 18, 18, 18, 18, 18, 18, 18, 18, 18, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19, 19}; const SCHAR kernels_28_to_71[MAX_HYBRID_BANDS] = { 0, 0, 1, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 17, 18, 18, 19, 19, 20, 20, 21, 21, 21, 22, 22, 22, 23, 23, 23, 23, 24, 24, 24, 24, 24, 25, 25, 25, 25, 25, 25, 26, 26, 26, 26, 26, 26, 26, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27, 27}; const SCHAR kernels_4_to_64[MAX_HYBRID_BANDS] = { 0, 0, 1, 1, 1, 1, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3}; const SCHAR kernels_5_to_64[MAX_HYBRID_BANDS] = { 0, 1, 1, 2, 2, 2, 2, 2, 2, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4}; const SCHAR kernels_7_to_64[MAX_HYBRID_BANDS] = { 0, 1, 2, 2, 3, 3, 3, 3, 4, 4, 4, 4, 4, 4, 5, 5, 5, 5, 5, 5, 5, 5, 5, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6}; const SCHAR kernels_9_to_64[MAX_HYBRID_BANDS] = { 0, 1, 2, 3, 3, 4, 4, 5, 5, 6, 6, 6, 6, 6, 7, 7, 7, 7, 7, 7, 7, 7, 7, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8}; const SCHAR kernels_12_to_64[MAX_HYBRID_BANDS] = { 0, 1, 2, 3, 4, 4, 5, 5, 6, 6, 6, 7, 7, 7, 8, 8, 8, 8, 9, 9, 9, 9, 9, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11}; const SCHAR kernels_15_to_64[MAX_HYBRID_BANDS] = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 9, 10, 10, 10, 11, 11, 11, 11, 12, 12, 12, 12, 12, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14}; const SCHAR kernels_23_to_64[MAX_HYBRID_BANDS] = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 12, 13, 13, 14, 14, 15, 15, 16, 16, 16, 17, 17, 17, 18, 18, 18, 18, 19, 19, 19, 19, 19, 20, 20, 20, 20, 20, 20, 21, 21, 21, 21, 21, 21, 21, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22}; const UCHAR mapping_15_to_23[MAX_PARAMETER_BANDS_LD] = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 9, 10, 10, 11, 11, 12, 12, 13, 13, 13, 14, 14, 14}; const UCHAR mapping_12_to_23[MAX_PARAMETER_BANDS_LD] = { 0, 1, 2, 3, 4, 4, 5, 5, 6, 6, 6, 7, 7, 8, 8, 9, 9, 10, 10, 10, 11, 11, 11}; const UCHAR mapping_9_to_23[MAX_PARAMETER_BANDS_LD] = { 0, 1, 2, 3, 3, 4, 4, 5, 5, 6, 6, 6, 6, 7, 7, 7, 7, 8, 8, 8, 8, 8, 8}; const UCHAR mapping_7_to_23[MAX_PARAMETER_BANDS_LD] = { 0, 1, 2, 2, 3, 3, 3, 3, 4, 4, 4, 4, 4, 5, 5, 5, 5, 6, 6, 6, 6, 6, 6}; const UCHAR mapping_5_to_23[MAX_PARAMETER_BANDS_LD] = { 0, 1, 1, 2, 2, 2, 2, 2, 2, 3, 3, 3, 3, 3, 3, 3, 3, 4, 4, 4, 4, 4, 4}; const UCHAR mapping_4_to_23[MAX_PARAMETER_BANDS_LD] = { 0, 0, 1, 1, 1, 1, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 3, 3, 3, 3, 3, 3}; const FIXP_CFG clipGainTable__FDK[] = { /*CLIP_PROTECT_GAIN_0(1.000000f)*/ FX_DBL2FX_CFG(MAXVAL_DBL), CLIP_PROTECT_GAIN_1(1.189207f), CLIP_PROTECT_GAIN_1(1.414213f), CLIP_PROTECT_GAIN_1(1.681792f), /*CLIP_PROTECT_GAIN_1(2.000000f)*/ FX_DBL2FX_CFG(MAXVAL_DBL), CLIP_PROTECT_GAIN_2(2.378414f), CLIP_PROTECT_GAIN_2(2.828427f), /*CLIP_PROTECT_GAIN_2(4.000000f)*/ FX_DBL2FX_CFG(MAXVAL_DBL)}; const UCHAR clipGainSFTable__FDK[] = {0, 1, 1, 1, 1, 2, 2, 2}; const UCHAR pbStrideTable[] = {1, 2, 5, 28}; const int smgTimeTable[] = {64, 128, 256, 512}; /* table is scaled by factor 0.5 */ const FIXP_CFG envShapeDataTable__FDK[5][2] = { {FL2FXCONST_CFG(0.25000000000000f), FL2FXCONST_CFG(0.25000000000000f)}, {FL2FXCONST_CFG(0.35355339059327f), FL2FXCONST_CFG(0.31498026247372f)}, {FL2FXCONST_CFG(0.50000000000000f), FL2FXCONST_CFG(0.39685026299205f)}, {FL2FXCONST_CFG(0.70710678118655f), FL2FXCONST_CFG(0.50000000000000f)}, {/*FL2FXCONST_CFG( 1.00000000000000f)*/ FX_DBL2FX_CFG(MAXVAL_DBL), FL2FXCONST_CFG(0.62996052494744f)}}; /* sac_calcM1andM2 */ const SCHAR row2channelSTP[][MAX_M2_INPUT] = {{0, 1}, {0, 3}, {0, 2}, {0, 4}, {0, 4}, {0, 2}, {-1, 2}, {0, 1}}; const SCHAR row2channelGES[][MAX_M2_INPUT] = {{0, 1}, {0, 3}, {0, 3}, {0, 5}, {0, 5}, {0, 2}, {-1, 2}, {0, 1}}; const SCHAR row2residual[][MAX_M2_INPUT] = {{-1, 0}, {-1, 0}, {-1, -1}, {-1, -1}, {-1, -1}, {-1, -1}, {-1, -1}, {-1, 0}}; /******************************************************************************* Functionname: sac_getCLDValues ******************************************************************************* Description: Get CLD values from table index. Arguments: index: Table index *cu, *cl : Pointer to locations where resulting values will be written to. Return: nothing *******************************************************************************/ void SpatialDequantGetCLDValues(int index, FIXP_DBL* cu, FIXP_DBL* cl) { *cu = FX_CFG2FX_DBL(dequantCLD_c_l[index]); *cl = FX_CFG2FX_DBL(dequantCLD_c_l[31 - 1 - index]); } void SpatialDequantGetCLD2Values(int idx, FIXP_DBL* x) { *x = FX_CFG2FX_DBL(dequantCLD__FDK[idx]); } fdk-aac-2.0.2/libSACdec/src/sac_rom.h000066400000000000000000000210371404225310700171360ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround decoder library ************************* Author(s): Description: SAC Dec tables *******************************************************************************/ #ifndef SAC_ROM_H #define SAC_ROM_H #include "FDK_archdef.h" #include "sac_dec_interface.h" #include "huff_nodes.h" #include "sac_bitdec.h" #include "machine_type.h" /* Global ROM table data type: */ #define FIXP_CFG FIXP_DBL #define FX_CFG2FX_DBL #define FX_CFG2FX_SGL FX_DBL2FX_SGL #define CFG(a) FIXP_DBL(a) #define FL2FXCONST_CFG FL2FXCONST_DBL #define FX_DBL2FX_CFG(x) ((FIXP_DBL)(x)) /* others */ #define SCALE_INV_ICC (2) #define G_dd_SCALE (2) #define QCC_SCALE 1 #define M1M2_DATA FIXP_DBL #define M1M2_CDATA FIXP_DBL #define M1M2_CDATA2FX_DBL(a) (a) #define FX_DBL2M1M2_CDATA(a) (a) #define CLIP_PROTECT_GAIN_0(x) FL2FXCONST_CFG(((x) / (float)(1 << 0))) #define CLIP_PROTECT_GAIN_1(x) FL2FXCONST_CFG(((x) / (float)(1 << 1))) #define CLIP_PROTECT_GAIN_2(x) FL2FXCONST_CFG(((x) / (float)(1 << 2))) #define SF_CLD_C1C2 (9) extern const FIXP_CFG dequantCPC__FDK[]; extern const FIXP_CFG dequantICC__FDK[8]; extern const FIXP_CFG dequantCLD__FDK[31]; #define IPD_SCALE (5) #define PI__IPD (FL2FXCONST_DBL(3.1415926535897932f / (float)(1 << IPD_SCALE))) /* Define for PI*2 for better precision in SpatialDecApplyPhase() */ #define PIx2__IPD \ (FL2FXCONST_DBL(3.1415926535897932f / (float)(1 << (IPD_SCALE - 1)))) extern const FIXP_CFG dequantIPD__FDK[16]; extern const FIXP_DBL dequantIPD_CLD_ICC_splitAngle__FDK[15][31][8]; extern const FIXP_CFG H11_nc[31][8]; extern const FIXP_CFG H12_nc[31][8]; extern const FIXP_DBL dequantCLD_c1[31]; extern const FIXP_CFG BP__FDK[]; extern const FIXP_CFG BP_GF__FDK[]; extern const SCHAR row2channelSTP[][MAX_M2_INPUT]; /* sac_bitdec */ extern const INT samplingFreqTable[16]; extern const UCHAR freqResTable[]; extern const UCHAR freqResTable_LD[]; extern const UCHAR tempShapeChanTable[2][8]; extern const TREEPROPERTIES treePropertyTable[]; extern const SCHAR kernels_4_to_71[MAX_HYBRID_BANDS]; extern const SCHAR kernels_5_to_71[MAX_HYBRID_BANDS]; extern const SCHAR kernels_7_to_71[MAX_HYBRID_BANDS]; extern const SCHAR kernels_10_to_71[MAX_HYBRID_BANDS]; extern const SCHAR kernels_14_to_71[MAX_HYBRID_BANDS]; extern const SCHAR kernels_20_to_71[MAX_HYBRID_BANDS]; extern const SCHAR kernels_28_to_71[MAX_HYBRID_BANDS]; extern const UCHAR mapping_4_to_28[MAX_PARAMETER_BANDS]; extern const UCHAR mapping_5_to_28[MAX_PARAMETER_BANDS]; extern const UCHAR mapping_7_to_28[MAX_PARAMETER_BANDS]; extern const UCHAR mapping_10_to_28[MAX_PARAMETER_BANDS]; extern const UCHAR mapping_14_to_28[MAX_PARAMETER_BANDS]; extern const UCHAR mapping_20_to_28[MAX_PARAMETER_BANDS]; extern const SCHAR kernels_4_to_64[MAX_HYBRID_BANDS]; extern const SCHAR kernels_5_to_64[MAX_HYBRID_BANDS]; extern const SCHAR kernels_7_to_64[MAX_HYBRID_BANDS]; extern const SCHAR kernels_9_to_64[MAX_HYBRID_BANDS]; extern const SCHAR kernels_12_to_64[MAX_HYBRID_BANDS]; extern const SCHAR kernels_15_to_64[MAX_HYBRID_BANDS]; extern const SCHAR kernels_23_to_64[MAX_HYBRID_BANDS]; extern const UCHAR mapping_15_to_23[MAX_PARAMETER_BANDS_LD]; extern const UCHAR mapping_12_to_23[MAX_PARAMETER_BANDS_LD]; extern const UCHAR mapping_9_to_23[MAX_PARAMETER_BANDS_LD]; extern const UCHAR mapping_7_to_23[MAX_PARAMETER_BANDS_LD]; extern const UCHAR mapping_5_to_23[MAX_PARAMETER_BANDS_LD]; extern const UCHAR mapping_4_to_23[MAX_PARAMETER_BANDS_LD]; extern const FIXP_CFG clipGainTable__FDK[]; extern const UCHAR clipGainSFTable__FDK[]; extern const UCHAR pbStrideTable[]; extern const int smgTimeTable[]; extern const FIXP_CFG envShapeDataTable__FDK[5][2]; extern const SCHAR row2channelGES[][MAX_M2_INPUT]; /* sac_calcM1andM2 */ extern const SCHAR row2residual[][MAX_M2_INPUT]; void SpatialDequantGetCLDValues(int index, FIXP_DBL* cu, FIXP_DBL* cl); void SpatialDequantGetCLD2Values(int index, FIXP_DBL* x); /* External helper functions */ static inline int SacGetHybridSubbands(int qmfSubbands) { return qmfSubbands - MAX_QMF_BANDS_TO_HYBRID + 10; } #endif /* SAC_ROM_H */ fdk-aac-2.0.2/libSACdec/src/sac_smoothing.cpp000066400000000000000000000255201404225310700207040ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround decoder library ************************* Author(s): Description: SAC Dec parameter smoothing *******************************************************************************/ #include "sac_dec.h" #include "sac_bitdec.h" #include "sac_smoothing.h" #include "sac_rom.h" /******************************************************************************* Functionname: calcFilterCoeff ******************************************************************************* Description: Arguments: Input: Output: *******************************************************************************/ static FIXP_DBL calcFilterCoeff__FDK(spatialDec *self, int ps, const SPATIAL_BS_FRAME *frame) { int dSlots; FIXP_DBL delta; dSlots = frame->paramSlot[ps] - self->smoothState->prevParamSlot; if (dSlots <= 0) { dSlots += self->timeSlots; } delta = fDivNorm(dSlots, self->smgTime[ps]); return delta; } /******************************************************************************* Functionname: getSmoothOnOff ******************************************************************************* Description: Arguments: Input: Output: *******************************************************************************/ static int getSmoothOnOff(spatialDec *self, int ps, int pb) { int smoothBand = 0; smoothBand = self->smgData[ps][pb]; return smoothBand; } void SpatialDecSmoothM1andM2(spatialDec *self, const SPATIAL_BS_FRAME *frame, int ps) { FIXP_DBL delta__FDK; FIXP_DBL one_minus_delta__FDK; int pb, row, col; int residualBands = 0; if (self->residualCoding) { int i; int boxes = self->numOttBoxes; for (i = 0; i < boxes; i++) { if (self->residualBands[i] > residualBands) { residualBands = self->residualBands[i]; } } } delta__FDK = calcFilterCoeff__FDK(self, ps, frame); if (delta__FDK == /*FL2FXCONST_DBL(1.0f)*/ (FIXP_DBL)MAXVAL_DBL) one_minus_delta__FDK = FL2FXCONST_DBL(0.0f); else if (delta__FDK == FL2FXCONST_DBL(0.0f)) one_minus_delta__FDK = /*FL2FXCONST_DBL(1.0f)*/ (FIXP_DBL)MAXVAL_DBL; else one_minus_delta__FDK = (FL2FXCONST_DBL(0.5f) - (delta__FDK >> 1)) << 1; for (pb = 0; pb < self->numParameterBands; pb++) { int smoothBand; smoothBand = getSmoothOnOff(self, ps, pb); if (smoothBand && (pb >= residualBands)) { for (row = 0; row < self->numM2rows; row++) { for (col = 0; col < self->numVChannels; col++) { self->M2Real__FDK[row][col][pb] = ((fMultDiv2(delta__FDK, self->M2Real__FDK[row][col][pb]) + fMultDiv2(one_minus_delta__FDK, self->M2RealPrev__FDK[row][col][pb])) << 1); if (0 || (self->phaseCoding == 3)) { self->M2Imag__FDK[row][col][pb] = ((fMultDiv2(delta__FDK, self->M2Imag__FDK[row][col][pb]) + fMultDiv2(one_minus_delta__FDK, self->M2ImagPrev__FDK[row][col][pb])) << 1); } } } } } self->smoothState->prevParamSlot = frame->paramSlot[ps]; } /* init states */ void initParameterSmoothing(spatialDec *self) { self->smoothState->prevParamSlot = 0; } void SpatialDecSmoothOPD(spatialDec *self, const SPATIAL_BS_FRAME *frame, int ps) { int pb; int dSlots; FIXP_DBL delta__FDK; FIXP_DBL one_minus_delta__FDK; FIXP_DBL *phaseLeftSmooth__FDK = self->smoothState->opdLeftState__FDK; FIXP_DBL *phaseRightSmooth__FDK = self->smoothState->opdRightState__FDK; int quantCoarse; quantCoarse = frame->IPDLosslessData[0].bsQuantCoarseXXX[ps]; if (frame->OpdSmoothingMode == 0) { FDKmemcpy(phaseLeftSmooth__FDK, self->PhaseLeft__FDK, self->numParameterBands * sizeof(FIXP_DBL)); FDKmemcpy(phaseRightSmooth__FDK, self->PhaseRight__FDK, self->numParameterBands * sizeof(FIXP_DBL)); } else { if (ps == 0) { dSlots = frame->paramSlot[ps] + 1; } else { dSlots = frame->paramSlot[ps] - frame->paramSlot[ps - 1]; } delta__FDK = (FIXP_DBL)((INT)(FL2FXCONST_DBL(0.0078125f)) * dSlots); if (delta__FDK == (FIXP_DBL)MAXVAL_DBL /*FL2FXCONST_DBL(1.0f)*/) one_minus_delta__FDK = FL2FXCONST_DBL(0.0f); else if (delta__FDK == FL2FXCONST_DBL(0.0f)) one_minus_delta__FDK = (FIXP_DBL)MAXVAL_DBL /*FL2FXCONST_DBL(1.0f)*/; else one_minus_delta__FDK = (FL2FXCONST_DBL(0.5f) - (delta__FDK >> 1)) << 1; for (pb = 0; pb < self->numParameterBands; pb++) { FIXP_DBL tmpL, tmpR, tmp; tmpL = self->PhaseLeft__FDK[pb]; tmpR = self->PhaseRight__FDK[pb]; while (tmpL > phaseLeftSmooth__FDK[pb] + PI__IPD) tmpL -= PI__IPD << 1; while (tmpL < phaseLeftSmooth__FDK[pb] - PI__IPD) tmpL += PI__IPD << 1; while (tmpR > phaseRightSmooth__FDK[pb] + PI__IPD) tmpR -= PI__IPD << 1; while (tmpR < phaseRightSmooth__FDK[pb] - PI__IPD) tmpR += PI__IPD << 1; phaseLeftSmooth__FDK[pb] = fMult(delta__FDK, tmpL) + fMult(one_minus_delta__FDK, phaseLeftSmooth__FDK[pb]); phaseRightSmooth__FDK[pb] = fMult(delta__FDK, tmpR) + fMult(one_minus_delta__FDK, phaseRightSmooth__FDK[pb]); tmp = (((tmpL >> 1) - (tmpR >> 1)) - ((phaseLeftSmooth__FDK[pb] >> 1) - (phaseRightSmooth__FDK[pb] >> 1))) << 1; while (tmp > PI__IPD) tmp -= PI__IPD << 1; while (tmp < -PI__IPD) tmp += PI__IPD << 1; if (fixp_abs(tmp) > fMult((quantCoarse ? FL2FXCONST_DBL(50.f / 180.f) : FL2FXCONST_DBL(25.f / 180.f)), PI__IPD)) { phaseLeftSmooth__FDK[pb] = tmpL; phaseRightSmooth__FDK[pb] = tmpR; } while (phaseLeftSmooth__FDK[pb] > PI__IPD << 1) phaseLeftSmooth__FDK[pb] -= PI__IPD << 1; while (phaseLeftSmooth__FDK[pb] < (FIXP_DBL)0) phaseLeftSmooth__FDK[pb] += PI__IPD << 1; while (phaseRightSmooth__FDK[pb] > PI__IPD << 1) phaseRightSmooth__FDK[pb] -= PI__IPD << 1; while (phaseRightSmooth__FDK[pb] < (FIXP_DBL)0) phaseRightSmooth__FDK[pb] += PI__IPD << 1; self->PhaseLeft__FDK[pb] = phaseLeftSmooth__FDK[pb]; self->PhaseRight__FDK[pb] = phaseRightSmooth__FDK[pb]; } } return; } fdk-aac-2.0.2/libSACdec/src/sac_smoothing.h000066400000000000000000000120771404225310700203540ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround decoder library ************************* Author(s): Description: SAC Dec parameter smoothing *******************************************************************************/ #ifndef SAC_SMOOTHING_H #define SAC_SMOOTHING_H #include "sac_dec.h" void initParameterSmoothing(spatialDec *self); void SpatialDecSmoothM1andM2(spatialDec *self, const SPATIAL_BS_FRAME *frame, int ps); void SpatialDecSmoothOPD(spatialDec *self, const SPATIAL_BS_FRAME *frame, int ps); #endif fdk-aac-2.0.2/libSACdec/src/sac_stp.cpp000066400000000000000000000533621404225310700175100ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround decoder library ************************* Author(s): Description: SAC Dec subband processing *******************************************************************************/ #include "sac_stp.h" #include "sac_calcM1andM2.h" #include "sac_bitdec.h" #include "FDK_matrixCalloc.h" #include "sac_rom.h" #define SF_FREQ_DOMAIN_HEADROOM (2 * (1)) #define BP_GF_START 6 #define BP_GF_SIZE 25 #define HP_SIZE 9 #define STP_UPDATE_ENERGY_RATE 32 #define SF_WET 5 #define SF_DRY \ 3 /* SF_DRY == 2 would produce good conformance test results as well */ #define SF_DRY_NRG \ (4 - 1) /* 8.495f = sum(BP_GF__FDK[i]) \ i=0,..,(sizeof(BP_GF__FDK)/sizeof(FIXP_CFG)-1) => energy \ calculation needs 4 bits headroom, headroom can be reduced by 1 \ bit due to fPow2Div2() usage */ #define SF_WET_NRG \ (4 - 1) /* 8.495f = sum(BP_GF__FDK[i]) \ i=0,..,(sizeof(BP_GF__FDK)/sizeof(FIXP_CFG)-1) => energy \ calculation needs 4 bits headroom, headroom can be reduced by 1 \ bit due to fPow2Div2() usage */ #define SF_PRODUCT_BP_GF 13 #define SF_PRODUCT_BP_GF_GF 26 #define SF_SCALE 2 #define SF_SCALE_LD64 FL2FXCONST_DBL(0.03125) /* LD64((1<= scale2 >= 1/STP_SCALE_LIMIT => STP_SCALE_LIMIT >= (0.1 + 0.9 * scale) >= 1/STP_SCALE_LIMIT => (STP_SCALE_LIMIT-0.1)/0.9 >= scale >= (1/STP_SCALE_LIMIT-0.1)/0.9 3. Limiting of scale factor before sqrt calculation ((STP_SCALE_LIMIT-0.1)/0.9)^2 >= (scale^2) >= ((1/STP_SCALE_LIMIT-0.1)/0.9)^2 (STP_SCALE_LIMIT_HI)^2 >= (scale^2) >= (STP_SCALE_LIMIT_LO)^2 4. Thresholds for limiting of scale factor STP_SCALE_LIMIT_HI = ((2.82-0.1)/0.9) STP_SCALE_LIMIT_LO = (((1.0/2.82)-0.1)/0.9) STP_SCALE_LIMIT_HI_LD64 = LD64(STP_SCALE_LIMIT_HI*STP_SCALE_LIMIT_HI) STP_SCALE_LIMIT_LO_LD64 = LD64(STP_SCALE_LIMIT_LO*STP_SCALE_LIMIT_LO) */ #define CALC_WET_SCALE(dryIdx, wetIdx) \ if ((DryEnerLD64[dryIdx] - STP_SCALE_LIMIT_HI_LD64) > WetEnerLD64[wetIdx]) { \ scale[wetIdx] = STP_SCALE_LIMIT_HI; \ } else if (DryEnerLD64[dryIdx] < \ (WetEnerLD64[wetIdx] - STP_SCALE_LIMIT_LO_LD64)) { \ scale[wetIdx] = STP_SCALE_LIMIT_LO; \ } else { \ tmp = ((DryEnerLD64[dryIdx] - WetEnerLD64[wetIdx]) >> 1) - SF_SCALE_LD64; \ scale[wetIdx] = CalcInvLdData(tmp); \ } struct STP_DEC { FIXP_DBL runDryEner[MAX_INPUT_CHANNELS]; FIXP_DBL runWetEner[MAX_OUTPUT_CHANNELS]; FIXP_DBL oldDryEnerLD64[MAX_INPUT_CHANNELS]; FIXP_DBL oldWetEnerLD64[MAX_OUTPUT_CHANNELS]; FIXP_DBL prev_tp_scale[MAX_OUTPUT_CHANNELS]; const FIXP_CFG *BP; const FIXP_CFG *BP_GF; int update_old_ener; }; inline void combineSignalCplx(FIXP_DBL *hybOutputRealDry, FIXP_DBL *hybOutputImagDry, FIXP_DBL *hybOutputRealWet, FIXP_DBL *hybOutputImagWet, int bands) { int n; for (n = bands - 1; n >= 0; n--) { *hybOutputRealDry = fAddSaturate(*hybOutputRealDry, *hybOutputRealWet); *hybOutputImagDry = fAddSaturate(*hybOutputImagDry, *hybOutputImagWet); hybOutputRealDry++, hybOutputRealWet++; hybOutputImagDry++, hybOutputImagWet++; } } inline void combineSignalCplxScale1(FIXP_DBL *hybOutputRealDry, FIXP_DBL *hybOutputImagDry, FIXP_DBL *hybOutputRealWet, FIXP_DBL *hybOutputImagWet, const FIXP_CFG *pBP, FIXP_DBL scaleX, int bands) { int n; FIXP_DBL scaleY; for (n = bands - 1; n >= 0; n--) { scaleY = fMultDiv2(scaleX, *pBP); *hybOutputRealDry = SATURATE_LEFT_SHIFT( (*hybOutputRealDry >> 1) + (fMultDiv2(*hybOutputRealWet, scaleY) << (SF_SCALE + 1)), 1, DFRACT_BITS); *hybOutputImagDry = SATURATE_LEFT_SHIFT( (*hybOutputImagDry >> 1) + (fMultDiv2(*hybOutputImagWet, scaleY) << (SF_SCALE + 1)), 1, DFRACT_BITS); hybOutputRealDry++, hybOutputRealWet++; hybOutputImagDry++, hybOutputImagWet++; pBP++; } } inline void combineSignalCplxScale2(FIXP_DBL *hybOutputRealDry, FIXP_DBL *hybOutputImagDry, FIXP_DBL *hybOutputRealWet, FIXP_DBL *hybOutputImagWet, FIXP_DBL scaleX, int bands) { int n; for (n = bands - 1; n >= 0; n--) { *hybOutputRealDry = *hybOutputRealDry + (fMultDiv2(*hybOutputRealWet, scaleX) << (SF_SCALE + 1)); *hybOutputImagDry = *hybOutputImagDry + (fMultDiv2(*hybOutputImagWet, scaleX) << (SF_SCALE + 1)); hybOutputRealDry++, hybOutputRealWet++; hybOutputImagDry++, hybOutputImagWet++; } } /******************************************************************************* Functionname: subbandTPCreate ******************************************************************************/ SACDEC_ERROR subbandTPCreate(HANDLE_STP_DEC *hStpDec) { HANDLE_STP_DEC self = NULL; FDK_ALLOCATE_MEMORY_1D(self, 1, struct STP_DEC) if (hStpDec != NULL) { *hStpDec = self; } return MPS_OK; bail: return MPS_OUTOFMEMORY; } SACDEC_ERROR subbandTPInit(HANDLE_STP_DEC self) { SACDEC_ERROR err = MPS_OK; int ch; for (ch = 0; ch < MAX_OUTPUT_CHANNELS; ch++) { self->prev_tp_scale[ch] = FL2FXCONST_DBL(1.0f / (1 << SF_SCALE)); self->oldWetEnerLD64[ch] = FL2FXCONST_DBL(0.0); } for (ch = 0; ch < MAX_INPUT_CHANNELS; ch++) { self->oldDryEnerLD64[ch] = FL2FXCONST_DBL(0.0); } self->BP = BP__FDK; self->BP_GF = BP_GF__FDK; self->update_old_ener = 0; return err; } /******************************************************************************* Functionname: subbandTPDestroy ******************************************************************************/ void subbandTPDestroy(HANDLE_STP_DEC *hStpDec) { if (hStpDec != NULL) { FDK_FREE_MEMORY_1D(*hStpDec); } } /******************************************************************************* Functionname: subbandTPApply ******************************************************************************/ SACDEC_ERROR subbandTPApply(spatialDec *self, const SPATIAL_BS_FRAME *frame) { FIXP_DBL *qmfOutputRealDry[MAX_OUTPUT_CHANNELS]; FIXP_DBL *qmfOutputImagDry[MAX_OUTPUT_CHANNELS]; FIXP_DBL *qmfOutputRealWet[MAX_OUTPUT_CHANNELS]; FIXP_DBL *qmfOutputImagWet[MAX_OUTPUT_CHANNELS]; FIXP_DBL DryEner[MAX_INPUT_CHANNELS]; FIXP_DBL scale[MAX_OUTPUT_CHANNELS]; FIXP_DBL DryEnerLD64[MAX_INPUT_CHANNELS]; FIXP_DBL WetEnerLD64[MAX_OUTPUT_CHANNELS]; FIXP_DBL DryEner0 = FL2FXCONST_DBL(0.0f); FIXP_DBL WetEnerX, damp, tmp; FIXP_DBL dmxReal0, dmxImag0; int skipChannels[MAX_OUTPUT_CHANNELS]; int n, ch, cplxBands, cplxHybBands; int dry_scale_dmx, wet_scale_dmx; int i_LF, i_RF; HANDLE_STP_DEC hStpDec; const FIXP_CFG *pBP; int nrgScale = (2 * self->clipProtectGainSF__FDK); hStpDec = self->hStpDec; /* set scalefactor and loop counter */ FDK_ASSERT(SF_DRY >= 1); { cplxBands = BP_GF_SIZE; cplxHybBands = self->hybridBands; if (self->treeConfig == TREE_212) { dry_scale_dmx = 2; /* 2 bits to compensate fMultDiv2() and fPow2Div2() used in energy calculation */ } else { dry_scale_dmx = (2 * SF_DRY) - 2; } wet_scale_dmx = 2; } /* setup pointer for forming the direct downmix signal */ for (ch = 0; ch < self->numOutputChannels; ch++) { qmfOutputRealDry[ch] = &self->hybOutputRealDry__FDK[ch][7]; qmfOutputRealWet[ch] = &self->hybOutputRealWet__FDK[ch][7]; qmfOutputImagDry[ch] = &self->hybOutputImagDry__FDK[ch][7]; qmfOutputImagWet[ch] = &self->hybOutputImagWet__FDK[ch][7]; } /* clear skipping flag for all output channels */ FDKmemset(skipChannels, 0, self->numOutputChannels * sizeof(int)); /* set scale values to zero */ FDKmemset(scale, 0, self->numOutputChannels * sizeof(FIXP_DBL)); /* update normalisation energy with latest smoothed energy */ if (hStpDec->update_old_ener == STP_UPDATE_ENERGY_RATE) { hStpDec->update_old_ener = 1; for (ch = 0; ch < self->numInputChannels; ch++) { hStpDec->oldDryEnerLD64[ch] = CalcLdData(hStpDec->runDryEner[ch] + ABS_THR__FDK); } for (ch = 0; ch < self->numOutputChannels; ch++) { if (self->treeConfig == TREE_212) hStpDec->oldWetEnerLD64[ch] = CalcLdData(hStpDec->runWetEner[ch] + ABS_THR__FDK); else hStpDec->oldWetEnerLD64[ch] = CalcLdData(hStpDec->runWetEner[ch] + ABS_THR2__FDK); } } else { hStpDec->update_old_ener++; } /* get channel configuration */ switch (self->treeConfig) { case TREE_212: i_LF = 0; i_RF = 1; break; default: return MPS_WRONG_TREECONFIG; } /* form the 'direct' downmix signal */ pBP = hStpDec->BP_GF - BP_GF_START; switch (self->treeConfig) { case TREE_212: INT sMin, sNorm, sReal, sImag; sReal = fMin(getScalefactor(&qmfOutputRealDry[i_LF][BP_GF_START], cplxBands - BP_GF_START), getScalefactor(&qmfOutputRealDry[i_RF][BP_GF_START], cplxBands - BP_GF_START)); sImag = fMin(getScalefactor(&qmfOutputImagDry[i_LF][BP_GF_START], cplxBands - BP_GF_START), getScalefactor(&qmfOutputImagDry[i_RF][BP_GF_START], cplxBands - BP_GF_START)); sMin = fMin(sReal, sImag) - 1; for (n = BP_GF_START; n < cplxBands; n++) { dmxReal0 = scaleValue(qmfOutputRealDry[i_LF][n], sMin) + scaleValue(qmfOutputRealDry[i_RF][n], sMin); dmxImag0 = scaleValue(qmfOutputImagDry[i_LF][n], sMin) + scaleValue(qmfOutputImagDry[i_RF][n], sMin); DryEner0 += (fMultDiv2(fPow2Div2(dmxReal0), pBP[n]) + fMultDiv2(fPow2Div2(dmxImag0), pBP[n])) >> SF_DRY_NRG; } sNorm = SF_FREQ_DOMAIN_HEADROOM + SF_DRY_NRG + dry_scale_dmx - (2 * sMin) + nrgScale; DryEner0 = scaleValueSaturate( DryEner0, fMax(fMin(sNorm, DFRACT_BITS - 1), -(DFRACT_BITS - 1))); break; default:; } DryEner[0] = DryEner0; /* normalise the 'direct' signals */ for (ch = 0; ch < self->numInputChannels; ch++) { if (self->treeConfig != TREE_212) DryEner[ch] = DryEner[ch] << nrgScale; hStpDec->runDryEner[ch] = fMult(STP_LPF_COEFF1__FDK, hStpDec->runDryEner[ch]) + fMult(ONE_MINUS_STP_LPF_COEFF1__FDK, DryEner[ch]); if (DryEner[ch] != FL2FXCONST_DBL(0.0f)) { DryEnerLD64[ch] = fixMax((CalcLdData(DryEner[ch]) - hStpDec->oldDryEnerLD64[ch]), FL2FXCONST_DBL(-0.484375f)); } else { DryEnerLD64[ch] = FL2FXCONST_DBL(-0.484375f); } } for (; ch < MAX_INPUT_CHANNELS; ch++) { DryEnerLD64[ch] = FL2FXCONST_DBL(-0.484375f); } /* normalise the 'diffuse' signals */ pBP = hStpDec->BP_GF - BP_GF_START; for (ch = 0; ch < self->numOutputChannels; ch++) { if (skipChannels[ch]) { continue; } WetEnerX = FL2FXCONST_DBL(0.0f); if (self->treeConfig == TREE_212) { INT sMin, sNorm; sMin = fMin(getScalefactor(&qmfOutputRealWet[ch][BP_GF_START], cplxBands - BP_GF_START), getScalefactor(&qmfOutputImagWet[ch][BP_GF_START], cplxBands - BP_GF_START)); for (n = BP_GF_START; n < cplxBands; n++) { WetEnerX += (fMultDiv2(fPow2Div2(scaleValue(qmfOutputRealWet[ch][n], sMin)), pBP[n]) + fMultDiv2(fPow2Div2(scaleValue(qmfOutputImagWet[ch][n], sMin)), pBP[n])) >> SF_WET_NRG; } sNorm = SF_FREQ_DOMAIN_HEADROOM + SF_WET_NRG + wet_scale_dmx - (2 * sMin) + nrgScale; WetEnerX = scaleValueSaturate( WetEnerX, fMax(fMin(sNorm, DFRACT_BITS - 1), -(DFRACT_BITS - 1))); } else FDK_ASSERT(self->treeConfig == TREE_212); hStpDec->runWetEner[ch] = fMult(STP_LPF_COEFF1__FDK, hStpDec->runWetEner[ch]) + fMult(ONE_MINUS_STP_LPF_COEFF1__FDK, WetEnerX); if (WetEnerX == FL2FXCONST_DBL(0.0f)) { WetEnerLD64[ch] = FL2FXCONST_DBL(-0.484375f); } else { WetEnerLD64[ch] = fixMax((CalcLdData(WetEnerX) - hStpDec->oldWetEnerLD64[ch]), FL2FXCONST_DBL(-0.484375f)); } } /* compute scale factor for the 'diffuse' signals */ switch (self->treeConfig) { case TREE_212: if (DryEner[0] != FL2FXCONST_DBL(0.0f)) { CALC_WET_SCALE(0, i_LF); CALC_WET_SCALE(0, i_RF); } break; default:; } damp = FL2FXCONST_DBL(0.1f / (1 << SF_SCALE)); for (ch = 0; ch < self->numOutputChannels; ch++) { /* damp the scaling factor */ scale[ch] = damp + fMult(FL2FXCONST_DBL(0.9f), scale[ch]); /* limiting the scale factor */ if (scale[ch] > STP_SCALE_LIMIT__FDK) { scale[ch] = STP_SCALE_LIMIT__FDK; } if (scale[ch] < ONE_DIV_STP_SCALE_LIMIT__FDK) { scale[ch] = ONE_DIV_STP_SCALE_LIMIT__FDK; } /* low pass filter the scaling factor */ scale[ch] = fMult(STP_LPF_COEFF2__FDK, scale[ch]) + fMult(ONE_MINUS_STP_LPF_COEFF2__FDK, hStpDec->prev_tp_scale[ch]); hStpDec->prev_tp_scale[ch] = scale[ch]; } /* combine 'direct' and scaled 'diffuse' signal */ FDK_ASSERT((HP_SIZE - 3 + 10 - 1) == PC_NUM_HYB_BANDS); const SCHAR *channlIndex = row2channelSTP[self->treeConfig]; for (ch = 0; ch < self->numOutputChannels; ch++) { int no_scaling; no_scaling = !frame->tempShapeEnableChannelSTP[channlIndex[ch]]; if (no_scaling) { combineSignalCplx( &self->hybOutputRealDry__FDK[ch][self->tp_hybBandBorder], &self->hybOutputImagDry__FDK[ch][self->tp_hybBandBorder], &self->hybOutputRealWet__FDK[ch][self->tp_hybBandBorder], &self->hybOutputImagWet__FDK[ch][self->tp_hybBandBorder], cplxHybBands - self->tp_hybBandBorder); } else { FIXP_DBL scaleX; scaleX = scale[ch]; pBP = hStpDec->BP - self->tp_hybBandBorder; /* Band[HP_SIZE-3+10-1] needs not to be processed in combineSignalCplxScale1(), because pB[HP_SIZE-3+10-1] would be 1.0 */ combineSignalCplxScale1( &self->hybOutputRealDry__FDK[ch][self->tp_hybBandBorder], &self->hybOutputImagDry__FDK[ch][self->tp_hybBandBorder], &self->hybOutputRealWet__FDK[ch][self->tp_hybBandBorder], &self->hybOutputImagWet__FDK[ch][self->tp_hybBandBorder], &pBP[self->tp_hybBandBorder], scaleX, (HP_SIZE - 3 + 10 - 1) - self->tp_hybBandBorder); { combineSignalCplxScale2( &self->hybOutputRealDry__FDK[ch][HP_SIZE - 3 + 10 - 1], &self->hybOutputImagDry__FDK[ch][HP_SIZE - 3 + 10 - 1], &self->hybOutputRealWet__FDK[ch][HP_SIZE - 3 + 10 - 1], &self->hybOutputImagWet__FDK[ch][HP_SIZE - 3 + 10 - 1], scaleX, cplxHybBands - (HP_SIZE - 3 + 10 - 1)); } } } return (SACDEC_ERROR)MPS_OK; ; } fdk-aac-2.0.2/libSACdec/src/sac_stp.h000066400000000000000000000120131404225310700171410ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround decoder library ************************* Author(s): Description: SAC Dec subband processing *******************************************************************************/ #ifndef SAC_STP_H #define SAC_STP_H #include "sac_dec.h" SACDEC_ERROR subbandTPCreate(HANDLE_STP_DEC *hStpDec); SACDEC_ERROR subbandTPInit(HANDLE_STP_DEC self); SACDEC_ERROR subbandTPApply(spatialDec *self, const SPATIAL_BS_FRAME *frame); void subbandTPDestroy(HANDLE_STP_DEC *hStpDec); #endif fdk-aac-2.0.2/libSACdec/src/sac_tsd.cpp000066400000000000000000000272071404225310700174730ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround decoder library ************************* Author(s): Matthias Hildenbrand Description: USAC MPS212 Transient Steering Decorrelator (TSD) *******************************************************************************/ #include "sac_tsd.h" #define TSD_START_BAND (7) #define SIZE_S (4) #define SIZE_C (5) /*** Tables ***/ RAM_ALIGN LNK_SECTION_CONSTDATA static const UCHAR nBitsTsdCW_32slots[32] = { 5, 9, 13, 16, 18, 20, 22, 24, 25, 26, 27, 28, 29, 29, 30, 30, 30, 29, 29, 28, 27, 26, 25, 24, 22, 20, 18, 16, 13, 9, 5, 0}; RAM_ALIGN LNK_SECTION_CONSTDATA static const UCHAR nBitsTsdCW_64slots[64] = { 6, 11, 16, 20, 23, 27, 30, 33, 35, 38, 40, 42, 44, 46, 48, 49, 51, 52, 53, 55, 56, 57, 58, 58, 59, 60, 60, 60, 61, 61, 61, 61, 61, 61, 61, 60, 60, 60, 59, 58, 58, 57, 56, 55, 53, 52, 51, 49, 48, 46, 44, 42, 40, 38, 35, 33, 30, 27, 23, 20, 16, 11, 6, 0}; RAM_ALIGN LNK_SECTION_CONSTDATA static const FIXP_DPK phiTsd[8] = { {{(FIXP_DBL)0x7fffffff, (FIXP_DBL)0x00000000}}, {{(FIXP_DBL)0x5a82799a, (FIXP_DBL)0x5a82799a}}, {{(FIXP_DBL)0x00000000, (FIXP_DBL)0x7fffffff}}, {{(FIXP_DBL)0xa57d8666, (FIXP_DBL)0x5a82799a}}, {{(FIXP_DBL)0x80000000, (FIXP_DBL)0x00000000}}, {{(FIXP_DBL)0xa57d8666, (FIXP_DBL)0xa57d8666}}, {{(FIXP_DBL)0x00000000, (FIXP_DBL)0x80000000}}, {{(FIXP_DBL)0x5a82799a, (FIXP_DBL)0xa57d8666}}}; /*** Static Functions ***/ static void longmult1(USHORT a[], USHORT b, USHORT d[], int len) { int k; ULONG tmp; ULONG b0 = (ULONG)b; tmp = ((ULONG)a[0]) * b0; d[0] = (USHORT)tmp; for (k = 1; k < len; k++) { tmp = (tmp >> 16) + ((ULONG)a[k]) * b0; d[k] = (USHORT)tmp; } } static void longdiv(USHORT b[], USHORT a, USHORT d[], USHORT *pr, int len) { ULONG r; ULONG tmp; int k; FDK_ASSERT(a != 0); r = 0; for (k = len - 1; k >= 0; k--) { tmp = ((ULONG)b[k]) + (r << 16); if (tmp) { d[k] = (USHORT)(tmp / a); r = tmp - d[k] * a; } else { d[k] = 0; } } *pr = (USHORT)r; } static void longsub(USHORT a[], USHORT b[], int lena, int lenb) { int h; LONG carry = 0; FDK_ASSERT(lena >= lenb); for (h = 0; h < lenb; h++) { carry += ((LONG)a[h]) - ((LONG)b[h]); a[h] = (USHORT)carry; carry = carry >> 16; } for (; h < lena; h++) { carry = ((LONG)a[h]) + carry; a[h] = (USHORT)carry; carry = carry >> 16; } FDK_ASSERT(carry == 0); /* carry != 0 indicates subtraction underflow, e.g. b > a */ return; } static int longcompare(USHORT a[], USHORT b[], int len) { int i; for (i = len - 1; i > 0; i--) { if (a[i] != b[i]) break; } return (a[i] >= b[i]) ? 1 : 0; } FDK_INLINE int isTrSlot(const TSD_DATA *pTsdData, const int ts) { return (pTsdData->bsTsdTrPhaseData[ts] >= 0); } /*** Public Functions ***/ int TsdRead(HANDLE_FDK_BITSTREAM hBs, const int numSlots, TSD_DATA *pTsdData) { int nBitsTrSlots = 0; int bsTsdNumTrSlots; const UCHAR *nBitsTsdCW_tab = NULL; switch (numSlots) { case 32: nBitsTrSlots = 4; nBitsTsdCW_tab = nBitsTsdCW_32slots; break; case 64: nBitsTrSlots = 5; nBitsTsdCW_tab = nBitsTsdCW_64slots; break; default: return 1; } /*** Read TempShapeData for bsTempShapeConfig == 3 ***/ pTsdData->bsTsdEnable = FDKreadBit(hBs); if (!pTsdData->bsTsdEnable) { return 0; } /*** Parse/Decode TsdData() ***/ pTsdData->numSlots = numSlots; bsTsdNumTrSlots = FDKreadBits(hBs, nBitsTrSlots); /* Decode transient slot positions */ { int nBitsTsdCW = (int)nBitsTsdCW_tab[bsTsdNumTrSlots]; SCHAR *phaseData = pTsdData->bsTsdTrPhaseData; int p = bsTsdNumTrSlots + 1; int k, h; USHORT s[SIZE_S] = {0}; USHORT c[SIZE_C] = {0}; USHORT r[1]; /* Init with TsdSepData[k] = 0 */ for (k = 0; k < numSlots; k++) { phaseData[k] = -1; /* means TsdSepData[] = 0 */ } for (h = (SIZE_S - 1); h >= 0; h--) { if (nBitsTsdCW > h * 16) { s[h] = (USHORT)FDKreadBits(hBs, nBitsTsdCW - h * 16); nBitsTsdCW = h * 16; } } /* c = prod_{h=1}^{p} (k-p+h)/h */ k = numSlots - 1; c[0] = k - p + 1; for (h = 2; h <= p; h++) { longmult1(c, (k - p + h), c, 5); /* c *= k - p + h; */ longdiv(c, h, c, r, 5); /* c /= h; */ FDK_ASSERT(*r == 0); } /* go through all slots */ for (; k >= 0; k--) { if (p > k) { for (; k >= 0; k--) { phaseData[k] = 1; /* means TsdSepData[] = 1 */ } break; } if (longcompare(s, c, 4)) { /* (s >= c) */ longsub(s, c, 4, 4); /* s -= c; */ phaseData[k] = 1; /* means TsdSepData[] = 1 */ if (p == 1) { break; } /* Update c for next iteration: c_new = c_old * p / k */ longmult1(c, p, c, 5); p--; } else { /* Update c for next iteration: c_new = c_old * (k-p) / k */ longmult1(c, (k - p), c, 5); } longdiv(c, k, c, r, 5); FDK_ASSERT(*r == 0); } /* Read phase data */ for (k = 0; k < numSlots; k++) { if (phaseData[k] == 1) { phaseData[k] = FDKreadBits(hBs, 3); } } } return 0; } void TsdGenerateNonTr(const int numHybridBands, const TSD_DATA *pTsdData, const int ts, FIXP_DBL *pVdirectReal, FIXP_DBL *pVdirectImag, FIXP_DBL *pVnonTrReal, FIXP_DBL *pVnonTrImag, FIXP_DBL **ppDecorrInReal, FIXP_DBL **ppDecorrInImag) { int k = 0; if (!isTrSlot(pTsdData, ts)) { /* Let allpass based decorrelator read from direct input. */ *ppDecorrInReal = pVdirectReal; *ppDecorrInImag = pVdirectImag; return; } /* Generate nonTr input signal for allpass based decorrelator */ for (; k < TSD_START_BAND; k++) { pVnonTrReal[k] = pVdirectReal[k]; pVnonTrImag[k] = pVdirectImag[k]; } for (; k < numHybridBands; k++) { pVnonTrReal[k] = (FIXP_DBL)0; pVnonTrImag[k] = (FIXP_DBL)0; } *ppDecorrInReal = pVnonTrReal; *ppDecorrInImag = pVnonTrImag; } void TsdApply(const int numHybridBands, const TSD_DATA *pTsdData, int *pTsdTs, const FIXP_DBL *pVdirectReal, const FIXP_DBL *pVdirectImag, FIXP_DBL *pDnonTrReal, FIXP_DBL *pDnonTrImag) { const int ts = *pTsdTs; if (isTrSlot(pTsdData, ts)) { int k; const FIXP_DPK *phi = &phiTsd[pTsdData->bsTsdTrPhaseData[ts]]; FDK_ASSERT((pTsdData->bsTsdTrPhaseData[ts] >= 0) && (pTsdData->bsTsdTrPhaseData[ts] < 8)); /* d = d_nonTr + v_direct * exp(j * bsTsdTrPhaseData[ts]/4 * pi ) */ for (k = TSD_START_BAND; k < numHybridBands; k++) { FIXP_DBL tempReal, tempImag; cplxMultDiv2(&tempReal, &tempImag, pVdirectReal[k], pVdirectImag[k], *phi); pDnonTrReal[k] = SATURATE_LEFT_SHIFT( (pDnonTrReal[k] >> 2) + (tempReal >> 1), 2, DFRACT_BITS); pDnonTrImag[k] = SATURATE_LEFT_SHIFT( (pDnonTrImag[k] >> 2) + (tempImag >> 1), 2, DFRACT_BITS); } } /* The modulo MAX_TSD_TIME_SLOTS operation is to avoid illegal memory accesses * in case of errors. */ *pTsdTs = (ts + 1) & (MAX_TSD_TIME_SLOTS - 1); return; } fdk-aac-2.0.2/libSACdec/src/sac_tsd.h000066400000000000000000000164231404225310700171360ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround decoder library ************************* Author(s): Matthias Hildenbrand Description: USAC MPS212 Transient Steering Decorrelator (TSD) *******************************************************************************/ #ifndef SAC_TSD_H #define SAC_TSD_H #include "FDK_bitstream.h" #include "common_fix.h" #define MAX_TSD_TIME_SLOTS (64) /** Structure which holds the data needed to apply TSD to current frame. */ typedef struct { UCHAR bsTsdEnable; /**< for current frame TSD is (0:disabled, 1:enabled) */ UCHAR numSlots; /**< total number of QMF slots per frame */ SCHAR bsTsdTrPhaseData[MAX_TSD_TIME_SLOTS]; /**< -1 => TsdSepData[ts]=0; 0-7: values of bsTsdTrPhaseData[ts] and TsdSepData[ts]=1 */ } TSD_DATA; FDK_INLINE int isTsdActive(const TSD_DATA *pTsdData) { return (int)pTsdData->bsTsdEnable; } /** * \brief Parse and Decode TSD data. * \param[in] hBs bitstream handle to read data from. * \param[in] numSlots number of QMF slots per frame. * \param[out] pTsdData pointer to TSD data structure. * \return 0 on succes, 1 on error. */ int TsdRead(HANDLE_FDK_BITSTREAM hBs, const int numSlots, TSD_DATA *pTsdData); /** * \brief Perform transient seperation (v_{x,nonTr} signal). * \param[in] numHybridBands number of hybrid bands. * \param[in] pTsdData pointer to TSD data structure. * \param[in] pVdirectReal pointer to array with direct signal. * \param[in] pVdirectImag pointer to array with direct signal. * \param[out] pVnonTrReal pointer to array with nonTr signal. * \param[out] pVnonTrImag pointer to array with nonTr signal. * \param[out] ppDecorrInReal handle to array where allpass based decorrelator * should read from (modified by this function). * \param[out] ppDecorrInImag handle to array where allpass based decorrelator * should read from (modified by this function). */ void TsdGenerateNonTr(const int numHybridBands, const TSD_DATA *pTsdData, const int ts, FIXP_DBL *pVdirectReal, FIXP_DBL *pVdirectImag, FIXP_DBL *pVnonTrReal, FIXP_DBL *pVnonTrImag, FIXP_DBL **ppDecorrInReal, FIXP_DBL **ppDecorrInImag); /** * \brief Generate d_{x,Tr} signal and add to d_{x,nonTr} signal * \param[in] numHybridBands * \param[in,out] pTsdData * \param pTsdTs pointer to persistent time slot counter * \param[in] pVdirectReal * \param[in] pVdirectImag * \param[out] pDnonTrReal * \param[out] pDnonTrImag */ void TsdApply(const int numHybridBands, const TSD_DATA *pTsdData, int *pTsdTs, const FIXP_DBL *pVdirectReal, const FIXP_DBL *pVdirectImag, FIXP_DBL *pDnonTrReal, FIXP_DBL *pDnonTrImag); #endif /* #ifndef SAC_TSD_H */ fdk-aac-2.0.2/libSACenc/000077500000000000000000000000001404225310700145625ustar00rootroot00000000000000fdk-aac-2.0.2/libSACenc/include/000077500000000000000000000000001404225310700162055ustar00rootroot00000000000000fdk-aac-2.0.2/libSACenc/include/sacenc_lib.h000066400000000000000000000352261404225310700204500ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround encoder library ************************* Author(s): Max Neuendorf Description: Encoder Library Interface Encoder API *******************************************************************************/ /**************************************************************************/ /** \file ******************************************************************************/ #ifndef SACENC_LIB_H #define SACENC_LIB_H /* Includes ******************************************************************/ #include "machine_type.h" #include "FDK_audio.h" #ifdef __cplusplus extern "C" { #endif /* Defines *******************************************************************/ /* Data Types ****************************************************************/ /** * Space encoder error codes. */ typedef enum { SACENC_OK = 0x00000000, /*!< No error happened. All fine. */ SACENC_INVALID_HANDLE = 0x00000080, /*!< Handle passed to function call was invalid. */ SACENC_MEMORY_ERROR = 0x00000800, /*!< Memory allocation failed. */ SACENC_INIT_ERROR = 0x00008000, /*!< General initialization error. */ SACENC_ENCODE_ERROR = 0x00080000, /*!< The encoding process was interrupted by an unexpected error. */ SACENC_PARAM_ERROR = 0x00800000, /*!< Invalid runtime parameter. */ SACENC_UNSUPPORTED_PARAMETER = 0x00800001, /*!< Parameter not available. */ SACENC_INVALID_CONFIG = 0x00800002, /*!< Configuration not provided. */ SACENC_UNKNOWN_ERROR = 0x08000000 /*!< Unknown error. */ } FDK_SACENC_ERROR; typedef enum { SACENC_INVALID_MODE = 0, SACENC_212 = 8, SACENC_ESCAPE = 15 } MP4SPACEENC_MODE; typedef enum { SACENC_BANDS_INVALID = 0, SACENC_BANDS_4 = 4, SACENC_BANDS_5 = 5, SACENC_BANDS_7 = 7, SACENC_BANDS_9 = 9, SACENC_BANDS_12 = 12, SACENC_BANDS_15 = 15, SACENC_BANDS_23 = 23 } MP4SPACEENC_BANDS_CONFIG; typedef enum { SACENC_QUANTMODE_INVALID = -1, SACENC_QUANTMODE_FINE = 0, SACENC_QUANTMODE_EBQ1 = 1, SACENC_QUANTMODE_EBQ2 = 2, SACENC_QUANTMODE_RSVD3 = 3 } MP4SPACEENC_QUANTMODE; typedef enum { SACENC_DMXGAIN_INVALID = -1, SACENC_DMXGAIN_0_dB = 0, SACENC_DMXGAIN_1_5_dB = 1, SACENC_DMXGAIN_3_dB = 2, SACENC_DMXGAIN_4_5_dB = 3, SACENC_DMXGAIN_6_dB = 4, SACENC_DMXGAIN_7_5_dB = 5, SACENC_DMXGAIN_9_dB = 6, SACENC_DMXGAIN_12_dB = 7 } MP4SPACEENC_DMX_GAIN; /** * \brief Space Encoder setting parameters. * * Use FDK_sacenc_setParam() function to configure the internal status of the * following parameters. */ typedef enum { SACENC_LOWDELAY, /*!< Configure lowdelay MPEG Surround. - 0: Disable Lowdelay. (default) - 1: Enable Lowdelay. - 2: Enable Lowdelay including keep frame. */ SACENC_ENC_MODE, /*!< Configure encoder tree mode. See ::MP4SPACEENC_MODE for available values. */ SACENC_SAMPLERATE, /*!< Configure encoder sampling rate. */ SACENC_FRAME_TIME_SLOTS, /*!< Configure number of slots per spatial frame. */ SACENC_PARAM_BANDS, /*!< Configure number of parameter bands. See ::MP4SPACEENC_BANDS_CONFIG for available values. */ SACENC_TIME_DOM_DMX, /*!< Configure time domain downmix. - 0: No time domain downmix. (default) - 1: Static time domain downmix. - 2: Enhanced time domain downmix, stereo to mono only. */ SACENC_DMX_GAIN, /*!< Configure downmix gain. See ::MP4SPACEENC_DMX_GAIN for available values. */ SACENC_COARSE_QUANT, /*!< Use coarse parameter quantization. - 0: No (default) - 1: Yes */ SACENC_QUANT_MODE, /*!< Configure quanitzation mode. See ::MP4SPACEENC_QUANTMODE for available values. */ SACENC_TIME_ALIGNMENT, /*!< Configure time alignment in samples. */ SACENC_INDEPENDENCY_COUNT, /*!< Configure the independency count. (count == 0 means independencyFlag == 1) */ SACENC_INDEPENDENCY_FACTOR, /*!< How often should we set the independency flag */ SACENC_NONE /*!< ------ */ } SPACEENC_PARAM; /** * Describes Spatial Specific Config. */ typedef struct { INT nSscSizeBits; /*!< Number of valid bits in pSsc buffer. */ UCHAR *pSsc; /*!< SpatialSpecificConfig buffer in binary format. */ } MPEG4SPACEENC_SSCBUF; /** * Provides some info about the encoder configuration. */ typedef struct { INT nSampleRate; /*!< Configured sampling rate.*/ INT nSamplesFrame; /*!< Frame length in samples. */ INT nTotalInputChannels; /*!< Number of expected audio input channels. */ INT nDmxDelay; /*!< Delay of the downmixed signal. */ INT nCodecDelay; /*!< Delay of the whole en-/decoded signal, including core-coder delay. */ INT nDecoderDelay; /*!< Delay added by the MP4SPACE decoder. */ INT nPayloadDelay; /*!< Delay of the payload. */ INT nDiscardOutFrames; /*!< Number of dmx frames to discard for alignment with bitstream. */ MPEG4SPACEENC_SSCBUF *pSscBuf; /*!< Pointer to Spatial Specific Config structure. */ } MP4SPACEENC_INFO; /** * MPEG Surround encoder handle. */ typedef struct MP4SPACE_ENCODER *HANDLE_MP4SPACE_ENCODER; /** * Defines the input arguments for a FDK_sacenc_encode() call. */ typedef struct { INT nInputSamples; /*!< Number of valid input audio samples (multiple of input channels). */ UINT inputBufferSizePerChannel; /*!< Size of input buffer (input audio samples) per channel. */ UINT isInputInterleaved; /*!< Indicates if input audio samples are represented in blocks or interleaved: - 0 : in blocks. - 1 : interleaved. */ } SACENC_InArgs; /** * Defines the output arguments for a FDK_sacenc_encode() call. */ typedef struct { INT nOutputBits; /*!< Number of valid payload bits generated during FDK_sacenc_encode(). */ INT nOutputSamples; /*!< Number of valid output audio samples generated during FDK_sacenc_encode(). */ UINT nSamplesConsumed; /*!< Number of input audio samples consumed in FDK_sacenc_encode(). */ } SACENC_OutArgs; /* Constants *****************************************************************/ /* Function / Class Declarations *********************************************/ /** * \brief Opens a new instace of the MPEG Surround encoder. * * \param phMp4SpaceEnc Pointer to the encoder handle to be deallocated. * * \return * - SACENC_OK, on success. * - SACENC_INVALID_HANDLE, SACENC_MEMORY_ERROR, on failure. */ FDK_SACENC_ERROR FDK_sacenc_open(HANDLE_MP4SPACE_ENCODER *phMp4SpaceEnc); /** * \brief Finalizes opening process of MPEG Surround encoder. * * Shows, how many samples are needed as input * * \param hMp4SpaceEnc A valid MPEG Surround encoder handle. * \param dmxDelay Downmix delay. * * \return * - SACENC_OK, on success. * - SACENC_INVALID_HANDLE, SACENC_INIT_ERROR, SACENC_INVALID_CONFIG, * on failure. */ FDK_SACENC_ERROR FDK_sacenc_init(HANDLE_MP4SPACE_ENCODER hMp4SpaceEnc, const INT dmxDelay); /** * \brief Close the MPEG Surround encoder instance. * * Deallocate encoder instance and free whole memory. * * \param phMp4SpaceEnc Pointer to the encoder handle to be deallocated. * * \return * - SACENC_OK, on success. * - SACENC_INVALID_HANDLE, on failure. */ FDK_SACENC_ERROR FDK_sacenc_close(HANDLE_MP4SPACE_ENCODER *phMp4SpaceEnc); /** * \brief MPEG surround parameter extraction, framwise. * * \param hMp4SpaceEnc A valid MPEG Surround encoder handle. * * \return * - SACENC_OK, on success. * - SACENC_INVALID_HANDLE, on failure. */ FDK_SACENC_ERROR FDK_sacenc_encode(const HANDLE_MP4SPACE_ENCODER hMp4SpaceEnc, const FDK_bufDescr *inBufDesc, const FDK_bufDescr *outBufDesc, const SACENC_InArgs *inargs, SACENC_OutArgs *outargs); /** * \brief Provides information on produced bitstream. * * \param hMp4SpaceEnc A valid MPEG Surround encoder handle. * \param pInfo Pointer to an encoder info struct, filled on * return. * * \return * - SACENC_OK, on success. * - SACENC_INVALID_HANDLE, on failure. */ FDK_SACENC_ERROR FDK_sacenc_getInfo(const HANDLE_MP4SPACE_ENCODER hMp4SpaceEnc, MP4SPACEENC_INFO *const pInfo); /** * \brief Set one single MPEG Surround encoder parameter. * * This function allows configuration of all encoder parameters specified in * ::SPACEENC_PARAM. Each parameter must be set with a separate function call. * An internal validation of the configuration value range will be done. * * \param hMp4SpaceEnc A valid MPEG Surround encoder handle. * \param param Parameter to be set. See ::SPACEENC_PARAM. * \param value Parameter value. See parameter description in * ::SPACEENC_PARAM. * * \return * - SACENC_OK, on success. * - SACENC_INVALID_HANDLE, SACENC_UNSUPPORTED_PARAMETER, * SACENC_INVALID_CONFIG, on failure. */ FDK_SACENC_ERROR FDK_sacenc_setParam(HANDLE_MP4SPACE_ENCODER hMp4SpaceEnc, const SPACEENC_PARAM param, const UINT value); /** * \brief Get information about MPEG Surround encoder library build. * * Fill a given LIB_INFO structure with library version information. * * \param info Pointer to an allocated LIB_INFO struct. * * \return * - SACENC_OK, on success. * - SACENC_INVALID_HANDLE, SACENC_INIT_ERROR, on failure. */ FDK_SACENC_ERROR FDK_sacenc_getLibInfo(LIB_INFO *info); #ifdef __cplusplus } #endif #endif /* SACENC_LIB_H */ fdk-aac-2.0.2/libSACenc/src/000077500000000000000000000000001404225310700153515ustar00rootroot00000000000000fdk-aac-2.0.2/libSACenc/src/sacenc_bitstream.cpp000066400000000000000000000703121404225310700213660ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround encoder library ************************* Author(s): Description: Encoder Library Interface Bitstream Writer *******************************************************************************/ /* Includes ******************************************************************/ #include "sacenc_bitstream.h" #include "sacenc_const.h" #include "genericStds.h" #include "common_fix.h" #include "FDK_matrixCalloc.h" #include "sacenc_nlc_enc.h" /* Defines *******************************************************************/ #define MAX_FREQ_RES_INDEX 8 #define MAX_SAMPLING_FREQUENCY_INDEX 13 #define SAMPLING_FREQUENCY_INDEX_ESCAPE 15 /* Data Types ****************************************************************/ typedef struct { SCHAR cld_old[SACENC_MAX_NUM_BOXES][MAX_NUM_BINS]; SCHAR icc_old[SACENC_MAX_NUM_BOXES][MAX_NUM_BINS]; UCHAR quantCoarseCldPrev[SACENC_MAX_NUM_BOXES][MAX_NUM_PARAMS]; UCHAR quantCoarseIccPrev[SACENC_MAX_NUM_BOXES][MAX_NUM_PARAMS]; } PREV_OTTDATA; typedef struct { PREV_OTTDATA prevOttData; } STATIC_SPATIALFRAME; typedef struct BSF_INSTANCE { SPATIALSPECIFICCONFIG spatialSpecificConfig; SPATIALFRAME frame; STATIC_SPATIALFRAME prevFrameData; } BSF_INSTANCE; /* Constants *****************************************************************/ static const INT SampleRateTable[MAX_SAMPLING_FREQUENCY_INDEX] = { 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000, 7350}; static const UCHAR FreqResBinTable_LD[MAX_FREQ_RES_INDEX] = {0, 23, 15, 12, 9, 7, 5, 4}; static const UCHAR FreqResStrideTable_LD[] = {1, 2, 5, 23}; /* Function / Class Declarations *********************************************/ /* Function / Class Definition ***********************************************/ static FDK_SACENC_ERROR DuplicateLosslessData( const INT startBox, const INT stopBox, const LOSSLESSDATA *const hLosslessDataFrom, const INT setFrom, LOSSLESSDATA *const hLosslessDataTo, const INT setTo) { FDK_SACENC_ERROR error = SACENC_OK; if ((NULL == hLosslessDataFrom) || (NULL == hLosslessDataTo)) { error = SACENC_INVALID_HANDLE; } else { int i; for (i = startBox; i < stopBox; i++) { hLosslessDataTo->bsXXXDataMode[i][setTo] = hLosslessDataFrom->bsXXXDataMode[i][setFrom]; hLosslessDataTo->bsDataPair[i][setTo] = hLosslessDataFrom->bsDataPair[i][setFrom]; hLosslessDataTo->bsQuantCoarseXXX[i][setTo] = hLosslessDataFrom->bsQuantCoarseXXX[i][setFrom]; hLosslessDataTo->bsFreqResStrideXXX[i][setTo] = hLosslessDataFrom->bsFreqResStrideXXX[i][setFrom]; } } return error; } FDK_SACENC_ERROR fdk_sacenc_duplicateParameterSet( const SPATIALFRAME *const hFrom, const INT setFrom, SPATIALFRAME *const hTo, const INT setTo) { FDK_SACENC_ERROR error = SACENC_OK; if ((NULL == hFrom) || (NULL == hTo)) { error = SACENC_INVALID_HANDLE; } else { int box; /* Only Copy Parameter Set selective stuff */ /* OTT-Data */ for (box = 0; box < SACENC_MAX_NUM_BOXES; box++) { FDKmemcpy(hTo->ottData.cld[box][setTo], hFrom->ottData.cld[box][setFrom], sizeof(hFrom->ottData.cld[0][0])); FDKmemcpy(hTo->ottData.icc[box][setTo], hFrom->ottData.icc[box][setFrom], sizeof(hFrom->ottData.icc[0][0])); } /* LOSSLESSDATA */ DuplicateLosslessData(0, SACENC_MAX_NUM_BOXES, &hFrom->CLDLosslessData, setFrom, &hTo->CLDLosslessData, setTo); DuplicateLosslessData(0, SACENC_MAX_NUM_BOXES, &hFrom->ICCLosslessData, setFrom, &hTo->ICCLosslessData, setTo); } /* valid handle */ return error; } /* set frame defaults */ static void clearFrame(SPATIALFRAME *const pFrame) { FDKmemclear(pFrame, sizeof(SPATIALFRAME)); pFrame->bsIndependencyFlag = 1; pFrame->framingInfo.numParamSets = 1; } static void fine2coarse(SCHAR *const data, const DATA_TYPE dataType, const INT startBand, const INT numBands) { int i; if (dataType == t_CLD) { for (i = startBand; i < startBand + numBands; i++) { data[i] /= 2; } } else { for (i = startBand; i < startBand + numBands; i++) { data[i] >>= 1; } } } static void coarse2fine(SCHAR *const data, const DATA_TYPE dataType, const INT startBand, const INT numBands) { int i; for (i = startBand; i < startBand + numBands; i++) { data[i] <<= 1; } if (dataType == t_CLD) { for (i = startBand; i < startBand + numBands; i++) { if (data[i] == -14) { data[i] = -15; } else if (data[i] == 14) { data[i] = 15; } } } /* (dataType == t_CLD) */ } static UCHAR getBsFreqResStride(const INT index) { const UCHAR *pFreqResStrideTable = NULL; int freqResStrideTableSize = 0; pFreqResStrideTable = FreqResStrideTable_LD; freqResStrideTableSize = sizeof(FreqResStrideTable_LD) / sizeof(*FreqResStrideTable_LD); return (((NULL != pFreqResStrideTable) && (index >= 0) && (index < freqResStrideTableSize)) ? pFreqResStrideTable[index] : 1); } /* write data to bitstream */ static void ecData(HANDLE_FDK_BITSTREAM bitstream, SCHAR data[MAX_NUM_PARAMS][MAX_NUM_BINS], SCHAR oldData[MAX_NUM_BINS], UCHAR quantCoarseXXXprev[MAX_NUM_PARAMS], LOSSLESSDATA *const losslessData, const DATA_TYPE dataType, const INT paramIdx, const INT numParamSets, const INT independencyFlag, const INT startBand, const INT stopBand, const INT defaultValue) { int ps, pb, strOffset, pbStride, dataBands, i; int aStrides[MAX_NUM_BINS + 1] = {0}; SHORT cmpIdxData[2][MAX_NUM_BINS] = {{0}}; SHORT cmpOldData[MAX_NUM_BINS] = {0}; /* bsXXXDataMode */ if (independencyFlag || (losslessData->bsQuantCoarseXXX[paramIdx][0] != quantCoarseXXXprev[paramIdx])) { losslessData->bsXXXDataMode[paramIdx][0] = FINECOARSE; } else { losslessData->bsXXXDataMode[paramIdx][0] = KEEP; for (i = startBand; i < stopBand; i++) { if (data[0][i] != oldData[i]) { losslessData->bsXXXDataMode[paramIdx][0] = FINECOARSE; break; } } } FDKwriteBits(bitstream, losslessData->bsXXXDataMode[paramIdx][0], 2); for (ps = 1; ps < numParamSets; ps++) { if (losslessData->bsQuantCoarseXXX[paramIdx][ps] != losslessData->bsQuantCoarseXXX[paramIdx][ps - 1]) { losslessData->bsXXXDataMode[paramIdx][ps] = FINECOARSE; } else { losslessData->bsXXXDataMode[paramIdx][ps] = KEEP; for (i = startBand; i < stopBand; i++) { if (data[ps][i] != data[ps - 1][i]) { losslessData->bsXXXDataMode[paramIdx][ps] = FINECOARSE; break; } } } FDKwriteBits(bitstream, losslessData->bsXXXDataMode[paramIdx][ps], 2); } /* for ps */ /* Create data pairs if possible */ for (ps = 0; ps < (numParamSets - 1); ps++) { if (losslessData->bsXXXDataMode[paramIdx][ps] == FINECOARSE) { /* Check if next parameter set is FINCOARSE */ if (losslessData->bsXXXDataMode[paramIdx][ps + 1] == FINECOARSE) { /* We have to check if ps and ps+1 use the same bsXXXQuantMode */ /* and also have the same stride */ if ((losslessData->bsQuantCoarseXXX[paramIdx][ps + 1] == losslessData->bsQuantCoarseXXX[paramIdx][ps]) && (losslessData->bsFreqResStrideXXX[paramIdx][ps + 1] == losslessData->bsFreqResStrideXXX[paramIdx][ps])) { losslessData->bsDataPair[paramIdx][ps] = 1; losslessData->bsDataPair[paramIdx][ps + 1] = 1; /* We have a data pair -> Jump to the ps after next ps*/ ps++; continue; } } /* dataMode of next ps is not FINECOARSE or does not use the same * bsXXXQuantMode/stride */ /* -> no dataPair possible */ losslessData->bsDataPair[paramIdx][ps] = 0; /* Initialize ps after next ps to Zero (only important for the last * parameter set) */ losslessData->bsDataPair[paramIdx][ps + 1] = 0; } else { /* No FINECOARSE -> no data pair possible */ losslessData->bsDataPair[paramIdx][ps] = 0; /* Initialize ps after next ps to Zero (only important for the last * parameter set) */ losslessData->bsDataPair[paramIdx][ps + 1] = 0; } } /* for ps */ for (ps = 0; ps < numParamSets; ps++) { if (losslessData->bsXXXDataMode[paramIdx][ps] == DEFAULT) { /* Prepare old data */ for (i = startBand; i < stopBand; i++) { oldData[i] = defaultValue; } quantCoarseXXXprev[paramIdx] = 0; /* Default data are always fine */ } if (losslessData->bsXXXDataMode[paramIdx][ps] == FINECOARSE) { FDKwriteBits(bitstream, losslessData->bsDataPair[paramIdx][ps], 1); FDKwriteBits(bitstream, losslessData->bsQuantCoarseXXX[paramIdx][ps], 1); FDKwriteBits(bitstream, losslessData->bsFreqResStrideXXX[paramIdx][ps], 2); if (losslessData->bsQuantCoarseXXX[paramIdx][ps] != quantCoarseXXXprev[paramIdx]) { if (quantCoarseXXXprev[paramIdx]) { coarse2fine(oldData, dataType, startBand, stopBand - startBand); } else { fine2coarse(oldData, dataType, startBand, stopBand - startBand); } } /* Handle strides */ pbStride = getBsFreqResStride(losslessData->bsFreqResStrideXXX[paramIdx][ps]); dataBands = (stopBand - startBand - 1) / pbStride + 1; aStrides[0] = startBand; for (pb = 1; pb <= dataBands; pb++) { aStrides[pb] = aStrides[pb - 1] + pbStride; } strOffset = 0; while (aStrides[dataBands] > stopBand) { if (strOffset < dataBands) { strOffset++; } for (i = strOffset; i <= dataBands; i++) { aStrides[i]--; } } /* while */ for (pb = 0; pb < dataBands; pb++) { cmpOldData[startBand + pb] = oldData[aStrides[pb]]; cmpIdxData[0][startBand + pb] = data[ps][aStrides[pb]]; if (losslessData->bsDataPair[paramIdx][ps]) { cmpIdxData[1][startBand + pb] = data[ps + 1][aStrides[pb]]; } } /* for pb*/ /* Finally encode */ if (losslessData->bsDataPair[paramIdx][ps]) { fdk_sacenc_ecDataPairEnc(bitstream, cmpIdxData, cmpOldData, dataType, 0, startBand, dataBands, losslessData->bsQuantCoarseXXX[paramIdx][ps], independencyFlag && (ps == 0)); } else { fdk_sacenc_ecDataSingleEnc(bitstream, cmpIdxData, cmpOldData, dataType, 0, startBand, dataBands, losslessData->bsQuantCoarseXXX[paramIdx][ps], independencyFlag && (ps == 0)); } /* Overwrite old data */ for (i = startBand; i < stopBand; i++) { if (losslessData->bsDataPair[paramIdx][ps]) { oldData[i] = data[ps + 1][i]; } else { oldData[i] = data[ps][i]; } } quantCoarseXXXprev[paramIdx] = losslessData->bsQuantCoarseXXX[paramIdx][ps]; /* Jump forward if we have encoded a data pair */ if (losslessData->bsDataPair[paramIdx][ps]) { ps++; } } /* if (losslessData->bsXXXDataMode[paramIdx][ps] == FINECOARSE ) */ } /* for ps */ } /****************************************************************************/ /* Bitstream formatter interface functions */ /****************************************************************************/ static FDK_SACENC_ERROR getBsFreqResIndex(const INT numBands, INT *const pbsFreqResIndex) { FDK_SACENC_ERROR error = SACENC_OK; if (NULL == pbsFreqResIndex) { error = SACENC_INVALID_HANDLE; } else { const UCHAR *pFreqResBinTable = FreqResBinTable_LD; int i; *pbsFreqResIndex = -1; for (i = 0; i < MAX_FREQ_RES_INDEX; i++) { if (numBands == pFreqResBinTable[i]) { *pbsFreqResIndex = i; break; } } if (*pbsFreqResIndex < 0 || *pbsFreqResIndex >= MAX_FREQ_RES_INDEX) { error = SACENC_INVALID_CONFIG; } } return error; } static FDK_SACENC_ERROR getSamplingFrequencyIndex( const INT bsSamplingFrequency, INT *const pbsSamplingFrequencyIndex) { FDK_SACENC_ERROR error = SACENC_OK; if (NULL == pbsSamplingFrequencyIndex) { error = SACENC_INVALID_HANDLE; } else { int i; *pbsSamplingFrequencyIndex = SAMPLING_FREQUENCY_INDEX_ESCAPE; for (i = 0; i < MAX_SAMPLING_FREQUENCY_INDEX; i++) { if (bsSamplingFrequency == SampleRateTable[i]) { /*spatial sampling rate*/ *pbsSamplingFrequencyIndex = i; break; } } } return error; } /* destroy encoder instance */ FDK_SACENC_ERROR fdk_sacenc_destroySpatialBitstreamEncoder( HANDLE_BSF_INSTANCE *selfPtr) { FDK_SACENC_ERROR error = SACENC_OK; if ((selfPtr == NULL) || (*selfPtr == NULL)) { error = SACENC_INVALID_HANDLE; } else { if (*selfPtr != NULL) { FDK_FREE_MEMORY_1D(*selfPtr); } } return error; } /* create encoder instance */ FDK_SACENC_ERROR fdk_sacenc_createSpatialBitstreamEncoder( HANDLE_BSF_INSTANCE *selfPtr) { FDK_SACENC_ERROR error = SACENC_OK; if (NULL == selfPtr) { error = SACENC_INVALID_HANDLE; } else { /* allocate encoder struct */ FDK_ALLOCATE_MEMORY_1D(*selfPtr, 1, BSF_INSTANCE); } return error; bail: fdk_sacenc_destroySpatialBitstreamEncoder(selfPtr); return ((SACENC_OK == error) ? SACENC_MEMORY_ERROR : error); } /* init encoder instance */ FDK_SACENC_ERROR fdk_sacenc_initSpatialBitstreamEncoder( HANDLE_BSF_INSTANCE selfPtr) { FDK_SACENC_ERROR error = SACENC_OK; if (selfPtr == NULL) { error = SACENC_INVALID_HANDLE; } else { /* init/clear */ clearFrame(&selfPtr->frame); } /* valid handle */ return error; } /* get SpatialSpecificConfig struct */ SPATIALSPECIFICCONFIG *fdk_sacenc_getSpatialSpecificConfig( HANDLE_BSF_INSTANCE selfPtr) { return ((selfPtr == NULL) ? NULL : &(selfPtr->spatialSpecificConfig)); } /* write SpatialSpecificConfig to stream */ FDK_SACENC_ERROR fdk_sacenc_writeSpatialSpecificConfig( SPATIALSPECIFICCONFIG *const spatialSpecificConfig, UCHAR *const pOutputBuffer, const INT outputBufferSize, INT *const pnOutputBits) { FDK_SACENC_ERROR error = SACENC_OK; INT bsSamplingFrequencyIndex = 0; INT bsFreqRes = 0; if ((spatialSpecificConfig == NULL) || (pOutputBuffer == NULL) || (pnOutputBits == NULL)) { error = SACENC_INVALID_HANDLE; } else { FDK_BITSTREAM bitstream; /* Find FreqRes */ if (SACENC_OK != (error = getBsFreqResIndex(spatialSpecificConfig->numBands, &bsFreqRes))) goto bail; /* Find SamplingFrequencyIndex */ if (SACENC_OK != (error = getSamplingFrequencyIndex( spatialSpecificConfig->bsSamplingFrequency, &bsSamplingFrequencyIndex))) goto bail; /* bind extern buffer to bitstream handle */ FDKinitBitStream(&bitstream, pOutputBuffer, outputBufferSize, 0, BS_WRITER); /****************************************************************************/ /* write to bitstream */ FDKwriteBits(&bitstream, bsSamplingFrequencyIndex, 4); if (bsSamplingFrequencyIndex == 15) { FDKwriteBits(&bitstream, spatialSpecificConfig->bsSamplingFrequency, 24); } FDKwriteBits(&bitstream, spatialSpecificConfig->bsFrameLength, 5); FDKwriteBits(&bitstream, bsFreqRes, 3); FDKwriteBits(&bitstream, spatialSpecificConfig->bsTreeConfig, 4); FDKwriteBits(&bitstream, spatialSpecificConfig->bsQuantMode, 2); FDKwriteBits(&bitstream, 0, 1); /* bsArbitraryDownmix */ FDKwriteBits(&bitstream, spatialSpecificConfig->bsFixedGainDMX, 3); FDKwriteBits(&bitstream, TEMPSHAPE_OFF, 2); FDKwriteBits(&bitstream, spatialSpecificConfig->bsDecorrConfig, 2); FDKbyteAlign(&bitstream, 0); /* byte alignment */ /* return number of valid bits in bitstream */ if ((*pnOutputBits = FDKgetValidBits(&bitstream)) > (outputBufferSize * 8)) { error = SACENC_INVALID_CONFIG; goto bail; } /* terminate buffer with alignment */ FDKbyteAlign(&bitstream, 0); } /* valid handle */ bail: return error; } /* get SpatialFrame struct */ SPATIALFRAME *fdk_sacenc_getSpatialFrame(HANDLE_BSF_INSTANCE selfPtr, const SPATIALFRAME_TYPE frameType) { int idx = -1; switch (frameType) { case READ_SPATIALFRAME: case WRITE_SPATIALFRAME: idx = 0; break; default: idx = -1; /* invalid configuration */ } /* switch frameType */ return (((selfPtr == NULL) || (idx == -1)) ? NULL : &selfPtr->frame); } static FDK_SACENC_ERROR writeFramingInfo(HANDLE_FDK_BITSTREAM hBitstream, const FRAMINGINFO *const pFramingInfo, const INT frameLength) { FDK_SACENC_ERROR error = SACENC_OK; if ((hBitstream == NULL) || (pFramingInfo == NULL)) { error = SACENC_INVALID_HANDLE; } else { FDKwriteBits(hBitstream, pFramingInfo->bsFramingType, 1); FDKwriteBits(hBitstream, pFramingInfo->numParamSets - 1, 1); if (pFramingInfo->bsFramingType) { int ps = 0; int numParamSets = pFramingInfo->numParamSets; { for (ps = 0; ps < numParamSets; ps++) { int bitsParamSlot = 0; while ((1 << bitsParamSlot) < (frameLength + 1)) bitsParamSlot++; if (bitsParamSlot > 0) FDKwriteBits(hBitstream, pFramingInfo->bsParamSlots[ps], bitsParamSlot); } } } /* pFramingInfo->bsFramingType */ } /* valid handle */ return error; } static FDK_SACENC_ERROR writeSmgData(HANDLE_FDK_BITSTREAM hBitstream, const SMGDATA *const pSmgData, const INT numParamSets, const INT dataBands) { FDK_SACENC_ERROR error = SACENC_OK; if ((hBitstream == NULL) || (pSmgData == NULL)) { error = SACENC_INVALID_HANDLE; } else { int i, j; for (i = 0; i < numParamSets; i++) { FDKwriteBits(hBitstream, pSmgData->bsSmoothMode[i], 2); if (pSmgData->bsSmoothMode[i] >= 2) { FDKwriteBits(hBitstream, pSmgData->bsSmoothTime[i], 2); } if (pSmgData->bsSmoothMode[i] == 3) { const int stride = getBsFreqResStride(pSmgData->bsFreqResStride[i]); FDKwriteBits(hBitstream, pSmgData->bsFreqResStride[i], 2); for (j = 0; j < dataBands; j += stride) { FDKwriteBits(hBitstream, pSmgData->bsSmgData[i][j], 1); } } } /* for i */ } /* valid handle */ return error; } static FDK_SACENC_ERROR writeOttData( HANDLE_FDK_BITSTREAM hBitstream, PREV_OTTDATA *const pPrevOttData, OTTDATA *const pOttData, const OTTCONFIG ottConfig[SACENC_MAX_NUM_BOXES], LOSSLESSDATA *const pCLDLosslessData, LOSSLESSDATA *const pICCLosslessData, const INT numOttBoxes, const INT numBands, const INT numParamSets, const INT bsIndependencyFlag) { FDK_SACENC_ERROR error = SACENC_OK; if ((hBitstream == NULL) || (pPrevOttData == NULL) || (pOttData == NULL) || (ottConfig == NULL) || (pCLDLosslessData == NULL) || (pICCLosslessData == NULL)) { error = SACENC_INVALID_HANDLE; } else { int i; for (i = 0; i < numOttBoxes; i++) { ecData(hBitstream, pOttData->cld[i], pPrevOttData->cld_old[i], pPrevOttData->quantCoarseCldPrev[i], pCLDLosslessData, t_CLD, i, numParamSets, bsIndependencyFlag, 0, ottConfig[i].bsOttBands, 15); } { for (i = 0; i < numOttBoxes; i++) { { ecData(hBitstream, pOttData->icc[i], pPrevOttData->icc_old[i], pPrevOttData->quantCoarseIccPrev[i], pICCLosslessData, t_ICC, i, numParamSets, bsIndependencyFlag, 0, numBands, 0); } } /* for i */ } } /* valid handle */ return error; } /* write extension frame data to stream */ static FDK_SACENC_ERROR WriteSpatialExtensionFrame( HANDLE_FDK_BITSTREAM bitstream, HANDLE_BSF_INSTANCE self) { FDK_SACENC_ERROR error = SACENC_OK; if ((bitstream == NULL) || (self == NULL)) { error = SACENC_INVALID_HANDLE; } else { FDKbyteAlign(bitstream, 0); } /* valid handle */ return error; } /* write frame data to stream */ FDK_SACENC_ERROR fdk_sacenc_writeSpatialFrame(UCHAR *const pOutputBuffer, const INT outputBufferSize, INT *const pnOutputBits, HANDLE_BSF_INSTANCE selfPtr) { FDK_SACENC_ERROR error = SACENC_OK; if ((pOutputBuffer == NULL) || (pnOutputBits == NULL) || (selfPtr == NULL)) { error = SACENC_INVALID_HANDLE; } else { SPATIALFRAME *frame = NULL; SPATIALSPECIFICCONFIG *config = NULL; FDK_BITSTREAM bitstream; int i, j, numParamSets, numOttBoxes; if ((NULL == (frame = fdk_sacenc_getSpatialFrame(selfPtr, READ_SPATIALFRAME))) || (NULL == (config = &(selfPtr->spatialSpecificConfig)))) { error = SACENC_INVALID_HANDLE; goto bail; } numOttBoxes = selfPtr->spatialSpecificConfig.treeDescription.numOttBoxes; numParamSets = frame->framingInfo.numParamSets; if (frame->bUseBBCues) { for (i = 0; i < SACENC_MAX_NUM_BOXES; i++) { /* If a transient was detected, force only the second ps broad band */ if (numParamSets == 1) { frame->CLDLosslessData.bsFreqResStrideXXX[i][0] = 3; frame->ICCLosslessData.bsFreqResStrideXXX[i][0] = 3; } else { for (j = 1; j < MAX_NUM_PARAMS; j++) { frame->CLDLosslessData.bsFreqResStrideXXX[i][j] = 3; frame->ICCLosslessData.bsFreqResStrideXXX[i][j] = 3; } } } } /* frame->bUseBBCues */ /* bind extern buffer to bitstream handle */ FDKinitBitStream(&bitstream, pOutputBuffer, outputBufferSize, 0, BS_WRITER); if (SACENC_OK != (error = writeFramingInfo( &bitstream, &(frame->framingInfo), selfPtr->spatialSpecificConfig.bsFrameLength))) { goto bail; } /* write bsIndependencyFlag */ FDKwriteBits(&bitstream, frame->bsIndependencyFlag, 1); /* write spatial data to bitstream */ if (SACENC_OK != (error = writeOttData(&bitstream, &selfPtr->prevFrameData.prevOttData, &frame->ottData, config->ottConfig, &frame->CLDLosslessData, &frame->ICCLosslessData, numOttBoxes, config->numBands, numParamSets, frame->bsIndependencyFlag))) { goto bail; } if (SACENC_OK != (error = writeSmgData(&bitstream, &frame->smgData, numParamSets, config->numBands))) { goto bail; } /* byte alignment */ FDKbyteAlign(&bitstream, 0); /* Write SpatialExtensionFrame */ if (SACENC_OK != (error = WriteSpatialExtensionFrame(&bitstream, selfPtr))) { goto bail; } if (NULL == (frame = fdk_sacenc_getSpatialFrame(selfPtr, WRITE_SPATIALFRAME))) { error = SACENC_INVALID_HANDLE; goto bail; } clearFrame(frame); /* return number of valid bits in bitstream */ if ((*pnOutputBits = FDKgetValidBits(&bitstream)) > (outputBufferSize * 8)) { error = SACENC_INVALID_CONFIG; goto bail; } /* terminate buffer with alignment */ FDKbyteAlign(&bitstream, 0); } /* valid handle */ bail: return error; } fdk-aac-2.0.2/libSACenc/src/sacenc_bitstream.h000066400000000000000000000230461404225310700210350ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround encoder library ************************* Author(s): Description: Encoder Library Interface Bitstream Writer *******************************************************************************/ #ifndef SACENC_BITSTREAM_H #define SACENC_BITSTREAM_H /* Includes ******************************************************************/ #include "FDK_bitstream.h" #include "FDK_matrixCalloc.h" #include "sacenc_lib.h" #include "sacenc_const.h" /* Defines *******************************************************************/ #define MAX_NUM_BINS 23 #define MAX_NUM_PARAMS 2 #define MAX_NUM_OUTPUTCHANNELS SACENC_MAX_OUTPUT_CHANNELS #define MAX_TIME_SLOTS 32 typedef enum { TREE_212 = 7, TREE_ESCAPE = 15 } TREECONFIG; typedef enum { FREQ_RES_40 = 0, FREQ_RES_20 = 1, FREQ_RES_10 = 2, FREQ_RES_5 = 3 } FREQ; typedef enum { QUANTMODE_INVALID = -1, QUANTMODE_FINE = 0, QUANTMODE_EBQ1 = 1, QUANTMODE_EBQ2 = 2 } QUANTMODE; typedef enum { TEMPSHAPE_OFF = 0 } TEMPSHAPECONFIG; typedef enum { FIXEDGAINDMX_INVALID = -1, FIXEDGAINDMX_0 = 0, FIXEDGAINDMX_1 = 1, FIXEDGAINDMX_2 = 2, FIXEDGAINDMX_3 = 3, FIXEDGAINDMX_4 = 4, FIXEDGAINDMX_5 = 5, FIXEDGAINDMX_6 = 6, FIXEDGAINDMX_7 = 7 } FIXEDGAINDMXCONFIG; typedef enum { DECORR_INVALID = -1, DECORR_QMFSPLIT0 = 0, /* QMF splitfreq: 3, 15, 24, 65 */ DECORR_QMFSPLIT1 = 1, /* QMF splitfreq: 3, 50, 65, 65 */ DECORR_QMFSPLIT2 = 2 /* QMF splitfreq: 0, 15, 65, 65 */ } DECORRCONFIG; typedef enum { DEFAULT = 0, KEEP = 1, INTERPOLATE = 2, FINECOARSE = 3 } DATA_MODE; typedef enum { READ_SPATIALFRAME = 0, WRITE_SPATIALFRAME = 1 } SPATIALFRAME_TYPE; /* Data Types ****************************************************************/ typedef struct { INT numOttBoxes; INT numInChan; INT numOutChan; } TREEDESCRIPTION; typedef struct { INT bsOttBands; } OTTCONFIG; typedef struct { INT bsSamplingFrequency; /* for bsSamplingFrequencyIndex */ INT bsFrameLength; INT numBands; /* for bsFreqRes */ TREECONFIG bsTreeConfig; QUANTMODE bsQuantMode; FIXEDGAINDMXCONFIG bsFixedGainDMX; int bsEnvQuantMode; DECORRCONFIG bsDecorrConfig; TREEDESCRIPTION treeDescription; OTTCONFIG ottConfig[SACENC_MAX_NUM_BOXES]; } SPATIALSPECIFICCONFIG; typedef struct { UCHAR bsFramingType; UCHAR numParamSets; UCHAR bsParamSlots[MAX_NUM_PARAMS]; } FRAMINGINFO; typedef struct { SCHAR cld[SACENC_MAX_NUM_BOXES][MAX_NUM_PARAMS][MAX_NUM_BINS]; SCHAR icc[SACENC_MAX_NUM_BOXES][MAX_NUM_PARAMS][MAX_NUM_BINS]; } OTTDATA; typedef struct { UCHAR bsSmoothMode[MAX_NUM_PARAMS]; UCHAR bsSmoothTime[MAX_NUM_PARAMS]; UCHAR bsFreqResStride[MAX_NUM_PARAMS]; UCHAR bsSmgData[MAX_NUM_PARAMS][MAX_NUM_BINS]; } SMGDATA; typedef struct { UCHAR bsEnvShapeChannel[MAX_NUM_OUTPUTCHANNELS]; UCHAR bsEnvShapeData[MAX_NUM_OUTPUTCHANNELS][MAX_TIME_SLOTS]; } TEMPSHAPEDATA; typedef struct { UCHAR bsXXXDataMode[SACENC_MAX_NUM_BOXES][MAX_NUM_PARAMS]; UCHAR bsDataPair[SACENC_MAX_NUM_BOXES][MAX_NUM_PARAMS]; UCHAR bsQuantCoarseXXX[SACENC_MAX_NUM_BOXES][MAX_NUM_PARAMS]; UCHAR bsFreqResStrideXXX[SACENC_MAX_NUM_BOXES][MAX_NUM_PARAMS]; } LOSSLESSDATA; typedef struct { FRAMINGINFO framingInfo; UCHAR bsIndependencyFlag; OTTDATA ottData; SMGDATA smgData; TEMPSHAPEDATA tempShapeData; LOSSLESSDATA CLDLosslessData; LOSSLESSDATA ICCLosslessData; UCHAR bUseBBCues; } SPATIALFRAME; typedef struct BSF_INSTANCE *HANDLE_BSF_INSTANCE; /* Constants *****************************************************************/ /* Function / Class Declarations *********************************************/ /* destroy encoder instance */ FDK_SACENC_ERROR fdk_sacenc_destroySpatialBitstreamEncoder( HANDLE_BSF_INSTANCE *selfPtr); /* create encoder instance */ FDK_SACENC_ERROR fdk_sacenc_createSpatialBitstreamEncoder( HANDLE_BSF_INSTANCE *selfPtr); FDK_SACENC_ERROR fdk_sacenc_initSpatialBitstreamEncoder( HANDLE_BSF_INSTANCE selfPtr); /* get SpatialSpecificConfig struct */ SPATIALSPECIFICCONFIG *fdk_sacenc_getSpatialSpecificConfig( HANDLE_BSF_INSTANCE selfPtr); /* write SpatialSpecificConfig to stream */ FDK_SACENC_ERROR fdk_sacenc_writeSpatialSpecificConfig( SPATIALSPECIFICCONFIG *const spatialSpecificConfig, UCHAR *const pOutputBuffer, const INT outputBufferSize, INT *const pnOutputBits); /* get SpatialFrame struct */ SPATIALFRAME *fdk_sacenc_getSpatialFrame(HANDLE_BSF_INSTANCE selfPtr, const SPATIALFRAME_TYPE frameType); /* write frame data to stream */ FDK_SACENC_ERROR fdk_sacenc_writeSpatialFrame(UCHAR *const pOutputBuffer, const INT outputBufferSize, INT *const pnOutputBits, HANDLE_BSF_INSTANCE selfPtr); /* Copy/Save spatial frame data for one parameter set */ FDK_SACENC_ERROR fdk_sacenc_duplicateParameterSet( const SPATIALFRAME *const hFrom, const INT setFrom, SPATIALFRAME *const hTo, const INT setTo); #endif /* SACENC_BITSTREAM_H */ fdk-aac-2.0.2/libSACenc/src/sacenc_const.h000066400000000000000000000127441404225310700201740ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround encoder library ************************* Author(s): Markus Multrus Description: Encoder Library Interface constants to MPEG-4 spatial encoder lib *******************************************************************************/ #ifndef SACENC_CONST_H #define SACENC_CONST_H /* Includes ******************************************************************/ #include "machine_type.h" /* Defines *******************************************************************/ #define NUM_QMF_BANDS 64 #define MAX_QMF_BANDS 128 #define SACENC_MAX_NUM_BOXES 1 #define SACENC_MAX_INPUT_CHANNELS 2 #define SACENC_MAX_OUTPUT_CHANNELS 1 #define SACENC_FLOAT_EPSILON (1e-9f) /* Data Types ****************************************************************/ /* Constants *****************************************************************/ /* Function / Class Declarations *********************************************/ #endif /* SACENC_CONST_H */ fdk-aac-2.0.2/libSACenc/src/sacenc_delay.cpp000066400000000000000000000435421404225310700204770ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround encoder library ************************* Author(s): Christian Goettlinger Description: Encoder Library Interface delay management of the encoder *******************************************************************************/ /**************************************************************************/ /** \file This file contains all delay infrastructure ******************************************************************************/ /* Includes ******************************************************************/ #include "sacenc_delay.h" #include "sacenc_const.h" #include "FDK_matrixCalloc.h" /* Defines *******************************************************************/ /* Data Types ****************************************************************/ struct DELAY { struct DELAY_CONFIG { /* Routing Config Switches*/ INT bDmxAlign; INT bTimeDomDmx; INT bMinimizeDelay; INT bSacTimeAlignmentDynamicOut; /* Needed Input Variables*/ INT nQmfLen; INT nFrameLen; INT nSurroundDelay; INT nArbDmxDelay; INT nLimiterDelay; INT nCoreCoderDelay; INT nSacStreamMuxDelay; INT nSacTimeAlignment; /* Overwritten, if bSacTimeAlignmentDynamicOut */ } config; /* Variable Delaybuffers -> Delays */ INT nDmxAlignBuffer; INT nSurroundAnalysisBuffer; INT nArbDmxAnalysisBuffer; INT nOutputAudioBuffer; INT nBitstreamFrameBuffer; INT nOutputAudioQmfFrameBuffer; INT nDiscardOutFrames; /* Variable Delaybuffers Computation Variables */ INT nBitstreamFrameBufferSize; /* Output: Infos */ INT nInfoDmxDelay; /* Delay of the downmixed signal after the space encoder */ INT nInfoCodecDelay; /* Delay of the whole en-/decoder including CoreCoder */ INT nInfoDecoderDelay; /* Delay of the Mpeg Surround decoder */ }; /* Constants *****************************************************************/ /* Function / Class Declarations *********************************************/ /* Function / Class Definition ***********************************************/ /*----------------------------------------------------------------------------- functionname: fdk_sacenc_delay_Open() description: initializes Delays returns: noError on success, an apropriate error code else -----------------------------------------------------------------------------*/ FDK_SACENC_ERROR fdk_sacenc_delay_Open(HANDLE_DELAY *phDelay) { FDK_SACENC_ERROR error = SACENC_OK; if (NULL == phDelay) { error = SACENC_INVALID_HANDLE; } else { FDK_ALLOCATE_MEMORY_1D(*phDelay, 1, struct DELAY); } return error; bail: fdk_sacenc_delay_Close(phDelay); return ((SACENC_OK == error) ? SACENC_MEMORY_ERROR : error); } /*----------------------------------------------------------------------------- functionname: fdk_sacenc_delay_Close() description: destructs Delay returns: noError on success, an apropriate error code else -----------------------------------------------------------------------------*/ FDK_SACENC_ERROR fdk_sacenc_delay_Close(HANDLE_DELAY *phDelay) { FDK_SACENC_ERROR error = SACENC_OK; if (NULL == phDelay) { error = SACENC_INVALID_HANDLE; } else { if (NULL != *phDelay) { FDK_FREE_MEMORY_1D(*phDelay); } } return error; } FDK_SACENC_ERROR fdk_sacenc_delay_Init(HANDLE_DELAY hDelay, const INT nQmfLen, const INT nFrameLen, const INT nCoreCoderDelay, const INT nSacStreamMuxDelay) { FDK_SACENC_ERROR error = SACENC_OK; if (NULL == hDelay) { error = SACENC_INVALID_HANDLE; } else { /* Fill structure before calculation */ FDKmemclear(&hDelay->config, sizeof(hDelay->config)); hDelay->config.nQmfLen = nQmfLen; hDelay->config.nFrameLen = nFrameLen; hDelay->config.nCoreCoderDelay = nCoreCoderDelay; hDelay->config.nSacStreamMuxDelay = nSacStreamMuxDelay; } return error; } /*----------------------------------------------------------------------------- functionname: fdk_sacenc_delay_SubCalulateBufferDelays() description: Calculates the Delays of the buffers returns: Error Code -----------------------------------------------------------------------------*/ FDK_SACENC_ERROR fdk_sacenc_delay_SubCalulateBufferDelays(HANDLE_DELAY hDel) { FDK_SACENC_ERROR error = SACENC_OK; if (NULL == hDel) { error = SACENC_INVALID_HANDLE; } else { int nEncoderAnDelay, nEncoderSynDelay, nEncoderWinDelay, nDecoderAnDelay, nDecoderSynDelay, nResidualCoderFrameDelay, nArbDmxResidualCoderFrameDelay; if (hDel->config.bSacTimeAlignmentDynamicOut > 0) { hDel->config.nSacTimeAlignment = 0; } { nEncoderAnDelay = 2 * hDel->config.nQmfLen + hDel->config.nQmfLen / 2; /* Only Ld-QMF Delay, no hybrid */ nEncoderSynDelay = 1 * hDel->config.nQmfLen + hDel->config.nQmfLen / 2; nDecoderAnDelay = 2 * hDel->config.nQmfLen + hDel->config.nQmfLen / 2; nDecoderSynDelay = 1 * hDel->config.nQmfLen + hDel->config.nQmfLen / 2; nEncoderWinDelay = hDel->config.nFrameLen / 2; /* WindowLookahead is just half a frame */ } { nResidualCoderFrameDelay = 0; } { nArbDmxResidualCoderFrameDelay = 0; } /* Calculate variable Buffer-Delays */ if (hDel->config.bTimeDomDmx == 0) { /* ArbitraryDmx and TdDmx off */ int tempDelay; hDel->nSurroundAnalysisBuffer = 0; hDel->nArbDmxAnalysisBuffer = 0; tempDelay = nEncoderSynDelay + hDel->config.nLimiterDelay + hDel->config.nCoreCoderDelay + hDel->config.nSacTimeAlignment + nDecoderAnDelay; tempDelay = (nResidualCoderFrameDelay * hDel->config.nFrameLen) + hDel->config.nSacStreamMuxDelay - tempDelay; if (tempDelay > 0) { hDel->nBitstreamFrameBuffer = 0; hDel->nOutputAudioBuffer = tempDelay; } else { tempDelay = -tempDelay; hDel->nBitstreamFrameBuffer = (tempDelay + hDel->config.nFrameLen - 1) / hDel->config.nFrameLen; hDel->nOutputAudioBuffer = (hDel->nBitstreamFrameBuffer * hDel->config.nFrameLen) - tempDelay; } hDel->nOutputAudioQmfFrameBuffer = (hDel->nOutputAudioBuffer + (hDel->config.nQmfLen / 2) - 1) / hDel->config.nQmfLen; if (hDel->config.bDmxAlign > 0) { tempDelay = nEncoderWinDelay + nEncoderAnDelay + nEncoderSynDelay + hDel->nOutputAudioBuffer + hDel->config.nLimiterDelay + hDel->config.nCoreCoderDelay; hDel->nDiscardOutFrames = (tempDelay + hDel->config.nFrameLen - 1) / hDel->config.nFrameLen; hDel->nDmxAlignBuffer = hDel->nDiscardOutFrames * hDel->config.nFrameLen - tempDelay; } else { hDel->nDiscardOutFrames = 0; hDel->nDmxAlignBuffer = 0; } /* Output: Info-Variables */ hDel->nInfoDmxDelay = hDel->nSurroundAnalysisBuffer + nEncoderAnDelay + nEncoderWinDelay + nEncoderSynDelay + hDel->nOutputAudioBuffer + hDel->config.nLimiterDelay; hDel->nInfoCodecDelay = hDel->nInfoDmxDelay + hDel->config.nCoreCoderDelay + hDel->config.nSacTimeAlignment + nDecoderAnDelay + nDecoderSynDelay; } else { /* ArbitraryDmx or TdDmx on */ int tempDelay1, tempDelay2, tempDelay12, tempDelay3; tempDelay1 = hDel->config.nArbDmxDelay - hDel->config.nSurroundDelay; if (tempDelay1 >= 0) { hDel->nSurroundAnalysisBuffer = tempDelay1; hDel->nArbDmxAnalysisBuffer = 0; } else { hDel->nSurroundAnalysisBuffer = 0; hDel->nArbDmxAnalysisBuffer = -tempDelay1; } tempDelay1 = nEncoderWinDelay + hDel->config.nSurroundDelay + hDel->nSurroundAnalysisBuffer + nEncoderAnDelay; /*Surround Path*/ tempDelay2 = nEncoderWinDelay + hDel->config.nArbDmxDelay + hDel->nArbDmxAnalysisBuffer + nEncoderAnDelay; /* ArbDmx Compare Path */ tempDelay3 = hDel->config.nArbDmxDelay + hDel->config.nLimiterDelay + hDel->config.nCoreCoderDelay + hDel->config.nSacTimeAlignment + nDecoderAnDelay; /* ArbDmx Passthrough*/ tempDelay12 = FDKmax(nResidualCoderFrameDelay, nArbDmxResidualCoderFrameDelay) * hDel->config.nFrameLen; tempDelay12 += hDel->config.nSacStreamMuxDelay; if (tempDelay1 > tempDelay2) { tempDelay12 += tempDelay1; } else { tempDelay12 += tempDelay2; } if (tempDelay3 > tempDelay12) { if (hDel->config.bMinimizeDelay > 0) { hDel->nBitstreamFrameBuffer = (tempDelay3 - tempDelay12) / hDel->config.nFrameLen; /*floor*/ hDel->nOutputAudioBuffer = 0; hDel->nSurroundAnalysisBuffer += (tempDelay3 - tempDelay12 - (hDel->nBitstreamFrameBuffer * hDel->config.nFrameLen)); hDel->nArbDmxAnalysisBuffer += (tempDelay3 - tempDelay12 - (hDel->nBitstreamFrameBuffer * hDel->config.nFrameLen)); } else { hDel->nBitstreamFrameBuffer = ((tempDelay3 - tempDelay12) + hDel->config.nFrameLen - 1) / hDel->config.nFrameLen; hDel->nOutputAudioBuffer = hDel->nBitstreamFrameBuffer * hDel->config.nFrameLen + tempDelay12 - tempDelay3; } } else { hDel->nBitstreamFrameBuffer = 0; hDel->nOutputAudioBuffer = tempDelay12 - tempDelay3; } if (hDel->config.bDmxAlign > 0) { int tempDelay = hDel->config.nArbDmxDelay + hDel->nOutputAudioBuffer + hDel->config.nLimiterDelay + hDel->config.nCoreCoderDelay; hDel->nDiscardOutFrames = (tempDelay + hDel->config.nFrameLen - 1) / hDel->config.nFrameLen; hDel->nDmxAlignBuffer = hDel->nDiscardOutFrames * hDel->config.nFrameLen - tempDelay; } else { hDel->nDiscardOutFrames = 0; hDel->nDmxAlignBuffer = 0; } /* Output: Info-Variables */ hDel->nInfoDmxDelay = hDel->config.nArbDmxDelay + hDel->nOutputAudioBuffer + hDel->config.nLimiterDelay; hDel->nInfoCodecDelay = hDel->nInfoDmxDelay + hDel->config.nCoreCoderDelay + hDel->config.nSacTimeAlignment + nDecoderAnDelay + nDecoderSynDelay; hDel->nInfoDecoderDelay = nDecoderAnDelay + nDecoderSynDelay; } /* ArbitraryDmx or TdDmx on */ /* Additonal Variables needed for Computation Issues */ hDel->nBitstreamFrameBufferSize = hDel->nBitstreamFrameBuffer + 1; } return error; } static FDK_SACENC_ERROR assignParameterInRange( const INT startRange, /* including startRange */ const INT stopRange, /* including stopRange */ const INT value, /* value to write*/ INT *const ptr /* destination pointer*/ ) { FDK_SACENC_ERROR error = SACENC_INVALID_CONFIG; if ((startRange <= value) && (value <= stopRange)) { *ptr = value; error = SACENC_OK; } return error; } FDK_SACENC_ERROR fdk_sacenc_delay_SetDmxAlign(HANDLE_DELAY hDelay, const INT bDmxAlignIn) { return (assignParameterInRange(0, 1, bDmxAlignIn, &hDelay->config.bDmxAlign)); } FDK_SACENC_ERROR fdk_sacenc_delay_SetTimeDomDmx(HANDLE_DELAY hDelay, const INT bTimeDomDmxIn) { return ( assignParameterInRange(0, 1, bTimeDomDmxIn, &hDelay->config.bTimeDomDmx)); } FDK_SACENC_ERROR fdk_sacenc_delay_SetSacTimeAlignmentDynamicOut( HANDLE_DELAY hDelay, const INT bSacTimeAlignmentDynamicOutIn) { return (assignParameterInRange(0, 1, bSacTimeAlignmentDynamicOutIn, &hDelay->config.bSacTimeAlignmentDynamicOut)); } FDK_SACENC_ERROR fdk_sacenc_delay_SetNSacTimeAlignment( HANDLE_DELAY hDelay, const INT nSacTimeAlignmentIn) { return (assignParameterInRange(-32768, 32767, nSacTimeAlignmentIn, &hDelay->config.nSacTimeAlignment)); } FDK_SACENC_ERROR fdk_sacenc_delay_SetMinimizeDelay(HANDLE_DELAY hDelay, const INT bMinimizeDelay) { return (assignParameterInRange(0, 1, bMinimizeDelay, &hDelay->config.bMinimizeDelay)); } INT fdk_sacenc_delay_GetOutputAudioBufferDelay(HANDLE_DELAY hDelay) { return (hDelay->nOutputAudioBuffer); } INT fdk_sacenc_delay_GetSurroundAnalysisBufferDelay(HANDLE_DELAY hDelay) { return (hDelay->nSurroundAnalysisBuffer); } INT fdk_sacenc_delay_GetArbDmxAnalysisBufferDelay(HANDLE_DELAY hDelay) { return (hDelay->nArbDmxAnalysisBuffer); } INT fdk_sacenc_delay_GetBitstreamFrameBufferSize(HANDLE_DELAY hDelay) { return (hDelay->nBitstreamFrameBufferSize); } INT fdk_sacenc_delay_GetDmxAlignBufferDelay(HANDLE_DELAY hDelay) { return (hDelay->nDmxAlignBuffer); } INT fdk_sacenc_delay_GetDiscardOutFrames(HANDLE_DELAY hDelay) { return (hDelay->nDiscardOutFrames); } INT fdk_sacenc_delay_GetInfoDmxDelay(HANDLE_DELAY hDelay) { return (hDelay->nInfoDmxDelay); } INT fdk_sacenc_delay_GetInfoCodecDelay(HANDLE_DELAY hDelay) { return (hDelay->nInfoCodecDelay); } INT fdk_sacenc_delay_GetInfoDecoderDelay(HANDLE_DELAY hDelay) { return (hDelay->nInfoDecoderDelay); } fdk-aac-2.0.2/libSACenc/src/sacenc_delay.h000066400000000000000000000170671404225310700201470ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround encoder library ************************* Author(s): Christian Goettlinger Description: Encoder Library Interface delay management of the encoder *******************************************************************************/ /**************************************************************************/ /** \file ******************************************************************************/ #ifndef SACENC_DELAY_H #define SACENC_DELAY_H /* Includes ******************************************************************/ #include "sacenc_lib.h" #include "machine_type.h" #include "FDK_matrixCalloc.h" /* Defines *******************************************************************/ #define MAX_DELAY_INPUT 1024 #define MAX_DELAY_OUTPUT 4096 /* bumped from 0 to 5. this should be equal or larger to the dualrate sbr * resampler filter length */ #define MAX_DELAY_SURROUND_ANALYSIS 5 #define MAX_BITSTREAM_DELAY 1 /* Data Types ****************************************************************/ typedef struct DELAY *HANDLE_DELAY; /* Constants *****************************************************************/ /* Function / Class Declarations *********************************************/ FDK_SACENC_ERROR fdk_sacenc_delay_Open(HANDLE_DELAY *phDelay); FDK_SACENC_ERROR fdk_sacenc_delay_Close(HANDLE_DELAY *phDelay); FDK_SACENC_ERROR fdk_sacenc_delay_Init(HANDLE_DELAY hDelay, const INT nQmfLen, const INT nFrameLen, const INT nCoreCoderDelay, const INT nSacStreamMuxDelay); FDK_SACENC_ERROR fdk_sacenc_delay_SubCalulateBufferDelays(HANDLE_DELAY hDel); /* Set Expert Config Parameters */ FDK_SACENC_ERROR fdk_sacenc_delay_SetDmxAlign(HANDLE_DELAY hDelay, const INT bDmxAlignIn); FDK_SACENC_ERROR fdk_sacenc_delay_SetTimeDomDmx(HANDLE_DELAY hDelay, const INT bTimeDomDmxIn); FDK_SACENC_ERROR fdk_sacenc_delay_SetSacTimeAlignmentDynamicOut( HANDLE_DELAY hDelay, const INT bSacTimeAlignmentDynamicOutIn); FDK_SACENC_ERROR fdk_sacenc_delay_SetNSacTimeAlignment( HANDLE_DELAY hDelay, const INT nSacTimeAlignmentIn); FDK_SACENC_ERROR fdk_sacenc_delay_SetMinimizeDelay(HANDLE_DELAY hDelay, const INT bMinimizeDelay); /* Get Internal Variables */ INT fdk_sacenc_delay_GetOutputAudioBufferDelay(HANDLE_DELAY hDelay); INT fdk_sacenc_delay_GetSurroundAnalysisBufferDelay(HANDLE_DELAY hDelay); INT fdk_sacenc_delay_GetArbDmxAnalysisBufferDelay(HANDLE_DELAY hDelay); INT fdk_sacenc_delay_GetBitstreamFrameBufferSize(HANDLE_DELAY hDelay); INT fdk_sacenc_delay_GetDmxAlignBufferDelay(HANDLE_DELAY hDelay); INT fdk_sacenc_delay_GetDiscardOutFrames(HANDLE_DELAY hDelay); INT fdk_sacenc_delay_GetInfoDmxDelay(HANDLE_DELAY hDelay); INT fdk_sacenc_delay_GetInfoCodecDelay(HANDLE_DELAY hDelay); INT fdk_sacenc_delay_GetInfoDecoderDelay(HANDLE_DELAY hDelay); #endif /* SACENC_DELAY_H */ fdk-aac-2.0.2/libSACenc/src/sacenc_dmx_tdom_enh.cpp000066400000000000000000000573201404225310700220450ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround encoder library ************************* Author(s): M. Luis Valero Description: Enhanced Time Domain Downmix *******************************************************************************/ /* Includes ******************************************************************/ #include "sacenc_dmx_tdom_enh.h" #include "FDK_matrixCalloc.h" #include "FDK_trigFcts.h" #include "fixpoint_math.h" /* Defines *******************************************************************/ #define PI_FLT 3.1415926535897931f #define ALPHA_FLT 0.0001f #define PI_E (2) #define PI_M (FL2FXCONST_DBL(PI_FLT / (1 << PI_E))) #define ALPHA_E (13) #define ALPHA_M (FL2FXCONST_DBL(ALPHA_FLT * (1 << ALPHA_E))) enum { L = 0, R = 1 }; /* Data Types ****************************************************************/ typedef struct T_ENHANCED_TIME_DOMAIN_DMX { int maxFramelength; int framelength; FIXP_DBL prev_gain_m[2]; INT prev_gain_e; FIXP_DBL prev_H1_m[2]; INT prev_H1_e; FIXP_DBL *sinusWindow_m; SCHAR sinusWindow_e; FIXP_DBL prev_Left_m; INT prev_Left_e; FIXP_DBL prev_Right_m; INT prev_Right_e; FIXP_DBL prev_XNrg_m; INT prev_XNrg_e; FIXP_DBL lin_bbCld_weight_m; INT lin_bbCld_weight_e; FIXP_DBL gain_weight_m[2]; INT gain_weight_e; } ENHANCED_TIME_DOMAIN_DMX; /* Constants *****************************************************************/ /* Function / Class Declarations *********************************************/ static void calculateRatio(const FIXP_DBL sqrt_linCld_m, const INT sqrt_linCld_e, const FIXP_DBL lin_Cld_m, const INT lin_Cld_e, const FIXP_DBL Icc_m, const INT Icc_e, FIXP_DBL G_m[2], INT *G_e); static void calculateDmxGains(const FIXP_DBL lin_Cld_m, const INT lin_Cld_e, const FIXP_DBL lin_Cld2_m, const INT lin_Cld2_e, const FIXP_DBL Icc_m, const INT Icc_e, const FIXP_DBL G_m[2], const INT G_e, FIXP_DBL H1_m[2], INT *pH1_e); /* Function / Class Definition ***********************************************/ static FIXP_DBL invSqrtNorm2(const FIXP_DBL op_m, const INT op_e, INT *const result_e) { FIXP_DBL src_m = op_m; int src_e = op_e; if (src_e & 1) { src_m >>= 1; src_e += 1; } src_m = invSqrtNorm2(src_m, result_e); *result_e = (*result_e) - (src_e >> 1); return src_m; } static FIXP_DBL sqrtFixp(const FIXP_DBL op_m, const INT op_e, INT *const result_e) { FIXP_DBL src_m = op_m; int src_e = op_e; if (src_e & 1) { src_m >>= 1; src_e += 1; } *result_e = (src_e >> 1); return sqrtFixp(src_m); } static FIXP_DBL fixpAdd(const FIXP_DBL src1_m, const INT src1_e, const FIXP_DBL src2_m, const INT src2_e, INT *const dst_e) { FIXP_DBL dst_m; if (src1_m == FL2FXCONST_DBL(0.f)) { *dst_e = src2_e; dst_m = src2_m; } else if (src2_m == FL2FXCONST_DBL(0.f)) { *dst_e = src1_e; dst_m = src1_m; } else { *dst_e = fixMax(src1_e, src2_e) + 1; dst_m = scaleValue(src1_m, fixMax((src1_e - (*dst_e)), -(DFRACT_BITS - 1))) + scaleValue(src2_m, fixMax((src2_e - (*dst_e)), -(DFRACT_BITS - 1))); } return dst_m; } /** * \brief Sum up fixpoint values with best possible accuracy. * * \param value1 First input value. * \param q1 Scaling factor of first input value. * \param pValue2 Pointer to second input value, will be modified on * return. * \param pQ2 Pointer to second scaling factor, will be modified on * return. * * \return void */ static void fixpAddNorm(const FIXP_DBL value1, const INT q1, FIXP_DBL *const pValue2, INT *const pQ2) { const int headroom1 = fNormz(fixp_abs(value1)) - 1; const int headroom2 = fNormz(fixp_abs(*pValue2)) - 1; int resultScale = fixMax(q1 - headroom1, (*pQ2) - headroom2); if ((value1 != FL2FXCONST_DBL(0.f)) && (*pValue2 != FL2FXCONST_DBL(0.f))) { resultScale++; } *pValue2 = scaleValue(value1, q1 - resultScale) + scaleValue(*pValue2, fixMax(-(DFRACT_BITS - 1), ((*pQ2) - resultScale))); *pQ2 = (*pValue2 != (FIXP_DBL)0) ? resultScale : DFRACT_BITS - 1; } FDK_SACENC_ERROR fdk_sacenc_open_enhancedTimeDomainDmx( HANDLE_ENHANCED_TIME_DOMAIN_DMX *phEnhancedTimeDmx, const INT framelength) { FDK_SACENC_ERROR error = SACENC_OK; HANDLE_ENHANCED_TIME_DOMAIN_DMX hEnhancedTimeDmx = NULL; if (NULL == phEnhancedTimeDmx) { error = SACENC_INVALID_HANDLE; } else { FDK_ALLOCATE_MEMORY_1D(hEnhancedTimeDmx, 1, ENHANCED_TIME_DOMAIN_DMX); FDK_ALLOCATE_MEMORY_1D(hEnhancedTimeDmx->sinusWindow_m, 1 + framelength, FIXP_DBL); hEnhancedTimeDmx->maxFramelength = framelength; *phEnhancedTimeDmx = hEnhancedTimeDmx; } return error; bail: fdk_sacenc_close_enhancedTimeDomainDmx(&hEnhancedTimeDmx); return ((SACENC_OK == error) ? SACENC_MEMORY_ERROR : error); } FDK_SACENC_ERROR fdk_sacenc_init_enhancedTimeDomainDmx( HANDLE_ENHANCED_TIME_DOMAIN_DMX hEnhancedTimeDmx, const FIXP_DBL *const pInputGain_m, const INT inputGain_e, const FIXP_DBL outputGain_m, const INT outputGain_e, const INT framelength) { FDK_SACENC_ERROR error = SACENC_OK; if (hEnhancedTimeDmx == NULL) { error = SACENC_INVALID_HANDLE; } else { int smp; if (framelength > hEnhancedTimeDmx->maxFramelength) { error = SACENC_INIT_ERROR; goto bail; } hEnhancedTimeDmx->framelength = framelength; INT deltax_e; FIXP_DBL deltax_m; deltax_m = fDivNormHighPrec( PI_M, (FIXP_DBL)(2 * hEnhancedTimeDmx->framelength), &deltax_e); deltax_m = scaleValue(deltax_m, PI_E + deltax_e - (DFRACT_BITS - 1) - 1); deltax_e = 1; for (smp = 0; smp < hEnhancedTimeDmx->framelength + 1; smp++) { hEnhancedTimeDmx->sinusWindow_m[smp] = fMult(ALPHA_M, fPow2(fixp_sin(smp * deltax_m, deltax_e))); } hEnhancedTimeDmx->sinusWindow_e = -ALPHA_E; hEnhancedTimeDmx->prev_Left_m = hEnhancedTimeDmx->prev_Right_m = hEnhancedTimeDmx->prev_XNrg_m = FL2FXCONST_DBL(0.f); hEnhancedTimeDmx->prev_Left_e = hEnhancedTimeDmx->prev_Right_e = hEnhancedTimeDmx->prev_XNrg_e = DFRACT_BITS - 1; hEnhancedTimeDmx->lin_bbCld_weight_m = fDivNormHighPrec(fPow2(pInputGain_m[L]), fPow2(pInputGain_m[R]), &hEnhancedTimeDmx->lin_bbCld_weight_e); hEnhancedTimeDmx->gain_weight_m[L] = fMult(pInputGain_m[L], outputGain_m); hEnhancedTimeDmx->gain_weight_m[R] = fMult(pInputGain_m[R], outputGain_m); hEnhancedTimeDmx->gain_weight_e = -fNorm(fixMax(hEnhancedTimeDmx->gain_weight_m[L], hEnhancedTimeDmx->gain_weight_m[R])); hEnhancedTimeDmx->gain_weight_m[L] = scaleValue( hEnhancedTimeDmx->gain_weight_m[L], -hEnhancedTimeDmx->gain_weight_e); hEnhancedTimeDmx->gain_weight_m[R] = scaleValue( hEnhancedTimeDmx->gain_weight_m[R], -hEnhancedTimeDmx->gain_weight_e); hEnhancedTimeDmx->gain_weight_e += inputGain_e + outputGain_e; hEnhancedTimeDmx->prev_gain_m[L] = hEnhancedTimeDmx->gain_weight_m[L] >> 1; hEnhancedTimeDmx->prev_gain_m[R] = hEnhancedTimeDmx->gain_weight_m[R] >> 1; hEnhancedTimeDmx->prev_gain_e = hEnhancedTimeDmx->gain_weight_e + 1; hEnhancedTimeDmx->prev_H1_m[L] = scaleValue(hEnhancedTimeDmx->gain_weight_m[L], -4); hEnhancedTimeDmx->prev_H1_m[R] = scaleValue(hEnhancedTimeDmx->gain_weight_m[R], -4); hEnhancedTimeDmx->prev_H1_e = 2 + 2 + hEnhancedTimeDmx->gain_weight_e; } bail: return error; } FDK_SACENC_ERROR fdk_sacenc_apply_enhancedTimeDomainDmx( HANDLE_ENHANCED_TIME_DOMAIN_DMX hEnhancedTimeDmx, const INT_PCM *const *const inputTime, INT_PCM *const outputTimeDmx, const INT InputDelay) { FDK_SACENC_ERROR error = SACENC_OK; if ((NULL == hEnhancedTimeDmx) || (NULL == inputTime) || (NULL == inputTime[L]) || (NULL == inputTime[R]) || (NULL == outputTimeDmx)) { error = SACENC_INVALID_HANDLE; } else { int smp; FIXP_DBL lin_bbCld_m, lin_Cld_m, bbCorr_m, sqrt_linCld_m, G_m[2], H1_m[2], gainLeft_m, gainRight_m; FIXP_DBL bbNrgLeft_m, bbNrgRight_m, bbXNrg_m, nrgLeft_m, nrgRight_m, nrgX_m; INT lin_bbCld_e, lin_Cld_e, bbCorr_e, sqrt_linCld_e, G_e, H1_e; INT bbNrgLeft_e, bbNrgRight_e, bbXNrg_e, nrgLeft_e, nrgRight_e, nrgX_e; /* Increase energy time resolution with shorter processing blocks. 128 is an * empiric value. */ const int granuleLength = fixMin(128, hEnhancedTimeDmx->framelength); int granuleShift = (granuleLength > 1) ? ((DFRACT_BITS - 1) - fNorm((FIXP_DBL)(granuleLength - 1))) : 0; granuleShift = fixMax( 3, granuleShift + 1); /* one bit more headroom for worst case accumulation */ smp = 0; /* Prevent division by zero. */ bbNrgLeft_m = bbNrgRight_m = bbXNrg_m = (FIXP_DBL)(1); bbNrgLeft_e = bbNrgRight_e = bbXNrg_e = 0; do { const int offset = smp; FIXP_DBL partialL, partialR, partialX; partialL = partialR = partialX = FL2FXCONST_DBL(0.f); int in_margin = FDKmin( getScalefactorPCM( &inputTime[L][offset], fixMin(offset + granuleLength, hEnhancedTimeDmx->framelength) - offset, 1), getScalefactorPCM( &inputTime[R][offset], fixMin(offset + granuleLength, hEnhancedTimeDmx->framelength) - offset, 1)); /* partial energy */ for (smp = offset; smp < fixMin(offset + granuleLength, hEnhancedTimeDmx->framelength); smp++) { FIXP_PCM inputL = scaleValue((FIXP_PCM)inputTime[L][smp], in_margin - 1); FIXP_PCM inputR = scaleValue((FIXP_PCM)inputTime[R][smp], in_margin - 1); partialL += fPow2Div2(inputL) >> (granuleShift - 3); partialR += fPow2Div2(inputR) >> (granuleShift - 3); partialX += fMultDiv2(inputL, inputR) >> (granuleShift - 3); } fixpAddNorm(partialL, granuleShift - 2 * in_margin, &bbNrgLeft_m, &bbNrgLeft_e); fixpAddNorm(partialR, granuleShift - 2 * in_margin, &bbNrgRight_m, &bbNrgRight_e); fixpAddNorm(partialX, granuleShift - 2 * in_margin, &bbXNrg_m, &bbXNrg_e); } while (smp < hEnhancedTimeDmx->framelength); nrgLeft_m = fixpAdd(hEnhancedTimeDmx->prev_Left_m, hEnhancedTimeDmx->prev_Left_e, bbNrgLeft_m, bbNrgLeft_e, &nrgLeft_e); nrgRight_m = fixpAdd(hEnhancedTimeDmx->prev_Right_m, hEnhancedTimeDmx->prev_Right_e, bbNrgRight_m, bbNrgRight_e, &nrgRight_e); nrgX_m = fixpAdd(hEnhancedTimeDmx->prev_XNrg_m, hEnhancedTimeDmx->prev_XNrg_e, bbXNrg_m, bbXNrg_e, &nrgX_e); lin_bbCld_m = fMult(hEnhancedTimeDmx->lin_bbCld_weight_m, fDivNorm(nrgLeft_m, nrgRight_m, &lin_bbCld_e)); lin_bbCld_e += hEnhancedTimeDmx->lin_bbCld_weight_e + nrgLeft_e - nrgRight_e; bbCorr_m = fMult(nrgX_m, invSqrtNorm2(fMult(nrgLeft_m, nrgRight_m), nrgLeft_e + nrgRight_e, &bbCorr_e)); bbCorr_e += nrgX_e; hEnhancedTimeDmx->prev_Left_m = bbNrgLeft_m; hEnhancedTimeDmx->prev_Left_e = bbNrgLeft_e; hEnhancedTimeDmx->prev_Right_m = bbNrgRight_m; hEnhancedTimeDmx->prev_Right_e = bbNrgRight_e; hEnhancedTimeDmx->prev_XNrg_m = bbXNrg_m; hEnhancedTimeDmx->prev_XNrg_e = bbXNrg_e; /* bbCld = 10.f*log10(lin_bbCld) lin_Cld = pow(10,bbCld/20) = pow(10,10.f*log10(lin_bbCld)/20.f) = sqrt(lin_bbCld) lin_Cld2 = lin_Cld*lin_Cld = sqrt(lin_bbCld)*sqrt(lin_bbCld) = lin_bbCld */ lin_Cld_m = sqrtFixp(lin_bbCld_m, lin_bbCld_e, &lin_Cld_e); sqrt_linCld_m = sqrtFixp(lin_Cld_m, lin_Cld_e, &sqrt_linCld_e); /*calculate how much right and how much left signal, to avoid signal * cancellations*/ calculateRatio(sqrt_linCld_m, sqrt_linCld_e, lin_Cld_m, lin_Cld_e, bbCorr_m, bbCorr_e, G_m, &G_e); /*calculate downmix gains*/ calculateDmxGains(lin_Cld_m, lin_Cld_e, lin_bbCld_m, lin_bbCld_e, bbCorr_m, bbCorr_e, G_m, G_e, H1_m, &H1_e); /*adapt output gains*/ H1_m[L] = fMult(H1_m[L], hEnhancedTimeDmx->gain_weight_m[L]); H1_m[R] = fMult(H1_m[R], hEnhancedTimeDmx->gain_weight_m[R]); H1_e += hEnhancedTimeDmx->gain_weight_e; gainLeft_m = hEnhancedTimeDmx->prev_gain_m[L]; gainRight_m = hEnhancedTimeDmx->prev_gain_m[R]; INT intermediate_gain_e = +hEnhancedTimeDmx->sinusWindow_e + H1_e - hEnhancedTimeDmx->prev_gain_e; for (smp = 0; smp < hEnhancedTimeDmx->framelength; smp++) { const INT N = hEnhancedTimeDmx->framelength; FIXP_DBL intermediate_gainLeft_m, intermediate_gainRight_m, tmp; intermediate_gainLeft_m = scaleValue((fMult(hEnhancedTimeDmx->sinusWindow_m[smp], H1_m[L]) + fMult(hEnhancedTimeDmx->sinusWindow_m[N - smp], hEnhancedTimeDmx->prev_H1_m[L])), intermediate_gain_e); intermediate_gainRight_m = scaleValue((fMult(hEnhancedTimeDmx->sinusWindow_m[smp], H1_m[R]) + fMult(hEnhancedTimeDmx->sinusWindow_m[N - smp], hEnhancedTimeDmx->prev_H1_m[R])), intermediate_gain_e); gainLeft_m = intermediate_gainLeft_m + fMult(FL2FXCONST_DBL(1.f - ALPHA_FLT), gainLeft_m); gainRight_m = intermediate_gainRight_m + fMult(FL2FXCONST_DBL(1.f - ALPHA_FLT), gainRight_m); tmp = fMultDiv2(gainLeft_m, (FIXP_PCM)inputTime[L][smp + InputDelay]) + fMultDiv2(gainRight_m, (FIXP_PCM)inputTime[R][smp + InputDelay]); outputTimeDmx[smp] = (INT_PCM)SATURATE_SHIFT( tmp, -(hEnhancedTimeDmx->prev_gain_e + 1 - (DFRACT_BITS - SAMPLE_BITS)), SAMPLE_BITS); } hEnhancedTimeDmx->prev_gain_m[L] = gainLeft_m; hEnhancedTimeDmx->prev_gain_m[R] = gainRight_m; hEnhancedTimeDmx->prev_H1_m[L] = H1_m[L]; hEnhancedTimeDmx->prev_H1_m[R] = H1_m[R]; hEnhancedTimeDmx->prev_H1_e = H1_e; } return error; } static void calculateRatio(const FIXP_DBL sqrt_linCld_m, const INT sqrt_linCld_e, const FIXP_DBL lin_Cld_m, const INT lin_Cld_e, const FIXP_DBL Icc_m, const INT Icc_e, FIXP_DBL G_m[2], INT *G_e) { #define G_SCALE_FACTOR (2) if (Icc_m >= FL2FXCONST_DBL(0.f)) { G_m[0] = G_m[1] = FL2FXCONST_DBL(1.f / (float)(1 << G_SCALE_FACTOR)); G_e[0] = G_SCALE_FACTOR; } else { const FIXP_DBL max_gain_factor = FL2FXCONST_DBL(2.f / (float)(1 << G_SCALE_FACTOR)); FIXP_DBL tmp1_m, tmp2_m, numerator_m, denominator_m, r_m, r4_m, q; INT tmp1_e, tmp2_e, numerator_e, denominator_e, r_e, r4_e; /* r = (lin_Cld + 1 + 2*Icc*sqrt_linCld) / (lin_Cld + 1 - * 2*Icc*sqrt_linCld) = (tmp1 + tmp2) / (tmp1 - tmp2) */ tmp1_m = fixpAdd(lin_Cld_m, lin_Cld_e, FL2FXCONST_DBL(1.f / 2.f), 1, &tmp1_e); tmp2_m = fMult(Icc_m, sqrt_linCld_m); tmp2_e = 1 + Icc_e + sqrt_linCld_e; numerator_m = fixpAdd(tmp1_m, tmp1_e, tmp2_m, tmp2_e, &numerator_e); denominator_m = fixpAdd(tmp1_m, tmp1_e, -tmp2_m, tmp2_e, &denominator_e); if ((numerator_m > FL2FXCONST_DBL(0.f)) && (denominator_m > FL2FXCONST_DBL(0.f))) { r_m = fDivNorm(numerator_m, denominator_m, &r_e); r_e += numerator_e - denominator_e; /* r_4 = sqrt( sqrt( r ) ) */ r4_m = sqrtFixp(r_m, r_e, &r4_e); r4_m = sqrtFixp(r4_m, r4_e, &r4_e); r4_e -= G_SCALE_FACTOR; /* q = min(r4_m, max_gain_factor) */ q = ((r4_e >= 0) && (r4_m >= (max_gain_factor >> r4_e))) ? max_gain_factor : scaleValue(r4_m, r4_e); } else { q = FL2FXCONST_DBL(0.f); } G_m[0] = max_gain_factor - q; G_m[1] = q; *G_e = G_SCALE_FACTOR; } } static void calculateDmxGains(const FIXP_DBL lin_Cld_m, const INT lin_Cld_e, const FIXP_DBL lin_Cld2_m, const INT lin_Cld2_e, const FIXP_DBL Icc_m, const INT Icc_e, const FIXP_DBL G_m[2], const INT G_e, FIXP_DBL H1_m[2], INT *pH1_e) { #define H1_SCALE_FACTOR (2) const FIXP_DBL max_gain_factor = FL2FXCONST_DBL(2.f / (float)(1 << H1_SCALE_FACTOR)); FIXP_DBL nrgRight_m, nrgLeft_m, crossNrg_m, inv_weight_num_m, inv_weight_denom_m, inverse_weight_m, inverse_weight_limited; INT nrgRight_e, nrgLeft_e, crossNrg_e, inv_weight_num_e, inv_weight_denom_e, inverse_weight_e; /* nrgRight = sqrt(1/(lin_Cld2 + 1) */ nrgRight_m = fixpAdd(lin_Cld2_m, lin_Cld2_e, FL2FXCONST_DBL(1.f / 2.f), 1, &nrgRight_e); nrgRight_m = invSqrtNorm2(nrgRight_m, nrgRight_e, &nrgRight_e); /* nrgLeft = lin_Cld * nrgRight */ nrgLeft_m = fMult(lin_Cld_m, nrgRight_m); nrgLeft_e = lin_Cld_e + nrgRight_e; /* crossNrg = sqrt(nrgLeft*nrgRight) */ crossNrg_m = sqrtFixp(fMult(nrgLeft_m, nrgRight_m), nrgLeft_e + nrgRight_e, &crossNrg_e); /* inverse_weight = sqrt((nrgLeft + nrgRight) / ( (G[0]*G[0]*nrgLeft) + * (G[1]*G[1]*nrgRight) + 2*G[0]*G[1]*Icc*crossNrg)) = sqrt(inv_weight_num / * inv_weight_denom) */ inv_weight_num_m = fixpAdd(nrgRight_m, nrgRight_e, nrgLeft_m, nrgLeft_e, &inv_weight_num_e); inv_weight_denom_m = fixpAdd(fMult(fPow2(G_m[0]), nrgLeft_m), 2 * G_e + nrgLeft_e, fMult(fPow2(G_m[1]), nrgRight_m), 2 * G_e + nrgRight_e, &inv_weight_denom_e); inv_weight_denom_m = fixpAdd(fMult(fMult(fMult(G_m[0], G_m[1]), crossNrg_m), Icc_m), 1 + 2 * G_e + crossNrg_e + Icc_e, inv_weight_denom_m, inv_weight_denom_e, &inv_weight_denom_e); if (inv_weight_denom_m > FL2FXCONST_DBL(0.f)) { inverse_weight_m = fDivNorm(inv_weight_num_m, inv_weight_denom_m, &inverse_weight_e); inverse_weight_m = sqrtFixp(inverse_weight_m, inverse_weight_e + inv_weight_num_e - inv_weight_denom_e, &inverse_weight_e); inverse_weight_e -= H1_SCALE_FACTOR; /* inverse_weight_limited = min(max_gain_factor, inverse_weight) */ inverse_weight_limited = ((inverse_weight_e >= 0) && (inverse_weight_m >= (max_gain_factor >> inverse_weight_e))) ? max_gain_factor : scaleValue(inverse_weight_m, inverse_weight_e); } else { inverse_weight_limited = max_gain_factor; } H1_m[0] = fMult(G_m[0], inverse_weight_limited); H1_m[1] = fMult(G_m[1], inverse_weight_limited); *pH1_e = G_e + H1_SCALE_FACTOR; } FDK_SACENC_ERROR fdk_sacenc_close_enhancedTimeDomainDmx( HANDLE_ENHANCED_TIME_DOMAIN_DMX *phEnhancedTimeDmx) { FDK_SACENC_ERROR error = SACENC_OK; if (phEnhancedTimeDmx == NULL) { error = SACENC_INVALID_HANDLE; } else { if (*phEnhancedTimeDmx != NULL) { if ((*phEnhancedTimeDmx)->sinusWindow_m != NULL) { FDK_FREE_MEMORY_1D((*phEnhancedTimeDmx)->sinusWindow_m); } FDK_FREE_MEMORY_1D(*phEnhancedTimeDmx); } } return error; } fdk-aac-2.0.2/libSACenc/src/sacenc_dmx_tdom_enh.h000066400000000000000000000140601404225310700215040ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround encoder library ************************* Author(s): M. Luis Valero Description: Enhanced Time Domain Downmix *******************************************************************************/ #ifndef SACENC_DMX_TDOM_ENH_H #define SACENC_DMX_TDOM_ENH_H /* Includes ******************************************************************/ #include "sacenc_lib.h" #include "common_fix.h" /* Defines *******************************************************************/ /* Data Types ****************************************************************/ typedef struct T_ENHANCED_TIME_DOMAIN_DMX *HANDLE_ENHANCED_TIME_DOMAIN_DMX; /* Constants *****************************************************************/ /* Function / Class Declarations *********************************************/ FDK_SACENC_ERROR fdk_sacenc_open_enhancedTimeDomainDmx( HANDLE_ENHANCED_TIME_DOMAIN_DMX *hEnhancedTimeDmx, const INT framelength); FDK_SACENC_ERROR fdk_sacenc_init_enhancedTimeDomainDmx( HANDLE_ENHANCED_TIME_DOMAIN_DMX hEnhancedTimeDmx, const FIXP_DBL *const pInputGain_m, const INT inputGain_e, const FIXP_DBL outputGain_m, const INT outputGain_e, const INT framelength); FDK_SACENC_ERROR fdk_sacenc_apply_enhancedTimeDomainDmx( HANDLE_ENHANCED_TIME_DOMAIN_DMX hEnhancedTimeDmx, const INT_PCM *const *const inputTime, INT_PCM *const outputTimeDmx, const INT InputDelay); FDK_SACENC_ERROR fdk_sacenc_close_enhancedTimeDomainDmx( HANDLE_ENHANCED_TIME_DOMAIN_DMX *hEnhancedTimeDmx); #endif /* SACENC_DMX_TDOM_ENH_H */ fdk-aac-2.0.2/libSACenc/src/sacenc_filter.cpp000066400000000000000000000172441404225310700206660ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround encoder library ************************* Author(s): M. Multrus Description: Encoder Library Filter functions *******************************************************************************/ /* Includes ******************************************************************/ #include "sacenc_filter.h" /* Defines *******************************************************************/ /* Data Types ****************************************************************/ typedef struct T_DC_FILTER { FIXP_DBL c__FDK; FIXP_DBL state__FDK; } DC_FILTER; /* Constants *****************************************************************/ /* Function / Class Declarations *********************************************/ /* Function / Class Definition ***********************************************/ FDK_SACENC_ERROR fdk_sacenc_createDCFilter(HANDLE_DC_FILTER *hDCFilter) { FDK_SACENC_ERROR error = SACENC_OK; if (NULL == hDCFilter) { error = SACENC_INVALID_HANDLE; } else { FDK_ALLOCATE_MEMORY_1D(*hDCFilter, 1, DC_FILTER); } return error; bail: fdk_sacenc_destroyDCFilter(hDCFilter); return ((SACENC_OK == error) ? SACENC_MEMORY_ERROR : error); } FDK_SACENC_ERROR fdk_sacenc_initDCFilter(HANDLE_DC_FILTER hDCFilter, const UINT sampleRate) { FDK_SACENC_ERROR error = SACENC_OK; FIXP_DBL expC; int s; /* Conversion for use of CalcInvLdData: e^x = 2^(x*log10(e)/log10(2) = CalcInvLdData(x*log10(e)/log10(2)/64.0) 1.44269504089 = log10(e)/log10(2) 0.5 = scale constant value with 1 Bits */ expC = fDivNormHighPrec((FIXP_DBL)20, (FIXP_DBL)sampleRate, &s); expC = fMultDiv2(FL2FXCONST_DBL(-1.44269504089 * 0.5), expC) >> (LD_DATA_SHIFT - 1 - 1); if (s < 0) expC = expC >> (-s); else expC = expC << (s); expC = CalcInvLdData(expC); hDCFilter->c__FDK = expC; hDCFilter->state__FDK = FL2FXCONST_DBL(0.0f); return error; } FDK_SACENC_ERROR fdk_sacenc_destroyDCFilter(HANDLE_DC_FILTER *hDCFilter) { FDK_SACENC_ERROR error = SACENC_OK; if ((hDCFilter != NULL) && (*hDCFilter != NULL)) { FDKfree(*hDCFilter); *hDCFilter = NULL; } return error; } FDK_SACENC_ERROR fdk_sacenc_applyDCFilter(HANDLE_DC_FILTER hDCFilter, const INT_PCM *const signalIn, INT_PCM *const signalOut, const INT signalLength) { FDK_SACENC_ERROR error = SACENC_OK; if ((hDCFilter == NULL) || (signalIn == NULL) || (signalOut == NULL)) { error = SACENC_INVALID_HANDLE; } else { const INT_PCM *const x = signalIn; INT_PCM *const y = signalOut; const FIXP_DBL c = hDCFilter->c__FDK; FIXP_DBL *const state = &hDCFilter->state__FDK; int i; FIXP_DBL x0, x1, y1; x1 = x0 = FX_PCM2FX_DBL(x[0]) >> DC_FILTER_SF; y1 = x0 + (*state); for (i = 1; i < signalLength; i++) { x0 = FX_PCM2FX_DBL(x[i]) >> DC_FILTER_SF; y[i - 1] = FX_DBL2FX_PCM(y1); y1 = x0 - x1 + fMult(c, y1); x1 = x0; } *state = fMult(c, y1) - x1; y[i - 1] = FX_DBL2FX_PCM(y1); } return error; } fdk-aac-2.0.2/libSACenc/src/sacenc_filter.h000066400000000000000000000136651404225310700203360ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround encoder library ************************* Author(s): M. Multrus Description: Encoder Library Interface Filter functions *******************************************************************************/ #ifndef SACENC_FILTER_H #define SACENC_FILTER_H /* Includes ******************************************************************/ #include "common_fix.h" #include "sacenc_lib.h" #include "FDK_matrixCalloc.h" /* Defines *******************************************************************/ #define DC_FILTER_SF 1 /* Data Types ****************************************************************/ typedef struct T_DC_FILTER *HANDLE_DC_FILTER; /* Constants *****************************************************************/ /* Function / Class Declarations *********************************************/ FDK_SACENC_ERROR fdk_sacenc_createDCFilter(HANDLE_DC_FILTER *hDCFilter); FDK_SACENC_ERROR fdk_sacenc_initDCFilter(HANDLE_DC_FILTER hDCFilter, const UINT sampleRate); FDK_SACENC_ERROR fdk_sacenc_destroyDCFilter(HANDLE_DC_FILTER *hDCFilter); FDK_SACENC_ERROR fdk_sacenc_applyDCFilter(HANDLE_DC_FILTER hDCFilter, const INT_PCM *const signalIn, INT_PCM *const signalOut, const INT signalLength); #endif /* SACENC_FILTER_H */ fdk-aac-2.0.2/libSACenc/src/sacenc_framewindowing.cpp000066400000000000000000000472071404225310700224230ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround encoder library ************************* Author(s): Max Neuendorf Description: Encoder Library Interface Get windows for framing *******************************************************************************/ /**************************************************************************/ /** \file Description of file contents ******************************************************************************/ /* Includes ******************************************************************/ #include "sacenc_framewindowing.h" #include "sacenc_vectorfunctions.h" /* Defines *******************************************************************/ /* Data Types ****************************************************************/ typedef struct T_FRAMEWINDOW { INT nTimeSlotsMax; INT bFrameKeep; INT startSlope; INT stopSlope; INT startRect; INT stopRect; INT taperAnaLen; INT taperSynLen; FIXP_WIN pTaperAna__FDK[MAX_TIME_SLOTS]; FIXP_WIN pTaperSyn__FDK[MAX_TIME_SLOTS]; } FRAMEWINDOW; typedef enum { FIX_INVALID = -1, FIX_RECT_SMOOTH = 0, FIX_SMOOTH_RECT = 1, FIX_LARGE_SMOOTH = 2, FIX_RECT_TRIANG = 3 } FIX_TYPE; typedef enum { VAR_INVALID = -1, VAR_HOLD = 0, VAR_ISOLATE = 1 } VAR_TYPE; /* Constants *****************************************************************/ /* Function / Class Declarations *********************************************/ /* Function / Class Definition ***********************************************/ static void calcTaperWin(FIXP_WIN *pTaperWin, INT timeSlots) { FIXP_DBL x; int i, scale; for (i = 0; i < timeSlots; i++) { x = fDivNormHighPrec((FIXP_DBL)i, (FIXP_DBL)timeSlots, &scale); if (scale < 0) { pTaperWin[i] = FX_DBL2FX_WIN(x >> (-scale)); } else { pTaperWin[i] = FX_DBL2FX_WIN(x << (scale)); } } pTaperWin[timeSlots] = FX_DBL2FX_WIN((FIXP_DBL)MAXVAL_DBL); } FDK_SACENC_ERROR fdk_sacenc_frameWindow_Create( HANDLE_FRAMEWINDOW *phFrameWindow) { FDK_SACENC_ERROR error = SACENC_OK; if (NULL == phFrameWindow) { error = SACENC_INVALID_HANDLE; } else { /* Memory Allocation */ FDK_ALLOCATE_MEMORY_1D(*phFrameWindow, 1, FRAMEWINDOW); } return error; bail: fdk_sacenc_frameWindow_Destroy(phFrameWindow); return ((SACENC_OK == error) ? SACENC_MEMORY_ERROR : error); } FDK_SACENC_ERROR fdk_sacenc_frameWindow_Init( HANDLE_FRAMEWINDOW hFrameWindow, const FRAMEWINDOW_CONFIG *const pFrameWindowConfig) { FDK_SACENC_ERROR error = SACENC_OK; if ((hFrameWindow == NULL) || (pFrameWindowConfig == NULL)) { error = SACENC_INVALID_HANDLE; } else if (pFrameWindowConfig->nTimeSlotsMax < 0) { error = SACENC_INIT_ERROR; } else { int ts; hFrameWindow->bFrameKeep = pFrameWindowConfig->bFrameKeep; hFrameWindow->nTimeSlotsMax = pFrameWindowConfig->nTimeSlotsMax; FIXP_WIN winMaxVal = FX_DBL2FX_WIN((FIXP_DBL)MAXVAL_DBL); int timeSlots = pFrameWindowConfig->nTimeSlotsMax; { hFrameWindow->startSlope = 0; hFrameWindow->stopSlope = ((3 * timeSlots) >> 1) - 1; hFrameWindow->startRect = timeSlots >> 1; hFrameWindow->stopRect = timeSlots; calcTaperWin(hFrameWindow->pTaperSyn__FDK, timeSlots >> 1); hFrameWindow->taperSynLen = timeSlots >> 1; } /* Calculate Taper for non-rect. ana. windows */ hFrameWindow->taperAnaLen = hFrameWindow->startRect - hFrameWindow->startSlope; for (ts = 0; ts < hFrameWindow->taperAnaLen; ts++) { { hFrameWindow->pTaperAna__FDK[ts] = winMaxVal; } } } return error; } FDK_SACENC_ERROR fdk_sacenc_frameWindow_Destroy( HANDLE_FRAMEWINDOW *phFrameWindow) { FDK_SACENC_ERROR error = SACENC_OK; if ((NULL != phFrameWindow) && (NULL != *phFrameWindow)) { FDKfree(*phFrameWindow); *phFrameWindow = NULL; } return error; } static FDK_SACENC_ERROR FrameWinList_Reset(FRAMEWIN_LIST *const pFrameWinList) { FDK_SACENC_ERROR error = SACENC_OK; if (NULL == pFrameWinList) { error = SACENC_INVALID_HANDLE; } else { int k = 0; for (k = 0; k < MAX_NUM_PARAMS; k++) { pFrameWinList->dat[k].slot = -1; pFrameWinList->dat[k].hold = FW_INTP; } pFrameWinList->n = 0; } return error; } static FDK_SACENC_ERROR FrameWindowList_Add(FRAMEWIN_LIST *const pFrameWinList, const INT slot, const FW_SLOTTYPE hold) { FDK_SACENC_ERROR error = SACENC_OK; if (NULL == pFrameWinList) { error = SACENC_INVALID_HANDLE; } else { if (pFrameWinList->n >= MAX_NUM_PARAMS) { /* Place left in List ?*/ error = SACENC_PARAM_ERROR; } else if (pFrameWinList->n > 0 && pFrameWinList->dat[pFrameWinList->n - 1].slot - slot > 0) { error = SACENC_PARAM_ERROR; } else { pFrameWinList->dat[pFrameWinList->n].slot = slot; pFrameWinList->dat[pFrameWinList->n].hold = hold; pFrameWinList->n++; } } return error; } static FDK_SACENC_ERROR FrameWindowList_Remove( FRAMEWIN_LIST *const pFrameWinList, const INT idx) { FDK_SACENC_ERROR error = SACENC_OK; if (NULL == pFrameWinList) { error = SACENC_INVALID_HANDLE; } else { int k = 0; if (idx < 0 || idx >= MAX_NUM_PARAMS) { error = SACENC_PARAM_ERROR; } else if (pFrameWinList->n > 0) { if (idx == MAX_NUM_PARAMS - 1) { pFrameWinList->dat[idx].slot = -1; pFrameWinList->dat[idx].hold = FW_INTP; } else { for (k = idx; k < MAX_NUM_PARAMS - 1; k++) { pFrameWinList->dat[k] = pFrameWinList->dat[k + 1]; } } pFrameWinList->n--; } } return error; } static FDK_SACENC_ERROR FrameWindowList_Limit( FRAMEWIN_LIST *const pFrameWinList, const INT ll /*lower limit*/, const INT ul /*upper limit*/ ) { FDK_SACENC_ERROR error = SACENC_OK; if (NULL == pFrameWinList) { error = SACENC_INVALID_HANDLE; } else { int k = 0; for (k = 0; k < pFrameWinList->n; k++) { if (pFrameWinList->dat[k].slot < ll || pFrameWinList->dat[k].slot > ul) { FrameWindowList_Remove(pFrameWinList, k); --k; } } } return error; } FDK_SACENC_ERROR fdk_sacenc_frameWindow_GetWindow( HANDLE_FRAMEWINDOW hFrameWindow, INT tr_pos[MAX_NUM_PARAMS], const INT timeSlots, FRAMINGINFO *const pFramingInfo, FIXP_WIN *pWindowAna__FDK[MAX_NUM_PARAMS], FRAMEWIN_LIST *const pFrameWinList, const INT avoid_keep) { FDK_SACENC_ERROR error = SACENC_OK; if ((hFrameWindow == NULL) || (tr_pos == NULL) || (pFramingInfo == NULL) || (pFrameWinList == NULL) || (pWindowAna__FDK == NULL)) { error = SACENC_INVALID_HANDLE; } else { const VAR_TYPE varType = VAR_HOLD; const int tranL = 4; int winCnt = 0; int w, ps; int startSlope = hFrameWindow->startSlope; int stopSlope = hFrameWindow->stopSlope; int startRect = hFrameWindow->startRect; int stopRect = hFrameWindow->stopRect; int taperAnaLen = hFrameWindow->taperAnaLen; FIXP_WIN winMaxVal = FX_DBL2FX_WIN((FIXP_DBL)MAXVAL_DBL); FIXP_WIN applyRightWindowGain__FDK[MAX_NUM_PARAMS]; FIXP_WIN *pTaperAna__FDK = hFrameWindow->pTaperAna__FDK; /* sanity check */ for (ps = 0; ps < MAX_NUM_PARAMS; ps++) { if (pWindowAna__FDK[ps] == NULL) { error = SACENC_INVALID_HANDLE; goto bail; } } if ((timeSlots > hFrameWindow->nTimeSlotsMax) || (timeSlots < 0)) { error = SACENC_INVALID_CONFIG; goto bail; } /* Reset */ if (SACENC_OK != (error = FrameWinList_Reset(pFrameWinList))) goto bail; FDKmemclear(applyRightWindowGain__FDK, sizeof(applyRightWindowGain__FDK)); if (tr_pos[0] > -1) { /* Transients in first (left) half? */ int p_l = tr_pos[0]; winCnt = 0; /* Create Parameter Positions */ switch (varType) { case VAR_HOLD: if (SACENC_OK != (error = FrameWindowList_Add(pFrameWinList, p_l - 1, FW_HOLD))) goto bail; if (SACENC_OK != (error = FrameWindowList_Add(pFrameWinList, p_l, FW_INTP))) goto bail; break; case VAR_ISOLATE: if (SACENC_OK != (error = FrameWindowList_Add(pFrameWinList, p_l - 1, FW_HOLD))) goto bail; if (SACENC_OK != (error = FrameWindowList_Add(pFrameWinList, p_l, FW_INTP))) goto bail; if (SACENC_OK != (error = FrameWindowList_Add(pFrameWinList, p_l + tranL, FW_HOLD))) goto bail; if (SACENC_OK != (error = FrameWindowList_Add( pFrameWinList, p_l + tranL + 1, FW_INTP))) goto bail; break; default: error = SACENC_INVALID_CONFIG; break; } /* Outside of frame? => Kick Out */ if (SACENC_OK != (error = FrameWindowList_Limit(pFrameWinList, 0, timeSlots - 1))) goto bail; /* Add timeSlots as temporary border for window creation */ if (SACENC_OK != (error = FrameWindowList_Add(pFrameWinList, timeSlots - 1, FW_HOLD))) goto bail; /* Create Windows */ for (ps = 0; ps < pFrameWinList->n - 1; ps++) { if (FW_HOLD != pFrameWinList->dat[ps].hold) { int const start = pFrameWinList->dat[ps].slot; int const stop = pFrameWinList->dat[ps + 1].slot; /* Analysis Window */ FDKmemset_flex(pWindowAna__FDK[winCnt], FX_DBL2FX_WIN((FIXP_DBL)0), start); FDKmemset_flex(&pWindowAna__FDK[winCnt][start], winMaxVal, stop - start + 1); FDKmemset_flex(&pWindowAna__FDK[winCnt][stop + 1], FX_DBL2FX_WIN((FIXP_DBL)0), timeSlots - stop - 1); applyRightWindowGain__FDK[winCnt] = pWindowAna__FDK[winCnt][timeSlots - 1]; winCnt++; } } /* ps */ /* Pop temporary frame border */ if (SACENC_OK != (error = FrameWindowList_Remove(pFrameWinList, pFrameWinList->n - 1))) goto bail; } else { /* No transient in left half of ana. window */ winCnt = 0; /* Add paramter set at end of frame */ if (SACENC_OK != (error = FrameWindowList_Add(pFrameWinList, timeSlots - 1, FW_INTP))) goto bail; /* Analysis Window */ FDKmemset_flex(pWindowAna__FDK[winCnt], FX_DBL2FX_WIN((FIXP_DBL)0), startSlope); FDKmemcpy_flex(&pWindowAna__FDK[winCnt][startSlope], 1, pTaperAna__FDK, 1, taperAnaLen); FDKmemset_flex(&pWindowAna__FDK[winCnt][startRect], winMaxVal, timeSlots - startRect); applyRightWindowGain__FDK[winCnt] = winMaxVal; winCnt++; } /* if (tr_pos[0] > -1) */ for (w = 0; w < winCnt; w++) { if (applyRightWindowGain__FDK[w] > (FIXP_WIN)0) { if (tr_pos[1] > -1) { /* Transients in second (right) half? */ int p_r = tr_pos[1]; /* Analysis Window */ FDKmemset_flex(&pWindowAna__FDK[w][timeSlots], winMaxVal, p_r - timeSlots); FDKmemset_flex(&pWindowAna__FDK[w][p_r], FX_DBL2FX_WIN((FIXP_DBL)0), 2 * timeSlots - p_r); } else { /* No transient in right half of ana. window */ /* Analysis Window */ FDKmemset_flex(&pWindowAna__FDK[w][timeSlots], winMaxVal, stopRect - timeSlots + 1); FDKmemcpy_flex(&pWindowAna__FDK[w][stopRect], 1, &pTaperAna__FDK[taperAnaLen - 1], -1, taperAnaLen); FDKmemset_flex(&pWindowAna__FDK[w][stopSlope + 1], FX_DBL2FX_WIN((FIXP_DBL)0), 2 * timeSlots - stopSlope - 1); } /* if (tr_pos[1] > -1) */ /* Weight */ if (applyRightWindowGain__FDK[w] < winMaxVal) { int ts; for (ts = 0; ts < timeSlots; ts++) { pWindowAna__FDK[w][timeSlots + ts] = FX_DBL2FX_WIN(fMult(pWindowAna__FDK[w][timeSlots + ts], applyRightWindowGain__FDK[w])); } } } /* if (applyRightWindowGain[w] > 0.0f) */ else { /* All Zero */ FDKmemset_flex(&pWindowAna__FDK[w][timeSlots], FX_DBL2FX_WIN((FIXP_DBL)0), timeSlots); } } /* loop over windows */ if (hFrameWindow->bFrameKeep == 1) { FDKmemcpy_flex(&pWindowAna__FDK[0][2 * timeSlots], 1, &pWindowAna__FDK[0][timeSlots], 1, timeSlots); FDKmemcpy_flex(&pWindowAna__FDK[0][timeSlots], 1, pWindowAna__FDK[0], 1, timeSlots); if (avoid_keep != 0) { FDKmemset_flex(pWindowAna__FDK[0], FX_DBL2FX_WIN((FIXP_DBL)0), timeSlots); } else { FDKmemset_flex(pWindowAna__FDK[0], winMaxVal, timeSlots); } } /* if (hFrameWindow->bFrameKeep==1) */ /* Feed Info to Bitstream Formatter */ pFramingInfo->numParamSets = pFrameWinList->n; pFramingInfo->bsFramingType = 1; /* variable framing */ for (ps = 0; ps < pFramingInfo->numParamSets; ps++) { pFramingInfo->bsParamSlots[ps] = pFrameWinList->dat[ps].slot; } /* if there is just one param set at last slot, use fixed framing to save some bits */ if ((pFramingInfo->numParamSets == 1) && (pFramingInfo->bsParamSlots[0] == timeSlots - 1)) { pFramingInfo->bsFramingType = 0; } } /* valid handle */ bail: return error; } FDK_SACENC_ERROR fdk_sacenc_analysisWindowing( const INT nTimeSlots, const INT startTimeSlot, FIXP_WIN *pFrameWindowAna__FDK, const FIXP_DPK *const *const ppDataIn__FDK, FIXP_DPK *const *const ppDataOut__FDK, const INT nHybridBands, const INT dim) { FDK_SACENC_ERROR error = SACENC_OK; if ((pFrameWindowAna__FDK == NULL) || (ppDataIn__FDK == NULL) || (ppDataOut__FDK == NULL)) { error = SACENC_INVALID_HANDLE; } else { int i, ts; FIXP_WIN maxVal = FX_DBL2FX_WIN((FIXP_DBL)MAXVAL_DBL); if (dim == FW_CHANGE_DIM) { for (ts = startTimeSlot; ts < nTimeSlots; ts++) { FIXP_WIN win = pFrameWindowAna__FDK[ts]; if (win == maxVal) { for (i = 0; i < nHybridBands; i++) { ppDataOut__FDK[i][ts].v.re = ppDataIn__FDK[ts][i].v.re; ppDataOut__FDK[i][ts].v.im = ppDataIn__FDK[ts][i].v.im; } } else { for (i = 0; i < nHybridBands; i++) { ppDataOut__FDK[i][ts].v.re = fMult(win, ppDataIn__FDK[ts][i].v.re); ppDataOut__FDK[i][ts].v.im = fMult(win, ppDataIn__FDK[ts][i].v.im); } } } /* ts */ } else { for (ts = startTimeSlot; ts < nTimeSlots; ts++) { FIXP_WIN win = pFrameWindowAna__FDK[ts]; if (win == maxVal) { for (i = 0; i < nHybridBands; i++) { ppDataOut__FDK[ts][i].v.re = ppDataIn__FDK[ts][i].v.re; ppDataOut__FDK[ts][i].v.im = ppDataIn__FDK[ts][i].v.im; } } else { for (i = 0; i < nHybridBands; i++) { ppDataOut__FDK[ts][i].v.re = fMult(win, ppDataIn__FDK[ts][i].v.re); ppDataOut__FDK[ts][i].v.im = fMult(win, ppDataIn__FDK[ts][i].v.im); } } } /* ts */ } } return error; } fdk-aac-2.0.2/libSACenc/src/sacenc_framewindowing.h000066400000000000000000000157001404225310700220610ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround encoder library ************************* Author(s): Max Neuendorf Description: Encoder Library Interface Get windows for framing *******************************************************************************/ #ifndef SACENC_FRAMEWINDOWING_H #define SACENC_FRAMEWINDOWING_H /**************************************************************************/ /** \file Description of file contents ******************************************************************************/ /* Includes ******************************************************************/ #include "genericStds.h" #include "common_fix.h" #include "sacenc_lib.h" #include "sacenc_bitstream.h" /* Defines *******************************************************************/ #define FIXP_WIN FIXP_DBL #define FX_DBL2FX_WIN(x) (x) #define DALDATATYPE_WIN DALDATATYPE_DFRACT typedef enum { FW_INTP = 0, FW_HOLD = 1 } FW_SLOTTYPE; typedef enum { FW_LEAVE_DIM = 0, FW_CHANGE_DIM = 1 } FW_DIMENSION; /* Data Types ****************************************************************/ typedef struct T_FRAMEWINDOW *HANDLE_FRAMEWINDOW; typedef struct T_FRAMEWINDOW_CONFIG { INT nTimeSlotsMax; INT bFrameKeep; } FRAMEWINDOW_CONFIG; typedef struct { INT slot; FW_SLOTTYPE hold; } FRAMEWIN_DATA; typedef struct { FRAMEWIN_DATA dat[MAX_NUM_PARAMS]; INT n; } FRAMEWIN_LIST; /* Constants *****************************************************************/ /* Function / Class Declarations *********************************************/ FDK_SACENC_ERROR fdk_sacenc_frameWindow_Create( HANDLE_FRAMEWINDOW *phFrameWindow); FDK_SACENC_ERROR fdk_sacenc_frameWindow_Init( HANDLE_FRAMEWINDOW hFrameWindow, const FRAMEWINDOW_CONFIG *const pFrameWindowConfig); FDK_SACENC_ERROR fdk_sacenc_frameWindow_Destroy( HANDLE_FRAMEWINDOW *phFrameWindow); FDK_SACENC_ERROR fdk_sacenc_frameWindow_GetWindow( HANDLE_FRAMEWINDOW hFrameWindow, INT tr_pos[MAX_NUM_PARAMS], const INT timeSlots, FRAMINGINFO *const pFramingInfo, FIXP_WIN *pWindowAna__FDK[MAX_NUM_PARAMS], FRAMEWIN_LIST *const pFrameWinList, const INT avoid_keep); FDK_SACENC_ERROR fdk_sacenc_analysisWindowing( const INT nTimeSlots, const INT startTimeSlot, FIXP_WIN *pFrameWindowAna__FDK, const FIXP_DPK *const *const ppDataIn__FDK, FIXP_DPK *const *const ppDataOut__FDK, const INT nHybridBands, const INT dim); #endif /* SACENC_FRAMEWINDOWING_H */ fdk-aac-2.0.2/libSACenc/src/sacenc_huff_tab.cpp000066400000000000000000001572151404225310700211620ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround encoder library ************************* Author(s): Markus Lohwasser Description: SAC-Encoder constant huffman tables *******************************************************************************/ /* Includes ******************************************************************/ #include "sacenc_huff_tab.h" /* Defines *******************************************************************/ /* Data Types ****************************************************************/ /* Constants *****************************************************************/ const HUFF_CLD_TABLE fdk_sacenc_huffCLDTab = { {/* h1D[2][31] */ {HUFF_PACK(0x00000000, 1), HUFF_PACK(0x00000002, 2), HUFF_PACK(0x00000006, 3), HUFF_PACK(0x0000000e, 4), HUFF_PACK(0x0000001e, 5), HUFF_PACK(0x0000003e, 6), HUFF_PACK(0x0000007e, 7), HUFF_PACK(0x000000fe, 8), HUFF_PACK(0x000001fe, 9), HUFF_PACK(0x000003fe, 10), HUFF_PACK(0x000007fe, 11), HUFF_PACK(0x00000ffe, 12), HUFF_PACK(0x00001ffe, 13), HUFF_PACK(0x00007ffe, 15), HUFF_PACK(0x00007ffc, 15), HUFF_PACK(0x0000fffe, 16), HUFF_PACK(0x0000fffa, 16), HUFF_PACK(0x0001fffe, 17), HUFF_PACK(0x0001fff6, 17), HUFF_PACK(0x0003fffe, 18), HUFF_PACK(0x0003ffff, 18), HUFF_PACK(0x0007ffde, 19), HUFF_PACK(0x0003ffee, 18), HUFF_PACK(0x000fffbe, 20), HUFF_PACK(0x001fff7e, 21), HUFF_PACK(0x00fffbfc, 24), HUFF_PACK(0x00fffbfd, 24), HUFF_PACK(0x00fffbfe, 24), HUFF_PACK(0x00fffbff, 24), HUFF_PACK(0x007ffdfc, 23), HUFF_PACK(0x007ffdfd, 23)}, {HUFF_PACK(0x00000000, 1), HUFF_PACK(0x00000002, 2), HUFF_PACK(0x00000006, 3), HUFF_PACK(0x0000000e, 4), HUFF_PACK(0x0000001e, 5), HUFF_PACK(0x0000003e, 6), HUFF_PACK(0x0000007e, 7), HUFF_PACK(0x000001fe, 9), HUFF_PACK(0x000001fc, 9), HUFF_PACK(0x000003fe, 10), HUFF_PACK(0x000003fa, 10), HUFF_PACK(0x000007fe, 11), HUFF_PACK(0x000007f6, 11), HUFF_PACK(0x00000ffe, 12), HUFF_PACK(0x00000fee, 12), HUFF_PACK(0x00001ffe, 13), HUFF_PACK(0x00001fde, 13), HUFF_PACK(0x00003ffe, 14), HUFF_PACK(0x00003fbe, 14), HUFF_PACK(0x00003fbf, 14), HUFF_PACK(0x00007ffe, 15), HUFF_PACK(0x0000fffe, 16), HUFF_PACK(0x0001fffe, 17), HUFF_PACK(0x0007fffe, 19), HUFF_PACK(0x0007fffc, 19), HUFF_PACK(0x000ffffa, 20), HUFF_PACK(0x001ffffc, 21), HUFF_PACK(0x001ffffd, 21), HUFF_PACK(0x001ffffe, 21), HUFF_PACK(0x001fffff, 21), HUFF_PACK(0x000ffffb, 20)}}, { /* HUFF_CLD_TAB_2D */ { /* HUFF_CLD_TAB_2D[0][] */ {/* HUFF_CLD_TAB_2D[0][0] */ { /* LAV3_2D */ {{HUFF_PACK(0x00000002, 2), HUFF_PACK(0x00000002, 3), HUFF_PACK(0x00000004, 5), HUFF_PACK(0x0000003e, 8)}, {HUFF_PACK(0x00000006, 4), HUFF_PACK(0x00000007, 4), HUFF_PACK(0x0000000e, 6), HUFF_PACK(0x000000fe, 10)}, {HUFF_PACK(0x0000007e, 9), HUFF_PACK(0x0000001e, 7), HUFF_PACK(0x0000000c, 6), HUFF_PACK(0x00000005, 5)}, {HUFF_PACK(0x000000ff, 10), HUFF_PACK(0x0000000d, 6), HUFF_PACK(0x00000000, 3), HUFF_PACK(0x00000003, 2)}}, HUFF_PACK(0x00000000, 0) /* escape */ }, { /* LAV5_2D */ {{HUFF_PACK(0x00000002, 3), HUFF_PACK(0x00000003, 3), HUFF_PACK(0x00000010, 5), HUFF_PACK(0x0000007c, 7), HUFF_PACK(0x000000d6, 8), HUFF_PACK(0x000003ee, 10)}, {HUFF_PACK(0x0000000a, 4), HUFF_PACK(0x0000000c, 4), HUFF_PACK(0x00000016, 5), HUFF_PACK(0x00000034, 6), HUFF_PACK(0x000000fe, 8), HUFF_PACK(0x00001f7e, 13)}, {HUFF_PACK(0x0000007e, 7), HUFF_PACK(0x00000036, 6), HUFF_PACK(0x00000026, 6), HUFF_PACK(0x00000046, 7), HUFF_PACK(0x0000011e, 9), HUFF_PACK(0x000001f6, 9)}, {HUFF_PACK(0x0000011f, 9), HUFF_PACK(0x000000d7, 8), HUFF_PACK(0x0000008e, 8), HUFF_PACK(0x000000ff, 8), HUFF_PACK(0x0000006a, 7), HUFF_PACK(0x0000004e, 7)}, {HUFF_PACK(0x00000fbe, 12), HUFF_PACK(0x000007de, 11), HUFF_PACK(0x0000004f, 7), HUFF_PACK(0x00000037, 6), HUFF_PACK(0x00000017, 5), HUFF_PACK(0x0000001e, 5)}, {HUFF_PACK(0x00001f7f, 13), HUFF_PACK(0x000000fa, 8), HUFF_PACK(0x00000022, 6), HUFF_PACK(0x00000012, 5), HUFF_PACK(0x0000000e, 4), HUFF_PACK(0x00000000, 2)}}, HUFF_PACK(0x00000000, 0) /* escape */ }, { /* LAV7_2D */ {{HUFF_PACK(0x0000000e, 4), HUFF_PACK(0x0000000a, 4), HUFF_PACK(0x0000000a, 5), HUFF_PACK(0x0000007c, 7), HUFF_PACK(0x000000be, 8), HUFF_PACK(0x0000017a, 9), HUFF_PACK(0x000000ee, 9), HUFF_PACK(0x000007b6, 11)}, {HUFF_PACK(0x00000006, 4), HUFF_PACK(0x0000000c, 4), HUFF_PACK(0x00000016, 5), HUFF_PACK(0x00000026, 6), HUFF_PACK(0x0000003e, 7), HUFF_PACK(0x0000002e, 7), HUFF_PACK(0x000001ec, 9), HUFF_PACK(0x000047ce, 15)}, {HUFF_PACK(0x00000016, 6), HUFF_PACK(0x0000003c, 6), HUFF_PACK(0x00000022, 6), HUFF_PACK(0x0000004e, 7), HUFF_PACK(0x0000003f, 7), HUFF_PACK(0x0000005e, 8), HUFF_PACK(0x000008fa, 12), HUFF_PACK(0x000008fb, 12)}, {HUFF_PACK(0x0000005f, 8), HUFF_PACK(0x000000fa, 8), HUFF_PACK(0x000000bf, 8), HUFF_PACK(0x0000003a, 7), HUFF_PACK(0x000001f6, 9), HUFF_PACK(0x000001de, 10), HUFF_PACK(0x000003da, 10), HUFF_PACK(0x000007b7, 11)}, {HUFF_PACK(0x000001df, 10), HUFF_PACK(0x000003ee, 10), HUFF_PACK(0x0000017b, 9), HUFF_PACK(0x000003ef, 10), HUFF_PACK(0x000001ee, 9), HUFF_PACK(0x0000008e, 8), HUFF_PACK(0x000001ef, 9), HUFF_PACK(0x000001fe, 9)}, {HUFF_PACK(0x000008f8, 12), HUFF_PACK(0x0000047e, 11), HUFF_PACK(0x0000047f, 11), HUFF_PACK(0x00000076, 8), HUFF_PACK(0x0000003c, 7), HUFF_PACK(0x00000046, 7), HUFF_PACK(0x0000007a, 7), HUFF_PACK(0x0000007e, 7)}, {HUFF_PACK(0x000023e6, 14), HUFF_PACK(0x000011f2, 13), HUFF_PACK(0x000001ff, 9), HUFF_PACK(0x0000003d, 7), HUFF_PACK(0x0000004f, 7), HUFF_PACK(0x0000002e, 6), HUFF_PACK(0x00000012, 5), HUFF_PACK(0x00000004, 4)}, {HUFF_PACK(0x000047cf, 15), HUFF_PACK(0x0000011e, 9), HUFF_PACK(0x000000bc, 8), HUFF_PACK(0x000000fe, 8), HUFF_PACK(0x0000001c, 6), HUFF_PACK(0x00000010, 5), HUFF_PACK(0x0000000d, 4), HUFF_PACK(0x00000000, 2)}}, HUFF_PACK(0x00000000, 0) /* escape */ }, { /* LAV9_2D */ {{HUFF_PACK(0x00000006, 4), HUFF_PACK(0x00000007, 4), HUFF_PACK(0x00000006, 5), HUFF_PACK(0x0000007e, 7), HUFF_PACK(0x0000000a, 7), HUFF_PACK(0x0000001e, 8), HUFF_PACK(0x0000008a, 9), HUFF_PACK(0x0000004e, 10), HUFF_PACK(0x00000276, 10), HUFF_PACK(0x000002e2, 11)}, {HUFF_PACK(0x00000000, 4), HUFF_PACK(0x0000000a, 4), HUFF_PACK(0x00000016, 5), HUFF_PACK(0x00000026, 6), HUFF_PACK(0x00000076, 7), HUFF_PACK(0x000000f2, 8), HUFF_PACK(0x00000012, 8), HUFF_PACK(0x0000005e, 8), HUFF_PACK(0x0000008b, 9), HUFF_PACK(0x00002e76, 15)}, {HUFF_PACK(0x00000012, 6), HUFF_PACK(0x00000007, 5), HUFF_PACK(0x00000038, 6), HUFF_PACK(0x0000007c, 7), HUFF_PACK(0x00000008, 7), HUFF_PACK(0x00000046, 8), HUFF_PACK(0x000000f6, 8), HUFF_PACK(0x000001ca, 9), HUFF_PACK(0x0000173a, 14), HUFF_PACK(0x00001738, 14)}, {HUFF_PACK(0x0000009e, 8), HUFF_PACK(0x0000004a, 7), HUFF_PACK(0x00000026, 7), HUFF_PACK(0x0000000c, 7), HUFF_PACK(0x0000004e, 8), HUFF_PACK(0x000000f7, 8), HUFF_PACK(0x0000013a, 9), HUFF_PACK(0x0000009e, 11), HUFF_PACK(0x000009fe, 12), HUFF_PACK(0x0000013e, 12)}, {HUFF_PACK(0x00000026, 9), HUFF_PACK(0x0000001a, 8), HUFF_PACK(0x000001e6, 9), HUFF_PACK(0x000001e2, 9), HUFF_PACK(0x000000ee, 8), HUFF_PACK(0x000001ce, 9), HUFF_PACK(0x00000277, 10), HUFF_PACK(0x000003ce, 10), HUFF_PACK(0x000002e6, 11), HUFF_PACK(0x000004fc, 11)}, {HUFF_PACK(0x000002e3, 11), HUFF_PACK(0x00000170, 10), HUFF_PACK(0x00000172, 10), HUFF_PACK(0x000000ba, 9), HUFF_PACK(0x0000003e, 9), HUFF_PACK(0x000001e3, 9), HUFF_PACK(0x0000001b, 8), HUFF_PACK(0x0000003f, 9), HUFF_PACK(0x0000009e, 9), HUFF_PACK(0x0000009f, 9)}, {HUFF_PACK(0x00000b9e, 13), HUFF_PACK(0x000009ff, 12), HUFF_PACK(0x000004fd, 11), HUFF_PACK(0x000004fe, 11), HUFF_PACK(0x000001cf, 9), HUFF_PACK(0x000000ef, 8), HUFF_PACK(0x00000044, 8), HUFF_PACK(0x0000005f, 8), HUFF_PACK(0x000000e4, 8), HUFF_PACK(0x000000f0, 8)}, {HUFF_PACK(0x00002e72, 15), HUFF_PACK(0x0000013f, 12), HUFF_PACK(0x00000b9f, 13), HUFF_PACK(0x0000013e, 9), HUFF_PACK(0x000000fe, 8), HUFF_PACK(0x00000047, 8), HUFF_PACK(0x0000000e, 7), HUFF_PACK(0x0000007d, 7), HUFF_PACK(0x00000010, 6), HUFF_PACK(0x00000024, 6)}, {HUFF_PACK(0x00002e77, 15), HUFF_PACK(0x00005ce6, 16), HUFF_PACK(0x000000bb, 9), HUFF_PACK(0x000000e6, 8), HUFF_PACK(0x00000016, 8), HUFF_PACK(0x000000ff, 8), HUFF_PACK(0x0000007a, 7), HUFF_PACK(0x0000003a, 6), HUFF_PACK(0x00000017, 5), HUFF_PACK(0x00000002, 4)}, {HUFF_PACK(0x00005ce7, 16), HUFF_PACK(0x000003cf, 10), HUFF_PACK(0x00000017, 8), HUFF_PACK(0x000001cb, 9), HUFF_PACK(0x0000009c, 8), HUFF_PACK(0x0000004b, 7), HUFF_PACK(0x00000016, 6), HUFF_PACK(0x0000000a, 5), HUFF_PACK(0x00000008, 4), HUFF_PACK(0x00000006, 3)}}, HUFF_PACK(0x00000000, 0) /* escape */ }}, {/* HUFF_CLD_TAB_2D[0][1] */ { /* LAV3_2D */ {{HUFF_PACK(0x00000000, 1), HUFF_PACK(0x0000003e, 6), HUFF_PACK(0x0000076e, 11), HUFF_PACK(0x00000ede, 12)}, {HUFF_PACK(0x00000006, 3), HUFF_PACK(0x0000003f, 6), HUFF_PACK(0x000003b6, 10), HUFF_PACK(0x0000003a, 6)}, {HUFF_PACK(0x0000001c, 5), HUFF_PACK(0x000000ee, 8), HUFF_PACK(0x000001da, 9), HUFF_PACK(0x0000001e, 5)}, {HUFF_PACK(0x000000ef, 8), HUFF_PACK(0x00000edf, 12), HUFF_PACK(0x000000ec, 8), HUFF_PACK(0x00000002, 2)}}, HUFF_PACK(0x00000000, 0) /* escape */ }, { /* LAV5_2D */ {{HUFF_PACK(0x00000006, 3), HUFF_PACK(0x0000001c, 5), HUFF_PACK(0x0000007e, 8), HUFF_PACK(0x00000efc, 12), HUFF_PACK(0x0000effe, 16), HUFF_PACK(0x0001dffe, 17)}, {HUFF_PACK(0x00000004, 3), HUFF_PACK(0x0000000a, 4), HUFF_PACK(0x0000003e, 7), HUFF_PACK(0x00000efe, 12), HUFF_PACK(0x000077fe, 15), HUFF_PACK(0x00000076, 7)}, {HUFF_PACK(0x00000006, 4), HUFF_PACK(0x00000016, 5), HUFF_PACK(0x000000be, 8), HUFF_PACK(0x00000efd, 12), HUFF_PACK(0x000000ee, 8), HUFF_PACK(0x0000000e, 5)}, {HUFF_PACK(0x0000003e, 6), HUFF_PACK(0x0000002e, 6), HUFF_PACK(0x000001de, 9), HUFF_PACK(0x000003be, 10), HUFF_PACK(0x0000007e, 7), HUFF_PACK(0x0000001e, 5)}, {HUFF_PACK(0x0000007f, 7), HUFF_PACK(0x0000005e, 7), HUFF_PACK(0x00003bfe, 14), HUFF_PACK(0x000000fe, 9), HUFF_PACK(0x0000001e, 6), HUFF_PACK(0x00000002, 3)}, {HUFF_PACK(0x000000bf, 8), HUFF_PACK(0x0001dfff, 17), HUFF_PACK(0x00001dfe, 13), HUFF_PACK(0x000000ff, 9), HUFF_PACK(0x0000003a, 6), HUFF_PACK(0x00000000, 2)}}, HUFF_PACK(0x00000000, 0) /* escape */ }, { /* LAV7_2D */ {{HUFF_PACK(0x00000002, 3), HUFF_PACK(0x0000001c, 5), HUFF_PACK(0x000000bc, 8), HUFF_PACK(0x000005fc, 11), HUFF_PACK(0x00005ffe, 15), HUFF_PACK(0x0002ffde, 18), HUFF_PACK(0x000bff7e, 20), HUFF_PACK(0x0017feff, 21)}, {HUFF_PACK(0x00000004, 3), HUFF_PACK(0x0000000a, 4), HUFF_PACK(0x0000000e, 7), HUFF_PACK(0x000002fa, 10), HUFF_PACK(0x000001fe, 13), HUFF_PACK(0x0000bff2, 16), HUFF_PACK(0x0005ffbe, 19), HUFF_PACK(0x000000ee, 8)}, {HUFF_PACK(0x00000002, 4), HUFF_PACK(0x00000016, 5), HUFF_PACK(0x000000f6, 8), HUFF_PACK(0x000005fe, 11), HUFF_PACK(0x000001ff, 13), HUFF_PACK(0x0000bff6, 16), HUFF_PACK(0x000001de, 9), HUFF_PACK(0x0000007e, 7)}, {HUFF_PACK(0x00000000, 5), HUFF_PACK(0x0000003c, 6), HUFF_PACK(0x0000000e, 8), HUFF_PACK(0x0000003e, 10), HUFF_PACK(0x00002ffe, 14), HUFF_PACK(0x000002fb, 10), HUFF_PACK(0x000000f7, 8), HUFF_PACK(0x0000002e, 6)}, {HUFF_PACK(0x00000006, 6), HUFF_PACK(0x0000007a, 7), HUFF_PACK(0x0000000a, 8), HUFF_PACK(0x0000007e, 11), HUFF_PACK(0x000000fe, 12), HUFF_PACK(0x00000016, 9), HUFF_PACK(0x00000006, 7), HUFF_PACK(0x00000002, 5)}, {HUFF_PACK(0x0000000f, 7), HUFF_PACK(0x00000076, 7), HUFF_PACK(0x00000017, 9), HUFF_PACK(0x00005ff8, 15), HUFF_PACK(0x00000bfe, 12), HUFF_PACK(0x0000001e, 9), HUFF_PACK(0x0000007f, 7), HUFF_PACK(0x00000003, 4)}, {HUFF_PACK(0x00000004, 7), HUFF_PACK(0x000000bd, 8), HUFF_PACK(0x0000bff3, 16), HUFF_PACK(0x00005fff, 15), HUFF_PACK(0x00000bfa, 12), HUFF_PACK(0x0000017c, 9), HUFF_PACK(0x0000003a, 6), HUFF_PACK(0x00000003, 3)}, {HUFF_PACK(0x0000017e, 9), HUFF_PACK(0x0017fefe, 21), HUFF_PACK(0x00017fee, 17), HUFF_PACK(0x00005ffa, 15), HUFF_PACK(0x00000bfb, 12), HUFF_PACK(0x000001df, 9), HUFF_PACK(0x0000003e, 6), HUFF_PACK(0x00000006, 3)}}, HUFF_PACK(0x0017feff, 21) /* escape */ }, { /* LAV9_2D */ {{HUFF_PACK(0x0000000e, 4), HUFF_PACK(0x00000014, 5), HUFF_PACK(0x0000008e, 8), HUFF_PACK(0x000004fe, 11), HUFF_PACK(0x000023fe, 14), HUFF_PACK(0x00008ffe, 16), HUFF_PACK(0x0005ffbc, 19), HUFF_PACK(0x0017fef7, 21), HUFF_PACK(0x0017fef7, 21), HUFF_PACK(0x0017fef7, 21)}, {HUFF_PACK(0x00000002, 3), HUFF_PACK(0x00000002, 4), HUFF_PACK(0x00000044, 7), HUFF_PACK(0x0000027e, 10), HUFF_PACK(0x000017fc, 13), HUFF_PACK(0x0000bff6, 16), HUFF_PACK(0x0005ffbe, 19), HUFF_PACK(0x00011ff8, 17), HUFF_PACK(0x000bff7a, 20), HUFF_PACK(0x000000bc, 8)}, {HUFF_PACK(0x00000006, 4), HUFF_PACK(0x00000016, 5), HUFF_PACK(0x0000001a, 7), HUFF_PACK(0x000000fe, 10), HUFF_PACK(0x000011f6, 13), HUFF_PACK(0x0000bffe, 16), HUFF_PACK(0x00011ff9, 17), HUFF_PACK(0x0017fef6, 21), HUFF_PACK(0x0000011e, 9), HUFF_PACK(0x00000056, 7)}, {HUFF_PACK(0x00000010, 5), HUFF_PACK(0x0000003e, 6), HUFF_PACK(0x0000009e, 8), HUFF_PACK(0x000007fe, 11), HUFF_PACK(0x000011f7, 13), HUFF_PACK(0x00005ff8, 15), HUFF_PACK(0x00017fee, 17), HUFF_PACK(0x000007ff, 11), HUFF_PACK(0x000000ae, 8), HUFF_PACK(0x0000001e, 7)}, {HUFF_PACK(0x00000026, 6), HUFF_PACK(0x0000000e, 6), HUFF_PACK(0x000001ee, 9), HUFF_PACK(0x0000047e, 11), HUFF_PACK(0x00000bfc, 12), HUFF_PACK(0x0000bfff, 16), HUFF_PACK(0x000008fa, 12), HUFF_PACK(0x0000006e, 9), HUFF_PACK(0x000001ef, 9), HUFF_PACK(0x0000007e, 7)}, {HUFF_PACK(0x0000007a, 7), HUFF_PACK(0x0000004e, 7), HUFF_PACK(0x0000007e, 9), HUFF_PACK(0x000000de, 10), HUFF_PACK(0x000011fe, 13), HUFF_PACK(0x00002ffe, 14), HUFF_PACK(0x000004ff, 11), HUFF_PACK(0x000000ff, 10), HUFF_PACK(0x000000bd, 8), HUFF_PACK(0x0000002e, 6)}, {HUFF_PACK(0x000000fe, 8), HUFF_PACK(0x000000af, 8), HUFF_PACK(0x000001ec, 9), HUFF_PACK(0x000001be, 11), HUFF_PACK(0x00011ffe, 17), HUFF_PACK(0x00002ffa, 14), HUFF_PACK(0x000008fe, 12), HUFF_PACK(0x000003fe, 10), HUFF_PACK(0x00000046, 7), HUFF_PACK(0x00000012, 5)}, {HUFF_PACK(0x0000003e, 8), HUFF_PACK(0x00000045, 7), HUFF_PACK(0x000002fe, 10), HUFF_PACK(0x000bff7e, 20), HUFF_PACK(0x00005ff9, 15), HUFF_PACK(0x00005ffa, 15), HUFF_PACK(0x00000bfd, 12), HUFF_PACK(0x0000013e, 9), HUFF_PACK(0x0000000c, 6), HUFF_PACK(0x00000007, 4)}, {HUFF_PACK(0x000000be, 8), HUFF_PACK(0x00000036, 8), HUFF_PACK(0x000bff7f, 20), HUFF_PACK(0x00023ffe, 18), HUFF_PACK(0x00011ffa, 17), HUFF_PACK(0x00005ffe, 15), HUFF_PACK(0x000001bf, 11), HUFF_PACK(0x000001ed, 9), HUFF_PACK(0x0000002a, 6), HUFF_PACK(0x00000000, 3)}, {HUFF_PACK(0x0000017e, 9), HUFF_PACK(0x0017fef7, 21), HUFF_PACK(0x00047ffe, 19), HUFF_PACK(0x00047fff, 19), HUFF_PACK(0x00011ffb, 17), HUFF_PACK(0x00002ffb, 14), HUFF_PACK(0x0000047c, 11), HUFF_PACK(0x000001fe, 9), HUFF_PACK(0x0000003c, 6), HUFF_PACK(0x00000006, 3)}}, HUFF_PACK(0x0017fef7, 21) /* escape */ }}}, { /* HUFF_CLD_TAB_2D[1][] */ {/* HUFF_CLD_TAB_2D[1][0] */ { /* LAV3_2D */ {{HUFF_PACK(0x00000000, 1), HUFF_PACK(0x0000001e, 5), HUFF_PACK(0x000003be, 10), HUFF_PACK(0x00000efe, 12)}, {HUFF_PACK(0x00000006, 3), HUFF_PACK(0x0000001c, 5), HUFF_PACK(0x000001de, 9), HUFF_PACK(0x000000ea, 8)}, {HUFF_PACK(0x00000074, 7), HUFF_PACK(0x000000ee, 8), HUFF_PACK(0x000000eb, 8), HUFF_PACK(0x0000001f, 5)}, {HUFF_PACK(0x0000077e, 11), HUFF_PACK(0x00000eff, 12), HUFF_PACK(0x00000076, 7), HUFF_PACK(0x00000002, 2)}}, HUFF_PACK(0x00000000, 0) /* escape */ }, { /* LAV5_2D */ {{HUFF_PACK(0x00000000, 2), HUFF_PACK(0x00000006, 4), HUFF_PACK(0x00000024, 7), HUFF_PACK(0x0000025e, 11), HUFF_PACK(0x00003cfe, 14), HUFF_PACK(0x000079fe, 15)}, {HUFF_PACK(0x00000006, 3), HUFF_PACK(0x00000007, 4), HUFF_PACK(0x00000078, 7), HUFF_PACK(0x000003ce, 10), HUFF_PACK(0x00001e7e, 13), HUFF_PACK(0x000000be, 9)}, {HUFF_PACK(0x00000008, 5), HUFF_PACK(0x0000003e, 6), HUFF_PACK(0x00000026, 7), HUFF_PACK(0x0000012e, 10), HUFF_PACK(0x000000bf, 9), HUFF_PACK(0x0000002e, 7)}, {HUFF_PACK(0x00000027, 7), HUFF_PACK(0x0000007a, 7), HUFF_PACK(0x000001e4, 9), HUFF_PACK(0x00000096, 9), HUFF_PACK(0x0000007b, 7), HUFF_PACK(0x0000003f, 6)}, {HUFF_PACK(0x000001e6, 9), HUFF_PACK(0x000001e5, 9), HUFF_PACK(0x00000f3e, 12), HUFF_PACK(0x0000005e, 8), HUFF_PACK(0x00000016, 6), HUFF_PACK(0x0000000e, 4)}, {HUFF_PACK(0x0000079e, 11), HUFF_PACK(0x000079ff, 15), HUFF_PACK(0x0000025f, 11), HUFF_PACK(0x0000004a, 8), HUFF_PACK(0x0000000a, 5), HUFF_PACK(0x00000002, 2)}}, HUFF_PACK(0x00000000, 0) /* escape */ }, { /* LAV7_2D */ {{HUFF_PACK(0x00000000, 2), HUFF_PACK(0x00000006, 4), HUFF_PACK(0x000000de, 8), HUFF_PACK(0x0000069e, 11), HUFF_PACK(0x000034fe, 14), HUFF_PACK(0x0001a7fe, 17), HUFF_PACK(0x00069ff6, 19), HUFF_PACK(0x00069ff7, 19)}, {HUFF_PACK(0x00000002, 3), HUFF_PACK(0x0000000c, 4), HUFF_PACK(0x0000006a, 7), HUFF_PACK(0x0000034e, 10), HUFF_PACK(0x00001fde, 13), HUFF_PACK(0x000069fe, 15), HUFF_PACK(0x0001a7fc, 17), HUFF_PACK(0x00000372, 10)}, {HUFF_PACK(0x0000003e, 6), HUFF_PACK(0x0000003c, 6), HUFF_PACK(0x000000df, 8), HUFF_PACK(0x000001ee, 10), HUFF_PACK(0x00000dde, 12), HUFF_PACK(0x000069fa, 15), HUFF_PACK(0x00000373, 10), HUFF_PACK(0x0000007a, 8)}, {HUFF_PACK(0x0000003e, 7), HUFF_PACK(0x00000068, 7), HUFF_PACK(0x000001ba, 9), HUFF_PACK(0x000003f6, 10), HUFF_PACK(0x00000d3e, 12), HUFF_PACK(0x0000034c, 10), HUFF_PACK(0x000001fa, 9), HUFF_PACK(0x000000d2, 8)}, {HUFF_PACK(0x0000007e, 8), HUFF_PACK(0x0000007f, 8), HUFF_PACK(0x000001f8, 9), HUFF_PACK(0x000006ee, 11), HUFF_PACK(0x000003de, 11), HUFF_PACK(0x000001b8, 9), HUFF_PACK(0x000001fc, 9), HUFF_PACK(0x0000006b, 7)}, {HUFF_PACK(0x000000f6, 9), HUFF_PACK(0x000001fe, 9), HUFF_PACK(0x0000034d, 10), HUFF_PACK(0x00003fbe, 14), HUFF_PACK(0x000007f6, 11), HUFF_PACK(0x000003fa, 10), HUFF_PACK(0x0000003c, 7), HUFF_PACK(0x0000003d, 6)}, {HUFF_PACK(0x000003f7, 10), HUFF_PACK(0x00000376, 10), HUFF_PACK(0x0001a7ff, 17), HUFF_PACK(0x00003fbf, 14), HUFF_PACK(0x00000ddf, 12), HUFF_PACK(0x000001f9, 9), HUFF_PACK(0x00000036, 6), HUFF_PACK(0x0000000e, 4)}, {HUFF_PACK(0x000003df, 11), HUFF_PACK(0x00034ffa, 18), HUFF_PACK(0x000069fb, 15), HUFF_PACK(0x000034fc, 14), HUFF_PACK(0x00000fee, 12), HUFF_PACK(0x000001ff, 9), HUFF_PACK(0x0000000e, 5), HUFF_PACK(0x00000002, 2)}}, HUFF_PACK(0x00000000, 0) /* escape */ }, { /* LAV9_2D */ {{HUFF_PACK(0x00000006, 3), HUFF_PACK(0x00000004, 4), HUFF_PACK(0x00000012, 7), HUFF_PACK(0x000007fe, 11), HUFF_PACK(0x00001f7e, 13), HUFF_PACK(0x0000fbfe, 16), HUFF_PACK(0x0001f7fe, 17), HUFF_PACK(0x000b7dfe, 21), HUFF_PACK(0x000b7dff, 21), HUFF_PACK(0x000b7dff, 21)}, {HUFF_PACK(0x00000000, 3), HUFF_PACK(0x00000006, 4), HUFF_PACK(0x0000007c, 7), HUFF_PACK(0x00000046, 9), HUFF_PACK(0x000007d0, 12), HUFF_PACK(0x00001f4e, 14), HUFF_PACK(0x0000b7fe, 17), HUFF_PACK(0x00005bee, 16), HUFF_PACK(0x00016fbe, 18), HUFF_PACK(0x000003ee, 10)}, {HUFF_PACK(0x00000006, 5), HUFF_PACK(0x0000000a, 5), HUFF_PACK(0x0000002e, 7), HUFF_PACK(0x000003fe, 10), HUFF_PACK(0x000007d2, 12), HUFF_PACK(0x00001f4f, 14), HUFF_PACK(0x00002dfe, 15), HUFF_PACK(0x0000b7de, 17), HUFF_PACK(0x000001fe, 10), HUFF_PACK(0x0000002e, 8)}, {HUFF_PACK(0x0000007a, 7), HUFF_PACK(0x0000007e, 7), HUFF_PACK(0x0000007a, 8), HUFF_PACK(0x000001fa, 10), HUFF_PACK(0x000007fe, 12), HUFF_PACK(0x00001f7c, 13), HUFF_PACK(0x000016fa, 14), HUFF_PACK(0x0000009e, 10), HUFF_PACK(0x00000020, 8), HUFF_PACK(0x00000021, 8)}, {HUFF_PACK(0x000000fe, 8), HUFF_PACK(0x00000016, 7), HUFF_PACK(0x000000fe, 9), HUFF_PACK(0x0000016e, 10), HUFF_PACK(0x0000009f, 10), HUFF_PACK(0x00000b7c, 13), HUFF_PACK(0x000003de, 11), HUFF_PACK(0x000000b6, 9), HUFF_PACK(0x000000be, 9), HUFF_PACK(0x0000007c, 8)}, {HUFF_PACK(0x0000005a, 8), HUFF_PACK(0x00000078, 8), HUFF_PACK(0x00000047, 9), HUFF_PACK(0x00000044, 9), HUFF_PACK(0x000007ff, 12), HUFF_PACK(0x000007d1, 12), HUFF_PACK(0x000001f6, 10), HUFF_PACK(0x000001f7, 10), HUFF_PACK(0x0000002f, 8), HUFF_PACK(0x0000002c, 7)}, {HUFF_PACK(0x000000fc, 9), HUFF_PACK(0x000001f6, 9), HUFF_PACK(0x000000f6, 9), HUFF_PACK(0x000007ff, 11), HUFF_PACK(0x000016fe, 14), HUFF_PACK(0x000002de, 11), HUFF_PACK(0x000003ea, 11), HUFF_PACK(0x000000bf, 9), HUFF_PACK(0x000000fa, 8), HUFF_PACK(0x0000000a, 6)}, {HUFF_PACK(0x0000004e, 9), HUFF_PACK(0x00000026, 8), HUFF_PACK(0x000001ee, 10), HUFF_PACK(0x00005bfe, 16), HUFF_PACK(0x00003efe, 14), HUFF_PACK(0x00000b7e, 13), HUFF_PACK(0x000003eb, 11), HUFF_PACK(0x000001fe, 9), HUFF_PACK(0x0000007b, 7), HUFF_PACK(0x00000007, 5)}, {HUFF_PACK(0x000001fb, 10), HUFF_PACK(0x00000045, 9), HUFF_PACK(0x00016ffe, 18), HUFF_PACK(0x0001f7ff, 17), HUFF_PACK(0x00002df6, 15), HUFF_PACK(0x00001f7d, 13), HUFF_PACK(0x000003fe, 11), HUFF_PACK(0x0000005e, 8), HUFF_PACK(0x0000003c, 6), HUFF_PACK(0x0000000e, 4)}, {HUFF_PACK(0x000003df, 11), HUFF_PACK(0x0005befe, 20), HUFF_PACK(0x0002df7e, 19), HUFF_PACK(0x00016fff, 18), HUFF_PACK(0x00007dfe, 15), HUFF_PACK(0x00000fa6, 13), HUFF_PACK(0x000007de, 11), HUFF_PACK(0x00000079, 8), HUFF_PACK(0x0000000e, 5), HUFF_PACK(0x00000002, 2)}}, HUFF_PACK(0x000b7dff, 21) /* escape */ }}, {/* HUFF_CLD_TAB_2D[1][1] */ { /* LAV3_2D */ {{HUFF_PACK(0x00000000, 1), HUFF_PACK(0x0000000e, 4), HUFF_PACK(0x000000fa, 8), HUFF_PACK(0x000007de, 11)}, {HUFF_PACK(0x0000000c, 4), HUFF_PACK(0x0000001e, 5), HUFF_PACK(0x000000fe, 8), HUFF_PACK(0x000001f6, 9)}, {HUFF_PACK(0x000000ff, 8), HUFF_PACK(0x0000007c, 7), HUFF_PACK(0x0000007e, 7), HUFF_PACK(0x0000001a, 5)}, {HUFF_PACK(0x000007df, 11), HUFF_PACK(0x000003ee, 10), HUFF_PACK(0x0000001b, 5), HUFF_PACK(0x00000002, 2)}}, HUFF_PACK(0x00000000, 0) /* escape */ }, { /* LAV5_2D */ {{HUFF_PACK(0x00000006, 3), HUFF_PACK(0x0000000e, 4), HUFF_PACK(0x0000007c, 7), HUFF_PACK(0x000003fe, 10), HUFF_PACK(0x00000fbe, 12), HUFF_PACK(0x00003efe, 14)}, {HUFF_PACK(0x00000000, 3), HUFF_PACK(0x00000001, 3), HUFF_PACK(0x0000003c, 6), HUFF_PACK(0x0000005e, 8), HUFF_PACK(0x000007de, 11), HUFF_PACK(0x000007be, 11)}, {HUFF_PACK(0x0000001e, 6), HUFF_PACK(0x0000000a, 5), HUFF_PACK(0x0000001f, 6), HUFF_PACK(0x0000005f, 8), HUFF_PACK(0x000001ee, 9), HUFF_PACK(0x000001f6, 9)}, {HUFF_PACK(0x000001fe, 9), HUFF_PACK(0x000000fe, 8), HUFF_PACK(0x000000f6, 8), HUFF_PACK(0x000000fa, 8), HUFF_PACK(0x0000007e, 7), HUFF_PACK(0x00000016, 6)}, {HUFF_PACK(0x000007bf, 11), HUFF_PACK(0x000003de, 10), HUFF_PACK(0x000003ee, 10), HUFF_PACK(0x0000007a, 7), HUFF_PACK(0x0000000e, 5), HUFF_PACK(0x00000006, 4)}, {HUFF_PACK(0x00003eff, 14), HUFF_PACK(0x00001f7e, 13), HUFF_PACK(0x000003ff, 10), HUFF_PACK(0x0000002e, 7), HUFF_PACK(0x00000004, 4), HUFF_PACK(0x00000002, 2)}}, HUFF_PACK(0x00000000, 0) /* escape */ }, { /* LAV7_2D */ {{HUFF_PACK(0x00000002, 3), HUFF_PACK(0x0000000a, 4), HUFF_PACK(0x0000001a, 6), HUFF_PACK(0x000001be, 9), HUFF_PACK(0x000006e6, 11), HUFF_PACK(0x0000067a, 12), HUFF_PACK(0x00000cf2, 13), HUFF_PACK(0x000033de, 15)}, {HUFF_PACK(0x0000000c, 4), HUFF_PACK(0x0000000e, 4), HUFF_PACK(0x0000000e, 5), HUFF_PACK(0x000000de, 8), HUFF_PACK(0x00000372, 10), HUFF_PACK(0x000003d6, 11), HUFF_PACK(0x00000678, 12), HUFF_PACK(0x00000cf6, 13)}, {HUFF_PACK(0x00000036, 6), HUFF_PACK(0x00000012, 5), HUFF_PACK(0x0000003e, 6), HUFF_PACK(0x0000003c, 7), HUFF_PACK(0x000001b8, 9), HUFF_PACK(0x000003d4, 11), HUFF_PACK(0x0000033e, 11), HUFF_PACK(0x0000033f, 11)}, {HUFF_PACK(0x0000007e, 8), HUFF_PACK(0x0000006a, 7), HUFF_PACK(0x0000004e, 7), HUFF_PACK(0x0000007e, 7), HUFF_PACK(0x000001ba, 9), HUFF_PACK(0x000000ce, 9), HUFF_PACK(0x000000f6, 9), HUFF_PACK(0x000001ee, 10)}, {HUFF_PACK(0x000001ef, 10), HUFF_PACK(0x0000013e, 9), HUFF_PACK(0x0000007f, 8), HUFF_PACK(0x00000066, 8), HUFF_PACK(0x000000d6, 8), HUFF_PACK(0x0000003e, 7), HUFF_PACK(0x000000d7, 8), HUFF_PACK(0x0000009e, 8)}, {HUFF_PACK(0x000007ae, 12), HUFF_PACK(0x000001e8, 10), HUFF_PACK(0x000001e9, 10), HUFF_PACK(0x0000027e, 10), HUFF_PACK(0x00000032, 7), HUFF_PACK(0x00000018, 6), HUFF_PACK(0x00000026, 6), HUFF_PACK(0x00000034, 6)}, {HUFF_PACK(0x00000cf3, 13), HUFF_PACK(0x000007aa, 12), HUFF_PACK(0x000007ab, 12), HUFF_PACK(0x0000027f, 10), HUFF_PACK(0x000001bf, 9), HUFF_PACK(0x0000001b, 6), HUFF_PACK(0x0000001e, 5), HUFF_PACK(0x0000000b, 4)}, {HUFF_PACK(0x000033df, 15), HUFF_PACK(0x000019ee, 14), HUFF_PACK(0x000007af, 12), HUFF_PACK(0x000006e7, 11), HUFF_PACK(0x000001bb, 9), HUFF_PACK(0x0000007f, 7), HUFF_PACK(0x00000008, 4), HUFF_PACK(0x00000000, 2)}}, HUFF_PACK(0x00000000, 0) /* escape */ }, { /* LAV9_2D */ {{HUFF_PACK(0x0000000e, 4), HUFF_PACK(0x00000008, 4), HUFF_PACK(0x0000007e, 7), HUFF_PACK(0x000001fe, 9), HUFF_PACK(0x000001ba, 10), HUFF_PACK(0x00000dbe, 12), HUFF_PACK(0x00000d7e, 13), HUFF_PACK(0x00001af6, 14), HUFF_PACK(0x00007fec, 15), HUFF_PACK(0x0001ffb6, 17)}, {HUFF_PACK(0x0000000a, 4), HUFF_PACK(0x0000000c, 4), HUFF_PACK(0x0000000c, 5), HUFF_PACK(0x00000036, 7), HUFF_PACK(0x000000de, 9), HUFF_PACK(0x000005fe, 11), HUFF_PACK(0x000006be, 12), HUFF_PACK(0x00001b7e, 13), HUFF_PACK(0x00007fee, 15), HUFF_PACK(0x00006dfe, 15)}, {HUFF_PACK(0x0000001e, 6), HUFF_PACK(0x0000000e, 5), HUFF_PACK(0x0000000a, 5), HUFF_PACK(0x0000006a, 7), HUFF_PACK(0x000001ae, 9), HUFF_PACK(0x000006fe, 11), HUFF_PACK(0x00000376, 11), HUFF_PACK(0x00000dfe, 13), HUFF_PACK(0x00000dff, 13), HUFF_PACK(0x00000d7f, 13)}, {HUFF_PACK(0x000000b6, 8), HUFF_PACK(0x0000005e, 7), HUFF_PACK(0x0000007c, 7), HUFF_PACK(0x0000006e, 7), HUFF_PACK(0x0000006a, 8), HUFF_PACK(0x0000016a, 9), HUFF_PACK(0x00000ffe, 12), HUFF_PACK(0x00000dfe, 12), HUFF_PACK(0x00000ffc, 12), HUFF_PACK(0x00001bfe, 13)}, {HUFF_PACK(0x0000035e, 10), HUFF_PACK(0x000001b6, 9), HUFF_PACK(0x0000005e, 8), HUFF_PACK(0x000000b4, 8), HUFF_PACK(0x0000006c, 7), HUFF_PACK(0x0000017e, 9), HUFF_PACK(0x0000036e, 10), HUFF_PACK(0x000003ee, 10), HUFF_PACK(0x0000037e, 11), HUFF_PACK(0x00000377, 11)}, {HUFF_PACK(0x00000fff, 12), HUFF_PACK(0x000001ae, 10), HUFF_PACK(0x000001be, 10), HUFF_PACK(0x000001f6, 9), HUFF_PACK(0x000001be, 9), HUFF_PACK(0x000000da, 8), HUFF_PACK(0x000000fe, 8), HUFF_PACK(0x0000016b, 9), HUFF_PACK(0x000000d6, 9), HUFF_PACK(0x0000037e, 10)}, {HUFF_PACK(0x000017fe, 13), HUFF_PACK(0x00000bfe, 12), HUFF_PACK(0x000007de, 11), HUFF_PACK(0x000006de, 11), HUFF_PACK(0x000001b8, 10), HUFF_PACK(0x000000d6, 8), HUFF_PACK(0x0000002e, 7), HUFF_PACK(0x00000034, 7), HUFF_PACK(0x000000de, 8), HUFF_PACK(0x000000be, 8)}, {HUFF_PACK(0x00007fef, 15), HUFF_PACK(0x000006bc, 12), HUFF_PACK(0x00001bff, 13), HUFF_PACK(0x00001ffa, 13), HUFF_PACK(0x000001b9, 10), HUFF_PACK(0x000003fe, 10), HUFF_PACK(0x000000fa, 8), HUFF_PACK(0x0000002e, 6), HUFF_PACK(0x00000034, 6), HUFF_PACK(0x0000001f, 6)}, {HUFF_PACK(0x00006dff, 15), HUFF_PACK(0x00001af7, 14), HUFF_PACK(0x000036fe, 14), HUFF_PACK(0x000006fe, 12), HUFF_PACK(0x00000fbe, 12), HUFF_PACK(0x0000035f, 10), HUFF_PACK(0x000000b7, 8), HUFF_PACK(0x0000002c, 6), HUFF_PACK(0x0000001e, 5), HUFF_PACK(0x00000009, 4)}, {HUFF_PACK(0x0001ffb7, 17), HUFF_PACK(0x0000ffda, 16), HUFF_PACK(0x00000d7a, 13), HUFF_PACK(0x000017ff, 13), HUFF_PACK(0x00000fbf, 12), HUFF_PACK(0x000002fe, 10), HUFF_PACK(0x0000005f, 8), HUFF_PACK(0x00000016, 6), HUFF_PACK(0x00000004, 4), HUFF_PACK(0x00000000, 2)}}, HUFF_PACK(0x00000000, 0) /* escape */ }}}}}; const HUFF_ICC_TABLE fdk_sacenc_huffICCTab = { {/* h1D[2][8] */ {HUFF_PACK(0x00000000, 1), HUFF_PACK(0x00000002, 2), HUFF_PACK(0x00000006, 3), HUFF_PACK(0x0000000e, 4), HUFF_PACK(0x0000001e, 5), HUFF_PACK(0x0000003e, 6), HUFF_PACK(0x0000007e, 7), HUFF_PACK(0x0000007f, 7)}, {HUFF_PACK(0x00000000, 1), HUFF_PACK(0x00000002, 2), HUFF_PACK(0x00000006, 3), HUFF_PACK(0x0000000e, 4), HUFF_PACK(0x0000001e, 5), HUFF_PACK(0x0000003e, 6), HUFF_PACK(0x0000007e, 7), HUFF_PACK(0x0000007f, 7)}}, { /* HUFF_ICC_TAB_2D */ { /* HUFF_ICC_TAB_2D[0][] */ {/* HUFF_ICC_TAB_2D[0][0] */ { /* LAV1_2D */ {{HUFF_PACK(0x00000000, 1), HUFF_PACK(0x00000006, 3)}, {HUFF_PACK(0x00000007, 3), HUFF_PACK(0x00000002, 2)}}, HUFF_PACK(0x00000000, 0) /* escape */ }, { /* LAV3_2D */ {{HUFF_PACK(0x00000002, 2), HUFF_PACK(0x00000000, 2), HUFF_PACK(0x0000000a, 5), HUFF_PACK(0x0000007e, 8)}, {HUFF_PACK(0x0000000e, 5), HUFF_PACK(0x00000004, 4), HUFF_PACK(0x00000016, 6), HUFF_PACK(0x000003fe, 11)}, {HUFF_PACK(0x000001fe, 10), HUFF_PACK(0x000000fe, 9), HUFF_PACK(0x0000003e, 7), HUFF_PACK(0x0000001e, 6)}, {HUFF_PACK(0x000003ff, 11), HUFF_PACK(0x00000017, 6), HUFF_PACK(0x00000006, 4), HUFF_PACK(0x00000003, 2)}}, HUFF_PACK(0x00000000, 0) /* escape */ }, { /* LAV5_2D */ {{HUFF_PACK(0x00000000, 2), HUFF_PACK(0x00000002, 3), HUFF_PACK(0x0000000c, 5), HUFF_PACK(0x0000006a, 7), HUFF_PACK(0x000000dc, 8), HUFF_PACK(0x000006ee, 11)}, {HUFF_PACK(0x0000001e, 5), HUFF_PACK(0x0000000c, 4), HUFF_PACK(0x0000000d, 5), HUFF_PACK(0x0000001e, 6), HUFF_PACK(0x000001ae, 9), HUFF_PACK(0x0000ddff, 16)}, {HUFF_PACK(0x000000de, 8), HUFF_PACK(0x0000007e, 7), HUFF_PACK(0x0000001f, 6), HUFF_PACK(0x000001be, 9), HUFF_PACK(0x00006efe, 15), HUFF_PACK(0x0000ddfe, 16)}, {HUFF_PACK(0x0000377e, 14), HUFF_PACK(0x00001bbe, 13), HUFF_PACK(0x00000dde, 12), HUFF_PACK(0x000001bf, 9), HUFF_PACK(0x000000d6, 8), HUFF_PACK(0x00000376, 10)}, {HUFF_PACK(0x0000ddff, 16), HUFF_PACK(0x0000ddff, 16), HUFF_PACK(0x000001ba, 9), HUFF_PACK(0x00000034, 6), HUFF_PACK(0x0000003e, 6), HUFF_PACK(0x0000000e, 5)}, {HUFF_PACK(0x0000ddff, 16), HUFF_PACK(0x000001af, 9), HUFF_PACK(0x0000007f, 7), HUFF_PACK(0x00000036, 6), HUFF_PACK(0x0000000e, 4), HUFF_PACK(0x00000002, 2)}}, HUFF_PACK(0x0000ddff, 16) /* escape */ }, { /* LAV7_2D */ {{HUFF_PACK(0x00000000, 2), HUFF_PACK(0x0000000c, 4), HUFF_PACK(0x0000002e, 6), HUFF_PACK(0x00000044, 7), HUFF_PACK(0x00000086, 8), HUFF_PACK(0x0000069e, 11), HUFF_PACK(0x0000043e, 11), HUFF_PACK(0x0000087a, 12)}, {HUFF_PACK(0x0000001e, 5), HUFF_PACK(0x0000000e, 4), HUFF_PACK(0x0000002a, 6), HUFF_PACK(0x00000046, 7), HUFF_PACK(0x0000015e, 9), HUFF_PACK(0x00000047, 7), HUFF_PACK(0x0000034a, 10), HUFF_PACK(0x0000087b, 12)}, {HUFF_PACK(0x000000d6, 8), HUFF_PACK(0x00000026, 6), HUFF_PACK(0x0000002f, 6), HUFF_PACK(0x000000d7, 8), HUFF_PACK(0x0000006a, 7), HUFF_PACK(0x0000034e, 10), HUFF_PACK(0x0000087b, 12), HUFF_PACK(0x0000087b, 12)}, {HUFF_PACK(0x000002be, 10), HUFF_PACK(0x000001a6, 9), HUFF_PACK(0x000001be, 9), HUFF_PACK(0x00000012, 5), HUFF_PACK(0x000001bf, 9), HUFF_PACK(0x0000087b, 12), HUFF_PACK(0x0000087b, 12), HUFF_PACK(0x0000087b, 12)}, {HUFF_PACK(0x0000087b, 12), HUFF_PACK(0x0000087b, 12), HUFF_PACK(0x0000087b, 12), HUFF_PACK(0x0000087b, 12), HUFF_PACK(0x00000036, 6), HUFF_PACK(0x000000d0, 8), HUFF_PACK(0x0000043c, 11), HUFF_PACK(0x0000043f, 11)}, {HUFF_PACK(0x0000087b, 12), HUFF_PACK(0x0000087b, 12), HUFF_PACK(0x0000087b, 12), HUFF_PACK(0x0000034b, 10), HUFF_PACK(0x00000027, 6), HUFF_PACK(0x00000020, 6), HUFF_PACK(0x00000042, 7), HUFF_PACK(0x000000d1, 8)}, {HUFF_PACK(0x0000087b, 12), HUFF_PACK(0x0000087b, 12), HUFF_PACK(0x000002bf, 10), HUFF_PACK(0x000000de, 8), HUFF_PACK(0x000000ae, 8), HUFF_PACK(0x00000056, 7), HUFF_PACK(0x00000016, 5), HUFF_PACK(0x00000014, 5)}, {HUFF_PACK(0x0000087b, 12), HUFF_PACK(0x0000069f, 11), HUFF_PACK(0x000001a4, 9), HUFF_PACK(0x0000010e, 9), HUFF_PACK(0x00000045, 7), HUFF_PACK(0x0000006e, 7), HUFF_PACK(0x0000001f, 5), HUFF_PACK(0x00000001, 2)}}, HUFF_PACK(0x0000087b, 12) /* escape */ }}, {/* HUFF_ICC_TAB_2D[0][1] */ { /* LAV1_2D */ {{HUFF_PACK(0x00000000, 1), HUFF_PACK(0x00000006, 3)}, {HUFF_PACK(0x00000007, 3), HUFF_PACK(0x00000002, 2)}}, HUFF_PACK(0x00000000, 0) /* escape */ }, { /* LAV3_2D */ {{HUFF_PACK(0x00000002, 2), HUFF_PACK(0x00000004, 4), HUFF_PACK(0x0000017e, 10), HUFF_PACK(0x000002fe, 11)}, {HUFF_PACK(0x00000000, 2), HUFF_PACK(0x0000000e, 5), HUFF_PACK(0x000000be, 9), HUFF_PACK(0x00000016, 6)}, {HUFF_PACK(0x0000000f, 5), HUFF_PACK(0x00000014, 6), HUFF_PACK(0x0000005e, 8), HUFF_PACK(0x00000006, 4)}, {HUFF_PACK(0x0000002e, 7), HUFF_PACK(0x000002ff, 11), HUFF_PACK(0x00000015, 6), HUFF_PACK(0x00000003, 2)}}, HUFF_PACK(0x00000000, 0) /* escape */ }, { /* LAV5_2D */ {{HUFF_PACK(0x00000000, 2), HUFF_PACK(0x0000001e, 5), HUFF_PACK(0x000003fc, 10), HUFF_PACK(0x0000fffa, 16), HUFF_PACK(0x000fff9e, 20), HUFF_PACK(0x000fff9f, 20)}, {HUFF_PACK(0x00000006, 3), HUFF_PACK(0x00000004, 4), HUFF_PACK(0x000000be, 9), HUFF_PACK(0x00007ffe, 15), HUFF_PACK(0x0007ffce, 19), HUFF_PACK(0x000000fe, 8)}, {HUFF_PACK(0x00000006, 4), HUFF_PACK(0x0000001e, 6), HUFF_PACK(0x000003fd, 10), HUFF_PACK(0x0000fffb, 16), HUFF_PACK(0x00000ffe, 12), HUFF_PACK(0x0000003e, 6)}, {HUFF_PACK(0x0000000a, 5), HUFF_PACK(0x0000007e, 7), HUFF_PACK(0x00001ffe, 13), HUFF_PACK(0x00007fff, 15), HUFF_PACK(0x0000005e, 8), HUFF_PACK(0x0000000e, 5)}, {HUFF_PACK(0x0000001f, 6), HUFF_PACK(0x000003fe, 10), HUFF_PACK(0x0001fff2, 17), HUFF_PACK(0x00000ffc, 12), HUFF_PACK(0x0000002e, 7), HUFF_PACK(0x0000000e, 4)}, {HUFF_PACK(0x000000bf, 9), HUFF_PACK(0x0003ffe6, 18), HUFF_PACK(0x0000fff8, 16), HUFF_PACK(0x00000ffd, 12), HUFF_PACK(0x00000016, 6), HUFF_PACK(0x00000002, 2)}}, HUFF_PACK(0x00000000, 0) /* escape */ }, { /* LAV7_2D */ {{HUFF_PACK(0x00000002, 2), HUFF_PACK(0x0000001e, 6), HUFF_PACK(0x00000ffe, 12), HUFF_PACK(0x0000ffff, 16), HUFF_PACK(0x0000fffe, 16), HUFF_PACK(0x0000ffff, 16), HUFF_PACK(0x0000ffff, 16), HUFF_PACK(0x0000ffff, 16)}, {HUFF_PACK(0x00000006, 3), HUFF_PACK(0x00000008, 5), HUFF_PACK(0x000007fe, 11), HUFF_PACK(0x0000ffff, 16), HUFF_PACK(0x0000ffff, 16), HUFF_PACK(0x0000ffff, 16), HUFF_PACK(0x0000ffff, 16), HUFF_PACK(0x0000005a, 8)}, {HUFF_PACK(0x00000006, 4), HUFF_PACK(0x0000007a, 7), HUFF_PACK(0x00000164, 10), HUFF_PACK(0x00007ffa, 15), HUFF_PACK(0x0000ffff, 16), HUFF_PACK(0x0000ffff, 16), HUFF_PACK(0x00001fee, 13), HUFF_PACK(0x0000003c, 6)}, {HUFF_PACK(0x0000000e, 5), HUFF_PACK(0x000000fe, 8), HUFF_PACK(0x000002ce, 11), HUFF_PACK(0x000002cf, 11), HUFF_PACK(0x00007ffb, 15), HUFF_PACK(0x00001fec, 13), HUFF_PACK(0x000000b0, 9), HUFF_PACK(0x0000002e, 7)}, {HUFF_PACK(0x0000003e, 6), HUFF_PACK(0x000003fe, 10), HUFF_PACK(0x00000165, 10), HUFF_PACK(0x00007ffc, 15), HUFF_PACK(0x00001fef, 13), HUFF_PACK(0x000007fa, 11), HUFF_PACK(0x000007f8, 11), HUFF_PACK(0x0000001f, 6)}, {HUFF_PACK(0x0000002f, 7), HUFF_PACK(0x000000f6, 8), HUFF_PACK(0x00001fed, 13), HUFF_PACK(0x0000ffff, 16), HUFF_PACK(0x00007ffd, 15), HUFF_PACK(0x00000ff2, 12), HUFF_PACK(0x000000b1, 9), HUFF_PACK(0x0000000a, 5)}, {HUFF_PACK(0x00000009, 5), HUFF_PACK(0x00000166, 10), HUFF_PACK(0x0000ffff, 16), HUFF_PACK(0x0000ffff, 16), HUFF_PACK(0x00007ffe, 15), HUFF_PACK(0x00003ffc, 14), HUFF_PACK(0x0000005b, 8), HUFF_PACK(0x0000000e, 4)}, {HUFF_PACK(0x0000007e, 7), HUFF_PACK(0x0000ffff, 16), HUFF_PACK(0x0000ffff, 16), HUFF_PACK(0x0000ffff, 16), HUFF_PACK(0x0000ffff, 16), HUFF_PACK(0x00000ff3, 12), HUFF_PACK(0x000000f7, 8), HUFF_PACK(0x00000000, 2)}}, HUFF_PACK(0x0000ffff, 16) /* escape */ }}}, { /* HUFF_ICC_TAB_2D[1][] */ {/* HUFF_ICC_TAB_2D[1][0] */ { /* LAV1_2D */ {{HUFF_PACK(0x00000000, 1), HUFF_PACK(0x00000006, 3)}, {HUFF_PACK(0x00000007, 3), HUFF_PACK(0x00000002, 2)}}, HUFF_PACK(0x00000000, 0) /* escape */ }, { /* LAV3_2D */ {{HUFF_PACK(0x00000002, 2), HUFF_PACK(0x0000000e, 4), HUFF_PACK(0x0000037e, 10), HUFF_PACK(0x00000dfe, 12)}, {HUFF_PACK(0x0000000f, 4), HUFF_PACK(0x0000000c, 4), HUFF_PACK(0x000001ba, 9), HUFF_PACK(0x000001bb, 9)}, {HUFF_PACK(0x000000de, 8), HUFF_PACK(0x000000dc, 8), HUFF_PACK(0x000001be, 9), HUFF_PACK(0x0000001a, 5)}, {HUFF_PACK(0x000006fe, 11), HUFF_PACK(0x00000dff, 12), HUFF_PACK(0x00000036, 6), HUFF_PACK(0x00000000, 1)}}, HUFF_PACK(0x00000000, 0) /* escape */ }, { /* LAV5_2D */ {{HUFF_PACK(0x00000000, 1), HUFF_PACK(0x0000000c, 4), HUFF_PACK(0x000001b6, 9), HUFF_PACK(0x00001b7c, 13), HUFF_PACK(0x0000dbfe, 16), HUFF_PACK(0x00036fff, 18)}, {HUFF_PACK(0x0000000e, 4), HUFF_PACK(0x0000001e, 5), HUFF_PACK(0x000001be, 9), HUFF_PACK(0x00000dfe, 12), HUFF_PACK(0x00036ffe, 18), HUFF_PACK(0x0000036e, 10)}, {HUFF_PACK(0x0000006e, 7), HUFF_PACK(0x000000fe, 8), HUFF_PACK(0x000000d8, 8), HUFF_PACK(0x000036fe, 14), HUFF_PACK(0x000006de, 11), HUFF_PACK(0x000000de, 8)}, {HUFF_PACK(0x000001fa, 9), HUFF_PACK(0x000000da, 8), HUFF_PACK(0x00000dff, 12), HUFF_PACK(0x00001b7e, 13), HUFF_PACK(0x000000d9, 8), HUFF_PACK(0x000000ff, 8)}, {HUFF_PACK(0x000003f6, 10), HUFF_PACK(0x000006fe, 11), HUFF_PACK(0x00006dfe, 15), HUFF_PACK(0x0000037e, 10), HUFF_PACK(0x000000fc, 8), HUFF_PACK(0x0000001a, 5)}, {HUFF_PACK(0x000007ee, 11), HUFF_PACK(0x0001b7fe, 17), HUFF_PACK(0x00001b7d, 13), HUFF_PACK(0x000007ef, 11), HUFF_PACK(0x0000003e, 6), HUFF_PACK(0x00000002, 2)}}, HUFF_PACK(0x00036fff, 18) /* escape */ }, { /* LAV7_2D */ {{HUFF_PACK(0x00000000, 1), HUFF_PACK(0x0000000c, 4), HUFF_PACK(0x000007ee, 11), HUFF_PACK(0x00001e7e, 13), HUFF_PACK(0x00003cfe, 14), HUFF_PACK(0x000079ff, 15), HUFF_PACK(0x000079ff, 15), HUFF_PACK(0x000079ff, 15)}, {HUFF_PACK(0x0000000e, 4), HUFF_PACK(0x0000001a, 5), HUFF_PACK(0x000001e6, 9), HUFF_PACK(0x00001fbe, 13), HUFF_PACK(0x000079fe, 15), HUFF_PACK(0x000079ff, 15), HUFF_PACK(0x000079ff, 15), HUFF_PACK(0x000006fc, 11)}, {HUFF_PACK(0x0000006c, 7), HUFF_PACK(0x000000f6, 8), HUFF_PACK(0x000001ba, 9), HUFF_PACK(0x00000dfc, 12), HUFF_PACK(0x00000dfd, 12), HUFF_PACK(0x000079ff, 15), HUFF_PACK(0x00000f3e, 12), HUFF_PACK(0x000001bb, 9)}, {HUFF_PACK(0x000000dc, 8), HUFF_PACK(0x000001fe, 9), HUFF_PACK(0x0000036e, 10), HUFF_PACK(0x000003fe, 10), HUFF_PACK(0x000079ff, 15), HUFF_PACK(0x00000fde, 12), HUFF_PACK(0x000001ee, 9), HUFF_PACK(0x000000f2, 8)}, {HUFF_PACK(0x000001fa, 9), HUFF_PACK(0x000003f6, 10), HUFF_PACK(0x000001be, 9), HUFF_PACK(0x000079ff, 15), HUFF_PACK(0x00001fbf, 13), HUFF_PACK(0x000003ce, 10), HUFF_PACK(0x000003ff, 10), HUFF_PACK(0x000000de, 8)}, {HUFF_PACK(0x00000078, 7), HUFF_PACK(0x000000da, 8), HUFF_PACK(0x000079ff, 15), HUFF_PACK(0x000079ff, 15), HUFF_PACK(0x000006fd, 11), HUFF_PACK(0x0000036c, 10), HUFF_PACK(0x000001ef, 9), HUFF_PACK(0x000000fe, 8)}, {HUFF_PACK(0x0000036f, 10), HUFF_PACK(0x00000dfe, 12), HUFF_PACK(0x000079ff, 15), HUFF_PACK(0x000079ff, 15), HUFF_PACK(0x000079ff, 15), HUFF_PACK(0x0000036d, 10), HUFF_PACK(0x000000fc, 8), HUFF_PACK(0x0000003e, 6)}, {HUFF_PACK(0x00000dff, 12), HUFF_PACK(0x000079ff, 15), HUFF_PACK(0x000079ff, 15), HUFF_PACK(0x000079ff, 15), HUFF_PACK(0x000079ff, 15), HUFF_PACK(0x0000079e, 11), HUFF_PACK(0x0000007a, 7), HUFF_PACK(0x00000002, 2)}}, HUFF_PACK(0x000079ff, 15) /* escape */ }}, {/* HUFF_ICC_TAB_2D[1][1] */ { /* LAV1_2D */ {{HUFF_PACK(0x00000000, 1), HUFF_PACK(0x00000006, 3)}, {HUFF_PACK(0x00000007, 3), HUFF_PACK(0x00000002, 2)}}, HUFF_PACK(0x00000000, 0) /* escape */ }, { /* LAV3_2D */ {{HUFF_PACK(0x00000002, 2), HUFF_PACK(0x0000000e, 4), HUFF_PACK(0x000000fc, 8), HUFF_PACK(0x00000fde, 12)}, {HUFF_PACK(0x0000000c, 4), HUFF_PACK(0x0000000d, 4), HUFF_PACK(0x000001fe, 9), HUFF_PACK(0x000007ee, 11)}, {HUFF_PACK(0x000001fa, 9), HUFF_PACK(0x000001ff, 9), HUFF_PACK(0x000000fe, 8), HUFF_PACK(0x0000003e, 6)}, {HUFF_PACK(0x00000fdf, 12), HUFF_PACK(0x000003f6, 10), HUFF_PACK(0x0000001e, 5), HUFF_PACK(0x00000000, 1)}}, HUFF_PACK(0x00000000, 0) /* escape */ }, { /* LAV5_2D */ {{HUFF_PACK(0x00000000, 2), HUFF_PACK(0x0000000e, 4), HUFF_PACK(0x0000003a, 7), HUFF_PACK(0x00000676, 11), HUFF_PACK(0x000019fe, 13), HUFF_PACK(0x0000cebe, 16)}, {HUFF_PACK(0x0000000f, 4), HUFF_PACK(0x00000002, 3), HUFF_PACK(0x0000001e, 6), HUFF_PACK(0x000000fe, 9), HUFF_PACK(0x000019d6, 13), HUFF_PACK(0x0000675e, 15)}, {HUFF_PACK(0x0000003e, 7), HUFF_PACK(0x00000032, 6), HUFF_PACK(0x00000018, 5), HUFF_PACK(0x0000033e, 10), HUFF_PACK(0x00000cfe, 12), HUFF_PACK(0x00000677, 11)}, {HUFF_PACK(0x00000674, 11), HUFF_PACK(0x0000019c, 9), HUFF_PACK(0x000000ff, 9), HUFF_PACK(0x0000003b, 7), HUFF_PACK(0x0000001c, 6), HUFF_PACK(0x0000007e, 8)}, {HUFF_PACK(0x000033fe, 14), HUFF_PACK(0x000033ff, 14), HUFF_PACK(0x00000cea, 12), HUFF_PACK(0x00000066, 7), HUFF_PACK(0x0000001a, 5), HUFF_PACK(0x00000006, 4)}, {HUFF_PACK(0x0000cebf, 16), HUFF_PACK(0x000033ae, 14), HUFF_PACK(0x0000067e, 11), HUFF_PACK(0x0000019e, 9), HUFF_PACK(0x0000001b, 5), HUFF_PACK(0x00000002, 2)}}, HUFF_PACK(0x00000000, 0) /* escape */ }, { /* LAV7_2D */ {{HUFF_PACK(0x00000002, 2), HUFF_PACK(0x00000002, 4), HUFF_PACK(0x000000fe, 9), HUFF_PACK(0x000007be, 12), HUFF_PACK(0x00000ffc, 13), HUFF_PACK(0x00000ffd, 13), HUFF_PACK(0x00001efe, 15), HUFF_PACK(0x00003dfe, 16)}, {HUFF_PACK(0x00000004, 4), HUFF_PACK(0x00000000, 3), HUFF_PACK(0x0000003c, 7), HUFF_PACK(0x000000f6, 10), HUFF_PACK(0x000001da, 11), HUFF_PACK(0x000003fe, 12), HUFF_PACK(0x00003dfe, 15), HUFF_PACK(0x00003dff, 16)}, {HUFF_PACK(0x0000003c, 8), HUFF_PACK(0x0000003e, 7), HUFF_PACK(0x0000000a, 5), HUFF_PACK(0x0000003a, 8), HUFF_PACK(0x000003de, 11), HUFF_PACK(0x000007be, 13), HUFF_PACK(0x00000f7e, 14), HUFF_PACK(0x00001efe, 14)}, {HUFF_PACK(0x000001de, 11), HUFF_PACK(0x000000ec, 10), HUFF_PACK(0x0000007e, 9), HUFF_PACK(0x0000000c, 5), HUFF_PACK(0x000001ee, 10), HUFF_PACK(0x00000f7e, 13), HUFF_PACK(0x000007fc, 12), HUFF_PACK(0x00003dff, 15)}, {HUFF_PACK(0x00007ffe, 16), HUFF_PACK(0x000003be, 12), HUFF_PACK(0x000000fe, 10), HUFF_PACK(0x000001fe, 10), HUFF_PACK(0x0000001a, 6), HUFF_PACK(0x0000001c, 7), HUFF_PACK(0x000007fd, 12), HUFF_PACK(0x00000ffe, 13)}, {HUFF_PACK(0x00003dff, 16), HUFF_PACK(0x000003bf, 12), HUFF_PACK(0x00001ffe, 14), HUFF_PACK(0x000003ff, 12), HUFF_PACK(0x0000003e, 8), HUFF_PACK(0x0000001b, 6), HUFF_PACK(0x0000007e, 8), HUFF_PACK(0x000000f6, 9)}, {HUFF_PACK(0x00007fff, 16), HUFF_PACK(0x00003dff, 16), HUFF_PACK(0x00003ffe, 15), HUFF_PACK(0x000001db, 11), HUFF_PACK(0x000000ee, 10), HUFF_PACK(0x0000007a, 8), HUFF_PACK(0x0000000e, 5), HUFF_PACK(0x0000000b, 5)}, {HUFF_PACK(0x00003dff, 16), HUFF_PACK(0x00003dff, 16), HUFF_PACK(0x000003de, 12), HUFF_PACK(0x000001fe, 11), HUFF_PACK(0x000001ee, 11), HUFF_PACK(0x0000007a, 9), HUFF_PACK(0x00000006, 5), HUFF_PACK(0x00000003, 2)}}, HUFF_PACK(0x00003dff, 16) /* escape */ }}}}}; const HUFF_PT0_TABLE fdk_sacenc_huffPart0Tab = { {/* CLD */ HUFF_PACK(0x00000052, 8), HUFF_PACK(0x000000ae, 9), HUFF_PACK(0x000000af, 9), HUFF_PACK(0x00000028, 7), HUFF_PACK(0x0000006e, 7), HUFF_PACK(0x00000036, 6), HUFF_PACK(0x0000001e, 5), HUFF_PACK(0x0000000e, 4), HUFF_PACK(0x0000000c, 4), HUFF_PACK(0x0000000a, 4), HUFF_PACK(0x00000002, 4), HUFF_PACK(0x00000016, 5), HUFF_PACK(0x00000012, 5), HUFF_PACK(0x00000017, 5), HUFF_PACK(0x00000000, 4), HUFF_PACK(0x00000004, 4), HUFF_PACK(0x00000006, 4), HUFF_PACK(0x00000008, 4), HUFF_PACK(0x00000007, 4), HUFF_PACK(0x00000003, 4), HUFF_PACK(0x00000001, 4), HUFF_PACK(0x0000001a, 5), HUFF_PACK(0x00000013, 5), HUFF_PACK(0x0000003e, 6), HUFF_PACK(0x00000016, 6), HUFF_PACK(0x00000017, 6), HUFF_PACK(0x0000006f, 7), HUFF_PACK(0x0000002a, 7), HUFF_PACK(0x00000056, 8), HUFF_PACK(0x00000053, 8), HUFF_PACK(0x0000003f, 6)}, {/* ICC */ HUFF_PACK(0x0000001e, 5), HUFF_PACK(0x0000000e, 4), HUFF_PACK(0x00000006, 3), HUFF_PACK(0x00000000, 2), HUFF_PACK(0x00000002, 2), HUFF_PACK(0x00000001, 2), HUFF_PACK(0x0000003e, 6), HUFF_PACK(0x0000003f, 6)}}; /* Function / Class Declarations *********************************************/ /* Function / Class Definition ***********************************************/ fdk-aac-2.0.2/libSACenc/src/sacenc_huff_tab.h000066400000000000000000000167031404225310700206230ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround encoder library ************************* Author(s): Markus Lohwasser Description: SAC-Encoder constant huffman tables *******************************************************************************/ #ifndef SACENC_HUFF_TAB_H #define SACENC_HUFF_TAB_H /* Includes ******************************************************************/ #include "machine_type.h" /* Defines *******************************************************************/ #define HUFF_PACK(a, b) \ { \ ((((ULONG)a) & 0x00FFFFFF) << 8) | (((ULONG)b) & 0xFF) \ } /*!< Pack huffman value and length information. */ #define HUFF_VALUE(a) \ (((a.packed >> 8) & 0x00FFFFFF)) /*!< Return value from packed table entry. \ */ #define HUFF_LENGTH(a) \ ((a.packed & 0xFF)) /*!< Return length from packed table entry. */ /* Data Types ****************************************************************/ /** * \brief This struct contains packed huffman entries. * * The packed entry consist of hffman value and length information. * * |---------------------------------| * | value | length | * |---------------------------------| * |<------- 31...8 ------->|< 7..0 >| */ typedef struct { ULONG packed; /*! Packed huffman entry: - lower 8 bit are reservoed for length information - upper 24 bit contains huffman value */ } HUFF_ENTRY; typedef struct { HUFF_ENTRY entry[2][2]; HUFF_ENTRY escape; } LAV1_2D; typedef struct { HUFF_ENTRY entry[4][4]; HUFF_ENTRY escape; } LAV3_2D; typedef struct { HUFF_ENTRY entry[6][6]; HUFF_ENTRY escape; } LAV5_2D; typedef struct { HUFF_ENTRY entry[7][7]; HUFF_ENTRY escape; } LAV6_2D; typedef struct { HUFF_ENTRY entry[8][8]; HUFF_ENTRY escape; } LAV7_2D; typedef struct { HUFF_ENTRY entry[10][10]; HUFF_ENTRY escape; } LAV9_2D; typedef struct { HUFF_ENTRY entry[13][13]; HUFF_ENTRY escape; } LAV12_2D; typedef struct { LAV3_2D lav3; LAV5_2D lav5; LAV7_2D lav7; LAV9_2D lav9; } HUFF_CLD_TAB_2D; typedef struct { LAV1_2D lav1; LAV3_2D lav3; LAV5_2D lav5; LAV7_2D lav7; } HUFF_ICC_TAB_2D; typedef struct { HUFF_ENTRY h1D[2][31]; HUFF_CLD_TAB_2D h2D[2][2]; } HUFF_CLD_TABLE; typedef struct { HUFF_ENTRY h1D[2][8]; HUFF_ICC_TAB_2D h2D[2][2]; } HUFF_ICC_TABLE; typedef struct { HUFF_ENTRY cld[31]; HUFF_ENTRY icc[8]; } HUFF_PT0_TABLE; typedef HUFF_ENTRY HUFF_RES_TABLE[5][8]; /* Constants *****************************************************************/ extern const HUFF_CLD_TABLE fdk_sacenc_huffCLDTab; extern const HUFF_ICC_TABLE fdk_sacenc_huffICCTab; extern const HUFF_PT0_TABLE fdk_sacenc_huffPart0Tab; /* Function / Class Declarations *********************************************/ #endif /* SACENC_HUFF_TAB_H */ fdk-aac-2.0.2/libSACenc/src/sacenc_lib.cpp000066400000000000000000002111511404225310700201400ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround encoder library ************************* Author(s): Max Neuendorf Description: Encoder Library Interface Interface to Spacial Audio Coding Encoder lib *******************************************************************************/ /**************************************************************************** \file Description of file contents ******************************************************************************/ /* Includes ******************************************************************/ #include "sacenc_lib.h" #include "sacenc_const.h" #include "genericStds.h" #include "FDK_core.h" #include "sacenc_tree.h" #include "sacenc_bitstream.h" #include "sacenc_onsetdetect.h" #include "sacenc_framewindowing.h" #include "sacenc_filter.h" #include "sacenc_paramextract.h" #include "sacenc_staticgain.h" #include "sacenc_delay.h" #include "sacenc_dmx_tdom_enh.h" #include "sacenc_vectorfunctions.h" #include "qmf.h" /* Defines *******************************************************************/ /* Encoder library info */ #define SACENC_LIB_VL0 2 #define SACENC_LIB_VL1 0 #define SACENC_LIB_VL2 0 #define SACENC_LIB_TITLE "MPEG Surround Encoder" #ifdef SUPPRESS_BUILD_DATE_INFO #define SACENC_LIB_BUILD_DATE "" #define SACENC_LIB_BUILD_TIME "" #else #define SACENC_LIB_BUILD_DATE __DATE__ #define SACENC_LIB_BUILD_TIME __TIME__ #endif #define MAX_MPEGS_BYTES (1 << 14) #define MAX_SSC_BYTES (1 << 6) #define MAX_SPACE_TREE_CHANNELS 2 #define NUM_KEEP_WINDOWS 3 /* Data Types ****************************************************************/ typedef struct { MP4SPACEENC_MODE encMode; MP4SPACEENC_BANDS_CONFIG nParamBands; MP4SPACEENC_QUANTMODE quantMode; UCHAR bUseCoarseQuant; UCHAR bLdMode; UCHAR bTimeDomainDmx; UINT sampleRate; UINT frameTimeSlots; /* e.g. 32 when used with HE-AAC */ UINT independencyFactor; /* how often should we set the independency flag */ INT timeAlignment; /* additional delay for downmix */ } MP4SPACEENC_SETUP, *HANDLE_MP4SPACEENC_SETUP; struct ENC_CONFIG_SETUP { UCHAR bEncMode_212; UCHAR maxHybridInStaticSlots; LONG maxSamplingrate; INT maxAnalysisLengthTimeSlots; INT maxHybridBands; INT maxQmfBands; INT maxChIn; INT maxFrameTimeSlots; INT maxFrameLength; INT maxChOut; INT maxChTotOut; }; struct MP4SPACE_ENCODER { MP4SPACEENC_SETUP user; ENC_CONFIG_SETUP setup; /* describe allocated instance */ HANDLE_FRAMEWINDOW hFrameWindow; /* Windowing, only created+updated, but not used */ INT nSamplesValid; /* Input Buffer Handling */ /* Routing Sensible Switches/Variables */ MP4SPACEENC_BANDS_CONFIG nParamBands; UCHAR useTimeDomDownmix; /* not Routing Sensible Switches/Varibles - must be contained in Check */ MP4SPACEENC_MODE encMode; UCHAR bEncMode_212_only; /* not Routing Sensible Switches/Varibles + lower Classes */ UCHAR useFrameKeep; UINT independencyFactor; UINT nSampleRate; UCHAR nInputChannels; UCHAR nOutputChannels; UCHAR nFrameTimeSlots; /* e.g. 32 when used with HE-AAC */ UCHAR nQmfBands; UCHAR nHybridBands; UINT nFrameLength; /* number of output waveform samples/channel/frame */ /* not Routing Sensible Switches/Varibles + lower Classes, secondary computed */ INT nSamplesNext; INT nAnalysisLengthTimeSlots; INT nAnalysisLookaheadTimeSlots; INT nUpdateHybridPositionTimeSlots; INT *pnOutputBits; INT nInputDelay; INT nOutputBufferDelay; INT nSurroundAnalysisBufferDelay; INT nBitstreamDelayBuffer; INT nBitstreamBufferRead; INT nBitstreamBufferWrite; INT nDiscardOutFrames; INT avoid_keep; /* not Routing Sensible Switches/Varibles -> moved to lower Classes */ UCHAR useCoarseQuantCld; /* Only Used in SpaceTreeSetup */ UCHAR useCoarseQuantIcc; /* Only Used in SpaceTreeSetup */ UCHAR useCoarseQuantCpc; /* Only Used in SpaceTreeSetup */ UCHAR useCoarseQuantArbDmx; /* ArbitraryDmx,... not available yet */ MP4SPACEENC_QUANTMODE quantMode; /* Used for quanitzation and in bitstream writer */ INT coreCoderDelay; /* Used in delay compensation */ INT timeAlignment; /* Used in delay compensation */ /* Local Processing Variables */ INT independencyCount; INT independencyFlag; INT **ppTrCurrPos; /* belongs somehow to Onset Detection */ INT trPrevPos[2 * MAX_NUM_TRANS]; /* belongs somehow to Onset Detection */ FRAMEWIN_LIST frameWinList; SPATIALFRAME saveFrame; /* Module-Handles */ SPACE_TREE_SETUP spaceTreeSetup; MPEG4SPACEENC_SSCBUF sscBuf; FIXP_WIN *pFrameWindowAna__FDK[MAX_NUM_PARAMS]; HANDLE_QMF_FILTER_BANK *phQmfFiltIn__FDK; HANDLE_DC_FILTER phDCFilterSigIn[SACENC_MAX_INPUT_CHANNELS]; HANDLE_ONSET_DETECT phOnset[SACENC_MAX_INPUT_CHANNELS]; HANDLE_SPACE_TREE hSpaceTree; HANDLE_BSF_INSTANCE hBitstreamFormatter; HANDLE_STATIC_GAIN_CONFIG hStaticGainConfig; HANDLE_STATIC_GAIN hStaticGain; HANDLE_DELAY hDelay; /* enhanced time domain downmix (for stereo input) */ HANDLE_ENHANCED_TIME_DOMAIN_DMX hEnhancedTimeDmx; /* Data Buffers */ INT_PCM **ppTimeSigIn__FDK; INT_PCM **ppTimeSigDelayIn__FDK; INT_PCM **ppTimeSigOut__FDK; FIXP_DPK ***pppHybridIn__FDK; FIXP_DPK ***pppHybridInStatic__FDK; FIXP_DPK ***pppProcDataIn__FDK; INT_PCM *pOutputDelayBuffer__FDK; UCHAR **ppBitstreamDelayBuffer; UCHAR *pParameterBand2HybridBandOffset; INT staticGainScale; INT *pEncoderInputChScale; INT *staticTimeDomainDmxInScale; }; /* Constants *****************************************************************/ static const UCHAR pValidBands_Ld[8] = {4, 5, 7, 9, 12, 15, 23, 40}; static const UCHAR qmf2qmf[] = /* Bypass the HybridAnylyis/Synthesis*/ {0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23, 24, 25, 26, 27, 28, 29, 30, 31, 32, 33, 34, 35, 36, 37, 38, 39, 40, 41, 42, 43, 44, 45, 46, 47, 48, 49, 50, 51, 52, 53, 54, 55, 56, 57, 58, 59, 60, 61, 62, 63, 64, 65, 66, 67, 68, 69, 70, 71, 72, 73, 74, 75, 76, 77, 78, 79, 80, 81, 82, 83, 84, 85, 86, 87, 88, 89, 90, 91, 92, 93, 94, 95, 96, 97, 98, 99, 100, 101, 102, 103, 104, 105, 106, 107, 108, 109, 110, 111, 112, 113, 114, 115, 116, 117, 118, 119, 120, 121, 122, 123, 124, 125, 126, 127}; /* Function / Class Declarations *********************************************/ static FDK_SACENC_ERROR mp4SpaceEnc_create( HANDLE_MP4SPACE_ENCODER *phMp4SpaceEnc); static FDK_SACENC_ERROR FillSpatialSpecificConfig( const HANDLE_MP4SPACE_ENCODER hEnc, SPATIALSPECIFICCONFIG *const hSsc); static FDK_SACENC_ERROR mp4SpaceEnc_FillSpaceTreeSetup( const HANDLE_MP4SPACE_ENCODER hEnc, SPACE_TREE_SETUP *const hSpaceTreeSetup); static FDK_SACENC_ERROR mp4SpaceEnc_InitDelayCompensation( HANDLE_MP4SPACE_ENCODER hMp4SpaceEnc, const INT coreCoderDelay); static FDK_SACENC_ERROR mp4SpaceEnc_InitDefault( HANDLE_MP4SPACE_ENCODER hMp4SpaceEnc); static DECORRCONFIG mp4SpaceEnc_GetDecorrConfig(const MP4SPACEENC_MODE encMode); static FDK_SACENC_ERROR mp4SpaceEnc_InitNumParamBands( HANDLE_MP4SPACE_ENCODER hEnc, const MP4SPACEENC_BANDS_CONFIG nParamBands); /* Function / Class Definition ***********************************************/ static UINT mp4SpaceEnc_GetNumQmfBands(const UINT nSampleRate) { UINT nQmfBands = 0; if (nSampleRate < 27713) nQmfBands = 32; else if (nSampleRate < 55426) nQmfBands = 64; return nQmfBands; } static UINT updateQmfFlags(const UINT flags, const INT keepStates) { UINT qmfFlags = flags; qmfFlags = (qmfFlags & (~(UINT)QMF_FLAG_LP)); qmfFlags = (qmfFlags | QMF_FLAG_MPSLDFB); qmfFlags = (keepStates) ? (qmfFlags | QMF_FLAG_KEEP_STATES) : (qmfFlags & (~(UINT)QMF_FLAG_KEEP_STATES)); return qmfFlags; } static INT freq2HybridBand(const UINT nFrequency, const UINT nSampleRate, const UINT nQmfBands) { /* nQmfSlotWidth = (nSampleRate/2) / nQmfBands; nQmfBand = nFrequency / nQmfSlotWidth; */ int nHybridBand = -1; int scale = 0; const FIXP_DBL temp = fDivNorm((FIXP_DBL)(2 * nFrequency * nQmfBands), (FIXP_DBL)nSampleRate, &scale); const int nQmfBand = scaleValue(temp, scale - (DFRACT_BITS - 1)); if ((nQmfBand > -1) && (nQmfBand < (int)nQmfBands)) { nHybridBand = qmf2qmf[nQmfBand]; } return nHybridBand; } /* * Examine buffer descriptor regarding choosen type. * * \param pBufDesc Pointer to buffer descriptor * \param type Buffer type to look for. * \return - Buffer descriptor index. * -1, if there is no entry available. */ static INT getBufDescIdx(const FDK_bufDescr *pBufDesc, const UINT type) { INT i, idx = -1; for (i = 0; i < (int)pBufDesc->numBufs; i++) { if (pBufDesc->pBufType[i] == type) { idx = i; break; } } return idx; } FDK_SACENC_ERROR FDK_sacenc_open(HANDLE_MP4SPACE_ENCODER *phMp4SpaceEnc) { return mp4SpaceEnc_create(phMp4SpaceEnc); } static FDK_SACENC_ERROR mp4SpaceEnc_create( HANDLE_MP4SPACE_ENCODER *phMp4SpaceEnc) { FDK_SACENC_ERROR error = SACENC_OK; HANDLE_MP4SPACE_ENCODER hEnc = NULL; ENC_CONFIG_SETUP setup; if (NULL == phMp4SpaceEnc) { error = SACENC_INVALID_HANDLE; } else { int i, ch; FDKmemclear(&setup, sizeof(ENC_CONFIG_SETUP)); /* Allocate Encoder Instance */ FDK_ALLOCATE_MEMORY_1D(hEnc, 1, struct MP4SPACE_ENCODER); /* Clear everything, also pointers. */ if (NULL != hEnc) { FDKmemclear(hEnc, sizeof(struct MP4SPACE_ENCODER)); } setup.maxSamplingrate = 48000; setup.maxFrameTimeSlots = 16; setup.maxAnalysisLengthTimeSlots = 3 * setup.maxFrameTimeSlots; setup.maxQmfBands = mp4SpaceEnc_GetNumQmfBands(setup.maxSamplingrate); ; setup.maxHybridBands = setup.maxQmfBands; setup.maxFrameLength = setup.maxQmfBands * setup.maxFrameTimeSlots; setup.maxChIn = 2; setup.maxChOut = 1; setup.maxChTotOut = setup.maxChOut; setup.bEncMode_212 = 1; setup.maxHybridInStaticSlots = 24; /* Open Static Gain*/ if (SACENC_OK != (error = fdk_sacenc_staticGain_OpenConfig(&hEnc->hStaticGainConfig))) { goto bail; } /* enhanced time domain downmix (for stereo input) */ if (SACENC_OK != (error = fdk_sacenc_open_enhancedTimeDomainDmx( &hEnc->hEnhancedTimeDmx, setup.maxFrameLength))) { goto bail; } FDK_ALLOCATE_MEMORY_1D(hEnc->pParameterBand2HybridBandOffset, MAX_NUM_PARAM_BANDS, UCHAR); /* Create Space Tree first, to get number of in-/output channels */ if (SACENC_OK != (error = fdk_sacenc_spaceTree_Open(&hEnc->hSpaceTree))) { goto bail; } FDK_ALLOCATE_MEMORY_1D(hEnc->pEncoderInputChScale, setup.maxChIn, INT); FDK_ALLOCATE_MEMORY_1D(hEnc->staticTimeDomainDmxInScale, setup.maxChIn, INT); FDK_ALLOCATE_MEMORY_1D(hEnc->phQmfFiltIn__FDK, setup.maxChIn, HANDLE_QMF_FILTER_BANK); /* Allocate Analysis Filterbank Structs */ for (ch = 0; ch < setup.maxChIn; ch++) { FDK_ALLOCATE_MEMORY_1D_INT(hEnc->phQmfFiltIn__FDK[ch], 1, struct QMF_FILTER_BANK, SECT_DATA_L2) FDK_ALLOCATE_MEMORY_1D_INT(hEnc->phQmfFiltIn__FDK[ch]->FilterStates, 2 * 5 * setup.maxQmfBands, FIXP_QAS, SECT_DATA_L2) } /* Allocate Synthesis Filterbank Structs for arbitrary downmix */ /* Allocate DC Filter Struct for normal signal input */ for (ch = 0; ch < setup.maxChIn; ch++) { if (SACENC_OK != (error = fdk_sacenc_createDCFilter(&hEnc->phDCFilterSigIn[ch]))) { goto bail; } } /* Open Onset Detection */ for (ch = 0; ch < setup.maxChIn; ch++) { if (SACENC_OK != (error = fdk_sacenc_onsetDetect_Open( &hEnc->phOnset[ch], setup.maxFrameTimeSlots))) { goto bail; } } FDK_ALLOCATE_MEMORY_2D(hEnc->ppTrCurrPos, setup.maxChIn, MAX_NUM_TRANS, INT); /* Create Windowing */ if (SACENC_OK != (error = fdk_sacenc_frameWindow_Create(&hEnc->hFrameWindow))) { goto bail; } /* Open static gain */ if (SACENC_OK != (error = fdk_sacenc_staticGain_Open(&hEnc->hStaticGain))) { goto bail; } /* create bitstream encoder */ if (SACENC_OK != (error = fdk_sacenc_createSpatialBitstreamEncoder( &hEnc->hBitstreamFormatter))) { goto bail; } FDK_ALLOCATE_MEMORY_1D(hEnc->sscBuf.pSsc, MAX_SSC_BYTES, UCHAR); { FDK_ALLOCATE_MEMORY_2D(hEnc->ppTimeSigIn__FDK, setup.maxChIn, setup.maxFrameLength + MAX_DELAY_SURROUND_ANALYSIS, INT_PCM); } FDK_ALLOCATE_MEMORY_2D(hEnc->ppTimeSigDelayIn__FDK, setup.maxChIn, MAX_DELAY_SURROUND_ANALYSIS, INT_PCM); /* Create new buffers for several signals (including arbitrary downmix) */ if (setup.bEncMode_212 == 0) { /* pOutputDelayBuffer__FDK buffer is not needed for SACENC_212 mode */ FDK_ALLOCATE_MEMORY_1D( hEnc->pOutputDelayBuffer__FDK, (setup.maxFrameLength + MAX_DELAY_OUTPUT) * setup.maxChOut, INT_PCM); } /* allocate buffers */ if (setup.bEncMode_212 == 0) { /* ppTimeSigOut__FDK buffer is not needed for SACENC_212 mode */ FDK_ALLOCATE_MEMORY_2D(hEnc->ppTimeSigOut__FDK, setup.maxChTotOut, setup.maxFrameLength, INT_PCM); } if (setup.bEncMode_212 == 1) { /* pppHybridIn__FDK buffer can be reduced by maxFrameTimeSlots/2 slots for * SACENC_212 mode */ FDK_ALLOCATE_MEMORY_3D( hEnc->pppHybridIn__FDK, setup.maxChIn, setup.maxAnalysisLengthTimeSlots - (setup.maxFrameTimeSlots >> 1), setup.maxHybridBands, FIXP_DPK); FDK_ALLOCATE_MEMORY_3D(hEnc->pppHybridInStatic__FDK, setup.maxChIn, setup.maxHybridInStaticSlots, setup.maxHybridBands, FIXP_DPK); } else { FDK_ALLOCATE_MEMORY_3D(hEnc->pppHybridIn__FDK, setup.maxChIn, setup.maxAnalysisLengthTimeSlots, setup.maxHybridBands, FIXP_DPK); } if (setup.bEncMode_212 == 0) { /* pppProcDataIn__FDK buffer is not needed for SACENC_212 mode */ FDK_ALLOCATE_MEMORY_3D(hEnc->pppProcDataIn__FDK, MAX_SPACE_TREE_CHANNELS, setup.maxAnalysisLengthTimeSlots, setup.maxHybridBands, FIXP_DPK); } for (i = 0; i < MAX_NUM_PARAMS; i++) { FDK_ALLOCATE_MEMORY_1D(hEnc->pFrameWindowAna__FDK[i], setup.maxAnalysisLengthTimeSlots, FIXP_WIN); } /* for i */ if (SACENC_OK != (error = fdk_sacenc_delay_Open(&hEnc->hDelay))) { goto bail; } if (setup.bEncMode_212 == 0) { /* ppBitstreamDelayBuffer buffer is not needed for SACENC_212 mode */ FDK_ALLOCATE_MEMORY_2D(hEnc->ppBitstreamDelayBuffer, MAX_BITSTREAM_DELAY, MAX_MPEGS_BYTES, UCHAR); } FDK_ALLOCATE_MEMORY_1D(hEnc->pnOutputBits, MAX_BITSTREAM_DELAY, INT); hEnc->setup = setup; /* save configuration used while encoder allocation. */ mp4SpaceEnc_InitDefault(hEnc); if (NULL != phMp4SpaceEnc) { *phMp4SpaceEnc = hEnc; /* return encoder handle */ } } /* valid handle */ return error; bail: if (NULL != hEnc) { hEnc->setup = setup; FDK_sacenc_close(&hEnc); } return ((SACENC_OK == error) ? SACENC_MEMORY_ERROR : error); } static FDK_SACENC_ERROR mp4SpaceEnc_InitDefault(HANDLE_MP4SPACE_ENCODER hEnc) { FDK_SACENC_ERROR err = SACENC_OK; /* Get default static gain configuration. */ if (SACENC_OK != (err = fdk_sacenc_staticGain_InitDefaultConfig( hEnc->hStaticGainConfig))) { goto bail; } bail: return err; } static FDK_SACENC_ERROR FDK_sacenc_configure( HANDLE_MP4SPACE_ENCODER hEnc, const HANDLE_MP4SPACEENC_SETUP hSetup) { FDK_SACENC_ERROR error = SACENC_OK; hEnc->nSampleRate = hSetup->sampleRate; hEnc->encMode = hSetup->encMode; hEnc->nQmfBands = mp4SpaceEnc_GetNumQmfBands(hEnc->nSampleRate); /* Make sure that we have set time domain downmix for 212 */ if (hSetup->encMode == SACENC_212 && hSetup->bTimeDomainDmx == 0) { error = SACENC_INVALID_CONFIG; } else { hEnc->useTimeDomDownmix = hSetup->bTimeDomainDmx; } hEnc->timeAlignment = hSetup->timeAlignment; hEnc->quantMode = hSetup->quantMode; hEnc->useCoarseQuantCld = hSetup->bUseCoarseQuant; hEnc->useCoarseQuantCpc = hSetup->bUseCoarseQuant; hEnc->useFrameKeep = (hSetup->bLdMode == 2); hEnc->useCoarseQuantIcc = 0; /* not available */ hEnc->useCoarseQuantArbDmx = 0; /* not available for user right now */ hEnc->independencyFactor = hSetup->independencyFactor; hEnc->independencyCount = 0; hEnc->independencyFlag = 1; /* set number of Hybrid bands */ hEnc->nHybridBands = hEnc->nQmfBands; hEnc->nFrameTimeSlots = hSetup->frameTimeSlots; mp4SpaceEnc_InitNumParamBands(hEnc, hSetup->nParamBands); return error; } FDK_SACENC_ERROR FDK_sacenc_init(HANDLE_MP4SPACE_ENCODER hEnc, const INT dmxDelay) { FDK_SACENC_ERROR error = SACENC_OK; /* Sanity Checks */ if (NULL == hEnc) { error = SACENC_INVALID_HANDLE; } else { const int initStatesFlag = 1; int ch; /* loop counter */ int nChInArbDmx; if (SACENC_OK != (error = FDK_sacenc_configure(hEnc, &hEnc->user))) { goto bail; } hEnc->bEncMode_212_only = hEnc->setup.bEncMode_212; /* Slots per Frame and Frame Length */ if (hEnc->nFrameTimeSlots < 1) { error = SACENC_INVALID_CONFIG; goto bail; } hEnc->nFrameLength = hEnc->nQmfBands * hEnc->nFrameTimeSlots; if (hEnc->useFrameKeep == 1) { hEnc->nAnalysisLengthTimeSlots = 3 * hEnc->nFrameTimeSlots; hEnc->nUpdateHybridPositionTimeSlots = hEnc->nFrameTimeSlots; } else { hEnc->nAnalysisLengthTimeSlots = 2 * hEnc->nFrameTimeSlots; hEnc->nUpdateHybridPositionTimeSlots = 0; } { hEnc->nAnalysisLookaheadTimeSlots = hEnc->nAnalysisLengthTimeSlots - 3 * hEnc->nFrameTimeSlots / 2; } /* init parameterBand2hybridBandOffset table */ fdk_sacenc_calcParameterBand2HybridBandOffset( (BOX_SUBBAND_CONFIG)hEnc->nParamBands, hEnc->nHybridBands, hEnc->pParameterBand2HybridBandOffset); /* Fill Setup structure for Space Tree */ if (SACENC_OK != (error = mp4SpaceEnc_FillSpaceTreeSetup(hEnc, &hEnc->spaceTreeSetup))) { goto bail; } /* Init space tree configuration */ if (SACENC_OK != (error = fdk_sacenc_spaceTree_Init( hEnc->hSpaceTree, &hEnc->spaceTreeSetup, hEnc->pParameterBand2HybridBandOffset, hEnc->useFrameKeep))) { goto bail; } /* Get space tree description and resulting number of input/output channels */ { SPACE_TREE_DESCRIPTION spaceTreeDescription; if (SACENC_OK != (error = fdk_sacenc_spaceTree_GetDescription( hEnc->hSpaceTree, &spaceTreeDescription))) { goto bail; } hEnc->nInputChannels = spaceTreeDescription.nOutChannels; /* space tree description describes decoder configuration */ hEnc->nOutputChannels = spaceTreeDescription.nInChannels; /* space tree description describes decoder configuration */ } nChInArbDmx = 0; /* INITIALIZATION */ for (ch = 0; ch < hEnc->nInputChannels; ch++) { /* scaling in analysis qmf filterbank (7) */ hEnc->pEncoderInputChScale[ch] = 7; { /* additional scaling in qmf prototype filter for low delay */ hEnc->pEncoderInputChScale[ch] += 1; } { hEnc->pEncoderInputChScale[ch] += DC_FILTER_SF; } } /* nInputChannels */ /* Init analysis filterbank */ for (ch = 0; ch < hEnc->nInputChannels; ch++) { hEnc->phQmfFiltIn__FDK[ch]->flags = updateQmfFlags(hEnc->phQmfFiltIn__FDK[ch]->flags, !initStatesFlag); if (0 != qmfInitAnalysisFilterBank( hEnc->phQmfFiltIn__FDK[ch], (FIXP_QAS *)hEnc->phQmfFiltIn__FDK[ch]->FilterStates, 1, hEnc->nQmfBands, hEnc->nQmfBands, hEnc->nQmfBands, hEnc->phQmfFiltIn__FDK[ch]->flags)) { error = SACENC_INIT_ERROR; goto bail; } } /* Initialize DC Filter. */ { for (ch = 0; ch < hEnc->nInputChannels; ch++) { if (SACENC_OK != (error = fdk_sacenc_initDCFilter( hEnc->phDCFilterSigIn[ch], hEnc->nSampleRate))) { goto bail; } } } /* Init onset detect. */ { /* init onset detect configuration struct */ ONSET_DETECT_CONFIG onsetDetectConfig; onsetDetectConfig.maxTimeSlots = hEnc->nFrameTimeSlots; onsetDetectConfig.lowerBoundOnsetDetection = freq2HybridBand(1725, hEnc->nSampleRate, hEnc->nQmfBands); onsetDetectConfig.upperBoundOnsetDetection = hEnc->nHybridBands; for (ch = 0; ch < hEnc->nInputChannels; ch++) { if (SACENC_OK != (error = fdk_sacenc_onsetDetect_Init( hEnc->phOnset[ch], &onsetDetectConfig, 1))) { goto bail; } } } { /* init windowing */ FRAMEWINDOW_CONFIG framewindowConfig; framewindowConfig.nTimeSlotsMax = hEnc->nFrameTimeSlots; framewindowConfig.bFrameKeep = hEnc->useFrameKeep; if (SACENC_OK != (error = fdk_sacenc_frameWindow_Init( hEnc->hFrameWindow, &framewindowConfig))) { goto bail; } } /* Set encoder mode for static gain initialization. */ if (SACENC_OK != (error = fdk_sacenc_staticGain_SetEncMode( hEnc->hStaticGainConfig, hEnc->encMode))) { goto bail; } /* Init static gain. */ if (SACENC_OK != (error = fdk_sacenc_staticGain_Init( hEnc->hStaticGain, hEnc->hStaticGainConfig, &(hEnc->staticGainScale)))) { goto bail; } for (ch = 0; ch < hEnc->nInputChannels; ch++) { hEnc->pEncoderInputChScale[ch] += hEnc->staticGainScale; } /* enhanced downmix for stereo input*/ if (hEnc->useTimeDomDownmix != 0) { if (SACENC_OK != (error = fdk_sacenc_init_enhancedTimeDomainDmx( hEnc->hEnhancedTimeDmx, fdk_sacenc_getPreGainPtrFDK(hEnc->hStaticGain), hEnc->staticGainScale, fdk_sacenc_getPostGainFDK(hEnc->hStaticGain), hEnc->staticGainScale, hEnc->nFrameLength))) { goto bail; } } /* Create config structure for bitstream formatter including arbitrary * downmix residual */ if (SACENC_OK != (error = fdk_sacenc_initSpatialBitstreamEncoder( hEnc->hBitstreamFormatter))) { goto bail; } if (SACENC_OK != (error = FillSpatialSpecificConfig( hEnc, fdk_sacenc_getSpatialSpecificConfig( hEnc->hBitstreamFormatter)))) { goto bail; } if (SACENC_OK != (error = fdk_sacenc_writeSpatialSpecificConfig( fdk_sacenc_getSpatialSpecificConfig(hEnc->hBitstreamFormatter), hEnc->sscBuf.pSsc, MAX_SSC_BYTES, &hEnc->sscBuf.nSscSizeBits))) { goto bail; } /* init delay compensation with dmx core coder delay; if no core coder is * used, many other buffers are initialized nevertheless */ if (SACENC_OK != (error = mp4SpaceEnc_InitDelayCompensation(hEnc, dmxDelay))) { goto bail; } /* How much input do we need? */ hEnc->nSamplesNext = hEnc->nFrameLength * (hEnc->nInputChannels + nChInArbDmx); hEnc->nSamplesValid = 0; } /* valid handle */ bail: return error; } static INT getAnalysisLengthTimeSlots(FIXP_WIN *pFrameWindowAna, INT nTimeSlots) { int i; for (i = nTimeSlots - 1; i >= 0; i--) { if (pFrameWindowAna[i] != (FIXP_WIN)0) { break; } } nTimeSlots = i + 1; return nTimeSlots; } static INT getAnalysisStartTimeSlot(FIXP_WIN *pFrameWindowAna, INT nTimeSlots) { int startTimeSlot = 0; int i; for (i = 0; i < nTimeSlots; i++) { if (pFrameWindowAna[i] != (FIXP_WIN)0) { break; } } startTimeSlot = i; return startTimeSlot; } static FDK_SACENC_ERROR __FeedDeinterPreScale( HANDLE_MP4SPACE_ENCODER hEnc, INT_PCM const *const pSamples, INT_PCM *const pOutputSamples, INT const nSamples, UINT const isInputInterleaved, UINT const inputBufferSizePerChannel, UINT *const pnSamplesFed) { FDK_SACENC_ERROR error = SACENC_OK; if ((hEnc == NULL) || (pSamples == NULL) || (pnSamplesFed == NULL)) { error = SACENC_INVALID_HANDLE; } else if (nSamples == 0) { error = SACENC_INVALID_CONFIG; /* Flushing not implemented */ } else { int ch; const INT nChIn = hEnc->nInputChannels; const INT nChInWithDmx = nChIn; const INT samplesToFeed = FDKmin(nSamples, hEnc->nSamplesNext - hEnc->nSamplesValid); const INT nSamplesPerChannel = samplesToFeed / nChInWithDmx; if ((samplesToFeed < 0) || (samplesToFeed % nChInWithDmx != 0) || (samplesToFeed > nChInWithDmx * (INT)hEnc->nFrameLength)) { error = SACENC_INVALID_CONFIG; goto bail; } int i; const INT_PCM *pInput__FDK; const INT_PCM *pInput2__FDK; { /* no dmx align = default*/ pInput__FDK = pSamples; pInput2__FDK = pSamples + (hEnc->nInputDelay * nChInWithDmx); } for (i = 0; i < hEnc->nInputChannels; i++) { hEnc->staticTimeDomainDmxInScale[i] = hEnc->staticGainScale; } /***** N-channel-input *****/ for (ch = 0; ch < nChIn; ch++) { /* Write delayed time signal into time signal buffer */ FDKmemcpy(&(hEnc->ppTimeSigIn__FDK[ch][0]), &(hEnc->ppTimeSigDelayIn__FDK[ch][0]), hEnc->nSurroundAnalysisBufferDelay * sizeof(INT_PCM)); if (isInputInterleaved) { /* Add the new frame de-interleaved. Apply nSurroundAnalysisBufferDelay. */ FDKmemcpy_flex( &(hEnc->ppTimeSigIn__FDK[ch][hEnc->nSurroundAnalysisBufferDelay]), 1, pInput__FDK + ch, nChInWithDmx, hEnc->nInputDelay); FDKmemcpy_flex( &(hEnc->ppTimeSigIn__FDK[ch][hEnc->nSurroundAnalysisBufferDelay + hEnc->nInputDelay]), 1, pInput2__FDK + ch, nChInWithDmx, nSamplesPerChannel - hEnc->nInputDelay); } else { /* Input is already deinterleaved, just copy */ FDKmemcpy( &(hEnc->ppTimeSigIn__FDK[ch][hEnc->nSurroundAnalysisBufferDelay]), pInput__FDK + ch * inputBufferSizePerChannel, hEnc->nInputDelay * sizeof(INT_PCM)); FDKmemcpy( &(hEnc->ppTimeSigIn__FDK[ch][hEnc->nSurroundAnalysisBufferDelay + hEnc->nInputDelay]), pInput2__FDK + ch * inputBufferSizePerChannel, (nSamplesPerChannel - hEnc->nInputDelay) * sizeof(INT_PCM)); } /* Update time signal delay buffer */ FDKmemcpy(&(hEnc->ppTimeSigDelayIn__FDK[ch][0]), &(hEnc->ppTimeSigIn__FDK[ch][hEnc->nFrameLength]), hEnc->nSurroundAnalysisBufferDelay * sizeof(INT_PCM)); } /* for ch */ /***** No Arbitrary Downmix *****/ /* "Crude TD Dmx": Time DomainDownmix + NO Arbitrary Downmix, Delay Added at * pOutputBuffer */ if ((hEnc->useTimeDomDownmix > 0)) { if ((hEnc->useTimeDomDownmix == 1) || (hEnc->nInputChannels != 2)) { error = SACENC_INVALID_CONFIG; goto bail; } else { /* enhanced time domain downmix (for stereo input) */ if (hEnc->encMode == SACENC_212) { if (pOutputSamples == NULL) { error = SACENC_INVALID_HANDLE; goto bail; } fdk_sacenc_apply_enhancedTimeDomainDmx( hEnc->hEnhancedTimeDmx, hEnc->ppTimeSigIn__FDK, pOutputSamples, hEnc->nSurroundAnalysisBufferDelay); } else { if (&hEnc->ppTimeSigOut__FDK[0][0] == NULL) { error = SACENC_INVALID_HANDLE; goto bail; } fdk_sacenc_apply_enhancedTimeDomainDmx( hEnc->hEnhancedTimeDmx, hEnc->ppTimeSigIn__FDK, &hEnc->ppTimeSigOut__FDK[0][0], hEnc->nSurroundAnalysisBufferDelay); } } } /* update number of samples still to process */ hEnc->nSamplesValid += samplesToFeed; /*return number of fed samples */ *pnSamplesFed = samplesToFeed; } bail: return error; } FDK_SACENC_ERROR FDK_sacenc_encode(const HANDLE_MP4SPACE_ENCODER hMp4SpaceEnc, const FDK_bufDescr *inBufDesc, const FDK_bufDescr *outBufDesc, const SACENC_InArgs *inargs, SACENC_OutArgs *outargs) { FDK_SACENC_ERROR error = SACENC_OK; const INT_PCM *pInputSamples = (const INT_PCM *)inBufDesc->ppBase[getBufDescIdx( inBufDesc, (FDK_BUF_TYPE_INPUT | FDK_BUF_TYPE_PCM_DATA))]; INT_PCM *const pOutputSamples = (INT_PCM *)outBufDesc->ppBase[getBufDescIdx( outBufDesc, (FDK_BUF_TYPE_OUTPUT | FDK_BUF_TYPE_PCM_DATA))]; const int nOutputSamplesBufferSize = outBufDesc->pBufSize[getBufDescIdx( outBufDesc, (FDK_BUF_TYPE_OUTPUT | FDK_BUF_TYPE_PCM_DATA))] / outBufDesc->pEleSize[getBufDescIdx( outBufDesc, (FDK_BUF_TYPE_OUTPUT | FDK_BUF_TYPE_PCM_DATA))]; if ((hMp4SpaceEnc == NULL) || (pInputSamples == NULL)) { error = SACENC_INVALID_HANDLE; } else { int nOutputSamples; int i, ch, ps, winCnt, ts, slot; INT currTransPos = -1; SPATIALFRAME *pFrameData = NULL; /* Improve Code Readability */ const int nChIn = hMp4SpaceEnc->nInputChannels; const int nChInWithDmx = nChIn; const int nChOut = hMp4SpaceEnc->nOutputChannels; const int nSamplesPerChannel = inargs->nInputSamples / nChInWithDmx; const int nOutputSamplesMax = nSamplesPerChannel * nChOut; const int nFrameTimeSlots = hMp4SpaceEnc->nFrameTimeSlots; INT encoderInputChScale[SACENC_MAX_INPUT_CHANNELS]; INT nFrameTimeSlotsReduction = 0; if (hMp4SpaceEnc->encMode == SACENC_212) { nFrameTimeSlotsReduction = hMp4SpaceEnc->nFrameTimeSlots >> 1; } for (i = 0; i < nChIn; i++) encoderInputChScale[i] = hMp4SpaceEnc->pEncoderInputChScale[i]; /* Sanity Check */ if ((0 != inargs->nInputSamples % nChInWithDmx)) { error = SACENC_INVALID_CONFIG; goto bail; } /* * Get Frame Data Handle. */ /* get bitstream handle (for storage of cld's, icc's and so on) * get spatialframe 2 frames in the future; NOTE: this is necessary to * synchronise spatial data and audio data */ if (NULL == (pFrameData = fdk_sacenc_getSpatialFrame( hMp4SpaceEnc->hBitstreamFormatter, WRITE_SPATIALFRAME))) { error = SACENC_INVALID_HANDLE; goto bail; } /* Independent Frames Counters*/ if (hMp4SpaceEnc->nDiscardOutFrames > 0) { /* Independent Frames if they should be discarded, Reset Counter*/ hMp4SpaceEnc->independencyCount = 0; /* Reset the counter, first valid frame is an independent one*/ hMp4SpaceEnc->independencyFlag = 1; } else { /*hMp4SpaceEnc->nDiscardOutFrames == 0*/ hMp4SpaceEnc->independencyFlag = (hMp4SpaceEnc->independencyCount == 0) ? 1 : 0; if (hMp4SpaceEnc->independencyFactor > 0) { hMp4SpaceEnc->independencyCount++; hMp4SpaceEnc->independencyCount = hMp4SpaceEnc->independencyCount % ((int)hMp4SpaceEnc->independencyFactor); } else { /* independencyFactor == 0 */ hMp4SpaceEnc->independencyCount = -1; } } /* * Time signal preprocessing: * - Feed input buffer * - Prescale time signal * - Apply DC filter on input signal */ /* Feed, Deinterleave, Pre-Scale the input time signals */ if (SACENC_OK != (error = __FeedDeinterPreScale( hMp4SpaceEnc, pInputSamples, pOutputSamples, inargs->nInputSamples, inargs->isInputInterleaved, inargs->inputBufferSizePerChannel, &outargs->nSamplesConsumed))) { goto bail; } if (hMp4SpaceEnc->nSamplesNext != hMp4SpaceEnc->nSamplesValid) { error = SACENC_INVALID_CONFIG; goto bail; } if (hMp4SpaceEnc->encMode == SACENC_212 && hMp4SpaceEnc->bEncMode_212_only) { for (ch = 0; ch < nChIn; ch++) { for (slot = 0; slot < nFrameTimeSlots; slot++) { setCplxVec( hMp4SpaceEnc->pppHybridIn__FDK [ch][hMp4SpaceEnc->nUpdateHybridPositionTimeSlots + nFrameTimeSlots - nFrameTimeSlotsReduction + slot], (FIXP_DBL)0, hMp4SpaceEnc->nHybridBands); } } } /* * Time / Frequency: * - T/F audio input channels * - T/F arbitrary downmix input channels */ for (ch = 0; ch < nChIn; ch++) { C_AALLOC_SCRATCH_START(pQmfInReal, FIXP_DBL, MAX_QMF_BANDS) C_AALLOC_SCRATCH_START(pQmfInImag, FIXP_DBL, MAX_QMF_BANDS) FIXP_GAIN *pPreGain = fdk_sacenc_getPreGainPtrFDK(hMp4SpaceEnc->hStaticGain); for (ts = 0; ts < nFrameTimeSlots; ts++) { FIXP_DBL *pSpecReal; FIXP_DBL *pSpecImag; INT_PCM *pTimeIn = &hMp4SpaceEnc->ppTimeSigIn__FDK[ch][(ts * hMp4SpaceEnc->nQmfBands)]; { /* Apply DC filter on input channels */ if (SACENC_OK != (error = fdk_sacenc_applyDCFilter( hMp4SpaceEnc->phDCFilterSigIn[ch], pTimeIn, pTimeIn, hMp4SpaceEnc->nQmfBands))) { goto bail; } } /* QMF filterbank */ C_ALLOC_SCRATCH_START(pWorkBuffer, FIXP_DBL, (MAX_QMF_BANDS << 1)); qmfAnalysisFilteringSlot(hMp4SpaceEnc->phQmfFiltIn__FDK[ch], pQmfInReal, pQmfInImag, pTimeIn, 1, pWorkBuffer); C_ALLOC_SCRATCH_END(pWorkBuffer, FIXP_DBL, (MAX_QMF_BANDS << 1)); pSpecReal = pQmfInReal; pSpecImag = pQmfInImag; /* Apply pre-scale after filterbank */ if (MAXVAL_GAIN != pPreGain[ch]) { for (i = 0; i < hMp4SpaceEnc->nHybridBands; i++) { hMp4SpaceEnc ->pppHybridIn__FDK[ch] [hMp4SpaceEnc->nAnalysisLookaheadTimeSlots + ts][i] .v.re = fMult(pSpecReal[i], pPreGain[ch]); hMp4SpaceEnc ->pppHybridIn__FDK[ch] [hMp4SpaceEnc->nAnalysisLookaheadTimeSlots + ts][i] .v.im = fMult(pSpecImag[i], pPreGain[ch]); } } else { for (i = 0; i < hMp4SpaceEnc->nHybridBands; i++) { hMp4SpaceEnc ->pppHybridIn__FDK[ch] [hMp4SpaceEnc->nAnalysisLookaheadTimeSlots + ts][i] .v.re = pSpecReal[i]; hMp4SpaceEnc ->pppHybridIn__FDK[ch] [hMp4SpaceEnc->nAnalysisLookaheadTimeSlots + ts][i] .v.im = pSpecImag[i]; } } } /* ts */ C_AALLOC_SCRATCH_END(pQmfInImag, FIXP_DBL, MAX_QMF_BANDS) C_AALLOC_SCRATCH_END(pQmfInReal, FIXP_DBL, MAX_QMF_BANDS) if (SACENC_OK != error) { goto bail; } } /* ch */ if (hMp4SpaceEnc->encMode == SACENC_212 && hMp4SpaceEnc->bEncMode_212_only) { for (ch = 0; ch < nChIn; ch++) { for (slot = 0; slot < (int)(hMp4SpaceEnc->nUpdateHybridPositionTimeSlots + nFrameTimeSlots - nFrameTimeSlotsReduction); slot++) { copyCplxVec(hMp4SpaceEnc->pppHybridIn__FDK[ch][slot], hMp4SpaceEnc->pppHybridInStatic__FDK[ch][slot], hMp4SpaceEnc->nHybridBands); } } for (ch = 0; ch < nChIn; ch++) { for (slot = 0; slot < (int)(hMp4SpaceEnc->nUpdateHybridPositionTimeSlots + nFrameTimeSlots - nFrameTimeSlotsReduction); slot++) { copyCplxVec( hMp4SpaceEnc->pppHybridInStatic__FDK[ch][slot], hMp4SpaceEnc->pppHybridIn__FDK[ch][nFrameTimeSlots + slot], hMp4SpaceEnc->nHybridBands); } } } /* * Onset Detection: * - detection of transients * - build framing */ for (ch = 0; ch < nChIn; ch++) { if (ch != 3) { /* !LFE */ if (SACENC_OK != (error = fdk_sacenc_onsetDetect_Apply( hMp4SpaceEnc->phOnset[ch], nFrameTimeSlots, hMp4SpaceEnc->nHybridBands, &hMp4SpaceEnc->pppHybridIn__FDK [ch][hMp4SpaceEnc->nAnalysisLookaheadTimeSlots], encoderInputChScale[ch], hMp4SpaceEnc->trPrevPos[1], /* contains previous Transient */ hMp4SpaceEnc->ppTrCurrPos[ch]))) { goto bail; } if ((1) && (hMp4SpaceEnc->useFrameKeep == 0)) { hMp4SpaceEnc->ppTrCurrPos[ch][0] = -1; } /* Find first Transient Position */ if ((hMp4SpaceEnc->ppTrCurrPos[ch][0] >= 0) && ((currTransPos < 0) || (hMp4SpaceEnc->ppTrCurrPos[ch][0] < currTransPos))) { currTransPos = hMp4SpaceEnc->ppTrCurrPos[ch][0]; } } /* !LFE */ } /* ch */ if (hMp4SpaceEnc->useFrameKeep == 1) { if ((currTransPos != -1) || (hMp4SpaceEnc->independencyFlag == 1)) { hMp4SpaceEnc->avoid_keep = NUM_KEEP_WINDOWS; currTransPos = -1; } } /* Save previous Transient Position */ hMp4SpaceEnc->trPrevPos[0] = FDKmax(-1, hMp4SpaceEnc->trPrevPos[1] - (INT)nFrameTimeSlots); hMp4SpaceEnc->trPrevPos[1] = currTransPos; /* Update Onset Detection Energy Buffer */ for (ch = 0; ch < nChIn; ch++) { if (SACENC_OK != (error = fdk_sacenc_onsetDetect_Update( hMp4SpaceEnc->phOnset[ch], nFrameTimeSlots))) { goto bail; } } /* Framing */ if (SACENC_OK != (error = fdk_sacenc_frameWindow_GetWindow( hMp4SpaceEnc->hFrameWindow, hMp4SpaceEnc->trPrevPos, nFrameTimeSlots, &pFrameData->framingInfo, hMp4SpaceEnc->pFrameWindowAna__FDK, &hMp4SpaceEnc->frameWinList, hMp4SpaceEnc->avoid_keep))) { goto bail; } /* * MPS Processing: */ for (ps = 0, winCnt = 0; ps < hMp4SpaceEnc->frameWinList.n; ++ps) { /* Analysis Windowing */ if (hMp4SpaceEnc->frameWinList.dat[ps].hold == FW_HOLD) { /* ************************************** */ /* ONLY COPY AND HOLD PREVIOUS PARAMETERS */ if (SACENC_OK != (error = fdk_sacenc_duplicateParameterSet( &hMp4SpaceEnc->saveFrame, 0, pFrameData, ps))) { goto bail; } } else { /* !FW_HOLD */ /* ************************************** */ /* NEW WINDOW */ INT nAnalysisLengthTimeSlots, analysisStartTimeSlot; nAnalysisLengthTimeSlots = getAnalysisLengthTimeSlots( hMp4SpaceEnc->pFrameWindowAna__FDK[winCnt], hMp4SpaceEnc->nAnalysisLengthTimeSlots); analysisStartTimeSlot = getAnalysisStartTimeSlot(hMp4SpaceEnc->pFrameWindowAna__FDK[winCnt], hMp4SpaceEnc->nAnalysisLengthTimeSlots); /* perform main signal analysis windowing in * fdk_sacenc_spaceTree_Apply() */ FIXP_WIN *pFrameWindowAna__FDK = hMp4SpaceEnc->pFrameWindowAna__FDK[winCnt]; FIXP_DPK ***pppHybridIn__FDK = hMp4SpaceEnc->pppHybridIn__FDK; FIXP_DPK ***pppProcDataIn__FDK = hMp4SpaceEnc->pppProcDataIn__FDK; if (hMp4SpaceEnc->encMode == SACENC_212 && hMp4SpaceEnc->bEncMode_212_only) { pppProcDataIn__FDK = pppHybridIn__FDK; } if (SACENC_OK != (error = fdk_sacenc_spaceTree_Apply( hMp4SpaceEnc->hSpaceTree, ps, nChIn, nAnalysisLengthTimeSlots, analysisStartTimeSlot, hMp4SpaceEnc->nHybridBands, pFrameWindowAna__FDK, pppHybridIn__FDK, pppProcDataIn__FDK, /* multi-channel input */ pFrameData, hMp4SpaceEnc->avoid_keep, encoderInputChScale))) { goto bail; } /* Save spatial frame for potential hold parameter set */ if (SACENC_OK != (error = fdk_sacenc_duplicateParameterSet( pFrameData, ps, &hMp4SpaceEnc->saveFrame, 0))) { goto bail; } ++winCnt; } if (hMp4SpaceEnc->avoid_keep > 0) { hMp4SpaceEnc->avoid_keep--; } } /* Loop over Parameter Sets */ /* ---- End of Processing Loop ---- */ /* * Update hybridInReal/Imag buffer and do the same for arbDmx * this means to move the hybrid data of the current frame to the beginning * of the 2*nFrameLength-long buffer */ if (!(hMp4SpaceEnc->encMode == SACENC_212 && hMp4SpaceEnc->bEncMode_212_only)) { for (ch = 0; ch < nChIn; ch++) { /* for automatic downmix */ for (slot = 0; slot < (int)(hMp4SpaceEnc->nUpdateHybridPositionTimeSlots + nFrameTimeSlots - nFrameTimeSlotsReduction); slot++) { copyCplxVec( hMp4SpaceEnc->pppHybridIn__FDK[ch][slot], hMp4SpaceEnc->pppHybridIn__FDK[ch][nFrameTimeSlots + slot], hMp4SpaceEnc->nHybridBands); } for (slot = 0; slot < nFrameTimeSlots; slot++) { setCplxVec( hMp4SpaceEnc->pppHybridIn__FDK [ch][hMp4SpaceEnc->nUpdateHybridPositionTimeSlots + nFrameTimeSlots - nFrameTimeSlotsReduction + slot], (FIXP_DBL)0, hMp4SpaceEnc->nHybridBands); } } } /* * Spatial Tonality: */ { /* Smooth config off. */ FDKmemclear(&pFrameData->smgData, sizeof(pFrameData->smgData)); } /* * Create bitstream * - control independecy flag * - write spatial frame * - return bitstream */ UCHAR *pBitstreamDelayBuffer; if (hMp4SpaceEnc->encMode == SACENC_212) { /* no bitstream delay buffer for SACENC_212 mode, write bitstream directly * into the sacOutBuffer buffer which is provided by the core routine */ pBitstreamDelayBuffer = (UCHAR *)outBufDesc->ppBase[1]; } else { /* bitstream delay is handled in ppBitstreamDelayBuffer buffer */ pBitstreamDelayBuffer = hMp4SpaceEnc ->ppBitstreamDelayBuffer[hMp4SpaceEnc->nBitstreamBufferWrite]; } if (pBitstreamDelayBuffer == NULL) { error = SACENC_INVALID_HANDLE; goto bail; } pFrameData->bsIndependencyFlag = hMp4SpaceEnc->independencyFlag; if (SACENC_OK != (error = fdk_sacenc_writeSpatialFrame( pBitstreamDelayBuffer, MAX_MPEGS_BYTES, &hMp4SpaceEnc->pnOutputBits[hMp4SpaceEnc->nBitstreamBufferWrite], hMp4SpaceEnc->hBitstreamFormatter))) { goto bail; } /* return bitstream info */ if ((hMp4SpaceEnc->nDiscardOutFrames == 0) && (getBufDescIdx(outBufDesc, (FDK_BUF_TYPE_OUTPUT | FDK_BUF_TYPE_BS_DATA)) != -1)) { const INT idx = getBufDescIdx( outBufDesc, (FDK_BUF_TYPE_OUTPUT | FDK_BUF_TYPE_BS_DATA)); const INT outBits = hMp4SpaceEnc->pnOutputBits[hMp4SpaceEnc->nBitstreamBufferRead]; if (((outBits + 7) / 8) > (INT)(outBufDesc->pBufSize[idx] / outBufDesc->pEleSize[idx])) { outargs->nOutputBits = 0; error = SACENC_ENCODE_ERROR; goto bail; } /* return bitstream buffer, copy delayed bitstream for all configurations * except for the SACENC_212 mode */ if (hMp4SpaceEnc->encMode != SACENC_212) { FDKmemcpy( outBufDesc->ppBase[idx], hMp4SpaceEnc ->ppBitstreamDelayBuffer[hMp4SpaceEnc->nBitstreamBufferRead], (outBits + 7) / 8); } /* return number of valid bits */ outargs->nOutputBits = outBits; } else { /* No spatial data should be returned if the current frame is to be discarded. */ outargs->nOutputBits = 0; } /* update pointers */ hMp4SpaceEnc->nBitstreamBufferRead = (hMp4SpaceEnc->nBitstreamBufferRead + 1) % hMp4SpaceEnc->nBitstreamDelayBuffer; hMp4SpaceEnc->nBitstreamBufferWrite = (hMp4SpaceEnc->nBitstreamBufferWrite + 1) % hMp4SpaceEnc->nBitstreamDelayBuffer; /* Set Output Parameters */ nOutputSamples = (hMp4SpaceEnc->nDiscardOutFrames == 0) ? (nOutputSamplesMax) : 0; /* don't output samples in case frames to be discarded */ if (nOutputSamples > nOutputSamplesBufferSize) { error = SACENC_INVALID_CONFIG; goto bail; } outargs->nOutputSamples = nOutputSamples; { /* !bQmfOutput */ if (hMp4SpaceEnc->encMode != SACENC_212) { /* delay output samples and interleave them */ /* note: in case of arbitrary downmix this will always be processed, * because nOutputSamples != 0, even if bDMXAlign is switched on */ /* always run copy-func, so nOutputSamplesMax instead of nOutputSamples */ for (ch = 0; ch < nChOut; ch++) { FDKmemcpy_flex( &hMp4SpaceEnc->pOutputDelayBuffer__FDK [ch + (hMp4SpaceEnc->nOutputBufferDelay) * nChOut], nChOut, hMp4SpaceEnc->ppTimeSigOut__FDK[ch], 1, nOutputSamplesMax / nChOut); } /* write delayed data in output pcm stream */ /* always calculate, limiter must have a lookahead!!! */ FDKmemcpy(pOutputSamples, hMp4SpaceEnc->pOutputDelayBuffer__FDK, nOutputSamplesMax * sizeof(INT_PCM)); /* update delay buffer (move back end to the beginning of the buffer) */ FDKmemmove( hMp4SpaceEnc->pOutputDelayBuffer__FDK, &hMp4SpaceEnc->pOutputDelayBuffer__FDK[nOutputSamplesMax], nChOut * (hMp4SpaceEnc->nOutputBufferDelay) * sizeof(INT_PCM)); } if (hMp4SpaceEnc->useTimeDomDownmix <= 0) { if (SACENC_OK != (error = fdk_sacenc_staticPostGain_ApplyFDK( hMp4SpaceEnc->hStaticGain, pOutputSamples, nOutputSamplesMax, 0))) { goto bail; } } } /* !bQmfOutput */ if (hMp4SpaceEnc->nDiscardOutFrames > 0) { hMp4SpaceEnc->nDiscardOutFrames--; } /* Invalidate Input Buffer */ hMp4SpaceEnc->nSamplesValid = 0; } /* valid handle */ bail: return error; } FDK_SACENC_ERROR FDK_sacenc_close(HANDLE_MP4SPACE_ENCODER *phMp4SpaceEnc) { FDK_SACENC_ERROR error = SACENC_OK; if (NULL != phMp4SpaceEnc) { if (NULL != *phMp4SpaceEnc) { int ch, i; HANDLE_MP4SPACE_ENCODER const hEnc = *phMp4SpaceEnc; if (hEnc->pParameterBand2HybridBandOffset != NULL) { FDK_FREE_MEMORY_1D(hEnc->pParameterBand2HybridBandOffset); } /* Free Analysis Filterbank Structs */ if (hEnc->pEncoderInputChScale != NULL) { FDK_FREE_MEMORY_1D(hEnc->pEncoderInputChScale); } if (hEnc->staticTimeDomainDmxInScale != NULL) { FDK_FREE_MEMORY_1D(hEnc->staticTimeDomainDmxInScale); } if (hEnc->phQmfFiltIn__FDK != NULL) { for (ch = 0; ch < hEnc->setup.maxChIn; ch++) { if (hEnc->phQmfFiltIn__FDK[ch] != NULL) { if (hEnc->phQmfFiltIn__FDK[ch]->FilterStates != NULL) { FDK_FREE_MEMORY_1D(hEnc->phQmfFiltIn__FDK[ch]->FilterStates); } FDK_FREE_MEMORY_1D(hEnc->phQmfFiltIn__FDK[ch]); } } FDK_FREE_MEMORY_1D(hEnc->phQmfFiltIn__FDK); } for (ch = 0; ch < hEnc->setup.maxChIn; ch++) { if (NULL != hEnc->phDCFilterSigIn[ch]) { fdk_sacenc_destroyDCFilter(&hEnc->phDCFilterSigIn[ch]); } } /* Close Onset Detection */ for (ch = 0; ch < hEnc->setup.maxChIn; ch++) { if (NULL != hEnc->phOnset[ch]) { fdk_sacenc_onsetDetect_Close(&hEnc->phOnset[ch]); } } if (hEnc->ppTrCurrPos) { FDK_FREE_MEMORY_2D(hEnc->ppTrCurrPos); } if (hEnc->hFrameWindow) { fdk_sacenc_frameWindow_Destroy(&hEnc->hFrameWindow); } /* Close Space Tree */ if (NULL != hEnc->hSpaceTree) { fdk_sacenc_spaceTree_Close(&hEnc->hSpaceTree); } if (NULL != hEnc->hEnhancedTimeDmx) { fdk_sacenc_close_enhancedTimeDomainDmx(&hEnc->hEnhancedTimeDmx); } /* Close Static Gain */ if (NULL != hEnc->hStaticGain) { fdk_sacenc_staticGain_Close(&hEnc->hStaticGain); } if (NULL != hEnc->hStaticGainConfig) { fdk_sacenc_staticGain_CloseConfig(&hEnc->hStaticGainConfig); } /* Close Delay*/ if (NULL != hEnc->hDelay) { fdk_sacenc_delay_Close(&hEnc->hDelay); } /* Delete Bitstream Stuff */ if (NULL != hEnc->hBitstreamFormatter) { fdk_sacenc_destroySpatialBitstreamEncoder(&(hEnc->hBitstreamFormatter)); } if (hEnc->pppHybridIn__FDK != NULL) { if (hEnc->setup.bEncMode_212 == 1) { FDK_FREE_MEMORY_3D(hEnc->pppHybridIn__FDK); FDK_FREE_MEMORY_3D(hEnc->pppHybridInStatic__FDK); } else { FDK_FREE_MEMORY_3D(hEnc->pppHybridIn__FDK); } } if (hEnc->pppProcDataIn__FDK != NULL) { FDK_FREE_MEMORY_3D(hEnc->pppProcDataIn__FDK); } if (hEnc->pOutputDelayBuffer__FDK != NULL) { FDK_FREE_MEMORY_1D(hEnc->pOutputDelayBuffer__FDK); } if (hEnc->ppTimeSigIn__FDK != NULL) { { FDK_FREE_MEMORY_2D(hEnc->ppTimeSigIn__FDK); } } if (hEnc->ppTimeSigDelayIn__FDK != NULL) { FDK_FREE_MEMORY_2D(hEnc->ppTimeSigDelayIn__FDK); } if (hEnc->ppTimeSigOut__FDK != NULL) { FDK_FREE_MEMORY_2D(hEnc->ppTimeSigOut__FDK); } for (i = 0; i < MAX_NUM_PARAMS; i++) { if (hEnc->pFrameWindowAna__FDK[i] != NULL) { FDK_FREE_MEMORY_1D(hEnc->pFrameWindowAna__FDK[i]); } } if (hEnc->pnOutputBits != NULL) { FDK_FREE_MEMORY_1D(hEnc->pnOutputBits); } if (hEnc->ppBitstreamDelayBuffer != NULL) { FDK_FREE_MEMORY_2D(hEnc->ppBitstreamDelayBuffer); } if (hEnc->sscBuf.pSsc != NULL) { FDK_FREE_MEMORY_1D(hEnc->sscBuf.pSsc); } FDK_FREE_MEMORY_1D(*phMp4SpaceEnc); } } return error; } /*----------------------------------------------------------------------------- functionname: mp4SpaceEnc_InitDelayCompensation() description: initialzes delay compensation returns: noError on success, an apropriate error code else -----------------------------------------------------------------------------*/ static FDK_SACENC_ERROR mp4SpaceEnc_InitDelayCompensation( HANDLE_MP4SPACE_ENCODER hMp4SpaceEnc, const INT coreCoderDelay) { FDK_SACENC_ERROR error = SACENC_OK; /* Sanity Check */ if (hMp4SpaceEnc == NULL) { error = SACENC_INVALID_HANDLE; } else { hMp4SpaceEnc->coreCoderDelay = coreCoderDelay; if (SACENC_OK != (error = fdk_sacenc_delay_Init( hMp4SpaceEnc->hDelay, hMp4SpaceEnc->nQmfBands, hMp4SpaceEnc->nFrameLength, coreCoderDelay, hMp4SpaceEnc->timeAlignment))) { goto bail; } fdk_sacenc_delay_SetDmxAlign(hMp4SpaceEnc->hDelay, 0); fdk_sacenc_delay_SetTimeDomDmx( hMp4SpaceEnc->hDelay, (hMp4SpaceEnc->useTimeDomDownmix >= 1) ? 1 : 0); fdk_sacenc_delay_SetMinimizeDelay(hMp4SpaceEnc->hDelay, 1); if (SACENC_OK != (error = fdk_sacenc_delay_SubCalulateBufferDelays( hMp4SpaceEnc->hDelay))) { goto bail; } /* init output delay compensation */ hMp4SpaceEnc->nBitstreamDelayBuffer = fdk_sacenc_delay_GetBitstreamFrameBufferSize(hMp4SpaceEnc->hDelay); hMp4SpaceEnc->nOutputBufferDelay = fdk_sacenc_delay_GetOutputAudioBufferDelay(hMp4SpaceEnc->hDelay); hMp4SpaceEnc->nSurroundAnalysisBufferDelay = fdk_sacenc_delay_GetSurroundAnalysisBufferDelay(hMp4SpaceEnc->hDelay); hMp4SpaceEnc->nBitstreamBufferRead = 0; hMp4SpaceEnc->nBitstreamBufferWrite = hMp4SpaceEnc->nBitstreamDelayBuffer - 1; if (hMp4SpaceEnc->encMode == SACENC_212) { /* mode 212 expects no bitstream delay */ if (hMp4SpaceEnc->nBitstreamBufferWrite != hMp4SpaceEnc->nBitstreamBufferRead) { error = SACENC_PARAM_ERROR; goto bail; } /* mode 212 expects no output buffer delay */ if (hMp4SpaceEnc->nOutputBufferDelay != 0) { error = SACENC_PARAM_ERROR; goto bail; } } /*** Input delay to obtain a net encoder delay that is a multiple of the used framelength to ensure synchronization of framing in artistic down-mix with the corresponding spatial data. ***/ hMp4SpaceEnc->nDiscardOutFrames = fdk_sacenc_delay_GetDiscardOutFrames(hMp4SpaceEnc->hDelay); hMp4SpaceEnc->nInputDelay = fdk_sacenc_delay_GetDmxAlignBufferDelay(hMp4SpaceEnc->hDelay); /* reset independency Flag counter */ hMp4SpaceEnc->independencyCount = 0; hMp4SpaceEnc->independencyFlag = 1; int i; /* write some parameters to bitstream */ for (i = 0; i < hMp4SpaceEnc->nBitstreamDelayBuffer - 1; i++) { SPATIALFRAME *pFrameData = NULL; if (NULL == (pFrameData = fdk_sacenc_getSpatialFrame( hMp4SpaceEnc->hBitstreamFormatter, READ_SPATIALFRAME))) { error = SACENC_INVALID_HANDLE; goto bail; } pFrameData->bsIndependencyFlag = 1; pFrameData->framingInfo.numParamSets = 1; pFrameData->framingInfo.bsFramingType = 0; fdk_sacenc_writeSpatialFrame( hMp4SpaceEnc->ppBitstreamDelayBuffer[i], MAX_MPEGS_BYTES, &hMp4SpaceEnc->pnOutputBits[i], hMp4SpaceEnc->hBitstreamFormatter); } if ((hMp4SpaceEnc->nInputDelay > MAX_DELAY_INPUT) || (hMp4SpaceEnc->nOutputBufferDelay > MAX_DELAY_OUTPUT) || (hMp4SpaceEnc->nSurroundAnalysisBufferDelay > MAX_DELAY_SURROUND_ANALYSIS) || (hMp4SpaceEnc->nBitstreamDelayBuffer > MAX_BITSTREAM_DELAY)) { error = SACENC_INIT_ERROR; goto bail; } } bail: return error; } static QUANTMODE __mapQuantMode(const MP4SPACEENC_QUANTMODE quantMode) { QUANTMODE bsQuantMode = QUANTMODE_INVALID; switch (quantMode) { case SACENC_QUANTMODE_FINE: bsQuantMode = QUANTMODE_FINE; break; case SACENC_QUANTMODE_EBQ1: bsQuantMode = QUANTMODE_EBQ1; break; case SACENC_QUANTMODE_EBQ2: bsQuantMode = QUANTMODE_EBQ2; break; case SACENC_QUANTMODE_RSVD3: case SACENC_QUANTMODE_INVALID: default: bsQuantMode = QUANTMODE_INVALID; } /* switch hEnc->quantMode */ return bsQuantMode; } static FDK_SACENC_ERROR FillSpatialSpecificConfig( const HANDLE_MP4SPACE_ENCODER hEnc, SPATIALSPECIFICCONFIG *const hSsc) { FDK_SACENC_ERROR error = SACENC_OK; if ((NULL == hEnc) || (NULL == hSsc)) { error = SACENC_INVALID_HANDLE; } else { SPACE_TREE_DESCRIPTION spaceTreeDescription; int i; /* Get tree description */ if (SACENC_OK != (error = fdk_sacenc_spaceTree_GetDescription( hEnc->hSpaceTree, &spaceTreeDescription))) { goto bail; } /* Fill SSC */ FDKmemclear(hSsc, sizeof(SPATIALSPECIFICCONFIG)); /* reset */ hSsc->numBands = hEnc->spaceTreeSetup.nParamBands; /* for bsFreqRes */ /* Fill tree configuration */ hSsc->treeDescription.numOttBoxes = spaceTreeDescription.nOttBoxes; hSsc->treeDescription.numInChan = spaceTreeDescription.nInChannels; hSsc->treeDescription.numOutChan = spaceTreeDescription.nOutChannels; for (i = 0; i < SACENC_MAX_NUM_BOXES; i++) { hSsc->ottConfig[i].bsOttBands = hSsc->numBands; } switch (hEnc->encMode) { case SACENC_212: hSsc->bsTreeConfig = TREE_212; break; case SACENC_INVALID_MODE: default: error = SACENC_INVALID_CONFIG; goto bail; } hSsc->bsSamplingFrequency = hEnc->nSampleRate; /* for bsSamplingFrequencyIndex */ hSsc->bsFrameLength = hEnc->nFrameTimeSlots - 1; /* map decorr type */ if (DECORR_INVALID == (hSsc->bsDecorrConfig = mp4SpaceEnc_GetDecorrConfig(hEnc->encMode))) { error = SACENC_INVALID_CONFIG; goto bail; } /* map quantMode */ if (QUANTMODE_INVALID == (hSsc->bsQuantMode = __mapQuantMode(hEnc->quantMode))) { error = SACENC_INVALID_CONFIG; goto bail; } /* Configure Gains*/ hSsc->bsFixedGainDMX = fdk_sacenc_staticGain_GetDmxGain(hEnc->hStaticGain); hSsc->bsEnvQuantMode = 0; } /* valid handle */ bail: return error; } static FDK_SACENC_ERROR mp4SpaceEnc_FillSpaceTreeSetup( const HANDLE_MP4SPACE_ENCODER hEnc, SPACE_TREE_SETUP *const hSpaceTreeSetup) { FDK_SACENC_ERROR error = SACENC_OK; /* Sanity Check */ if (NULL == hEnc || NULL == hSpaceTreeSetup) { error = SACENC_INVALID_HANDLE; } else { QUANTMODE tmpQuantmode = QUANTMODE_INVALID; /* map quantMode */ if (QUANTMODE_INVALID == (tmpQuantmode = __mapQuantMode(hEnc->quantMode))) { error = SACENC_INVALID_CONFIG; goto bail; } hSpaceTreeSetup->nParamBands = hEnc->nParamBands; hSpaceTreeSetup->bUseCoarseQuantTtoCld = hEnc->useCoarseQuantCld; hSpaceTreeSetup->bUseCoarseQuantTtoIcc = hEnc->useCoarseQuantIcc; hSpaceTreeSetup->quantMode = tmpQuantmode; hSpaceTreeSetup->nHybridBandsMax = hEnc->nHybridBands; switch (hEnc->encMode) { case SACENC_212: hSpaceTreeSetup->mode = SPACETREE_212; hSpaceTreeSetup->nChannelsInMax = 2; break; case SACENC_INVALID_MODE: default: error = SACENC_INVALID_CONFIG; goto bail; } /* switch hEnc->encMode */ } /* valid handle */ bail: return error; } FDK_SACENC_ERROR FDK_sacenc_getInfo(const HANDLE_MP4SPACE_ENCODER hMp4SpaceEnc, MP4SPACEENC_INFO *const pInfo) { FDK_SACENC_ERROR error = SACENC_OK; if ((NULL == hMp4SpaceEnc) || (NULL == pInfo)) { error = SACENC_INVALID_HANDLE; } else { pInfo->nSampleRate = hMp4SpaceEnc->nSampleRate; pInfo->nSamplesFrame = hMp4SpaceEnc->nFrameLength; pInfo->nTotalInputChannels = hMp4SpaceEnc->nInputChannels; pInfo->nDmxDelay = fdk_sacenc_delay_GetInfoDmxDelay(hMp4SpaceEnc->hDelay); pInfo->nCodecDelay = fdk_sacenc_delay_GetInfoCodecDelay(hMp4SpaceEnc->hDelay); pInfo->nDecoderDelay = fdk_sacenc_delay_GetInfoDecoderDelay(hMp4SpaceEnc->hDelay); pInfo->nPayloadDelay = fdk_sacenc_delay_GetBitstreamFrameBufferSize(hMp4SpaceEnc->hDelay) - 1; pInfo->nDiscardOutFrames = hMp4SpaceEnc->nDiscardOutFrames; pInfo->pSscBuf = &hMp4SpaceEnc->sscBuf; } return error; } FDK_SACENC_ERROR FDK_sacenc_setParam(HANDLE_MP4SPACE_ENCODER hMp4SpaceEnc, const SPACEENC_PARAM param, const UINT value) { FDK_SACENC_ERROR error = SACENC_OK; /* check encoder handle */ if (hMp4SpaceEnc == NULL) { error = SACENC_INVALID_HANDLE; goto bail; } /* apply param value */ switch (param) { case SACENC_LOWDELAY: if (!((value == 0) || (value == 1) || (value == 2))) { error = SACENC_INVALID_CONFIG; break; } hMp4SpaceEnc->user.bLdMode = value; break; case SACENC_ENC_MODE: switch ((MP4SPACEENC_MODE)value) { case SACENC_212: hMp4SpaceEnc->user.encMode = (MP4SPACEENC_MODE)value; break; default: error = SACENC_INVALID_CONFIG; } break; case SACENC_SAMPLERATE: if (((int)value < 0) || ((int)value > hMp4SpaceEnc->setup.maxSamplingrate)) { error = SACENC_INVALID_CONFIG; break; } hMp4SpaceEnc->user.sampleRate = value; break; case SACENC_FRAME_TIME_SLOTS: if (((int)value < 0) || ((int)value > hMp4SpaceEnc->setup.maxFrameTimeSlots)) { error = SACENC_INVALID_CONFIG; break; } hMp4SpaceEnc->user.frameTimeSlots = value; break; case SACENC_PARAM_BANDS: switch ((MP4SPACEENC_BANDS_CONFIG)value) { case SACENC_BANDS_4: case SACENC_BANDS_5: case SACENC_BANDS_7: case SACENC_BANDS_9: case SACENC_BANDS_12: case SACENC_BANDS_15: case SACENC_BANDS_23: hMp4SpaceEnc->user.nParamBands = (MP4SPACEENC_BANDS_CONFIG)value; break; default: error = SACENC_INVALID_CONFIG; } break; case SACENC_TIME_DOM_DMX: if (!((value == 0) || (value == 2))) { error = SACENC_INVALID_CONFIG; break; } hMp4SpaceEnc->user.bTimeDomainDmx = value; break; case SACENC_DMX_GAIN: if (!((value == 0) || (value == 1) || (value == 2) || (value == 3) || (value == 4) || (value == 5) || (value == 6) || (value == 7))) { error = SACENC_INVALID_CONFIG; break; } error = fdk_sacenc_staticGain_SetDmxGain(hMp4SpaceEnc->hStaticGainConfig, (MP4SPACEENC_DMX_GAIN)value); break; case SACENC_COARSE_QUANT: if (!((value == 0) || (value == 1))) { error = SACENC_INVALID_CONFIG; break; } hMp4SpaceEnc->user.bUseCoarseQuant = value; break; case SACENC_QUANT_MODE: switch ((MP4SPACEENC_QUANTMODE)value) { case SACENC_QUANTMODE_FINE: case SACENC_QUANTMODE_EBQ1: case SACENC_QUANTMODE_EBQ2: hMp4SpaceEnc->user.quantMode = (MP4SPACEENC_QUANTMODE)value; break; default: error = SACENC_INVALID_CONFIG; } break; case SACENC_TIME_ALIGNMENT: if ((INT)value < -32768 || (INT)value > 32767) { error = SACENC_INVALID_CONFIG; break; } hMp4SpaceEnc->user.timeAlignment = value; break; case SACENC_INDEPENDENCY_COUNT: hMp4SpaceEnc->independencyCount = value; break; case SACENC_INDEPENDENCY_FACTOR: hMp4SpaceEnc->user.independencyFactor = value; break; default: error = SACENC_UNSUPPORTED_PARAMETER; break; } /* switch(param) */ bail: return error; } FDK_SACENC_ERROR FDK_sacenc_getLibInfo(LIB_INFO *info) { int i = 0; if (info == NULL) { return SACENC_INVALID_HANDLE; } FDK_toolsGetLibInfo(info); /* search for next free tab */ for (i = 0; i < FDK_MODULE_LAST; i++) { if (info[i].module_id == FDK_NONE) break; } if (i == FDK_MODULE_LAST) { return SACENC_INIT_ERROR; } info[i].module_id = FDK_MPSENC; info[i].build_date = SACENC_LIB_BUILD_DATE; info[i].build_time = SACENC_LIB_BUILD_TIME; info[i].title = SACENC_LIB_TITLE; info[i].version = LIB_VERSION(SACENC_LIB_VL0, SACENC_LIB_VL1, SACENC_LIB_VL2); LIB_VERSION_STRING(&info[i]); /* Capability flags */ info[i].flags = 0; /* End of flags */ return SACENC_OK; } static DECORRCONFIG mp4SpaceEnc_GetDecorrConfig( const MP4SPACEENC_MODE encMode) { DECORRCONFIG decorrConfig = DECORR_INVALID; /* set decorrConfig dependent on tree mode */ switch (encMode) { case SACENC_212: decorrConfig = DECORR_QMFSPLIT0; break; case SACENC_INVALID_MODE: default: decorrConfig = DECORR_INVALID; } return decorrConfig; } static FDK_SACENC_ERROR mp4SpaceEnc_InitNumParamBands( HANDLE_MP4SPACE_ENCODER hEnc, const MP4SPACEENC_BANDS_CONFIG nParamBands) { FDK_SACENC_ERROR error = SACENC_OK; /* Set/Check nParamBands */ int k = 0; const int n = sizeof(pValidBands_Ld) / sizeof(UCHAR); const UCHAR *pBands = pValidBands_Ld; while (k < n && pBands[k] != (UCHAR)nParamBands) ++k; if (k == n) { hEnc->nParamBands = SACENC_BANDS_INVALID; } else { hEnc->nParamBands = nParamBands; } return error; } fdk-aac-2.0.2/libSACenc/src/sacenc_nlc_enc.cpp000066400000000000000000001224101404225310700207720ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround encoder library ************************* Author(s): Karsten Linzmeier Description: Noiseless Coding Huffman encoder *******************************************************************************/ /* Includes ******************************************************************/ #include "sacenc_nlc_enc.h" #include "genericStds.h" #include "fixpoint_math.h" #include "sacenc_const.h" #include "sacenc_huff_tab.h" #include "sacenc_paramextract.h" /* Defines *******************************************************************/ #define PAIR_SHIFT 4 #define PAIR_MASK 0xf #define PBC_MIN_BANDS 5 typedef enum { BACKWARDS = 0x0, FORWARDS = 0x1 } DIRECTION; typedef enum { DIFF_FREQ = 0x0, DIFF_TIME = 0x1 } DIFF_TYPE; typedef enum { HUFF_1D = 0x0, HUFF_2D = 0x1 } CODING_SCHEME; typedef enum { FREQ_PAIR = 0x0, TIME_PAIR = 0x1 } PAIRING; /* Data Types ****************************************************************/ /* Constants *****************************************************************/ static const UCHAR lavHuffVal[4] = {0, 2, 6, 7}; static const UCHAR lavHuffLen[4] = {1, 2, 3, 3}; static const UCHAR lav_step_CLD[] = {0, 0, 0, 0, 1, 1, 2, 2, 3, 3}; static const UCHAR lav_step_ICC[] = {0, 0, 1, 1, 2, 2, 3, 3}; /* Function / Class Declarations *********************************************/ /* Function / Class Definition ***********************************************/ static void split_lsb(const SHORT *const in_data, SHORT offset, const INT num_val, SHORT *const out_data_lsb, SHORT *const out_data_msb) { int i; for (i = 0; i < num_val; i++) { SHORT val = in_data[i] + offset; if (out_data_lsb != NULL) out_data_lsb[i] = val & 0x0001; if (out_data_msb != NULL) out_data_msb[i] = val >> 1; } } static void apply_lsb_coding(HANDLE_FDK_BITSTREAM strm, const SHORT *const in_data_lsb, const UINT num_lsb, const INT num_val) { int i; for (i = 0; i < num_val; i++) { FDKwriteBits(strm, in_data_lsb[i], num_lsb); } } static void calc_diff_freq(const SHORT *const in_data, SHORT *const out_data, const INT num_val) { int i; out_data[0] = in_data[0]; for (i = 1; i < num_val; i++) { out_data[i] = in_data[i] - in_data[i - 1]; } } static void calc_diff_time(const SHORT *const in_data, const SHORT *const prev_data, SHORT *const out_data, const INT num_val) { int i; out_data[0] = in_data[0]; out_data[1] = prev_data[0]; for (i = 0; i < num_val; i++) { out_data[i + 2] = in_data[i] - prev_data[i]; } } static INT sym_check(SHORT data[2], const INT lav, SHORT *const pSym_bits) { UCHAR symBits = 0; int sum_val = data[0] + data[1]; int diff_val = data[0] - data[1]; int num_sbits = 0; if (sum_val != 0) { int sum_neg = (sum_val < 0) ? 1 : 0; if (sum_neg) { sum_val = -sum_val; diff_val = -diff_val; } symBits = (symBits << 1) | sum_neg; num_sbits++; } if (diff_val != 0) { int diff_neg = (diff_val < 0) ? 1 : 0; if (diff_neg) { diff_val = -diff_val; } symBits = (symBits << 1) | diff_neg; num_sbits++; } if (pSym_bits != NULL) { *pSym_bits = symBits; } if (sum_val % 2) { data[0] = lav - sum_val / 2; data[1] = lav - diff_val / 2; } else { data[0] = sum_val / 2; data[1] = diff_val / 2; } return num_sbits; } static INT ilog2(UINT i) { int l = 0; if (i) i--; while (i > 0) { i >>= 1; l++; } return l; } static SHORT calc_pcm_bits(const SHORT num_val, const SHORT num_levels) { SHORT num_complete_chunks = 0, rest_chunk_size = 0; SHORT max_grp_len = 0, bits_pcm = 0; int chunk_levels, i; switch (num_levels) { case 3: max_grp_len = 5; break; case 6: max_grp_len = 5; break; case 7: max_grp_len = 6; break; case 11: max_grp_len = 2; break; case 13: max_grp_len = 4; break; case 19: max_grp_len = 4; break; case 25: max_grp_len = 3; break; case 51: max_grp_len = 4; break; default: max_grp_len = 1; } num_complete_chunks = num_val / max_grp_len; rest_chunk_size = num_val % max_grp_len; chunk_levels = 1; for (i = 1; i <= max_grp_len; i++) { chunk_levels *= num_levels; } bits_pcm = (SHORT)(ilog2(chunk_levels) * num_complete_chunks); bits_pcm += (SHORT)(ilog2(num_levels) * rest_chunk_size); return bits_pcm; } static void apply_pcm_coding(HANDLE_FDK_BITSTREAM strm, const SHORT *const in_data_1, const SHORT *const in_data_2, const SHORT offset, const SHORT num_val, const SHORT num_levels) { SHORT i = 0, j = 0, idx = 0; SHORT max_grp_len = 0, grp_len = 0, next_val = 0; int grp_val = 0, chunk_levels = 0; SHORT pcm_chunk_size[7] = {0}; switch (num_levels) { case 3: max_grp_len = 5; break; case 5: max_grp_len = 3; break; case 6: max_grp_len = 5; break; case 7: max_grp_len = 6; break; case 9: max_grp_len = 5; break; case 11: max_grp_len = 2; break; case 13: max_grp_len = 4; break; case 19: max_grp_len = 4; break; case 25: max_grp_len = 3; break; case 51: max_grp_len = 4; break; default: max_grp_len = 1; } chunk_levels = 1; for (i = 1; i <= max_grp_len; i++) { chunk_levels *= num_levels; pcm_chunk_size[i] = ilog2(chunk_levels); } for (i = 0; i < num_val; i += max_grp_len) { grp_len = FDKmin(max_grp_len, num_val - i); grp_val = 0; for (j = 0; j < grp_len; j++) { idx = i + j; if (in_data_2 == NULL) { next_val = in_data_1[idx]; } else if (in_data_1 == NULL) { next_val = in_data_2[idx]; } else { next_val = ((idx % 2) ? in_data_2[idx / 2] : in_data_1[idx / 2]); } next_val += offset; grp_val = grp_val * num_levels + next_val; } FDKwriteBits(strm, grp_val, pcm_chunk_size[grp_len]); } } static UINT huff_enc_1D(HANDLE_FDK_BITSTREAM strm, const DATA_TYPE data_type, const INT dim1, SHORT *const in_data, const SHORT num_val, const SHORT p0_flag) { int i, offset = 0; UINT huffBits = 0; HUFF_ENTRY part0 = {0}; const HUFF_ENTRY *pHuffTab = NULL; switch (data_type) { case t_CLD: pHuffTab = fdk_sacenc_huffCLDTab.h1D[dim1]; break; case t_ICC: pHuffTab = fdk_sacenc_huffICCTab.h1D[dim1]; break; } if (p0_flag) { switch (data_type) { case t_CLD: part0 = fdk_sacenc_huffPart0Tab.cld[in_data[0]]; break; case t_ICC: part0 = fdk_sacenc_huffPart0Tab.icc[in_data[0]]; break; } huffBits += FDKwriteBits(strm, HUFF_VALUE(part0), HUFF_LENGTH(part0)); offset = 1; } for (i = offset; i < num_val; i++) { int id_sign = 0; int id = in_data[i]; if (id != 0) { id_sign = 0; if (id < 0) { id = -id; id_sign = 1; } } huffBits += FDKwriteBits(strm, HUFF_VALUE(pHuffTab[id]), HUFF_LENGTH(pHuffTab[id])); if (id != 0) { huffBits += FDKwriteBits(strm, id_sign, 1); } } /* for i */ return huffBits; } static void getHuffEntry(const INT lav, const DATA_TYPE data_type, const INT i, const SHORT tab_idx_2D[2], const SHORT in_data[][2], HUFF_ENTRY *const pEntry, HUFF_ENTRY *const pEscape) { const HUFF_CLD_TAB_2D *pCLD2dTab = &fdk_sacenc_huffCLDTab.h2D[tab_idx_2D[0]][tab_idx_2D[1]]; const HUFF_ICC_TAB_2D *pICC2dTab = &fdk_sacenc_huffICCTab.h2D[tab_idx_2D[0]][tab_idx_2D[1]]; switch (lav) { case 1: { const LAV1_2D *pLav1 = NULL; switch (data_type) { case t_CLD: pLav1 = NULL; break; case t_ICC: pLav1 = &pICC2dTab->lav1; break; } if (pLav1 != NULL) { *pEntry = pLav1->entry[in_data[i][0]][in_data[i][1]]; *pEscape = pLav1->escape; } } break; case 3: { const LAV3_2D *pLav3 = NULL; switch (data_type) { case t_CLD: pLav3 = &pCLD2dTab->lav3; break; case t_ICC: pLav3 = &pICC2dTab->lav3; break; } if (pLav3 != NULL) { *pEntry = pLav3->entry[in_data[i][0]][in_data[i][1]]; *pEscape = pLav3->escape; } } break; case 5: { const LAV5_2D *pLav5 = NULL; switch (data_type) { case t_CLD: pLav5 = &pCLD2dTab->lav5; break; case t_ICC: pLav5 = &pICC2dTab->lav5; break; } if (pLav5 != NULL) { *pEntry = pLav5->entry[in_data[i][0]][in_data[i][1]]; *pEscape = pLav5->escape; } } break; case 7: { const LAV7_2D *pLav7 = NULL; switch (data_type) { case t_CLD: pLav7 = &pCLD2dTab->lav7; break; case t_ICC: pLav7 = &pICC2dTab->lav7; break; } if (pLav7 != NULL) { *pEntry = pLav7->entry[in_data[i][0]][in_data[i][1]]; *pEscape = pLav7->escape; } } break; case 9: { const LAV9_2D *pLav9 = NULL; switch (data_type) { case t_CLD: pLav9 = &pCLD2dTab->lav9; break; case t_ICC: pLav9 = NULL; break; } if (pLav9 != NULL) { *pEntry = pLav9->entry[in_data[i][0]][in_data[i][1]]; *pEscape = pLav9->escape; } } break; } } static UINT huff_enc_2D(HANDLE_FDK_BITSTREAM strm, const DATA_TYPE data_type, SHORT tab_idx_2D[2], SHORT lav_idx, SHORT in_data[][2], SHORT num_val, SHORT stride, SHORT *p0_data[2]) { SHORT i = 0, lav = 0, num_sbits = 0, sym_bits = 0, escIdx = 0; SHORT esc_data[2][MAXBANDS] = {{0}}; UINT huffBits = 0; const HUFF_ENTRY *pHuffEntry = NULL; switch (data_type) { case t_CLD: lav = 2 * lav_idx + 3; /* LAV */ pHuffEntry = fdk_sacenc_huffPart0Tab.cld; break; case t_ICC: lav = 2 * lav_idx + 1; /* LAV */ pHuffEntry = fdk_sacenc_huffPart0Tab.icc; break; } /* Partition 0 */ if (p0_data[0] != NULL) { HUFF_ENTRY entry = pHuffEntry[*p0_data[0]]; huffBits += FDKwriteBits(strm, HUFF_VALUE(entry), HUFF_LENGTH(entry)); } if (p0_data[1] != NULL) { HUFF_ENTRY entry = pHuffEntry[*p0_data[1]]; huffBits += FDKwriteBits(strm, HUFF_VALUE(entry), HUFF_LENGTH(entry)); } for (i = 0; i < num_val; i += stride) { HUFF_ENTRY entry = {0}; HUFF_ENTRY escape = {0}; esc_data[0][escIdx] = in_data[i][0] + lav; esc_data[1][escIdx] = in_data[i][1] + lav; num_sbits = sym_check(in_data[i], lav, &sym_bits); getHuffEntry(lav, data_type, i, tab_idx_2D, in_data, &entry, &escape); huffBits += FDKwriteBits(strm, HUFF_VALUE(entry), HUFF_LENGTH(entry)); if ((HUFF_VALUE(entry) == HUFF_VALUE(escape)) && (HUFF_LENGTH(entry) == HUFF_LENGTH(escape))) { escIdx++; } else { huffBits += FDKwriteBits(strm, sym_bits, num_sbits); } } /* for i */ if (escIdx > 0) { huffBits += calc_pcm_bits(2 * escIdx, (2 * lav + 1)); if (strm != NULL) { apply_pcm_coding(strm, esc_data[0], esc_data[1], 0 /*offset*/, 2 * escIdx, (2 * lav + 1)); } } return huffBits; } static SCHAR get_next_lav_step(const INT lav, const DATA_TYPE data_type) { SCHAR lav_step = 0; switch (data_type) { case t_CLD: lav_step = (lav > 9) ? -1 : lav_step_CLD[lav]; break; case t_ICC: lav_step = (lav > 7) ? -1 : lav_step_ICC[lav]; break; } return lav_step; } static INT diff_type_offset(const DIFF_TYPE diff_type) { int offset = 0; switch (diff_type) { case DIFF_FREQ: offset = 0; break; case DIFF_TIME: offset = 2; break; } return offset; } static SHORT calc_huff_bits(SHORT *in_data_1, SHORT *in_data_2, const DATA_TYPE data_type, const DIFF_TYPE diff_type_1, const DIFF_TYPE diff_type_2, const SHORT num_val, SHORT *const lav_idx, SHORT *const cdg_scheme) { SHORT tab_idx_2D[2][2] = {{0}}; SHORT tab_idx_1D[2] = {0}; SHORT df_rest_flag[2] = {0}; SHORT p0_flag[2] = {0}; SHORT pair_vec[MAXBANDS][2] = {{0}}; SHORT *p0_data_1[2] = {NULL}; SHORT *p0_data_2[2] = {NULL}; SHORT i = 0; SHORT lav_fp[2] = {0}; SHORT bit_count_1D = 0; SHORT bit_count_2D_freq = 0; SHORT bit_count_min = 0; SHORT num_val_1_short = 0; SHORT num_val_2_short = 0; SHORT *in_data_1_short = NULL; SHORT *in_data_2_short = NULL; /* 1D Huffman coding */ bit_count_1D = 1; /* HUFF_1D */ num_val_1_short = num_val; num_val_2_short = num_val; if (in_data_1 != NULL) { in_data_1_short = in_data_1 + diff_type_offset(diff_type_1); } if (in_data_2 != NULL) { in_data_2_short = in_data_2 + diff_type_offset(diff_type_2); } p0_flag[0] = (diff_type_1 == DIFF_FREQ); p0_flag[1] = (diff_type_2 == DIFF_FREQ); tab_idx_1D[0] = (diff_type_1 == DIFF_FREQ) ? 0 : 1; tab_idx_1D[1] = (diff_type_2 == DIFF_FREQ) ? 0 : 1; if (in_data_1 != NULL) { bit_count_1D += huff_enc_1D(NULL, data_type, tab_idx_1D[0], in_data_1_short, num_val_1_short, p0_flag[0]); } if (in_data_2 != NULL) { bit_count_1D += huff_enc_1D(NULL, data_type, tab_idx_1D[1], in_data_2_short, num_val_2_short, p0_flag[1]); } bit_count_min = bit_count_1D; *cdg_scheme = HUFF_1D << PAIR_SHIFT; lav_idx[0] = lav_idx[1] = -1; /* Huffman 2D frequency pairs */ bit_count_2D_freq = 1; /* HUFF_2D */ num_val_1_short = num_val; num_val_2_short = num_val; if (in_data_1 != NULL) { in_data_1_short = in_data_1 + diff_type_offset(diff_type_1); } if (in_data_2 != NULL) { in_data_2_short = in_data_2 + diff_type_offset(diff_type_2); } lav_fp[0] = lav_fp[1] = 0; p0_data_1[0] = NULL; p0_data_1[1] = NULL; p0_data_2[0] = NULL; p0_data_2[1] = NULL; if (in_data_1 != NULL) { if (diff_type_1 == DIFF_FREQ) { p0_data_1[0] = &in_data_1[0]; p0_data_1[1] = NULL; num_val_1_short -= 1; in_data_1_short += 1; } df_rest_flag[0] = num_val_1_short % 2; if (df_rest_flag[0]) num_val_1_short -= 1; for (i = 0; i < num_val_1_short - 1; i += 2) { pair_vec[i][0] = in_data_1_short[i]; pair_vec[i][1] = in_data_1_short[i + 1]; lav_fp[0] = FDKmax(lav_fp[0], fAbs(pair_vec[i][0])); lav_fp[0] = FDKmax(lav_fp[0], fAbs(pair_vec[i][1])); } tab_idx_2D[0][0] = (diff_type_1 == DIFF_TIME) ? 1 : 0; tab_idx_2D[0][1] = 0; tab_idx_1D[0] = (diff_type_1 == DIFF_FREQ) ? 0 : 1; lav_fp[0] = get_next_lav_step(lav_fp[0], data_type); if (lav_fp[0] != -1) bit_count_2D_freq += lavHuffLen[lav_fp[0]]; } if (in_data_2 != NULL) { if (diff_type_2 == DIFF_FREQ) { p0_data_2[0] = NULL; p0_data_2[1] = &in_data_2[0]; num_val_2_short -= 1; in_data_2_short += 1; } df_rest_flag[1] = num_val_2_short % 2; if (df_rest_flag[1]) num_val_2_short -= 1; for (i = 0; i < num_val_2_short - 1; i += 2) { pair_vec[i + 1][0] = in_data_2_short[i]; pair_vec[i + 1][1] = in_data_2_short[i + 1]; lav_fp[1] = FDKmax(lav_fp[1], fAbs(pair_vec[i + 1][0])); lav_fp[1] = FDKmax(lav_fp[1], fAbs(pair_vec[i + 1][1])); } tab_idx_2D[1][0] = (diff_type_2 == DIFF_TIME) ? 1 : 0; tab_idx_2D[1][1] = 0; tab_idx_1D[1] = (diff_type_2 == DIFF_FREQ) ? 0 : 1; lav_fp[1] = get_next_lav_step(lav_fp[1], data_type); if (lav_fp[1] != -1) bit_count_2D_freq += lavHuffLen[lav_fp[1]]; } if ((lav_fp[0] != -1) && (lav_fp[1] != -1)) { if (in_data_1 != NULL) { bit_count_2D_freq += huff_enc_2D(NULL, data_type, tab_idx_2D[0], lav_fp[0], pair_vec, num_val_1_short, 2, p0_data_1); } if (in_data_2 != NULL) { bit_count_2D_freq += huff_enc_2D(NULL, data_type, tab_idx_2D[1], lav_fp[1], pair_vec + 1, num_val_2_short, 2, p0_data_2); } if (in_data_1 != NULL) { if (df_rest_flag[0]) bit_count_2D_freq += huff_enc_1D(NULL, data_type, tab_idx_1D[0], in_data_1_short + num_val_1_short, 1, 0); } if (in_data_2 != NULL) { if (df_rest_flag[1]) bit_count_2D_freq += huff_enc_1D(NULL, data_type, tab_idx_1D[1], in_data_2_short + num_val_2_short, 1, 0); } if (bit_count_2D_freq < bit_count_min) { bit_count_min = bit_count_2D_freq; *cdg_scheme = HUFF_2D << PAIR_SHIFT | FREQ_PAIR; lav_idx[0] = lav_fp[0]; lav_idx[1] = lav_fp[1]; } } return bit_count_min; } static void apply_huff_coding(HANDLE_FDK_BITSTREAM strm, SHORT *const in_data_1, SHORT *const in_data_2, const DATA_TYPE data_type, const DIFF_TYPE diff_type_1, const DIFF_TYPE diff_type_2, const SHORT num_val, const SHORT *const lav_idx, const SHORT cdg_scheme) { SHORT tab_idx_2D[2][2] = {{0}}; SHORT tab_idx_1D[2] = {0}; SHORT df_rest_flag[2] = {0}; SHORT p0_flag[2] = {0}; SHORT pair_vec[MAXBANDS][2] = {{0}}; SHORT *p0_data_1[2] = {NULL}; SHORT *p0_data_2[2] = {NULL}; SHORT i = 0; SHORT num_val_1_short = num_val; SHORT num_val_2_short = num_val; SHORT *in_data_1_short = NULL; SHORT *in_data_2_short = NULL; /* Offset */ if (in_data_1 != NULL) { in_data_1_short = in_data_1 + diff_type_offset(diff_type_1); } if (in_data_2 != NULL) { in_data_2_short = in_data_2 + diff_type_offset(diff_type_2); } /* Signalize coding scheme */ FDKwriteBits(strm, cdg_scheme >> PAIR_SHIFT, 1); switch (cdg_scheme >> PAIR_SHIFT) { case HUFF_1D: p0_flag[0] = (diff_type_1 == DIFF_FREQ); p0_flag[1] = (diff_type_2 == DIFF_FREQ); tab_idx_1D[0] = (diff_type_1 == DIFF_FREQ) ? 0 : 1; tab_idx_1D[1] = (diff_type_2 == DIFF_FREQ) ? 0 : 1; if (in_data_1 != NULL) { huff_enc_1D(strm, data_type, tab_idx_1D[0], in_data_1_short, num_val_1_short, p0_flag[0]); } if (in_data_2 != NULL) { huff_enc_1D(strm, data_type, tab_idx_1D[1], in_data_2_short, num_val_2_short, p0_flag[1]); } break; /* HUFF_1D */ case HUFF_2D: switch (cdg_scheme & PAIR_MASK) { case FREQ_PAIR: if (in_data_1 != NULL) { if (diff_type_1 == DIFF_FREQ) { p0_data_1[0] = &in_data_1[0]; p0_data_1[1] = NULL; num_val_1_short -= 1; in_data_1_short += 1; } df_rest_flag[0] = num_val_1_short % 2; if (df_rest_flag[0]) num_val_1_short -= 1; for (i = 0; i < num_val_1_short - 1; i += 2) { pair_vec[i][0] = in_data_1_short[i]; pair_vec[i][1] = in_data_1_short[i + 1]; } tab_idx_2D[0][0] = (diff_type_1 == DIFF_TIME) ? 1 : 0; tab_idx_2D[0][1] = 0; tab_idx_1D[0] = (diff_type_1 == DIFF_FREQ) ? 0 : 1; } /* if( in_data_1 != NULL ) */ if (in_data_2 != NULL) { if (diff_type_2 == DIFF_FREQ) { p0_data_2[0] = NULL; p0_data_2[1] = &in_data_2[0]; num_val_2_short -= 1; in_data_2_short += 1; } df_rest_flag[1] = num_val_2_short % 2; if (df_rest_flag[1]) num_val_2_short -= 1; for (i = 0; i < num_val_2_short - 1; i += 2) { pair_vec[i + 1][0] = in_data_2_short[i]; pair_vec[i + 1][1] = in_data_2_short[i + 1]; } tab_idx_2D[1][0] = (diff_type_2 == DIFF_TIME) ? 1 : 0; tab_idx_2D[1][1] = 0; tab_idx_1D[1] = (diff_type_2 == DIFF_FREQ) ? 0 : 1; } /* if( in_data_2 != NULL ) */ if (in_data_1 != NULL) { FDKwriteBits(strm, lavHuffVal[lav_idx[0]], lavHuffLen[lav_idx[0]]); huff_enc_2D(strm, data_type, tab_idx_2D[0], lav_idx[0], pair_vec, num_val_1_short, 2, p0_data_1); if (df_rest_flag[0]) { huff_enc_1D(strm, data_type, tab_idx_1D[0], in_data_1_short + num_val_1_short, 1, 0); } } if (in_data_2 != NULL) { FDKwriteBits(strm, lavHuffVal[lav_idx[1]], lavHuffLen[lav_idx[1]]); huff_enc_2D(strm, data_type, tab_idx_2D[1], lav_idx[1], pair_vec + 1, num_val_2_short, 2, p0_data_2); if (df_rest_flag[1]) { huff_enc_1D(strm, data_type, tab_idx_1D[1], in_data_2_short + num_val_2_short, 1, 0); } } break; /* FREQ_PAIR */ case TIME_PAIR: if ((diff_type_1 == DIFF_FREQ) || (diff_type_2 == DIFF_FREQ)) { p0_data_1[0] = &in_data_1[0]; p0_data_1[1] = &in_data_2[0]; in_data_1_short += 1; in_data_2_short += 1; num_val_1_short -= 1; } for (i = 0; i < num_val_1_short; i++) { pair_vec[i][0] = in_data_1_short[i]; pair_vec[i][1] = in_data_2_short[i]; } tab_idx_2D[0][0] = ((diff_type_1 == DIFF_TIME) || (diff_type_2 == DIFF_TIME)) ? 1 : 0; tab_idx_2D[0][1] = 1; FDKwriteBits(strm, lavHuffVal[lav_idx[0]], lavHuffLen[lav_idx[0]]); huff_enc_2D(strm, data_type, tab_idx_2D[0], lav_idx[0], pair_vec, num_val_1_short, 1, p0_data_1); break; /* TIME_PAIR */ } /* switch( cdg_scheme & PAIR_MASK ) */ break; /* HUFF_2D */ default: break; } /* switch( cdg_scheme >> PAIR_SHIFT ) */ } INT fdk_sacenc_ecDataPairEnc(HANDLE_FDK_BITSTREAM strm, SHORT aaInData[][MAXBANDS], SHORT aHistory[MAXBANDS], const DATA_TYPE data_type, const INT setIdx, const INT startBand, const INT dataBands, const INT coarse_flag, const INT independency_flag) { SHORT reset = 0, pb = 0; SHORT quant_levels = 0, quant_offset = 0, num_pcm_val = 0; SHORT splitLsb_flag = 0; SHORT pcmCoding_flag = 0; SHORT allowDiffTimeBack_flag = !independency_flag || (setIdx > 0); SHORT num_lsb_bits = -1; SHORT num_pcm_bits = -1; SHORT quant_data_lsb[2][MAXBANDS]; SHORT quant_data_msb[2][MAXBANDS]; SHORT quant_data_hist_lsb[MAXBANDS]; SHORT quant_data_hist_msb[MAXBANDS]; SHORT data_diff_freq[2][MAXBANDS]; SHORT data_diff_time[2][MAXBANDS + 2]; SHORT *p_quant_data_msb[2]; SHORT *p_quant_data_hist_msb = NULL; SHORT min_bits_all = 0; SHORT min_found = 0; SHORT min_bits_df_df = -1; SHORT min_bits_df_dt = -1; SHORT min_bits_dtbw_df = -1; SHORT min_bits_dt_dt = -1; SHORT lav_df_df[2] = {-1, -1}; SHORT lav_df_dt[2] = {-1, -1}; SHORT lav_dtbw_df[2] = {-1, -1}; SHORT lav_dt_dt[2] = {-1, -1}; SHORT coding_scheme_df_df = 0; SHORT coding_scheme_df_dt = 0; SHORT coding_scheme_dtbw_df = 0; SHORT coding_scheme_dt_dt = 0; switch (data_type) { case t_CLD: if (coarse_flag) { splitLsb_flag = 0; quant_levels = 15; quant_offset = 7; } else { splitLsb_flag = 0; quant_levels = 31; quant_offset = 15; } break; case t_ICC: if (coarse_flag) { splitLsb_flag = 0; quant_levels = 4; quant_offset = 0; } else { splitLsb_flag = 0; quant_levels = 8; quant_offset = 0; } break; } /* switch( data_type ) */ /* Split off LSB */ if (splitLsb_flag) { split_lsb(aaInData[setIdx] + startBand, quant_offset, dataBands, quant_data_lsb[0], quant_data_msb[0]); split_lsb(aaInData[setIdx + 1] + startBand, quant_offset, dataBands, quant_data_lsb[1], quant_data_msb[1]); p_quant_data_msb[0] = quant_data_msb[0]; p_quant_data_msb[1] = quant_data_msb[1]; num_lsb_bits = 2 * dataBands; } else if (quant_offset != 0) { for (pb = 0; pb < dataBands; pb++) { quant_data_msb[0][pb] = aaInData[setIdx][startBand + pb] + quant_offset; quant_data_msb[1][pb] = aaInData[setIdx + 1][startBand + pb] + quant_offset; } p_quant_data_msb[0] = quant_data_msb[0]; p_quant_data_msb[1] = quant_data_msb[1]; num_lsb_bits = 0; } else { p_quant_data_msb[0] = aaInData[setIdx] + startBand; p_quant_data_msb[1] = aaInData[setIdx + 1] + startBand; num_lsb_bits = 0; } if (allowDiffTimeBack_flag) { if (splitLsb_flag) { split_lsb(aHistory + startBand, quant_offset, dataBands, quant_data_hist_lsb, quant_data_hist_msb); p_quant_data_hist_msb = quant_data_hist_msb; } else if (quant_offset != 0) { for (pb = 0; pb < dataBands; pb++) { quant_data_hist_msb[pb] = aHistory[startBand + pb] + quant_offset; } p_quant_data_hist_msb = quant_data_hist_msb; } else { p_quant_data_hist_msb = aHistory + startBand; } } /* Calculate frequency differences */ calc_diff_freq(p_quant_data_msb[0], data_diff_freq[0], dataBands); calc_diff_freq(p_quant_data_msb[1], data_diff_freq[1], dataBands); /* Calculate time differences */ if (allowDiffTimeBack_flag) { calc_diff_time(p_quant_data_msb[0], p_quant_data_hist_msb, data_diff_time[0], dataBands); } calc_diff_time(p_quant_data_msb[1], p_quant_data_msb[0], data_diff_time[1], dataBands); /* Calculate coding scheme with minumum bit consumption */ /**********************************************************/ num_pcm_bits = calc_pcm_bits(2 * dataBands, quant_levels); num_pcm_val = 2 * dataBands; /**********************************************************/ min_bits_all = num_pcm_bits; /**********************************************************/ /**********************************************************/ /**********************************************************/ min_bits_df_df = calc_huff_bits(data_diff_freq[0], data_diff_freq[1], data_type, DIFF_FREQ, DIFF_FREQ, dataBands, lav_df_df, &coding_scheme_df_df); min_bits_df_df += 2; min_bits_df_df += num_lsb_bits; if (min_bits_df_df < min_bits_all) { min_bits_all = min_bits_df_df; } /**********************************************************/ /**********************************************************/ min_bits_df_dt = calc_huff_bits(data_diff_freq[0], data_diff_time[1], data_type, DIFF_FREQ, DIFF_TIME, dataBands, lav_df_dt, &coding_scheme_df_dt); min_bits_df_dt += 2; min_bits_df_dt += num_lsb_bits; if (min_bits_df_dt < min_bits_all) { min_bits_all = min_bits_df_dt; } /**********************************************************/ /**********************************************************/ /**********************************************************/ if (allowDiffTimeBack_flag) { /**********************************************************/ min_bits_dtbw_df = calc_huff_bits( data_diff_time[0], data_diff_freq[1], data_type, DIFF_TIME, DIFF_FREQ, dataBands, lav_dtbw_df, &coding_scheme_dtbw_df); min_bits_dtbw_df += 2; min_bits_dtbw_df += num_lsb_bits; if (min_bits_dtbw_df < min_bits_all) { min_bits_all = min_bits_dtbw_df; } /**********************************************************/ /**********************************************************/ min_bits_dt_dt = calc_huff_bits(data_diff_time[0], data_diff_time[1], data_type, DIFF_TIME, DIFF_TIME, dataBands, lav_dt_dt, &coding_scheme_dt_dt); min_bits_dt_dt += 2; min_bits_dt_dt += num_lsb_bits; if (min_bits_dt_dt < min_bits_all) { min_bits_all = min_bits_dt_dt; } /**********************************************************/ } /* if( allowDiffTimeBack_flag ) */ /***************************/ /* Start actual coding now */ /***************************/ /* PCM or Diff/Huff Coding? */ pcmCoding_flag = (min_bits_all == num_pcm_bits); FDKwriteBits(strm, pcmCoding_flag, 1); if (pcmCoding_flag) { /* Grouped PCM Coding */ apply_pcm_coding(strm, aaInData[setIdx] + startBand, aaInData[setIdx + 1] + startBand, quant_offset, num_pcm_val, quant_levels); } else { /* Diff/Huff Coding */ min_found = 0; /*******************************************/ if (min_bits_all == min_bits_df_df) { FDKwriteBits(strm, DIFF_FREQ, 1); FDKwriteBits(strm, DIFF_FREQ, 1); apply_huff_coding(strm, data_diff_freq[0], data_diff_freq[1], data_type, DIFF_FREQ, DIFF_FREQ, dataBands, lav_df_df, coding_scheme_df_df); min_found = 1; } /*******************************************/ /*******************************************/ if (!min_found && (min_bits_all == min_bits_df_dt)) { FDKwriteBits(strm, DIFF_FREQ, 1); FDKwriteBits(strm, DIFF_TIME, 1); apply_huff_coding(strm, data_diff_freq[0], data_diff_time[1], data_type, DIFF_FREQ, DIFF_TIME, dataBands, lav_df_dt, coding_scheme_df_dt); min_found = 1; } /*******************************************/ /*******************************************/ /*******************************************/ if (allowDiffTimeBack_flag) { /*******************************************/ if (!min_found && (min_bits_all == min_bits_dtbw_df)) { FDKwriteBits(strm, DIFF_TIME, 1); FDKwriteBits(strm, DIFF_FREQ, 1); apply_huff_coding(strm, data_diff_time[0], data_diff_freq[1], data_type, DIFF_TIME, DIFF_FREQ, dataBands, lav_dtbw_df, coding_scheme_dtbw_df); min_found = 1; } /*******************************************/ /*******************************************/ if (!min_found && (min_bits_all == min_bits_dt_dt)) { FDKwriteBits(strm, DIFF_TIME, 1); FDKwriteBits(strm, DIFF_TIME, 1); apply_huff_coding(strm, data_diff_time[0], data_diff_time[1], data_type, DIFF_TIME, DIFF_TIME, dataBands, lav_dt_dt, coding_scheme_dt_dt); } /*******************************************/ } /* if( allowDiffTimeBack_flag ) */ /* LSB coding */ if (splitLsb_flag) { apply_lsb_coding(strm, quant_data_lsb[0], 1, dataBands); apply_lsb_coding(strm, quant_data_lsb[1], 1, dataBands); } } /* Diff/Huff/LSB coding */ return reset; } INT fdk_sacenc_ecDataSingleEnc(HANDLE_FDK_BITSTREAM strm, SHORT aaInData[][MAXBANDS], SHORT aHistory[MAXBANDS], const DATA_TYPE data_type, const INT setIdx, const INT startBand, const INT dataBands, const INT coarse_flag, const INT independency_flag) { SHORT reset = 0, pb = 0; SHORT quant_levels = 0, quant_offset = 0, num_pcm_val = 0; SHORT splitLsb_flag = 0; SHORT pcmCoding_flag = 0; SHORT allowDiffTimeBack_flag = !independency_flag || (setIdx > 0); SHORT num_lsb_bits = -1; SHORT num_pcm_bits = -1; SHORT quant_data_lsb[MAXBANDS]; SHORT quant_data_msb[MAXBANDS]; SHORT quant_data_hist_lsb[MAXBANDS]; SHORT quant_data_hist_msb[MAXBANDS]; SHORT data_diff_freq[MAXBANDS]; SHORT data_diff_time[MAXBANDS + 2]; SHORT *p_quant_data_msb; SHORT *p_quant_data_hist_msb = NULL; SHORT min_bits_all = 0; SHORT min_found = 0; SHORT min_bits_df = -1; SHORT min_bits_dt = -1; SHORT lav_df[2] = {-1, -1}; SHORT lav_dt[2] = {-1, -1}; SHORT coding_scheme_df = 0; SHORT coding_scheme_dt = 0; switch (data_type) { case t_CLD: if (coarse_flag) { splitLsb_flag = 0; quant_levels = 15; quant_offset = 7; } else { splitLsb_flag = 0; quant_levels = 31; quant_offset = 15; } break; case t_ICC: if (coarse_flag) { splitLsb_flag = 0; quant_levels = 4; quant_offset = 0; } else { splitLsb_flag = 0; quant_levels = 8; quant_offset = 0; } break; } /* switch( data_type ) */ /* Split off LSB */ if (splitLsb_flag) { split_lsb(aaInData[setIdx] + startBand, quant_offset, dataBands, quant_data_lsb, quant_data_msb); p_quant_data_msb = quant_data_msb; num_lsb_bits = dataBands; } else if (quant_offset != 0) { for (pb = 0; pb < dataBands; pb++) { quant_data_msb[pb] = aaInData[setIdx][startBand + pb] + quant_offset; } p_quant_data_msb = quant_data_msb; num_lsb_bits = 0; } else { p_quant_data_msb = aaInData[setIdx] + startBand; num_lsb_bits = 0; } if (allowDiffTimeBack_flag) { if (splitLsb_flag) { split_lsb(aHistory + startBand, quant_offset, dataBands, quant_data_hist_lsb, quant_data_hist_msb); p_quant_data_hist_msb = quant_data_hist_msb; } else if (quant_offset != 0) { for (pb = 0; pb < dataBands; pb++) { quant_data_hist_msb[pb] = aHistory[startBand + pb] + quant_offset; } p_quant_data_hist_msb = quant_data_hist_msb; } else { p_quant_data_hist_msb = aHistory + startBand; } } /* Calculate frequency differences */ calc_diff_freq(p_quant_data_msb, data_diff_freq, dataBands); /* Calculate time differences */ if (allowDiffTimeBack_flag) { calc_diff_time(p_quant_data_msb, p_quant_data_hist_msb, data_diff_time, dataBands); } /* Calculate coding scheme with minumum bit consumption */ /**********************************************************/ num_pcm_bits = calc_pcm_bits(dataBands, quant_levels); num_pcm_val = dataBands; /**********************************************************/ min_bits_all = num_pcm_bits; /**********************************************************/ /**********************************************************/ /**********************************************************/ min_bits_df = calc_huff_bits(data_diff_freq, NULL, data_type, DIFF_FREQ, DIFF_FREQ, dataBands, lav_df, &coding_scheme_df); if (allowDiffTimeBack_flag) min_bits_df += 1; min_bits_df += num_lsb_bits; if (min_bits_df < min_bits_all) { min_bits_all = min_bits_df; } /**********************************************************/ /**********************************************************/ if (allowDiffTimeBack_flag) { min_bits_dt = calc_huff_bits(data_diff_time, NULL, data_type, DIFF_TIME, DIFF_TIME, dataBands, lav_dt, &coding_scheme_dt); min_bits_dt += 1; min_bits_dt += num_lsb_bits; if (min_bits_dt < min_bits_all) { min_bits_all = min_bits_dt; } } /* if( allowDiffTimeBack_flag ) */ /***************************/ /* Start actual coding now */ /***************************/ /* PCM or Diff/Huff Coding? */ pcmCoding_flag = (min_bits_all == num_pcm_bits); FDKwriteBits(strm, pcmCoding_flag, 1); if (pcmCoding_flag) { /* Grouped PCM Coding */ apply_pcm_coding(strm, aaInData[setIdx] + startBand, NULL, quant_offset, num_pcm_val, quant_levels); } else { /* Diff/Huff Coding */ min_found = 0; /*******************************************/ if (min_bits_all == min_bits_df) { if (allowDiffTimeBack_flag) { FDKwriteBits(strm, DIFF_FREQ, 1); } apply_huff_coding(strm, data_diff_freq, NULL, data_type, DIFF_FREQ, DIFF_FREQ, dataBands, lav_df, coding_scheme_df); min_found = 1; } /* if( min_bits_all == min_bits_df ) */ /*******************************************/ /*******************************************/ if (allowDiffTimeBack_flag) { /*******************************************/ if (!min_found && (min_bits_all == min_bits_dt)) { FDKwriteBits(strm, DIFF_TIME, 1); apply_huff_coding(strm, data_diff_time, NULL, data_type, DIFF_TIME, DIFF_TIME, dataBands, lav_dt, coding_scheme_dt); } /*******************************************/ } /* if( allowDiffTimeBack_flag ) */ /* LSB coding */ if (splitLsb_flag) { apply_lsb_coding(strm, quant_data_lsb, 1, dataBands); } } /* Diff/Huff/LSB coding */ return reset; } fdk-aac-2.0.2/libSACenc/src/sacenc_nlc_enc.h000066400000000000000000000144261404225310700204460ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround encoder library ************************* Author(s): Karsten Linzmeier Description: Noiseless Coding Huffman encoder *******************************************************************************/ #ifndef SACENC_NLC_ENC_H #define SACENC_NLC_ENC_H /* Includes ******************************************************************/ #include "sacenc_const.h" #include "FDK_bitstream.h" #include "sacenc_bitstream.h" /* Defines *******************************************************************/ #define MAXBANDS MAX_NUM_BINS /* maximum number of frequency bands */ /* Data Types ****************************************************************/ typedef enum { t_CLD, t_ICC } DATA_TYPE; /* Constants *****************************************************************/ /* Function / Class Declarations *********************************************/ INT fdk_sacenc_ecDataPairEnc(HANDLE_FDK_BITSTREAM strm, SHORT aaInData[][MAXBANDS], SHORT aHistory[MAXBANDS], const DATA_TYPE data_type, const INT setIdx, const INT startBand, const INT dataBands, const INT coarse_flag, const INT independency_flag); INT fdk_sacenc_ecDataSingleEnc(HANDLE_FDK_BITSTREAM strm, SHORT aaInData[][MAXBANDS], SHORT aHistory[MAXBANDS], const DATA_TYPE data_type, const INT setIdx, const INT startBand, const INT dataBands, const INT coarse_flag, const INT independency_flag); #endif /* SACENC_NLC_ENC_H */ fdk-aac-2.0.2/libSACenc/src/sacenc_onsetdetect.cpp000066400000000000000000000337371404225310700217270ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround encoder library ************************* Author(s): Max Neuendorf Description: Encoder Library Interface Detect Onset in current frame *******************************************************************************/ /**************************************************************************/ /** \file Description of file contents ******************************************************************************/ /* Includes ******************************************************************/ #include "sacenc_onsetdetect.h" #include "genericStds.h" #include "sacenc_vectorfunctions.h" /* Defines *******************************************************************/ #define SPACE_ONSET_THRESHOLD (3.0) #define SPACE_ONSET_THRESHOLD_SF (3) #define SPACE_ONSET_THRESHOLD_SQUARE \ (FL2FXCONST_DBL((1.0 / (SPACE_ONSET_THRESHOLD * SPACE_ONSET_THRESHOLD)) * \ (float)(1 << SPACE_ONSET_THRESHOLD_SF))) /* Data Types ****************************************************************/ struct ONSET_DETECT { INT maxTimeSlots; INT minTransientDistance; INT avgEnergyDistance; INT lowerBoundOnsetDetection; INT upperBoundOnsetDetection; FIXP_DBL *pEnergyHist__FDK; SCHAR *pEnergyHistScale; SCHAR avgEnergyDistanceScale; }; /* Constants *****************************************************************/ /* Function / Class Declarations *********************************************/ /* Function / Class Definition ***********************************************/ FDK_SACENC_ERROR fdk_sacenc_onsetDetect_Open(HANDLE_ONSET_DETECT *phOnset, const UINT maxTimeSlots) { FDK_SACENC_ERROR error = SACENC_OK; HANDLE_ONSET_DETECT hOnset = NULL; if (NULL == phOnset) { error = SACENC_INVALID_HANDLE; } else { /* Memory Allocation */ FDK_ALLOCATE_MEMORY_1D(hOnset, 1, struct ONSET_DETECT); FDK_ALLOCATE_MEMORY_1D(hOnset->pEnergyHist__FDK, 16 + maxTimeSlots, FIXP_DBL); FDK_ALLOCATE_MEMORY_1D(hOnset->pEnergyHistScale, 16 + maxTimeSlots, SCHAR); hOnset->maxTimeSlots = maxTimeSlots; hOnset->minTransientDistance = 8; /* minimum distance between detected transients */ hOnset->avgEnergyDistance = 16; /* average energy distance */ hOnset->avgEnergyDistanceScale = 4; *phOnset = hOnset; } return error; bail: fdk_sacenc_onsetDetect_Close(&hOnset); return ((SACENC_OK == error) ? SACENC_MEMORY_ERROR : error); } FDK_SACENC_ERROR fdk_sacenc_onsetDetect_Init( HANDLE_ONSET_DETECT hOnset, const ONSET_DETECT_CONFIG *const pOnsetDetectConfig, const UINT initFlags) { FDK_SACENC_ERROR error = SACENC_OK; if ((NULL == hOnset) || (pOnsetDetectConfig == NULL)) { error = SACENC_INVALID_HANDLE; } else { if ((pOnsetDetectConfig->maxTimeSlots > hOnset->maxTimeSlots) || (pOnsetDetectConfig->upperBoundOnsetDetection < hOnset->lowerBoundOnsetDetection)) { error = SACENC_INVALID_CONFIG; goto bail; } hOnset->maxTimeSlots = pOnsetDetectConfig->maxTimeSlots; hOnset->lowerBoundOnsetDetection = pOnsetDetectConfig->lowerBoundOnsetDetection; hOnset->upperBoundOnsetDetection = pOnsetDetectConfig->upperBoundOnsetDetection; hOnset->minTransientDistance = 8; /* minimum distance between detected transients */ hOnset->avgEnergyDistance = 16; /* average energy distance */ hOnset->avgEnergyDistanceScale = 4; /* Init / Reset */ if (initFlags) { int i; for (i = 0; i < hOnset->avgEnergyDistance + hOnset->maxTimeSlots; i++) hOnset->pEnergyHistScale[i] = -(DFRACT_BITS - 3); FDKmemset_flex( hOnset->pEnergyHist__FDK, FL2FXCONST_DBL(SACENC_FLOAT_EPSILON * (1 << (DFRACT_BITS - 3))), hOnset->avgEnergyDistance + hOnset->maxTimeSlots); } } bail: return error; } /**************************************************************************/ FDK_SACENC_ERROR fdk_sacenc_onsetDetect_Close(HANDLE_ONSET_DETECT *phOnset) { FDK_SACENC_ERROR error = SACENC_OK; if ((NULL != phOnset) && (NULL != *phOnset)) { if (NULL != (*phOnset)->pEnergyHist__FDK) { FDKfree((*phOnset)->pEnergyHist__FDK); } (*phOnset)->pEnergyHist__FDK = NULL; if (NULL != (*phOnset)->pEnergyHistScale) { FDKfree((*phOnset)->pEnergyHistScale); } (*phOnset)->pEnergyHistScale = NULL; FDKfree(*phOnset); *phOnset = NULL; } return error; } /**************************************************************************/ FDK_SACENC_ERROR fdk_sacenc_onsetDetect_Update(HANDLE_ONSET_DETECT hOnset, const INT timeSlots) { FDK_SACENC_ERROR error = SACENC_OK; if (NULL == hOnset) { error = SACENC_INVALID_HANDLE; } else { if (timeSlots > hOnset->maxTimeSlots) { error = SACENC_INVALID_CONFIG; } else { int i; /* Shift old data */ for (i = 0; i < hOnset->avgEnergyDistance; i++) { hOnset->pEnergyHist__FDK[i] = hOnset->pEnergyHist__FDK[i + timeSlots]; hOnset->pEnergyHistScale[i] = hOnset->pEnergyHistScale[i + timeSlots]; } /* Clear for new data */ FDKmemset_flex(&hOnset->pEnergyHist__FDK[hOnset->avgEnergyDistance], FL2FXCONST_DBL(SACENC_FLOAT_EPSILON), timeSlots); } } return error; } /**************************************************************************/ FDK_SACENC_ERROR fdk_sacenc_onsetDetect_Apply( HANDLE_ONSET_DETECT hOnset, const INT nTimeSlots, const INT nHybridBands, FIXP_DPK *const *const ppHybridData__FDK, const INT hybridDataScale, const INT prevPos, INT pTransientPos[MAX_NUM_TRANS]) { FDK_SACENC_ERROR error = SACENC_OK; C_ALLOC_SCRATCH_START(envs, FIXP_DBL, (16 + MAX_TIME_SLOTS)) FDKmemclear(envs, (16 + MAX_TIME_SLOTS) * sizeof(FIXP_DBL)); if ((hOnset == NULL) || (pTransientPos == NULL) || (ppHybridData__FDK == NULL)) { error = SACENC_INVALID_HANDLE; } else { int i, ts, trCnt, currPos; if ((nTimeSlots < 0) || (nTimeSlots > hOnset->maxTimeSlots) || (hOnset->lowerBoundOnsetDetection < -1) || (hOnset->upperBoundOnsetDetection > nHybridBands)) { error = SACENC_INVALID_CONFIG; goto bail; } const int lowerBoundOnsetDetection = hOnset->lowerBoundOnsetDetection; const int upperBoundOnsetDetection = hOnset->upperBoundOnsetDetection; const int M = hOnset->avgEnergyDistance; { SCHAR *envScale = hOnset->pEnergyHistScale; FIXP_DBL *env = hOnset->pEnergyHist__FDK; const FIXP_DBL threshold_square = SPACE_ONSET_THRESHOLD_SQUARE; trCnt = 0; /* reset transient array */ FDKmemset_flex(pTransientPos, -1, MAX_NUM_TRANS); /* minimum transient distance of minTransDist QMF samples */ if (prevPos > 0) { currPos = FDKmax(nTimeSlots, prevPos - nTimeSlots + hOnset->minTransientDistance); } else { currPos = nTimeSlots; } /* get energy and scalefactor for each time slot */ int outScale; int inScale = 3; /* scale factor determined empirically */ for (ts = 0; ts < nTimeSlots; ts++) { env[M + ts] = sumUpCplxPow2( &ppHybridData__FDK[ts][lowerBoundOnsetDetection + 1], SUM_UP_DYNAMIC_SCALE, inScale, &outScale, upperBoundOnsetDetection - lowerBoundOnsetDetection - 1); envScale[M + ts] = outScale + (hybridDataScale << 1); } /* calculate common scale for all time slots */ SCHAR maxScale = -(DFRACT_BITS - 1); for (i = 0; i < (nTimeSlots + M); i++) { maxScale = fixMax(maxScale, envScale[i]); } /* apply common scale and store energy in temporary buffer */ for (i = 0; i < (nTimeSlots + M); i++) { envs[i] = env[i] >> fixMin((maxScale - envScale[i]), (DFRACT_BITS - 1)); } FIXP_DBL maxVal = FL2FXCONST_DBL(0.0f); for (i = 0; i < (nTimeSlots + M); i++) { maxVal |= fAbs(envs[i]); } int s = fixMax(0, CntLeadingZeros(maxVal) - 1); for (i = 0; i < (nTimeSlots + M); i++) { envs[i] = envs[i] << s; } int currPosPrev = currPos; FIXP_DBL p1, p2; p2 = FL2FXCONST_DBL(0.0f); for (; (currPos < (nTimeSlots << 1)) && (trCnt < MAX_NUM_TRANS); currPos++) { p1 = fMultDiv2(envs[currPos - nTimeSlots + M], threshold_square) >> (SPACE_ONSET_THRESHOLD_SF - 1); /* Calculate average of past M energy values */ if (currPosPrev == (currPos - 1)) { /* remove last and add new element */ p2 -= (envs[currPosPrev - nTimeSlots] >> (int)hOnset->avgEnergyDistanceScale); p2 += (envs[currPos - nTimeSlots + M - 1] >> (int)hOnset->avgEnergyDistanceScale); } else { /* calculate complete vector */ p2 = FL2FXCONST_DBL(0.0f); for (ts = 0; ts < M; ts++) { p2 += (envs[currPos - nTimeSlots + ts] >> (int)hOnset->avgEnergyDistanceScale); } } currPosPrev = currPos; { /* save position if transient found */ if (p1 > p2) { pTransientPos[trCnt++] = currPos; currPos += hOnset->minTransientDistance; } } } /* for currPos */ } } /* valid handle*/ bail: C_ALLOC_SCRATCH_END(envs, FIXP_DBL, (16 + MAX_TIME_SLOTS)) return error; } /**************************************************************************/ fdk-aac-2.0.2/libSACenc/src/sacenc_onsetdetect.h000066400000000000000000000152761404225310700213720ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround encoder library ************************* Author(s): Max Neuendorf Description: Encoder Library Interface Determine TES flags *******************************************************************************/ #ifndef SACENC_ONSETDETECT_H #define SACENC_ONSETDETECT_H /**************************************************************************/ /** \file Description of file contents ******************************************************************************/ /* Includes ******************************************************************/ #include "common_fix.h" #include "FDK_matrixCalloc.h" #include "sacenc_lib.h" #include "sacenc_bitstream.h" /* for def. of MAX_NUM_PARAMS */ /* Defines *******************************************************************/ #define MAX_NUM_TRANS (MAX_NUM_PARAMS / 2) /* Data Types ****************************************************************/ typedef struct T_ONSET_DETECT_CONFIG { INT maxTimeSlots; /* calc transien detection in ]lowerBoundOnsetDetection; * upperBoundOnsetDetection[ */ INT lowerBoundOnsetDetection; INT upperBoundOnsetDetection; } ONSET_DETECT_CONFIG; typedef struct ONSET_DETECT *HANDLE_ONSET_DETECT; /* Constants *****************************************************************/ /* Function / Class Declarations *********************************************/ FDK_SACENC_ERROR fdk_sacenc_onsetDetect_Open(HANDLE_ONSET_DETECT *phOnset, const UINT maxTimeSlots); FDK_SACENC_ERROR fdk_sacenc_onsetDetect_Init( HANDLE_ONSET_DETECT hOnset, const ONSET_DETECT_CONFIG *const pOnsetDetectConfig, const UINT initFlags); FDK_SACENC_ERROR fdk_sacenc_onsetDetect_Close(HANDLE_ONSET_DETECT *phOnset); FDK_SACENC_ERROR fdk_sacenc_onsetDetect_Update(HANDLE_ONSET_DETECT hOnset, const INT timeSlots); FDK_SACENC_ERROR fdk_sacenc_onsetDetect_Apply( HANDLE_ONSET_DETECT hOnset, const INT nTimeSlots, const INT nHybridBands, FIXP_DPK *const *const ppHybridData__FDK, const INT hybridDataScale, const INT prevPos, INT pTransientPos[MAX_NUM_TRANS]); #endif /* SACENC_ONSETDETECT_H */ fdk-aac-2.0.2/libSACenc/src/sacenc_paramextract.cpp000066400000000000000000000715351404225310700220770ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround encoder library ************************* Author(s): M. Multrus Description: Parameter Extraction *******************************************************************************/ /* Includes ******************************************************************/ #include "sacenc_paramextract.h" #include "sacenc_tree.h" #include "sacenc_vectorfunctions.h" /* Defines *******************************************************************/ #define LOG10_2_10 (3.01029995664f) /* 10.0f*log10(2.f) */ #define SCALE_CLDE_SF (7) /* maxVal in Quant tab is +/- 50 */ #define SCALE_CLDD_SF (8) /* maxVal in Quant tab is +/- 150 */ /* Data Types ****************************************************************/ typedef struct T_TTO_BOX { FIXP_DBL pCld__FDK[MAX_NUM_PARAM_BANDS]; FIXP_DBL pIcc__FDK[MAX_NUM_PARAM_BANDS]; FIXP_DBL pCldQuant__FDK[MAX_NUM_PARAM_BANDS]; const FIXP_DBL *pIccQuantTable__FDK; const FIXP_DBL *pCldQuantTableDec__FDK; const FIXP_DBL *pCldQuantTableEnc__FDK; SCHAR pCldEbQIdx[MAX_NUM_PARAM_BANDS]; SCHAR pIccDownmixIdx[MAX_NUM_PARAM_BANDS]; UCHAR *pParameterBand2HybridBandOffset; const INT *pSubbandImagSign; UCHAR nHybridBandsMax; UCHAR nParameterBands; UCHAR bFrameKeep; UCHAR iccCorrelationCoherenceBorder; BOX_QUANTMODE boxQuantMode; UCHAR nIccQuantSteps; UCHAR nIccQuantOffset; UCHAR nCldQuantSteps; UCHAR nCldQuantOffset; UCHAR bUseCoarseQuantCld; UCHAR bUseCoarseQuantIcc; } TTO_BOX; struct BOX_SUBBAND_SETUP { BOX_SUBBAND_CONFIG subbandConfig; UCHAR nParameterBands; const UCHAR *pSubband2ParameterIndexLd; UCHAR iccCorrelationCoherenceBorder; }; /* Constants *****************************************************************/ static const UCHAR subband2Parameter4_Ld[NUM_QMF_BANDS] = { 0, 0, 1, 1, 1, 1, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3}; static const UCHAR subband2Parameter5_Ld[NUM_QMF_BANDS] = { 0, 1, 1, 2, 2, 2, 2, 2, 2, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4}; static const UCHAR subband2Parameter7_Ld[NUM_QMF_BANDS] = { 0, 1, 2, 2, 3, 3, 3, 3, 4, 4, 4, 4, 4, 4, 5, 5, 5, 5, 5, 5, 5, 5, 5, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6}; static const UCHAR subband2Parameter9_Ld[NUM_QMF_BANDS] = { 0, 1, 2, 3, 3, 4, 4, 5, 5, 6, 6, 6, 6, 6, 7, 7, 7, 7, 7, 7, 7, 7, 7, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8}; static const UCHAR subband2Parameter12_Ld[NUM_QMF_BANDS] = { 0, 1, 2, 3, 4, 4, 5, 5, 6, 6, 6, 7, 7, 7, 8, 8, 8, 8, 9, 9, 9, 9, 9, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11, 11}; static const UCHAR subband2Parameter15_Ld[NUM_QMF_BANDS] = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 9, 10, 10, 10, 11, 11, 11, 11, 12, 12, 12, 12, 12, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 13, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14}; static const UCHAR subband2Parameter23_Ld[NUM_QMF_BANDS] = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 12, 13, 13, 14, 14, 15, 15, 16, 16, 16, 17, 17, 17, 18, 18, 18, 18, 19, 19, 19, 19, 19, 20, 20, 20, 20, 20, 20, 21, 21, 21, 21, 21, 21, 21, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22, 22}; static const INT subbandImagSign_Ld[NUM_QMF_BANDS] = { 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, }; #define SCALE_CLDE(a) (FL2FXCONST_DBL(a / (float)(1 << SCALE_CLDE_SF))) static const FIXP_DBL cldQuantTableFineEnc__FDK[MAX_CLD_QUANT_FINE] = { SCALE_CLDE(-50.0), SCALE_CLDE(-45.0), SCALE_CLDE(-40.0), SCALE_CLDE(-35.0), SCALE_CLDE(-30.0), SCALE_CLDE(-25.0), SCALE_CLDE(-22.0), SCALE_CLDE(-19.0), SCALE_CLDE(-16.0), SCALE_CLDE(-13.0), SCALE_CLDE(-10.0), SCALE_CLDE(-8.0), SCALE_CLDE(-6.0), SCALE_CLDE(-4.0), SCALE_CLDE(-2.0), SCALE_CLDE(0.0), SCALE_CLDE(2.0), SCALE_CLDE(4.0), SCALE_CLDE(6.0), SCALE_CLDE(8.0), SCALE_CLDE(10.0), SCALE_CLDE(13.0), SCALE_CLDE(16.0), SCALE_CLDE(19.0), SCALE_CLDE(22.0), SCALE_CLDE(25.0), SCALE_CLDE(30.0), SCALE_CLDE(35.0), SCALE_CLDE(40.0), SCALE_CLDE(45.0), SCALE_CLDE(50.0)}; static const FIXP_DBL cldQuantTableCoarseEnc__FDK[MAX_CLD_QUANT_COARSE] = { SCALE_CLDE(-50.0), SCALE_CLDE(-35.0), SCALE_CLDE(-25.0), SCALE_CLDE(-19.0), SCALE_CLDE(-13.0), SCALE_CLDE(-8.0), SCALE_CLDE(-4.0), SCALE_CLDE(0.0), SCALE_CLDE(4.0), SCALE_CLDE(8.0), SCALE_CLDE(13.0), SCALE_CLDE(19.0), SCALE_CLDE(25.0), SCALE_CLDE(35.0), SCALE_CLDE(50.0)}; #define SCALE_CLDD(a) (FL2FXCONST_DBL(a / (float)(1 << SCALE_CLDD_SF))) static const FIXP_DBL cldQuantTableFineDec__FDK[MAX_CLD_QUANT_FINE] = { SCALE_CLDD(-150.0), SCALE_CLDD(-45.0), SCALE_CLDD(-40.0), SCALE_CLDD(-35.0), SCALE_CLDD(-30.0), SCALE_CLDD(-25.0), SCALE_CLDD(-22.0), SCALE_CLDD(-19.0), SCALE_CLDD(-16.0), SCALE_CLDD(-13.0), SCALE_CLDD(-10.0), SCALE_CLDD(-8.0), SCALE_CLDD(-6.0), SCALE_CLDD(-4.0), SCALE_CLDD(-2.0), SCALE_CLDD(0.0), SCALE_CLDD(2.0), SCALE_CLDD(4.0), SCALE_CLDD(6.0), SCALE_CLDD(8.0), SCALE_CLDD(10.0), SCALE_CLDD(13.0), SCALE_CLDD(16.0), SCALE_CLDD(19.0), SCALE_CLDD(22.0), SCALE_CLDD(25.0), SCALE_CLDD(30.0), SCALE_CLDD(35.0), SCALE_CLDD(40.0), SCALE_CLDD(45.0), SCALE_CLDD(150.0)}; static const FIXP_DBL cldQuantTableCoarseDec__FDK[MAX_CLD_QUANT_COARSE] = { SCALE_CLDD(-150.0), SCALE_CLDD(-35.0), SCALE_CLDD(-25.0), SCALE_CLDD(-19.0), SCALE_CLDD(-13.0), SCALE_CLDD(-8.0), SCALE_CLDD(-4.0), SCALE_CLDD(0.0), SCALE_CLDD(4.0), SCALE_CLDD(8.0), SCALE_CLDD(13.0), SCALE_CLDD(19.0), SCALE_CLDD(25.0), SCALE_CLDD(35.0), SCALE_CLDD(150.0)}; #define SCALE_ICC(a) (FL2FXCONST_DBL(a)) static const FIXP_DBL iccQuantTableFine__FDK[MAX_ICC_QUANT_FINE] = { SCALE_ICC(0.99999999953), SCALE_ICC(0.937f), SCALE_ICC(0.84118f), SCALE_ICC(0.60092f), SCALE_ICC(0.36764f), SCALE_ICC(0.0f), SCALE_ICC(-0.589f), SCALE_ICC(-0.99f)}; static const FIXP_DBL iccQuantTableCoarse__FDK[MAX_ICC_QUANT_COARSE] = { SCALE_ICC(0.99999999953), SCALE_ICC(0.84118f), SCALE_ICC(0.36764f), SCALE_ICC(-0.5890f)}; static const BOX_SUBBAND_SETUP boxSubbandSetup[] = { {BOX_SUBBANDS_4, 4, subband2Parameter4_Ld, 1}, {BOX_SUBBANDS_5, 5, subband2Parameter5_Ld, 2}, {BOX_SUBBANDS_7, 7, subband2Parameter7_Ld, 3}, {BOX_SUBBANDS_9, 9, subband2Parameter9_Ld, 4}, {BOX_SUBBANDS_12, 12, subband2Parameter12_Ld, 4}, {BOX_SUBBANDS_15, 15, subband2Parameter15_Ld, 5}, {BOX_SUBBANDS_23, 23, subband2Parameter23_Ld, 8}}; /* Function / Class Declarations *********************************************/ /* Function / Class Definition ***********************************************/ static const BOX_SUBBAND_SETUP *getBoxSubbandSetup( const BOX_SUBBAND_CONFIG subbandConfig) { int i; const BOX_SUBBAND_SETUP *setup = NULL; for (i = 0; i < (int)(sizeof(boxSubbandSetup) / sizeof(BOX_SUBBAND_SETUP)); i++) { if (boxSubbandSetup[i].subbandConfig == subbandConfig) { setup = &boxSubbandSetup[i]; break; } } return setup; } static inline void ApplyBBCuesFDK(FIXP_DBL *const pData, const INT nParamBands) { int i, s; FIXP_DBL tmp, invParamBands; invParamBands = fDivNormHighPrec((FIXP_DBL)1, (FIXP_DBL)nParamBands, &s); s = -s; tmp = fMult(pData[0], invParamBands) >> s; for (i = 1; i < nParamBands; i++) { tmp += fMult(pData[i], invParamBands) >> s; } for (i = 0; i < nParamBands; i++) { pData[i] = tmp; } } static INT getNumberParameterBands(const BOX_SUBBAND_CONFIG subbandConfig) { const BOX_SUBBAND_SETUP *setup = getBoxSubbandSetup(subbandConfig); return ((setup == NULL) ? 0 : setup->nParameterBands); } static const UCHAR *getSubband2ParameterIndex( const BOX_SUBBAND_CONFIG subbandConfig) { const BOX_SUBBAND_SETUP *setup = getBoxSubbandSetup(subbandConfig); return ((setup == NULL) ? NULL : (setup->pSubband2ParameterIndexLd)); } void fdk_sacenc_calcParameterBand2HybridBandOffset( const BOX_SUBBAND_CONFIG subbandConfig, const INT nHybridBands, UCHAR *pParameterBand2HybridBandOffset) { const BOX_SUBBAND_SETUP *setup = getBoxSubbandSetup(subbandConfig); const UCHAR *pSubband2ParameterIndex; int i, pb; pSubband2ParameterIndex = setup->pSubband2ParameterIndexLd; for (pb = 0, i = 0; i < nHybridBands - 1; i++) { if (pSubband2ParameterIndex[i + 1] - pSubband2ParameterIndex[i]) { pParameterBand2HybridBandOffset[pb++] = (i + 1); } } pParameterBand2HybridBandOffset[pb++] = (i + 1); } const INT *fdk_sacenc_getSubbandImagSign() { const INT *pImagSign = NULL; pImagSign = subbandImagSign_Ld; return (pImagSign); } static INT getIccCorrelationCoherenceBorder( const BOX_SUBBAND_CONFIG subbandConfig, const INT bUseCoherenceOnly) { const BOX_SUBBAND_SETUP *setup = getBoxSubbandSetup(subbandConfig); return ( (setup == NULL) ? 0 : ((bUseCoherenceOnly) ? 0 : setup->iccCorrelationCoherenceBorder)); } FDK_SACENC_ERROR fdk_sacenc_createTtoBox(HANDLE_TTO_BOX *hTtoBox) { FDK_SACENC_ERROR error = SACENC_OK; if (NULL == hTtoBox) { error = SACENC_INVALID_HANDLE; } else { FDK_ALLOCATE_MEMORY_1D(*hTtoBox, 1, TTO_BOX); } return error; bail: fdk_sacenc_destroyTtoBox(hTtoBox); return ((SACENC_OK == error) ? SACENC_MEMORY_ERROR : error); } FDK_SACENC_ERROR fdk_sacenc_initTtoBox(HANDLE_TTO_BOX hTtoBox, const TTO_BOX_CONFIG *const ttoBoxConfig, UCHAR *pParameterBand2HybridBandOffset) { FDK_SACENC_ERROR error = SACENC_OK; if ((hTtoBox == NULL) || (ttoBoxConfig == NULL) || (pParameterBand2HybridBandOffset == NULL)) { error = SACENC_INVALID_HANDLE; } else { FDKmemclear(hTtoBox, sizeof(TTO_BOX)); hTtoBox->bUseCoarseQuantCld = ttoBoxConfig->bUseCoarseQuantCld; hTtoBox->bUseCoarseQuantIcc = ttoBoxConfig->bUseCoarseQuantIcc; hTtoBox->boxQuantMode = ttoBoxConfig->boxQuantMode; hTtoBox->iccCorrelationCoherenceBorder = getIccCorrelationCoherenceBorder( ttoBoxConfig->subbandConfig, ttoBoxConfig->bUseCoherenceIccOnly); hTtoBox->nHybridBandsMax = ttoBoxConfig->nHybridBandsMax; hTtoBox->nParameterBands = getNumberParameterBands(ttoBoxConfig->subbandConfig); hTtoBox->bFrameKeep = ttoBoxConfig->bFrameKeep; hTtoBox->nIccQuantSteps = fdk_sacenc_getNumberIccQuantLevels(hTtoBox->bUseCoarseQuantIcc); hTtoBox->nIccQuantOffset = fdk_sacenc_getIccQuantOffset(hTtoBox->bUseCoarseQuantIcc); hTtoBox->pIccQuantTable__FDK = hTtoBox->bUseCoarseQuantIcc ? iccQuantTableCoarse__FDK : iccQuantTableFine__FDK; hTtoBox->pCldQuantTableDec__FDK = hTtoBox->bUseCoarseQuantCld ? cldQuantTableCoarseDec__FDK : cldQuantTableFineDec__FDK; hTtoBox->pCldQuantTableEnc__FDK = hTtoBox->bUseCoarseQuantCld ? cldQuantTableCoarseEnc__FDK : cldQuantTableFineEnc__FDK; hTtoBox->nCldQuantSteps = fdk_sacenc_getNumberCldQuantLevels(hTtoBox->bUseCoarseQuantCld); hTtoBox->nCldQuantOffset = fdk_sacenc_getCldQuantOffset(hTtoBox->bUseCoarseQuantCld); /* sanity */ if (NULL == (hTtoBox->pParameterBand2HybridBandOffset = pParameterBand2HybridBandOffset)) { error = SACENC_INIT_ERROR; goto bail; } if (NULL == (hTtoBox->pSubbandImagSign = fdk_sacenc_getSubbandImagSign())) { error = SACENC_INIT_ERROR; } if ((hTtoBox->boxQuantMode != BOX_QUANTMODE_FINE) && (hTtoBox->boxQuantMode != BOX_QUANTMODE_EBQ1) && (hTtoBox->boxQuantMode != BOX_QUANTMODE_EBQ2)) { error = SACENC_INIT_ERROR; goto bail; } } bail: return error; } FDK_SACENC_ERROR fdk_sacenc_destroyTtoBox(HANDLE_TTO_BOX *hTtoBox) { FDK_SACENC_ERROR error = SACENC_OK; if (*hTtoBox != NULL) { FDKfree(*hTtoBox); *hTtoBox = NULL; } return error; } static FDK_SACENC_ERROR calculateIccFDK(const INT nParamBand, const INT correlationCoherenceBorder, const FIXP_DBL *const pPwr1, const FIXP_DBL *const pPwr2, const FIXP_DBL *const pProdReal, FIXP_DBL const *const pProdImag, FIXP_DBL *const pIcc) { FDK_SACENC_ERROR error = SACENC_OK; if ((pPwr1 == NULL) || (pPwr2 == NULL) || (pProdReal == NULL) || (pProdImag == NULL) || (pIcc == NULL)) { error = SACENC_INVALID_HANDLE; } else { /* sanity check border */ if (correlationCoherenceBorder > nParamBand) { error = SACENC_INVALID_CONFIG; } else { /* correlation */ FDKcalcCorrelationVec(pIcc, pProdReal, pPwr1, pPwr2, correlationCoherenceBorder); /* coherence */ calcCoherenceVec(&pIcc[correlationCoherenceBorder], &pProdReal[correlationCoherenceBorder], &pProdImag[correlationCoherenceBorder], &pPwr1[correlationCoherenceBorder], &pPwr2[correlationCoherenceBorder], 0, 0, nParamBand - correlationCoherenceBorder); } /* valid configuration */ } /* valid handle */ return error; } static void QuantizeCoefFDK(const FIXP_DBL *const input, const INT nBands, const FIXP_DBL *const quantTable, const INT idxOffset, const INT nQuantSteps, SCHAR *const quantOut) { int band; const int reverse = (quantTable[0] > quantTable[1]); for (band = 0; band < nBands; band++) { FIXP_DBL qVal; FIXP_DBL curVal = input[band]; int lower = 0; int upper = nQuantSteps - 1; if (reverse) { while (upper - lower > 1) { int idx = (lower + upper) >> 1; qVal = quantTable[idx]; if (curVal >= qVal) { upper = idx; } else { lower = idx; } } /* while */ if ((curVal - quantTable[lower]) >= (quantTable[upper] - curVal)) { quantOut[band] = lower - idxOffset; } else { quantOut[band] = upper - idxOffset; } } /* if reverse */ else { while (upper - lower > 1) { int idx = (lower + upper) >> 1; qVal = quantTable[idx]; if (curVal <= qVal) { upper = idx; } else { lower = idx; } } /* while */ if ((curVal - quantTable[lower]) <= (quantTable[upper] - curVal)) { quantOut[band] = lower - idxOffset; } else { quantOut[band] = upper - idxOffset; } } /* else reverse */ } /* for band */ } static void deQuantizeCoefFDK(const SCHAR *const input, const INT nBands, const FIXP_DBL *const quantTable, const INT idxOffset, FIXP_DBL *const dequantOut) { int band; for (band = 0; band < nBands; band++) { dequantOut[band] = quantTable[input[band] + idxOffset]; } } static void CalculateCldFDK(FIXP_DBL *const pCld, const FIXP_DBL *const pPwr1, const FIXP_DBL *const pPwr2, const INT scaleCh1, const INT *const pbScaleCh1, const INT scaleCh2, const INT *const pbScaleCh2, const int nParamBand) { INT i; FIXP_DBL ldPwr1, ldPwr2, cld; FIXP_DBL maxPwr = FL2FXCONST_DBL( 30.0f / (1 << (LD_DATA_SHIFT + 1))); /* consider SACENC_FLOAT_EPSILON in power calculation */ for (i = 0; i < nParamBand; i++) { ldPwr1 = (CalcLdData(pPwr1[i]) >> 1) + ((FIXP_DBL)(scaleCh1 + pbScaleCh1[i]) << (DFRACT_BITS - 1 - LD_DATA_SHIFT)); ldPwr2 = (CalcLdData(pPwr2[i]) >> 1) + ((FIXP_DBL)(scaleCh2 + pbScaleCh2[i]) << (DFRACT_BITS - 1 - LD_DATA_SHIFT)); ldPwr1 = fixMax(fixMin(ldPwr1, maxPwr), -maxPwr); ldPwr2 = fixMax(fixMin(ldPwr2, maxPwr), -maxPwr); /* ldPwr1 and ldPwr2 are scaled by LD_DATA_SHIFT and additional 1 bit; 1 bit * scale by fMultDiv2() */ cld = fMultDiv2(FL2FXCONST_DBL(LOG10_2_10 / (1 << SCALE_CLDE_SF)), ldPwr1 - ldPwr2); cld = fixMin(cld, (FIXP_DBL)(((FIXP_DBL)MAXVAL_DBL) >> (LD_DATA_SHIFT + 2))); cld = fixMax(cld, (FIXP_DBL)(((FIXP_DBL)MINVAL_DBL) >> (LD_DATA_SHIFT + 2))); pCld[i] = cld << (LD_DATA_SHIFT + 2); } } FDK_SACENC_ERROR fdk_sacenc_applyTtoBox( HANDLE_TTO_BOX hTtoBox, const INT nTimeSlots, const INT startTimeSlot, const INT nHybridBands, const FIXP_DPK *const *const ppHybridData1__FDK, const FIXP_DPK *const *const ppHybridData2__FDK, SCHAR *const pIccIdx, UCHAR *const pbIccQuantCoarse, SCHAR *const pCldIdx, UCHAR *const pbCldQuantCoarse, const INT bUseBBCues, INT *scaleCh1, INT *scaleCh2) { FDK_SACENC_ERROR error = SACENC_OK; C_ALLOC_SCRATCH_START(powerHybridData1__FDK, FIXP_DBL, MAX_NUM_PARAM_BANDS) C_ALLOC_SCRATCH_START(powerHybridData2__FDK, FIXP_DBL, MAX_NUM_PARAM_BANDS) C_ALLOC_SCRATCH_START(prodHybridDataReal__FDK, FIXP_DBL, MAX_NUM_PARAM_BANDS) C_ALLOC_SCRATCH_START(prodHybridDataImag__FDK, FIXP_DBL, MAX_NUM_PARAM_BANDS) C_ALLOC_SCRATCH_START(IccDownmix__FDK, FIXP_DBL, MAX_NUM_PARAM_BANDS) C_ALLOC_SCRATCH_START(IccDownmixQuant__FDK, FIXP_DBL, MAX_NUM_PARAM_BANDS) C_ALLOC_SCRATCH_START(pbScaleCh1, INT, MAX_NUM_PARAM_BANDS) C_ALLOC_SCRATCH_START(pbScaleCh2, INT, MAX_NUM_PARAM_BANDS) if ((hTtoBox == NULL) || (pCldIdx == NULL) || (pbCldQuantCoarse == NULL) || (ppHybridData1__FDK == NULL) || (ppHybridData2__FDK == NULL) || (pIccIdx == NULL) || (pbIccQuantCoarse == NULL)) { error = SACENC_INVALID_HANDLE; } else { int j, pb; const int nParamBands = hTtoBox->nParameterBands; const int bUseEbQ = (hTtoBox->boxQuantMode == BOX_QUANTMODE_EBQ1) || (hTtoBox->boxQuantMode == BOX_QUANTMODE_EBQ2); /* sanity check */ if ((nHybridBands < 0) || (nHybridBands > hTtoBox->nHybridBandsMax)) { error = SACENC_INVALID_CONFIG; goto bail; } int outScale; /* scalefactor will not be evaluated */ int inScale = 5; /* scale factor determined empirically */ /* calculate the headroom of the hybrid data for each parameter band */ FDKcalcPbScaleFactor(ppHybridData1__FDK, hTtoBox->pParameterBand2HybridBandOffset, pbScaleCh1, startTimeSlot, nTimeSlots, nParamBands); FDKcalcPbScaleFactor(ppHybridData2__FDK, hTtoBox->pParameterBand2HybridBandOffset, pbScaleCh2, startTimeSlot, nTimeSlots, nParamBands); for (j = 0, pb = 0; pb < nParamBands; pb++) { FIXP_DBL data1, data2; data1 = data2 = (FIXP_DBL)0; for (; j < hTtoBox->pParameterBand2HybridBandOffset[pb]; j++) { data1 += sumUpCplxPow2Dim2(ppHybridData1__FDK, SUM_UP_STATIC_SCALE, inScale + pbScaleCh1[pb], &outScale, startTimeSlot, nTimeSlots, j, j + 1); data2 += sumUpCplxPow2Dim2(ppHybridData2__FDK, SUM_UP_STATIC_SCALE, inScale + pbScaleCh2[pb], &outScale, startTimeSlot, nTimeSlots, j, j + 1); } /* for j */ powerHybridData1__FDK[pb] = data1; powerHybridData2__FDK[pb] = data2; } /* pb */ { for (j = 0, pb = 0; pb < nParamBands; pb++) { FIXP_DBL dataReal, dataImag; dataReal = dataImag = (FIXP_DBL)0; for (; j < hTtoBox->pParameterBand2HybridBandOffset[pb]; j++) { FIXP_DPK scalarProd; cplx_cplxScalarProduct(&scalarProd, ppHybridData1__FDK, ppHybridData2__FDK, inScale + pbScaleCh1[pb], inScale + pbScaleCh2[pb], &outScale, startTimeSlot, nTimeSlots, j, j + 1); dataReal += scalarProd.v.re; if (hTtoBox->pSubbandImagSign[j] < 0) { dataImag -= scalarProd.v.im; } else { dataImag += scalarProd.v.im; } } /* for j */ prodHybridDataReal__FDK[pb] = dataReal; prodHybridDataImag__FDK[pb] = dataImag; } /* pb */ if (SACENC_OK != (error = calculateIccFDK( nParamBands, hTtoBox->iccCorrelationCoherenceBorder, powerHybridData1__FDK, powerHybridData2__FDK, prodHybridDataReal__FDK, prodHybridDataImag__FDK, hTtoBox->pIcc__FDK))) { goto bail; } /* calculate correlation based Icc for downmix */ if (SACENC_OK != (error = calculateIccFDK( nParamBands, nParamBands, powerHybridData1__FDK, powerHybridData2__FDK, prodHybridDataReal__FDK, prodHybridDataImag__FDK, IccDownmix__FDK))) { goto bail; } } if (!bUseEbQ) { CalculateCldFDK(hTtoBox->pCld__FDK, powerHybridData1__FDK, powerHybridData2__FDK, *scaleCh1 + inScale + 1, pbScaleCh1, *scaleCh2 + inScale + 1, pbScaleCh2, nParamBands); } if (bUseBBCues) { ApplyBBCuesFDK(&hTtoBox->pCld__FDK[0], nParamBands); { ApplyBBCuesFDK(&hTtoBox->pIcc__FDK[0], nParamBands); } } /* bUseBBCues */ /* quantize/de-quantize icc */ { QuantizeCoefFDK(hTtoBox->pIcc__FDK, nParamBands, hTtoBox->pIccQuantTable__FDK, hTtoBox->nIccQuantOffset, hTtoBox->nIccQuantSteps, pIccIdx); QuantizeCoefFDK(IccDownmix__FDK, nParamBands, hTtoBox->pIccQuantTable__FDK, hTtoBox->nIccQuantOffset, hTtoBox->nIccQuantSteps, hTtoBox->pIccDownmixIdx); deQuantizeCoefFDK(hTtoBox->pIccDownmixIdx, nParamBands, hTtoBox->pIccQuantTable__FDK, hTtoBox->nIccQuantOffset, IccDownmixQuant__FDK); *pbIccQuantCoarse = hTtoBox->bUseCoarseQuantIcc; } /* quantize/de-quantize cld */ if (!bUseEbQ) { QuantizeCoefFDK(hTtoBox->pCld__FDK, nParamBands, hTtoBox->pCldQuantTableEnc__FDK, hTtoBox->nCldQuantOffset, hTtoBox->nCldQuantSteps, pCldIdx); deQuantizeCoefFDK(pCldIdx, nParamBands, hTtoBox->pCldQuantTableDec__FDK, hTtoBox->nCldQuantOffset, hTtoBox->pCldQuant__FDK); } else { FDKmemcpy(pCldIdx, hTtoBox->pCldEbQIdx, nParamBands * sizeof(SCHAR)); } *pbCldQuantCoarse = hTtoBox->bUseCoarseQuantCld; } /* valid handle */ bail: C_ALLOC_SCRATCH_END(pbScaleCh2, INT, MAX_NUM_PARAM_BANDS) C_ALLOC_SCRATCH_END(pbScaleCh1, INT, MAX_NUM_PARAM_BANDS) C_ALLOC_SCRATCH_END(IccDownmixQuant__FDK, FIXP_DBL, MAX_NUM_PARAM_BANDS) C_ALLOC_SCRATCH_END(IccDownmix__FDK, FIXP_DBL, MAX_NUM_PARAM_BANDS) C_ALLOC_SCRATCH_END(prodHybridDataImag__FDK, FIXP_DBL, MAX_NUM_PARAM_BANDS) C_ALLOC_SCRATCH_END(prodHybridDataReal__FDK, FIXP_DBL, MAX_NUM_PARAM_BANDS) C_ALLOC_SCRATCH_END(powerHybridData2__FDK, FIXP_DBL, MAX_NUM_PARAM_BANDS) C_ALLOC_SCRATCH_END(powerHybridData1__FDK, FIXP_DBL, MAX_NUM_PARAM_BANDS) return error; } INT fdk_sacenc_subband2ParamBand(const BOX_SUBBAND_CONFIG boxSubbandConfig, const INT nSubband) { INT nParamBand = -1; const UCHAR *pSubband2ParameterIndex = getSubband2ParameterIndex(boxSubbandConfig); if (pSubband2ParameterIndex != NULL) { const int hybrid_resolution = 64; if ((nSubband > -1) && (nSubband < hybrid_resolution)) { nParamBand = pSubband2ParameterIndex[nSubband]; } } return nParamBand; } fdk-aac-2.0.2/libSACenc/src/sacenc_paramextract.h000066400000000000000000000204131404225310700215310ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround encoder library ************************* Author(s): M. Multrus Description: Parameter Extraction *******************************************************************************/ #ifndef SACENC_PARAMEXTRACT_H #define SACENC_PARAMEXTRACT_H /* Includes ******************************************************************/ #include "common_fix.h" #include "sacenc_lib.h" #include "sacenc_const.h" #include "sacenc_bitstream.h" /* Defines *******************************************************************/ #define MAX_CLD_QUANT_FINE (31) #define MAX_CLD_QUANT_COARSE (15) #define OFFSET_CLD_QUANT_COARSE (7) #define OFFSET_CLD_QUANT_FINE (15) #define MAX_ICC_QUANT_COARSE (4) #define MAX_ICC_QUANT_FINE (8) #define OFFSET_ICC_QUANT_COARSE (0) #define OFFSET_ICC_QUANT_FINE (0) #define MAX_NUM_PARAM_BANDS (28) #define NUM_MAPPED_HYBRID_BANDS (16) /* Data Types ****************************************************************/ typedef struct T_TTO_BOX *HANDLE_TTO_BOX; typedef enum { BOX_SUBBANDS_INVALID = 0, BOX_SUBBANDS_4 = 4, BOX_SUBBANDS_5 = 5, BOX_SUBBANDS_7 = 7, BOX_SUBBANDS_9 = 9, BOX_SUBBANDS_12 = 12, BOX_SUBBANDS_15 = 15, BOX_SUBBANDS_23 = 23 } BOX_SUBBAND_CONFIG; typedef enum { BOX_QUANTMODE_INVALID = -1, BOX_QUANTMODE_FINE = 0, BOX_QUANTMODE_EBQ1 = 1, BOX_QUANTMODE_EBQ2 = 2, BOX_QUANTMODE_RESERVED3 = 3, BOX_QUANTMODE_RESERVED4 = 4, BOX_QUANTMODE_RESERVED5 = 5, BOX_QUANTMODE_RESERVED6 = 6, BOX_QUANTMODE_RESERVED7 = 7 } BOX_QUANTMODE; typedef struct T_TTO_BOX_CONFIG { UCHAR bUseCoarseQuantCld; UCHAR bUseCoarseQuantIcc; UCHAR bUseCoherenceIccOnly; BOX_SUBBAND_CONFIG subbandConfig; BOX_QUANTMODE boxQuantMode; UCHAR nHybridBandsMax; UCHAR bFrameKeep; } TTO_BOX_CONFIG; /* Constants *****************************************************************/ /* Function / Class Declarations *********************************************/ FDK_SACENC_ERROR fdk_sacenc_createTtoBox(HANDLE_TTO_BOX *hTtoBox); FDK_SACENC_ERROR fdk_sacenc_initTtoBox(HANDLE_TTO_BOX hTtoBox, const TTO_BOX_CONFIG *const ttoBoxConfig, UCHAR *pParameterBand2HybridBandOffset); FDK_SACENC_ERROR fdk_sacenc_destroyTtoBox(HANDLE_TTO_BOX *hTtoBox); FDK_SACENC_ERROR fdk_sacenc_applyTtoBox( HANDLE_TTO_BOX hTtoBox, const INT nTimeSlots, const INT startTimeSlot, const INT nHybridBands, const FIXP_DPK *const *const ppHybridData1__FDK, const FIXP_DPK *const *const ppHybridData2__FDK, SCHAR *const pIccIdx, UCHAR *const pbIccQuantCoarse, SCHAR *const pCldIdx, UCHAR *const pbCldQuantCoarse, const INT bUseBBCues, INT *scaleCh0, INT *scaleCh1); INT fdk_sacenc_subband2ParamBand(const BOX_SUBBAND_CONFIG boxSubbandConfig, const INT nSubband); const INT *fdk_sacenc_getSubbandImagSign(); void fdk_sacenc_calcParameterBand2HybridBandOffset( const BOX_SUBBAND_CONFIG subbandConfig, const INT nHybridBands, UCHAR *pParameterBand2HybridBandOffset); /* Function / Class Definition ***********************************************/ static inline UCHAR fdk_sacenc_getCldQuantOffset(const INT bUseCoarseQuant) { return ((bUseCoarseQuant) ? OFFSET_CLD_QUANT_COARSE : OFFSET_CLD_QUANT_FINE); } static inline UCHAR fdk_sacenc_getIccQuantOffset(const INT bUseCoarseQuant) { return ((bUseCoarseQuant) ? OFFSET_ICC_QUANT_COARSE : OFFSET_ICC_QUANT_FINE); } static inline UCHAR fdk_sacenc_getNumberCldQuantLevels( const INT bUseCoarseQuant) { return ((bUseCoarseQuant) ? MAX_CLD_QUANT_COARSE : MAX_CLD_QUANT_FINE); } static inline UCHAR fdk_sacenc_getNumberIccQuantLevels( const INT bUseCoarseQuant) { return ((bUseCoarseQuant) ? MAX_ICC_QUANT_COARSE : MAX_ICC_QUANT_FINE); } #endif /* SACENC_PARAMEXTRACT_H */ fdk-aac-2.0.2/libSACenc/src/sacenc_staticgain.cpp000066400000000000000000000377511404225310700215340ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround encoder library ************************* Author(s): Christian Goettlinger Description: Encoder Library Interface gain management of the encoder *******************************************************************************/ /***************************************************************************** \file This file contains all static gain infrastructure ******************************************************************************/ /* Includes ******************************************************************/ #include "sacenc_staticgain.h" /* Defines *******************************************************************/ #define MP4SPACEENC_DMX_GAIN_DEFAULT SACENC_DMXGAIN_3_dB #define GAINCF_SF (4) #define GAINCT1(x) FL2FXCONST_DBL(x) #define GAINCF(x) FL2FXCONST_DBL(x) #define GAINCT2(x) FL2FXCONST_DBL(x) #define FX_DBL2FX_GAIN(x) (x) /* Data Types ****************************************************************/ struct STATIC_GAIN { /* External Config Values */ MP4SPACEENC_MODE encMode; MP4SPACEENC_DMX_GAIN fixedGainDMX; INT preGainFactorDb; /* Internal Values */ FIXP_GAIN PostGain__FDK; FIXP_GAIN pPreGain__FDK[SACENC_MAX_INPUT_CHANNELS]; }; /* Constants *****************************************************************/ /* preGainFactorTable: pre calculation: (float)pow(10.f,(((float) x)/20.f))/(float)(1<encMode = SACENC_INVALID_MODE; /* Optional Configs Set to Default Values */ hStaticGainConfig->fixedGainDMX = MP4SPACEENC_DMX_GAIN_DEFAULT; hStaticGainConfig->preGainFactorDb = 0; } return error; } /*----------------------------------------------------------------------------- functionname: fdk_sacenc_staticGain_CloseConfig() description: destructs Static Gain Config Structure returns: noError on success, an apropriate error code else -----------------------------------------------------------------------------*/ FDK_SACENC_ERROR fdk_sacenc_staticGain_CloseConfig( HANDLE_STATIC_GAIN_CONFIG *phStaticGainConfig) { FDK_SACENC_ERROR error = SACENC_OK; if ((phStaticGainConfig == NULL) || (*phStaticGainConfig == NULL)) { error = SACENC_INVALID_HANDLE; } else { FDKfree(*phStaticGainConfig); *phStaticGainConfig = NULL; } return error; } /*----------------------------------------------------------------------------- functionname: fdk_sacenc_staticGain_Open() description: initializes Static Gains returns: noError on success, an apropriate error code else -----------------------------------------------------------------------------*/ FDK_SACENC_ERROR fdk_sacenc_staticGain_Open(HANDLE_STATIC_GAIN *phStaticGain) { FDK_SACENC_ERROR error = SACENC_OK; if (NULL == phStaticGain) { error = SACENC_INVALID_HANDLE; } else { /* Allocate Instance */ FDK_ALLOCATE_MEMORY_1D(*phStaticGain, 1, struct STATIC_GAIN); } return error; bail: return ((SACENC_OK == error) ? SACENC_MEMORY_ERROR : error); } FDK_SACENC_ERROR fdk_sacenc_staticGain_Init( HANDLE_STATIC_GAIN hStaticGain, const HANDLE_STATIC_GAIN_CONFIG hStaticGainConfig, INT *const scale) { FDK_SACENC_ERROR error = SACENC_OK; if ((hStaticGain == NULL) || (hStaticGainConfig == NULL)) { error = SACENC_INVALID_HANDLE; } else { hStaticGain->encMode = hStaticGainConfig->encMode; hStaticGain->fixedGainDMX = hStaticGainConfig->fixedGainDMX; hStaticGain->preGainFactorDb = hStaticGainConfig->preGainFactorDb; if ((hStaticGain->preGainFactorDb < -20) || (hStaticGain->preGainFactorDb > 20)) { error = SACENC_INVALID_CONFIG; goto bail; } FIXP_DBL fPreGainFactor__FDK; if (hStaticGain->preGainFactorDb == 0) { fPreGainFactor__FDK = (FIXP_DBL)MAXVAL_DBL; *scale = 0; } else { int s; fPreGainFactor__FDK = preGainFactorTable__FDK[hStaticGain->preGainFactorDb + 20]; s = fixMax(0, CntLeadingZeros(fPreGainFactor__FDK) - 1); fPreGainFactor__FDK = fPreGainFactor__FDK << (s); *scale = GAINCF_SF - s; } if (hStaticGain->fixedGainDMX == 0) hStaticGain->PostGain__FDK = MAXVAL_GAIN; else hStaticGain->PostGain__FDK = dmxGainTable__FDK[hStaticGain->fixedGainDMX - 1]; FDKmemclear( hStaticGain->pPreGain__FDK, sizeof(hStaticGain->pPreGain__FDK)); /* zero all input channels */ /* Configure PreGain-Vector */ if (hStaticGain->encMode == SACENC_212) { hStaticGain->pPreGain__FDK[0] = FX_DBL2FX_GAIN(fPreGainFactor__FDK); /* L */ hStaticGain->pPreGain__FDK[1] = FX_DBL2FX_GAIN(fPreGainFactor__FDK); /* R */ } else { error = SACENC_INVALID_CONFIG; } } /* valid handle */ bail: return error; } /*----------------------------------------------------------------------------- functionname: fdk_sacenc_staticGain_Close() description: destructs Static Gains returns: noError on success, an apropriate error code else -----------------------------------------------------------------------------*/ FDK_SACENC_ERROR fdk_sacenc_staticGain_Close(HANDLE_STATIC_GAIN *phStaticGain) { FDK_SACENC_ERROR error = SACENC_OK; if ((phStaticGain == NULL) || (*phStaticGain == NULL)) { error = SACENC_INVALID_HANDLE; } else { FDKfree(*phStaticGain); *phStaticGain = NULL; } return error; } /*----------------------------------------------------------------------------- functionname: fdk_sacenc_staticPostGain_Apply description: multiply the Output samples with the PostGain returns: noError on success, an apropriate error code else -----------------------------------------------------------------------------*/ FDK_SACENC_ERROR fdk_sacenc_staticPostGain_ApplyFDK( const HANDLE_STATIC_GAIN hStaticGain, INT_PCM *const pOutputSamples, const INT nOutputSamples, const INT scale) { FDK_SACENC_ERROR error = SACENC_OK; if (NULL == hStaticGain) { error = SACENC_INVALID_HANDLE; } else { int i; FIXP_GAIN postGain = hStaticGain->PostGain__FDK; if (scale < 0) { if (postGain == MAXVAL_GAIN) { for (i = 0; i < nOutputSamples; i++) { pOutputSamples[i] = pOutputSamples[i] >> (-scale); } } else { for (i = 0; i < nOutputSamples; i++) { pOutputSamples[i] = FX_DBL2FX_PCM( fMult(postGain, FX_PCM2FX_DBL(pOutputSamples[i])) >> (-scale)); } } } else { if (postGain == MAXVAL_GAIN) { for (i = 0; i < nOutputSamples; i++) { pOutputSamples[i] = FX_DBL2FX_PCM(SATURATE_LEFT_SHIFT( FX_PCM2FX_DBL(pOutputSamples[i]), scale, DFRACT_BITS)); } } else { for (i = 0; i < nOutputSamples; i++) { pOutputSamples[i] = FX_DBL2FX_PCM(SATURATE_LEFT_SHIFT( fMult(postGain, FX_PCM2FX_DBL(pOutputSamples[i])), scale, DFRACT_BITS)); } } } } return error; } /*----------------------------------------------------------------------------- functionname: fdk_sacenc_getPreGainPtr()/ fdk_sacenc_getPostGain() description: get Gain-Pointers from struct returns: Pointer to PreGain or postGain -----------------------------------------------------------------------------*/ FIXP_GAIN *fdk_sacenc_getPreGainPtrFDK(HANDLE_STATIC_GAIN hStaticGain) { return ((hStaticGain == NULL) ? NULL : hStaticGain->pPreGain__FDK); } FIXP_GAIN fdk_sacenc_getPostGainFDK(HANDLE_STATIC_GAIN hStaticGain) { return (hStaticGain->PostGain__FDK); } /* get fixed downmix gain and map it to bitstream enum */ FIXEDGAINDMXCONFIG fdk_sacenc_staticGain_GetDmxGain( const HANDLE_STATIC_GAIN hStaticGain) { FIXEDGAINDMXCONFIG dmxGain = FIXEDGAINDMX_INVALID; switch (hStaticGain->fixedGainDMX) { case 0: dmxGain = FIXEDGAINDMX_0; break; case 1: dmxGain = FIXEDGAINDMX_1; break; case 2: dmxGain = FIXEDGAINDMX_2; break; case 3: dmxGain = FIXEDGAINDMX_3; break; case 4: dmxGain = FIXEDGAINDMX_4; break; case 5: dmxGain = FIXEDGAINDMX_5; break; case 6: dmxGain = FIXEDGAINDMX_6; break; case 7: dmxGain = FIXEDGAINDMX_7; break; default: dmxGain = FIXEDGAINDMX_INVALID; } return dmxGain; } FDK_SACENC_ERROR fdk_sacenc_staticGain_SetDmxGain( HANDLE_STATIC_GAIN_CONFIG hStaticGainCfg, const MP4SPACEENC_DMX_GAIN dmxGain) { FDK_SACENC_ERROR error = SACENC_OK; if (NULL == hStaticGainCfg) { error = SACENC_INVALID_HANDLE; } else { hStaticGainCfg->fixedGainDMX = dmxGain; } return error; } FDK_SACENC_ERROR fdk_sacenc_staticGain_SetEncMode( HANDLE_STATIC_GAIN_CONFIG hStaticGainCfg, const MP4SPACEENC_MODE encMode) { FDK_SACENC_ERROR error = SACENC_OK; if (NULL == hStaticGainCfg) { error = SACENC_INVALID_HANDLE; } else { hStaticGainCfg->encMode = encMode; } return error; } fdk-aac-2.0.2/libSACenc/src/sacenc_staticgain.h000066400000000000000000000167321404225310700211750ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround encoder library ************************* Author(s): Christian Goettlinger Description: Encoder Library Interfac gain management of the encoder *******************************************************************************/ /**************************************************************************/ /** \file ******************************************************************************/ #ifndef SACENC_STATICGAIN_H #define SACENC_STATICGAIN_H /* Includes ******************************************************************/ #include "common_fix.h" #include "sacenc_lib.h" #include "sacenc_const.h" #include "sacenc_bitstream.h" /* Defines *******************************************************************/ #define FIXP_GAIN FIXP_DBL #define MAXVAL_GAIN ((FIXP_DBL)MAXVAL_DBL) /* Data Types ****************************************************************/ struct STATIC_GAIN_CONFIG { MP4SPACEENC_MODE encMode; MP4SPACEENC_DMX_GAIN fixedGainDMX; INT preGainFactorDb; }; typedef struct STATIC_GAIN_CONFIG *HANDLE_STATIC_GAIN_CONFIG; typedef struct STATIC_GAIN *HANDLE_STATIC_GAIN; /* Constants *****************************************************************/ /* Function / Class Declarations *********************************************/ /* Initializes Static Gain Computation Config */ FDK_SACENC_ERROR fdk_sacenc_staticGain_OpenConfig( HANDLE_STATIC_GAIN_CONFIG *phStaticGainConfig); FDK_SACENC_ERROR fdk_sacenc_staticGain_InitDefaultConfig( HANDLE_STATIC_GAIN_CONFIG hStaticGainConfig); /* Deletes Static Gain Computation Config ~Destructor */ FDK_SACENC_ERROR fdk_sacenc_staticGain_CloseConfig( HANDLE_STATIC_GAIN_CONFIG *phStaticGainConfig); /* Initializes Static Gain Computation ~Constructor */ FDK_SACENC_ERROR fdk_sacenc_staticGain_Open(HANDLE_STATIC_GAIN *phStaticGain); FDK_SACENC_ERROR fdk_sacenc_staticGain_Init( HANDLE_STATIC_GAIN hStaticGain, const HANDLE_STATIC_GAIN_CONFIG hStaticGainConfig, INT *const scale); /* Deletes Static Gain Computation Infrastucture ~Destructor */ FDK_SACENC_ERROR fdk_sacenc_staticGain_Close(HANDLE_STATIC_GAIN *phStaticGain); /* Apply PostGain to the output PCM Downmix-Signal */ FDK_SACENC_ERROR fdk_sacenc_staticPostGain_ApplyFDK( const HANDLE_STATIC_GAIN hStaticGain, INT_PCM *const pOutputSamples, const INT nOutputSamples, const INT scale); /* Get Pointer to PreGain-vector */ FIXP_GAIN *fdk_sacenc_getPreGainPtrFDK(HANDLE_STATIC_GAIN hStaticGain); /* Get Pointer to PostGain-coef */ FIXP_GAIN fdk_sacenc_getPostGainFDK(HANDLE_STATIC_GAIN hStaticGain); FIXEDGAINDMXCONFIG fdk_sacenc_staticGain_GetDmxGain( const HANDLE_STATIC_GAIN hStaticGain); FDK_SACENC_ERROR fdk_sacenc_staticGain_SetDmxGain( HANDLE_STATIC_GAIN_CONFIG hStaticGainCfg, const MP4SPACEENC_DMX_GAIN dmxGain); FDK_SACENC_ERROR fdk_sacenc_staticGain_SetEncMode( HANDLE_STATIC_GAIN_CONFIG hStaticGainCfg, const MP4SPACEENC_MODE encMode); #endif /* SACENC_STATICGAIN_H */ fdk-aac-2.0.2/libSACenc/src/sacenc_tree.cpp000066400000000000000000000377751404225310700203530ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround encoder library ************************* Author(s): Max Neuendorf Description: Encoder Library Interface Tree Structure for Space Encoder *******************************************************************************/ /* Includes ******************************************************************/ #include "sacenc_tree.h" #include "genericStds.h" #include "sacenc_const.h" #include "sacenc_paramextract.h" #include "sacenc_framewindowing.h" #include "FDK_matrixCalloc.h" /* Defines *******************************************************************/ enum { BOX_0 = 0, BOX_1 = 1 }; enum { CH_L = 0, CH_R = 1 }; enum { TTO_CH_0 = 0, TTO_CH_1 = 1 }; enum { WIN_INACTIV = 0, WIN_ACTIV = 1 }; enum { MAX_KEEP_FRAMECOUNT = 100 }; /* Data Types ****************************************************************/ struct SPACE_TREE { SPACETREE_MODE mode; SPACE_TREE_DESCRIPTION descr; HANDLE_TTO_BOX ttoBox[SACENC_MAX_NUM_BOXES]; UCHAR nParamBands; UCHAR bUseCoarseQuantTtoIcc; UCHAR bUseCoarseQuantTtoCld; QUANTMODE quantMode; INT frameCount; UCHAR bFrameKeep; /* Intermediate buffers */ UCHAR pCld_prev[SACENC_MAX_NUM_BOXES][MAX_NUM_PARAM_BANDS]; UCHAR pIcc_prev[SACENC_MAX_NUM_BOXES][MAX_NUM_PARAM_BANDS]; UCHAR nChannelsInMax; UCHAR nHybridBandsMax; }; typedef struct { UCHAR boxId; UCHAR inCh1; UCHAR inCh2; UCHAR inCh3; UCHAR inCh4; UCHAR wCh1; UCHAR wCh2; } TTO_DESCRIPTOR; typedef struct { SPACETREE_MODE mode; SPACE_TREE_DESCRIPTION treeDescription; } TREE_CONFIG; typedef struct { SPACETREE_MODE mode; UCHAR nChannelsIn; UCHAR nChannelsOut; UCHAR nTtoBoxes; TTO_DESCRIPTOR tto_descriptor[1]; } TREE_SETUP; /* Constants *****************************************************************/ static const TREE_CONFIG treeConfigTable[] = { {SPACETREE_INVALID_MODE, {0, 0, 0}}, {SPACETREE_212, {1, 1, 2}}}; static const TREE_SETUP treeSetupTable[] = { {SPACETREE_INVALID_MODE, 0, 0, 0, {{0, 0, 0, 0, 0, 0, 0}}}, {SPACETREE_212, 2, 1, 1, {{BOX_0, CH_L, CH_R, TTO_CH_0, TTO_CH_1, WIN_ACTIV, WIN_ACTIV}}}}; /* Function / Class Declarations *********************************************/ /* Function / Class Definition ***********************************************/ static FDK_SACENC_ERROR getTreeConfig( const SPACETREE_MODE mode, SPACE_TREE_DESCRIPTION *pTreeDescription) { FDK_SACENC_ERROR error = SACENC_INIT_ERROR; if (pTreeDescription == NULL) { error = SACENC_INVALID_HANDLE; } else { int i; for (i = 0; i < (int)(sizeof(treeConfigTable) / sizeof(TREE_CONFIG)); i++) { if (treeConfigTable[i].mode == mode) { *pTreeDescription = treeConfigTable[i].treeDescription; error = SACENC_OK; break; } } } /* valid handle */ return error; } static const TREE_SETUP *getTreeSetup(const SPACETREE_MODE mode) { int i; const TREE_SETUP *setup = NULL; for (i = 0; i < (int)(sizeof(treeSetupTable) / sizeof(TREE_SETUP)); i++) { if (treeSetupTable[i].mode == mode) { setup = &treeSetupTable[i]; break; } } return setup; } FDK_SACENC_ERROR fdk_sacenc_spaceTree_Open(HANDLE_SPACE_TREE *phSpaceTree) { FDK_SACENC_ERROR error = SACENC_OK; HANDLE_SPACE_TREE hSpaceTree = NULL; if (NULL == phSpaceTree) { error = SACENC_INVALID_HANDLE; } else { int box; FDK_ALLOCATE_MEMORY_1D(hSpaceTree, 1, struct SPACE_TREE); for (box = 0; box < SACENC_MAX_NUM_BOXES; box++) { HANDLE_TTO_BOX ttoBox = NULL; if (SACENC_OK != (error = fdk_sacenc_createTtoBox(&ttoBox))) { goto bail; } if (NULL != hSpaceTree) { hSpaceTree->ttoBox[box] = ttoBox; } } *phSpaceTree = hSpaceTree; } return error; bail: fdk_sacenc_spaceTree_Close(&hSpaceTree); return ((SACENC_OK == error) ? SACENC_MEMORY_ERROR : error); } FDK_SACENC_ERROR fdk_sacenc_spaceTree_Init( HANDLE_SPACE_TREE hST, const SPACE_TREE_SETUP *const hSetup, UCHAR *pParameterBand2HybridBandOffset, const INT bFrameKeep) { FDK_SACENC_ERROR error = SACENC_OK; if ((hST == NULL) || (hSetup == NULL)) { error = SACENC_INVALID_HANDLE; } else { int bTtoBoxFrontBackCombin[SACENC_MAX_NUM_BOXES] = {0}; int box = 0; hST->frameCount = 0; hST->bFrameKeep = bFrameKeep; /* Init */ hST->mode = hSetup->mode; hST->nParamBands = hSetup->nParamBands; hST->bUseCoarseQuantTtoIcc = hSetup->bUseCoarseQuantTtoIcc; hST->bUseCoarseQuantTtoCld = hSetup->bUseCoarseQuantTtoCld; hST->quantMode = hSetup->quantMode; hST->nChannelsInMax = hSetup->nChannelsInMax; hST->nHybridBandsMax = hSetup->nHybridBandsMax; if (SACENC_OK != (error = getTreeConfig(hST->mode, &hST->descr))) { goto bail; } switch (hST->mode) { case SPACETREE_212: bTtoBoxFrontBackCombin[BOX_0] = 0; break; case SPACETREE_INVALID_MODE: default: error = SACENC_INIT_ERROR; goto bail; } /* switch (hST->mode) */ if (hST->descr.nOttBoxes > SACENC_MAX_NUM_BOXES) { error = SACENC_INIT_ERROR; goto bail; } for (box = 0; box < hST->descr.nOttBoxes; box++) { TTO_BOX_CONFIG boxConfig; boxConfig.subbandConfig = (BOX_SUBBAND_CONFIG)hST->nParamBands; boxConfig.bUseCoarseQuantCld = hST->bUseCoarseQuantTtoCld; boxConfig.bUseCoarseQuantIcc = hST->bUseCoarseQuantTtoIcc; boxConfig.bUseCoherenceIccOnly = bTtoBoxFrontBackCombin[box]; boxConfig.boxQuantMode = (BOX_QUANTMODE)hST->quantMode; boxConfig.nHybridBandsMax = hST->nHybridBandsMax; boxConfig.bFrameKeep = hST->bFrameKeep; if (SACENC_OK != (error = fdk_sacenc_initTtoBox(hST->ttoBox[box], &boxConfig, pParameterBand2HybridBandOffset))) { goto bail; } } /* for box */ } /* valid handle */ bail: return error; } static void SpaceTree_FrameKeep212(const HANDLE_SPACE_TREE hST, SPATIALFRAME *const hSTOut, const INT avoid_keep) { int pb; if (avoid_keep == 0) { if (hST->frameCount % 2 == 0) { for (pb = 0; pb < hST->nParamBands; pb++) { hST->pIcc_prev[BOX_0][pb] = hSTOut->ottData.icc[BOX_0][0][pb]; hSTOut->ottData.cld[BOX_0][0][pb] = hST->pCld_prev[BOX_0][pb]; } } else { for (pb = 0; pb < hST->nParamBands; pb++) { hSTOut->ottData.icc[BOX_0][0][pb] = hST->pIcc_prev[BOX_0][pb]; hST->pCld_prev[BOX_0][pb] = hSTOut->ottData.cld[BOX_0][0][pb]; } } } else { for (pb = 0; pb < hST->nParamBands; pb++) { hST->pIcc_prev[BOX_0][pb] = hSTOut->ottData.icc[BOX_0][0][pb]; hST->pCld_prev[BOX_0][pb] = hSTOut->ottData.cld[BOX_0][0][pb]; } } hST->frameCount++; if (hST->frameCount == MAX_KEEP_FRAMECOUNT) { hST->frameCount = 0; } } static FDK_SACENC_ERROR SpaceTree_FrameKeep(const HANDLE_SPACE_TREE hST, SPATIALFRAME *const hSTOut, const INT avoid_keep) { FDK_SACENC_ERROR error = SACENC_OK; switch (hST->mode) { case SPACETREE_212: SpaceTree_FrameKeep212(hST, hSTOut, avoid_keep); break; case SPACETREE_INVALID_MODE: default: error = SACENC_INVALID_CONFIG; break; } return error; } FDK_SACENC_ERROR fdk_sacenc_spaceTree_Apply( HANDLE_SPACE_TREE hST, const INT paramSet, const INT nChannelsIn, const INT nTimeSlots, const INT startTimeSlot, const INT nHybridBands, FIXP_WIN *pFrameWindowAna__FDK, FIXP_DPK *const *const *const pppHybrid__FDK, FIXP_DPK *const *const *const pppHybridIn__FDK, SPATIALFRAME *const hSTOut, const INT avoid_keep, INT *pEncoderInputChScale) { /** \verbatim ============================================================================================================================= TREE_212 ============================================================================================================================= _______ L -- TTO_CH_0 --| | | TTO_0 |-- TTO_CH_0 R -- TTO_CH_1 --|_______| \endverbatim */ FDK_SACENC_ERROR error = SACENC_OK; int k; const TREE_SETUP *treeSetup = NULL; if ((hST == NULL) || (hSTOut == NULL) || (pppHybrid__FDK == NULL) || (pppHybridIn__FDK == NULL)) { error = SACENC_INVALID_HANDLE; goto bail; } if ((treeSetup = getTreeSetup(hST->mode)) == NULL) { error = SACENC_INVALID_CONFIG; goto bail; } /* Sanity Checks */ if ((nChannelsIn != treeSetup->nChannelsIn) || (nChannelsIn > hST->nChannelsInMax) || (nHybridBands > hST->nHybridBandsMax)) { error = SACENC_INVALID_CONFIG; goto bail; } /* Apply all TTO boxes. */ for (k = 0; k < treeSetup->nTtoBoxes; k++) { const TTO_DESCRIPTOR *pTTO = &treeSetup->tto_descriptor[k]; int i, inCh[2], outCh[2], win[2]; inCh[0] = pTTO->inCh1; outCh[0] = pTTO->inCh3; win[0] = pTTO->wCh1; inCh[1] = pTTO->inCh2; outCh[1] = pTTO->inCh4; win[1] = pTTO->wCh2; for (i = 0; i < 2; i++) { if (win[i] == WIN_ACTIV) { fdk_sacenc_analysisWindowing( nTimeSlots, startTimeSlot, pFrameWindowAna__FDK, pppHybrid__FDK[inCh[i]], pppHybridIn__FDK[outCh[i]], nHybridBands, FW_LEAVE_DIM); } } /* Calculate output downmix within last TTO box, if no TTT box is applied. */ if (SACENC_OK != (error = fdk_sacenc_applyTtoBox( hST->ttoBox[pTTO->boxId], nTimeSlots, startTimeSlot, nHybridBands, pppHybridIn__FDK[pTTO->inCh3], pppHybridIn__FDK[pTTO->inCh4], hSTOut->ottData.icc[pTTO->boxId][paramSet], &(hSTOut->ICCLosslessData.bsQuantCoarseXXX[pTTO->boxId][paramSet]), hSTOut->ottData.cld[pTTO->boxId][paramSet], &(hSTOut->CLDLosslessData.bsQuantCoarseXXX[pTTO->boxId][paramSet]), hSTOut->bUseBBCues, &pEncoderInputChScale[inCh[0]], &pEncoderInputChScale[inCh[1]]))) { goto bail; } } if (hST->bFrameKeep == 1) { if (SACENC_OK != (error = SpaceTree_FrameKeep(hST, hSTOut, avoid_keep))) { goto bail; } } bail: return error; } FDK_SACENC_ERROR fdk_sacenc_spaceTree_Close(HANDLE_SPACE_TREE *phSpaceTree) { FDK_SACENC_ERROR error = SACENC_OK; if ((phSpaceTree == NULL) || (*phSpaceTree == NULL)) { error = SACENC_INVALID_HANDLE; } else { int box; HANDLE_SPACE_TREE const hST = *phSpaceTree; /* for (box = 0; box < hST->descr.nOttBoxes; ++box) { */ for (box = 0; box < SACENC_MAX_NUM_BOXES; ++box) { if (SACENC_OK != (error = fdk_sacenc_destroyTtoBox(&hST->ttoBox[box]))) { goto bail; } } FDKfree(*phSpaceTree); *phSpaceTree = NULL; } bail: return error; } FDK_SACENC_ERROR fdk_sacenc_spaceTree_GetDescription( const HANDLE_SPACE_TREE hSpaceTree, SPACE_TREE_DESCRIPTION *pSpaceTreeDescription) { FDK_SACENC_ERROR error = SACENC_OK; if ((hSpaceTree == NULL) || (pSpaceTreeDescription == NULL)) { error = SACENC_INVALID_HANDLE; } else { *pSpaceTreeDescription = hSpaceTree->descr; } return error; } INT fdk_sacenc_spaceTree_Hybrid2ParamBand(const INT nParamBands, const INT nHybridBand) { return fdk_sacenc_subband2ParamBand((BOX_SUBBAND_CONFIG)nParamBands, nHybridBand); } /***************************************************************************** ******************************************************************************/ fdk-aac-2.0.2/libSACenc/src/sacenc_tree.h000066400000000000000000000154331404225310700200030ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround encoder library ************************* Author(s): Max Neuendorf Description: Encoder Library Interface Tree Structure for Space Encoder *******************************************************************************/ #ifndef SACENC_TREE_H #define SACENC_TREE_H /* Includes ******************************************************************/ #include "sacenc_framewindowing.h" #include "sacenc_lib.h" #include "sacenc_bitstream.h" /* Defines *******************************************************************/ /* Data Types ****************************************************************/ typedef enum { SPACETREE_INVALID_MODE = 0, SPACETREE_212 = 8 } SPACETREE_MODE; typedef struct SPACE_TREE *HANDLE_SPACE_TREE; typedef struct { UCHAR nParamBands; UCHAR bUseCoarseQuantTtoCld; UCHAR bUseCoarseQuantTtoIcc; QUANTMODE quantMode; SPACETREE_MODE mode; UCHAR nChannelsInMax; UCHAR nHybridBandsMax; } SPACE_TREE_SETUP; typedef struct { UCHAR nOttBoxes; UCHAR nInChannels; UCHAR nOutChannels; } SPACE_TREE_DESCRIPTION; /* Constants *****************************************************************/ /* Function / Class Declarations *********************************************/ FDK_SACENC_ERROR fdk_sacenc_spaceTree_Open(HANDLE_SPACE_TREE *phSpaceTree); FDK_SACENC_ERROR fdk_sacenc_spaceTree_Init( HANDLE_SPACE_TREE hST, const SPACE_TREE_SETUP *const hSetup, UCHAR *pParameterBand2HybridBandOffset, const INT bFrameKeep); FDK_SACENC_ERROR fdk_sacenc_spaceTree_Apply( HANDLE_SPACE_TREE hST, const INT paramSet, const INT nChannelsIn, const INT nTimeSlots, const INT startTimeSlot, const INT nHybridBands, FIXP_WIN *pFrameWindowAna__FDK, FIXP_DPK *const *const *const pppHybrid__FDK, FIXP_DPK *const *const *const pppHybridIn__FDK, SPATIALFRAME *const hSTOut, const INT avoid_keep, INT *pEncoderInputChScale); FDK_SACENC_ERROR fdk_sacenc_spaceTree_Close(HANDLE_SPACE_TREE *phSpaceTree); FDK_SACENC_ERROR fdk_sacenc_spaceTree_GetDescription( const HANDLE_SPACE_TREE hSpaceTree, SPACE_TREE_DESCRIPTION *pSpaceTreeDescription); INT fdk_sacenc_spaceTree_Hybrid2ParamBand(const INT nParamBands, const INT nHybridBand); #endif /* SACENC_TREE_H */ fdk-aac-2.0.2/libSACenc/src/sacenc_vectorfunctions.cpp000066400000000000000000000364071404225310700226360ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround encoder library ************************* Author(s): Josef Hoepfl Description: Encoder Library Interface vector functions *******************************************************************************/ /***************************************************************************** \file This file contains vector functions ******************************************************************************/ /* Includes ******************************************************************/ #include "sacenc_vectorfunctions.h" /* Defines *******************************************************************/ /* Data Types ****************************************************************/ /* Constants *****************************************************************/ /* Function / Class Declarations *********************************************/ /* Function / Class Definition ***********************************************/ FIXP_DBL sumUpCplxPow2(const FIXP_DPK *const x, const INT scaleMode, const INT inScaleFactor, INT *const outScaleFactor, const INT n) { int i, cs; if (scaleMode == SUM_UP_DYNAMIC_SCALE) { /* calculate headroom */ FIXP_DBL maxVal = FL2FXCONST_DBL(0.0f); for (i = 0; i < n; i++) { maxVal |= fAbs(x[i].v.re); maxVal |= fAbs(x[i].v.im); } cs = inScaleFactor - fixMax(0, CntLeadingZeros(maxVal) - 1); } else { cs = inScaleFactor; } /* consider scaling of energy and scaling in fPow2Div2 and addition */ *outScaleFactor = 2 * cs + 2; /* make sure that the scalefactor is in the range of -(DFRACT_BITS-1), ... , * (DFRACT_BITS-1) */ cs = fixMax(fixMin(cs, DFRACT_BITS - 1), -(DFRACT_BITS - 1)); /* sum up complex energy samples */ FIXP_DBL re, im, sum; re = im = sum = FL2FXCONST_DBL(0.0); if (cs < 0) { cs = -cs; for (i = 0; i < n; i++) { re += fPow2Div2(x[i].v.re << cs); im += fPow2Div2(x[i].v.im << cs); } } else { cs = 2 * cs; for (i = 0; i < n; i++) { re += fPow2Div2(x[i].v.re) >> cs; im += fPow2Div2(x[i].v.im) >> cs; } } sum = (re >> 1) + (im >> 1); return (sum); } FIXP_DBL sumUpCplxPow2Dim2(const FIXP_DPK *const *const x, const INT scaleMode, const INT inScaleFactor, INT *const outScaleFactor, const INT sDim1, const INT nDim1, const INT sDim2, const INT nDim2) { int i, j, cs; if (scaleMode == SUM_UP_DYNAMIC_SCALE) { /* calculate headroom */ FIXP_DBL maxVal = FL2FXCONST_DBL(0.0f); for (i = sDim1; i < nDim1; i++) { for (j = sDim2; j < nDim2; j++) { maxVal |= fAbs(x[i][j].v.re); maxVal |= fAbs(x[i][j].v.im); } } cs = inScaleFactor - fixMax(0, CntLeadingZeros(maxVal) - 1); } else { cs = inScaleFactor; } /* consider scaling of energy and scaling in fPow2Div2 and addition */ *outScaleFactor = 2 * cs + 2; /* make sure that the scalefactor is in the range of -(DFRACT_BITS-1), ... , * (DFRACT_BITS-1) */ cs = fixMax(fixMin(cs, DFRACT_BITS - 1), -(DFRACT_BITS - 1)); /* sum up complex energy samples */ FIXP_DBL re, im, sum; re = im = sum = FL2FXCONST_DBL(0.0); if (cs < 0) { cs = -cs; for (i = sDim1; i < nDim1; i++) { for (j = sDim2; j < nDim2; j++) { re += fPow2Div2(x[i][j].v.re << cs); im += fPow2Div2(x[i][j].v.im << cs); } } } else { cs = 2 * cs; for (i = sDim1; i < nDim1; i++) { for (j = sDim2; j < nDim2; j++) { re += fPow2Div2(x[i][j].v.re) >> cs; im += fPow2Div2(x[i][j].v.im) >> cs; } } } sum = (re >> 1) + (im >> 1); return (sum); } void copyCplxVec(FIXP_DPK *const Z, const FIXP_DPK *const X, const INT n) { FDKmemmove(Z, X, sizeof(FIXP_DPK) * n); } void setCplxVec(FIXP_DPK *const Z, const FIXP_DBL a, const INT n) { int i; for (i = 0; i < n; i++) { Z[i].v.re = a; Z[i].v.im = a; } } void cplx_cplxScalarProduct(FIXP_DPK *const Z, const FIXP_DPK *const *const X, const FIXP_DPK *const *const Y, const INT scaleX, const INT scaleY, INT *const scaleZ, const INT sDim1, const INT nDim1, const INT sDim2, const INT nDim2) { int i, j, sx, sy; FIXP_DBL xre, yre, xim, yim, re, im; /* make sure that the scalefactor is in the range of -(DFRACT_BITS-1), ... , * (DFRACT_BITS-1) */ sx = fixMax(fixMin(scaleX, DFRACT_BITS - 1), -(DFRACT_BITS - 1)); sy = fixMax(fixMin(scaleY, DFRACT_BITS - 1), -(DFRACT_BITS - 1)); /* consider scaling of energy and scaling in fMultDiv2 and shift of result * values */ *scaleZ = sx + sy + 2; re = (FIXP_DBL)0; im = (FIXP_DBL)0; if ((sx < 0) && (sy < 0)) { sx = -sx; sy = -sy; for (i = sDim1; i < nDim1; i++) { for (j = sDim2; j < nDim2; j++) { xre = X[i][j].v.re << sx; xim = X[i][j].v.im << sx; yre = Y[i][j].v.re << sy; yim = Y[i][j].v.im << sy; re += fMultDiv2(xre, yre) + fMultDiv2(xim, yim); im += fMultDiv2(xim, yre) - fMultDiv2(xre, yim); } } } else if ((sx >= 0) && (sy >= 0)) { for (i = sDim1; i < nDim1; i++) { for (j = sDim2; j < nDim2; j++) { xre = X[i][j].v.re; xim = X[i][j].v.im; yre = Y[i][j].v.re; yim = Y[i][j].v.im; re += (fMultDiv2(xre, yre) + fMultDiv2(xim, yim)) >> (sx + sy); im += (fMultDiv2(xim, yre) - fMultDiv2(xre, yim)) >> (sx + sy); } } } else if ((sx < 0) && (sy >= 0)) { sx = -sx; for (i = sDim1; i < nDim1; i++) { for (j = sDim2; j < nDim2; j++) { xre = X[i][j].v.re << sx; xim = X[i][j].v.im << sx; yre = Y[i][j].v.re; yim = Y[i][j].v.im; re += (fMultDiv2(xre, yre) + fMultDiv2(xim, yim)) >> sy; im += (fMultDiv2(xim, yre) - fMultDiv2(xre, yim)) >> sy; } } } else { sy = -sy; for (i = sDim1; i < nDim1; i++) { for (j = sDim2; j < nDim2; j++) { xre = X[i][j].v.re; xim = X[i][j].v.im; yre = Y[i][j].v.re << sy; yim = Y[i][j].v.im << sy; re += (fMultDiv2(xre, yre) + fMultDiv2(xim, yim)) >> sx; im += (fMultDiv2(xim, yre) - fMultDiv2(xre, yim)) >> sx; } } } Z->v.re = re >> 1; Z->v.im = im >> 1; } void FDKcalcCorrelationVec(FIXP_DBL *const z, const FIXP_DBL *const pr12, const FIXP_DBL *const p1, const FIXP_DBL *const p2, const INT n) { int i, s; FIXP_DBL p12, cor; /* correlation */ for (i = 0; i < n; i++) { p12 = fMult(p1[i], p2[i]); if (p12 > FL2FXCONST_DBL(0.0f)) { p12 = invSqrtNorm2(p12, &s); cor = fMult(pr12[i], p12); z[i] = SATURATE_LEFT_SHIFT(cor, s, DFRACT_BITS); } else { z[i] = (FIXP_DBL)MAXVAL_DBL; } } } void calcCoherenceVec(FIXP_DBL *const z, const FIXP_DBL *const p12r, const FIXP_DBL *const p12i, const FIXP_DBL *const p1, const FIXP_DBL *const p2, const INT scaleP12, const INT scaleP, const INT n) { int i, s, s1, s2; FIXP_DBL coh, p12, p12ri; for (i = 0; i < n; i++) { s2 = fixMin(fixMax(0, CountLeadingBits(p12r[i]) - 1), fixMax(0, CountLeadingBits(p12i[i]) - 1)); p12ri = sqrtFixp(fPow2Div2(p12r[i] << s2) + fPow2Div2(p12i[i] << s2)); s1 = fixMin(fixMax(0, CountLeadingBits(p1[i]) - 1), fixMax(0, CountLeadingBits(p2[i]) - 1)); p12 = fMultDiv2(p1[i] << s1, p2[i] << s1); if (p12 > FL2FXCONST_DBL(0.0f)) { p12 = invSqrtNorm2(p12, &s); coh = fMult(p12ri, p12); s = fixMax(fixMin((scaleP12 - scaleP + s + s1 - s2), DFRACT_BITS - 1), -(DFRACT_BITS - 1)); if (s < 0) { z[i] = coh >> (-s); } else { z[i] = SATURATE_LEFT_SHIFT(coh, s, DFRACT_BITS); } } else { z[i] = (FIXP_DBL)MAXVAL_DBL; } } } void addWeightedCplxVec(FIXP_DPK *const *const Z, const FIXP_DBL *const a, const FIXP_DPK *const *const X, const FIXP_DBL *const b, const FIXP_DPK *const *const Y, const INT scale, INT *const scaleCh1, const INT scaleCh2, const UCHAR *const pParameterBand2HybridBandOffset, const INT nParameterBands, const INT nTimeSlots, const INT startTimeSlot) { int pb, j, i; int cs, s1, s2; /* determine maximum scale of both channels */ cs = fixMax(*scaleCh1, scaleCh2); s1 = cs - (*scaleCh1); s2 = cs - scaleCh2; /* scalefactor 1 is updated with common scale of channel 1 and channel2 */ *scaleCh1 = cs; /* scale of a and b; additional scale for fMultDiv2() */ for (j = 0, pb = 0; pb < nParameterBands; pb++) { FIXP_DBL aPb, bPb; aPb = a[pb], bPb = b[pb]; for (; j < pParameterBand2HybridBandOffset[pb]; j++) { for (i = startTimeSlot; i < nTimeSlots; i++) { Z[j][i].v.re = ((fMultDiv2(aPb, X[j][i].v.re) >> s1) + (fMultDiv2(bPb, Y[j][i].v.re) >> s2)) << (scale + 1); Z[j][i].v.im = ((fMultDiv2(aPb, X[j][i].v.im) >> s1) + (fMultDiv2(bPb, Y[j][i].v.im) >> s2)) << (scale + 1); } } } } void FDKcalcPbScaleFactor(const FIXP_DPK *const *const x, const UCHAR *const pParameterBand2HybridBandOffset, INT *const outScaleFactor, const INT startTimeSlot, const INT nTimeSlots, const INT nParamBands) { int i, j, pb; /* calculate headroom */ for (j = 0, pb = 0; pb < nParamBands; pb++) { FIXP_DBL maxVal = FL2FXCONST_DBL(0.0f); for (; j < pParameterBand2HybridBandOffset[pb]; j++) { for (i = startTimeSlot; i < nTimeSlots; i++) { maxVal |= fAbs(x[i][j].v.re); maxVal |= fAbs(x[i][j].v.im); } } outScaleFactor[pb] = -fixMax(0, CntLeadingZeros(maxVal) - 1); } } INT FDKcalcScaleFactor(const FIXP_DBL *const x, const FIXP_DBL *const y, const INT n) { int i; /* calculate headroom */ FIXP_DBL maxVal = FL2FXCONST_DBL(0.0f); if (x != NULL) { for (i = 0; i < n; i++) { maxVal |= fAbs(x[i]); } } if (y != NULL) { for (i = 0; i < n; i++) { maxVal |= fAbs(y[i]); } } if (maxVal == (FIXP_DBL)0) return (-(DFRACT_BITS - 1)); else return (-CountLeadingBits(maxVal)); } INT FDKcalcScaleFactorDPK(const FIXP_DPK *RESTRICT x, const INT startBand, const INT bands) { INT qs, clz; FIXP_DBL maxVal = FL2FXCONST_DBL(0.0f); for (qs = startBand; qs < bands; qs++) { maxVal |= fAbs(x[qs].v.re); maxVal |= fAbs(x[qs].v.im); } clz = -fixMax(0, CntLeadingZeros(maxVal) - 1); return (clz); } fdk-aac-2.0.2/libSACenc/src/sacenc_vectorfunctions.h000066400000000000000000000426231404225310700223000ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /*********************** MPEG surround encoder library ************************* Author(s): Josef Hoepfl Description: Encoder Library Interface vector functions *******************************************************************************/ /***************************************************************************** \file This file contains vector functions ******************************************************************************/ #ifndef SACENC_VECTORFUNCTIONS_H #define SACENC_VECTORFUNCTIONS_H /* Includes ******************************************************************/ #include "common_fix.h" /* Defines *******************************************************************/ #define SUM_UP_STATIC_SCALE 0 #define SUM_UP_DYNAMIC_SCALE 1 /* Data Types ****************************************************************/ /* Constants *****************************************************************/ /* Function / Class Declarations *********************************************/ /** * \brief Vector function : Sum up complex power * * Description : ret = sum( re{X[i]} * re{X[i]} + im{X[i]} * * im{X[i]} ), i=0,...,n-1 ret is scaled by outScaleFactor * * \param const FIXP_DPK x[] * Input: complex vector of the length n * * \param int scaleMode * Input: choose static or dynamic scaling * (SUM_UP_DYNAMIC_SCALE/SUM_UP_STATIC_SCALE) * * \param int inScaleFactor * Input: determine headroom bits for the complex input vector * * \param int outScaleFactor * Output: complete scaling in energy calculation * * \return FIXP_DBL ret */ FIXP_DBL sumUpCplxPow2(const FIXP_DPK *const x, const INT scaleMode, const INT inScaleFactor, INT *const outScaleFactor, const INT n); /** * \brief Vector function : Sum up complex power * * Description : ret = sum( re{X[i][j]} * re{X[i][]} + * im{X[i][]} * im{X[i][]} ), i=sDim1,...,nDim1-1 i=sDim2,...,nDim2-1 ret is * scaled by outScaleFactor * * \param const FIXP_DPK x[] * Input: complex vector of the length n * * \param int scaleMode * Input: choose static or dynamic scaling * (SUM_UP_DYNAMIC_SCALE/SUM_UP_STATIC_SCALE) * * \param int inScaleFactor * Input: determine headroom bits for the complex input vector * * \param int outScaleFactor * Output: complete scaling in energy calculation * * \param int sDim1 * Input: start index for loop counter in dimension 1 * * \param int nDim1 * Input: loop counter in dimension 1 * * \param int sDim2 * Input: start index for loop counter in dimension 2 * * \param int nDim2 * Input: loop counter in dimension 2 * * \return FIXP_DBL ret */ FIXP_DBL sumUpCplxPow2Dim2(const FIXP_DPK *const *const x, const INT scaleMode, const INT inScaleFactor, INT *const outScaleFactor, const INT sDim1, const INT nDim1, const INT sDim2, const INT nDim2); /** * \brief Vector function : Z[i] = X[i], i=0,...,n-1 * * Description : re{Z[i]} = re{X[i]}, i=0,...,n-1 * im{Z[i]} = im{X[i]}, i=0,...,n-1 * * Copy complex vector X[] to complex vector Z[]. * It is allowed to overlay X[] with Z[]. * * \param FIXP_DPK Z[] * Output: vector of the length n * * \param const FIXP_DPK X[] * Input: vector of the length n * * \param int n * Input: length of vector Z[] and X[] * * \return void */ void copyCplxVec(FIXP_DPK *const Z, const FIXP_DPK *const X, const INT n); /** * \brief Vector function : Z[i] = a, i=0,...,n-1 * * Description : re{Z[i]} = a, i=0,...,n-1 * im{Z[i]} = a, i=0,...,n-1 * * Set real and imaginary part of the complex value Z to a. * * \param FIPX_DPK Z[] * Output: vector of the length n * * \param const FIXP_DBL a * Input: constant value * * \param int n * Input: length of vector Z[] * * \return void */ void setCplxVec(FIXP_DPK *const Z, const FIXP_DBL a, const INT n); /** * \brief Vector function : Calculate complex-valued result of complex * scalar product * * Description : re{Z} = sum( re{X[i]} * re{Y[i]} + im{X[i]} * * im{Y[i]}, i=0,...,n-1 ) im{Z} = sum( im{X[i]} * re{Y[i]} - re{X[i]} * * im{Y[i]}, i=0,...,n-1 ) * * The function returns the complex-valued result of the complex * scalar product at the address of Z. The result is scaled by scaleZ. * * \param FIXP_DPK *Z * Output: pointer to Z * * \param const FIXP_DPK *const *const X * Input: vector of the length n * * \param const FIXP_DPK *const *const Y * Input: vector of the length n * * \param int scaleX * Input: scalefactor of vector X[] * * \param int scaleY * Input: scalefactor of vector Y[] * * \param int scaleZ * Output: scalefactor of vector Z[] * * \param int sDim1 * Input: start index for loop counter in dimension 1 * * \param int nDim1 * Input: loop counter in dimension 1 * * \param int sDim2 * Input: start index for loop counter in dimension 2 * * \param int nDim2 * Input: loop counter in dimension 2 * * \return void */ void cplx_cplxScalarProduct(FIXP_DPK *const Z, const FIXP_DPK *const *const X, const FIXP_DPK *const *const Y, const INT scaleX, const INT scaleY, INT *const scaleZ, const INT sDim1, const INT nDim1, const INT sDim2, const INT nDim2); /** * \brief Vector function : Calculate correlation * * Description : z[i] = pr12[i] / sqrt(p1[i]*p2[i]) , * i=0,...,n-1 * * \param FIXP_DBL z[] * Output: vector of length n * * \param const FIXP_DBL pr12[] * Input: vector of the length n * * \param const FIXP_DBL p1[] * Input: vector of the length n * * \param const FIXP_DBL p2[] * Input: vector of the length n * * \param int n * Input: length of vector pr12[], p1[] and p2[] * * \return void */ void FDKcalcCorrelationVec(FIXP_DBL *const z, const FIXP_DBL *const pr12, const FIXP_DBL *const p1, const FIXP_DBL *const p2, const INT n); /** * \brief Vector function : Calculate coherence * * Description : z[i] = sqrt( (p12r[i]*p12r[i] + * p12i[i]*p12i[i]) / (p1[i]*p2[i]) ), i=0,...,n-1 * * \param FIXP_DBL z[] * Output: vector of length n * * \param const FIXP_DBL p12r[] * Input: vector of the length n * * \param const FIXP_DBL p12i[] * Input: vector of the length n * * \param const FIXP_DBL p1[] * Input: vector of the length n * * \param const FIXP_DBL p2[] * Input: vector of the length n * * \param int scaleP12[] * Input: scalefactor of p12r and p12i * * \param int scaleP * Input: scalefactor of p1 and p2 * * \param int n * Input: length of vector p12r[], p12i[], p1[] and p2[] * * \return void */ void calcCoherenceVec(FIXP_DBL *const z, const FIXP_DBL *const p12r, const FIXP_DBL *const p12i, const FIXP_DBL *const p1, const FIXP_DBL *const p2, const INT scaleP12, const INT scaleP, const INT n); /** * \brief Vector function : Z[j][i] = a[pb] * X[j][i] + b[pb] * * Y[j][i], j=0,...,nHybridBands-1; i=startTimeSlot,...,nTimeSlots-1; * pb=0,...,nParameterBands-1 * * Description : re{Z[j][i]} = a[pb] * re{X[j][i]} + b[pb] * * re{Y[j][i]}, j=0,...,nHybridBands-1; i=startTimeSlot,...,nTimeSlots-1; * pb=0,...,nParameterBands-1 im{Z[j][i]} = a[pb] * im{X[j][i]} + b[pb] * * im{Y[j][i]}, j=0,...,nHybridBands-1; * i=startTimeSlot,...,nTimeSlots-1; pb=0,...,nParameterBands-1 * * It is allowed to overlay X[] or Y[] with Z[]. The scalefactor * of channel 1 is updated with the common scalefactor of channel 1 and * channel 2. * * \param FIXP_DPK **Z * Output: vector of the length nHybridBands*nTimeSlots * * \param const FIXP_DBL *a * Input: vector of length nParameterBands * * \param const FIXP_DPK **X * Input: vector of the length nHybridBands*nTimeSlots * * \param const FIXP_DBL *b * Input: vector of length nParameterBands * * \param const FIXP_DPK **Y * Input: vector of the length nHybridBands*nTimeSlots * * \param int scale * Input: scale of vector a and b * * \param int *scaleCh1 * Input: scale of ch1 * * \param int scaleCh2 * Input: scale of ch2 * * \param UCHAR *pParameterBand2HybridBandOffset * Input: vector of length nParameterBands * * \param int nTimeSlots * Input: number of time slots * * \param int startTimeSlot * Input: start time slot * * \return void */ void addWeightedCplxVec(FIXP_DPK *const *const Z, const FIXP_DBL *const a, const FIXP_DPK *const *const X, const FIXP_DBL *const b, const FIXP_DPK *const *const Y, const INT scale, INT *const scaleCh1, const INT scaleCh2, const UCHAR *const pParameterBand2HybridBandOffset, const INT nParameterBands, const INT nTimeSlots, const INT startTimeSlot); /** * \brief Vector function : Calculate the headroom of a complex vector * in a parameter band grid * * \param FIXP_DPK **x * Input: pointer to complex input vector * * \param UCHAR *pParameterBand2HybridBandOffset * Input: pointer to hybrid band offsets * * \param int *outScaleFactor * Input: pointer to ouput scalefactor * * \param int startTimeSlot * Input: start time slot * * \param int nTimeSlots * Input: number of time slot * * \param int nParamBands * Input: number of parameter bands * * \return void */ void FDKcalcPbScaleFactor(const FIXP_DPK *const *const x, const UCHAR *const pParameterBand2HybridBandOffset, INT *const outScaleFactor, const INT startTimeSlot, const INT nTimeSlots, const INT nParamBands); /** * \brief Vector function : Calculate the common headroom of two * sparate vectors * * \param FIXP_DBL *x * Input: pointer to first input vector * * \param FIXP_DBL *y * Input: pointer to second input vector * * \param int n * Input: number of samples * * \return int headromm in bits */ INT FDKcalcScaleFactor(const FIXP_DBL *const x, const FIXP_DBL *const y, const INT n); /** * \brief Vector function : Calculate the headroom of a complex vector * * \param FIXP_DPK *x * Input: pointer to complex input vector * * \param INT startBand * Input: start band * * \param INT bands * Input: number of bands * * \return int headromm in bits */ INT FDKcalcScaleFactorDPK(const FIXP_DPK *RESTRICT x, const INT startBand, const INT bands); /* Function / Class Definition ***********************************************/ template inline void FDKmemcpy_flex(T *const dst, const INT dstStride, const T *const src, const INT srcStride, const INT nSamples) { int i; for (i = 0; i < nSamples; i++) { dst[i * dstStride] = src[i * srcStride]; } } template inline void FDKmemset_flex(T *const x, const T c, const INT nSamples) { int i; for (i = 0; i < nSamples; i++) { x[i] = c; } } #endif /* SACENC_VECTORFUNCTIONS_H */ fdk-aac-2.0.2/libSBRdec/000077500000000000000000000000001404225310700145705ustar00rootroot00000000000000fdk-aac-2.0.2/libSBRdec/include/000077500000000000000000000000001404225310700162135ustar00rootroot00000000000000fdk-aac-2.0.2/libSBRdec/include/sbrdecoder.h000066400000000000000000000416431404225310700205100ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Description: SBR decoder front-end prototypes and definitions. *******************************************************************************/ #ifndef SBRDECODER_H #define SBRDECODER_H #include "common_fix.h" #include "FDK_bitstream.h" #include "FDK_audio.h" #include "FDK_qmf_domain.h" #define SBR_DEBUG_EXTHLP \ "\ --- SBR ---\n\ 0x00000010 Ancillary data and SBR-Header\n\ 0x00000020 SBR-Side info\n\ 0x00000040 Decoded SBR-bitstream data, e.g. envelope data\n\ 0x00000080 SBR-Bitstream statistics\n\ 0x00000100 Miscellaneous SBR-messages\n\ 0x00000200 SBR-Energies and gains in the adjustor\n\ 0x00000400 Fatal SBR errors\n\ 0x00000800 Transposer coefficients for inverse filtering\n\ " /* Capability flags */ #define CAPF_SBR_LP \ 0x00000001 /*!< Flag indicating library's capability of Low Power mode. */ #define CAPF_SBR_HQ \ 0x00000002 /*!< Flag indicating library's capability of High Quality mode. \ */ #define CAPF_SBR_DRM_BS \ 0x00000004 /*!< Flag indicating library's capability to decode DRM SBR data. \ */ #define CAPF_SBR_CONCEALMENT \ 0x00000008 /*!< Flag indicating library's capability to conceal erroneous \ frames. */ #define CAPF_SBR_DRC \ 0x00000010 /*!< Flag indicating library's capability for Dynamic Range \ Control. */ #define CAPF_SBR_PS_MPEG \ 0x00000020 /*!< Flag indicating library's capability to do MPEG Parametric \ Stereo. */ #define CAPF_SBR_PS_DRM \ 0x00000040 /*!< Flag indicating library's capability to do DRM Parametric \ Stereo. */ #define CAPF_SBR_ELD_DOWNSCALE \ 0x00000080 /*!< Flag indicating library's capability to do ELD decoding in \ downscaled mode */ #define CAPF_SBR_HBEHQ \ 0x00000100 /*!< Flag indicating library's capability to do HQ Harmonic \ transposing */ typedef enum { SBRDEC_OK = 0, /*!< All fine. */ /* SBRDEC_CONCEAL, */ /* SBRDEC_NOSYNCH, */ /* SBRDEC_ILLEGAL_PROGRAM, */ /* SBRDEC_ILLEGAL_TAG, */ /* SBRDEC_ILLEGAL_CHN_CONFIG, */ /* SBRDEC_ILLEGAL_SECTION, */ /* SBRDEC_ILLEGAL_SCFACTORS, */ /* SBRDEC_ILLEGAL_PULSE_DATA, */ /* SBRDEC_MAIN_PROFILE_NOT_IMPLEMENTED, */ /* SBRDEC_GC_NOT_IMPLEMENTED, */ /* SBRDEC_ILLEGAL_PLUS_ELE_ID, */ SBRDEC_INVALID_ARGUMENT, /*!< */ SBRDEC_CREATE_ERROR, /*!< */ SBRDEC_NOT_INITIALIZED, /*!< */ SBRDEC_MEM_ALLOC_FAILED, /*!< Memory allocation failed. Probably not enough memory available. */ SBRDEC_PARSE_ERROR, /*!< */ SBRDEC_UNSUPPORTED_CONFIG, /*!< */ SBRDEC_SET_PARAM_FAIL, /*!< */ SBRDEC_OUTPUT_BUFFER_TOO_SMALL /*!< */ } SBR_ERROR; typedef enum { SBR_SYSTEM_BITSTREAM_DELAY, /*!< System: Switch to enable an additional SBR bitstream delay of one frame. */ SBR_QMF_MODE, /*!< Set QMF mode, either complex or low power. */ SBR_LD_QMF_TIME_ALIGN, /*!< Set QMF type, either LD-MPS or CLDFB. Relevant for ELD streams only. */ SBR_FLUSH_DATA, /*!< Set internal state to flush the decoder with the next process call. */ SBR_CLEAR_HISTORY, /*!< Clear all internal states (delay lines, QMF states, ...). */ SBR_BS_INTERRUPTION /*!< Signal bit stream interruption. Value is ignored. */ , SBR_SKIP_QMF /*!< Enable skipping of QMF step: 1 skip analysis, 2 skip synthesis */ } SBRDEC_PARAM; typedef struct SBR_DECODER_INSTANCE *HANDLE_SBRDECODER; #ifdef __cplusplus extern "C" { #endif /** * \brief Allocates and initializes one SBR decoder instance. * \param pSelf Pointer to where a SBR decoder handle is copied into. * \param pQmfDomain Pointer to QMF domain data structure. * * \return Error code. */ SBR_ERROR sbrDecoder_Open(HANDLE_SBRDECODER *pSelf, HANDLE_FDK_QMF_DOMAIN pQmfDomain); /** * \brief Initialize a SBR decoder runtime instance. Must be called before * decoding starts. * * \param self Handle to a SBR decoder instance. * \param sampleRateIn Input samplerate of the SBR decoder instance. * \param sampleRateOut Output samplerate of the SBR decoder instance. * \param samplesPerFrame Number of samples per frames. * \param coreCodec Audio Object Type (AOT) of the core codec. * \param elementID Table with MPEG-4 element Ids in canonical order. * \param elementIndex SBR element index * \param harmonicSBR * \param stereoConfigIndex * \param downscaleFactor ELD downscale factor * \param configMode Table with MPEG-4 element Ids in canonical order. * \param configChanged Flag that enforces a complete decoder reset. * * \return Error code. */ SBR_ERROR sbrDecoder_InitElement( HANDLE_SBRDECODER self, const int sampleRateIn, const int sampleRateOut, const int samplesPerFrame, const AUDIO_OBJECT_TYPE coreCodec, const MP4_ELEMENT_ID elementID, const int elementIndex, const UCHAR harmonicSBR, const UCHAR stereoConfigIndex, const UCHAR configMode, UCHAR *configChanged, const INT downscaleFactor); /** * \brief Free config dependent SBR memory. * \param self SBR decoder instance handle */ SBR_ERROR sbrDecoder_FreeMem(HANDLE_SBRDECODER *self); /** * \brief pass out of band SBR header to SBR decoder * * \param self Handle to a SBR decoder instance. * \param hBs bit stream handle data source. * \param sampleRateIn SBR input sampling rate * \param sampleRateOut SBR output sampling rate * \param samplesPerFrame frame length * \param elementID SBR element ID. * \param elementIndex SBR element index. * \param harmonicSBR * \param stereoConfigIndex * \param downscaleFactor ELD downscale factor * * \return Error code. */ INT sbrDecoder_Header(HANDLE_SBRDECODER self, HANDLE_FDK_BITSTREAM hBs, const INT sampleRateIn, const INT sampleRateOut, const INT samplesPerFrame, const AUDIO_OBJECT_TYPE coreCodec, const MP4_ELEMENT_ID elementID, const INT elementIndex, const UCHAR harmonicSBR, const UCHAR stereoConfigIndex, const UCHAR configMode, UCHAR *configChanged, const INT downscaleFactor); /** * \brief Set a parameter of the SBR decoder runtime instance. * \param self SBR decoder handle. * \param param Parameter which will be set if successfull. * \param value New parameter value. * \return Error code. */ SBR_ERROR sbrDecoder_SetParam(HANDLE_SBRDECODER self, const SBRDEC_PARAM param, const INT value); /** * \brief Feed DRC channel data into a SBR decoder runtime instance. * * \param self SBR decoder handle. * \param ch Channel number to which the DRC data is * associated to. * \param numBands Number of DRC bands. * \param pNextFact_mag Pointer to a table with the DRC factor * magnitudes. * \param nextFact_exp Exponent for all DRC factors. * \param drcInterpolationScheme DRC interpolation scheme. * \param winSequence Window sequence from core coder (eight short * or one long window). * \param pBandTop Pointer to a table with the top borders for * all DRC bands. * * \return Error code. */ SBR_ERROR sbrDecoder_drcFeedChannel(HANDLE_SBRDECODER self, INT ch, UINT numBands, FIXP_DBL *pNextFact_mag, INT nextFact_exp, SHORT drcInterpolationScheme, UCHAR winSequence, USHORT *pBandTop); /** * \brief Disable SBR DRC for a certain channel. * * \param hSbrDecoder SBR decoder handle. * \param ch Number of the channel that has to be disabled. * * \return None. */ void sbrDecoder_drcDisable(HANDLE_SBRDECODER self, INT ch); /** * \brief Parse one SBR element data extension data block. The bit stream * position will be placed at the end of the SBR payload block. The remaining * bits will be returned into *count if a payload length is given * (byPayLen > 0). If no SBR payload length is given (bsPayLen < 0) then * the bit stream position on return will be random after this function * call in case of errors, and any further decoding will be completely * pointless. This function accepts either normal ordered SBR data or reverse * ordered DRM SBR data. * * \param self SBR decoder handle. * \param hBs Bit stream handle as data source. * \param count Pointer to an integer where the amount of parsed SBR * payload bits is stored into. * \param bsPayLen If > 0 this value is the SBR payload length. If < 0, * the SBR payload length is unknown. * \param flags CRC flag (0: EXT_SBR_DATA; 1: EXT_SBR_DATA_CRC) * \param prev_element Previous MPEG-4 element ID. * \param element_index Index of the current element. * \param acFlags Audio codec flags * * \return Error code. */ SBR_ERROR sbrDecoder_Parse(HANDLE_SBRDECODER self, HANDLE_FDK_BITSTREAM hBs, UCHAR *pDrmBsBuffer, USHORT drmBsBufferSize, int *count, int bsPayLen, int crcFlag, MP4_ELEMENT_ID prev_element, int element_index, UINT acFlags, UINT acElFlags[]); /** * \brief This function decodes the given SBR bitstreams and applies SBR to the * given time data. * * SBR-processing works InPlace. I.e. the calling function has to provide * a time domain buffer timeData which can hold the completely decoded * result. * * Left and right channel are read and stored according to the * interleaving flag, frame length and number of channels. * * \param self Handle of an open SBR decoder instance. * \param hSbrBs SBR Bitstream handle. * \param input Pointer to input data. * \param timeData Pointer to upsampled output data. * \param timeDataSize Size of timeData. * \param numChannels Pointer to a buffer holding the number of channels in * time data buffer. * \param sampleRate Output samplerate. * \param channelMapping Channel mapping indices. * \param coreDecodedOk Flag indicating if the core decoder did not find any * error (0: core decoder found errors, 1: no errors). * \param psDecoded Pointer to a buffer holding a flag. Input: PS is * possible, Output: PS has been rendered. * \param inDataHeadroom Headroom of the SBR input time signal to prevent * clipping. * \param outDataHeadroom Pointer to headroom of the SBR output time signal to * prevent clipping. * * \return Error code. */ SBR_ERROR sbrDecoder_Apply(HANDLE_SBRDECODER self, LONG *input, LONG *timeData, const int timeDataSize, int *numChannels, int *sampleRate, const FDK_channelMapDescr *const mapDescr, const int mapIdx, const int coreDecodedOk, UCHAR *psDecoded, const INT inDataHeadroom, INT *outDataHeadroom); /** * \brief Close SBR decoder instance and free memory. * \param self SBR decoder handle. * \return Error Code. */ SBR_ERROR sbrDecoder_Close(HANDLE_SBRDECODER *self); /** * \brief Get SBR decoder library information. * \param info Pointer to a LIB_INFO struct, where library information is * written to. * \return 0 on success, -1 if invalid handle or if no free element is * available to write information to. */ INT sbrDecoder_GetLibInfo(LIB_INFO *info); /** * \brief Determine the modules output signal delay in samples. * \param self SBR decoder handle. * \return The number of samples signal delay added by the module. */ UINT sbrDecoder_GetDelay(const HANDLE_SBRDECODER self); #ifdef __cplusplus } #endif #endif fdk-aac-2.0.2/libSBRdec/src/000077500000000000000000000000001404225310700153575ustar00rootroot00000000000000fdk-aac-2.0.2/libSBRdec/src/HFgen_preFlat.cpp000066400000000000000000001106731404225310700205370ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Oliver Moser, Manuel Jander, Matthias Hildenbrand Description: QMF frequency pre-whitening for SBR. In the documentation the terms "scale factor" and "exponent" mean the same. Variables containing such information have the suffix "_sf". *******************************************************************************/ #include "HFgen_preFlat.h" #define POLY_ORDER 3 #define MAXLOWBANDS 32 #define LOG10FAC 0.752574989159953f /* == 10/log2(10) * 2^-2 */ #define LOG10FAC_INV 0.664385618977472f /* == log2(10)/20 * 2^2 */ #define FIXP_CHB FIXP_SGL /* STB sinus Tab used in transformation */ #define CHC(a) (FX_DBL2FXCONST_SGL(a)) #define FX_CHB2FX_DBL(a) FX_SGL2FX_DBL(a) typedef struct backsubst_data { FIXP_CHB Lnorm1d[3]; /*!< Normalized L matrix */ SCHAR Lnorm1d_sf[3]; FIXP_CHB Lnormii [3]; /*!< The diagonal data points [i][i] of the normalized L matrix */ SCHAR Lnormii_sf[3]; FIXP_CHB Bmul0 [4]; /*!< To normalize L*x=b, Bmul0 is what we need to multiply b with. */ SCHAR Bmul0_sf[4]; FIXP_CHB LnormInv1d[6]; /*!< Normalized inverted L matrix (L') */ SCHAR LnormInv1d_sf[6]; FIXP_CHB Bmul1[4]; /*!< To normalize L'*x=b, Bmul1 is what we need to multiply b with. */ SCHAR Bmul1_sf[4]; } backsubst_data; /* for each element n do, f(n) = trunc(log2(n))+1 */ const UCHAR getLog2[32] = {0, 1, 2, 2, 3, 3, 3, 3, 4, 4, 4, 4, 4, 4, 4, 4, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5}; /** \def BSD_IDX_OFFSET * * bsd[] begins at index 0 with data for numBands=5. The correct bsd[] is * indexed like bsd[numBands-BSD_IDX_OFFSET]. */ #define BSD_IDX_OFFSET 5 #define N_NUMBANDS \ MAXLOWBANDS - BSD_IDX_OFFSET + \ 1 /*!< Number of backsubst_data elements in bsd */ const backsubst_data bsd[N_NUMBANDS] = { { /* numBands=5 */ {CHC(0x66c85a52), CHC(0x4278e587), CHC(0x697dcaff)}, {-1, 0, 0}, {CHC(0x66a61789), CHC(0x5253b8e3), CHC(0x5addad81)}, {3, 4, 1}, {CHC(0x7525ee90), CHC(0x6e2a1210), CHC(0x6523bb40), CHC(0x59822ead)}, {-6, -4, -2, 0}, {CHC(0x609e4cad), CHC(0x59c7e312), CHC(0x681eecac), CHC(0x440ea893), CHC(0x4a214bb3), CHC(0x53c345a1)}, {1, 0, -1, -1, -3, -5}, {CHC(0x7525ee90), CHC(0x58587936), CHC(0x410d0b38), CHC(0x7f1519d6)}, {-6, -1, 2, 0}, }, { /* numBands=6 */ {CHC(0x68943285), CHC(0x4841d2c3), CHC(0x6a6214c7)}, {-1, 0, 0}, {CHC(0x63c5923e), CHC(0x4e906e18), CHC(0x6285af8a)}, {3, 4, 1}, {CHC(0x7263940b), CHC(0x424a69a5), CHC(0x4ae8383a), CHC(0x517b7730)}, {-7, -4, -2, 0}, {CHC(0x518aee5f), CHC(0x4823a096), CHC(0x43764a39), CHC(0x6e6faf23), CHC(0x61bba44f), CHC(0x59d8b132)}, {1, 0, -1, -2, -4, -6}, {CHC(0x7263940b), CHC(0x6757bff2), CHC(0x5bf40fe0), CHC(0x7d6f4292)}, {-7, -2, 1, 0}, }, { /* numBands=7 */ {CHC(0x699b4c3c), CHC(0x4b8b702f), CHC(0x6ae51a4f)}, {-1, 0, 0}, {CHC(0x623a7f49), CHC(0x4ccc91fc), CHC(0x68f048dd)}, {3, 4, 1}, {CHC(0x7e6ebe18), CHC(0x5701daf2), CHC(0x74a8198b), CHC(0x4b399aa1)}, {-8, -5, -3, 0}, {CHC(0x464a64a6), CHC(0x78e42633), CHC(0x5ee174ba), CHC(0x5d0008c8), CHC(0x455cff0f), CHC(0x6b9100e7)}, {1, -1, -2, -2, -4, -7}, {CHC(0x7e6ebe18), CHC(0x42c52efe), CHC(0x45fe401f), CHC(0x7b5808ef)}, {-8, -2, 1, 0}, }, { /* numBands=8 */ {CHC(0x6a3fd9b4), CHC(0x4d99823f), CHC(0x6b372a94)}, {-1, 0, 0}, {CHC(0x614c6ef7), CHC(0x4bd06699), CHC(0x6e59cfca)}, {3, 4, 1}, {CHC(0x4c389cc5), CHC(0x79686681), CHC(0x5e2544c2), CHC(0x46305b43)}, {-8, -6, -3, 0}, {CHC(0x7b4ca7c6), CHC(0x68270ac5), CHC(0x467c644c), CHC(0x505c1b0f), CHC(0x67a14778), CHC(0x45801767)}, {0, -1, -2, -2, -5, -7}, {CHC(0x4c389cc5), CHC(0x5c499ceb), CHC(0x6f863c9f), CHC(0x79059bfc)}, {-8, -3, 0, 0}, }, { /* numBands=9 */ {CHC(0x6aad9988), CHC(0x4ef8ac18), CHC(0x6b6df116)}, {-1, 0, 0}, {CHC(0x60b159b0), CHC(0x4b33f772), CHC(0x72f5573d)}, {3, 4, 1}, {CHC(0x6206cb18), CHC(0x58a7d8dc), CHC(0x4e0b2d0b), CHC(0x4207ad84)}, {-9, -6, -3, 0}, {CHC(0x6dadadae), CHC(0x5b8b2cfc), CHC(0x6cf61db2), CHC(0x46c3c90b), CHC(0x506314ea), CHC(0x5f034acd)}, {0, -1, -3, -2, -5, -8}, {CHC(0x6206cb18), CHC(0x42f8b8de), CHC(0x5bb4776f), CHC(0x769acc79)}, {-9, -3, 0, 0}, }, { /* numBands=10 */ {CHC(0x6afa7252), CHC(0x4feed3ed), CHC(0x6b94504d)}, {-1, 0, 0}, {CHC(0x60467899), CHC(0x4acbafba), CHC(0x76eb327f)}, {3, 4, 1}, {CHC(0x42415b15), CHC(0x431080da), CHC(0x420f1c32), CHC(0x7d0c1aeb)}, {-9, -6, -3, -1}, {CHC(0x62b2c7a4), CHC(0x51b040a6), CHC(0x56caddb4), CHC(0x7e74a2c8), CHC(0x4030adf5), CHC(0x43d1dc4f)}, {0, -1, -3, -3, -5, -8}, {CHC(0x42415b15), CHC(0x64e299b3), CHC(0x4d33b5e8), CHC(0x742cee5f)}, {-9, -4, 0, 0}, }, { /* numBands=11 */ {CHC(0x6b3258bb), CHC(0x50a21233), CHC(0x6bb03c19)}, {-1, 0, 0}, {CHC(0x5ff997c6), CHC(0x4a82706e), CHC(0x7a5aae36)}, {3, 4, 1}, {CHC(0x5d2fb4fb), CHC(0x685bddd8), CHC(0x71b5e983), CHC(0x7708c90b)}, {-10, -7, -4, -1}, {CHC(0x59aceea2), CHC(0x49c428a0), CHC(0x46ca5527), CHC(0x724be884), CHC(0x68e586da), CHC(0x643485b6)}, {0, -1, -3, -3, -6, -9}, {CHC(0x5d2fb4fb), CHC(0x4e3fad1a), CHC(0x42310ba2), CHC(0x71c8b3ce)}, {-10, -4, 0, 0}, }, { /* numBands=12 */ {CHC(0x6b5c4726), CHC(0x5128a4a8), CHC(0x6bc52ee1)}, {-1, 0, 0}, {CHC(0x5fc06618), CHC(0x4a4ce559), CHC(0x7d5c16e9)}, {3, 4, 1}, {CHC(0x43af8342), CHC(0x531533d3), CHC(0x633660a6), CHC(0x71ce6052)}, {-10, -7, -4, -1}, {CHC(0x522373d7), CHC(0x434150cb), CHC(0x75b58afc), CHC(0x68474f2d), CHC(0x575348a5), CHC(0x4c20973f)}, {0, -1, -4, -3, -6, -9}, {CHC(0x43af8342), CHC(0x7c4d3d11), CHC(0x732e13db), CHC(0x6f756ac4)}, {-10, -5, -1, 0}, }, { /* numBands=13 */ {CHC(0x6b7c8953), CHC(0x51903fcd), CHC(0x6bd54d2e)}, {-1, 0, 0}, {CHC(0x5f94abf0), CHC(0x4a2480fa), CHC(0x40013553)}, {3, 4, 2}, {CHC(0x6501236e), CHC(0x436b9c4e), CHC(0x578d7881), CHC(0x6d34f92e)}, {-11, -7, -4, -1}, {CHC(0x4bc0e2b2), CHC(0x7b9d12ac), CHC(0x636c1c1b), CHC(0x5fe15c2b), CHC(0x49d54879), CHC(0x7662cfa5)}, {0, -2, -4, -3, -6, -10}, {CHC(0x6501236e), CHC(0x64b059fe), CHC(0x656d8359), CHC(0x6d370900)}, {-11, -5, -1, 0}, }, { /* numBands=14 */ {CHC(0x6b95e276), CHC(0x51e1b637), CHC(0x6be1f7ed)}, {-1, 0, 0}, {CHC(0x5f727a1c), CHC(0x4a053e9c), CHC(0x412e528c)}, {3, 4, 2}, {CHC(0x4d178bd4), CHC(0x6f33b4e8), CHC(0x4e028f7f), CHC(0x691ee104)}, {-11, -8, -4, -1}, {CHC(0x46473d3f), CHC(0x725bd0a6), CHC(0x55199885), CHC(0x58bcc56b), CHC(0x7e7e6288), CHC(0x5ddef6eb)}, {0, -2, -4, -3, -7, -10}, {CHC(0x4d178bd4), CHC(0x52ebd467), CHC(0x5a395a6e), CHC(0x6b0f724f)}, {-11, -5, -1, 0}, }, { /* numBands=15 */ {CHC(0x6baa2a22), CHC(0x5222eb91), CHC(0x6bec1a86)}, {-1, 0, 0}, {CHC(0x5f57393b), CHC(0x49ec8934), CHC(0x423b5b58)}, {3, 4, 2}, {CHC(0x77fd2486), CHC(0x5cfbdf2c), CHC(0x46153bd1), CHC(0x65757ed9)}, {-12, -8, -4, -1}, {CHC(0x41888ee6), CHC(0x6a661db3), CHC(0x49abc8c8), CHC(0x52965848), CHC(0x6d9301b7), CHC(0x4bb04721)}, {0, -2, -4, -3, -7, -10}, {CHC(0x77fd2486), CHC(0x45424c68), CHC(0x50f33cc6), CHC(0x68ff43f0)}, {-12, -5, -1, 0}, }, { /* numBands=16 */ {CHC(0x6bbaa499), CHC(0x5257ed94), CHC(0x6bf456e4)}, {-1, 0, 0}, {CHC(0x5f412594), CHC(0x49d8a766), CHC(0x432d1dbd)}, {3, 4, 2}, {CHC(0x5ef5cfde), CHC(0x4eafcd2d), CHC(0x7ed36893), CHC(0x62274b45)}, {-12, -8, -5, -1}, {CHC(0x7ac438f5), CHC(0x637aab21), CHC(0x4067617a), CHC(0x4d3c6ec7), CHC(0x5fd6e0dd), CHC(0x7bd5f024)}, {-1, -2, -4, -3, -7, -11}, {CHC(0x5ef5cfde), CHC(0x751d0d4f), CHC(0x492b3c41), CHC(0x67065409)}, {-12, -6, -1, 0}, }, { /* numBands=17 */ {CHC(0x6bc836c9), CHC(0x5283997e), CHC(0x6bfb1f5e)}, {-1, 0, 0}, {CHC(0x5f2f02b6), CHC(0x49c868e9), CHC(0x44078151)}, {3, 4, 2}, {CHC(0x4c43b65a), CHC(0x4349dcf6), CHC(0x73799e2d), CHC(0x5f267274)}, {-12, -8, -5, -1}, {CHC(0x73726394), CHC(0x5d68511a), CHC(0x7191bbcc), CHC(0x48898c70), CHC(0x548956e1), CHC(0x66981ce8)}, {-1, -2, -5, -3, -7, -11}, {CHC(0x4c43b65a), CHC(0x64131116), CHC(0x429028e2), CHC(0x65240211)}, {-12, -6, -1, 0}, }, { /* numBands=18 */ {CHC(0x6bd3860d), CHC(0x52a80156), CHC(0x6c00c68d)}, {-1, 0, 0}, {CHC(0x5f1fed86), CHC(0x49baf636), CHC(0x44cdb9dc)}, {3, 4, 2}, {CHC(0x7c189389), CHC(0x742666d8), CHC(0x69b8c776), CHC(0x5c67e27d)}, {-13, -9, -5, -1}, {CHC(0x6cf1ea76), CHC(0x58095703), CHC(0x64e351a9), CHC(0x4460da90), CHC(0x4b1f8083), CHC(0x55f2d3e1)}, {-1, -2, -5, -3, -7, -11}, {CHC(0x7c189389), CHC(0x5651792a), CHC(0x79cb9b3d), CHC(0x635769c0)}, {-13, -6, -2, 0}, }, { /* numBands=19 */ {CHC(0x6bdd0c40), CHC(0x52c6abf6), CHC(0x6c058950)}, {-1, 0, 0}, {CHC(0x5f133f88), CHC(0x49afb305), CHC(0x45826d73)}, {3, 4, 2}, {CHC(0x6621a164), CHC(0x6512528e), CHC(0x61449fc8), CHC(0x59e2a0c0)}, {-13, -9, -5, -1}, {CHC(0x6721cadb), CHC(0x53404cd4), CHC(0x5a389e91), CHC(0x40abcbd2), CHC(0x43332f01), CHC(0x48b82e46)}, {-1, -2, -5, -3, -7, -11}, {CHC(0x6621a164), CHC(0x4b12cc28), CHC(0x6ffd4df8), CHC(0x619f835e)}, {-13, -6, -2, 0}, }, { /* numBands=20 */ {CHC(0x6be524c5), CHC(0x52e0beb3), CHC(0x6c099552)}, {-1, 0, 0}, {CHC(0x5f087c68), CHC(0x49a62bb5), CHC(0x4627d175)}, {3, 4, 2}, {CHC(0x54ec6afe), CHC(0x58991a42), CHC(0x59e23e8c), CHC(0x578f4ef4)}, {-13, -9, -5, -1}, {CHC(0x61e78f6f), CHC(0x4ef5e1e9), CHC(0x5129c3b8), CHC(0x7ab0f7b2), CHC(0x78efb076), CHC(0x7c2567ea)}, {-1, -2, -5, -4, -8, -12}, {CHC(0x54ec6afe), CHC(0x41c7812c), CHC(0x676f6f8d), CHC(0x5ffb383f)}, {-13, -6, -2, 0}, }, { /* numBands=21 */ {CHC(0x6bec1542), CHC(0x52f71929), CHC(0x6c0d0d5e)}, {-1, 0, 0}, {CHC(0x5eff45c5), CHC(0x499e092d), CHC(0x46bfc0c9)}, {3, 4, 2}, {CHC(0x47457a78), CHC(0x4e2d99b3), CHC(0x53637ea5), CHC(0x5567d0e9)}, {-13, -9, -5, -1}, {CHC(0x5d2dc61b), CHC(0x4b1760c8), CHC(0x4967cf39), CHC(0x74b113d8), CHC(0x6d6676b6), CHC(0x6ad114e9)}, {-1, -2, -5, -4, -8, -12}, {CHC(0x47457a78), CHC(0x740accaa), CHC(0x5feb6609), CHC(0x5e696f95)}, {-13, -7, -2, 0}, }, { /* numBands=22 */ {CHC(0x6bf21387), CHC(0x530a683c), CHC(0x6c100c59)}, {-1, 0, 0}, {CHC(0x5ef752ea), CHC(0x499708c6), CHC(0x474bcd1b)}, {3, 4, 2}, {CHC(0x78a21ab7), CHC(0x45658aec), CHC(0x4da3c4fe), CHC(0x5367094b)}, {-14, -9, -5, -1}, {CHC(0x58e2df6a), CHC(0x4795990e), CHC(0x42b5e0f7), CHC(0x6f408c64), CHC(0x6370bebf), CHC(0x5c91ca85)}, {-1, -2, -5, -4, -8, -12}, {CHC(0x78a21ab7), CHC(0x66f951d6), CHC(0x594605bb), CHC(0x5ce91657)}, {-14, -7, -2, 0}, }, { /* numBands=23 */ {CHC(0x6bf749b2), CHC(0x531b3348), CHC(0x6c12a750)}, {-1, 0, 0}, {CHC(0x5ef06b17), CHC(0x4990f6c9), CHC(0x47cd4c5b)}, {3, 4, 2}, {CHC(0x66dede36), CHC(0x7bdf90a9), CHC(0x4885b2b9), CHC(0x5188a6b7)}, {-14, -10, -5, -1}, {CHC(0x54f85812), CHC(0x446414ae), CHC(0x79c8d519), CHC(0x6a4c2f31), CHC(0x5ac8325f), CHC(0x50bf9200)}, {-1, -2, -6, -4, -8, -12}, {CHC(0x66dede36), CHC(0x5be0d90e), CHC(0x535cc453), CHC(0x5b7923f0)}, {-14, -7, -2, 0}, }, { /* numBands=24 */ {CHC(0x6bfbd91d), CHC(0x5329e580), CHC(0x6c14eeed)}, {-1, 0, 0}, {CHC(0x5eea6179), CHC(0x498baa90), CHC(0x4845635d)}, {3, 4, 2}, {CHC(0x58559b7e), CHC(0x6f1b231f), CHC(0x43f1789b), CHC(0x4fc8fcb8)}, {-14, -10, -5, -1}, {CHC(0x51621775), CHC(0x417881a3), CHC(0x6f9ba9b6), CHC(0x65c412b2), CHC(0x53352c61), CHC(0x46db9caf)}, {-1, -2, -6, -4, -8, -12}, {CHC(0x58559b7e), CHC(0x52636003), CHC(0x4e13b316), CHC(0x5a189cdf)}, {-14, -7, -2, 0}, }, { /* numBands=25 */ {CHC(0x6bffdc73), CHC(0x5336d4af), CHC(0x6c16f084)}, {-1, 0, 0}, {CHC(0x5ee51249), CHC(0x498703cc), CHC(0x48b50e4f)}, {3, 4, 2}, {CHC(0x4c5616cf), CHC(0x641b9fad), CHC(0x7fa735e0), CHC(0x4e24e57a)}, {-14, -10, -6, -1}, {CHC(0x4e15f47a), CHC(0x7d9481d6), CHC(0x66a82f8a), CHC(0x619ae971), CHC(0x4c8b2f5f), CHC(0x7d09ec11)}, {-1, -3, -6, -4, -8, -13}, {CHC(0x4c5616cf), CHC(0x4a3770fb), CHC(0x495402de), CHC(0x58c693fa)}, {-14, -7, -2, 0}, }, { /* numBands=26 */ {CHC(0x6c036943), CHC(0x53424625), CHC(0x6c18b6dc)}, {-1, 0, 0}, {CHC(0x5ee060aa), CHC(0x4982e88a), CHC(0x491d277f)}, {3, 4, 2}, {CHC(0x425ada5b), CHC(0x5a9368ac), CHC(0x78380a42), CHC(0x4c99aa05)}, {-14, -10, -6, -1}, {CHC(0x4b0b569c), CHC(0x78a420da), CHC(0x5ebdf203), CHC(0x5dc57e63), CHC(0x46a650ff), CHC(0x6ee13fb8)}, {-1, -3, -6, -4, -8, -13}, {CHC(0x425ada5b), CHC(0x4323073c), CHC(0x450ae92b), CHC(0x57822ad5)}, {-14, -7, -2, 0}, }, { /* numBands=27 */ {CHC(0x6c06911a), CHC(0x534c7261), CHC(0x6c1a4aba)}, {-1, 0, 0}, {CHC(0x5edc3524), CHC(0x497f43c0), CHC(0x497e6cd8)}, {3, 4, 2}, {CHC(0x73fb550e), CHC(0x5244894f), CHC(0x717aad78), CHC(0x4b24ef6c)}, {-15, -10, -6, -1}, {CHC(0x483aebe4), CHC(0x74139116), CHC(0x57b58037), CHC(0x5a3a4f3c), CHC(0x416950fe), CHC(0x62c7f4f2)}, {-1, -3, -6, -4, -8, -13}, {CHC(0x73fb550e), CHC(0x79efb994), CHC(0x4128cab7), CHC(0x564a919a)}, {-15, -8, -2, 0}, }, { /* numBands=28 */ {CHC(0x6c096264), CHC(0x535587cd), CHC(0x6c1bb355)}, {-1, 0, 0}, {CHC(0x5ed87c76), CHC(0x497c0439), CHC(0x49d98452)}, {3, 4, 2}, {CHC(0x65dec5bf), CHC(0x4afd1ba3), CHC(0x6b58b4b3), CHC(0x49c4a7b0)}, {-15, -10, -6, -1}, {CHC(0x459e6eb1), CHC(0x6fd850b7), CHC(0x516e7be9), CHC(0x56f13d05), CHC(0x79785594), CHC(0x58617de7)}, {-1, -3, -6, -4, -9, -13}, {CHC(0x65dec5bf), CHC(0x6f2168aa), CHC(0x7b41310f), CHC(0x551f0692)}, {-15, -8, -3, 0}, }, { /* numBands=29 */ {CHC(0x6c0be913), CHC(0x535dacd5), CHC(0x6c1cf6a3)}, {-1, 0, 0}, {CHC(0x5ed526b4), CHC(0x49791bc5), CHC(0x4a2eff99)}, {3, 4, 2}, {CHC(0x59e44afe), CHC(0x44949ada), CHC(0x65bf36f5), CHC(0x487705a0)}, {-15, -10, -6, -1}, {CHC(0x43307779), CHC(0x6be959c4), CHC(0x4bce2122), CHC(0x53e34d89), CHC(0x7115ff82), CHC(0x4f6421a1)}, {-1, -3, -6, -4, -9, -13}, {CHC(0x59e44afe), CHC(0x659eab7d), CHC(0x74cea459), CHC(0x53fed574)}, {-15, -8, -3, 0}, }, { /* numBands=30 */ {CHC(0x6c0e2f17), CHC(0x53650181), CHC(0x6c1e199d)}, {-1, 0, 0}, {CHC(0x5ed2269f), CHC(0x49767e9e), CHC(0x4a7f5f0b)}, {3, 4, 2}, {CHC(0x4faa4ae6), CHC(0x7dd3bf11), CHC(0x609e2732), CHC(0x473a72e9)}, {-15, -11, -6, -1}, {CHC(0x40ec57c6), CHC(0x683ee147), CHC(0x46be261d), CHC(0x510a7983), CHC(0x698a84cb), CHC(0x4794a927)}, {-1, -3, -6, -4, -9, -13}, {CHC(0x4faa4ae6), CHC(0x5d3615ad), CHC(0x6ee74773), CHC(0x52e956a1)}, {-15, -8, -3, 0}, }, { /* numBands=31 */ {CHC(0x6c103cc9), CHC(0x536ba0ac), CHC(0x6c1f2070)}, {-1, 0, 0}, {CHC(0x5ecf711e), CHC(0x497422ea), CHC(0x4acb1438)}, {3, 4, 2}, {CHC(0x46e322ad), CHC(0x73c32f3c), CHC(0x5be7d172), CHC(0x460d8800)}, {-15, -11, -6, -1}, {CHC(0x7d9bf8ad), CHC(0x64d22351), CHC(0x422bdc81), CHC(0x4e6184aa), CHC(0x62ba2375), CHC(0x40c325de)}, {-2, -3, -6, -4, -9, -13}, {CHC(0x46e322ad), CHC(0x55bef2a3), CHC(0x697b3135), CHC(0x51ddee4d)}, {-15, -8, -3, 0}, }, { // numBands=32 {CHC(0x6c121933), CHC(0x5371a104), CHC(0x6c200ea0)}, {-1, 0, 0}, {CHC(0x5eccfcd3), CHC(0x49720060), CHC(0x4b1283f0)}, {3, 4, 2}, {CHC(0x7ea12a52), CHC(0x6aca3303), CHC(0x579072bf), CHC(0x44ef056e)}, {-16, -11, -6, -1}, {CHC(0x79a3a9ab), CHC(0x619d38fc), CHC(0x7c0f0734), CHC(0x4be3dd5d), CHC(0x5c8d7163), CHC(0x7591065f)}, {-2, -3, -7, -4, -9, -14}, {CHC(0x7ea12a52), CHC(0x4f1782a6), CHC(0x647cbcb2), CHC(0x50dc0bb1)}, {-16, -8, -3, 0}, }, }; /** \def SUM_SAFETY * * SUM_SAFTEY defines the bits needed to right-shift every summand in * order to be overflow-safe. In the two backsubst functions we sum up 4 * values. Since one of which is definitely not MAXVAL_DBL (the L[x][y]), * we spare just 2 safety bits instead of 3. */ #define SUM_SAFETY 2 /** * \brief Solves L*x=b via backsubstitution according to the following * structure: * * x[0] = b[0]; * x[1] = (b[1] - x[0]) / L[1][1]; * x[2] = (b[2] - x[1]*L[2][1] - x[0]) / L[2][2]; * x[3] = (b[3] - x[2]*L[3][2] - x[1]*L[3][1] - x[0]) / L[3][3]; * * \param[in] numBands SBR crossover band index * \param[in] b the b in L*x=b (one-dimensional) * \param[out] x output polynomial coefficients (mantissa) * \param[out] x_sf exponents of x[] */ static void backsubst_fw(const int numBands, const FIXP_DBL *const b, FIXP_DBL *RESTRICT x, int *RESTRICT x_sf) { int i, k; int m; /* the trip counter that indexes incrementally through Lnorm1d[] */ const FIXP_CHB *RESTRICT pLnorm1d = bsd[numBands - BSD_IDX_OFFSET].Lnorm1d; const SCHAR *RESTRICT pLnorm1d_sf = bsd[numBands - BSD_IDX_OFFSET].Lnorm1d_sf; const FIXP_CHB *RESTRICT pLnormii = bsd[numBands - BSD_IDX_OFFSET].Lnormii; const SCHAR *RESTRICT pLnormii_sf = bsd[numBands - BSD_IDX_OFFSET].Lnormii_sf; x[0] = b[0]; for (i = 1, m = 0; i <= POLY_ORDER; ++i) { FIXP_DBL sum = b[i] >> SUM_SAFETY; int sum_sf = x_sf[i]; for (k = i - 1; k > 0; --k, ++m) { int e; FIXP_DBL mult = fMultNorm(FX_CHB2FX_DBL(pLnorm1d[m]), x[k], &e); int mult_sf = pLnorm1d_sf[m] + x_sf[k] + e; /* check if the new summand mult has a different sf than the sum currently * has */ int diff = mult_sf - sum_sf; if (diff > 0) { /* yes, and it requires the sum to be adjusted (scaled down) */ sum >>= diff; sum_sf = mult_sf; } else if (diff < 0) { /* yes, but here mult needs to be scaled down */ mult >>= -diff; } sum -= (mult >> SUM_SAFETY); } /* - x[0] */ if (x_sf[0] > sum_sf) { sum >>= (x_sf[0] - sum_sf); sum_sf = x_sf[0]; } sum -= (x[0] >> (sum_sf - x_sf[0] + SUM_SAFETY)); /* instead of the division /L[i][i], we multiply by the inverse */ int e; x[i] = fMultNorm(sum, FX_CHB2FX_DBL(pLnormii[i - 1]), &e); x_sf[i] = sum_sf + pLnormii_sf[i - 1] + e + SUM_SAFETY; } } /** * \brief Solves L*x=b via backsubstitution according to the following * structure: * * x[3] = b[3]; * x[2] = b[2] - L[2][3]*x[3]; * x[1] = b[1] - L[1][2]*x[2] - L[1][3]*x[3]; * x[0] = b[0] - L[0][1]*x[1] - L[0][2]*x[2] - L[0][3]*x[3]; * * \param[in] numBands SBR crossover band index * \param[in] b the b in L*x=b (one-dimensional) * \param[out] x solution vector * \param[out] x_sf exponents of x[] */ static void backsubst_bw(const int numBands, const FIXP_DBL *const b, FIXP_DBL *RESTRICT x, int *RESTRICT x_sf) { int i, k; int m; /* the trip counter that indexes incrementally through LnormInv1d[] */ const FIXP_CHB *RESTRICT pLnormInv1d = bsd[numBands - BSD_IDX_OFFSET].LnormInv1d; const SCHAR *RESTRICT pLnormInv1d_sf = bsd[numBands - BSD_IDX_OFFSET].LnormInv1d_sf; x[POLY_ORDER] = b[POLY_ORDER]; for (i = POLY_ORDER - 1, m = 0; i >= 0; i--) { FIXP_DBL sum = b[i] >> SUM_SAFETY; int sum_sf = x_sf[i]; /* sum's sf but disregarding SUM_SAFETY (added at the iteration's end) */ for (k = i + 1; k <= POLY_ORDER; ++k, ++m) { int e; FIXP_DBL mult = fMultNorm(FX_CHB2FX_DBL(pLnormInv1d[m]), x[k], &e); int mult_sf = pLnormInv1d_sf[m] + x_sf[k] + e; /* check if the new summand mult has a different sf than sum currently has */ int diff = mult_sf - sum_sf; if (diff > 0) { /* yes, and it requires the sum v to be adjusted (scaled down) */ sum >>= diff; sum_sf = mult_sf; } else if (diff < 0) { /* yes, but here mult needs to be scaled down */ mult >>= -diff; } /* mult has now the same sf than what it is about to be added to. */ /* scale mult down additionally so that building the sum is overflow-safe. */ sum -= (mult >> SUM_SAFETY); } x_sf[i] = sum_sf + SUM_SAFETY; x[i] = sum; } } /** * \brief Solves a system of linear equations (L*x=b) with the Cholesky * algorithm. * * \param[in] numBands SBR crossover band index * \param[in,out] b input: vector b, output: solution vector p. * \param[in,out] b_sf input: exponent of b; output: exponent of solution * p. */ static void choleskySolve(const int numBands, FIXP_DBL *RESTRICT b, int *RESTRICT b_sf) { int i, e; const FIXP_CHB *RESTRICT pBmul0 = bsd[numBands - BSD_IDX_OFFSET].Bmul0; const SCHAR *RESTRICT pBmul0_sf = bsd[numBands - BSD_IDX_OFFSET].Bmul0_sf; const FIXP_CHB *RESTRICT pBmul1 = bsd[numBands - BSD_IDX_OFFSET].Bmul1; const SCHAR *RESTRICT pBmul1_sf = bsd[numBands - BSD_IDX_OFFSET].Bmul1_sf; /* normalize b */ FIXP_DBL bnormed[POLY_ORDER + 1]; for (i = 0; i <= POLY_ORDER; ++i) { bnormed[i] = fMultNorm(b[i], FX_CHB2FX_DBL(pBmul0[i]), &e); b_sf[i] += pBmul0_sf[i] + e; } backsubst_fw(numBands, bnormed, b, b_sf); /* normalize b again */ for (i = 0; i <= POLY_ORDER; ++i) { bnormed[i] = fMultNorm(b[i], FX_CHB2FX_DBL(pBmul1[i]), &e); b_sf[i] += pBmul1_sf[i] + e; } backsubst_bw(numBands, bnormed, b, b_sf); } /** * \brief Find polynomial approximation of vector y with implicit abscisas * x=0,1,2,3..n-1 * * The problem (V^T * V * p = V^T * y) is solved with Cholesky. * V is the Vandermode Matrix constructed with x = 0...n-1; * A = V^T * V; b = V^T * y; * * \param[in] numBands SBR crossover band index (BSD_IDX_OFFSET <= numBands <= * MAXLOWBANDS) * \param[in] y input vector (mantissa) * \param[in] y_sf exponents of y[] * \param[out] p output polynomial coefficients (mantissa) * \param[out] p_sf exponents of p[] */ static void polyfit(const int numBands, const FIXP_DBL *const y, const int y_sf, FIXP_DBL *RESTRICT p, int *RESTRICT p_sf) { int i, k; LONG v[POLY_ORDER + 1]; int sum_saftey = getLog2[numBands - 1]; FDK_ASSERT((numBands >= BSD_IDX_OFFSET) && (numBands <= MAXLOWBANDS)); /* construct vector b[] temporarily stored in array p[] */ FDKmemclear(p, (POLY_ORDER + 1) * sizeof(FIXP_DBL)); /* p[] are the sums over n values and each p[i] has its own sf */ for (i = 0; i <= POLY_ORDER; ++i) p_sf[i] = 1 - DFRACT_BITS; for (k = 0; k < numBands; k++) { v[0] = (LONG)1; for (i = 1; i <= POLY_ORDER; i++) { v[i] = k * v[i - 1]; } for (i = 0; i <= POLY_ORDER; i++) { if (v[POLY_ORDER - i] != 0 && y[k] != FIXP_DBL(0)) { int e; FIXP_DBL mult = fMultNorm((FIXP_DBL)v[POLY_ORDER - i], y[k], &e); int sf = DFRACT_BITS - 1 + y_sf + e; /* check if the new summand has a different sf than the sum p[i] * currently has */ int diff = sf - p_sf[i]; if (diff > 0) { /* yes, and it requires the sum p[i] to be adjusted (scaled down) */ p[i] >>= fMin(DFRACT_BITS - 1, diff); p_sf[i] = sf; } else if (diff < 0) { /* yes, but here mult needs to be scaled down */ mult >>= -diff; } /* mult has now the same sf than what it is about to be added to. scale mult down additionally so that building the sum is overflow-safe. */ p[i] += mult >> sum_saftey; } } } p_sf[0] += sum_saftey; p_sf[1] += sum_saftey; p_sf[2] += sum_saftey; p_sf[3] += sum_saftey; choleskySolve(numBands, p, p_sf); } /** * \brief Calculates the output of a POLY_ORDER-degree polynomial function * with Horner scheme: * * y(x) = p3 + p2*x + p1*x^2 + p0*x^3 * = p3 + x*(p2 + x*(p1 + x*p0)) * * The for loop iterates through the mult/add parts in y(x) as above, * during which regular upscaling ensures a stable exponent of the * result. * * \param[in] p coefficients as in y(x) * \param[in] p_sf exponents of p[] * \param[in] x_int non-fractional integer representation of x as in y(x) * \param[out] out_sf exponent of return value * * \return result y(x) */ static FIXP_DBL polyval(const FIXP_DBL *const p, const int *const p_sf, const int x_int, int *out_sf) { FDK_ASSERT(x_int <= 31); /* otherwise getLog2[] needs more elements */ int k, x_sf; int result_sf; /* working space to compute return value *out_sf */ FIXP_DBL x; /* fractional value of x_int */ FIXP_DBL result; /* return value */ /* if x == 0, then y(x) is just p3 */ if (x_int != 0) { x_sf = getLog2[x_int]; x = (FIXP_DBL)x_int << (DFRACT_BITS - 1 - x_sf); } else { *out_sf = p_sf[3]; return p[3]; } result = p[0]; result_sf = p_sf[0]; for (k = 1; k <= POLY_ORDER; ++k) { FIXP_DBL mult = fMult(x, result); int mult_sf = x_sf + result_sf; int room = CountLeadingBits(mult); mult <<= room; mult_sf -= room; FIXP_DBL pp = p[k]; int pp_sf = p_sf[k]; /* equalize the shift factors of pp and mult so that we can sum them up */ int diff = pp_sf - mult_sf; if (diff > 0) { diff = fMin(diff, DFRACT_BITS - 1); mult >>= diff; } else if (diff < 0) { diff = fMax(diff, 1 - DFRACT_BITS); pp >>= -diff; } /* downshift by 1 to ensure safe summation */ mult >>= 1; mult_sf++; pp >>= 1; pp_sf++; result_sf = fMax(pp_sf, mult_sf); result = mult + pp; /* rarely, mult and pp happen to be almost equal except their sign, and then upon summation, result becomes so small, that it is within the inaccuracy range of a few bits, and then the relative error produced by this function may become HUGE */ } *out_sf = result_sf; return result; } void sbrDecoder_calculateGainVec(FIXP_DBL **sourceBufferReal, FIXP_DBL **sourceBufferImag, int sourceBuf_e_overlap, int sourceBuf_e_current, int overlap, FIXP_DBL *RESTRICT GainVec, int *GainVec_exp, int numBands, const int startSample, const int stopSample) { FIXP_DBL p[POLY_ORDER + 1]; FIXP_DBL meanNrg; FIXP_DBL LowEnv[MAXLOWBANDS]; FIXP_DBL invNumBands = GetInvInt(numBands); FIXP_DBL invNumSlots = GetInvInt(stopSample - startSample); int i, loBand, exp, scale_nrg, scale_nrg_ov; int sum_scale = 5, sum_scale_ov = 3; if (overlap > 8) { FDK_ASSERT(overlap <= 16); sum_scale_ov += 1; sum_scale += 1; } /* exponents of energy values */ sourceBuf_e_overlap = sourceBuf_e_overlap * 2 + sum_scale_ov; sourceBuf_e_current = sourceBuf_e_current * 2 + sum_scale; exp = fMax(sourceBuf_e_overlap, sourceBuf_e_current); scale_nrg = sourceBuf_e_current - exp; scale_nrg_ov = sourceBuf_e_overlap - exp; meanNrg = (FIXP_DBL)0; /* Calculate the spectral envelope in dB over the current copy-up frame. */ for (loBand = 0; loBand < numBands; loBand++) { FIXP_DBL nrg_ov, nrg; INT reserve = 0, exp_new; FIXP_DBL maxVal = FL2FX_DBL(0.0f); for (i = startSample; i < stopSample; i++) { maxVal |= (FIXP_DBL)((LONG)(sourceBufferReal[i][loBand]) ^ ((LONG)sourceBufferReal[i][loBand] >> (DFRACT_BITS - 1))); maxVal |= (FIXP_DBL)((LONG)(sourceBufferImag[i][loBand]) ^ ((LONG)sourceBufferImag[i][loBand] >> (DFRACT_BITS - 1))); } if (maxVal != FL2FX_DBL(0.0f)) { reserve = CntLeadingZeros(maxVal) - 2; } nrg_ov = nrg = (FIXP_DBL)0; if (scale_nrg_ov > -31) { for (i = startSample; i < overlap; i++) { nrg_ov += (fPow2Div2(scaleValue(sourceBufferReal[i][loBand], reserve)) + fPow2Div2(scaleValue(sourceBufferImag[i][loBand], reserve))) >> sum_scale_ov; } } else { scale_nrg_ov = 0; } if (scale_nrg > -31) { for (i = overlap; i < stopSample; i++) { nrg += (fPow2Div2(scaleValue(sourceBufferReal[i][loBand], reserve)) + fPow2Div2(scaleValue(sourceBufferImag[i][loBand], reserve))) >> sum_scale; } } else { scale_nrg = 0; } nrg = (scaleValue(nrg_ov, scale_nrg_ov) >> 1) + (scaleValue(nrg, scale_nrg) >> 1); nrg = fMult(nrg, invNumSlots); exp_new = exp - (2 * reserve) + 2; /* +1 for addition directly above, +1 for fPow2Div2 in loops above */ /* LowEnv = 10*log10(nrg) = log2(nrg) * 10/log2(10) */ /* exponent of logarithmic energy is 8 */ if (nrg > (FIXP_DBL)0) { int exp_log2; nrg = CalcLog2(nrg, exp_new, &exp_log2); nrg = scaleValue(nrg, exp_log2 - 6); nrg = fMult(FL2FXCONST_SGL(LOG10FAC), nrg); } else { nrg = (FIXP_DBL)0; } LowEnv[loBand] = nrg; meanNrg += fMult(nrg, invNumBands); } exp = 6 + 2; /* exponent of LowEnv: +2 is exponent of LOG10FAC */ /* subtract mean before polynomial approximation to reduce dynamic of p[] */ for (loBand = 0; loBand < numBands; loBand++) { LowEnv[loBand] = meanNrg - LowEnv[loBand]; } /* For numBands < BSD_IDX_OFFSET (== POLY_ORDER+2) we dont get an overdetermined equation system. The calculated polynomial will exactly fit the input data and evaluating the polynomial will lead to the same vector than the original input vector: lowEnvSlope[] == lowEnv[] */ if (numBands > POLY_ORDER + 1) { /* Find polynomial approximation of LowEnv */ int p_sf[POLY_ORDER + 1]; polyfit(numBands, LowEnv, exp, p, p_sf); for (i = 0; i < numBands; i++) { int sf; /* lowBandEnvSlope[i] = tmp; */ FIXP_DBL tmp = polyval(p, p_sf, i, &sf); /* GainVec = 10^((mean(y)-y)/20) = 2^( (mean(y)-y) * log2(10)/20 ) */ tmp = fMult(tmp, FL2FXCONST_SGL(LOG10FAC_INV)); GainVec[i] = f2Pow(tmp, sf - 2, &GainVec_exp[i]); /* -2 is exponent of LOG10FAC_INV */ } } else { /* numBands <= POLY_ORDER+1 */ for (i = 0; i < numBands; i++) { int sf = exp; /* exponent of LowEnv[] */ /* lowBandEnvSlope[i] = LowEnv[i]; */ FIXP_DBL tmp = LowEnv[i]; /* GainVec = 10^((mean(y)-y)/20) = 2^( (mean(y)-y) * log2(10)/20 ) */ tmp = fMult(tmp, FL2FXCONST_SGL(LOG10FAC_INV)); GainVec[i] = f2Pow(tmp, sf - 2, &GainVec_exp[i]); /* -2 is exponent of LOG10FAC_INV */ } } } fdk-aac-2.0.2/libSBRdec/src/HFgen_preFlat.h000066400000000000000000000140641404225310700202010ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Manuel Jander, Matthias Hildenbrand Description: QMF frequency pre whitening for SBR *******************************************************************************/ #include "common_fix.h" #ifndef HFGEN_PREFLAT_H #define HFGEN_PREFLAT_H #define GAIN_VEC_EXP 6 /* exponent of GainVec[] */ /** * \brief Find gain vector to flatten the QMF frequency bands whithout loosing * the fine structure. * \param[in] sourceBufferReal real part of QMF domain data. * \param[in] sourceBufferImag imaginary part of QMF domain data. * \param[in] sourceBuffer_e_overlap exponent of sourceBufferReal. * \param[in] sourceBuffer_e_current exponent of sourceBufferImag. * \param[in] overlap number of overlap samples. * \param[out] GainVec array of gain values (one for each QMF band). * \param[out] GainVec_exp exponents of GainVec (one for each QMF band). * \param[in] numBands number of low bands (k_0). * \param[in] startSample time slot start. * \param[in] stopSample time slot stop. */ void sbrDecoder_calculateGainVec(FIXP_DBL **sourceBufferReal, FIXP_DBL **sourceBufferImag, int sourceBuffer_e_overlap, int sourceBuffer_e_current, int overlap, FIXP_DBL GainVec[], int GainVec_exp[], const int numBands, const int startSample, const int stopSample); #endif /* __HFGEN_PREFLAT_H */ fdk-aac-2.0.2/libSBRdec/src/arm/000077500000000000000000000000001404225310700161365ustar00rootroot00000000000000fdk-aac-2.0.2/libSBRdec/src/arm/lpp_tran_arm.cpp000066400000000000000000000153061404225310700213250ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Arthur Tritthart Description: (ARM optimised) LPP transposer subroutines *******************************************************************************/ #if defined(__arm__) #define FUNCTION_LPPTRANSPOSER_func1 #ifdef FUNCTION_LPPTRANSPOSER_func1 /* Note: This code requires only 43 cycles per iteration instead of 61 on * ARM926EJ-S */ static void lppTransposer_func1(FIXP_DBL *lowBandReal, FIXP_DBL *lowBandImag, FIXP_DBL **qmfBufferReal, FIXP_DBL **qmfBufferImag, int loops, int hiBand, int dynamicScale, int descale, FIXP_SGL a0r, FIXP_SGL a0i, FIXP_SGL a1r, FIXP_SGL a1i, const int fPreWhitening, FIXP_DBL preWhiteningGain, int preWhiteningGains_sf) { FIXP_DBL real1, real2, imag1, imag2, accu1, accu2; real2 = lowBandReal[-2]; real1 = lowBandReal[-1]; imag2 = lowBandImag[-2]; imag1 = lowBandImag[-1]; for (int i = 0; i < loops; i++) { accu1 = fMultDiv2(a0r, real1); accu2 = fMultDiv2(a0i, imag1); accu1 = fMultAddDiv2(accu1, a1r, real2); accu2 = fMultAddDiv2(accu2, a1i, imag2); real2 = fMultDiv2(a1i, real2); accu1 = accu1 - accu2; accu1 = accu1 >> dynamicScale; accu2 = fMultAddDiv2(real2, a1r, imag2); real2 = real1; imag2 = imag1; accu2 = fMultAddDiv2(accu2, a0i, real1); real1 = lowBandReal[i]; accu2 = fMultAddDiv2(accu2, a0r, imag1); imag1 = lowBandImag[i]; accu2 = accu2 >> dynamicScale; accu1 <<= 1; accu2 <<= 1; accu1 += (real1 >> descale); accu2 += (imag1 >> descale); if (fPreWhitening) { accu1 = scaleValueSaturate(fMultDiv2(accu1, preWhiteningGain), preWhiteningGains_sf); accu2 = scaleValueSaturate(fMultDiv2(accu2, preWhiteningGain), preWhiteningGains_sf); } qmfBufferReal[i][hiBand] = accu1; qmfBufferImag[i][hiBand] = accu2; } } #endif /* #ifdef FUNCTION_LPPTRANSPOSER_func1 */ #endif /* __arm__ */ fdk-aac-2.0.2/libSBRdec/src/env_calc.cpp000066400000000000000000003521711404225310700176460ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Envelope calculation The envelope adjustor compares the energies present in the transposed highband to the reference energies conveyed with the bitstream. The highband is amplified (sometimes) or attenuated (mostly) to the desired level. The spectral shape of the reference energies can be changed several times per frame if necessary. Each set of energy values corresponding to a certain range in time will be called an envelope here. The bitstream supports several frequency scales and two resolutions. Normally, one or more QMF-subbands are grouped to one SBR-band. An envelope contains reference energies for each SBR-band. In addition to the energy envelopes, noise envelopes are transmitted that define the ratio of energy which is generated by adding noise instead of transposing the lowband. The noise envelopes are given in a coarser time and frequency resolution. If a signal contains strong tonal components, synthetic sines can be generated in individual SBR bands. An overlap buffer of 6 QMF-timeslots is used to allow a more flexible alignment of the envelopes in time that is not restricted to the core codec's frame borders. Therefore the envelope adjustor has access to the spectral data of the current frame as well as the last 6 QMF-timeslots of the previous frame. However, in average only the data of 1 frame is being processed as the adjustor is called once per frame. Depending on the frequency range set in the bitstream, only QMF-subbands between lowSubband and highSubband are adjusted. Scaling of spectral data to maximize SNR (see #QMF_SCALE_FACTOR) as well as a special Mantissa-Exponent format ( see calculateSbrEnvelope() ) are being used. The main entry point for this modules is calculateSbrEnvelope(). \sa sbr_scale.h, #QMF_SCALE_FACTOR, calculateSbrEnvelope(), \ref documentationOverview */ #include "env_calc.h" #include "sbrdec_freq_sca.h" #include "env_extr.h" #include "transcendent.h" #include "sbr_ram.h" #include "sbr_rom.h" #include "genericStds.h" /* need FDKpow() for debug outputs */ #define MAX_SFB_NRG_HEADROOM (1) #define MAX_VAL_NRG_HEADROOM ((((FIXP_DBL)MAXVAL_DBL) >> MAX_SFB_NRG_HEADROOM)) typedef struct { FIXP_DBL nrgRef[MAX_FREQ_COEFFS]; FIXP_DBL nrgEst[MAX_FREQ_COEFFS]; FIXP_DBL nrgGain[MAX_FREQ_COEFFS]; FIXP_DBL noiseLevel[MAX_FREQ_COEFFS]; FIXP_DBL nrgSine[MAX_FREQ_COEFFS]; SCHAR nrgRef_e[MAX_FREQ_COEFFS]; SCHAR nrgEst_e[MAX_FREQ_COEFFS]; SCHAR nrgGain_e[MAX_FREQ_COEFFS]; SCHAR noiseLevel_e[MAX_FREQ_COEFFS]; SCHAR nrgSine_e[MAX_FREQ_COEFFS]; /* yet another exponent [0]: for ts < no_cols; [1]: for ts >= no_cols */ SCHAR exponent[2]; } ENV_CALC_NRGS; static void equalizeFiltBufferExp(FIXP_DBL *filtBuffer, SCHAR *filtBuffer_e, FIXP_DBL *NrgGain, SCHAR *NrgGain_e, int subbands); static void calcNrgPerSubband(FIXP_DBL **analysBufferReal, FIXP_DBL **analysBufferImag, int lowSubband, int highSubband, int start_pos, int next_pos, SCHAR frameExp, FIXP_DBL *nrgEst, SCHAR *nrgEst_e); static void calcNrgPerSfb(FIXP_DBL **analysBufferReal, FIXP_DBL **analysBufferImag, int nSfb, UCHAR *freqBandTable, int start_pos, int next_pos, SCHAR input_e, FIXP_DBL *nrg_est, SCHAR *nrg_est_e); static void calcSubbandGain(FIXP_DBL nrgRef, SCHAR nrgRef_e, ENV_CALC_NRGS *nrgs, int c, FIXP_DBL tmpNoise, SCHAR tmpNoise_e, UCHAR sinePresentFlag, UCHAR sineMapped, int noNoiseFlag); static void calcAvgGain(ENV_CALC_NRGS *nrgs, int lowSubband, int highSubband, FIXP_DBL *sumRef_m, SCHAR *sumRef_e, FIXP_DBL *ptrAvgGain_m, SCHAR *ptrAvgGain_e); static void adjustTimeSlot_EldGrid(FIXP_DBL *ptrReal, ENV_CALC_NRGS *nrgs, UCHAR *ptrHarmIndex, int lowSubbands, int noSubbands, int scale_change, int noNoiseFlag, int *ptrPhaseIndex, int scale_diff_low); static void adjustTimeSlotLC(FIXP_DBL *ptrReal, ENV_CALC_NRGS *nrgs, UCHAR *ptrHarmIndex, int lowSubbands, int noSubbands, int scale_change, int noNoiseFlag, int *ptrPhaseIndex); /** * \brief Variant of adjustTimeSlotHQ() which only regards gain and noise but no * additional harmonics */ static void adjustTimeSlotHQ_GainAndNoise( FIXP_DBL *ptrReal, FIXP_DBL *ptrImag, HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env, ENV_CALC_NRGS *nrgs, int lowSubbands, int noSubbands, int scale_change, FIXP_SGL smooth_ratio, int noNoiseFlag, int filtBufferNoiseShift); /** * \brief Variant of adjustTimeSlotHQ() which only adds the additional harmonics */ static void adjustTimeSlotHQ_AddHarmonics( FIXP_DBL *ptrReal, FIXP_DBL *ptrImag, HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env, ENV_CALC_NRGS *nrgs, int lowSubbands, int noSubbands, int scale_change); static void adjustTimeSlotHQ(FIXP_DBL *ptrReal, FIXP_DBL *ptrImag, HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env, ENV_CALC_NRGS *nrgs, int lowSubbands, int noSubbands, int scale_change, FIXP_SGL smooth_ratio, int noNoiseFlag, int filtBufferNoiseShift); /*! \brief Map sine flags from bitstream to QMF bands The bitstream carries only 1 sine flag per band (Sfb) and frame. This function maps every sine flag from the bitstream to a specific QMF subband and to a specific envelope where the sine shall start. The result is stored in the vector sineMapped which contains one entry per QMF subband. The value of an entry specifies the envelope where a sine shall start. A value of 32 indicates that no sine is present in the subband. The missing harmonics flags from the previous frame (harmFlagsPrev) determine if a sine starts at the beginning of the frame or at the transient position. Additionally, the flags in harmFlagsPrev are being updated by this function for the next frame. */ static void mapSineFlags( UCHAR *freqBandTable, /*!< Band borders (there's only 1 flag per band) */ int nSfb, /*!< Number of bands in the table */ ULONG *addHarmonics, /*!< Packed addHarmonics of current frame (aligned to the MSB) */ ULONG *harmFlagsPrev, /*!< Packed addHarmonics of previous frame (aligned to the LSB) */ ULONG *harmFlagsPrevActive, /*!< Packed sineMapped of previous frame (aligned to the LSB) */ int tranEnv, /*!< Transient position */ SCHAR *sineMapped) /*!< Resulting vector of sine start positions for each QMF band */ { int i; int bitcount = 31; ULONG harmFlagsQmfBands[ADD_HARMONICS_FLAGS_SIZE] = {0}; ULONG *curFlags = addHarmonics; /* Format of addHarmonics (aligned to MSB): Up to MAX_FREQ_COEFFS sfb bands can be flagged for a sign. first word = flags for lowest 32 sfb bands in use second word = flags for higest 32 sfb bands (if present) Format of harmFlagsPrev (aligned to LSB): Index is absolute (not relative to lsb) so it is correct even if lsb changes first word = flags for lowest 32 qmf bands (0...31) second word = flags for next higher 32 qmf bands (32...63) */ /* Reset the output vector first */ FDKmemset(sineMapped, 32, MAX_FREQ_COEFFS * sizeof(SCHAR)); /* 32 means 'no sine' */ FDKmemclear(harmFlagsPrevActive, ADD_HARMONICS_FLAGS_SIZE * sizeof(ULONG)); for (i = 0; i < nSfb; i++) { ULONG maskSfb = 1 << bitcount; /* mask to extract addHarmonics flag of current Sfb */ if (*curFlags & maskSfb) { /* There is a sine in this band */ const int lsb = freqBandTable[0]; /* start of sbr range */ /* qmf band to which sine should be added */ const int qmfBand = (freqBandTable[i] + freqBandTable[i + 1]) >> 1; const int qmfBandDiv32 = qmfBand >> 5; const int maskQmfBand = 1 << (qmfBand & 31); /* mask to extract harmonic flag from prevFlags */ /* mapping of sfb with sine to a certain qmf band -> for harmFlagsPrev */ harmFlagsQmfBands[qmfBandDiv32] |= maskQmfBand; /* If there was a sine in the last frame, let it continue from the first envelope on else start at the transient position. Indexing of sineMapped starts relative to lsb. */ sineMapped[qmfBand - lsb] = (harmFlagsPrev[qmfBandDiv32] & maskQmfBand) ? 0 : tranEnv; if (sineMapped[qmfBand - lsb] < PVC_NTIMESLOT) { harmFlagsPrevActive[qmfBandDiv32] |= maskQmfBand; } } if (bitcount-- == 0) { bitcount = 31; curFlags++; } } FDKmemcpy(harmFlagsPrev, harmFlagsQmfBands, sizeof(ULONG) * ADD_HARMONICS_FLAGS_SIZE); } /*! \brief Restore sineMapped of previous frame For PVC it might happen that the PVC framing (always 0) is out of sync with the SBR framing. The adding of additional harmonics is done based on the SBR framing. If the SBR framing is trailing the PVC framing the sine mapping of the previous SBR frame needs to be used for the overlapping time slots. */ /*static*/ void mapSineFlagsPvc( UCHAR *freqBandTable, /*!< Band borders (there's only 1 flag per band) */ int nSfb, /*!< Number of bands in the table */ ULONG *harmFlagsPrev, /*!< Packed addHarmonics of previous frame (aligned to the MSB) */ ULONG *harmFlagsPrevActive, /*!< Packed sineMapped of previous frame (aligned to the LSB) */ SCHAR *sineMapped, /*!< Resulting vector of sine start positions for each QMF band */ int sinusoidalPos, /*!< sinusoidal position */ SCHAR *sinusoidalPosPrev, /*!< sinusoidal position of previous frame */ int trailingSbrFrame) /*!< indication if the SBR framing is trailing the PVC framing */ { /* Reset the output vector first */ FDKmemset(sineMapped, 32, MAX_FREQ_COEFFS); /* 32 means 'no sine' */ if (trailingSbrFrame) { /* restore sineMapped[] of previous frame */ int i; const int lsb = freqBandTable[0]; const int usb = freqBandTable[nSfb]; for (i = lsb; i < usb; i++) { const int qmfBandDiv32 = i >> 5; const int maskQmfBand = 1 << (i & 31); /* mask to extract harmonic flag from prevFlags */ /* Two cases need to be distinguished ... */ if (harmFlagsPrevActive[qmfBandDiv32] & maskQmfBand) { /* the sine mapping already started last PVC frame -> seamlessly * continue */ sineMapped[i - lsb] = 0; } else if (harmFlagsPrev[qmfBandDiv32] & maskQmfBand) { /* sinusoidalPos of prev PVC frame was >= PVC_NTIMESLOT -> sine starts * in this frame */ sineMapped[i - lsb] = *sinusoidalPosPrev - PVC_NTIMESLOT; /* we are 16 sbr time slots ahead of last frame now */ } } } *sinusoidalPosPrev = sinusoidalPos; } /*! \brief Reduce gain-adjustment induced aliasing for real valued filterbank. */ /*static*/ void aliasingReduction( FIXP_DBL *degreeAlias, /*!< estimated aliasing for each QMF channel */ ENV_CALC_NRGS *nrgs, UCHAR *useAliasReduction, /*!< synthetic sine energy for each subband, used as flag */ int noSubbands) /*!< number of QMF channels to process */ { FIXP_DBL *nrgGain = nrgs->nrgGain; /*!< subband gains to be modified */ SCHAR *nrgGain_e = nrgs->nrgGain_e; /*!< subband gains to be modified (exponents) */ FIXP_DBL *nrgEst = nrgs->nrgEst; /*!< subband energy before amplification */ SCHAR *nrgEst_e = nrgs->nrgEst_e; /*!< subband energy before amplification (exponents) */ int grouping = 0, index = 0, noGroups, k; int groupVector[MAX_FREQ_COEFFS]; /* Calculate grouping*/ for (k = 0; k < noSubbands - 1; k++) { if ((degreeAlias[k + 1] != FL2FXCONST_DBL(0.0f)) && useAliasReduction[k]) { if (grouping == 0) { groupVector[index++] = k; grouping = 1; } else { if (groupVector[index - 1] + 3 == k) { groupVector[index++] = k + 1; grouping = 0; } } } else { if (grouping) { if (useAliasReduction[k]) groupVector[index++] = k + 1; else groupVector[index++] = k; grouping = 0; } } } if (grouping) { groupVector[index++] = noSubbands; } noGroups = index >> 1; /*Calculate new gain*/ for (int group = 0; group < noGroups; group++) { FIXP_DBL nrgOrig = FL2FXCONST_DBL( 0.0f); /* Original signal energy in current group of bands */ SCHAR nrgOrig_e = 0; FIXP_DBL nrgAmp = FL2FXCONST_DBL( 0.0f); /* Amplified signal energy in group (using current gains) */ SCHAR nrgAmp_e = 0; FIXP_DBL nrgMod = FL2FXCONST_DBL( 0.0f); /* Signal energy in group when applying modified gains */ SCHAR nrgMod_e = 0; FIXP_DBL groupGain; /* Total energy gain in group */ SCHAR groupGain_e; FIXP_DBL compensation; /* Compensation factor for the energy change when applying modified gains */ SCHAR compensation_e; int startGroup = groupVector[2 * group]; int stopGroup = groupVector[2 * group + 1]; /* Calculate total energy in group before and after amplification with * current gains: */ for (k = startGroup; k < stopGroup; k++) { /* Get original band energy */ FIXP_DBL tmp = nrgEst[k]; SCHAR tmp_e = nrgEst_e[k]; FDK_add_MantExp(tmp, tmp_e, nrgOrig, nrgOrig_e, &nrgOrig, &nrgOrig_e); /* Multiply band energy with current gain */ tmp = fMult(tmp, nrgGain[k]); tmp_e = tmp_e + nrgGain_e[k]; FDK_add_MantExp(tmp, tmp_e, nrgAmp, nrgAmp_e, &nrgAmp, &nrgAmp_e); } /* Calculate total energy gain in group */ FDK_divide_MantExp(nrgAmp, nrgAmp_e, nrgOrig, nrgOrig_e, &groupGain, &groupGain_e); for (k = startGroup; k < stopGroup; k++) { FIXP_DBL tmp; SCHAR tmp_e; FIXP_DBL alpha = degreeAlias[k]; if (k < noSubbands - 1) { if (degreeAlias[k + 1] > alpha) alpha = degreeAlias[k + 1]; } /* Modify gain depending on the degree of aliasing */ FDK_add_MantExp( fMult(alpha, groupGain), groupGain_e, fMult(/*FL2FXCONST_DBL(1.0f)*/ (FIXP_DBL)MAXVAL_DBL - alpha, nrgGain[k]), nrgGain_e[k], &nrgGain[k], &nrgGain_e[k]); /* Apply modified gain to original energy */ tmp = fMult(nrgGain[k], nrgEst[k]); tmp_e = nrgGain_e[k] + nrgEst_e[k]; /* Accumulate energy with modified gains applied */ FDK_add_MantExp(tmp, tmp_e, nrgMod, nrgMod_e, &nrgMod, &nrgMod_e); } /* Calculate compensation factor to retain the energy of the amplified * signal */ FDK_divide_MantExp(nrgAmp, nrgAmp_e, nrgMod, nrgMod_e, &compensation, &compensation_e); /* Apply compensation factor to all gains of the group */ for (k = startGroup; k < stopGroup; k++) { nrgGain[k] = fMult(nrgGain[k], compensation); nrgGain_e[k] = nrgGain_e[k] + compensation_e; } } } #define INTER_TES_SF_CHANGE 4 typedef struct { FIXP_DBL subsample_power_low[(((1024) / (32) * (4) / 2) + (3 * (4)))]; FIXP_DBL subsample_power_high[(((1024) / (32) * (4) / 2) + (3 * (4)))]; FIXP_DBL gain[(((1024) / (32) * (4) / 2) + (3 * (4)))]; SCHAR subsample_power_low_sf[(((1024) / (32) * (4) / 2) + (3 * (4)))]; SCHAR subsample_power_high_sf[(((1024) / (32) * (4) / 2) + (3 * (4)))]; } ITES_TEMP; static void apply_inter_tes(FIXP_DBL **qmfReal, FIXP_DBL **qmfImag, const QMF_SCALE_FACTOR *sbrScaleFactor, const SCHAR exp[2], const int RATE, const int startPos, const int stopPos, const int lowSubband, const int nbSubband, const UCHAR gamma_idx) { int highSubband = lowSubband + nbSubband; FIXP_DBL *subsample_power_high, *subsample_power_low; SCHAR *subsample_power_high_sf, *subsample_power_low_sf; FIXP_DBL total_power_high = (FIXP_DBL)0; FIXP_DBL total_power_low = (FIXP_DBL)0; FIXP_DBL *gain; int gain_sf[(((1024) / (32) * (4) / 2) + (3 * (4)))]; /* gamma[gamma_idx] = {0.0f, 1.0f, 2.0f, 4.0f} */ int gamma_sf = (int)gamma_idx - 1; /* perhaps +1 to save one bit? (0.99999f vs 1.f) */ int nbSubsample = stopPos - startPos; int i, j; C_ALLOC_SCRATCH_START(pTmp, ITES_TEMP, 1); subsample_power_high = pTmp->subsample_power_high; subsample_power_low = pTmp->subsample_power_low; subsample_power_high_sf = pTmp->subsample_power_high_sf; subsample_power_low_sf = pTmp->subsample_power_low_sf; gain = pTmp->gain; if (gamma_idx > 0) { int preShift2 = 32 - fNormz((FIXP_DBL)nbSubsample); int total_power_low_sf = 1 - DFRACT_BITS; int total_power_high_sf = 1 - DFRACT_BITS; for (i = 0; i < nbSubsample; ++i) { FIXP_DBL bufferReal[(((1024) / (32) * (4) / 2) + (3 * (4)))]; FIXP_DBL bufferImag[(((1024) / (32) * (4) / 2) + (3 * (4)))]; FIXP_DBL maxVal = (FIXP_DBL)0; int ts = startPos + i; int low_sf = (ts < 3 * RATE) ? sbrScaleFactor->ov_lb_scale : sbrScaleFactor->lb_scale; low_sf = 15 - low_sf; for (j = 0; j < lowSubband; ++j) { bufferImag[j] = qmfImag[startPos + i][j]; maxVal |= (FIXP_DBL)((LONG)(bufferImag[j]) ^ ((LONG)bufferImag[j] >> (DFRACT_BITS - 1))); bufferReal[j] = qmfReal[startPos + i][j]; maxVal |= (FIXP_DBL)((LONG)(bufferReal[j]) ^ ((LONG)bufferReal[j] >> (DFRACT_BITS - 1))); } subsample_power_low[i] = (FIXP_DBL)0; subsample_power_low_sf[i] = 0; if (maxVal != FL2FXCONST_DBL(0.f)) { /* multiply first, then shift for safe summation */ int preShift = 1 - CntLeadingZeros(maxVal); int postShift = 32 - fNormz((FIXP_DBL)lowSubband); /* reduce preShift because otherwise we risk to square -1.f */ if (preShift != 0) preShift++; subsample_power_low_sf[i] += (low_sf + preShift) * 2 + postShift + 1; scaleValues(bufferReal, lowSubband, -preShift); scaleValues(bufferImag, lowSubband, -preShift); for (j = 0; j < lowSubband; ++j) { FIXP_DBL addme; addme = fPow2Div2(bufferReal[j]); subsample_power_low[i] += addme >> postShift; addme = fPow2Div2(bufferImag[j]); subsample_power_low[i] += addme >> postShift; } } /* now get high */ maxVal = (FIXP_DBL)0; int high_sf = exp[(ts < 16 * RATE) ? 0 : 1]; for (j = lowSubband; j < highSubband; ++j) { bufferImag[j] = qmfImag[startPos + i][j]; maxVal |= (FIXP_DBL)((LONG)(bufferImag[j]) ^ ((LONG)bufferImag[j] >> (DFRACT_BITS - 1))); bufferReal[j] = qmfReal[startPos + i][j]; maxVal |= (FIXP_DBL)((LONG)(bufferReal[j]) ^ ((LONG)bufferReal[j] >> (DFRACT_BITS - 1))); } subsample_power_high[i] = (FIXP_DBL)0; subsample_power_high_sf[i] = 0; if (maxVal != FL2FXCONST_DBL(0.f)) { int preShift = 1 - CntLeadingZeros(maxVal); /* reduce preShift because otherwise we risk to square -1.f */ if (preShift != 0) preShift++; int postShift = 32 - fNormz((FIXP_DBL)(highSubband - lowSubband)); subsample_power_high_sf[i] += (high_sf + preShift) * 2 + postShift + 1; scaleValues(&bufferReal[lowSubband], highSubband - lowSubband, -preShift); scaleValues(&bufferImag[lowSubband], highSubband - lowSubband, -preShift); for (j = lowSubband; j < highSubband; j++) { subsample_power_high[i] += fPow2Div2(bufferReal[j]) >> postShift; subsample_power_high[i] += fPow2Div2(bufferImag[j]) >> postShift; } } /* sum all together */ FIXP_DBL new_summand = subsample_power_low[i]; int new_summand_sf = subsample_power_low_sf[i]; /* make sure the current sum, and the new summand have the same SF */ if (new_summand_sf > total_power_low_sf) { int diff = fMin(DFRACT_BITS - 1, new_summand_sf - total_power_low_sf); total_power_low >>= diff; total_power_low_sf = new_summand_sf; } else if (new_summand_sf < total_power_low_sf) { new_summand >>= fMin(DFRACT_BITS - 1, total_power_low_sf - new_summand_sf); } total_power_low += (new_summand >> preShift2); new_summand = subsample_power_high[i]; new_summand_sf = subsample_power_high_sf[i]; if (new_summand_sf > total_power_high_sf) { total_power_high >>= fMin(DFRACT_BITS - 1, new_summand_sf - total_power_high_sf); total_power_high_sf = new_summand_sf; } else if (new_summand_sf < total_power_high_sf) { new_summand >>= fMin(DFRACT_BITS - 1, total_power_high_sf - new_summand_sf); } total_power_high += (new_summand >> preShift2); } total_power_low_sf += preShift2; total_power_high_sf += preShift2; /* gain[i] = e_LOW[i] */ for (i = 0; i < nbSubsample; ++i) { int sf2; FIXP_DBL mult = fMultNorm(subsample_power_low[i], (FIXP_DBL)nbSubsample, &sf2); int mult_sf = subsample_power_low_sf[i] + DFRACT_BITS - 1 + sf2; if (total_power_low != FIXP_DBL(0)) { gain[i] = fDivNorm(mult, total_power_low, &sf2); gain_sf[i] = mult_sf - total_power_low_sf + sf2; gain[i] = sqrtFixp_lookup(gain[i], &gain_sf[i]); if (gain_sf[i] < 0) { gain[i] >>= -gain_sf[i]; gain_sf[i] = 0; } } else { if (mult == FIXP_DBL(0)) { gain[i] = FIXP_DBL(0); gain_sf[i] = 0; } else { gain[i] = (FIXP_DBL)MAXVAL_DBL; gain_sf[i] = 0; } } } FIXP_DBL total_power_high_after = (FIXP_DBL)0; int total_power_high_after_sf = 1 - DFRACT_BITS; /* gain[i] = g_inter[i] */ for (i = 0; i < nbSubsample; ++i) { if (gain_sf[i] < 0) { gain[i] >>= -gain_sf[i]; gain_sf[i] = 0; } /* calculate: gain[i] = 1.0f + gamma * (gain[i] - 1.0f); */ FIXP_DBL one = (FIXP_DBL)MAXVAL_DBL >> gain_sf[i]; /* to substract this from gain[i] */ /* gamma is actually always 1 according to the table, so skip the * fMultDiv2 */ FIXP_DBL mult = (gain[i] - one) >> 1; int mult_sf = gain_sf[i] + gamma_sf; one = FL2FXCONST_DBL(0.5f) >> mult_sf; gain[i] = one + mult; gain_sf[i] += gamma_sf + 1; /* +1 because of fMultDiv2() */ /* set gain to at least 0.2f */ /* limit and calculate gain[i]^2 too */ FIXP_DBL gain_pow2; int gain_pow2_sf; if (fIsLessThan(gain[i], gain_sf[i], FL2FXCONST_DBL(0.2f), 0)) { gain[i] = FL2FXCONST_DBL(0.8f); gain_sf[i] = -2; gain_pow2 = FL2FXCONST_DBL(0.64f); gain_pow2_sf = -4; } else { /* this upscaling seems quite important */ int r = CountLeadingBits(gain[i]); gain[i] <<= r; gain_sf[i] -= r; gain_pow2 = fPow2(gain[i]); gain_pow2_sf = gain_sf[i] << 1; } int room; subsample_power_high[i] = fMultNorm(subsample_power_high[i], gain_pow2, &room); subsample_power_high_sf[i] = subsample_power_high_sf[i] + gain_pow2_sf + room; int new_summand_sf = subsample_power_high_sf[i]; /* + gain_pow2_sf; */ if (new_summand_sf > total_power_high_after_sf) { total_power_high_after >>= fMin(DFRACT_BITS - 1, new_summand_sf - total_power_high_after_sf); total_power_high_after_sf = new_summand_sf; } else if (new_summand_sf < total_power_high_after_sf) { subsample_power_high[i] >>= fMin(DFRACT_BITS - 1, total_power_high_after_sf - new_summand_sf); } total_power_high_after += subsample_power_high[i] >> preShift2; } total_power_high_after_sf += preShift2; int sf2 = 0; FIXP_DBL gain_adj_2 = FL2FX_DBL(0.5f); int gain_adj_2_sf = 1; if ((total_power_high != (FIXP_DBL)0) && (total_power_high_after != (FIXP_DBL)0)) { gain_adj_2 = fDivNorm(total_power_high, total_power_high_after, &sf2); gain_adj_2_sf = total_power_high_sf - total_power_high_after_sf + sf2; } FIXP_DBL gain_adj = sqrtFixp_lookup(gain_adj_2, &gain_adj_2_sf); int gain_adj_sf = gain_adj_2_sf; for (i = 0; i < nbSubsample; ++i) { gain[i] = fMult(gain[i], gain_adj); gain_sf[i] += gain_adj_sf; /* limit gain */ if (gain_sf[i] > INTER_TES_SF_CHANGE) { gain[i] = (FIXP_DBL)MAXVAL_DBL; gain_sf[i] = INTER_TES_SF_CHANGE; } } for (i = 0; i < nbSubsample; ++i) { /* equalize gain[]'s scale factors */ gain[i] >>= INTER_TES_SF_CHANGE - gain_sf[i]; for (j = lowSubband; j < highSubband; j++) { qmfReal[startPos + i][j] = fMult(qmfReal[startPos + i][j], gain[i]); qmfImag[startPos + i][j] = fMult(qmfImag[startPos + i][j], gain[i]); } } } else { /* gamma_idx == 0 */ /* Inter-TES is not active. Still perform the scale change to have a * consistent scaling for all envelopes of this frame. */ for (i = 0; i < nbSubsample; ++i) { for (j = lowSubband; j < highSubband; j++) { qmfReal[startPos + i][j] >>= INTER_TES_SF_CHANGE; qmfImag[startPos + i][j] >>= INTER_TES_SF_CHANGE; } } } C_ALLOC_SCRATCH_END(pTmp, ITES_TEMP, 1); } /*! \brief Apply spectral envelope to subband samples This function is called from sbr_dec.cpp in each frame. To enhance accuracy and due to the usage of tables for squareroots and inverse, some calculations are performed with the operands being split into mantissa and exponent. The variable names in the source code carry the suffixes _m and _e respectively. The control data in #hFrameData containts envelope data which is represented by this format but stored in single words. (See requantizeEnvelopeData() for details). This data is unpacked within calculateSbrEnvelope() to follow the described suffix convention. The actual value (comparable to the corresponding float-variable in the research-implementation) of a mantissa/exponent-pair can be calculated as \f$ value = value\_m * 2^{value\_e} \f$ All energies and noise levels decoded from the bitstream suit for an original signal magnitude of \f$\pm 32768 \f$ rather than \f$ \pm 1\f$. Therefore, the scale factor hb_scale passed into this function will be converted to an 'input exponent' (#input_e), which fits the internal representation. Before the actual processing, an exponent #adj_e for resulting adjusted samples is derived from the maximum reference energy. Then, for each envelope, the following steps are performed: \li Calculate energy in the signal to be adjusted. Depending on the the value of #interpolFreq (interpolation mode), this is either done seperately for each QMF-subband or for each SBR-band. The resulting energies are stored in #nrgEst_m[#MAX_FREQ_COEFFS] (mantissas) and #nrgEst_e[#MAX_FREQ_COEFFS] (exponents). \li Calculate gain and noise level for each subband:
\f$ gain = \sqrt{ \frac{nrgRef}{nrgEst} \cdot (1 - noiseRatio) } \hspace{2cm} noise = \sqrt{ nrgRef \cdot noiseRatio } \f$
where noiseRatio and nrgRef are extracted from the bitstream and nrgEst is the subband energy before adjustment. The resulting gains are stored in #nrgGain_m[#MAX_FREQ_COEFFS] (mantissas) and #nrgGain_e[#MAX_FREQ_COEFFS] (exponents), the noise levels are stored in #noiseLevel_m[#MAX_FREQ_COEFFS] and #noiseLevel_e[#MAX_FREQ_COEFFS] (exponents). The sine levels are stored in #nrgSine_m[#MAX_FREQ_COEFFS] and #nrgSine_e[#MAX_FREQ_COEFFS]. \li Noise limiting: The gain for each subband is limited both absolutely and relatively compared to the total gain over all subbands. \li Boost gain: Calculate and apply boost factor for each limiter band in order to compensate for the energy loss imposed by the limiting. \li Apply gains and add noise: The gains and noise levels are applied to all timeslots of the current envelope. A short FIR-filter (length 4 QMF-timeslots) can be used to smooth the sudden change at the envelope borders. Each complex subband sample of the current timeslot is multiplied by the smoothed gain, then random noise with the calculated level is added. \note To reduce the stack size, some of the local arrays could be located within the time output buffer. Of the 512 samples temporarily available there, about half the size is already used by #SBR_FRAME_DATA. A pointer to the remaining free memory could be supplied by an additional argument to calculateSbrEnvelope() in sbr_dec: \par \code calculateSbrEnvelope (&hSbrDec->sbrScaleFactor, &hSbrDec->SbrCalculateEnvelope, hHeaderData, hFrameData, QmfBufferReal, QmfBufferImag, timeOutPtr + sizeof(SBR_FRAME_DATA)/sizeof(Float) + 1); \endcode \par Within calculateSbrEnvelope(), some pointers could be defined instead of the arrays #nrgRef_m, #nrgRef_e, #nrgEst_m, #nrgEst_e, #noiseLevel_m: \par \code fract* nrgRef_m = timeOutPtr; SCHAR* nrgRef_e = nrgRef_m + MAX_FREQ_COEFFS; fract* nrgEst_m = nrgRef_e + MAX_FREQ_COEFFS; SCHAR* nrgEst_e = nrgEst_m + MAX_FREQ_COEFFS; fract* noiseLevel_m = nrgEst_e + MAX_FREQ_COEFFS; \endcode
*/ void calculateSbrEnvelope( QMF_SCALE_FACTOR *sbrScaleFactor, /*!< Scaling factors */ HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env, /*!< Handle to struct filled by the create-function */ HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ HANDLE_SBR_FRAME_DATA hFrameData, /*!< Control data of current frame */ PVC_DYNAMIC_DATA *pPvcDynamicData, FIXP_DBL * *analysBufferReal, /*!< Real part of subband samples to be processed */ FIXP_DBL * *analysBufferImag, /*!< Imag part of subband samples to be processed */ const int useLP, FIXP_DBL *degreeAlias, /*!< Estimated aliasing for each QMF channel */ const UINT flags, const int frameErrorFlag) { int c, i, i_stop, j, envNoise = 0; UCHAR *borders = hFrameData->frameInfo.borders; UCHAR *bordersPvc = hFrameData->frameInfo.pvcBorders; int pvc_mode = pPvcDynamicData->pvc_mode; int first_start = ((pvc_mode > 0) ? bordersPvc[0] : borders[0]) * hHeaderData->timeStep; FIXP_SGL *noiseLevels = hFrameData->sbrNoiseFloorLevel; HANDLE_FREQ_BAND_DATA hFreq = &hHeaderData->freqBandData; UCHAR **pFreqBandTable = hFreq->freqBandTable; UCHAR *pFreqBandTableNoise = hFreq->freqBandTableNoise; int lowSubband = hFreq->lowSubband; int highSubband = hFreq->highSubband; int noSubbands = highSubband - lowSubband; /* old high subband before headerchange we asume no headerchange here */ int ov_highSubband = hFreq->highSubband; int noNoiseBands = hFreq->nNfb; UCHAR *noSubFrameBands = hFreq->nSfb; int no_cols = hHeaderData->numberTimeSlots * hHeaderData->timeStep; SCHAR sineMapped[MAX_FREQ_COEFFS]; SCHAR ov_adj_e = SCALE2EXP(sbrScaleFactor->ov_hb_scale); SCHAR adj_e = 0; SCHAR output_e; SCHAR final_e = 0; /* inter-TES is active in one or more envelopes of the current SBR frame */ const int iTES_enable = hFrameData->iTESactive; const int iTES_scale_change = (iTES_enable) ? INTER_TES_SF_CHANGE : 0; SCHAR maxGainLimit_e = (frameErrorFlag) ? MAX_GAIN_CONCEAL_EXP : MAX_GAIN_EXP; UCHAR smooth_length = 0; FIXP_SGL *pIenv = hFrameData->iEnvelope; C_ALLOC_SCRATCH_START(useAliasReduction, UCHAR, 64) /* if values differ we had a headerchange; if old highband is bigger then new one we need to patch overlap-highband-scaling for this frame (see use of ov_highSubband) as overlap contains higher frequency components which would get lost */ if (hFreq->highSubband < hFreq->ov_highSubband) { ov_highSubband = hFreq->ov_highSubband; } if (pvc_mode > 0) { if (hFrameData->frameInfo.bordersNoise[0] > bordersPvc[0]) { /* noise envelope of previous frame is trailing into current PVC frame */ envNoise = -1; noiseLevels = h_sbr_cal_env->prevSbrNoiseFloorLevel; noNoiseBands = h_sbr_cal_env->prevNNfb; noSubFrameBands = h_sbr_cal_env->prevNSfb; lowSubband = h_sbr_cal_env->prevLoSubband; highSubband = h_sbr_cal_env->prevHiSubband; noSubbands = highSubband - lowSubband; ov_highSubband = highSubband; if (highSubband < h_sbr_cal_env->prev_ov_highSubband) { ov_highSubband = h_sbr_cal_env->prev_ov_highSubband; } pFreqBandTable[0] = h_sbr_cal_env->prevFreqBandTableLo; pFreqBandTable[1] = h_sbr_cal_env->prevFreqBandTableHi; pFreqBandTableNoise = h_sbr_cal_env->prevFreqBandTableNoise; } mapSineFlagsPvc(pFreqBandTable[1], noSubFrameBands[1], h_sbr_cal_env->harmFlagsPrev, h_sbr_cal_env->harmFlagsPrevActive, sineMapped, hFrameData->sinusoidal_position, &h_sbr_cal_env->sinusoidal_positionPrev, (borders[0] > bordersPvc[0]) ? 1 : 0); } else { /* Extract sine flags for all QMF bands */ mapSineFlags(pFreqBandTable[1], noSubFrameBands[1], hFrameData->addHarmonics, h_sbr_cal_env->harmFlagsPrev, h_sbr_cal_env->harmFlagsPrevActive, hFrameData->frameInfo.tranEnv, sineMapped); } /* Scan for maximum in bufferd noise levels. This is needed in case that we had strong noise in the previous frame which is smoothed into the current frame. The resulting exponent is used as start value for the maximum search in reference energies */ if (!useLP) adj_e = h_sbr_cal_env->filtBufferNoise_e - getScalefactor(h_sbr_cal_env->filtBufferNoise, noSubbands) + (INT)MAX_SFB_NRG_HEADROOM; /* Scan for maximum reference energy to be able to select appropriate values for adj_e and final_e. */ if (pvc_mode > 0) { INT maxSfbNrg_e = pPvcDynamicData->predEsg_expMax; /* Energy -> magnitude (sqrt halfens exponent) */ maxSfbNrg_e = (maxSfbNrg_e + 1) >> 1; /* +1 to go safe (round to next higher int) */ /* Some safety margin is needed for 2 reasons: - The signal energy is not equally spread over all subband samples in a specific sfb of an envelope (Nrg could be too high by a factor of envWidth * sfbWidth) - Smoothing can smear high gains of the previous envelope into the current */ maxSfbNrg_e += (6 + MAX_SFB_NRG_HEADROOM); adj_e = maxSfbNrg_e; // final_e should not exist for PVC fixfix framing } else { for (i = 0; i < hFrameData->frameInfo.nEnvelopes; i++) { INT maxSfbNrg_e = -FRACT_BITS + NRG_EXP_OFFSET; /* start value for maximum search */ /* Fetch frequency resolution for current envelope: */ for (j = noSubFrameBands[hFrameData->frameInfo.freqRes[i]]; j != 0; j--) { maxSfbNrg_e = fixMax(maxSfbNrg_e, (INT)((LONG)(*pIenv++) & MASK_E)); } maxSfbNrg_e -= NRG_EXP_OFFSET; /* Energy -> magnitude (sqrt halfens exponent) */ maxSfbNrg_e = (maxSfbNrg_e + 1) >> 1; /* +1 to go safe (round to next higher int) */ /* Some safety margin is needed for 2 reasons: - The signal energy is not equally spread over all subband samples in a specific sfb of an envelope (Nrg could be too high by a factor of envWidth * sfbWidth) - Smoothing can smear high gains of the previous envelope into the current */ maxSfbNrg_e += (6 + MAX_SFB_NRG_HEADROOM); if (borders[i] < hHeaderData->numberTimeSlots) /* This envelope affects timeslots that belong to the output frame */ adj_e = fMax(maxSfbNrg_e, adj_e); if (borders[i + 1] > hHeaderData->numberTimeSlots) /* This envelope affects timeslots after the output frame */ final_e = fMax(maxSfbNrg_e, final_e); } } /* Calculate adjustment factors and apply them for every envelope. */ pIenv = hFrameData->iEnvelope; if (pvc_mode > 0) { /* iterate over SBR time slots starting with bordersPvc[i] */ i = bordersPvc[0]; /* usually 0; can be >0 if switching from legacy SBR to PVC */ i_stop = PVC_NTIMESLOT; FDK_ASSERT(bordersPvc[hFrameData->frameInfo.nEnvelopes] == PVC_NTIMESLOT); } else { /* iterate over SBR envelopes starting with 0 */ i = 0; i_stop = hFrameData->frameInfo.nEnvelopes; } for (; i < i_stop; i++) { int k, noNoiseFlag; SCHAR noise_e, input_e = SCALE2EXP(sbrScaleFactor->hb_scale); C_ALLOC_SCRATCH_START(pNrgs, ENV_CALC_NRGS, 1); /* Helper variables. */ int start_pos, stop_pos, freq_res; if (pvc_mode > 0) { start_pos = hHeaderData->timeStep * i; /* Start-position in time (subband sample) for current envelope. */ stop_pos = hHeaderData->timeStep * (i + 1); /* Stop-position in time (subband sample) for current envelope. */ freq_res = hFrameData->frameInfo .freqRes[0]; /* Frequency resolution for current envelope. */ FDK_ASSERT( freq_res == hFrameData->frameInfo.freqRes[hFrameData->frameInfo.nEnvelopes - 1]); } else { start_pos = hHeaderData->timeStep * borders[i]; /* Start-position in time (subband sample) for current envelope. */ stop_pos = hHeaderData->timeStep * borders[i + 1]; /* Stop-position in time (subband sample) for current envelope. */ freq_res = hFrameData->frameInfo .freqRes[i]; /* Frequency resolution for current envelope. */ } /* Always fully initialize the temporary energy table. This prevents negative energies and extreme gain factors in cases where the number of limiter bands exceeds the number of subbands. The latter can be caused by undetected bit errors and is tested by some streams from the certification set. */ FDKmemclear(pNrgs, sizeof(ENV_CALC_NRGS)); if (pvc_mode > 0) { /* get predicted energy values from PVC module */ expandPredEsg(pPvcDynamicData, i, (int)MAX_FREQ_COEFFS, pNrgs->nrgRef, pNrgs->nrgRef_e); if (i == borders[0]) { mapSineFlags(pFreqBandTable[1], noSubFrameBands[1], hFrameData->addHarmonics, h_sbr_cal_env->harmFlagsPrev, h_sbr_cal_env->harmFlagsPrevActive, hFrameData->sinusoidal_position, sineMapped); } if (i >= hFrameData->frameInfo.bordersNoise[envNoise + 1]) { if (envNoise >= 0) { noiseLevels += noNoiseBands; /* The noise floor data is stored in a row [noiseFloor1 noiseFloor2...].*/ } else { /* leave trailing noise envelope of past frame */ noNoiseBands = hFreq->nNfb; noSubFrameBands = hFreq->nSfb; noiseLevels = hFrameData->sbrNoiseFloorLevel; lowSubband = hFreq->lowSubband; highSubband = hFreq->highSubband; noSubbands = highSubband - lowSubband; ov_highSubband = highSubband; if (highSubband < hFreq->ov_highSubband) { ov_highSubband = hFreq->ov_highSubband; } pFreqBandTable[0] = hFreq->freqBandTableLo; pFreqBandTable[1] = hFreq->freqBandTableHi; pFreqBandTableNoise = hFreq->freqBandTableNoise; } envNoise++; } } else { /* If the start-pos of the current envelope equals the stop pos of the current noise envelope, increase the pointer (i.e. choose the next noise-floor).*/ if (borders[i] == hFrameData->frameInfo.bordersNoise[envNoise + 1]) { noiseLevels += noNoiseBands; /* The noise floor data is stored in a row [noiseFloor1 noiseFloor2...].*/ envNoise++; } } if (i == hFrameData->frameInfo.tranEnv || i == h_sbr_cal_env->prevTranEnv) /* attack */ { noNoiseFlag = 1; if (!useLP) smooth_length = 0; /* No smoothing on attacks! */ } else { noNoiseFlag = 0; if (!useLP) smooth_length = (1 - hHeaderData->bs_data.smoothingLength) << 2; /* can become either 0 or 4 */ } /* Energy estimation in transposed highband. */ if (hHeaderData->bs_data.interpolFreq) calcNrgPerSubband(analysBufferReal, (useLP) ? NULL : analysBufferImag, lowSubband, highSubband, start_pos, stop_pos, input_e, pNrgs->nrgEst, pNrgs->nrgEst_e); else calcNrgPerSfb(analysBufferReal, (useLP) ? NULL : analysBufferImag, noSubFrameBands[freq_res], pFreqBandTable[freq_res], start_pos, stop_pos, input_e, pNrgs->nrgEst, pNrgs->nrgEst_e); /* Calculate subband gains */ { UCHAR *table = pFreqBandTable[freq_res]; UCHAR *pUiNoise = &pFreqBandTableNoise[1]; /*! Upper limit of the current noise floor band. */ FIXP_SGL *pNoiseLevels = noiseLevels; FIXP_DBL tmpNoise = FX_SGL2FX_DBL((FIXP_SGL)((LONG)(*pNoiseLevels) & MASK_M)); SCHAR tmpNoise_e = (UCHAR)((LONG)(*pNoiseLevels++) & MASK_E) - NOISE_EXP_OFFSET; int cc = 0; c = 0; if (pvc_mode > 0) { for (j = 0; j < noSubFrameBands[freq_res]; j++) { UCHAR sinePresentFlag = 0; int li = table[j]; int ui = table[j + 1]; for (k = li; k < ui; k++) { sinePresentFlag |= (i >= sineMapped[cc]); cc++; } for (k = li; k < ui; k++) { FIXP_DBL refNrg = pNrgs->nrgRef[k - lowSubband]; SCHAR refNrg_e = pNrgs->nrgRef_e[k - lowSubband]; if (k >= *pUiNoise) { tmpNoise = FX_SGL2FX_DBL((FIXP_SGL)((LONG)(*pNoiseLevels) & MASK_M)); tmpNoise_e = (SCHAR)((LONG)(*pNoiseLevels++) & MASK_E) - NOISE_EXP_OFFSET; pUiNoise++; } FDK_ASSERT(k >= lowSubband); if (useLP) useAliasReduction[k - lowSubband] = !sinePresentFlag; pNrgs->nrgSine[c] = FL2FXCONST_DBL(0.0f); pNrgs->nrgSine_e[c] = 0; calcSubbandGain(refNrg, refNrg_e, pNrgs, c, tmpNoise, tmpNoise_e, sinePresentFlag, i >= sineMapped[c], noNoiseFlag); c++; } } } else { for (j = 0; j < noSubFrameBands[freq_res]; j++) { FIXP_DBL refNrg = FX_SGL2FX_DBL((FIXP_SGL)((LONG)(*pIenv) & MASK_M)); SCHAR refNrg_e = (SCHAR)((LONG)(*pIenv) & MASK_E) - NRG_EXP_OFFSET; UCHAR sinePresentFlag = 0; int li = table[j]; int ui = table[j + 1]; for (k = li; k < ui; k++) { sinePresentFlag |= (i >= sineMapped[cc]); cc++; } for (k = li; k < ui; k++) { if (k >= *pUiNoise) { tmpNoise = FX_SGL2FX_DBL((FIXP_SGL)((LONG)(*pNoiseLevels) & MASK_M)); tmpNoise_e = (SCHAR)((LONG)(*pNoiseLevels++) & MASK_E) - NOISE_EXP_OFFSET; pUiNoise++; } FDK_ASSERT(k >= lowSubband); if (useLP) useAliasReduction[k - lowSubband] = !sinePresentFlag; pNrgs->nrgSine[c] = FL2FXCONST_DBL(0.0f); pNrgs->nrgSine_e[c] = 0; calcSubbandGain(refNrg, refNrg_e, pNrgs, c, tmpNoise, tmpNoise_e, sinePresentFlag, i >= sineMapped[c], noNoiseFlag); pNrgs->nrgRef[c] = refNrg; pNrgs->nrgRef_e[c] = refNrg_e; c++; } pIenv++; } } } /* Noise limiting */ for (c = 0; c < hFreq->noLimiterBands; c++) { FIXP_DBL sumRef, boostGain, maxGain; FIXP_DBL accu = FL2FXCONST_DBL(0.0f); SCHAR sumRef_e, boostGain_e, maxGain_e, accu_e = 0; int maxGainLimGainSum_e = 0; calcAvgGain(pNrgs, hFreq->limiterBandTable[c], hFreq->limiterBandTable[c + 1], &sumRef, &sumRef_e, &maxGain, &maxGain_e); /* Multiply maxGain with limiterGain: */ maxGain = fMult( maxGain, FDK_sbrDecoder_sbr_limGains_m[hHeaderData->bs_data.limiterGains]); /* maxGain_e += * FDK_sbrDecoder_sbr_limGains_e[hHeaderData->bs_data.limiterGains]; */ /* The addition of maxGain_e and FDK_sbrDecoder_sbr_limGains_e[3] might yield values greater than 127 which doesn't fit into an SCHAR! In these rare situations limit maxGain_e to 127. */ maxGainLimGainSum_e = maxGain_e + FDK_sbrDecoder_sbr_limGains_e[hHeaderData->bs_data.limiterGains]; maxGain_e = (maxGainLimGainSum_e > 127) ? (SCHAR)127 : (SCHAR)maxGainLimGainSum_e; /* Scale mantissa of MaxGain into range between 0.5 and 1: */ if (maxGain == FL2FXCONST_DBL(0.0f)) maxGain_e = -FRACT_BITS; else { SCHAR charTemp = CountLeadingBits(maxGain); maxGain_e -= charTemp; maxGain <<= (int)charTemp; } if (maxGain_e >= maxGainLimit_e) { /* upper limit (e.g. 96 dB) */ maxGain = FL2FXCONST_DBL(0.5f); maxGain_e = maxGainLimit_e; } /* Every subband gain is compared to the scaled "average gain" and limited if necessary: */ for (k = hFreq->limiterBandTable[c]; k < hFreq->limiterBandTable[c + 1]; k++) { if ((pNrgs->nrgGain_e[k] > maxGain_e) || (pNrgs->nrgGain_e[k] == maxGain_e && pNrgs->nrgGain[k] > maxGain)) { FIXP_DBL noiseAmp; SCHAR noiseAmp_e; FDK_divide_MantExp(maxGain, maxGain_e, pNrgs->nrgGain[k], pNrgs->nrgGain_e[k], &noiseAmp, &noiseAmp_e); pNrgs->noiseLevel[k] = fMult(pNrgs->noiseLevel[k], noiseAmp); pNrgs->noiseLevel_e[k] += noiseAmp_e; pNrgs->nrgGain[k] = maxGain; pNrgs->nrgGain_e[k] = maxGain_e; } } /* -- Boost gain Calculate and apply boost factor for each limiter band: 1. Check how much energy would be present when using the limited gain 2. Calculate boost factor by comparison with reference energy 3. Apply boost factor to compensate for the energy loss due to limiting */ for (k = hFreq->limiterBandTable[c]; k < hFreq->limiterBandTable[c + 1]; k++) { /* 1.a Add energy of adjusted signal (using preliminary gain) */ FIXP_DBL tmp = fMult(pNrgs->nrgGain[k], pNrgs->nrgEst[k]); SCHAR tmp_e = pNrgs->nrgGain_e[k] + pNrgs->nrgEst_e[k]; FDK_add_MantExp(tmp, tmp_e, accu, accu_e, &accu, &accu_e); /* 1.b Add sine energy (if present) */ if (pNrgs->nrgSine[k] != FL2FXCONST_DBL(0.0f)) { FDK_add_MantExp(pNrgs->nrgSine[k], pNrgs->nrgSine_e[k], accu, accu_e, &accu, &accu_e); } else { /* 1.c Add noise energy (if present) */ if (noNoiseFlag == 0) { FDK_add_MantExp(pNrgs->noiseLevel[k], pNrgs->noiseLevel_e[k], accu, accu_e, &accu, &accu_e); } } } /* 2.a Calculate ratio of wanted energy and accumulated energy */ if (accu == (FIXP_DBL)0) { /* If divisor is 0, limit quotient to +4 dB */ boostGain = FL2FXCONST_DBL(0.6279716f); boostGain_e = 2; } else { INT div_e; boostGain = fDivNorm(sumRef, accu, &div_e); boostGain_e = sumRef_e - accu_e + div_e; } /* 2.b Result too high? --> Limit the boost factor to +4 dB */ if ((boostGain_e > 3) || (boostGain_e == 2 && boostGain > FL2FXCONST_DBL(0.6279716f)) || (boostGain_e == 3 && boostGain > FL2FXCONST_DBL(0.3139858f))) { boostGain = FL2FXCONST_DBL(0.6279716f); boostGain_e = 2; } /* 3. Multiply all signal components with the boost factor */ for (k = hFreq->limiterBandTable[c]; k < hFreq->limiterBandTable[c + 1]; k++) { pNrgs->nrgGain[k] = fMultDiv2(pNrgs->nrgGain[k], boostGain); pNrgs->nrgGain_e[k] = pNrgs->nrgGain_e[k] + boostGain_e + 1; pNrgs->nrgSine[k] = fMultDiv2(pNrgs->nrgSine[k], boostGain); pNrgs->nrgSine_e[k] = pNrgs->nrgSine_e[k] + boostGain_e + 1; pNrgs->noiseLevel[k] = fMultDiv2(pNrgs->noiseLevel[k], boostGain); pNrgs->noiseLevel_e[k] = pNrgs->noiseLevel_e[k] + boostGain_e + 1; } } /* End of noise limiting */ if (useLP) aliasingReduction(degreeAlias + lowSubband, pNrgs, useAliasReduction, noSubbands); /* For the timeslots within the range for the output frame, use the same scale for the noise levels. Drawback: If the envelope exceeds the frame border, the noise levels will have to be rescaled later to fit final_e of the gain-values. */ noise_e = (start_pos < no_cols) ? adj_e : final_e; /* Convert energies to amplitude levels */ for (k = 0; k < noSubbands; k++) { FDK_sqrt_MantExp(&pNrgs->nrgSine[k], &pNrgs->nrgSine_e[k], &noise_e); FDK_sqrt_MantExp(&pNrgs->nrgGain[k], &pNrgs->nrgGain_e[k], &pNrgs->nrgGain_e[k]); FDK_sqrt_MantExp(&pNrgs->noiseLevel[k], &pNrgs->noiseLevel_e[k], &noise_e); } /* Apply calculated gains and adaptive noise */ /* assembleHfSignals() */ { int scale_change, sc_change; FIXP_SGL smooth_ratio; int filtBufferNoiseShift = 0; /* Initialize smoothing buffers with the first valid values */ if (h_sbr_cal_env->startUp) { if (!useLP) { h_sbr_cal_env->filtBufferNoise_e = noise_e; FDKmemcpy(h_sbr_cal_env->filtBuffer_e, pNrgs->nrgGain_e, noSubbands * sizeof(SCHAR)); FDKmemcpy(h_sbr_cal_env->filtBufferNoise, pNrgs->noiseLevel, noSubbands * sizeof(FIXP_DBL)); FDKmemcpy(h_sbr_cal_env->filtBuffer, pNrgs->nrgGain, noSubbands * sizeof(FIXP_DBL)); } h_sbr_cal_env->startUp = 0; } if (!useLP) { equalizeFiltBufferExp(h_sbr_cal_env->filtBuffer, /* buffered */ h_sbr_cal_env->filtBuffer_e, /* buffered */ pNrgs->nrgGain, /* current */ pNrgs->nrgGain_e, /* current */ noSubbands); /* Adapt exponent of buffered noise levels to the current exponent so they can easily be smoothed */ if ((h_sbr_cal_env->filtBufferNoise_e - noise_e) >= 0) { int shift = fixMin(DFRACT_BITS - 1, (int)(h_sbr_cal_env->filtBufferNoise_e - noise_e)); for (k = 0; k < noSubbands; k++) h_sbr_cal_env->filtBufferNoise[k] <<= shift; } else { int shift = fixMin(DFRACT_BITS - 1, -(int)(h_sbr_cal_env->filtBufferNoise_e - noise_e)); for (k = 0; k < noSubbands; k++) h_sbr_cal_env->filtBufferNoise[k] >>= shift; } h_sbr_cal_env->filtBufferNoise_e = noise_e; } /* find best scaling! */ scale_change = -(DFRACT_BITS - 1); for (k = 0; k < noSubbands; k++) { scale_change = fixMax(scale_change, (int)pNrgs->nrgGain_e[k]); } sc_change = (start_pos < no_cols) ? adj_e - input_e : final_e - input_e; if ((scale_change - sc_change + 1) < 0) scale_change -= (scale_change - sc_change + 1); scale_change = (scale_change - sc_change) + 1; for (k = 0; k < noSubbands; k++) { int sc = scale_change - pNrgs->nrgGain_e[k] + (sc_change - 1); pNrgs->nrgGain[k] >>= fixMin(sc, DFRACT_BITS - 1); pNrgs->nrgGain_e[k] += sc; } if (!useLP) { for (k = 0; k < noSubbands; k++) { int sc = scale_change - h_sbr_cal_env->filtBuffer_e[k] + (sc_change - 1); h_sbr_cal_env->filtBuffer[k] >>= fixMin(sc, DFRACT_BITS - 1); } } for (j = start_pos; j < stop_pos; j++) { /* This timeslot is located within the first part of the processing buffer and will be fed into the QMF-synthesis for the current frame. adj_e - input_e This timeslot will not yet be fed into the QMF so we do not care about the adj_e. sc_change = final_e - input_e */ if ((j == no_cols) && (start_pos < no_cols)) { int shift = (int)(noise_e - final_e); if (!useLP) filtBufferNoiseShift = shift; /* shifting of h_sbr_cal_env->filtBufferNoise[k] will be applied in function adjustTimeSlotHQ() */ if (shift >= 0) { shift = fixMin(DFRACT_BITS - 1, shift); for (k = 0; k < noSubbands; k++) { pNrgs->nrgSine[k] <<= shift; pNrgs->noiseLevel[k] <<= shift; /* if (!useLP) h_sbr_cal_env->filtBufferNoise[k] <<= shift; */ } } else { shift = fixMin(DFRACT_BITS - 1, -shift); for (k = 0; k < noSubbands; k++) { pNrgs->nrgSine[k] >>= shift; pNrgs->noiseLevel[k] >>= shift; /* if (!useLP) h_sbr_cal_env->filtBufferNoise[k] >>= shift; */ } } /* update noise scaling */ noise_e = final_e; if (!useLP) h_sbr_cal_env->filtBufferNoise_e = noise_e; /* scaling value unused! */ /* update gain buffer*/ sc_change -= (final_e - input_e); if (sc_change < 0) { for (k = 0; k < noSubbands; k++) { pNrgs->nrgGain[k] >>= -sc_change; pNrgs->nrgGain_e[k] += -sc_change; } if (!useLP) { for (k = 0; k < noSubbands; k++) { h_sbr_cal_env->filtBuffer[k] >>= -sc_change; } } } else { scale_change += sc_change; } } /* if */ if (!useLP) { /* Prevent the smoothing filter from running on constant levels */ if (j - start_pos < smooth_length) smooth_ratio = FDK_sbrDecoder_sbr_smoothFilter[j - start_pos]; else smooth_ratio = FL2FXCONST_SGL(0.0f); if (iTES_enable) { /* adjustTimeSlotHQ() without adding of additional harmonics */ adjustTimeSlotHQ_GainAndNoise( &analysBufferReal[j][lowSubband], &analysBufferImag[j][lowSubband], h_sbr_cal_env, pNrgs, lowSubband, noSubbands, fMin(scale_change, DFRACT_BITS - 1), smooth_ratio, noNoiseFlag, filtBufferNoiseShift); } else { adjustTimeSlotHQ(&analysBufferReal[j][lowSubband], &analysBufferImag[j][lowSubband], h_sbr_cal_env, pNrgs, lowSubband, noSubbands, fMin(scale_change, DFRACT_BITS - 1), smooth_ratio, noNoiseFlag, filtBufferNoiseShift); } } else { FDK_ASSERT(!iTES_enable); /* not supported */ if (flags & SBRDEC_ELD_GRID) { /* FDKmemset(analysBufferReal[j], 0, 64 * sizeof(FIXP_DBL)); */ adjustTimeSlot_EldGrid( &analysBufferReal[j][lowSubband], pNrgs, &h_sbr_cal_env->harmIndex, lowSubband, noSubbands, fMin(scale_change, DFRACT_BITS - 1), noNoiseFlag, &h_sbr_cal_env->phaseIndex, fMax(EXP2SCALE(adj_e) - sbrScaleFactor->lb_scale, -(DFRACT_BITS - 1))); } else { adjustTimeSlotLC(&analysBufferReal[j][lowSubband], pNrgs, &h_sbr_cal_env->harmIndex, lowSubband, noSubbands, fMin(scale_change, DFRACT_BITS - 1), noNoiseFlag, &h_sbr_cal_env->phaseIndex); } } /* In case the envelope spans accross the no_cols border both exponents * are needed. */ /* nrgGain_e[0...(noSubbands-1)] are equalized by * equalizeFiltBufferExp() */ pNrgs->exponent[(j < no_cols) ? 0 : 1] = (SCHAR)((15 - sbrScaleFactor->hb_scale) + pNrgs->nrgGain_e[0] + 1 - scale_change); } /* for */ if (iTES_enable) { apply_inter_tes( analysBufferReal, /* pABufR, */ analysBufferImag, /* pABufI, */ sbrScaleFactor, pNrgs->exponent, hHeaderData->timeStep, start_pos, stop_pos, lowSubband, noSubbands, hFrameData ->interTempShapeMode[i] /* frameData->interTempShapeMode[env] */ ); /* add additional harmonics */ for (j = start_pos; j < stop_pos; j++) { /* match exponent of additional harmonics to scale change of QMF data * caused by apply_inter_tes() */ scale_change = 0; if ((start_pos <= no_cols) && (stop_pos > no_cols)) { /* Scaling of analysBuffers was potentially changed within this envelope. The pNrgs->nrgSine_e match the second part of the envelope. For (j<=no_cols) the exponent of the sine energies has to be adapted. */ scale_change = pNrgs->exponent[1] - pNrgs->exponent[0]; } adjustTimeSlotHQ_AddHarmonics( &analysBufferReal[j][lowSubband], &analysBufferImag[j][lowSubband], h_sbr_cal_env, pNrgs, lowSubband, noSubbands, -iTES_scale_change + ((j < no_cols) ? scale_change : 0)); } } if (!useLP) { /* Update time-smoothing-buffers for gains and noise levels The gains and the noise values of the current envelope are copied into the buffer. This has to be done at the end of each envelope as the values are required for a smooth transition to the next envelope. */ FDKmemcpy(h_sbr_cal_env->filtBuffer, pNrgs->nrgGain, noSubbands * sizeof(FIXP_DBL)); FDKmemcpy(h_sbr_cal_env->filtBuffer_e, pNrgs->nrgGain_e, noSubbands * sizeof(SCHAR)); FDKmemcpy(h_sbr_cal_env->filtBufferNoise, pNrgs->noiseLevel, noSubbands * sizeof(FIXP_DBL)); } } C_ALLOC_SCRATCH_END(pNrgs, ENV_CALC_NRGS, 1); } /* adapt adj_e to the scale change caused by apply_inter_tes() */ adj_e += iTES_scale_change; /* Rescale output samples */ { FIXP_DBL maxVal; int ov_reserve, reserve; /* Determine headroom in old adjusted samples */ maxVal = maxSubbandSample(analysBufferReal, (useLP) ? NULL : analysBufferImag, lowSubband, ov_highSubband, 0, first_start); ov_reserve = fNorm(maxVal); /* Determine headroom in new adjusted samples */ maxVal = maxSubbandSample(analysBufferReal, (useLP) ? NULL : analysBufferImag, lowSubband, highSubband, first_start, no_cols); reserve = fNorm(maxVal); /* Determine common output exponent */ output_e = fMax(ov_adj_e - ov_reserve, adj_e - reserve); /* Rescale old samples */ rescaleSubbandSamples(analysBufferReal, (useLP) ? NULL : analysBufferImag, lowSubband, ov_highSubband, 0, first_start, ov_adj_e - output_e); /* Rescale new samples */ rescaleSubbandSamples(analysBufferReal, (useLP) ? NULL : analysBufferImag, lowSubband, highSubband, first_start, no_cols, adj_e - output_e); } /* Update hb_scale */ sbrScaleFactor->hb_scale = EXP2SCALE(output_e); /* Save the current final exponent for the next frame: */ /* adapt final_e to the scale change caused by apply_inter_tes() */ sbrScaleFactor->ov_hb_scale = EXP2SCALE(final_e + iTES_scale_change); /* We need to remember to the next frame that the transient will occur in the first envelope (if tranEnv == nEnvelopes). */ if (hFrameData->frameInfo.tranEnv == hFrameData->frameInfo.nEnvelopes) h_sbr_cal_env->prevTranEnv = 0; else h_sbr_cal_env->prevTranEnv = -1; if (pvc_mode > 0) { /* Not more than just the last noise envelope reaches into the next PVC frame! This should be true because bs_noise_position is <= 15 */ FDK_ASSERT(hFrameData->frameInfo .bordersNoise[hFrameData->frameInfo.nNoiseEnvelopes - 1] < PVC_NTIMESLOT); if (hFrameData->frameInfo .bordersNoise[hFrameData->frameInfo.nNoiseEnvelopes] > PVC_NTIMESLOT) { FDK_ASSERT(noiseLevels == (hFrameData->sbrNoiseFloorLevel + (hFrameData->frameInfo.nNoiseEnvelopes - 1) * noNoiseBands)); h_sbr_cal_env->prevNNfb = noNoiseBands; h_sbr_cal_env->prevNSfb[0] = noSubFrameBands[0]; h_sbr_cal_env->prevNSfb[1] = noSubFrameBands[1]; h_sbr_cal_env->prevLoSubband = lowSubband; h_sbr_cal_env->prevHiSubband = highSubband; h_sbr_cal_env->prev_ov_highSubband = ov_highSubband; FDKmemcpy(h_sbr_cal_env->prevFreqBandTableLo, pFreqBandTable[0], noSubFrameBands[0] + 1); FDKmemcpy(h_sbr_cal_env->prevFreqBandTableHi, pFreqBandTable[1], noSubFrameBands[1] + 1); FDKmemcpy(h_sbr_cal_env->prevFreqBandTableNoise, hFreq->freqBandTableNoise, sizeof(hFreq->freqBandTableNoise)); FDKmemcpy(h_sbr_cal_env->prevSbrNoiseFloorLevel, noiseLevels, MAX_NOISE_COEFFS * sizeof(FIXP_SGL)); } } C_ALLOC_SCRATCH_END(useAliasReduction, UCHAR, 64) } /*! \brief Create envelope instance Must be called once for each channel before calculateSbrEnvelope() can be used. \return errorCode, 0 if successful */ SBR_ERROR createSbrEnvelopeCalc( HANDLE_SBR_CALCULATE_ENVELOPE hs, /*!< pointer to envelope instance */ HANDLE_SBR_HEADER_DATA hHeaderData, /*!< static SBR control data, initialized with defaults */ const int chan, /*!< Channel for which to assign buffers */ const UINT flags) { SBR_ERROR err = SBRDEC_OK; int i; /* Clear previous missing harmonics flags */ for (i = 0; i < ADD_HARMONICS_FLAGS_SIZE; i++) { hs->harmFlagsPrev[i] = 0; hs->harmFlagsPrevActive[i] = 0; } hs->harmIndex = 0; FDKmemclear(hs->prevSbrNoiseFloorLevel, sizeof(hs->prevSbrNoiseFloorLevel)); hs->prevNNfb = 0; FDKmemclear(hs->prevFreqBandTableNoise, sizeof(hs->prevFreqBandTableNoise)); hs->sinusoidal_positionPrev = 0; /* Setup pointers for time smoothing. The buffer itself will be initialized later triggered by the startUp-flag. */ hs->prevTranEnv = -1; /* initialization */ resetSbrEnvelopeCalc(hs); if (chan == 0) { /* do this only once */ err = resetFreqBandTables(hHeaderData, flags); } return err; } /*! \brief Create envelope instance Must be called once for each channel before calculateSbrEnvelope() can be used. \return errorCode, 0 if successful */ int deleteSbrEnvelopeCalc(HANDLE_SBR_CALCULATE_ENVELOPE hs) { return 0; } /*! \brief Reset envelope instance This function must be called for each channel on a change of configuration. Note that resetFreqBandTables should also be called in this case. \return errorCode, 0 if successful */ void resetSbrEnvelopeCalc( HANDLE_SBR_CALCULATE_ENVELOPE hCalEnv) /*!< pointer to envelope instance */ { hCalEnv->phaseIndex = 0; /* Noise exponent needs to be reset because the output exponent for the next * frame depends on it */ hCalEnv->filtBufferNoise_e = 0; hCalEnv->startUp = 1; } /*! \brief Equalize exponents of the buffered gain values and the new ones After equalization of exponents, the FIR-filter addition for smoothing can be performed. This function is called once for each envelope before adjusting. */ static void equalizeFiltBufferExp( FIXP_DBL *filtBuffer, /*!< bufferd gains */ SCHAR *filtBuffer_e, /*!< exponents of bufferd gains */ FIXP_DBL *nrgGain, /*!< gains for current envelope */ SCHAR *nrgGain_e, /*!< exponents of gains for current envelope */ int subbands) /*!< Number of QMF subbands */ { int band; int diff; for (band = 0; band < subbands; band++) { diff = (int)(nrgGain_e[band] - filtBuffer_e[band]); if (diff > 0) { filtBuffer[band] >>= fMin(diff, DFRACT_BITS - 1); /* Compensate for the scale change by shifting the mantissa. */ filtBuffer_e[band] += diff; /* New gain is bigger, use its exponent */ } else if (diff < 0) { /* The buffered gains seem to be larger, but maybe there are some unused bits left in the mantissa */ int reserve = CntLeadingZeros(fixp_abs(filtBuffer[band])) - 1; if ((-diff) <= reserve) { /* There is enough space in the buffered mantissa so that we can take the new exponent as common. */ filtBuffer[band] <<= (-diff); filtBuffer_e[band] += diff; /* becomes equal to *ptrNewExp */ } else { filtBuffer[band] <<= reserve; /* Shift the mantissa as far as possible: */ filtBuffer_e[band] -= reserve; /* Compensate in the exponent: */ /* For the remaining difference, change the new gain value */ diff = -(reserve + diff); nrgGain[band] >>= fMin(diff, DFRACT_BITS - 1); nrgGain_e[band] += diff; } } } } /*! \brief Shift left the mantissas of all subband samples in the giventime and frequency range by the specified number of bits. This function is used to rescale the audio data in the overlap buffer which has already been envelope adjusted with the last frame. */ void rescaleSubbandSamples( FIXP_DBL **re, /*!< Real part of input and output subband samples */ FIXP_DBL **im, /*!< Imaginary part of input and output subband samples */ int lowSubband, /*!< Begin of frequency range to process */ int highSubband, /*!< End of frequency range to process */ int start_pos, /*!< Begin of time rage (QMF-timeslot) */ int next_pos, /*!< End of time rage (QMF-timeslot) */ int shift) /*!< number of bits to shift */ { int width = highSubband - lowSubband; if ((width > 0) && (shift != 0)) { if (im != NULL) { for (int l = start_pos; l < next_pos; l++) { scaleValues(&re[l][lowSubband], width, shift); scaleValues(&im[l][lowSubband], width, shift); } } else { for (int l = start_pos; l < next_pos; l++) { scaleValues(&re[l][lowSubband], width, shift); } } } } static inline FIXP_DBL FDK_get_maxval_real(FIXP_DBL maxVal, FIXP_DBL *reTmp, INT width) { maxVal = (FIXP_DBL)0; while (width-- != 0) { FIXP_DBL tmp = *(reTmp++); maxVal |= (FIXP_DBL)((LONG)(tmp) ^ ((LONG)tmp >> (DFRACT_BITS - 1))); } return maxVal; } /*! \brief Determine headroom for shifting Determine by how much the spectrum can be shifted left for better accuracy in later processing. \return Number of free bits in the biggest spectral value */ FIXP_DBL maxSubbandSample( FIXP_DBL **re, /*!< Real part of input and output subband samples */ FIXP_DBL **im, /*!< Real part of input and output subband samples */ int lowSubband, /*!< Begin of frequency range to process */ int highSubband, /*!< Number of QMF bands to process */ int start_pos, /*!< Begin of time rage (QMF-timeslot) */ int next_pos /*!< End of time rage (QMF-timeslot) */ ) { FIXP_DBL maxVal = FL2FX_DBL(0.0f); unsigned int width = highSubband - lowSubband; FDK_ASSERT(width <= (64)); if (width > 0) { if (im != NULL) { for (int l = start_pos; l < next_pos; l++) { int k = width; FIXP_DBL *reTmp = &re[l][lowSubband]; FIXP_DBL *imTmp = &im[l][lowSubband]; do { FIXP_DBL tmp1 = *(reTmp++); FIXP_DBL tmp2 = *(imTmp++); maxVal |= (FIXP_DBL)((LONG)(tmp1) ^ ((LONG)tmp1 >> (DFRACT_BITS - 1))); maxVal |= (FIXP_DBL)((LONG)(tmp2) ^ ((LONG)tmp2 >> (DFRACT_BITS - 1))); } while (--k != 0); } } else { for (int l = start_pos; l < next_pos; l++) { maxVal |= FDK_get_maxval_real(maxVal, &re[l][lowSubband], width); } } } if (maxVal > (FIXP_DBL)0) { /* For negative input values, maxVal is too small by 1. Add 1 only when * necessary: if maxVal is a power of 2 */ FIXP_DBL lowerPow2 = (FIXP_DBL)(1 << (DFRACT_BITS - 1 - CntLeadingZeros(maxVal))); if (maxVal == lowerPow2) maxVal += (FIXP_DBL)1; } return (maxVal); } /* #define SHIFT_BEFORE_SQUARE (3) */ /* (7/2) */ /* Avoid assertion failures triggerd by overflows which occured in robustness tests. Setting the SHIFT_BEFORE_SQUARE to 4 has negligible effect on (USAC) conformance results. */ #define SHIFT_BEFORE_SQUARE (4) /* ((8 - 0) / 2) */ /*!< If the accumulator does not provide enough overflow bits or does not provide a high dynamic range, the below energy calculation requires an additional shift operation for each sample. On the other hand, doing the shift allows using a single-precision multiplication for the square (at least 16bit x 16bit). For even values of OVRFLW_BITS (0, 2, 4, 6), saturated arithmetic is required for the energy accumulation. Theoretically, the sample-squares can sum up to a value of 76, requiring 7 overflow bits. However since such situations are *very* rare, accu can be limited to 64. In case native saturated arithmetic is not available, overflows can be prevented by replacing the above #define by #define SHIFT_BEFORE_SQUARE ((8 - OVRFLW_BITS) / 2) which will result in slightly reduced accuracy. */ /*! \brief Estimates the mean energy of each filter-bank channel for the duration of the current envelope This function is used when interpolFreq is true. */ static void calcNrgPerSubband( FIXP_DBL **analysBufferReal, /*!< Real part of subband samples */ FIXP_DBL **analysBufferImag, /*!< Imaginary part of subband samples */ int lowSubband, /*!< Begin of the SBR frequency range */ int highSubband, /*!< High end of the SBR frequency range */ int start_pos, /*!< First QMF-slot of current envelope */ int next_pos, /*!< Last QMF-slot of current envelope + 1 */ SCHAR frameExp, /*!< Common exponent for all input samples */ FIXP_DBL *nrgEst, /*!< resulting Energy (0..1) */ SCHAR *nrgEst_e) /*!< Exponent of resulting Energy */ { FIXP_SGL invWidth; SCHAR preShift; SCHAR shift; FIXP_DBL sum; int k; /* Divide by width of envelope later: */ invWidth = FX_DBL2FX_SGL(GetInvInt(next_pos - start_pos)); /* The common exponent needs to be doubled because all mantissas are squared: */ frameExp = frameExp << 1; for (k = lowSubband; k < highSubband; k++) { FIXP_DBL bufferReal[(((1024) / (32) * (4) / 2) + (3 * (4)))]; FIXP_DBL bufferImag[(((1024) / (32) * (4) / 2) + (3 * (4)))]; FIXP_DBL maxVal; if (analysBufferImag != NULL) { int l; maxVal = FL2FX_DBL(0.0f); for (l = start_pos; l < next_pos; l++) { bufferImag[l] = analysBufferImag[l][k]; maxVal |= (FIXP_DBL)((LONG)(bufferImag[l]) ^ ((LONG)bufferImag[l] >> (DFRACT_BITS - 1))); bufferReal[l] = analysBufferReal[l][k]; maxVal |= (FIXP_DBL)((LONG)(bufferReal[l]) ^ ((LONG)bufferReal[l] >> (DFRACT_BITS - 1))); } } else { int l; maxVal = FL2FX_DBL(0.0f); for (l = start_pos; l < next_pos; l++) { bufferReal[l] = analysBufferReal[l][k]; maxVal |= (FIXP_DBL)((LONG)(bufferReal[l]) ^ ((LONG)bufferReal[l] >> (DFRACT_BITS - 1))); } } if (maxVal != FL2FXCONST_DBL(0.f)) { /* If the accu does not provide enough overflow bits, we cannot shift the samples up to the limit. Instead, keep up to 3 free bits in each sample, i.e. up to 6 bits after calculation of square. Please note the comment on saturated arithmetic above! */ FIXP_DBL accu; preShift = CntLeadingZeros(maxVal) - 1; preShift -= SHIFT_BEFORE_SQUARE; /* Limit preShift to a maximum value to prevent accumulator overflow in exceptional situations where the signal in the analysis-buffer is very small (small maxVal). */ preShift = fMin(preShift, (SCHAR)25); accu = FL2FXCONST_DBL(0.0f); if (preShift >= 0) { int l; if (analysBufferImag != NULL) { for (l = start_pos; l < next_pos; l++) { FIXP_DBL temp1 = bufferReal[l] << (int)preShift; FIXP_DBL temp2 = bufferImag[l] << (int)preShift; accu = fPow2AddDiv2(accu, temp1); accu = fPow2AddDiv2(accu, temp2); } } else { for (l = start_pos; l < next_pos; l++) { FIXP_DBL temp = bufferReal[l] << (int)preShift; accu = fPow2AddDiv2(accu, temp); } } } else { /* if negative shift value */ int l; int negpreShift = -preShift; if (analysBufferImag != NULL) { for (l = start_pos; l < next_pos; l++) { FIXP_DBL temp1 = bufferReal[l] >> (int)negpreShift; FIXP_DBL temp2 = bufferImag[l] >> (int)negpreShift; accu = fPow2AddDiv2(accu, temp1); accu = fPow2AddDiv2(accu, temp2); } } else { for (l = start_pos; l < next_pos; l++) { FIXP_DBL temp = bufferReal[l] >> (int)negpreShift; accu = fPow2AddDiv2(accu, temp); } } } accu <<= 1; /* Convert double precision to Mantissa/Exponent: */ shift = fNorm(accu); sum = accu << (int)shift; /* Divide by width of envelope and apply frame scale: */ *nrgEst++ = fMult(sum, invWidth); shift += 2 * preShift; if (analysBufferImag != NULL) *nrgEst_e++ = frameExp - shift; else *nrgEst_e++ = frameExp - shift + 1; /* +1 due to missing imag. part */ } /* maxVal!=0 */ else { /* Prevent a zero-mantissa-number from being misinterpreted due to its exponent. */ *nrgEst++ = FL2FXCONST_DBL(0.0f); *nrgEst_e++ = 0; } } } /*! \brief Estimates the mean energy of each Scale factor band for the duration of the current envelope. This function is used when interpolFreq is false. */ static void calcNrgPerSfb( FIXP_DBL **analysBufferReal, /*!< Real part of subband samples */ FIXP_DBL **analysBufferImag, /*!< Imaginary part of subband samples */ int nSfb, /*!< Number of scale factor bands */ UCHAR *freqBandTable, /*!< First Subband for each Sfb */ int start_pos, /*!< First QMF-slot of current envelope */ int next_pos, /*!< Last QMF-slot of current envelope + 1 */ SCHAR input_e, /*!< Common exponent for all input samples */ FIXP_DBL *nrgEst, /*!< resulting Energy (0..1) */ SCHAR *nrgEst_e) /*!< Exponent of resulting Energy */ { FIXP_SGL invWidth; FIXP_DBL temp; SCHAR preShift; SCHAR shift, sum_e; FIXP_DBL sum; int j, k, l, li, ui; FIXP_DBL sumAll, sumLine; /* Single precision would be sufficient, but overflow bits are required for accumulation */ /* Divide by width of envelope later: */ invWidth = FX_DBL2FX_SGL(GetInvInt(next_pos - start_pos)); /* The common exponent needs to be doubled because all mantissas are squared: */ input_e = input_e << 1; for (j = 0; j < nSfb; j++) { li = freqBandTable[j]; ui = freqBandTable[j + 1]; FIXP_DBL maxVal = maxSubbandSample(analysBufferReal, analysBufferImag, li, ui, start_pos, next_pos); if (maxVal != FL2FXCONST_DBL(0.f)) { preShift = CntLeadingZeros(maxVal) - 1; /* If the accu does not provide enough overflow bits, we cannot shift the samples up to the limit. Instead, keep up to 3 free bits in each sample, i.e. up to 6 bits after calculation of square. Please note the comment on saturated arithmetic above! */ preShift -= SHIFT_BEFORE_SQUARE; sumAll = FL2FXCONST_DBL(0.0f); for (k = li; k < ui; k++) { sumLine = FL2FXCONST_DBL(0.0f); if (analysBufferImag != NULL) { if (preShift >= 0) { for (l = start_pos; l < next_pos; l++) { temp = analysBufferReal[l][k] << (int)preShift; sumLine += fPow2Div2(temp); temp = analysBufferImag[l][k] << (int)preShift; sumLine += fPow2Div2(temp); } } else { for (l = start_pos; l < next_pos; l++) { temp = analysBufferReal[l][k] >> -(int)preShift; sumLine += fPow2Div2(temp); temp = analysBufferImag[l][k] >> -(int)preShift; sumLine += fPow2Div2(temp); } } } else { if (preShift >= 0) { for (l = start_pos; l < next_pos; l++) { temp = analysBufferReal[l][k] << (int)preShift; sumLine += fPow2Div2(temp); } } else { for (l = start_pos; l < next_pos; l++) { temp = analysBufferReal[l][k] >> -(int)preShift; sumLine += fPow2Div2(temp); } } } /* The number of QMF-channels per SBR bands may be up to 15. Shift right to avoid overflows in sum over all channels. */ sumLine = sumLine >> (4 - 1); sumAll += sumLine; } /* Convert double precision to Mantissa/Exponent: */ shift = fNorm(sumAll); sum = sumAll << (int)shift; /* Divide by width of envelope: */ sum = fMult(sum, invWidth); /* Divide by width of Sfb: */ sum = fMult(sum, FX_DBL2FX_SGL(GetInvInt(ui - li))); /* Set all Subband energies in the Sfb to the average energy: */ if (analysBufferImag != NULL) sum_e = input_e + 4 - shift; /* -4 to compensate right-shift */ else sum_e = input_e + 4 + 1 - shift; /* -4 to compensate right-shift; +1 due to missing imag. part */ sum_e -= 2 * preShift; } /* maxVal!=0 */ else { /* Prevent a zero-mantissa-number from being misinterpreted due to its exponent. */ sum = FL2FXCONST_DBL(0.0f); sum_e = 0; } for (k = li; k < ui; k++) { *nrgEst++ = sum; *nrgEst_e++ = sum_e; } } } /*! \brief Calculate gain, noise, and additional sine level for one subband. The resulting energy gain is given by mantissa and exponent. */ static void calcSubbandGain( FIXP_DBL nrgRef, /*!< Reference Energy according to envelope data */ SCHAR nrgRef_e, /*!< Reference Energy according to envelope data (exponent) */ ENV_CALC_NRGS *nrgs, int i, FIXP_DBL tmpNoise, /*!< Relative noise level */ SCHAR tmpNoise_e, /*!< Relative noise level (exponent) */ UCHAR sinePresentFlag, /*!< Indicates if sine is present on band */ UCHAR sineMapped, /*!< Indicates if sine must be added */ int noNoiseFlag) /*!< Flag to suppress noise addition */ { FIXP_DBL nrgEst = nrgs->nrgEst[i]; /*!< Energy in transposed signal */ SCHAR nrgEst_e = nrgs->nrgEst_e[i]; /*!< Energy in transposed signal (exponent) */ FIXP_DBL *ptrNrgGain = &nrgs->nrgGain[i]; /*!< Resulting energy gain */ SCHAR *ptrNrgGain_e = &nrgs->nrgGain_e[i]; /*!< Resulting energy gain (exponent) */ FIXP_DBL *ptrNoiseLevel = &nrgs->noiseLevel[i]; /*!< Resulting absolute noise energy */ SCHAR *ptrNoiseLevel_e = &nrgs->noiseLevel_e[i]; /*!< Resulting absolute noise energy (exponent) */ FIXP_DBL *ptrNrgSine = &nrgs->nrgSine[i]; /*!< Additional sine energy */ SCHAR *ptrNrgSine_e = &nrgs->nrgSine_e[i]; /*!< Additional sine energy (exponent) */ FIXP_DBL a, b, c; SCHAR a_e, b_e, c_e; /* This addition of 1 prevents divisions by zero in the reference code. For very small energies in nrgEst, it prevents the gains from becoming very high which could cause some trouble due to the smoothing. */ b_e = (int)(nrgEst_e - 1); if (b_e >= 0) { nrgEst = (FL2FXCONST_DBL(0.5f) >> (INT)fixMin(b_e + 1, DFRACT_BITS - 1)) + (nrgEst >> 1); nrgEst_e += 1; /* shift by 1 bit to avoid overflow */ } else { nrgEst = (nrgEst >> (INT)(fixMin(-b_e + 1, DFRACT_BITS - 1))) + (FL2FXCONST_DBL(0.5f) >> 1); nrgEst_e = 2; /* shift by 1 bit to avoid overflow */ } /* A = NrgRef * TmpNoise */ a = fMult(nrgRef, tmpNoise); a_e = nrgRef_e + tmpNoise_e; /* B = 1 + TmpNoise */ b_e = (int)(tmpNoise_e - 1); if (b_e >= 0) { b = (FL2FXCONST_DBL(0.5f) >> (INT)fixMin(b_e + 1, DFRACT_BITS - 1)) + (tmpNoise >> 1); b_e = tmpNoise_e + 1; /* shift by 1 bit to avoid overflow */ } else { b = (tmpNoise >> (INT)(fixMin(-b_e + 1, DFRACT_BITS - 1))) + (FL2FXCONST_DBL(0.5f) >> 1); b_e = 2; /* shift by 1 bit to avoid overflow */ } /* noiseLevel = A / B = (NrgRef * TmpNoise) / (1 + TmpNoise) */ FDK_divide_MantExp(a, a_e, b, b_e, ptrNoiseLevel, ptrNoiseLevel_e); if (sinePresentFlag) { /* C = (1 + TmpNoise) * NrgEst */ c = fMult(b, nrgEst); c_e = b_e + nrgEst_e; /* gain = A / C = (NrgRef * TmpNoise) / (1 + TmpNoise) * NrgEst */ FDK_divide_MantExp(a, a_e, c, c_e, ptrNrgGain, ptrNrgGain_e); if (sineMapped) { /* sineLevel = nrgRef/ (1 + TmpNoise) */ FDK_divide_MantExp(nrgRef, nrgRef_e, b, b_e, ptrNrgSine, ptrNrgSine_e); } } else { if (noNoiseFlag) { /* B = NrgEst */ b = nrgEst; b_e = nrgEst_e; } else { /* B = NrgEst * (1 + TmpNoise) */ b = fMult(b, nrgEst); b_e = b_e + nrgEst_e; } /* gain = nrgRef / B */ INT result_exp = 0; *ptrNrgGain = fDivNorm(nrgRef, b, &result_exp); *ptrNrgGain_e = (SCHAR)result_exp + (nrgRef_e - b_e); /* There could be a one bit diffs. This is important to compensate, because later in the code values are compared by exponent only. */ int headroom = CountLeadingBits(*ptrNrgGain); *ptrNrgGain <<= headroom; *ptrNrgGain_e -= headroom; } } /*! \brief Calculate "average gain" for the specified subband range. This is rather a gain of the average magnitude than the average of gains! The result is used as a relative limit for all gains within the current "limiter band" (a certain frequency range). */ static void calcAvgGain( ENV_CALC_NRGS *nrgs, int lowSubband, /*!< Begin of the limiter band */ int highSubband, /*!< High end of the limiter band */ FIXP_DBL *ptrSumRef, SCHAR *ptrSumRef_e, FIXP_DBL *ptrAvgGain, /*!< Resulting overall gain (mantissa) */ SCHAR *ptrAvgGain_e) /*!< Resulting overall gain (exponent) */ { FIXP_DBL *nrgRef = nrgs->nrgRef; /*!< Reference Energy according to envelope data */ SCHAR *nrgRef_e = nrgs->nrgRef_e; /*!< Reference Energy according to envelope data (exponent) */ FIXP_DBL *nrgEst = nrgs->nrgEst; /*!< Energy in transposed signal */ SCHAR *nrgEst_e = nrgs->nrgEst_e; /*!< Energy in transposed signal (exponent) */ FIXP_DBL sumRef = 1; FIXP_DBL sumEst = 1; SCHAR sumRef_e = -FRACT_BITS; SCHAR sumEst_e = -FRACT_BITS; int k; for (k = lowSubband; k < highSubband; k++) { /* Add nrgRef[k] to sumRef: */ FDK_add_MantExp(sumRef, sumRef_e, nrgRef[k], nrgRef_e[k], &sumRef, &sumRef_e); /* Add nrgEst[k] to sumEst: */ FDK_add_MantExp(sumEst, sumEst_e, nrgEst[k], nrgEst_e[k], &sumEst, &sumEst_e); } FDK_divide_MantExp(sumRef, sumRef_e, sumEst, sumEst_e, ptrAvgGain, ptrAvgGain_e); *ptrSumRef = sumRef; *ptrSumRef_e = sumRef_e; } static void adjustTimeSlot_EldGrid( FIXP_DBL *RESTRICT ptrReal, /*!< Subband samples to be adjusted, real part */ ENV_CALC_NRGS *nrgs, UCHAR *ptrHarmIndex, /*!< Harmonic index */ int lowSubband, /*!< Lowest QMF-channel in the currently used SBR range. */ int noSubbands, /*!< Number of QMF subbands */ int scale_change, /*!< Number of bits to shift adjusted samples */ int noNoiseFlag, /*!< Flag to suppress noise addition */ int *ptrPhaseIndex, /*!< Start index to random number array */ int scale_diff_low) /*!< */ { int k; FIXP_DBL signalReal, sbNoise; int tone_count = 0; FIXP_DBL *pGain = nrgs->nrgGain; /*!< Gains of current envelope */ FIXP_DBL *RESTRICT pNoiseLevel = nrgs->noiseLevel; /*!< Noise levels of current envelope */ FIXP_DBL *RESTRICT pSineLevel = nrgs->nrgSine; /*!< Sine levels */ int phaseIndex = *ptrPhaseIndex; UCHAR harmIndex = *ptrHarmIndex; static const INT harmonicPhase[4][2] = {{1, 0}, {0, 1}, {-1, 0}, {0, -1}}; static const FIXP_DBL harmonicPhaseX[4][2] = { {FL2FXCONST_DBL(2.0 * 1.245183154539139e-001), FL2FXCONST_DBL(2.0 * 1.245183154539139e-001)}, {FL2FXCONST_DBL(2.0 * -1.123767859325028e-001), FL2FXCONST_DBL(2.0 * 1.123767859325028e-001)}, {FL2FXCONST_DBL(2.0 * -1.245183154539139e-001), FL2FXCONST_DBL(2.0 * -1.245183154539139e-001)}, {FL2FXCONST_DBL(2.0 * 1.123767859325028e-001), FL2FXCONST_DBL(2.0 * -1.123767859325028e-001)}}; const FIXP_DBL *p_harmonicPhaseX = &harmonicPhaseX[harmIndex][0]; const INT *p_harmonicPhase = &harmonicPhase[harmIndex][0]; const FIXP_DBL max_val = MAX_VAL_NRG_HEADROOM >> scale_change; const FIXP_DBL min_val = -max_val; *(ptrReal - 1) = fAddSaturate( *(ptrReal - 1), SATURATE_SHIFT(fMultDiv2(p_harmonicPhaseX[lowSubband & 1], pSineLevel[0]), scale_diff_low, DFRACT_BITS)); FIXP_DBL pSineLevel_prev = (FIXP_DBL)0; int idx_k = lowSubband & 1; for (k = 0; k < noSubbands; k++) { FIXP_DBL sineLevel_curr = *pSineLevel++; phaseIndex = (phaseIndex + 1) & (SBR_NF_NO_RANDOM_VAL - 1); signalReal = fMax(fMin(fMultDiv2(*ptrReal, *pGain++), max_val), min_val) << scale_change; sbNoise = *pNoiseLevel++; if (((INT)sineLevel_curr | noNoiseFlag) == 0) { signalReal += fMult(FDK_sbrDecoder_sbr_randomPhase[phaseIndex][0], sbNoise); } signalReal += sineLevel_curr * p_harmonicPhase[0]; signalReal = fMultAddDiv2(signalReal, pSineLevel_prev, p_harmonicPhaseX[idx_k]); pSineLevel_prev = sineLevel_curr; idx_k = !idx_k; if (k < noSubbands - 1) { signalReal = fMultAddDiv2(signalReal, pSineLevel[0], p_harmonicPhaseX[idx_k]); } else /* (k == noSubbands - 1) */ { if (k + lowSubband + 1 < 63) { *(ptrReal + 1) += fMultDiv2(pSineLevel_prev, p_harmonicPhaseX[idx_k]); } } *ptrReal++ = signalReal; if (pSineLevel_prev != FL2FXCONST_DBL(0.0f)) { if (++tone_count == 16) { k++; break; } } } /* Run again, if previous loop got breaked with tone_count = 16 */ for (; k < noSubbands; k++) { FIXP_DBL sineLevel_curr = *pSineLevel++; phaseIndex = (phaseIndex + 1) & (SBR_NF_NO_RANDOM_VAL - 1); signalReal = fMax(fMin(fMultDiv2(*ptrReal, *pGain++), max_val), min_val) << scale_change; sbNoise = *pNoiseLevel++; if (((INT)sineLevel_curr | noNoiseFlag) == 0) { signalReal += fMult(FDK_sbrDecoder_sbr_randomPhase[phaseIndex][0], sbNoise); } signalReal += sineLevel_curr * p_harmonicPhase[0]; *ptrReal++ = signalReal; } *ptrHarmIndex = (harmIndex + 1) & 3; *ptrPhaseIndex = phaseIndex & (SBR_NF_NO_RANDOM_VAL - 1); } /*! \brief Amplify one timeslot of the signal with the calculated gains and add the noisefloor. */ static void adjustTimeSlotLC( FIXP_DBL *ptrReal, /*!< Subband samples to be adjusted, real part */ ENV_CALC_NRGS *nrgs, UCHAR *ptrHarmIndex, /*!< Harmonic index */ int lowSubband, /*!< Lowest QMF-channel in the currently used SBR range. */ int noSubbands, /*!< Number of QMF subbands */ int scale_change, /*!< Number of bits to shift adjusted samples */ int noNoiseFlag, /*!< Flag to suppress noise addition */ int *ptrPhaseIndex) /*!< Start index to random number array */ { FIXP_DBL *pGain = nrgs->nrgGain; /*!< Gains of current envelope */ FIXP_DBL *pNoiseLevel = nrgs->noiseLevel; /*!< Noise levels of current envelope */ FIXP_DBL *pSineLevel = nrgs->nrgSine; /*!< Sine levels */ int k; int index = *ptrPhaseIndex; UCHAR harmIndex = *ptrHarmIndex; UCHAR freqInvFlag = (lowSubband & 1); FIXP_DBL signalReal, sineLevel, sineLevelNext, sineLevelPrev; int tone_count = 0; int sineSign = 1; const FIXP_DBL max_val = MAX_VAL_NRG_HEADROOM >> scale_change; const FIXP_DBL min_val = -max_val; #define C1 ((FIXP_SGL)FL2FXCONST_SGL(2.f * 0.00815f)) #define C1_CLDFB ((FIXP_SGL)FL2FXCONST_SGL(2.f * 0.16773f)) /* First pass for k=0 pulled out of the loop: */ index = (index + 1) & (SBR_NF_NO_RANDOM_VAL - 1); /* The next multiplication constitutes the actual envelope adjustment of the signal and should be carried out with full accuracy (supplying #FRACT_BITS valid bits). */ signalReal = fMax(fMin(fMultDiv2(*ptrReal, *pGain++), max_val), min_val) << scale_change; sineLevel = *pSineLevel++; sineLevelNext = (noSubbands > 1) ? pSineLevel[0] : FL2FXCONST_DBL(0.0f); if (sineLevel != FL2FXCONST_DBL(0.0f)) tone_count++; else if (!noNoiseFlag) /* Add noisefloor to the amplified signal */ signalReal += fMult(FDK_sbrDecoder_sbr_randomPhase[index][0], pNoiseLevel[0]); { if (!(harmIndex & 0x1)) { /* harmIndex 0,2 */ signalReal += (harmIndex & 0x2) ? -sineLevel : sineLevel; *ptrReal++ = signalReal; } else { /* harmIndex 1,3 in combination with freqInvFlag */ int shift = (int)(scale_change + 1); shift = (shift >= 0) ? fixMin(DFRACT_BITS - 1, shift) : fixMax(-(DFRACT_BITS - 1), shift); FIXP_DBL tmp1 = (shift >= 0) ? (fMultDiv2(C1, sineLevel) >> shift) : (fMultDiv2(C1, sineLevel) << (-shift)); FIXP_DBL tmp2 = fMultDiv2(C1, sineLevelNext); /* save switch and compare operations and reduce to XOR statement */ if (((harmIndex >> 1) & 0x1) ^ freqInvFlag) { *(ptrReal - 1) = fAddSaturate(*(ptrReal - 1), tmp1); signalReal -= tmp2; } else { *(ptrReal - 1) = fAddSaturate(*(ptrReal - 1), -tmp1); signalReal += tmp2; } *ptrReal++ = signalReal; freqInvFlag = !freqInvFlag; } } pNoiseLevel++; if (noSubbands > 2) { if (!(harmIndex & 0x1)) { /* harmIndex 0,2 */ if (!harmIndex) { sineSign = 0; } for (k = noSubbands - 2; k != 0; k--) { FIXP_DBL sinelevel = *pSineLevel++; index++; if (((signalReal = (sineSign ? -sinelevel : sinelevel)) == FL2FXCONST_DBL(0.0f)) && !noNoiseFlag) { /* Add noisefloor to the amplified signal */ index &= (SBR_NF_NO_RANDOM_VAL - 1); signalReal += fMult(FDK_sbrDecoder_sbr_randomPhase[index][0], pNoiseLevel[0]); } /* The next multiplication constitutes the actual envelope adjustment of * the signal. */ signalReal += fMax(fMin(fMultDiv2(*ptrReal, *pGain++), max_val), min_val) << scale_change; pNoiseLevel++; *ptrReal++ = signalReal; } /* for ... */ } else { /* harmIndex 1,3 in combination with freqInvFlag */ if (harmIndex == 1) freqInvFlag = !freqInvFlag; for (k = noSubbands - 2; k != 0; k--) { index++; /* The next multiplication constitutes the actual envelope adjustment of * the signal. */ signalReal = fMax(fMin(fMultDiv2(*ptrReal, *pGain++), max_val), min_val) << scale_change; if (*pSineLevel++ != FL2FXCONST_DBL(0.0f)) tone_count++; else if (!noNoiseFlag) { /* Add noisefloor to the amplified signal */ index &= (SBR_NF_NO_RANDOM_VAL - 1); signalReal += fMult(FDK_sbrDecoder_sbr_randomPhase[index][0], pNoiseLevel[0]); } pNoiseLevel++; if (tone_count <= 16) { FIXP_DBL addSine = fMultDiv2((pSineLevel[-2] - pSineLevel[0]), C1); signalReal += (freqInvFlag) ? (-addSine) : (addSine); } *ptrReal++ = signalReal; freqInvFlag = !freqInvFlag; } /* for ... */ } } if (noSubbands > -1) { index++; /* The next multiplication constitutes the actual envelope adjustment of the * signal. */ signalReal = fMax(fMin(fMultDiv2(*ptrReal, *pGain), max_val), min_val) << scale_change; sineLevelPrev = fMultDiv2(pSineLevel[-1], FL2FX_SGL(0.0163f)); sineLevel = pSineLevel[0]; if (pSineLevel[0] != FL2FXCONST_DBL(0.0f)) tone_count++; else if (!noNoiseFlag) { /* Add noisefloor to the amplified signal */ index &= (SBR_NF_NO_RANDOM_VAL - 1); signalReal = signalReal + fMult(FDK_sbrDecoder_sbr_randomPhase[index][0], pNoiseLevel[0]); } if (!(harmIndex & 0x1)) { /* harmIndex 0,2 */ *ptrReal = signalReal + ((sineSign) ? -sineLevel : sineLevel); } else { /* harmIndex 1,3 in combination with freqInvFlag */ if (tone_count <= 16) { if (freqInvFlag) { *ptrReal++ = signalReal - sineLevelPrev; if (noSubbands + lowSubband < 63) *ptrReal = *ptrReal + fMultDiv2(C1, sineLevel); } else { *ptrReal++ = signalReal + sineLevelPrev; if (noSubbands + lowSubband < 63) *ptrReal = *ptrReal - fMultDiv2(C1, sineLevel); } } else *ptrReal = signalReal; } } *ptrHarmIndex = (harmIndex + 1) & 3; *ptrPhaseIndex = index & (SBR_NF_NO_RANDOM_VAL - 1); } static void adjustTimeSlotHQ_GainAndNoise( FIXP_DBL *RESTRICT ptrReal, /*!< Subband samples to be adjusted, real part */ FIXP_DBL *RESTRICT ptrImag, /*!< Subband samples to be adjusted, imag part */ HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env, ENV_CALC_NRGS *nrgs, int lowSubband, /*!< Lowest QMF-channel in the currently used SBR range. */ int noSubbands, /*!< Number of QMF subbands */ int scale_change, /*!< Number of bits to shift adjusted samples */ FIXP_SGL smooth_ratio, /*!< Impact of last envelope */ int noNoiseFlag, /*!< Start index to random number array */ int filtBufferNoiseShift) /*!< Shift factor of filtBufferNoise */ { FIXP_DBL *RESTRICT gain = nrgs->nrgGain; /*!< Gains of current envelope */ FIXP_DBL *RESTRICT noiseLevel = nrgs->noiseLevel; /*!< Noise levels of current envelope */ FIXP_DBL *RESTRICT pSineLevel = nrgs->nrgSine; /*!< Sine levels */ FIXP_DBL *RESTRICT filtBuffer = h_sbr_cal_env->filtBuffer; /*!< Gains of last envelope */ FIXP_DBL *RESTRICT filtBufferNoise = h_sbr_cal_env->filtBufferNoise; /*!< Noise levels of last envelope */ int *RESTRICT ptrPhaseIndex = &h_sbr_cal_env->phaseIndex; /*!< Start index to random number array */ int k; FIXP_DBL signalReal, signalImag; FIXP_DBL noiseReal, noiseImag; FIXP_DBL smoothedGain, smoothedNoise; FIXP_SGL direct_ratio = /*FL2FXCONST_SGL(1.0f) */ (FIXP_SGL)MAXVAL_SGL - smooth_ratio; int index = *ptrPhaseIndex; int shift; FIXP_DBL max_val_noise = 0, min_val_noise = 0; const FIXP_DBL max_val = MAX_VAL_NRG_HEADROOM >> scale_change; const FIXP_DBL min_val = -max_val; *ptrPhaseIndex = (index + noSubbands) & (SBR_NF_NO_RANDOM_VAL - 1); filtBufferNoiseShift += 1; /* due to later use of fMultDiv2 instead of fMult */ if (filtBufferNoiseShift < 0) { shift = fixMin(DFRACT_BITS - 1, -filtBufferNoiseShift); } else { shift = fixMin(DFRACT_BITS - 1, filtBufferNoiseShift); max_val_noise = MAX_VAL_NRG_HEADROOM >> shift; min_val_noise = -max_val_noise; } if (smooth_ratio > FL2FXCONST_SGL(0.0f)) { for (k = 0; k < noSubbands; k++) { /* Smoothing: The old envelope has been bufferd and a certain ratio of the old gains and noise levels is used. */ smoothedGain = fMult(smooth_ratio, filtBuffer[k]) + fMult(direct_ratio, gain[k]); if (filtBufferNoiseShift < 0) { smoothedNoise = (fMultDiv2(smooth_ratio, filtBufferNoise[k]) >> shift) + fMult(direct_ratio, noiseLevel[k]); } else { smoothedNoise = fMultDiv2(smooth_ratio, filtBufferNoise[k]); smoothedNoise = (fMax(fMin(smoothedNoise, max_val_noise), min_val_noise) << shift) + fMult(direct_ratio, noiseLevel[k]); } /* The next 2 multiplications constitute the actual envelope adjustment of the signal and should be carried out with full accuracy (supplying #DFRACT_BITS valid bits). */ signalReal = fMax(fMin(fMultDiv2(*ptrReal, smoothedGain), max_val), min_val) << scale_change; signalImag = fMax(fMin(fMultDiv2(*ptrImag, smoothedGain), max_val), min_val) << scale_change; index++; if ((pSineLevel[k] != FL2FXCONST_DBL(0.0f)) || noNoiseFlag) { /* Just the amplified signal is saved */ *ptrReal++ = signalReal; *ptrImag++ = signalImag; } else { /* Add noisefloor to the amplified signal */ index &= (SBR_NF_NO_RANDOM_VAL - 1); noiseReal = fMult(FDK_sbrDecoder_sbr_randomPhase[index][0], smoothedNoise); noiseImag = fMult(FDK_sbrDecoder_sbr_randomPhase[index][1], smoothedNoise); *ptrReal++ = (signalReal + noiseReal); *ptrImag++ = (signalImag + noiseImag); } } } else { for (k = 0; k < noSubbands; k++) { smoothedGain = gain[k]; signalReal = fMax(fMin(fMultDiv2(*ptrReal, smoothedGain), max_val), min_val) << scale_change; signalImag = fMax(fMin(fMultDiv2(*ptrImag, smoothedGain), max_val), min_val) << scale_change; index++; if ((pSineLevel[k] == FL2FXCONST_DBL(0.0f)) && (noNoiseFlag == 0)) { /* Add noisefloor to the amplified signal */ smoothedNoise = noiseLevel[k]; index &= (SBR_NF_NO_RANDOM_VAL - 1); noiseReal = fMult(FDK_sbrDecoder_sbr_randomPhase[index][0], smoothedNoise); noiseImag = fMult(FDK_sbrDecoder_sbr_randomPhase[index][1], smoothedNoise); signalReal += noiseReal; signalImag += noiseImag; } *ptrReal++ = signalReal; *ptrImag++ = signalImag; } } } static void adjustTimeSlotHQ_AddHarmonics( FIXP_DBL *RESTRICT ptrReal, /*!< Subband samples to be adjusted, real part */ FIXP_DBL *RESTRICT ptrImag, /*!< Subband samples to be adjusted, imag part */ HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env, ENV_CALC_NRGS *nrgs, int lowSubband, /*!< Lowest QMF-channel in the currently used SBR range. */ int noSubbands, /*!< Number of QMF subbands */ int scale_change /*!< Scale mismatch between QMF input and sineLevel exponent. */ ) { FIXP_DBL *RESTRICT pSineLevel = nrgs->nrgSine; /*!< Sine levels */ UCHAR *RESTRICT ptrHarmIndex = &h_sbr_cal_env->harmIndex; /*!< Harmonic index */ int k; FIXP_DBL signalReal, signalImag; UCHAR harmIndex = *ptrHarmIndex; int freqInvFlag = (lowSubband & 1); FIXP_DBL sineLevel; *ptrHarmIndex = (harmIndex + 1) & 3; for (k = 0; k < noSubbands; k++) { sineLevel = pSineLevel[k]; freqInvFlag ^= 1; if (sineLevel != FL2FXCONST_DBL(0.f)) { signalReal = ptrReal[k]; signalImag = ptrImag[k]; sineLevel = scaleValue(sineLevel, scale_change); if (harmIndex & 2) { /* case 2,3 */ sineLevel = -sineLevel; } if (!(harmIndex & 1)) { /* case 0,2: */ ptrReal[k] = signalReal + sineLevel; } else { /* case 1,3 */ if (!freqInvFlag) sineLevel = -sineLevel; ptrImag[k] = signalImag + sineLevel; } } } } static void adjustTimeSlotHQ( FIXP_DBL *RESTRICT ptrReal, /*!< Subband samples to be adjusted, real part */ FIXP_DBL *RESTRICT ptrImag, /*!< Subband samples to be adjusted, imag part */ HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env, ENV_CALC_NRGS *nrgs, int lowSubband, /*!< Lowest QMF-channel in the currently used SBR range. */ int noSubbands, /*!< Number of QMF subbands */ int scale_change, /*!< Number of bits to shift adjusted samples */ FIXP_SGL smooth_ratio, /*!< Impact of last envelope */ int noNoiseFlag, /*!< Start index to random number array */ int filtBufferNoiseShift) /*!< Shift factor of filtBufferNoise */ { FIXP_DBL *RESTRICT gain = nrgs->nrgGain; /*!< Gains of current envelope */ FIXP_DBL *RESTRICT noiseLevel = nrgs->noiseLevel; /*!< Noise levels of current envelope */ FIXP_DBL *RESTRICT pSineLevel = nrgs->nrgSine; /*!< Sine levels */ FIXP_DBL *RESTRICT filtBuffer = h_sbr_cal_env->filtBuffer; /*!< Gains of last envelope */ FIXP_DBL *RESTRICT filtBufferNoise = h_sbr_cal_env->filtBufferNoise; /*!< Noise levels of last envelope */ UCHAR *RESTRICT ptrHarmIndex = &h_sbr_cal_env->harmIndex; /*!< Harmonic index */ int *RESTRICT ptrPhaseIndex = &h_sbr_cal_env->phaseIndex; /*!< Start index to random number array */ int k; FIXP_DBL signalReal, signalImag; FIXP_DBL noiseReal, noiseImag; FIXP_DBL smoothedGain, smoothedNoise; FIXP_SGL direct_ratio = /*FL2FXCONST_SGL(1.0f) */ (FIXP_SGL)MAXVAL_SGL - smooth_ratio; int index = *ptrPhaseIndex; UCHAR harmIndex = *ptrHarmIndex; int freqInvFlag = (lowSubband & 1); FIXP_DBL sineLevel; int shift; FIXP_DBL max_val_noise = 0, min_val_noise = 0; const FIXP_DBL max_val = MAX_VAL_NRG_HEADROOM >> scale_change; const FIXP_DBL min_val = -max_val; *ptrPhaseIndex = (index + noSubbands) & (SBR_NF_NO_RANDOM_VAL - 1); *ptrHarmIndex = (harmIndex + 1) & 3; /* Possible optimization: smooth_ratio and harmIndex stay constant during the loop. It might be faster to include a separate loop in each path. the check for smooth_ratio is now outside the loop and the workload of the whole function decreased by about 20 % */ filtBufferNoiseShift += 1; /* due to later use of fMultDiv2 instead of fMult */ if (filtBufferNoiseShift < 0) { shift = fixMin(DFRACT_BITS - 1, -filtBufferNoiseShift); } else { shift = fixMin(DFRACT_BITS - 1, filtBufferNoiseShift); max_val_noise = MAX_VAL_NRG_HEADROOM >> shift; min_val_noise = -max_val_noise; } if (smooth_ratio > FL2FXCONST_SGL(0.0f)) { for (k = 0; k < noSubbands; k++) { /* Smoothing: The old envelope has been bufferd and a certain ratio of the old gains and noise levels is used. */ smoothedGain = fMult(smooth_ratio, filtBuffer[k]) + fMult(direct_ratio, gain[k]); if (filtBufferNoiseShift < 0) { smoothedNoise = (fMultDiv2(smooth_ratio, filtBufferNoise[k]) >> shift) + fMult(direct_ratio, noiseLevel[k]); } else { smoothedNoise = fMultDiv2(smooth_ratio, filtBufferNoise[k]); smoothedNoise = (fMax(fMin(smoothedNoise, max_val_noise), min_val_noise) << shift) + fMult(direct_ratio, noiseLevel[k]); } /* The next 2 multiplications constitute the actual envelope adjustment of the signal and should be carried out with full accuracy (supplying #DFRACT_BITS valid bits). */ signalReal = fMax(fMin(fMultDiv2(*ptrReal, smoothedGain), max_val), min_val) << scale_change; signalImag = fMax(fMin(fMultDiv2(*ptrImag, smoothedGain), max_val), min_val) << scale_change; index++; if (pSineLevel[k] != FL2FXCONST_DBL(0.0f)) { sineLevel = pSineLevel[k]; switch (harmIndex) { case 0: *ptrReal++ = (signalReal + sineLevel); *ptrImag++ = (signalImag); break; case 2: *ptrReal++ = (signalReal - sineLevel); *ptrImag++ = (signalImag); break; case 1: *ptrReal++ = (signalReal); if (freqInvFlag) *ptrImag++ = (signalImag - sineLevel); else *ptrImag++ = (signalImag + sineLevel); break; case 3: *ptrReal++ = signalReal; if (freqInvFlag) *ptrImag++ = (signalImag + sineLevel); else *ptrImag++ = (signalImag - sineLevel); break; } } else { if (noNoiseFlag) { /* Just the amplified signal is saved */ *ptrReal++ = (signalReal); *ptrImag++ = (signalImag); } else { /* Add noisefloor to the amplified signal */ index &= (SBR_NF_NO_RANDOM_VAL - 1); noiseReal = fMult(FDK_sbrDecoder_sbr_randomPhase[index][0], smoothedNoise); noiseImag = fMult(FDK_sbrDecoder_sbr_randomPhase[index][1], smoothedNoise); *ptrReal++ = (signalReal + noiseReal); *ptrImag++ = (signalImag + noiseImag); } } freqInvFlag ^= 1; } } else { for (k = 0; k < noSubbands; k++) { smoothedGain = gain[k]; signalReal = fMax(fMin(fMultDiv2(*ptrReal, smoothedGain), max_val), min_val) << scale_change; signalImag = fMax(fMin(fMultDiv2(*ptrImag, smoothedGain), max_val), min_val) << scale_change; index++; if ((sineLevel = pSineLevel[k]) != FL2FXCONST_DBL(0.0f)) { switch (harmIndex) { case 0: signalReal += sineLevel; break; case 1: if (freqInvFlag) signalImag -= sineLevel; else signalImag += sineLevel; break; case 2: signalReal -= sineLevel; break; case 3: if (freqInvFlag) signalImag += sineLevel; else signalImag -= sineLevel; break; } } else { if (noNoiseFlag == 0) { /* Add noisefloor to the amplified signal */ smoothedNoise = noiseLevel[k]; index &= (SBR_NF_NO_RANDOM_VAL - 1); noiseReal = fMult(FDK_sbrDecoder_sbr_randomPhase[index][0], smoothedNoise); noiseImag = fMult(FDK_sbrDecoder_sbr_randomPhase[index][1], smoothedNoise); signalReal += noiseReal; signalImag += noiseImag; } } *ptrReal++ = signalReal; *ptrImag++ = signalImag; freqInvFlag ^= 1; } } } /*! \brief Reset limiter bands. Build frequency band table for the gain limiter dependent on the previously generated transposer patch areas. \return SBRDEC_OK if ok, SBRDEC_UNSUPPORTED_CONFIG on error */ SBR_ERROR ResetLimiterBands( UCHAR *limiterBandTable, /*!< Resulting band borders in QMF channels */ UCHAR *noLimiterBands, /*!< Resulting number of limiter band */ UCHAR *freqBandTable, /*!< Table with possible band borders */ int noFreqBands, /*!< Number of bands in freqBandTable */ const PATCH_PARAM *patchParam, /*!< Transposer patch parameters */ int noPatches, /*!< Number of transposer patches */ int limiterBands, /*!< Selected 'band density' from bitstream */ UCHAR sbrPatchingMode, int xOverQmf[MAX_NUM_PATCHES], int b41Sbr) { int i, k, isPatchBorder[2], loLimIndex, hiLimIndex, tempNoLim, nBands; UCHAR workLimiterBandTable[MAX_FREQ_COEFFS / 2 + MAX_NUM_PATCHES + 1]; int patchBorders[MAX_NUM_PATCHES + 1]; int kx, k2; int lowSubband = freqBandTable[0]; int highSubband = freqBandTable[noFreqBands]; /* 1 limiter band. */ if (limiterBands == 0) { limiterBandTable[0] = 0; limiterBandTable[1] = highSubband - lowSubband; nBands = 1; } else { if (!sbrPatchingMode && xOverQmf != NULL) { noPatches = 0; if (b41Sbr == 1) { for (i = 1; i < MAX_NUM_PATCHES_HBE; i++) if (xOverQmf[i] != 0) noPatches++; } else { for (i = 1; i < MAX_STRETCH_HBE; i++) if (xOverQmf[i] != 0) noPatches++; } for (i = 0; i < noPatches; i++) { patchBorders[i] = xOverQmf[i] - lowSubband; } } else { for (i = 0; i < noPatches; i++) { patchBorders[i] = patchParam[i].guardStartBand - lowSubband; } } patchBorders[i] = highSubband - lowSubband; /* 1.2, 2, or 3 limiter bands/octave plus bandborders at patchborders. */ for (k = 0; k <= noFreqBands; k++) { workLimiterBandTable[k] = freqBandTable[k] - lowSubband; } for (k = 1; k < noPatches; k++) { workLimiterBandTable[noFreqBands + k] = patchBorders[k]; } tempNoLim = nBands = noFreqBands + noPatches - 1; shellsort(workLimiterBandTable, tempNoLim + 1); loLimIndex = 0; hiLimIndex = 1; while (hiLimIndex <= tempNoLim) { FIXP_DBL div_m, oct_m, temp; INT div_e = 0, oct_e = 0, temp_e = 0; k2 = workLimiterBandTable[hiLimIndex] + lowSubband; kx = workLimiterBandTable[loLimIndex] + lowSubband; div_m = fDivNorm(k2, kx, &div_e); /* calculate number of octaves */ oct_m = fLog2(div_m, div_e, &oct_e); /* multiply with limiterbands per octave */ /* values 1, 1.2, 2, 3 -> scale factor of 2 */ temp = fMultNorm( oct_m, FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4_DBL[limiterBands], &temp_e); /* overall scale factor of temp ist addition of scalefactors from log2 calculation, limiter bands scalefactor (2) and limiter bands multiplication */ temp_e += oct_e + 2; /* div can be a maximum of 64 (k2 = 64 and kx = 1) -> oct can be a maximum of 6 -> temp can be a maximum of 18 (as limiterBandsPerOctoave is a maximum factor of 3) -> we need a scale factor of 5 for comparisson */ if (temp >> (5 - temp_e) < FL2FXCONST_DBL(0.49f) >> 5) { if (workLimiterBandTable[hiLimIndex] == workLimiterBandTable[loLimIndex]) { workLimiterBandTable[hiLimIndex] = highSubband; nBands--; hiLimIndex++; continue; } isPatchBorder[0] = isPatchBorder[1] = 0; for (k = 0; k <= noPatches; k++) { if (workLimiterBandTable[hiLimIndex] == patchBorders[k]) { isPatchBorder[1] = 1; break; } } if (!isPatchBorder[1]) { workLimiterBandTable[hiLimIndex] = highSubband; nBands--; hiLimIndex++; continue; } for (k = 0; k <= noPatches; k++) { if (workLimiterBandTable[loLimIndex] == patchBorders[k]) { isPatchBorder[0] = 1; break; } } if (!isPatchBorder[0]) { workLimiterBandTable[loLimIndex] = highSubband; nBands--; } } loLimIndex = hiLimIndex; hiLimIndex++; } shellsort(workLimiterBandTable, tempNoLim + 1); /* Test if algorithm exceeded maximum allowed limiterbands */ if (nBands > MAX_NUM_LIMITERS || nBands <= 0) { return SBRDEC_UNSUPPORTED_CONFIG; } /* Restrict maximum value of limiter band table */ if (workLimiterBandTable[tempNoLim] > highSubband) { return SBRDEC_UNSUPPORTED_CONFIG; } /* Copy limiterbands from working buffer into final destination */ for (k = 0; k <= nBands; k++) { limiterBandTable[k] = workLimiterBandTable[k]; } } *noLimiterBands = nBands; return SBRDEC_OK; } fdk-aac-2.0.2/libSBRdec/src/env_calc.h000066400000000000000000000175361404225310700173160ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Envelope calculation prototypes */ #ifndef ENV_CALC_H #define ENV_CALC_H #include "sbrdecoder.h" #include "env_extr.h" /* for HANDLE_SBR_HEADER_DATA */ typedef struct { FIXP_DBL filtBuffer[MAX_FREQ_COEFFS]; /*!< previous gains (required for smoothing) */ FIXP_DBL filtBufferNoise[MAX_FREQ_COEFFS]; /*!< previous noise levels (required for smoothing) */ SCHAR filtBuffer_e[MAX_FREQ_COEFFS]; /*!< Exponents of previous gains */ SCHAR filtBufferNoise_e; /*!< Common exponent of previous noise levels */ int startUp; /*!< flag to signal initial conditions in buffers */ int phaseIndex; /*!< Index for randomPase array */ int prevTranEnv; /*!< The transient envelope of the previous frame. */ ULONG harmFlagsPrev[ADD_HARMONICS_FLAGS_SIZE]; /*!< Words with 16 flags each indicating where a sine was added in the * previous frame.*/ UCHAR harmIndex; /*!< Current phase of synthetic sine */ int sbrPatchingMode; /*!< Current patching mode */ FIXP_SGL prevSbrNoiseFloorLevel[MAX_NOISE_COEFFS]; UCHAR prevNNfb; UCHAR prevNSfb[2]; UCHAR prevLoSubband; UCHAR prevHiSubband; UCHAR prev_ov_highSubband; UCHAR *prevFreqBandTable[2]; UCHAR prevFreqBandTableLo[MAX_FREQ_COEFFS / 2 + 1]; UCHAR prevFreqBandTableHi[MAX_FREQ_COEFFS + 1]; UCHAR prevFreqBandTableNoise[MAX_NOISE_COEFFS + 1]; SCHAR sinusoidal_positionPrev; ULONG harmFlagsPrevActive[ADD_HARMONICS_FLAGS_SIZE]; } SBR_CALCULATE_ENVELOPE; typedef SBR_CALCULATE_ENVELOPE *HANDLE_SBR_CALCULATE_ENVELOPE; void calculateSbrEnvelope( QMF_SCALE_FACTOR *sbrScaleFactor, HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env, HANDLE_SBR_HEADER_DATA hHeaderData, HANDLE_SBR_FRAME_DATA hFrameData, PVC_DYNAMIC_DATA *pPvcDynamicData, FIXP_DBL **analysBufferReal, FIXP_DBL * *analysBufferImag, /*!< Imag part of subband samples to be processed */ const int useLP, FIXP_DBL *degreeAlias, /*!< Estimated aliasing for each QMF channel */ const UINT flags, const int frameErrorFlag); SBR_ERROR createSbrEnvelopeCalc(HANDLE_SBR_CALCULATE_ENVELOPE hSbrCalculateEnvelope, HANDLE_SBR_HEADER_DATA hHeaderData, const int chan, const UINT flags); int deleteSbrEnvelopeCalc(HANDLE_SBR_CALCULATE_ENVELOPE hSbrCalculateEnvelope); void resetSbrEnvelopeCalc(HANDLE_SBR_CALCULATE_ENVELOPE hCalEnv); SBR_ERROR ResetLimiterBands(UCHAR *limiterBandTable, UCHAR *noLimiterBands, UCHAR *freqBandTable, int noFreqBands, const PATCH_PARAM *patchParam, int noPatches, int limiterBands, UCHAR sbrPatchingMode, int xOverQmf[MAX_NUM_PATCHES], int sbrRatio); void rescaleSubbandSamples(FIXP_DBL **re, FIXP_DBL **im, int lowSubband, int noSubbands, int start_pos, int next_pos, int shift); FIXP_DBL maxSubbandSample(FIXP_DBL **analysBufferReal_m, FIXP_DBL **analysBufferImag_m, int lowSubband, int highSubband, int start_pos, int stop_pos); #endif // ENV_CALC_H fdk-aac-2.0.2/libSBRdec/src/env_dec.cpp000066400000000000000000000776741404225310700175130ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief envelope decoding This module provides envelope decoding and error concealment algorithms. The main entry point is decodeSbrData(). \sa decodeSbrData(),\ref documentationOverview */ #include "env_dec.h" #include "env_extr.h" #include "transcendent.h" #include "genericStds.h" static void decodeEnvelope(HANDLE_SBR_HEADER_DATA hHeaderData, HANDLE_SBR_FRAME_DATA h_sbr_data, HANDLE_SBR_PREV_FRAME_DATA h_prev_data, HANDLE_SBR_PREV_FRAME_DATA h_prev_data_otherChannel); static void sbr_envelope_unmapping(HANDLE_SBR_HEADER_DATA hHeaderData, HANDLE_SBR_FRAME_DATA h_data_left, HANDLE_SBR_FRAME_DATA h_data_right); static void requantizeEnvelopeData(HANDLE_SBR_FRAME_DATA h_sbr_data, int ampResolution); static void deltaToLinearPcmEnvelopeDecoding( HANDLE_SBR_HEADER_DATA hHeaderData, HANDLE_SBR_FRAME_DATA h_sbr_data, HANDLE_SBR_PREV_FRAME_DATA h_prev_data); static void decodeNoiseFloorlevels(HANDLE_SBR_HEADER_DATA hHeaderData, HANDLE_SBR_FRAME_DATA h_sbr_data, HANDLE_SBR_PREV_FRAME_DATA h_prev_data); static void timeCompensateFirstEnvelope(HANDLE_SBR_HEADER_DATA hHeaderData, HANDLE_SBR_FRAME_DATA h_sbr_data, HANDLE_SBR_PREV_FRAME_DATA h_prev_data); static int checkEnvelopeData(HANDLE_SBR_HEADER_DATA hHeaderData, HANDLE_SBR_FRAME_DATA h_sbr_data, HANDLE_SBR_PREV_FRAME_DATA h_prev_data); #define SBR_ENERGY_PAN_OFFSET (12 << ENV_EXP_FRACT) #define SBR_MAX_ENERGY (35 << ENV_EXP_FRACT) #define DECAY (1 << ENV_EXP_FRACT) #if ENV_EXP_FRACT #define DECAY_COUPLING \ (1 << (ENV_EXP_FRACT - 1)) /*!< corresponds to a value of 0.5 */ #else #define DECAY_COUPLING \ 1 /*!< If the energy data is not shifted, use 1 instead of 0.5 */ #endif /*! \brief Convert table index */ static int indexLow2High(int offset, /*!< mapping factor */ int index, /*!< index to scalefactor band */ int res) /*!< frequency resolution */ { if (res == 0) { if (offset >= 0) { if (index < offset) return (index); else return (2 * index - offset); } else { offset = -offset; if (index < offset) return (2 * index + index); else return (2 * index + offset); } } else return (index); } /*! \brief Update previous envelope value for delta-coding The current envelope values needs to be stored for delta-coding in the next frame. The stored envelope is always represented with the high frequency resolution. If the current envelope uses the low frequency resolution, the energy value will be mapped to the corresponding high-res bands. */ static void mapLowResEnergyVal( FIXP_SGL currVal, /*!< current energy value */ FIXP_SGL *prevData, /*!< pointer to previous data vector */ int offset, /*!< mapping factor */ int index, /*!< index to scalefactor band */ int res) /*!< frequeny resolution */ { if (res == 0) { if (offset >= 0) { if (index < offset) prevData[index] = currVal; else { prevData[2 * index - offset] = currVal; prevData[2 * index + 1 - offset] = currVal; } } else { offset = -offset; if (index < offset) { prevData[3 * index] = currVal; prevData[3 * index + 1] = currVal; prevData[3 * index + 2] = currVal; } else { prevData[2 * index + offset] = currVal; prevData[2 * index + 1 + offset] = currVal; } } } else prevData[index] = currVal; } /*! \brief Convert raw envelope and noisefloor data to energy levels This function is being called by sbrDecoder_ParseElement() and provides two important algorithms: First the function decodes envelopes and noise floor levels as described in requantizeEnvelopeData() and sbr_envelope_unmapping(). The function also implements concealment algorithms in case there are errors within the sbr data. For both operations fractional arithmetic is used. Therefore you might encounter different output values on your target system compared to the reference implementation. */ void decodeSbrData( HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ HANDLE_SBR_FRAME_DATA h_data_left, /*!< pointer to left channel frame data */ HANDLE_SBR_PREV_FRAME_DATA h_prev_data_left, /*!< pointer to left channel previous frame data */ HANDLE_SBR_FRAME_DATA h_data_right, /*!< pointer to right channel frame data */ HANDLE_SBR_PREV_FRAME_DATA h_prev_data_right) /*!< pointer to right channel previous frame data */ { FIXP_SGL tempSfbNrgPrev[MAX_FREQ_COEFFS]; int errLeft; /* Save previous energy values to be able to reuse them later for concealment. */ FDKmemcpy(tempSfbNrgPrev, h_prev_data_left->sfb_nrg_prev, MAX_FREQ_COEFFS * sizeof(FIXP_SGL)); if (hHeaderData->frameErrorFlag || hHeaderData->bs_info.pvc_mode == 0) { decodeEnvelope(hHeaderData, h_data_left, h_prev_data_left, h_prev_data_right); } else { FDK_ASSERT(h_data_right == NULL); } decodeNoiseFloorlevels(hHeaderData, h_data_left, h_prev_data_left); if (h_data_right != NULL) { errLeft = hHeaderData->frameErrorFlag; decodeEnvelope(hHeaderData, h_data_right, h_prev_data_right, h_prev_data_left); decodeNoiseFloorlevels(hHeaderData, h_data_right, h_prev_data_right); if (!errLeft && hHeaderData->frameErrorFlag) { /* If an error occurs in the right channel where the left channel seemed ok, we apply concealment also on the left channel. This ensures that the coupling modes of both channels match and that we have the same number of envelopes in coupling mode. However, as the left channel has already been processed before, the resulting energy levels are not the same as if the left channel had been concealed during the first call of decodeEnvelope(). */ /* Restore previous energy values for concealment, because the values have been overwritten by the first call of decodeEnvelope(). */ FDKmemcpy(h_prev_data_left->sfb_nrg_prev, tempSfbNrgPrev, MAX_FREQ_COEFFS * sizeof(FIXP_SGL)); /* Do concealment */ decodeEnvelope(hHeaderData, h_data_left, h_prev_data_left, h_prev_data_right); } if (h_data_left->coupling) { sbr_envelope_unmapping(hHeaderData, h_data_left, h_data_right); } } /* Display the data for debugging: */ } /*! \brief Convert from coupled channels to independent L/R data */ static void sbr_envelope_unmapping( HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ HANDLE_SBR_FRAME_DATA h_data_left, /*!< pointer to left channel */ HANDLE_SBR_FRAME_DATA h_data_right) /*!< pointer to right channel */ { int i; FIXP_SGL tempL_m, tempR_m, tempRplus1_m, newL_m, newR_m; SCHAR tempL_e, tempR_e, tempRplus1_e, newL_e, newR_e; /* 1. Unmap (already dequantized) coupled envelope energies */ for (i = 0; i < h_data_left->nScaleFactors; i++) { tempR_m = (FIXP_SGL)((LONG)h_data_right->iEnvelope[i] & MASK_M); tempR_e = (SCHAR)((LONG)h_data_right->iEnvelope[i] & MASK_E); tempR_e -= (18 + NRG_EXP_OFFSET); /* -18 = ld(UNMAPPING_SCALE / h_data_right->nChannels) */ tempL_m = (FIXP_SGL)((LONG)h_data_left->iEnvelope[i] & MASK_M); tempL_e = (SCHAR)((LONG)h_data_left->iEnvelope[i] & MASK_E); tempL_e -= NRG_EXP_OFFSET; /* Calculate tempRight+1 */ FDK_add_MantExp(tempR_m, tempR_e, FL2FXCONST_SGL(0.5f), 1, /* 1.0 */ &tempRplus1_m, &tempRplus1_e); FDK_divide_MantExp(tempL_m, tempL_e + 1, /* 2 * tempLeft */ tempRplus1_m, tempRplus1_e, &newR_m, &newR_e); if (newR_m >= ((FIXP_SGL)MAXVAL_SGL - ROUNDING)) { newR_m >>= 1; newR_e += 1; } newL_m = FX_DBL2FX_SGL(fMult(tempR_m, newR_m)); newL_e = tempR_e + newR_e; h_data_right->iEnvelope[i] = ((FIXP_SGL)((SHORT)(FIXP_SGL)(newR_m + ROUNDING) & MASK_M)) + (FIXP_SGL)((SHORT)(FIXP_SGL)(newR_e + NRG_EXP_OFFSET) & MASK_E); h_data_left->iEnvelope[i] = ((FIXP_SGL)((SHORT)(FIXP_SGL)(newL_m + ROUNDING) & MASK_M)) + (FIXP_SGL)((SHORT)(FIXP_SGL)(newL_e + NRG_EXP_OFFSET) & MASK_E); } /* 2. Dequantize and unmap coupled noise floor levels */ for (i = 0; i < hHeaderData->freqBandData.nNfb * h_data_left->frameInfo.nNoiseEnvelopes; i++) { tempL_e = (SCHAR)(6 - (LONG)h_data_left->sbrNoiseFloorLevel[i]); tempR_e = (SCHAR)((LONG)h_data_right->sbrNoiseFloorLevel[i] - 12) /*SBR_ENERGY_PAN_OFFSET*/; /* Calculate tempR+1 */ FDK_add_MantExp(FL2FXCONST_SGL(0.5f), 1 + tempR_e, /* tempR */ FL2FXCONST_SGL(0.5f), 1, /* 1.0 */ &tempRplus1_m, &tempRplus1_e); /* Calculate 2*tempLeft/(tempR+1) */ FDK_divide_MantExp(FL2FXCONST_SGL(0.5f), tempL_e + 2, /* 2 * tempLeft */ tempRplus1_m, tempRplus1_e, &newR_m, &newR_e); /* if (newR_m >= ((FIXP_SGL)MAXVAL_SGL - ROUNDING)) { newR_m >>= 1; newR_e += 1; } */ /* L = tempR * R */ newL_m = newR_m; newL_e = newR_e + tempR_e; h_data_right->sbrNoiseFloorLevel[i] = ((FIXP_SGL)((SHORT)(FIXP_SGL)(newR_m + ROUNDING) & MASK_M)) + (FIXP_SGL)((SHORT)(FIXP_SGL)(newR_e + NOISE_EXP_OFFSET) & MASK_E); h_data_left->sbrNoiseFloorLevel[i] = ((FIXP_SGL)((SHORT)(FIXP_SGL)(newL_m + ROUNDING) & MASK_M)) + (FIXP_SGL)((SHORT)(FIXP_SGL)(newL_e + NOISE_EXP_OFFSET) & MASK_E); } } /*! \brief Simple alternative to the real SBR concealment If the real frameInfo is not available due to a frame loss, a replacement will be constructed with 1 envelope spanning the whole frame (FIX-FIX). The delta-coded energies are set to negative values, resulting in a fade-down. In case of coupling, the balance-channel will move towards the center. */ static void leanSbrConcealment( HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ HANDLE_SBR_FRAME_DATA h_sbr_data, /*!< pointer to current data */ HANDLE_SBR_PREV_FRAME_DATA h_prev_data /*!< pointer to data of last frame */ ) { FIXP_SGL target; /* targeted level for sfb_nrg_prev during fade-down */ FIXP_SGL step; /* speed of fade */ int i; int currentStartPos = fMax(0, h_prev_data->stopPos - hHeaderData->numberTimeSlots); int currentStopPos = hHeaderData->numberTimeSlots; /* Use some settings of the previous frame */ h_sbr_data->ampResolutionCurrentFrame = h_prev_data->ampRes; h_sbr_data->coupling = h_prev_data->coupling; for (i = 0; i < MAX_INVF_BANDS; i++) h_sbr_data->sbr_invf_mode[i] = h_prev_data->sbr_invf_mode[i]; /* Generate concealing control data */ h_sbr_data->frameInfo.nEnvelopes = 1; h_sbr_data->frameInfo.borders[0] = currentStartPos; h_sbr_data->frameInfo.borders[1] = currentStopPos; h_sbr_data->frameInfo.freqRes[0] = 1; h_sbr_data->frameInfo.tranEnv = -1; /* no transient */ h_sbr_data->frameInfo.nNoiseEnvelopes = 1; h_sbr_data->frameInfo.bordersNoise[0] = currentStartPos; h_sbr_data->frameInfo.bordersNoise[1] = currentStopPos; h_sbr_data->nScaleFactors = hHeaderData->freqBandData.nSfb[1]; /* Generate fake envelope data */ h_sbr_data->domain_vec[0] = 1; if (h_sbr_data->coupling == COUPLING_BAL) { target = (FIXP_SGL)SBR_ENERGY_PAN_OFFSET; step = (FIXP_SGL)DECAY_COUPLING; } else { target = FL2FXCONST_SGL(0.0f); step = (FIXP_SGL)DECAY; } if (hHeaderData->bs_info.ampResolution == 0) { target <<= 1; step <<= 1; } for (i = 0; i < h_sbr_data->nScaleFactors; i++) { if (h_prev_data->sfb_nrg_prev[i] > target) h_sbr_data->iEnvelope[i] = -step; else h_sbr_data->iEnvelope[i] = step; } /* Noisefloor levels are always cleared ... */ h_sbr_data->domain_vec_noise[0] = 1; FDKmemclear(h_sbr_data->sbrNoiseFloorLevel, sizeof(h_sbr_data->sbrNoiseFloorLevel)); /* ... and so are the sines */ FDKmemclear(h_sbr_data->addHarmonics, sizeof(ULONG) * ADD_HARMONICS_FLAGS_SIZE); } /*! \brief Build reference energies and noise levels from bitstream elements */ static void decodeEnvelope( HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ HANDLE_SBR_FRAME_DATA h_sbr_data, /*!< pointer to current data */ HANDLE_SBR_PREV_FRAME_DATA h_prev_data, /*!< pointer to data of last frame */ HANDLE_SBR_PREV_FRAME_DATA otherChannel /*!< other channel's last frame data */ ) { int i; int fFrameError = hHeaderData->frameErrorFlag; FIXP_SGL tempSfbNrgPrev[MAX_FREQ_COEFFS]; if (!fFrameError) { /* To avoid distortions after bad frames, set the error flag if delta coding in time occurs. However, SBR can take a little longer to come up again. */ if (h_prev_data->frameErrorFlag) { if (h_sbr_data->domain_vec[0] != 0) { fFrameError = 1; } } else { /* Check that the previous stop position and the current start position match. (Could be done in checkFrameInfo(), but the previous frame data is not available there) */ if (h_sbr_data->frameInfo.borders[0] != h_prev_data->stopPos - hHeaderData->numberTimeSlots) { /* Both the previous as well as the current frame are flagged to be ok, * but they do not match! */ if (h_sbr_data->domain_vec[0] == 1) { /* Prefer concealment over delta-time coding between the mismatching * frames */ fFrameError = 1; } else { /* Close the gap in time by triggering timeCompensateFirstEnvelope() */ fFrameError = 1; } } } } if (fFrameError) /* Error is detected */ { leanSbrConcealment(hHeaderData, h_sbr_data, h_prev_data); /* decode the envelope data to linear PCM */ deltaToLinearPcmEnvelopeDecoding(hHeaderData, h_sbr_data, h_prev_data); } else /*Do a temporary dummy decoding and check that the envelope values are within limits */ { if (h_prev_data->frameErrorFlag) { timeCompensateFirstEnvelope(hHeaderData, h_sbr_data, h_prev_data); if (h_sbr_data->coupling != h_prev_data->coupling) { /* Coupling mode has changed during concealment. The stored energy levels need to be converted. */ for (i = 0; i < hHeaderData->freqBandData.nSfb[1]; i++) { /* Former Level-Channel will be used for both channels */ if (h_prev_data->coupling == COUPLING_BAL) { h_prev_data->sfb_nrg_prev[i] = (otherChannel != NULL) ? otherChannel->sfb_nrg_prev[i] : (FIXP_SGL)SBR_ENERGY_PAN_OFFSET; } /* Former L/R will be combined as the new Level-Channel */ else if (h_sbr_data->coupling == COUPLING_LEVEL && otherChannel != NULL) { h_prev_data->sfb_nrg_prev[i] = (h_prev_data->sfb_nrg_prev[i] + otherChannel->sfb_nrg_prev[i]) >> 1; } else if (h_sbr_data->coupling == COUPLING_BAL) { h_prev_data->sfb_nrg_prev[i] = (FIXP_SGL)SBR_ENERGY_PAN_OFFSET; } } } } FDKmemcpy(tempSfbNrgPrev, h_prev_data->sfb_nrg_prev, MAX_FREQ_COEFFS * sizeof(FIXP_SGL)); deltaToLinearPcmEnvelopeDecoding(hHeaderData, h_sbr_data, h_prev_data); fFrameError = checkEnvelopeData(hHeaderData, h_sbr_data, h_prev_data); if (fFrameError) { hHeaderData->frameErrorFlag = 1; FDKmemcpy(h_prev_data->sfb_nrg_prev, tempSfbNrgPrev, MAX_FREQ_COEFFS * sizeof(FIXP_SGL)); decodeEnvelope(hHeaderData, h_sbr_data, h_prev_data, otherChannel); return; } } requantizeEnvelopeData(h_sbr_data, h_sbr_data->ampResolutionCurrentFrame); hHeaderData->frameErrorFlag = fFrameError; } /*! \brief Verify that envelope energies are within the allowed range \return 0 if all is fine, 1 if an envelope value was too high */ static int checkEnvelopeData( HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ HANDLE_SBR_FRAME_DATA h_sbr_data, /*!< pointer to current data */ HANDLE_SBR_PREV_FRAME_DATA h_prev_data /*!< pointer to data of last frame */ ) { FIXP_SGL *iEnvelope = h_sbr_data->iEnvelope; FIXP_SGL *sfb_nrg_prev = h_prev_data->sfb_nrg_prev; int i = 0, errorFlag = 0; FIXP_SGL sbr_max_energy = (h_sbr_data->ampResolutionCurrentFrame == 1) ? SBR_MAX_ENERGY : (SBR_MAX_ENERGY << 1); /* Range check for current energies */ for (i = 0; i < h_sbr_data->nScaleFactors; i++) { if (iEnvelope[i] > sbr_max_energy) { errorFlag = 1; } if (iEnvelope[i] < FL2FXCONST_SGL(0.0f)) { errorFlag = 1; /* iEnvelope[i] = FL2FXCONST_SGL(0.0f); */ } } /* Range check for previous energies */ for (i = 0; i < hHeaderData->freqBandData.nSfb[1]; i++) { sfb_nrg_prev[i] = fixMax(sfb_nrg_prev[i], FL2FXCONST_SGL(0.0f)); sfb_nrg_prev[i] = fixMin(sfb_nrg_prev[i], sbr_max_energy); } return (errorFlag); } /*! \brief Verify that the noise levels are within the allowed range The function is equivalent to checkEnvelopeData(). When the noise-levels are being decoded, it is already too late for concealment. Therefore the noise levels are simply limited here. */ static void limitNoiseLevels( HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ HANDLE_SBR_FRAME_DATA h_sbr_data) /*!< pointer to current data */ { int i; int nNfb = hHeaderData->freqBandData.nNfb; /* Set range limits. The exact values depend on the coupling mode. However this limitation is primarily intended to avoid unlimited accumulation of the delta-coded noise levels. */ #define lowerLimit \ ((FIXP_SGL)0) /* lowerLimit actually refers to the _highest_ noise energy */ #define upperLimit \ ((FIXP_SGL)35) /* upperLimit actually refers to the _lowest_ noise energy */ /* Range check for current noise levels */ for (i = 0; i < h_sbr_data->frameInfo.nNoiseEnvelopes * nNfb; i++) { h_sbr_data->sbrNoiseFloorLevel[i] = fixMin(h_sbr_data->sbrNoiseFloorLevel[i], upperLimit); h_sbr_data->sbrNoiseFloorLevel[i] = fixMax(h_sbr_data->sbrNoiseFloorLevel[i], lowerLimit); } } /*! \brief Compensate for the wrong timing that might occur after a frame error. */ static void timeCompensateFirstEnvelope( HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ HANDLE_SBR_FRAME_DATA h_sbr_data, /*!< pointer to actual data */ HANDLE_SBR_PREV_FRAME_DATA h_prev_data) /*!< pointer to data of last frame */ { int i, nScalefactors; FRAME_INFO *pFrameInfo = &h_sbr_data->frameInfo; UCHAR *nSfb = hHeaderData->freqBandData.nSfb; int estimatedStartPos = fMax(0, h_prev_data->stopPos - hHeaderData->numberTimeSlots); int refLen, newLen, shift; FIXP_SGL deltaExp; /* Original length of first envelope according to bitstream */ refLen = pFrameInfo->borders[1] - pFrameInfo->borders[0]; /* Corrected length of first envelope (concealing can make the first envelope * longer) */ newLen = pFrameInfo->borders[1] - estimatedStartPos; if (newLen <= 0) { /* An envelope length of <= 0 would not work, so we don't use it. May occur if the previous frame was flagged bad due to a mismatch of the old and new frame infos. */ newLen = refLen; estimatedStartPos = pFrameInfo->borders[0]; } deltaExp = FDK_getNumOctavesDiv8(newLen, refLen); /* Shift by -3 to rescale ld-table, ampRes-1 to enable coarser steps */ shift = (FRACT_BITS - 1 - ENV_EXP_FRACT - 1 + h_sbr_data->ampResolutionCurrentFrame - 3); deltaExp = deltaExp >> shift; pFrameInfo->borders[0] = estimatedStartPos; pFrameInfo->bordersNoise[0] = estimatedStartPos; if (h_sbr_data->coupling != COUPLING_BAL) { nScalefactors = (pFrameInfo->freqRes[0]) ? nSfb[1] : nSfb[0]; for (i = 0; i < nScalefactors; i++) h_sbr_data->iEnvelope[i] = h_sbr_data->iEnvelope[i] + deltaExp; } } /*! \brief Convert each envelope value from logarithmic to linear domain Energy levels are transmitted in powers of 2, i.e. only the exponent is extracted from the bitstream. Therefore, normally only integer exponents can occur. However during fading (in case of a corrupt bitstream), a fractional part can also occur. The data in the array iEnvelope is shifted left by ENV_EXP_FRACT compared to an integer representation so that numbers smaller than 1 can be represented. This function calculates a mantissa corresponding to the fractional part of the exponent for each reference energy. The array iEnvelope is converted in place to save memory. Input and output data must be interpreted differently, as shown in the below figure: \image html EnvelopeData.png The data is then used in calculateSbrEnvelope(). */ static void requantizeEnvelopeData(HANDLE_SBR_FRAME_DATA h_sbr_data, int ampResolution) { int i; FIXP_SGL mantissa; int ampShift = 1 - ampResolution; int exponent; /* In case that ENV_EXP_FRACT is changed to something else but 0 or 8, the initialization of this array has to be adapted! */ #if ENV_EXP_FRACT static const FIXP_SGL pow2[ENV_EXP_FRACT] = { FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 1))), /* 0.7071 */ FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 2))), /* 0.5946 */ FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 3))), FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 4))), FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 5))), FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 6))), FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 7))), FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 8))) /* 0.5013 */ }; int bit, mask; #endif for (i = 0; i < h_sbr_data->nScaleFactors; i++) { exponent = (LONG)h_sbr_data->iEnvelope[i]; #if ENV_EXP_FRACT exponent = exponent >> ampShift; mantissa = 0.5f; /* Amplify mantissa according to the fractional part of the exponent (result will be between 0.500000 and 0.999999) */ mask = 1; /* begin with lowest bit of exponent */ for (bit = ENV_EXP_FRACT - 1; bit >= 0; bit--) { if (exponent & mask) { /* The current bit of the exponent is set, multiply mantissa with the corresponding factor: */ mantissa = (FIXP_SGL)((mantissa * pow2[bit]) << 1); } /* Advance to next bit */ mask = mask << 1; } /* Make integer part of exponent right aligned */ exponent = exponent >> ENV_EXP_FRACT; #else /* In case of the high amplitude resolution, 1 bit of the exponent gets lost by the shift. This will be compensated by a mantissa of 0.5*sqrt(2) instead of 0.5 if that bit is 1. */ mantissa = (exponent & ampShift) ? FL2FXCONST_SGL(0.707106781186548f) : FL2FXCONST_SGL(0.5f); exponent = exponent >> ampShift; #endif /* Mantissa was set to 0.5 (instead of 1.0, therefore increase exponent by 1). Multiply by L=nChannels=64 by increasing exponent by another 6. => Increase exponent by 7 */ exponent += 7 + NRG_EXP_OFFSET; /* Combine mantissa and exponent and write back the result */ h_sbr_data->iEnvelope[i] = ((FIXP_SGL)((SHORT)(FIXP_SGL)mantissa & MASK_M)) + (FIXP_SGL)((SHORT)(FIXP_SGL)exponent & MASK_E); } } /*! \brief Build new reference energies from old ones and delta coded data */ static void deltaToLinearPcmEnvelopeDecoding( HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ HANDLE_SBR_FRAME_DATA h_sbr_data, /*!< pointer to current data */ HANDLE_SBR_PREV_FRAME_DATA h_prev_data) /*!< pointer to previous data */ { int i, domain, no_of_bands, band, freqRes; FIXP_SGL *sfb_nrg_prev = h_prev_data->sfb_nrg_prev; FIXP_SGL *ptr_nrg = h_sbr_data->iEnvelope; int offset = 2 * hHeaderData->freqBandData.nSfb[0] - hHeaderData->freqBandData.nSfb[1]; for (i = 0; i < h_sbr_data->frameInfo.nEnvelopes; i++) { domain = h_sbr_data->domain_vec[i]; freqRes = h_sbr_data->frameInfo.freqRes[i]; FDK_ASSERT(freqRes >= 0 && freqRes <= 1); no_of_bands = hHeaderData->freqBandData.nSfb[freqRes]; FDK_ASSERT(no_of_bands < (64)); if (domain == 0) { mapLowResEnergyVal(*ptr_nrg, sfb_nrg_prev, offset, 0, freqRes); ptr_nrg++; for (band = 1; band < no_of_bands; band++) { *ptr_nrg = *ptr_nrg + *(ptr_nrg - 1); mapLowResEnergyVal(*ptr_nrg, sfb_nrg_prev, offset, band, freqRes); ptr_nrg++; } } else { for (band = 0; band < no_of_bands; band++) { *ptr_nrg = *ptr_nrg + sfb_nrg_prev[indexLow2High(offset, band, freqRes)]; mapLowResEnergyVal(*ptr_nrg, sfb_nrg_prev, offset, band, freqRes); ptr_nrg++; } } } } /*! \brief Build new noise levels from old ones and delta coded data */ static void decodeNoiseFloorlevels( HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ HANDLE_SBR_FRAME_DATA h_sbr_data, /*!< pointer to current data */ HANDLE_SBR_PREV_FRAME_DATA h_prev_data) /*!< pointer to previous data */ { int i; int nNfb = hHeaderData->freqBandData.nNfb; int nNoiseFloorEnvelopes = h_sbr_data->frameInfo.nNoiseEnvelopes; /* Decode first noise envelope */ if (h_sbr_data->domain_vec_noise[0] == 0) { FIXP_SGL noiseLevel = h_sbr_data->sbrNoiseFloorLevel[0]; for (i = 1; i < nNfb; i++) { noiseLevel += h_sbr_data->sbrNoiseFloorLevel[i]; h_sbr_data->sbrNoiseFloorLevel[i] = noiseLevel; } } else { for (i = 0; i < nNfb; i++) { h_sbr_data->sbrNoiseFloorLevel[i] += h_prev_data->prevNoiseLevel[i]; } } /* If present, decode the second noise envelope Note: nNoiseFloorEnvelopes can only be 1 or 2 */ if (nNoiseFloorEnvelopes > 1) { if (h_sbr_data->domain_vec_noise[1] == 0) { FIXP_SGL noiseLevel = h_sbr_data->sbrNoiseFloorLevel[nNfb]; for (i = nNfb + 1; i < 2 * nNfb; i++) { noiseLevel += h_sbr_data->sbrNoiseFloorLevel[i]; h_sbr_data->sbrNoiseFloorLevel[i] = noiseLevel; } } else { for (i = 0; i < nNfb; i++) { h_sbr_data->sbrNoiseFloorLevel[i + nNfb] += h_sbr_data->sbrNoiseFloorLevel[i]; } } } limitNoiseLevels(hHeaderData, h_sbr_data); /* Update prevNoiseLevel with the last noise envelope */ for (i = 0; i < nNfb; i++) h_prev_data->prevNoiseLevel[i] = h_sbr_data->sbrNoiseFloorLevel[i + nNfb * (nNoiseFloorEnvelopes - 1)]; /* Requantize the noise floor levels in COUPLING_OFF-mode */ if (!h_sbr_data->coupling) { int nf_e; for (i = 0; i < nNoiseFloorEnvelopes * nNfb; i++) { nf_e = 6 - (LONG)h_sbr_data->sbrNoiseFloorLevel[i] + 1 + NOISE_EXP_OFFSET; /* +1 to compensate for a mantissa of 0.5 instead of 1.0 */ h_sbr_data->sbrNoiseFloorLevel[i] = (FIXP_SGL)(((LONG)FL2FXCONST_SGL(0.5f)) + /* mantissa */ (nf_e & MASK_E)); /* exponent */ } } } fdk-aac-2.0.2/libSBRdec/src/env_dec.h000066400000000000000000000121621404225310700171350ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Envelope decoding */ #ifndef ENV_DEC_H #define ENV_DEC_H #include "sbrdecoder.h" #include "env_extr.h" void decodeSbrData(HANDLE_SBR_HEADER_DATA hHeaderData, HANDLE_SBR_FRAME_DATA h_data_left, HANDLE_SBR_PREV_FRAME_DATA h_prev_data_left, HANDLE_SBR_FRAME_DATA h_data_right, HANDLE_SBR_PREV_FRAME_DATA h_prev_data_right); #endif fdk-aac-2.0.2/libSBRdec/src/env_extr.cpp000066400000000000000000001613041404225310700177220ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Envelope extraction The functions provided by this module are mostly called by applySBR(). After it is determined that there is valid SBR data, sbrGetHeaderData() might be called if the current SBR data contains an \ref SBR_HEADER_ELEMENT as opposed to a \ref SBR_STANDARD_ELEMENT. This function may return various error codes as defined in #SBR_HEADER_STATUS . Most importantly it returns HEADER_RESET when decoder settings need to be recalculated according to the SBR specifications. In that case applySBR() will initiatite the required re-configuration. The header data is stored in a #SBR_HEADER_DATA structure. The actual SBR data for the current frame is decoded into SBR_FRAME_DATA stuctures by sbrGetChannelPairElement() [for stereo streams] and sbrGetSingleChannelElement() [for mono streams]. There is no fractional arithmetic involved. Once the information is extracted, the data needs to be further prepared before the actual decoding process. This is done in decodeSbrData(). \sa Description of buffer management in applySBR(). \ref documentationOverview

About the SBR data format:

Each frame includes SBR data (side chain information), and can be either the \ref SBR_HEADER_ELEMENT or the \ref SBR_STANDARD_ELEMENT. Parts of the data can be protected by a CRC checksum. \anchor SBR_HEADER_ELEMENT

The SBR_HEADER_ELEMENT

The SBR_HEADER_ELEMENT can be transmitted with every frame, however, it typically is send every second or so. It contains fundamental information such as SBR sampling frequency and frequency range as well as control signals that do not require frequent changes. It also includes the \ref SBR_STANDARD_ELEMENT. Depending on the changes between the information in a current SBR_HEADER_ELEMENT and the previous SBR_HEADER_ELEMENT, the SBR decoder might need to be reset and reconfigured (e.g. new tables need to be calculated). \anchor SBR_STANDARD_ELEMENT

The SBR_STANDARD_ELEMENT

This data can be subdivided into "side info" and "raw data", where side info is defined as signals needed to decode the raw data and some decoder tuning signals. Raw data is referred to as PCM and Huffman coded envelope and noise floor estimates. The side info also includes information about the time-frequency grid for the current frame. \sa \ref documentationOverview */ #include "env_extr.h" #include "sbr_ram.h" #include "sbr_rom.h" #include "huff_dec.h" #include "psbitdec.h" #define DRM_PARAMETRIC_STEREO 0 #define EXTENSION_ID_PS_CODING 2 static int extractPvcFrameInfo( HANDLE_FDK_BITSTREAM hBs, /*!< bitbuffer handle */ HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ HANDLE_SBR_FRAME_DATA h_frame_data, /*!< pointer to memory where the frame-info will be stored */ HANDLE_SBR_PREV_FRAME_DATA h_prev_frame_data, /*!< pointer to memory where the previous frame-info will be stored */ UCHAR pvc_mode_last, /**< PVC mode of last frame */ const UINT flags); static int extractFrameInfo(HANDLE_FDK_BITSTREAM hBs, HANDLE_SBR_HEADER_DATA hHeaderData, HANDLE_SBR_FRAME_DATA h_frame_data, const UINT nrOfChannels, const UINT flags); static int sbrGetPvcEnvelope(HANDLE_SBR_HEADER_DATA hHeaderData, HANDLE_SBR_FRAME_DATA h_frame_data, HANDLE_FDK_BITSTREAM hBs, const UINT flags, const UINT pvcMode); static int sbrGetEnvelope(HANDLE_SBR_HEADER_DATA hHeaderData, HANDLE_SBR_FRAME_DATA h_frame_data, HANDLE_FDK_BITSTREAM hBs, const UINT flags); static void sbrGetDirectionControlData(HANDLE_SBR_FRAME_DATA hFrameData, HANDLE_FDK_BITSTREAM hBs, const UINT flags, const int bs_pvc_mode); static void sbrGetNoiseFloorData(HANDLE_SBR_HEADER_DATA hHeaderData, HANDLE_SBR_FRAME_DATA h_frame_data, HANDLE_FDK_BITSTREAM hBs); static int checkFrameInfo(FRAME_INFO *pFrameInfo, int numberOfTimeSlots, int overlap, int timeStep); /* Mapping to std samplerate table according to 14496-3 (4.6.18.2.6) */ typedef struct SR_MAPPING { UINT fsRangeLo; /* If fsRangeLo(n+1)>fs>=fsRangeLo(n), it will be mapped to... */ UINT fsMapped; /* fsMapped. */ } SR_MAPPING; static const SR_MAPPING stdSampleRatesMapping[] = { {0, 8000}, {9391, 11025}, {11502, 12000}, {13856, 16000}, {18783, 22050}, {23004, 24000}, {27713, 32000}, {37566, 44100}, {46009, 48000}, {55426, 64000}, {75132, 88200}, {92017, 96000}}; static const SR_MAPPING stdSampleRatesMappingUsac[] = { {0, 16000}, {18783, 22050}, {23004, 24000}, {27713, 32000}, {35777, 40000}, {42000, 44100}, {46009, 48000}, {55426, 64000}, {75132, 88200}, {92017, 96000}}; UINT sbrdec_mapToStdSampleRate(UINT fs, UINT isUsac) /*!< Output sampling frequency */ { UINT fsMapped = fs, tableSize = 0; const SR_MAPPING *mappingTable; int i; if (!isUsac) { mappingTable = stdSampleRatesMapping; tableSize = sizeof(stdSampleRatesMapping) / sizeof(SR_MAPPING); } else { mappingTable = stdSampleRatesMappingUsac; tableSize = sizeof(stdSampleRatesMappingUsac) / sizeof(SR_MAPPING); } for (i = tableSize - 1; i >= 0; i--) { if (fs >= mappingTable[i].fsRangeLo) { fsMapped = mappingTable[i].fsMapped; break; } } return (fsMapped); } SBR_ERROR initHeaderData(HANDLE_SBR_HEADER_DATA hHeaderData, const int sampleRateIn, const int sampleRateOut, const INT downscaleFactor, const int samplesPerFrame, const UINT flags, const int setDefaultHdr) { HANDLE_FREQ_BAND_DATA hFreq = &hHeaderData->freqBandData; SBR_ERROR sbrError = SBRDEC_OK; int numAnalysisBands; int sampleRateProc; if (!(flags & (SBRDEC_SYNTAX_USAC | SBRDEC_SYNTAX_RSVD50))) { sampleRateProc = sbrdec_mapToStdSampleRate(sampleRateOut * downscaleFactor, 0); } else { sampleRateProc = sampleRateOut * downscaleFactor; } if (sampleRateIn == sampleRateOut) { hHeaderData->sbrProcSmplRate = sampleRateProc << 1; numAnalysisBands = 32; } else { hHeaderData->sbrProcSmplRate = sampleRateProc; if ((sampleRateOut >> 1) == sampleRateIn) { /* 1:2 */ numAnalysisBands = 32; } else if ((sampleRateOut >> 2) == sampleRateIn) { /* 1:4 */ numAnalysisBands = 16; } else if ((sampleRateOut * 3) >> 3 == (sampleRateIn * 8) >> 3) { /* 3:8, 3/4 core frame length */ numAnalysisBands = 24; } else { sbrError = SBRDEC_UNSUPPORTED_CONFIG; goto bail; } } numAnalysisBands /= downscaleFactor; if (setDefaultHdr) { /* Fill in default values first */ hHeaderData->syncState = SBR_NOT_INITIALIZED; hHeaderData->status = 0; hHeaderData->frameErrorFlag = 0; hHeaderData->bs_info.ampResolution = 1; hHeaderData->bs_info.xover_band = 0; hHeaderData->bs_info.sbr_preprocessing = 0; hHeaderData->bs_info.pvc_mode = 0; hHeaderData->bs_data.startFreq = 5; hHeaderData->bs_data.stopFreq = 0; hHeaderData->bs_data.freqScale = 0; /* previously 2; for ELD reduced delay bitstreams /samplerates initializing of the sbr decoder instance fails if freqScale is set to 2 because no master table can be generated; in ELD reduced delay bitstreams this value is always 0; gets overwritten when header is read */ hHeaderData->bs_data.alterScale = 1; hHeaderData->bs_data.noise_bands = 2; hHeaderData->bs_data.limiterBands = 2; hHeaderData->bs_data.limiterGains = 2; hHeaderData->bs_data.interpolFreq = 1; hHeaderData->bs_data.smoothingLength = 1; /* Patch some entries */ if (sampleRateOut * downscaleFactor >= 96000) { hHeaderData->bs_data.startFreq = 4; /* having read these frequency values from bit stream before. */ hHeaderData->bs_data.stopFreq = 3; } else if (sampleRateOut * downscaleFactor > 24000) { /* Trigger an error if SBR is going to be processed without */ hHeaderData->bs_data.startFreq = 7; /* having read these frequency values from bit stream before. */ hHeaderData->bs_data.stopFreq = 3; } } if ((sampleRateOut >> 2) == sampleRateIn) { hHeaderData->timeStep = 4; } else { hHeaderData->timeStep = (flags & SBRDEC_ELD_GRID) ? 1 : 2; } /* Setup pointers to frequency band tables */ hFreq->freqBandTable[0] = hFreq->freqBandTableLo; hFreq->freqBandTable[1] = hFreq->freqBandTableHi; /* One SBR timeslot corresponds to the amount of samples equal to the amount * of analysis bands, divided by the timestep. */ hHeaderData->numberTimeSlots = (samplesPerFrame / numAnalysisBands) >> (hHeaderData->timeStep - 1); if (hHeaderData->numberTimeSlots > (16)) { sbrError = SBRDEC_UNSUPPORTED_CONFIG; } hHeaderData->numberOfAnalysisBands = numAnalysisBands; if ((sampleRateOut >> 2) == sampleRateIn) { hHeaderData->numberTimeSlots <<= 1; } bail: return sbrError; } /*! \brief Initialize the SBR_PREV_FRAME_DATA struct */ void initSbrPrevFrameData( HANDLE_SBR_PREV_FRAME_DATA h_prev_data, /*!< handle to struct SBR_PREV_FRAME_DATA */ int timeSlots) /*!< Framelength in SBR-timeslots */ { int i; /* Set previous energy and noise levels to 0 for the case that decoding starts in the middle of a bitstream */ for (i = 0; i < MAX_FREQ_COEFFS; i++) h_prev_data->sfb_nrg_prev[i] = (FIXP_DBL)0; for (i = 0; i < MAX_NOISE_COEFFS; i++) h_prev_data->prevNoiseLevel[i] = (FIXP_DBL)0; for (i = 0; i < MAX_INVF_BANDS; i++) h_prev_data->sbr_invf_mode[i] = INVF_OFF; h_prev_data->stopPos = timeSlots; h_prev_data->coupling = COUPLING_OFF; h_prev_data->ampRes = 0; FDKmemclear(&h_prev_data->prevFrameInfo, sizeof(h_prev_data->prevFrameInfo)); } /*! \brief Read header data from bitstream \return error status - 0 if ok */ SBR_HEADER_STATUS sbrGetHeaderData(HANDLE_SBR_HEADER_DATA hHeaderData, HANDLE_FDK_BITSTREAM hBs, const UINT flags, const int fIsSbrData, const UCHAR configMode) { SBR_HEADER_DATA_BS *pBsData; SBR_HEADER_DATA_BS lastHeader; SBR_HEADER_DATA_BS_INFO lastInfo; int headerExtra1 = 0, headerExtra2 = 0; /* Read and discard new header in config change detection mode */ if (configMode & AC_CM_DET_CFG_CHANGE) { if (!(flags & (SBRDEC_SYNTAX_RSVD50 | SBRDEC_SYNTAX_USAC))) { /* ampResolution */ FDKreadBits(hBs, 1); } /* startFreq, stopFreq */ FDKpushFor(hBs, 8); if (!(flags & (SBRDEC_SYNTAX_RSVD50 | SBRDEC_SYNTAX_USAC))) { /* xover_band */ FDKreadBits(hBs, 3); /* reserved bits */ FDKreadBits(hBs, 2); } headerExtra1 = FDKreadBit(hBs); headerExtra2 = FDKreadBit(hBs); FDKpushFor(hBs, 5 * headerExtra1 + 6 * headerExtra2); return HEADER_OK; } /* Copy SBR bit stream header to temporary header */ lastHeader = hHeaderData->bs_data; lastInfo = hHeaderData->bs_info; /* Read new header from bitstream */ if ((flags & (SBRDEC_SYNTAX_RSVD50 | SBRDEC_SYNTAX_USAC)) && !fIsSbrData) { pBsData = &hHeaderData->bs_dflt; } else { pBsData = &hHeaderData->bs_data; } if (!(flags & (SBRDEC_SYNTAX_RSVD50 | SBRDEC_SYNTAX_USAC))) { hHeaderData->bs_info.ampResolution = FDKreadBits(hBs, 1); } pBsData->startFreq = FDKreadBits(hBs, 4); pBsData->stopFreq = FDKreadBits(hBs, 4); if (!(flags & (SBRDEC_SYNTAX_RSVD50 | SBRDEC_SYNTAX_USAC))) { hHeaderData->bs_info.xover_band = FDKreadBits(hBs, 3); FDKreadBits(hBs, 2); } headerExtra1 = FDKreadBits(hBs, 1); headerExtra2 = FDKreadBits(hBs, 1); /* Handle extra header information */ if (headerExtra1) { pBsData->freqScale = FDKreadBits(hBs, 2); pBsData->alterScale = FDKreadBits(hBs, 1); pBsData->noise_bands = FDKreadBits(hBs, 2); } else { pBsData->freqScale = 2; pBsData->alterScale = 1; pBsData->noise_bands = 2; } if (headerExtra2) { pBsData->limiterBands = FDKreadBits(hBs, 2); pBsData->limiterGains = FDKreadBits(hBs, 2); pBsData->interpolFreq = FDKreadBits(hBs, 1); pBsData->smoothingLength = FDKreadBits(hBs, 1); } else { pBsData->limiterBands = 2; pBsData->limiterGains = 2; pBsData->interpolFreq = 1; pBsData->smoothingLength = 1; } /* Look for new settings. IEC 14496-3, 4.6.18.3.1 */ if (hHeaderData->syncState < SBR_HEADER || lastHeader.startFreq != pBsData->startFreq || lastHeader.stopFreq != pBsData->stopFreq || lastHeader.freqScale != pBsData->freqScale || lastHeader.alterScale != pBsData->alterScale || lastHeader.noise_bands != pBsData->noise_bands || lastInfo.xover_band != hHeaderData->bs_info.xover_band) { return HEADER_RESET; /* New settings */ } return HEADER_OK; } /*! \brief Get missing harmonics parameters (only used for AAC+SBR) \return error status - 0 if ok */ int sbrGetSyntheticCodedData(HANDLE_SBR_HEADER_DATA hHeaderData, HANDLE_SBR_FRAME_DATA hFrameData, HANDLE_FDK_BITSTREAM hBs, const UINT flags) { int i, bitsRead = 0; int add_harmonic_flag = FDKreadBits(hBs, 1); bitsRead++; if (add_harmonic_flag) { int nSfb = hHeaderData->freqBandData.nSfb[1]; for (i = 0; i < ADD_HARMONICS_FLAGS_SIZE; i++) { /* read maximum 32 bits and align them to the MSB */ int readBits = fMin(32, nSfb); nSfb -= readBits; if (readBits > 0) { hFrameData->addHarmonics[i] = FDKreadBits(hBs, readBits) << (32 - readBits); } else { hFrameData->addHarmonics[i] = 0; } bitsRead += readBits; } /* bs_pvc_mode = 0 for Rsvd50 */ if (flags & SBRDEC_SYNTAX_USAC) { if (hHeaderData->bs_info.pvc_mode) { int bs_sinusoidal_position = 31; if (FDKreadBit(hBs) /* bs_sinusoidal_position_flag */) { bs_sinusoidal_position = FDKreadBits(hBs, 5); } hFrameData->sinusoidal_position = bs_sinusoidal_position; } } } else { for (i = 0; i < ADD_HARMONICS_FLAGS_SIZE; i++) hFrameData->addHarmonics[i] = 0; } return (bitsRead); } /*! \brief Reads extension data from the bitstream The bitstream format allows up to 4 kinds of extended data element. Extended data may contain several elements, each identified by a 2-bit-ID. So far, no extended data elements are defined hence the first 2 parameters are unused. The data should be skipped in order to update the number of read bits for the consistency check in applySBR(). */ static int extractExtendedData( HANDLE_SBR_HEADER_DATA hHeaderData, /*!< handle to SBR header */ HANDLE_FDK_BITSTREAM hBs /*!< Handle to the bit buffer */ , HANDLE_PS_DEC hParametricStereoDec /*!< Parametric Stereo Decoder */ ) { INT nBitsLeft; int extended_data; int i, frameOk = 1; extended_data = FDKreadBits(hBs, 1); if (extended_data) { int cnt; int bPsRead = 0; cnt = FDKreadBits(hBs, 4); if (cnt == (1 << 4) - 1) cnt += FDKreadBits(hBs, 8); nBitsLeft = 8 * cnt; /* sanity check for cnt */ if (nBitsLeft > (INT)FDKgetValidBits(hBs)) { /* limit nBitsLeft */ nBitsLeft = (INT)FDKgetValidBits(hBs); /* set frame error */ frameOk = 0; } while (nBitsLeft > 7) { int extension_id = FDKreadBits(hBs, 2); nBitsLeft -= 2; switch (extension_id) { case EXTENSION_ID_PS_CODING: /* Read PS data from bitstream */ if (hParametricStereoDec != NULL) { if (bPsRead && !hParametricStereoDec->bsData[hParametricStereoDec->bsReadSlot] .mpeg.bPsHeaderValid) { cnt = nBitsLeft >> 3; /* number of remaining bytes */ for (i = 0; i < cnt; i++) FDKreadBits(hBs, 8); nBitsLeft -= cnt * 8; } else { nBitsLeft -= (INT)ReadPsData(hParametricStereoDec, hBs, nBitsLeft); bPsRead = 1; } } /* parametric stereo detected, could set channelMode accordingly here */ /* */ /* "The usage of this parametric stereo extension to HE-AAC is */ /* signalled implicitly in the bitstream. Hence, if an sbr_extension() */ /* with bs_extension_id==EXTENSION_ID_PS is found in the SBR part of */ /* the bitstream, a decoder supporting the combination of SBR and PS */ /* shall operate the PS tool to generate a stereo output signal." */ /* source: ISO/IEC 14496-3:2001/FDAM 2:2004(E) */ break; default: cnt = nBitsLeft >> 3; /* number of remaining bytes */ for (i = 0; i < cnt; i++) FDKreadBits(hBs, 8); nBitsLeft -= cnt * 8; break; } } if (nBitsLeft < 0) { frameOk = 0; goto bail; } else { /* Read fill bits for byte alignment */ FDKreadBits(hBs, nBitsLeft); } } bail: return (frameOk); } /*! \brief Read bitstream elements of a SBR channel element \return SbrFrameOK */ int sbrGetChannelElement(HANDLE_SBR_HEADER_DATA hHeaderData, HANDLE_SBR_FRAME_DATA hFrameDataLeft, HANDLE_SBR_FRAME_DATA hFrameDataRight, HANDLE_SBR_PREV_FRAME_DATA hFrameDataLeftPrev, UCHAR pvc_mode_last, HANDLE_FDK_BITSTREAM hBs, HANDLE_PS_DEC hParametricStereoDec, const UINT flags, const int overlap) { int i, bs_coupling = COUPLING_OFF; const int nCh = (hFrameDataRight == NULL) ? 1 : 2; if (!(flags & (SBRDEC_SYNTAX_USAC | SBRDEC_SYNTAX_RSVD50))) { /* Reserved bits */ if (FDKreadBits(hBs, 1)) { /* bs_data_extra */ FDKreadBits(hBs, 4); if ((flags & SBRDEC_SYNTAX_SCAL) || (nCh == 2)) { FDKreadBits(hBs, 4); } } } if (nCh == 2) { /* Read coupling flag */ bs_coupling = FDKreadBits(hBs, 1); if (bs_coupling) { hFrameDataLeft->coupling = COUPLING_LEVEL; hFrameDataRight->coupling = COUPLING_BAL; } else { hFrameDataLeft->coupling = COUPLING_OFF; hFrameDataRight->coupling = COUPLING_OFF; } } else { if (flags & SBRDEC_SYNTAX_SCAL) { FDKreadBits(hBs, 1); /* bs_coupling */ } hFrameDataLeft->coupling = COUPLING_OFF; } if (flags & (SBRDEC_SYNTAX_USAC | SBRDEC_SYNTAX_RSVD50)) { if (flags & SBRDEC_USAC_HARMONICSBR) { hFrameDataLeft->sbrPatchingMode = FDKreadBit(hBs); if (hFrameDataLeft->sbrPatchingMode == 0) { hFrameDataLeft->sbrOversamplingFlag = FDKreadBit(hBs); if (FDKreadBit(hBs)) { /* sbrPitchInBinsFlag */ hFrameDataLeft->sbrPitchInBins = FDKreadBits(hBs, 7); } else { hFrameDataLeft->sbrPitchInBins = 0; } } else { hFrameDataLeft->sbrOversamplingFlag = 0; hFrameDataLeft->sbrPitchInBins = 0; } if (nCh == 2) { if (bs_coupling) { hFrameDataRight->sbrPatchingMode = hFrameDataLeft->sbrPatchingMode; hFrameDataRight->sbrOversamplingFlag = hFrameDataLeft->sbrOversamplingFlag; hFrameDataRight->sbrPitchInBins = hFrameDataLeft->sbrPitchInBins; } else { hFrameDataRight->sbrPatchingMode = FDKreadBit(hBs); if (hFrameDataRight->sbrPatchingMode == 0) { hFrameDataRight->sbrOversamplingFlag = FDKreadBit(hBs); if (FDKreadBit(hBs)) { /* sbrPitchInBinsFlag */ hFrameDataRight->sbrPitchInBins = FDKreadBits(hBs, 7); } else { hFrameDataRight->sbrPitchInBins = 0; } } else { hFrameDataRight->sbrOversamplingFlag = 0; hFrameDataRight->sbrPitchInBins = 0; } } } } else { if (nCh == 2) { hFrameDataRight->sbrPatchingMode = 1; hFrameDataRight->sbrOversamplingFlag = 0; hFrameDataRight->sbrPitchInBins = 0; } hFrameDataLeft->sbrPatchingMode = 1; hFrameDataLeft->sbrOversamplingFlag = 0; hFrameDataLeft->sbrPitchInBins = 0; } } else { if (nCh == 2) { hFrameDataRight->sbrPatchingMode = 1; hFrameDataRight->sbrOversamplingFlag = 0; hFrameDataRight->sbrPitchInBins = 0; } hFrameDataLeft->sbrPatchingMode = 1; hFrameDataLeft->sbrOversamplingFlag = 0; hFrameDataLeft->sbrPitchInBins = 0; } /* sbr_grid(): Grid control */ if (hHeaderData->bs_info.pvc_mode) { FDK_ASSERT(nCh == 1); /* PVC not possible for CPE */ if (!extractPvcFrameInfo(hBs, hHeaderData, hFrameDataLeft, hFrameDataLeftPrev, pvc_mode_last, flags)) return 0; if (!checkFrameInfo(&hFrameDataLeft->frameInfo, hHeaderData->numberTimeSlots, overlap, hHeaderData->timeStep)) return 0; } else { if (!extractFrameInfo(hBs, hHeaderData, hFrameDataLeft, 1, flags)) return 0; if (!checkFrameInfo(&hFrameDataLeft->frameInfo, hHeaderData->numberTimeSlots, overlap, hHeaderData->timeStep)) return 0; } if (nCh == 2) { if (hFrameDataLeft->coupling) { FDKmemcpy(&hFrameDataRight->frameInfo, &hFrameDataLeft->frameInfo, sizeof(FRAME_INFO)); hFrameDataRight->ampResolutionCurrentFrame = hFrameDataLeft->ampResolutionCurrentFrame; } else { if (!extractFrameInfo(hBs, hHeaderData, hFrameDataRight, 2, flags)) return 0; if (!checkFrameInfo(&hFrameDataRight->frameInfo, hHeaderData->numberTimeSlots, overlap, hHeaderData->timeStep)) return 0; } } /* sbr_dtdf(): Fetch domain vectors (time or frequency direction for delta-coding) */ sbrGetDirectionControlData(hFrameDataLeft, hBs, flags, hHeaderData->bs_info.pvc_mode); if (nCh == 2) { sbrGetDirectionControlData(hFrameDataRight, hBs, flags, 0); } /* sbr_invf() */ for (i = 0; i < hHeaderData->freqBandData.nInvfBands; i++) { hFrameDataLeft->sbr_invf_mode[i] = (INVF_MODE)FDKreadBits(hBs, 2); } if (nCh == 2) { if (hFrameDataLeft->coupling) { for (i = 0; i < hHeaderData->freqBandData.nInvfBands; i++) { hFrameDataRight->sbr_invf_mode[i] = hFrameDataLeft->sbr_invf_mode[i]; } } else { for (i = 0; i < hHeaderData->freqBandData.nInvfBands; i++) { hFrameDataRight->sbr_invf_mode[i] = (INVF_MODE)FDKreadBits(hBs, 2); } } } if (nCh == 1) { if (hHeaderData->bs_info.pvc_mode) { if (!sbrGetPvcEnvelope(hHeaderData, hFrameDataLeft, hBs, flags, hHeaderData->bs_info.pvc_mode)) return 0; } else if (!sbrGetEnvelope(hHeaderData, hFrameDataLeft, hBs, flags)) return 0; sbrGetNoiseFloorData(hHeaderData, hFrameDataLeft, hBs); } else if (hFrameDataLeft->coupling) { if (!sbrGetEnvelope(hHeaderData, hFrameDataLeft, hBs, flags)) { return 0; } sbrGetNoiseFloorData(hHeaderData, hFrameDataLeft, hBs); if (!sbrGetEnvelope(hHeaderData, hFrameDataRight, hBs, flags)) { return 0; } sbrGetNoiseFloorData(hHeaderData, hFrameDataRight, hBs); } else { /* nCh == 2 && no coupling */ if (!sbrGetEnvelope(hHeaderData, hFrameDataLeft, hBs, flags)) return 0; if (!sbrGetEnvelope(hHeaderData, hFrameDataRight, hBs, flags)) return 0; sbrGetNoiseFloorData(hHeaderData, hFrameDataLeft, hBs); sbrGetNoiseFloorData(hHeaderData, hFrameDataRight, hBs); } sbrGetSyntheticCodedData(hHeaderData, hFrameDataLeft, hBs, flags); if (nCh == 2) { sbrGetSyntheticCodedData(hHeaderData, hFrameDataRight, hBs, flags); } if (!(flags & (SBRDEC_SYNTAX_USAC | SBRDEC_SYNTAX_RSVD50))) { if (!extractExtendedData(hHeaderData, hBs, hParametricStereoDec)) { return 0; } } return 1; } /*! \brief Read direction control data from bitstream */ void sbrGetDirectionControlData( HANDLE_SBR_FRAME_DATA h_frame_data, /*!< handle to struct SBR_FRAME_DATA */ HANDLE_FDK_BITSTREAM hBs, /*!< handle to struct BIT_BUF */ const UINT flags, const int bs_pvc_mode) { int i; int indepFlag = 0; if (flags & (SBRDEC_SYNTAX_USAC | SBRDEC_SYNTAX_RSVD50)) { indepFlag = flags & SBRDEC_USAC_INDEP; } if (bs_pvc_mode == 0) { i = 0; if (indepFlag) { h_frame_data->domain_vec[i++] = 0; } for (; i < h_frame_data->frameInfo.nEnvelopes; i++) { h_frame_data->domain_vec[i] = FDKreadBits(hBs, 1); } } i = 0; if (indepFlag) { h_frame_data->domain_vec_noise[i++] = 0; } for (; i < h_frame_data->frameInfo.nNoiseEnvelopes; i++) { h_frame_data->domain_vec_noise[i] = FDKreadBits(hBs, 1); } } /*! \brief Read noise-floor-level data from bitstream */ void sbrGetNoiseFloorData( HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ HANDLE_SBR_FRAME_DATA h_frame_data, /*!< handle to struct SBR_FRAME_DATA */ HANDLE_FDK_BITSTREAM hBs) /*!< handle to struct BIT_BUF */ { int i, j; int delta; COUPLING_MODE coupling; int noNoiseBands = hHeaderData->freqBandData.nNfb; Huffman hcb_noiseF; Huffman hcb_noise; int envDataTableCompFactor; coupling = h_frame_data->coupling; /* Select huffman codebook depending on coupling mode */ if (coupling == COUPLING_BAL) { hcb_noise = (Huffman)&FDK_sbrDecoder_sbr_huffBook_NoiseBalance11T; hcb_noiseF = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvBalance11F; /* "sbr_huffBook_NoiseBalance11F" */ envDataTableCompFactor = 1; } else { hcb_noise = (Huffman)&FDK_sbrDecoder_sbr_huffBook_NoiseLevel11T; hcb_noiseF = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvLevel11F; /* "sbr_huffBook_NoiseLevel11F" */ envDataTableCompFactor = 0; } /* Read raw noise-envelope data */ for (i = 0; i < h_frame_data->frameInfo.nNoiseEnvelopes; i++) { if (h_frame_data->domain_vec_noise[i] == 0) { if (coupling == COUPLING_BAL) { h_frame_data->sbrNoiseFloorLevel[i * noNoiseBands] = (FIXP_SGL)(((int)FDKreadBits(hBs, 5)) << envDataTableCompFactor); } else { h_frame_data->sbrNoiseFloorLevel[i * noNoiseBands] = (FIXP_SGL)(int)FDKreadBits(hBs, 5); } for (j = 1; j < noNoiseBands; j++) { delta = DecodeHuffmanCW(hcb_noiseF, hBs); h_frame_data->sbrNoiseFloorLevel[i * noNoiseBands + j] = (FIXP_SGL)(delta << envDataTableCompFactor); } } else { for (j = 0; j < noNoiseBands; j++) { delta = DecodeHuffmanCW(hcb_noise, hBs); h_frame_data->sbrNoiseFloorLevel[i * noNoiseBands + j] = (FIXP_SGL)(delta << envDataTableCompFactor); } } } } /* ns = mapNsMode2ns[pvcMode-1][nsMode] */ static const UCHAR mapNsMode2ns[2][2] = { {16, 4}, /* pvcMode = 1 */ {12, 3} /* pvcMode = 2 */ }; static int sbrGetPvcEnvelope( HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ HANDLE_SBR_FRAME_DATA h_frame_data, /*!< handle to struct SBR_FRAME_DATA */ HANDLE_FDK_BITSTREAM hBs, /*!< handle to struct BIT_BUF */ const UINT flags, const UINT pvcMode) { int divMode, nsMode; int indepFlag = flags & SBRDEC_USAC_INDEP; UCHAR *pvcID = h_frame_data->pvcID; divMode = FDKreadBits(hBs, PVC_DIVMODE_BITS); nsMode = FDKreadBit(hBs); FDK_ASSERT((pvcMode == 1) || (pvcMode == 2)); h_frame_data->ns = mapNsMode2ns[pvcMode - 1][nsMode]; if (divMode <= 3) { int i, k = 1, sum_length = 0, reuse_pcvID; /* special treatment for first time slot k=0 */ indepFlag ? (reuse_pcvID = 0) : (reuse_pcvID = FDKreadBit(hBs)); if (reuse_pcvID) { pvcID[0] = hHeaderData->pvcIDprev; } else { pvcID[0] = FDKreadBits(hBs, PVC_PVCID_BITS); } /* other time slots k>0 */ for (i = 0; i < divMode; i++) { int length, numBits = 4; if (sum_length >= 13) { numBits = 1; } else if (sum_length >= 11) { numBits = 2; } else if (sum_length >= 7) { numBits = 3; } length = FDKreadBits(hBs, numBits); sum_length += length + 1; if (sum_length >= PVC_NTIMESLOT) { return 0; /* parse error */ } for (; length--; k++) { pvcID[k] = pvcID[k - 1]; } pvcID[k++] = FDKreadBits(hBs, PVC_PVCID_BITS); } for (; k < 16; k++) { pvcID[k] = pvcID[k - 1]; } } else { /* divMode >= 4 */ int num_grid_info, fixed_length, grid_info, j, k = 0; divMode -= 4; num_grid_info = 2 << divMode; fixed_length = 8 >> divMode; FDK_ASSERT(num_grid_info * fixed_length == PVC_NTIMESLOT); /* special treatment for first time slot k=0 */ indepFlag ? (grid_info = 1) : (grid_info = FDKreadBit(hBs)); if (grid_info) { pvcID[k++] = FDKreadBits(hBs, PVC_PVCID_BITS); } else { pvcID[k++] = hHeaderData->pvcIDprev; } j = fixed_length - 1; for (; j--; k++) { pvcID[k] = pvcID[k - 1]; } num_grid_info--; /* other time slots k>0 */ for (; num_grid_info--;) { j = fixed_length; grid_info = FDKreadBit(hBs); if (grid_info) { pvcID[k++] = FDKreadBits(hBs, PVC_PVCID_BITS); j--; } for (; j--; k++) { pvcID[k] = pvcID[k - 1]; } } } hHeaderData->pvcIDprev = pvcID[PVC_NTIMESLOT - 1]; /* usage of PVC excludes inter-TES tool */ h_frame_data->iTESactive = (UCHAR)0; return 1; } /*! \brief Read envelope data from bitstream */ static int sbrGetEnvelope( HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ HANDLE_SBR_FRAME_DATA h_frame_data, /*!< handle to struct SBR_FRAME_DATA */ HANDLE_FDK_BITSTREAM hBs, /*!< handle to struct BIT_BUF */ const UINT flags) { int i, j; UCHAR no_band[MAX_ENVELOPES]; int delta = 0; int offset = 0; COUPLING_MODE coupling = h_frame_data->coupling; int ampRes = hHeaderData->bs_info.ampResolution; int nEnvelopes = h_frame_data->frameInfo.nEnvelopes; int envDataTableCompFactor; int start_bits, start_bits_balance; Huffman hcb_t, hcb_f; h_frame_data->nScaleFactors = 0; if ((h_frame_data->frameInfo.frameClass == 0) && (nEnvelopes == 1)) { if (flags & SBRDEC_ELD_GRID) ampRes = h_frame_data->ampResolutionCurrentFrame; else ampRes = 0; } h_frame_data->ampResolutionCurrentFrame = ampRes; /* Set number of bits for first value depending on amplitude resolution */ if (ampRes == 1) { start_bits = 6; start_bits_balance = 5; } else { start_bits = 7; start_bits_balance = 6; } /* Calculate number of values for each envelope and alltogether */ for (i = 0; i < nEnvelopes; i++) { no_band[i] = hHeaderData->freqBandData.nSfb[h_frame_data->frameInfo.freqRes[i]]; h_frame_data->nScaleFactors += no_band[i]; } if (h_frame_data->nScaleFactors > MAX_NUM_ENVELOPE_VALUES) return 0; /* Select Huffman codebook depending on coupling mode and amplitude resolution */ if (coupling == COUPLING_BAL) { envDataTableCompFactor = 1; if (ampRes == 0) { hcb_t = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvBalance10T; hcb_f = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvBalance10F; } else { hcb_t = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvBalance11T; hcb_f = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvBalance11F; } } else { envDataTableCompFactor = 0; if (ampRes == 0) { hcb_t = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvLevel10T; hcb_f = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvLevel10F; } else { hcb_t = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvLevel11T; hcb_f = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvLevel11F; } } h_frame_data->iTESactive = (UCHAR)0; /* disable inter-TES by default */ /* Now read raw envelope data */ for (j = 0, offset = 0; j < nEnvelopes; j++) { if (h_frame_data->domain_vec[j] == 0) { if (coupling == COUPLING_BAL) { h_frame_data->iEnvelope[offset] = (FIXP_SGL)(((int)FDKreadBits(hBs, start_bits_balance)) << envDataTableCompFactor); } else { h_frame_data->iEnvelope[offset] = (FIXP_SGL)(int)FDKreadBits(hBs, start_bits); } } for (i = (1 - h_frame_data->domain_vec[j]); i < no_band[j]; i++) { if (h_frame_data->domain_vec[j] == 0) { delta = DecodeHuffmanCW(hcb_f, hBs); } else { delta = DecodeHuffmanCW(hcb_t, hBs); } h_frame_data->iEnvelope[offset + i] = (FIXP_SGL)(delta << envDataTableCompFactor); } if ((flags & SBRDEC_SYNTAX_USAC) && (flags & SBRDEC_USAC_ITES)) { int bs_temp_shape = FDKreadBit(hBs); FDK_ASSERT(j < 8); h_frame_data->iTESactive |= (UCHAR)(bs_temp_shape << j); if (bs_temp_shape) { h_frame_data->interTempShapeMode[j] = FDKread2Bits(hBs); /* bs_inter_temp_shape_mode */ } else { h_frame_data->interTempShapeMode[j] = 0; } } offset += no_band[j]; } #if ENV_EXP_FRACT /* Convert from int to scaled fract (ENV_EXP_FRACT bits for the fractional * part) */ for (i = 0; i < h_frame_data->nScaleFactors; i++) { h_frame_data->iEnvelope[i] <<= ENV_EXP_FRACT; } #endif return 1; } /***************************************************************************/ /*! \brief Generates frame info for FIXFIXonly frame class used for low delay version \return zero for error, one for correct. ****************************************************************************/ static int generateFixFixOnly(FRAME_INFO *hSbrFrameInfo, int tranPosInternal, int numberTimeSlots, const UINT flags) { int nEnv, i, tranIdx; const int *pTable; if (tranPosInternal >= numberTimeSlots) { return 0; } switch (numberTimeSlots) { case 8: pTable = FDK_sbrDecoder_envelopeTable_8[tranPosInternal]; break; case 15: pTable = FDK_sbrDecoder_envelopeTable_15[tranPosInternal]; break; case 16: pTable = FDK_sbrDecoder_envelopeTable_16[tranPosInternal]; break; default: return 0; } /* look number of envelopes in table */ nEnv = pTable[0]; /* look up envelope distribution in table */ for (i = 1; i < nEnv; i++) hSbrFrameInfo->borders[i] = pTable[i + 2]; /* open and close frame border */ hSbrFrameInfo->borders[0] = 0; hSbrFrameInfo->borders[nEnv] = numberTimeSlots; hSbrFrameInfo->nEnvelopes = nEnv; /* transient idx */ tranIdx = hSbrFrameInfo->tranEnv = pTable[1]; /* add noise floors */ hSbrFrameInfo->bordersNoise[0] = 0; hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[tranIdx ? tranIdx : 1]; hSbrFrameInfo->bordersNoise[2] = numberTimeSlots; /* nEnv is always > 1, so nNoiseEnvelopes is always 2 (IEC 14496-3 4.6.19.3.2) */ hSbrFrameInfo->nNoiseEnvelopes = 2; return 1; } /*! \brief Extracts LowDelaySBR control data from the bitstream. \return zero for bitstream error, one for correct. */ static int extractLowDelayGrid( HANDLE_FDK_BITSTREAM hBitBuf, /*!< bitbuffer handle */ HANDLE_SBR_HEADER_DATA hHeaderData, HANDLE_SBR_FRAME_DATA h_frame_data, /*!< contains the FRAME_INFO struct to be filled */ int timeSlots, const UINT flags) { FRAME_INFO *pFrameInfo = &h_frame_data->frameInfo; INT numberTimeSlots = hHeaderData->numberTimeSlots; INT temp = 0, k; /* FIXFIXonly framing case */ h_frame_data->frameInfo.frameClass = 0; /* get the transient position from the bitstream */ switch (timeSlots) { case 8: /* 3bit transient position (temp={0;..;7}) */ temp = FDKreadBits(hBitBuf, 3); break; case 16: case 15: /* 4bit transient position (temp={0;..;15}) */ temp = FDKreadBits(hBitBuf, 4); break; default: return 0; } /* For "case 15" only*/ if (temp >= timeSlots) { return 0; } /* calculate borders according to the transient position */ if (!generateFixFixOnly(pFrameInfo, temp, numberTimeSlots, flags)) { return 0; } /* decode freq res: */ for (k = 0; k < pFrameInfo->nEnvelopes; k++) { pFrameInfo->freqRes[k] = (UCHAR)FDKreadBits(hBitBuf, 1); /* f = F [1 bits] */ } return 1; } /*! \brief Extract the PVC frame information (structure FRAME_INFO) from the bitstream \return Zero for bitstream error, one for correct. */ int extractPvcFrameInfo( HANDLE_FDK_BITSTREAM hBs, /*!< bitbuffer handle */ HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ HANDLE_SBR_FRAME_DATA h_frame_data, /*!< pointer to memory where the frame-info will be stored */ HANDLE_SBR_PREV_FRAME_DATA h_prev_frame_data, /*!< pointer to memory where the previous frame-info will be stored */ UCHAR pvc_mode_last, /**< PVC mode of last frame */ const UINT flags) { FRAME_INFO *pFrameInfo = &h_frame_data->frameInfo; FRAME_INFO *pPrevFrameInfo = &h_prev_frame_data->prevFrameInfo; int bs_var_len_hf, bs_noise_position; bs_noise_position = FDKreadBits(hBs, 4); /* SBR_PVC_NOISEPOSITION_BITS 4 */ bs_var_len_hf = FDKreadBit(hBs); pFrameInfo->noisePosition = bs_noise_position; pFrameInfo->tranEnv = -1; /* Init for bs_noise_position == 0 in case a parse error is found below. */ pFrameInfo->nEnvelopes = 1; pFrameInfo->nNoiseEnvelopes = 1; pFrameInfo->freqRes[0] = 0; if (bs_var_len_hf) { /* 1 or 3 Bits */ pFrameInfo->varLength = FDKreadBits(hBs, 2) + 1; if (pFrameInfo->varLength > 3) { pFrameInfo->varLength = 0; /* assume bs_var_len_hf == 0 in case of error */ return 0; /* reserved value -> parse error */ } } else { pFrameInfo->varLength = 0; } if (bs_noise_position) { pFrameInfo->nEnvelopes = 2; pFrameInfo->nNoiseEnvelopes = 2; FDKmemclear(pFrameInfo->freqRes, sizeof(pFrameInfo->freqRes)); } /* frame border calculation */ if (hHeaderData->bs_info.pvc_mode > 0) { /* See "7.5.1.4 HF adjustment of SBR envelope scalefactors" for reference. */ FDK_ASSERT((pFrameInfo->nEnvelopes == 1) || (pFrameInfo->nEnvelopes == 2)); /* left timeborder-offset: use the timeborder of prev SBR frame */ if (pPrevFrameInfo->nEnvelopes > 0) { pFrameInfo->borders[0] = pPrevFrameInfo->borders[pPrevFrameInfo->nEnvelopes] - PVC_NTIMESLOT; FDK_ASSERT(pFrameInfo->borders[0] <= 3); } else { pFrameInfo->borders[0] = 0; } /* right timeborder-offset: */ pFrameInfo->borders[pFrameInfo->nEnvelopes] = 16 + pFrameInfo->varLength; if (pFrameInfo->nEnvelopes == 2) { pFrameInfo->borders[1] = pFrameInfo->noisePosition; } /* Calculation of PVC time borders t_EPVC */ if (pvc_mode_last == 0) { /* there was a legacy SBR frame before this frame => use bs_var_len' for * first PVC timeslot */ pFrameInfo->pvcBorders[0] = pFrameInfo->borders[0]; } else { pFrameInfo->pvcBorders[0] = 0; } if (pFrameInfo->nEnvelopes == 2) { pFrameInfo->pvcBorders[1] = pFrameInfo->borders[1]; } pFrameInfo->pvcBorders[pFrameInfo->nEnvelopes] = 16; /* calculation of SBR noise-floor time-border vector: */ for (INT i = 0; i <= pFrameInfo->nNoiseEnvelopes; i++) { pFrameInfo->bordersNoise[i] = pFrameInfo->borders[i]; } pFrameInfo->tranEnv = -1; /* tranEnv not used */ } return 1; } /*! \brief Extract the frame information (structure FRAME_INFO) from the bitstream \return Zero for bitstream error, one for correct. */ int extractFrameInfo( HANDLE_FDK_BITSTREAM hBs, /*!< bitbuffer handle */ HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ HANDLE_SBR_FRAME_DATA h_frame_data, /*!< pointer to memory where the frame-info will be stored */ const UINT nrOfChannels, const UINT flags) { FRAME_INFO *pFrameInfo = &h_frame_data->frameInfo; int numberTimeSlots = hHeaderData->numberTimeSlots; int pointer_bits = 0, nEnv = 0, b = 0, border, i, n = 0, k, p, aL, aR, nL, nR, temp = 0, staticFreqRes; UCHAR frameClass; if (flags & SBRDEC_ELD_GRID) { /* CODEC_AACLD (LD+SBR) only uses the normal 0 Grid for non-transient Frames * and the LowDelayGrid for transient Frames */ frameClass = FDKreadBits(hBs, 1); /* frameClass = [1 bit] */ if (frameClass == 1) { /* if frameClass == 1, extract LowDelaySbrGrid, otherwise extract normal * SBR-Grid for FIXIFX */ /* extract the AACLD-Sbr-Grid */ pFrameInfo->frameClass = frameClass; int err = 1; err = extractLowDelayGrid(hBs, hHeaderData, h_frame_data, numberTimeSlots, flags); return err; } } else { frameClass = FDKreadBits(hBs, 2); /* frameClass = C [2 bits] */ } switch (frameClass) { case 0: temp = FDKreadBits(hBs, 2); /* E [2 bits ] */ nEnv = (int)(1 << temp); /* E -> e */ if ((flags & SBRDEC_ELD_GRID) && (nEnv == 1)) h_frame_data->ampResolutionCurrentFrame = FDKreadBits(hBs, 1); /* new ELD Syntax 07-11-09 */ staticFreqRes = FDKreadBits(hBs, 1); if (flags & (SBRDEC_SYNTAX_USAC | SBRDEC_SYNTAX_RSVD50)) { if (nEnv > MAX_ENVELOPES_USAC) return 0; } else b = nEnv + 1; switch (nEnv) { case 1: switch (numberTimeSlots) { case 15: FDKmemcpy(pFrameInfo, &FDK_sbrDecoder_sbr_frame_info1_15, sizeof(FRAME_INFO)); break; case 16: FDKmemcpy(pFrameInfo, &FDK_sbrDecoder_sbr_frame_info1_16, sizeof(FRAME_INFO)); break; default: FDK_ASSERT(0); } break; case 2: switch (numberTimeSlots) { case 15: FDKmemcpy(pFrameInfo, &FDK_sbrDecoder_sbr_frame_info2_15, sizeof(FRAME_INFO)); break; case 16: FDKmemcpy(pFrameInfo, &FDK_sbrDecoder_sbr_frame_info2_16, sizeof(FRAME_INFO)); break; default: FDK_ASSERT(0); } break; case 4: switch (numberTimeSlots) { case 15: FDKmemcpy(pFrameInfo, &FDK_sbrDecoder_sbr_frame_info4_15, sizeof(FRAME_INFO)); break; case 16: FDKmemcpy(pFrameInfo, &FDK_sbrDecoder_sbr_frame_info4_16, sizeof(FRAME_INFO)); break; default: FDK_ASSERT(0); } break; case 8: #if (MAX_ENVELOPES >= 8) switch (numberTimeSlots) { case 15: FDKmemcpy(pFrameInfo, &FDK_sbrDecoder_sbr_frame_info8_15, sizeof(FRAME_INFO)); break; case 16: FDKmemcpy(pFrameInfo, &FDK_sbrDecoder_sbr_frame_info8_16, sizeof(FRAME_INFO)); break; default: FDK_ASSERT(0); } break; #else return 0; #endif } /* Apply correct freqRes (High is default) */ if (!staticFreqRes) { for (i = 0; i < nEnv; i++) pFrameInfo->freqRes[i] = 0; } break; case 1: case 2: temp = FDKreadBits(hBs, 2); /* A [2 bits] */ n = FDKreadBits(hBs, 2); /* n = N [2 bits] */ nEnv = n + 1; /* # envelopes */ b = nEnv + 1; /* # borders */ break; } switch (frameClass) { case 1: /* Decode borders: */ pFrameInfo->borders[0] = 0; /* first border */ border = temp + numberTimeSlots; /* A -> aR */ i = b - 1; /* frame info index for last border */ pFrameInfo->borders[i] = border; /* last border */ for (k = 0; k < n; k++) { temp = FDKreadBits(hBs, 2); /* R [2 bits] */ border -= (2 * temp + 2); /* R -> r */ pFrameInfo->borders[--i] = border; } /* Decode pointer: */ pointer_bits = DFRACT_BITS - 1 - CountLeadingBits((FIXP_DBL)(n + 1)); p = FDKreadBits(hBs, pointer_bits); /* p = P [pointer_bits bits] */ if (p > n + 1) return 0; pFrameInfo->tranEnv = p ? n + 2 - p : -1; /* Decode freq res: */ for (k = n; k >= 0; k--) { pFrameInfo->freqRes[k] = FDKreadBits(hBs, 1); /* f = F [1 bits] */ } /* Calculate noise floor middle border: */ if (p == 0 || p == 1) pFrameInfo->bordersNoise[1] = pFrameInfo->borders[n]; else pFrameInfo->bordersNoise[1] = pFrameInfo->borders[pFrameInfo->tranEnv]; break; case 2: /* Decode borders: */ border = temp; /* A -> aL */ pFrameInfo->borders[0] = border; /* first border */ for (k = 1; k <= n; k++) { temp = FDKreadBits(hBs, 2); /* R [2 bits] */ border += (2 * temp + 2); /* R -> r */ pFrameInfo->borders[k] = border; } pFrameInfo->borders[k] = numberTimeSlots; /* last border */ /* Decode pointer: */ pointer_bits = DFRACT_BITS - 1 - CountLeadingBits((FIXP_DBL)(n + 1)); p = FDKreadBits(hBs, pointer_bits); /* p = P [pointer_bits bits] */ if (p > n + 1) return 0; if (p == 0 || p == 1) pFrameInfo->tranEnv = -1; else pFrameInfo->tranEnv = p - 1; /* Decode freq res: */ for (k = 0; k <= n; k++) { pFrameInfo->freqRes[k] = FDKreadBits(hBs, 1); /* f = F [1 bits] */ } /* Calculate noise floor middle border: */ switch (p) { case 0: pFrameInfo->bordersNoise[1] = pFrameInfo->borders[1]; break; case 1: pFrameInfo->bordersNoise[1] = pFrameInfo->borders[n]; break; default: pFrameInfo->bordersNoise[1] = pFrameInfo->borders[pFrameInfo->tranEnv]; break; } break; case 3: /* v_ctrlSignal = [frameClass,aL,aR,nL,nR,v_rL,v_rR,p,v_fLR]; */ aL = FDKreadBits(hBs, 2); /* AL [2 bits], AL -> aL */ aR = FDKreadBits(hBs, 2) + numberTimeSlots; /* AR [2 bits], AR -> aR */ nL = FDKreadBits(hBs, 2); /* nL = NL [2 bits] */ nR = FDKreadBits(hBs, 2); /* nR = NR [2 bits] */ /*------------------------------------------------------------------------- Calculate help variables --------------------------------------------------------------------------*/ /* general: */ nEnv = nL + nR + 1; /* # envelopes */ if (nEnv > MAX_ENVELOPES) return 0; b = nEnv + 1; /* # borders */ /*------------------------------------------------------------------------- Decode envelopes --------------------------------------------------------------------------*/ /* L-borders: */ border = aL; /* first border */ pFrameInfo->borders[0] = border; for (k = 1; k <= nL; k++) { temp = FDKreadBits(hBs, 2); /* R [2 bits] */ border += (2 * temp + 2); /* R -> r */ pFrameInfo->borders[k] = border; } /* R-borders: */ border = aR; /* last border */ i = nEnv; pFrameInfo->borders[i] = border; for (k = 0; k < nR; k++) { temp = FDKreadBits(hBs, 2); /* R [2 bits] */ border -= (2 * temp + 2); /* R -> r */ pFrameInfo->borders[--i] = border; } /* decode pointer: */ pointer_bits = DFRACT_BITS - 1 - CountLeadingBits((FIXP_DBL)(nL + nR + 1)); p = FDKreadBits(hBs, pointer_bits); /* p = P [pointer_bits bits] */ if (p > nL + nR + 1) return 0; pFrameInfo->tranEnv = p ? b - p : -1; /* decode freq res: */ for (k = 0; k < nEnv; k++) { pFrameInfo->freqRes[k] = FDKreadBits(hBs, 1); /* f = F [1 bits] */ } /*------------------------------------------------------------------------- Decode noise floors --------------------------------------------------------------------------*/ pFrameInfo->bordersNoise[0] = aL; if (nEnv == 1) { /* 1 noise floor envelope: */ pFrameInfo->bordersNoise[1] = aR; } else { /* 2 noise floor envelopes */ if (p == 0 || p == 1) pFrameInfo->bordersNoise[1] = pFrameInfo->borders[nEnv - 1]; else pFrameInfo->bordersNoise[1] = pFrameInfo->borders[pFrameInfo->tranEnv]; pFrameInfo->bordersNoise[2] = aR; } break; } /* Store number of envelopes, noise floor envelopes and frame class */ pFrameInfo->nEnvelopes = nEnv; if (nEnv == 1) pFrameInfo->nNoiseEnvelopes = 1; else pFrameInfo->nNoiseEnvelopes = 2; pFrameInfo->frameClass = frameClass; if (pFrameInfo->frameClass == 2 || pFrameInfo->frameClass == 1) { /* calculate noise floor first and last borders: */ pFrameInfo->bordersNoise[0] = pFrameInfo->borders[0]; pFrameInfo->bordersNoise[pFrameInfo->nNoiseEnvelopes] = pFrameInfo->borders[nEnv]; } return 1; } /*! \brief Check if the frameInfo vector has reasonable values. \return Zero for error, one for correct */ static int checkFrameInfo( FRAME_INFO *pFrameInfo, /*!< pointer to frameInfo */ int numberOfTimeSlots, /*!< QMF time slots per frame */ int overlap, /*!< Amount of overlap QMF time slots */ int timeStep) /*!< QMF slots to SBR slots step factor */ { int maxPos, i, j; int startPos; int stopPos; int tranEnv; int startPosNoise; int stopPosNoise; int nEnvelopes = pFrameInfo->nEnvelopes; int nNoiseEnvelopes = pFrameInfo->nNoiseEnvelopes; if (nEnvelopes < 1 || nEnvelopes > MAX_ENVELOPES) return 0; if (nNoiseEnvelopes > MAX_NOISE_ENVELOPES) return 0; startPos = pFrameInfo->borders[0]; stopPos = pFrameInfo->borders[nEnvelopes]; tranEnv = pFrameInfo->tranEnv; startPosNoise = pFrameInfo->bordersNoise[0]; stopPosNoise = pFrameInfo->bordersNoise[nNoiseEnvelopes]; if (overlap < 0 || overlap > (3 * (4))) { return 0; } if (timeStep < 1 || timeStep > (4)) { return 0; } maxPos = numberOfTimeSlots + (overlap / timeStep); /* Check that the start and stop positions of the frame are reasonable values. */ if ((startPos < 0) || (startPos >= stopPos)) return 0; if (startPos > maxPos - numberOfTimeSlots) /* First env. must start in or directly after the overlap buffer */ return 0; if (stopPos < numberOfTimeSlots) /* One complete frame must be ready for output after processing */ return 0; if (stopPos > maxPos) return 0; /* Check that the start border for every envelope is strictly later in time */ for (i = 0; i < nEnvelopes; i++) { if (pFrameInfo->borders[i] >= pFrameInfo->borders[i + 1]) return 0; } /* Check that the envelope to be shortened is actually among the envelopes */ if (tranEnv > nEnvelopes) return 0; /* Check the noise borders */ if (nEnvelopes == 1 && nNoiseEnvelopes > 1) return 0; if (startPos != startPosNoise || stopPos != stopPosNoise) return 0; /* Check that the start border for every noise-envelope is strictly later in * time*/ for (i = 0; i < nNoiseEnvelopes; i++) { if (pFrameInfo->bordersNoise[i] >= pFrameInfo->bordersNoise[i + 1]) return 0; } /* Check that every noise border is the same as an envelope border*/ for (i = 0; i < nNoiseEnvelopes; i++) { startPosNoise = pFrameInfo->bordersNoise[i]; for (j = 0; j < nEnvelopes; j++) { if (pFrameInfo->borders[j] == startPosNoise) break; } if (j == nEnvelopes) return 0; } return 1; } fdk-aac-2.0.2/libSBRdec/src/env_extr.h000066400000000000000000000434461404225310700173750ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Envelope extraction prototypes */ #ifndef ENV_EXTR_H #define ENV_EXTR_H #include "sbrdecoder.h" #include "FDK_bitstream.h" #include "lpp_tran.h" #include "psdec.h" #include "pvc_dec.h" #define ENV_EXP_FRACT 0 /*!< Shift raw envelope data to support fractional numbers. Can be set to 8 instead of 0 to enhance accuracy during concealment. This is not required for conformance and #requantizeEnvelopeData() will become more expensive. */ #define EXP_BITS 6 /*!< Size of exponent-part of a pseudo float envelope value (should be at least 6). The remaining bits in each word are used for the mantissa (should be at least 10). This format is used in the arrays iEnvelope[] and sbrNoiseFloorLevel[] in the FRAME_DATA struct which must fit in a certain part of the output buffer (See buffer management in sbr_dec.cpp). Exponents and mantissas could also be stored in separate arrays. Accessing the exponent or the mantissa would be simplified and the masks #MASK_E resp. #MASK_M would no longer be required. */ #define MASK_M \ (((1 << (FRACT_BITS - EXP_BITS)) - 1) \ << EXP_BITS) /*!< Mask for extracting the mantissa of a pseudo float \ envelope value */ #define MASK_E \ ((1 << EXP_BITS) - 1) /*!< Mask for extracting the exponent of a pseudo \ float envelope value */ #define SIGN_EXT \ (((SCHAR)-1) ^ \ MASK_E) /*!< a CHAR-constant with all bits above our sign-bit set */ #define ROUNDING \ ((FIXP_SGL)( \ 1 << (EXP_BITS - 1))) /*!< 0.5-offset for rounding the mantissa of a \ pseudo-float envelope value */ #define NRG_EXP_OFFSET \ 16 /*!< Will be added to the reference energy's exponent to prevent negative \ numbers */ #define NOISE_EXP_OFFSET \ 38 /*!< Will be added to the noise level exponent to prevent negative \ numbers */ #define ADD_HARMONICS_FLAGS_SIZE 2 /* ceil(MAX_FREQ_COEFFS/32) */ typedef enum { HEADER_NOT_PRESENT, HEADER_ERROR, HEADER_OK, HEADER_RESET } SBR_HEADER_STATUS; typedef enum { SBR_NOT_INITIALIZED = 0, UPSAMPLING = 1, SBR_HEADER = 2, SBR_ACTIVE = 3 } SBR_SYNC_STATE; typedef enum { COUPLING_OFF = 0, COUPLING_LEVEL, COUPLING_BAL } COUPLING_MODE; typedef struct { UCHAR nSfb[2]; /*!< Number of SBR-bands for low and high freq-resolution */ UCHAR nNfb; /*!< Actual number of noise bands to read from the bitstream*/ UCHAR numMaster; /*!< Number of SBR-bands in v_k_master */ UCHAR lowSubband; /*!< QMF-band where SBR frequency range starts */ UCHAR highSubband; /*!< QMF-band where SBR frequency range ends */ UCHAR ov_highSubband; /*!< if headerchange applies this value holds the old highband value -> highband value of overlap area; required for overlap in usac when headerchange occurs between XVAR and VARX frame */ UCHAR limiterBandTable[MAX_NUM_LIMITERS + 1]; /*!< Limiter band table. */ UCHAR noLimiterBands; /*!< Number of limiter bands. */ UCHAR nInvfBands; /*!< Number of bands for inverse filtering */ UCHAR *freqBandTable[2]; /*!< Pointers to freqBandTableLo and freqBandTableHi */ UCHAR freqBandTableLo[MAX_FREQ_COEFFS / 2 + 1]; /*!< Mapping of SBR bands to QMF bands for low frequency resolution */ UCHAR freqBandTableHi[MAX_FREQ_COEFFS + 1]; /*!< Mapping of SBR bands to QMF bands for high frequency resolution */ UCHAR freqBandTableNoise[MAX_NOISE_COEFFS + 1]; /*!< Mapping of SBR noise bands to QMF bands */ UCHAR v_k_master[MAX_FREQ_COEFFS + 1]; /*!< Master BandTable which freqBandTable is derived from */ } FREQ_BAND_DATA; typedef FREQ_BAND_DATA *HANDLE_FREQ_BAND_DATA; #define SBRDEC_ELD_GRID 1 #define SBRDEC_SYNTAX_SCAL 2 #define SBRDEC_SYNTAX_USAC 4 #define SBRDEC_SYNTAX_RSVD50 8 #define SBRDEC_USAC_INDEP \ 16 /* Flag indicating that USAC global independency flag is active. */ #define SBRDEC_LOW_POWER \ 32 /* Flag indicating that Low Power QMF mode shall be used. */ #define SBRDEC_PS_DECODED \ 64 /* Flag indicating that PS was decoded and rendered. */ #define SBRDEC_QUAD_RATE \ 128 /* Flag indicating that USAC SBR 4:1 is active. \ */ #define SBRDEC_USAC_HARMONICSBR \ 256 /* Flag indicating that USAC HBE tool is active. */ #define SBRDEC_LD_MPS_QMF \ 512 /* Flag indicating that the LD-MPS QMF shall be used. */ #define SBRDEC_USAC_ITES \ 1024 /* Flag indicating that USAC inter TES tool is active. */ #define SBRDEC_SYNTAX_DRM \ 2048 /* Flag indicating that DRM30/DRM+ reverse syntax is being used. */ #define SBRDEC_ELD_DOWNSCALE \ 4096 /* Flag indicating that ELD downscaled mode decoding is used */ #define SBRDEC_DOWNSAMPLE \ 8192 /* Flag indicating that the downsampling mode is used. */ #define SBRDEC_FLUSH 16384 /* Flag is used to flush all elements in use. */ #define SBRDEC_FORCE_RESET \ 32768 /* Flag is used to force a reset of all elements in use. */ #define SBRDEC_SKIP_QMF_ANA \ (1 << 21) /* Flag indicating that the input data is provided in the QMF \ domain. */ #define SBRDEC_SKIP_QMF_SYN \ (1 << 22) /* Flag indicating that the output data is exported in the QMF \ domain. */ #define SBRDEC_HDR_STAT_RESET 1 #define SBRDEC_HDR_STAT_UPDATE 2 typedef struct { UCHAR ampResolution; /*!< Amplitude resolution of envelope values (0: 1.5dB, 1: 3dB) */ UCHAR xover_band; /*!< Start index in #v_k_master[] used for dynamic crossover frequency */ UCHAR sbr_preprocessing; /*!< SBR prewhitening flag. */ UCHAR pvc_mode; /*!< Predictive vector coding mode */ } SBR_HEADER_DATA_BS_INFO; typedef struct { /* Changes in these variables causes a reset of the decoder */ UCHAR startFreq; /*!< Index for SBR start frequency */ UCHAR stopFreq; /*!< Index for SBR highest frequency */ UCHAR freqScale; /*!< 0: linear scale, 1-3 logarithmic scales */ UCHAR alterScale; /*!< Flag for coarser frequency resolution */ UCHAR noise_bands; /*!< Noise bands per octave, read from bitstream*/ /* don't require reset */ UCHAR limiterBands; /*!< Index for number of limiter bands per octave */ UCHAR limiterGains; /*!< Index to select gain limit */ UCHAR interpolFreq; /*!< Select gain calculation method (1: per QMF channel, 0: per SBR band) */ UCHAR smoothingLength; /*!< Smoothing of gains over time (0: on 1: off) */ } SBR_HEADER_DATA_BS; typedef struct { SBR_SYNC_STATE syncState; /*!< The current initialization status of the header */ UCHAR status; /*!< Flags field used for signaling a reset right before the processing starts and an update from config (e.g. ASC). */ UCHAR frameErrorFlag; /*!< Frame data valid flag. CAUTION: This variable will be overwritten by the flag stored in the element structure. This is necessary because of the frame delay. There it might happen that different slots use the same header. */ UCHAR numberTimeSlots; /*!< AAC: 16,15 */ UCHAR numberOfAnalysisBands; /*!< Number of QMF analysis bands */ UCHAR timeStep; /*!< Time resolution of SBR in QMF-slots */ UINT sbrProcSmplRate; /*!< SBR processing sampling frequency (!= OutputSamplingRate) (always: CoreSamplingRate * UpSamplingFactor; even in single rate mode) */ SBR_HEADER_DATA_BS bs_data; /*!< current SBR header. */ SBR_HEADER_DATA_BS bs_dflt; /*!< Default sbr header. */ SBR_HEADER_DATA_BS_INFO bs_info; /*!< SBR info. */ FREQ_BAND_DATA freqBandData; /*!< Pointer to struct #FREQ_BAND_DATA */ UCHAR pvcIDprev; } SBR_HEADER_DATA; typedef SBR_HEADER_DATA *HANDLE_SBR_HEADER_DATA; typedef struct { UCHAR frameClass; /*!< Select grid type */ UCHAR nEnvelopes; /*!< Number of envelopes */ UCHAR borders[MAX_ENVELOPES + 1]; /*!< Envelope borders (in SBR-timeslots, e.g. mp3PRO: 0..11) */ UCHAR freqRes[MAX_ENVELOPES]; /*!< Frequency resolution for each envelope (0=low, 1=high) */ SCHAR tranEnv; /*!< Transient envelope, -1 if none */ UCHAR nNoiseEnvelopes; /*!< Number of noise envelopes */ UCHAR bordersNoise[MAX_NOISE_ENVELOPES + 1]; /*!< borders of noise envelopes */ UCHAR pvcBorders[MAX_PVC_ENVELOPES + 1]; UCHAR noisePosition; UCHAR varLength; } FRAME_INFO; typedef struct { FIXP_SGL sfb_nrg_prev[MAX_FREQ_COEFFS]; /*!< Previous envelope (required for differential-coded values) */ FIXP_SGL prevNoiseLevel[MAX_NOISE_COEFFS]; /*!< Previous noise envelope (required for differential-coded values) */ COUPLING_MODE coupling; /*!< Stereo-mode of previous frame */ INVF_MODE sbr_invf_mode[MAX_INVF_BANDS]; /*!< Previous strength of filtering in transposer */ UCHAR ampRes; /*!< Previous amplitude resolution (0: 1.5dB, 1: 3dB) */ UCHAR stopPos; /*!< Position in time where last envelope ended */ UCHAR frameErrorFlag; /*!< Previous frame status */ UCHAR prevSbrPitchInBins; /*!< Previous frame pitchInBins */ FRAME_INFO prevFrameInfo; } SBR_PREV_FRAME_DATA; typedef SBR_PREV_FRAME_DATA *HANDLE_SBR_PREV_FRAME_DATA; typedef struct { int nScaleFactors; /*!< total number of scalefactors in frame */ FRAME_INFO frameInfo; /*!< time grid for current frame */ UCHAR domain_vec[MAX_ENVELOPES]; /*!< Bitfield containing direction of delta-coding for each envelope (0:frequency, 1:time) */ UCHAR domain_vec_noise [MAX_NOISE_ENVELOPES]; /*!< Same as above, but for noise envelopes */ INVF_MODE sbr_invf_mode[MAX_INVF_BANDS]; /*!< Strength of filtering in transposer */ COUPLING_MODE coupling; /*!< Stereo-mode */ int ampResolutionCurrentFrame; /*!< Amplitude resolution of envelope values (0: 1.5dB, 1: 3dB) */ ULONG addHarmonics[ADD_HARMONICS_FLAGS_SIZE]; /*!< Flags for synthetic sine addition (aligned to MSB) */ FIXP_SGL iEnvelope[MAX_NUM_ENVELOPE_VALUES]; /*!< Envelope data */ FIXP_SGL sbrNoiseFloorLevel[MAX_NUM_NOISE_VALUES]; /*!< Noise envelope data */ UCHAR iTESactive; /*!< One flag for each envelope to enable USAC inter-TES */ UCHAR interTempShapeMode[MAX_ENVELOPES]; /*!< USAC inter-TES: bs_inter_temp_shape_mode[ch][env] value */ UCHAR pvcID[PVC_NTIMESLOT]; /*!< One PVC ID value for each time slot */ UCHAR ns; UCHAR sinusoidal_position; UCHAR sbrPatchingMode; UCHAR sbrOversamplingFlag; UCHAR sbrPitchInBins; } SBR_FRAME_DATA; typedef SBR_FRAME_DATA *HANDLE_SBR_FRAME_DATA; /*! \brief Maps sampling frequencies to frequencies for which setup tables are available Maps arbitary sampling frequency to nearest neighbors for which setup tables are available (e.g. 25600 -> 24000). Used for startFreq calculation. The mapping is defined in 14496-3 (4.6.18.2.6), fs(SBR), and table 4.82 \return mapped sampling frequency */ UINT sbrdec_mapToStdSampleRate(UINT fs, UINT isUsac); /*!< Output sampling frequency */ void initSbrPrevFrameData(HANDLE_SBR_PREV_FRAME_DATA h_prev_data, int timeSlots); int sbrGetChannelElement(HANDLE_SBR_HEADER_DATA hHeaderData, HANDLE_SBR_FRAME_DATA hFrameDataLeft, HANDLE_SBR_FRAME_DATA hFrameDataRight, HANDLE_SBR_PREV_FRAME_DATA hFrameDataLeftPrev, UCHAR pvc_mode_last, HANDLE_FDK_BITSTREAM hBitBuf, HANDLE_PS_DEC hParametricStereoDec, const UINT flags, const int overlap); SBR_HEADER_STATUS sbrGetHeaderData(HANDLE_SBR_HEADER_DATA headerData, HANDLE_FDK_BITSTREAM hBitBuf, const UINT flags, const int fIsSbrData, const UCHAR configMode); /*! \brief Initialize SBR header data Copy default values to the header data struct and patch some entries depending on the core codec. */ SBR_ERROR initHeaderData(HANDLE_SBR_HEADER_DATA hHeaderData, const int sampleRateIn, const int sampleRateOut, const INT downscaleFactor, const int samplesPerFrame, const UINT flags, const int setDefaultHdr); #endif /* Convert headroom bits to exponent */ #define SCALE2EXP(s) (15 - (s)) #define EXP2SCALE(e) (15 - (e)) fdk-aac-2.0.2/libSBRdec/src/hbe.cpp000066400000000000000000003010541404225310700166240ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Fast FFT routines prototypes \author Fabian Haussel */ #include "hbe.h" #include "qmf.h" #include "env_extr.h" #define HBE_MAX_QMF_BANDS (40) #define HBE_MAX_OUT_SLOTS (11) #define QMF_WIN_LEN \ (12 + 6 - 4 - 1) /* 6 subband slots extra delay to align with HQ - 4 slots \ to compensate for critical sampling delay - 1 slot to \ align critical sampling exactly (w additional time \ domain delay)*/ #ifndef PI #define PI 3.14159265358979323846 #endif static const int xProducts[MAX_STRETCH_HBE - 1] = { 1, 1, 1}; /* Cross products on(1)/off(0) for T=2,3,4. */ static const int startSubband2kL[33] = { 0, 0, 0, 0, 0, 0, 0, 2, 2, 2, 4, 4, 4, 4, 4, 6, 6, 6, 8, 8, 8, 8, 8, 10, 10, 10, 12, 12, 12, 12, 12, 12, 12}; static const int pmin = 12; static const FIXP_DBL hintReal_F[4][3] = { {FL2FXCONST_DBL(0.39840335f), FL2FXCONST_DBL(0.39840335f), FL2FXCONST_DBL(-0.39840335f)}, {FL2FXCONST_DBL(0.39840335f), FL2FXCONST_DBL(-0.39840335f), FL2FXCONST_DBL(-0.39840335f)}, {FL2FXCONST_DBL(-0.39840335f), FL2FXCONST_DBL(-0.39840335f), FL2FXCONST_DBL(0.39840335f)}, {FL2FXCONST_DBL(-0.39840335f), FL2FXCONST_DBL(0.39840335f), FL2FXCONST_DBL(0.39840335f)}}; static const FIXP_DBL factors[4] = { FL2FXCONST_DBL(0.39840335f), FL2FXCONST_DBL(-0.39840335f), FL2FXCONST_DBL(-0.39840335f), FL2FXCONST_DBL(0.39840335f)}; #define PSCALE 32 static const FIXP_DBL p_F[128] = {FL2FXCONST_DBL(0.f / (PSCALE * 12.f)), FL2FXCONST_DBL(1.f / (PSCALE * 12.f)), FL2FXCONST_DBL(2.f / (PSCALE * 12.f)), FL2FXCONST_DBL(3.f / (PSCALE * 12.f)), FL2FXCONST_DBL(4.f / (PSCALE * 12.f)), FL2FXCONST_DBL(5.f / (PSCALE * 12.f)), FL2FXCONST_DBL(6.f / (PSCALE * 12.f)), FL2FXCONST_DBL(7.f / (PSCALE * 12.f)), FL2FXCONST_DBL(8.f / (PSCALE * 12.f)), FL2FXCONST_DBL(9.f / (PSCALE * 12.f)), FL2FXCONST_DBL(10.f / (PSCALE * 12.f)), FL2FXCONST_DBL(11.f / (PSCALE * 12.f)), FL2FXCONST_DBL(12.f / (PSCALE * 12.f)), FL2FXCONST_DBL(13.f / (PSCALE * 12.f)), FL2FXCONST_DBL(14.f / (PSCALE * 12.f)), FL2FXCONST_DBL(15.f / (PSCALE * 12.f)), FL2FXCONST_DBL(16.f / (PSCALE * 12.f)), FL2FXCONST_DBL(17.f / (PSCALE * 12.f)), FL2FXCONST_DBL(18.f / (PSCALE * 12.f)), FL2FXCONST_DBL(19.f / (PSCALE * 12.f)), FL2FXCONST_DBL(20.f / (PSCALE * 12.f)), FL2FXCONST_DBL(21.f / (PSCALE * 12.f)), FL2FXCONST_DBL(22.f / (PSCALE * 12.f)), FL2FXCONST_DBL(23.f / (PSCALE * 12.f)), FL2FXCONST_DBL(24.f / (PSCALE * 12.f)), FL2FXCONST_DBL(25.f / (PSCALE * 12.f)), FL2FXCONST_DBL(26.f / (PSCALE * 12.f)), FL2FXCONST_DBL(27.f / (PSCALE * 12.f)), FL2FXCONST_DBL(28.f / (PSCALE * 12.f)), FL2FXCONST_DBL(29.f / (PSCALE * 12.f)), FL2FXCONST_DBL(30.f / (PSCALE * 12.f)), FL2FXCONST_DBL(31.f / (PSCALE * 12.f)), FL2FXCONST_DBL(32.f / (PSCALE * 12.f)), FL2FXCONST_DBL(33.f / (PSCALE * 12.f)), FL2FXCONST_DBL(34.f / (PSCALE * 12.f)), FL2FXCONST_DBL(35.f / (PSCALE * 12.f)), FL2FXCONST_DBL(36.f / (PSCALE * 12.f)), FL2FXCONST_DBL(37.f / (PSCALE * 12.f)), FL2FXCONST_DBL(38.f / (PSCALE * 12.f)), FL2FXCONST_DBL(39.f / (PSCALE * 12.f)), FL2FXCONST_DBL(40.f / (PSCALE * 12.f)), FL2FXCONST_DBL(41.f / (PSCALE * 12.f)), FL2FXCONST_DBL(42.f / (PSCALE * 12.f)), FL2FXCONST_DBL(43.f / (PSCALE * 12.f)), FL2FXCONST_DBL(44.f / (PSCALE * 12.f)), FL2FXCONST_DBL(45.f / (PSCALE * 12.f)), FL2FXCONST_DBL(46.f / (PSCALE * 12.f)), FL2FXCONST_DBL(47.f / (PSCALE * 12.f)), FL2FXCONST_DBL(48.f / (PSCALE * 12.f)), FL2FXCONST_DBL(49.f / (PSCALE * 12.f)), FL2FXCONST_DBL(50.f / (PSCALE * 12.f)), FL2FXCONST_DBL(51.f / (PSCALE * 12.f)), FL2FXCONST_DBL(52.f / (PSCALE * 12.f)), FL2FXCONST_DBL(53.f / (PSCALE * 12.f)), FL2FXCONST_DBL(54.f / (PSCALE * 12.f)), FL2FXCONST_DBL(55.f / (PSCALE * 12.f)), FL2FXCONST_DBL(56.f / (PSCALE * 12.f)), FL2FXCONST_DBL(57.f / (PSCALE * 12.f)), FL2FXCONST_DBL(58.f / (PSCALE * 12.f)), FL2FXCONST_DBL(59.f / (PSCALE * 12.f)), FL2FXCONST_DBL(60.f / (PSCALE * 12.f)), FL2FXCONST_DBL(61.f / (PSCALE * 12.f)), FL2FXCONST_DBL(62.f / (PSCALE * 12.f)), FL2FXCONST_DBL(63.f / (PSCALE * 12.f)), FL2FXCONST_DBL(64.f / (PSCALE * 12.f)), FL2FXCONST_DBL(65.f / (PSCALE * 12.f)), FL2FXCONST_DBL(66.f / (PSCALE * 12.f)), FL2FXCONST_DBL(67.f / (PSCALE * 12.f)), FL2FXCONST_DBL(68.f / (PSCALE * 12.f)), FL2FXCONST_DBL(69.f / (PSCALE * 12.f)), FL2FXCONST_DBL(70.f / (PSCALE * 12.f)), FL2FXCONST_DBL(71.f / (PSCALE * 12.f)), FL2FXCONST_DBL(72.f / (PSCALE * 12.f)), FL2FXCONST_DBL(73.f / (PSCALE * 12.f)), FL2FXCONST_DBL(74.f / (PSCALE * 12.f)), FL2FXCONST_DBL(75.f / (PSCALE * 12.f)), FL2FXCONST_DBL(76.f / (PSCALE * 12.f)), FL2FXCONST_DBL(77.f / (PSCALE * 12.f)), FL2FXCONST_DBL(78.f / (PSCALE * 12.f)), FL2FXCONST_DBL(79.f / (PSCALE * 12.f)), FL2FXCONST_DBL(80.f / (PSCALE * 12.f)), FL2FXCONST_DBL(81.f / (PSCALE * 12.f)), FL2FXCONST_DBL(82.f / (PSCALE * 12.f)), FL2FXCONST_DBL(83.f / (PSCALE * 12.f)), FL2FXCONST_DBL(84.f / (PSCALE * 12.f)), FL2FXCONST_DBL(85.f / (PSCALE * 12.f)), FL2FXCONST_DBL(86.f / (PSCALE * 12.f)), FL2FXCONST_DBL(87.f / (PSCALE * 12.f)), FL2FXCONST_DBL(88.f / (PSCALE * 12.f)), FL2FXCONST_DBL(89.f / (PSCALE * 12.f)), FL2FXCONST_DBL(90.f / (PSCALE * 12.f)), FL2FXCONST_DBL(91.f / (PSCALE * 12.f)), FL2FXCONST_DBL(92.f / (PSCALE * 12.f)), FL2FXCONST_DBL(93.f / (PSCALE * 12.f)), FL2FXCONST_DBL(94.f / (PSCALE * 12.f)), FL2FXCONST_DBL(95.f / (PSCALE * 12.f)), FL2FXCONST_DBL(96.f / (PSCALE * 12.f)), FL2FXCONST_DBL(97.f / (PSCALE * 12.f)), FL2FXCONST_DBL(98.f / (PSCALE * 12.f)), FL2FXCONST_DBL(99.f / (PSCALE * 12.f)), FL2FXCONST_DBL(100.f / (PSCALE * 12.f)), FL2FXCONST_DBL(101.f / (PSCALE * 12.f)), FL2FXCONST_DBL(102.f / (PSCALE * 12.f)), FL2FXCONST_DBL(103.f / (PSCALE * 12.f)), FL2FXCONST_DBL(104.f / (PSCALE * 12.f)), FL2FXCONST_DBL(105.f / (PSCALE * 12.f)), FL2FXCONST_DBL(106.f / (PSCALE * 12.f)), FL2FXCONST_DBL(107.f / (PSCALE * 12.f)), FL2FXCONST_DBL(108.f / (PSCALE * 12.f)), FL2FXCONST_DBL(109.f / (PSCALE * 12.f)), FL2FXCONST_DBL(110.f / (PSCALE * 12.f)), FL2FXCONST_DBL(111.f / (PSCALE * 12.f)), FL2FXCONST_DBL(112.f / (PSCALE * 12.f)), FL2FXCONST_DBL(113.f / (PSCALE * 12.f)), FL2FXCONST_DBL(114.f / (PSCALE * 12.f)), FL2FXCONST_DBL(115.f / (PSCALE * 12.f)), FL2FXCONST_DBL(116.f / (PSCALE * 12.f)), FL2FXCONST_DBL(117.f / (PSCALE * 12.f)), FL2FXCONST_DBL(118.f / (PSCALE * 12.f)), FL2FXCONST_DBL(119.f / (PSCALE * 12.f)), FL2FXCONST_DBL(120.f / (PSCALE * 12.f)), FL2FXCONST_DBL(121.f / (PSCALE * 12.f)), FL2FXCONST_DBL(122.f / (PSCALE * 12.f)), FL2FXCONST_DBL(123.f / (PSCALE * 12.f)), FL2FXCONST_DBL(124.f / (PSCALE * 12.f)), FL2FXCONST_DBL(125.f / (PSCALE * 12.f)), FL2FXCONST_DBL(126.f / (PSCALE * 12.f)), FL2FXCONST_DBL(127.f / (PSCALE * 12.f))}; static const FIXP_DBL band_F[64] = { FL2FXCONST_DBL((0.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((1.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((2.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((3.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((4.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((5.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((6.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((7.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((8.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((9.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((10.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((11.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((12.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((13.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((14.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((15.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((16.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((17.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((18.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((19.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((20.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((21.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((22.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((23.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((24.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((25.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((26.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((27.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((28.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((29.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((30.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((31.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((32.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((33.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((34.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((35.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((36.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((37.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((38.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((39.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((40.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((41.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((42.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((43.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((44.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((45.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((46.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((47.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((48.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((49.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((50.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((51.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((52.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((53.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((54.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((55.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((56.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((57.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((58.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((59.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((60.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((61.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((62.f * 2.f + 1) / (PSCALE << 2)), FL2FXCONST_DBL((63.f * 2.f + 1) / (PSCALE << 2))}; static const FIXP_DBL tr_str[3] = {FL2FXCONST_DBL(1.f / 4.f), FL2FXCONST_DBL(2.f / 4.f), FL2FXCONST_DBL(3.f / 4.f)}; static const FIXP_DBL stretchfac[3] = {FL2FXCONST_DBL(1.f / 2.f), FL2FXCONST_DBL(1.f / 3.f), FL2FXCONST_DBL(1.f / 4.f)}; static const FIXP_DBL cos_F[64] = { 26353028, -79043208, 131685776, -184244944, 236697216, -289006912, 341142496, -393072608, 444773984, -496191392, 547325824, -598114752, 648559104, -698597248, 748230016, -797411904, 846083200, -894275136, 941928192, -989013760, 1035474624, -1081340672, 1126555136, -1171063296, 1214893696, -1257992192, 1300332544, -1341889408, 1382612736, -1422503808, 1461586944, -1499741440, 1537039104, -1573364864, 1608743808, -1643196672, 1676617344, -1709028992, 1740450560, -1770784896, 1800089472, -1828273536, 1855357440, -1881356288, 1906190080, -1929876608, 1952428928, -1973777664, 1993962880, -2012922240, 2030670208, -2047216000, 2062508288, -2076559488, 2089376128, -2100932224, 2111196800, -2120214784, 2127953792, -2134394368, 2139565056, -2143444864, 2146026624, -2147321856}; static const FIXP_DBL twiddle[121] = {1073741824, 1071442860, 1064555814, 1053110176, 1037154959, 1016758484, 992008094, 963009773, 929887697, 892783698, 851856663, 807281846, 759250125, 707967178, 653652607, 596538995, 536870912, 474903865, 410903207, 345142998, 277904834, 209476638, 140151432, 70226075, 0, -70226075, -140151432, -209476638, -277904834, -345142998, -410903207, -474903865, -536870912, -596538995, -653652607, -707967178, -759250125, -807281846, -851856663, -892783698, -929887697, -963009773, -992008094, -1016758484, -1037154959, -1053110176, -1064555814, -1071442860, -1073741824, -1071442860, -1064555814, -1053110176, -1037154959, -1016758484, -992008094, -963009773, -929887697, -892783698, -851856663, -807281846, -759250125, -707967178, -653652607, -596538995, -536870912, -474903865, -410903207, -345142998, -277904834, -209476638, -140151432, -70226075, 0, 70226075, 140151432, 209476638, 277904834, 345142998, 410903207, 474903865, 536870912, 596538995, 653652607, 707967178, 759250125, 807281846, 851856663, 892783698, 929887697, 963009773, 992008094, 1016758484, 1037154959, 1053110176, 1064555814, 1071442860, 1073741824, 1071442860, 1064555814, 1053110176, 1037154959, 1016758484, 992008094, 963009773, 929887697, 892783698, 851856663, 807281846, 759250125, 707967178, 653652607, 596538995, 536870912, 474903865, 410903207, 345142998, 277904834, 209476638, 140151432, 70226075, 0}; #if FIXP_QTW == FIXP_SGL #define HTW(x) (x) #else #define HTW(x) FX_DBL2FX_QTW(FX_SGL2FX_DBL((const FIXP_SGL)x)) #endif static const FIXP_QTW post_twiddle_cos_8[8] = { HTW(-1606), HTW(4756), HTW(-7723), HTW(10394), HTW(-12665), HTW(14449), HTW(-15679), HTW(16305)}; static const FIXP_QTW post_twiddle_cos_16[16] = { HTW(-804), HTW(2404), HTW(-3981), HTW(5520), HTW(-7005), HTW(8423), HTW(-9760), HTW(11003), HTW(-12140), HTW(13160), HTW(-14053), HTW(14811), HTW(-15426), HTW(15893), HTW(-16207), HTW(16364)}; static const FIXP_QTW post_twiddle_cos_24[24] = { HTW(-536), HTW(1606), HTW(-2669), HTW(3720), HTW(-4756), HTW(5771), HTW(-6762), HTW(7723), HTW(-8652), HTW(9543), HTW(-10394), HTW(11200), HTW(-11958), HTW(12665), HTW(-13318), HTW(13913), HTW(-14449), HTW(14924), HTW(-15334), HTW(15679), HTW(-15956), HTW(16165), HTW(-16305), HTW(16375)}; static const FIXP_QTW post_twiddle_cos_32[32] = { HTW(-402), HTW(1205), HTW(-2006), HTW(2801), HTW(-3590), HTW(4370), HTW(-5139), HTW(5897), HTW(-6639), HTW(7366), HTW(-8076), HTW(8765), HTW(-9434), HTW(10080), HTW(-10702), HTW(11297), HTW(-11866), HTW(12406), HTW(-12916), HTW(13395), HTW(-13842), HTW(14256), HTW(-14635), HTW(14978), HTW(-15286), HTW(15557), HTW(-15791), HTW(15986), HTW(-16143), HTW(16261), HTW(-16340), HTW(16379)}; static const FIXP_QTW post_twiddle_cos_40[40] = { HTW(-322), HTW(965), HTW(-1606), HTW(2245), HTW(-2880), HTW(3511), HTW(-4137), HTW(4756), HTW(-5368), HTW(5971), HTW(-6566), HTW(7150), HTW(-7723), HTW(8285), HTW(-8833), HTW(9368), HTW(-9889), HTW(10394), HTW(-10883), HTW(11356), HTW(-11810), HTW(12247), HTW(-12665), HTW(13063), HTW(-13441), HTW(13799), HTW(-14135), HTW(14449), HTW(-14741), HTW(15011), HTW(-15257), HTW(15480), HTW(-15679), HTW(15853), HTW(-16003), HTW(16129), HTW(-16229), HTW(16305), HTW(-16356), HTW(16381)}; static const FIXP_QTW post_twiddle_sin_8[8] = { HTW(16305), HTW(-15679), HTW(14449), HTW(-12665), HTW(10394), HTW(-7723), HTW(4756), HTW(-1606)}; static const FIXP_QTW post_twiddle_sin_16[16] = { HTW(16364), HTW(-16207), HTW(15893), HTW(-15426), HTW(14811), HTW(-14053), HTW(13160), HTW(-12140), HTW(11003), HTW(-9760), HTW(8423), HTW(-7005), HTW(5520), HTW(-3981), HTW(2404), HTW(-804)}; static const FIXP_QTW post_twiddle_sin_24[24] = { HTW(16375), HTW(-16305), HTW(16165), HTW(-15956), HTW(15679), HTW(-15334), HTW(14924), HTW(-14449), HTW(13913), HTW(-13318), HTW(12665), HTW(-11958), HTW(11200), HTW(-10394), HTW(9543), HTW(-8652), HTW(7723), HTW(-6762), HTW(5771), HTW(-4756), HTW(3720), HTW(-2669), HTW(1606), HTW(-536)}; static const FIXP_QTW post_twiddle_sin_32[32] = { HTW(16379), HTW(-16340), HTW(16261), HTW(-16143), HTW(15986), HTW(-15791), HTW(15557), HTW(-15286), HTW(14978), HTW(-14635), HTW(14256), HTW(-13842), HTW(13395), HTW(-12916), HTW(12406), HTW(-11866), HTW(11297), HTW(-10702), HTW(10080), HTW(-9434), HTW(8765), HTW(-8076), HTW(7366), HTW(-6639), HTW(5897), HTW(-5139), HTW(4370), HTW(-3590), HTW(2801), HTW(-2006), HTW(1205), HTW(-402)}; static const FIXP_QTW post_twiddle_sin_40[40] = { HTW(16381), HTW(-16356), HTW(16305), HTW(-16229), HTW(16129), HTW(-16003), HTW(15853), HTW(-15679), HTW(15480), HTW(-15257), HTW(15011), HTW(-14741), HTW(14449), HTW(-14135), HTW(13799), HTW(-13441), HTW(13063), HTW(-12665), HTW(12247), HTW(-11810), HTW(11356), HTW(-10883), HTW(10394), HTW(-9889), HTW(9368), HTW(-8833), HTW(8285), HTW(-7723), HTW(7150), HTW(-6566), HTW(5971), HTW(-5368), HTW(4756), HTW(-4137), HTW(3511), HTW(-2880), HTW(2245), HTW(-1606), HTW(965), HTW(-322)}; static const FIXP_DBL preModCos[32] = { -749875776, 786681536, 711263552, -821592064, -670937792, 854523392, 628995648, -885396032, -585538240, 914135680, 540670208, -940673088, -494499680, 964944384, 447137824, -986891008, -398698816, 1006460096, 349299264, -1023604544, -299058240, 1038283072, 248096752, -1050460288, -196537584, 1060106816, 144504928, -1067199488, -92124160, 1071721152, 39521456, -1073660992}; static const FIXP_DBL preModSin[32] = { 768510144, 730789760, -804379072, -691308864, 838310208, 650162560, -870221760, -607449920, 900036928, 563273856, -927683776, -517740896, 953095808, 470960608, -976211712, -423045728, 996975808, 374111712, -1015338112, -324276416, 1031254400, 273659904, -1044686336, -222384144, 1055601472, 170572640, -1063973632, -118350192, 1069782528, 65842640, -1073014208, -13176464}; /* The cube root function */ /***************************************************************************** functionname: invCubeRootNorm2 description: delivers 1/cuberoot(op) in Q1.31 format and modified exponent *****************************************************************************/ #define CUBE_ROOT_BITS 7 #define CUBE_ROOT_VALUES (128 + 2) #define CUBE_ROOT_BITS_MASK 0x7f #define CUBE_ROOT_FRACT_BITS_MASK 0x007FFFFF /* Inverse cube root table for operands running from 0.5 to 1.0 */ /* (INT) (1.0/cuberoot((op))); */ /* Implicit exponent is 1. */ LNK_SECTION_CONSTDATA static const FIXP_DBL invCubeRootTab[CUBE_ROOT_VALUES] = { (0x50a28be6), (0x506d1172), (0x503823c4), (0x5003c05a), (0x4fcfe4c0), (0x4f9c8e92), (0x4f69bb7d), (0x4f37693b), (0x4f059594), (0x4ed43e5f), (0x4ea36181), (0x4e72fcea), (0x4e430e98), (0x4e139495), (0x4de48cf5), (0x4db5f5db), (0x4d87cd73), (0x4d5a11f2), (0x4d2cc19c), (0x4cffdabb), (0x4cd35ba4), (0x4ca742b7), (0x4c7b8e5c), (0x4c503d05), (0x4c254d2a), (0x4bfabd50), (0x4bd08c00), (0x4ba6b7cd), (0x4b7d3f53), (0x4b542134), (0x4b2b5c18), (0x4b02eeb1), (0x4adad7b8), (0x4ab315ea), (0x4a8ba80d), (0x4a648cec), (0x4a3dc35b), (0x4a174a30), (0x49f1204a), (0x49cb448d), (0x49a5b5e2), (0x49807339), (0x495b7b86), (0x4936cdc2), (0x491268ec), (0x48ee4c08), (0x48ca761f), (0x48a6e63e), (0x48839b76), (0x486094de), (0x483dd190), (0x481b50ad), (0x47f91156), (0x47d712b3), (0x47b553f0), (0x4793d43c), (0x477292c9), (0x47518ece), (0x4730c785), (0x47103c2d), (0x46efec06), (0x46cfd655), (0x46affa61), (0x46905777), (0x4670ece4), (0x4651b9f9), (0x4632be0b), (0x4613f871), (0x45f56885), (0x45d70da5), (0x45b8e72f), (0x459af487), (0x457d3511), (0x455fa835), (0x45424d5d), (0x452523f6), (0x45082b6e), (0x44eb6337), (0x44cecac5), (0x44b2618d), (0x44962708), (0x447a1ab1), (0x445e3c02), (0x44428a7c), (0x4427059e), (0x440bacec), (0x43f07fe9), (0x43d57e1c), (0x43baa70e), (0x439ffa48), (0x43857757), (0x436b1dc8), (0x4350ed2b), (0x4336e511), (0x431d050c), (0x43034cb2), (0x42e9bb98), (0x42d05156), (0x42b70d85), (0x429defc0), (0x4284f7a2), (0x426c24cb), (0x425376d8), (0x423aed6a), (0x42228823), (0x420a46a6), (0x41f22898), (0x41da2d9f), (0x41c25561), (0x41aa9f86), (0x41930bba), (0x417b99a5), (0x416448f5), (0x414d1956), (0x41360a76), (0x411f1c06), (0x41084db5), (0x40f19f35), (0x40db1039), (0x40c4a074), (0x40ae4f9b), (0x40981d64), (0x40820985), (0x406c13b6), (0x40563bb1), (0x4040812e), (0x402ae3e7), (0x40156399), (0x40000000), (0x3FEAB8D9)}; /* n.a. */ static const FIXP_DBL invCubeRootCorrection[3] = {0x40000000, 0x50A28BE6, 0x6597FA95}; /***************************************************************************** * \brief calculate 1.0/cube_root(op), op contains mantissa and exponent * \param op_m: (i) mantissa of operand, must not be zero (0x0000.0000) or * negative * \param op_e: (i) pointer to the exponent of the operand (must be initialized) * and .. (o) pointer to the exponent of the result * \return: (o) mantissa of the result * \description: * This routine calculates the cube root of the input operand, that is * given with its mantissa in Q31 format (FIXP_DBL) and its exponent (INT). * The resulting mantissa is returned in format Q31. The exponent (*op_e) * is modified accordingly. It is not assured, that the result is fully * left-aligned but assumed to have not more than 2 bits headroom. There is one * macro to activate the use of this algorithm: FUNCTION_invCubeRootNorm2 By * means of activating the macro INVCUBEROOTNORM2_LINEAR_INTERPOLATE_HQ, a * slightly higher precision is reachable (by default, not active). For DEBUG * purpose only: a FDK_ASSERT macro validates, if the input mantissa is greater * zero. * */ static #ifdef __arm__ FIXP_DBL FDK_FORCEINLINE invCubeRootNorm2(FIXP_DBL op_m, INT* op_e) #else FIXP_DBL invCubeRootNorm2(FIXP_DBL op_m, INT* op_e) #endif { FDK_ASSERT(op_m > FIXP_DBL(0)); /* normalize input, calculate shift value */ INT exponent = (INT)fNormz(op_m) - 1; op_m <<= exponent; INT index = (INT)(op_m >> (DFRACT_BITS - 1 - (CUBE_ROOT_BITS + 1))) & CUBE_ROOT_BITS_MASK; FIXP_DBL fract = (FIXP_DBL)(((INT)op_m & CUBE_ROOT_FRACT_BITS_MASK) << (CUBE_ROOT_BITS + 1)); FIXP_DBL diff = invCubeRootTab[index + 1] - invCubeRootTab[index]; op_m = fMultAddDiv2(invCubeRootTab[index], diff << 1, fract); #if defined(INVCUBEROOTNORM2_LINEAR_INTERPOLATE_HQ) /* reg1 = t[i] + (t[i+1]-t[i])*fract ... already computed ... + * (1-fract)fract*(t[i+2]-t[i+1])/2 */ if (fract != (FIXP_DBL)0) { /* fract = fract * (1 - fract) */ fract = fMultDiv2(fract, (FIXP_DBL)((LONG)0x80000000 - (LONG)fract)) << 1; diff = diff - (invCubeRootTab[index + 2] - invCubeRootTab[index + 1]); op_m = fMultAddDiv2(op_m, fract, diff); } #endif /* INVCUBEROOTNORM2_LINEAR_INTERPOLATE_HQ */ /* calculate the output exponent = input * exp/3 = cubicroot(m)*2^(exp/3) * where 2^(exp/3) = 2^k'*2 or 2^k'*2^(1/3) or 2^k'*2^(2/3) */ exponent = exponent - *op_e + 3; INT shift_tmp = ((INT)fMultDiv2((FIXP_SGL)fAbs(exponent), (FIXP_SGL)0x5556)) >> 16; if (exponent < 0) { shift_tmp = -shift_tmp; } INT rem = exponent - 3 * shift_tmp; if (rem < 0) { rem += 3; shift_tmp--; } *op_e = shift_tmp; op_m = fMultDiv2(op_m, invCubeRootCorrection[rem]) << 2; return (op_m); } /***************************************************************************** functionname: invFourthRootNorm2 description: delivers 1/FourthRoot(op) in Q1.31 format and modified exponent *****************************************************************************/ #define FOURTHROOT_BITS 7 #define FOURTHROOT_VALUES (128 + 2) #define FOURTHROOT_BITS_MASK 0x7f #define FOURTHROOT_FRACT_BITS_MASK 0x007FFFFF LNK_SECTION_CONSTDATA static const FIXP_DBL invFourthRootTab[FOURTHROOT_VALUES] = { (0x4c1bf829), (0x4bf61977), (0x4bd09843), (0x4bab72ef), (0x4b86a7eb), (0x4b6235ac), (0x4b3e1ab6), (0x4b1a5592), (0x4af6e4d4), (0x4ad3c718), (0x4ab0fb03), (0x4a8e7f42), (0x4a6c5288), (0x4a4a7393), (0x4a28e126), (0x4a079a0c), (0x49e69d16), (0x49c5e91f), (0x49a57d04), (0x498557ac), (0x49657802), (0x4945dcf9), (0x49268588), (0x490770ac), (0x48e89d6a), (0x48ca0ac9), (0x48abb7d6), (0x488da3a6), (0x486fcd4f), (0x485233ed), (0x4834d6a3), (0x4817b496), (0x47faccf0), (0x47de1ee0), (0x47c1a999), (0x47a56c51), (0x47896643), (0x476d96af), (0x4751fcd6), (0x473697ff), (0x471b6773), (0x47006a81), (0x46e5a079), (0x46cb08ae), (0x46b0a279), (0x46966d34), (0x467c683d), (0x466292f4), (0x4648ecbc), (0x462f74fe), (0x46162b20), (0x45fd0e91), (0x45e41ebe), (0x45cb5b19), (0x45b2c315), (0x459a562a), (0x458213cf), (0x4569fb81), (0x45520cbc), (0x453a4701), (0x4522a9d1), (0x450b34b0), (0x44f3e726), (0x44dcc0ba), (0x44c5c0f7), (0x44aee768), (0x4498339e), (0x4481a527), (0x446b3b96), (0x4454f67e), (0x443ed576), (0x4428d815), (0x4412fdf3), (0x43fd46ad), (0x43e7b1de), (0x43d23f23), (0x43bcee1e), (0x43a7be6f), (0x4392afb8), (0x437dc19d), (0x4368f3c5), (0x435445d6), (0x433fb779), (0x432b4856), (0x4316f81a), (0x4302c66f), (0x42eeb305), (0x42dabd8a), (0x42c6e5ad), (0x42b32b21), (0x429f8d96), (0x428c0cc2), (0x4278a859), (0x42656010), (0x4252339e), (0x423f22bc), (0x422c2d23), (0x4219528b), (0x420692b2), (0x41f3ed51), (0x41e16228), (0x41cef0f2), (0x41bc9971), (0x41aa5b62), (0x41983687), (0x41862aa2), (0x41743775), (0x41625cc3), (0x41509a50), (0x413eefe2), (0x412d5d3e), (0x411be22b), (0x410a7e70), (0x40f931d5), (0x40e7fc23), (0x40d6dd24), (0x40c5d4a2), (0x40b4e268), (0x40a40642), (0x40933ffc), (0x40828f64), (0x4071f447), (0x40616e73), (0x4050fdb9), (0x4040a1e6), (0x40305acc), (0x4020283c), (0x40100a08), (0x40000000), (0x3ff009f9), }; static const FIXP_DBL invFourthRootCorrection[4] = {0x40000000, 0x4C1BF829, 0x5A82799A, 0x6BA27E65}; /* The fourth root function */ /***************************************************************************** * \brief calculate 1.0/fourth_root(op), op contains mantissa and exponent * \param op_m: (i) mantissa of operand, must not be zero (0x0000.0000) or * negative * \param op_e: (i) pointer to the exponent of the operand (must be initialized) * and .. (o) pointer to the exponent of the result * \return: (o) mantissa of the result * \description: * This routine calculates the cube root of the input operand, that is * given with its mantissa in Q31 format (FIXP_DBL) and its exponent (INT). * The resulting mantissa is returned in format Q31. The exponent (*op_e) * is modified accordingly. It is not assured, that the result is fully * left-aligned but assumed to have not more than 2 bits headroom. There is one * macro to activate the use of this algorithm: FUNCTION_invFourthRootNorm2 By * means of activating the macro INVFOURTHROOTNORM2_LINEAR_INTERPOLATE_HQ, a * slightly higher precision is reachable (by default, not active). For DEBUG * purpose only: a FDK_ASSERT macro validates, if the input mantissa is greater * zero. * */ /* #define INVFOURTHROOTNORM2_LINEAR_INTERPOLATE_HQ */ static #ifdef __arm__ FIXP_DBL FDK_FORCEINLINE invFourthRootNorm2(FIXP_DBL op_m, INT* op_e) #else FIXP_DBL invFourthRootNorm2(FIXP_DBL op_m, INT* op_e) #endif { FDK_ASSERT(op_m > FL2FXCONST_DBL(0.0)); /* normalize input, calculate shift value */ INT exponent = (INT)fNormz(op_m) - 1; op_m <<= exponent; INT index = (INT)(op_m >> (DFRACT_BITS - 1 - (FOURTHROOT_BITS + 1))) & FOURTHROOT_BITS_MASK; FIXP_DBL fract = (FIXP_DBL)(((INT)op_m & FOURTHROOT_FRACT_BITS_MASK) << (FOURTHROOT_BITS + 1)); FIXP_DBL diff = invFourthRootTab[index + 1] - invFourthRootTab[index]; op_m = invFourthRootTab[index] + (fMultDiv2(diff, fract) << 1); #if defined(INVFOURTHROOTNORM2_LINEAR_INTERPOLATE_HQ) /* reg1 = t[i] + (t[i+1]-t[i])*fract ... already computed ... + * (1-fract)fract*(t[i+2]-t[i+1])/2 */ if (fract != (FIXP_DBL)0) { /* fract = fract * (1 - fract) */ fract = fMultDiv2(fract, (FIXP_DBL)((LONG)0x80000000 - (LONG)fract)) << 1; diff = diff - (invFourthRootTab[index + 2] - invFourthRootTab[index + 1]); op_m = fMultAddDiv2(op_m, fract, diff); } #endif /* INVFOURTHROOTNORM2_LINEAR_INTERPOLATE_HQ */ exponent = exponent - *op_e + 4; INT rem = exponent & 0x00000003; INT shift_tmp = (exponent >> 2); *op_e = shift_tmp; op_m = fMultDiv2(op_m, invFourthRootCorrection[rem]) << 2; return (op_m); } /***************************************************************************** functionname: inv3EigthRootNorm2 description: delivers 1/cubert(op) normalized to .5...1 and the shift value of the OUTPUT *****************************************************************************/ #define THREEIGTHROOT_BITS 7 #define THREEIGTHROOT_VALUES (128 + 2) #define THREEIGTHROOT_BITS_MASK 0x7f #define THREEIGTHROOT_FRACT_BITS_MASK 0x007FFFFF LNK_SECTION_CONSTDATA static const FIXP_DBL inv3EigthRootTab[THREEIGTHROOT_VALUES] = { (0x45cae0f2), (0x45b981bf), (0x45a8492a), (0x45973691), (0x45864959), (0x457580e6), (0x4564dca4), (0x45545c00), (0x4543fe6b), (0x4533c35a), (0x4523aa44), (0x4513b2a4), (0x4503dbf7), (0x44f425be), (0x44e48f7b), (0x44d518b6), (0x44c5c0f7), (0x44b687c8), (0x44a76cb8), (0x44986f58), (0x44898f38), (0x447acbef), (0x446c2514), (0x445d9a3f), (0x444f2b0d), (0x4440d71a), (0x44329e07), (0x44247f73), (0x44167b04), (0x4408905e), (0x43fabf28), (0x43ed070b), (0x43df67b0), (0x43d1e0c5), (0x43c471f7), (0x43b71af6), (0x43a9db71), (0x439cb31c), (0x438fa1ab), (0x4382a6d2), (0x4375c248), (0x4368f3c5), (0x435c3b03), (0x434f97bc), (0x434309ac), (0x43369091), (0x432a2c28), (0x431ddc30), (0x4311a06c), (0x4305789c), (0x42f96483), (0x42ed63e5), (0x42e17688), (0x42d59c30), (0x42c9d4a6), (0x42be1fb1), (0x42b27d1a), (0x42a6ecac), (0x429b6e2f), (0x42900172), (0x4284a63f), (0x42795c64), (0x426e23b0), (0x4262fbf2), (0x4257e4f9), (0x424cde96), (0x4241e89a), (0x423702d8), (0x422c2d23), (0x4221674d), (0x4216b12c), (0x420c0a94), (0x4201735b), (0x41f6eb57), (0x41ec725f), (0x41e2084b), (0x41d7acf3), (0x41cd6030), (0x41c321db), (0x41b8f1ce), (0x41aecfe5), (0x41a4bbf8), (0x419ab5e6), (0x4190bd89), (0x4186d2bf), (0x417cf565), (0x41732558), (0x41696277), (0x415faca1), (0x415603b4), (0x414c6792), (0x4142d818), (0x4139552a), (0x412fdea6), (0x41267470), (0x411d1668), (0x4113c472), (0x410a7e70), (0x41014445), (0x40f815d4), (0x40eef302), (0x40e5dbb4), (0x40dccfcd), (0x40d3cf33), (0x40cad9cb), (0x40c1ef7b), (0x40b9102a), (0x40b03bbd), (0x40a7721c), (0x409eb32e), (0x4095feda), (0x408d5508), (0x4084b5a0), (0x407c208b), (0x407395b2), (0x406b14fd), (0x40629e56), (0x405a31a6), (0x4051ced8), (0x404975d5), (0x40412689), (0x4038e0dd), (0x4030a4bd), (0x40287215), (0x402048cf), (0x401828d7), (0x4010121a), (0x40080483), (0x40000000), (0x3ff8047d), }; /* The last value is rounded in order to avoid any overflow due to the values * range of the root table */ static const FIXP_DBL inv3EigthRootCorrection[8] = { 0x40000000, 0x45CAE0F2, 0x4C1BF829, 0x52FF6B55, 0x5A82799A, 0x62B39509, 0x6BA27E65, 0x75606373}; /* The 3/8 root function */ /***************************************************************************** * \brief calculate 1.0/3Eigth_root(op) = 1.0/(x)^(3/8), op contains mantissa * and exponent * \param op_m: (i) mantissa of operand, must not be zero (0x0000.0000) or * negative * \param op_e: (i) pointer to the exponent of the operand (must be initialized) * and .. (o) pointer to the exponent of the result * \return: (o) mantissa of the result * \description: * This routine calculates the cube root of the input operand, that is * given with its mantissa in Q31 format (FIXP_DBL) and its exponent (INT). * The resulting mantissa is returned in format Q31. The exponent (*op_e) * is modified accordingly. It is not assured, that the result is fully * left-aligned but assumed to have not more than 2 bits headroom. There is one * macro to activate the use of this algorithm: FUNCTION_inv3EigthRootNorm2 By * means of activating the macro INVTHREEIGTHROOTNORM2_LINEAR_INTERPOLATE_HQ, a * slightly higher precision is reachable (by default, not active). For DEBUG * purpose only: a FDK_ASSERT macro validates, if the input mantissa is greater * zero. * */ /* #define INVTHREEIGTHROOTNORM2_LINEAR_INTERPOLATE_HQ */ static #ifdef __arm__ FIXP_DBL FDK_FORCEINLINE inv3EigthRootNorm2(FIXP_DBL op_m, INT* op_e) #else FIXP_DBL inv3EigthRootNorm2(FIXP_DBL op_m, INT* op_e) #endif { FDK_ASSERT(op_m > FL2FXCONST_DBL(0.0)); /* normalize input, calculate shift op_mue */ INT exponent = (INT)fNormz(op_m) - 1; op_m <<= exponent; INT index = (INT)(op_m >> (DFRACT_BITS - 1 - (THREEIGTHROOT_BITS + 1))) & THREEIGTHROOT_BITS_MASK; FIXP_DBL fract = (FIXP_DBL)(((INT)op_m & THREEIGTHROOT_FRACT_BITS_MASK) << (THREEIGTHROOT_BITS + 1)); FIXP_DBL diff = inv3EigthRootTab[index + 1] - inv3EigthRootTab[index]; op_m = inv3EigthRootTab[index] + (fMultDiv2(diff, fract) << 1); #if defined(INVTHREEIGTHROOTNORM2_LINEAR_INTERPOLATE_HQ) /* op_m = t[i] + (t[i+1]-t[i])*fract ... already computed ... + * (1-fract)fract*(t[i+2]-t[i+1])/2 */ if (fract != (FIXP_DBL)0) { /* fract = fract * (1 - fract) */ fract = fMultDiv2(fract, (FIXP_DBL)((LONG)0x80000000 - (LONG)fract)) << 1; diff = diff - (inv3EigthRootTab[index + 2] - inv3EigthRootTab[index + 1]); op_m = fMultAddDiv2(op_m, fract, diff); } #endif /* INVTHREEIGTHROOTNORM2_LINEAR_INTERPOLATE_HQ */ exponent = exponent - *op_e + 8; INT rem = exponent & 0x00000007; INT shift_tmp = (exponent >> 3); *op_e = shift_tmp * 3; op_m = fMultDiv2(op_m, inv3EigthRootCorrection[rem]) << 2; return (fMult(op_m, fMult(op_m, op_m))); } SBR_ERROR QmfTransposerCreate(HANDLE_HBE_TRANSPOSER* hQmfTransposer, const int frameSize, int bDisableCrossProducts, int bSbr41) { HANDLE_HBE_TRANSPOSER hQmfTran = NULL; int i; if (hQmfTransposer != NULL) { /* Memory allocation */ /*--------------------------------------------------------------------------------------------*/ hQmfTran = (HANDLE_HBE_TRANSPOSER)FDKcalloc(1, sizeof(struct hbeTransposer)); if (hQmfTran == NULL) { return SBRDEC_MEM_ALLOC_FAILED; } for (i = 0; i < MAX_STRETCH_HBE - 1; i++) { hQmfTran->bXProducts[i] = (bDisableCrossProducts ? 0 : xProducts[i]); } hQmfTran->timeDomainWinLen = frameSize; if (frameSize == 768) { hQmfTran->noCols = (8 * frameSize / 3) / QMF_SYNTH_CHANNELS; /* 32 for 24:64 */ } else { hQmfTran->noCols = (bSbr41 + 1) * 2 * frameSize / QMF_SYNTH_CHANNELS; /* 32 for 32:64 and 64 for 16:64 -> identical to sbrdec->no_cols */ } hQmfTran->noChannels = frameSize / hQmfTran->noCols; hQmfTran->qmfInBufSize = QMF_WIN_LEN; hQmfTran->qmfOutBufSize = 2 * (hQmfTran->noCols / 2 + QMF_WIN_LEN - 1); hQmfTran->inBuf_F = (LONG*)FDKcalloc(QMF_SYNTH_CHANNELS + 20 + 1, sizeof(LONG)); /* buffered time signal needs to be delayed by synthesis_size; max * synthesis_size = 20; */ if (hQmfTran->inBuf_F == NULL) { QmfTransposerClose(hQmfTran); return SBRDEC_MEM_ALLOC_FAILED; } hQmfTran->qmfInBufReal_F = (FIXP_DBL**)FDKcalloc(hQmfTran->qmfInBufSize, sizeof(FIXP_DBL*)); hQmfTran->qmfInBufImag_F = (FIXP_DBL**)FDKcalloc(hQmfTran->qmfInBufSize, sizeof(FIXP_DBL*)); if (hQmfTran->qmfInBufReal_F == NULL) { QmfTransposerClose(hQmfTran); return SBRDEC_MEM_ALLOC_FAILED; } if (hQmfTran->qmfInBufImag_F == NULL) { QmfTransposerClose(hQmfTran); return SBRDEC_MEM_ALLOC_FAILED; } for (i = 0; i < hQmfTran->qmfInBufSize; i++) { hQmfTran->qmfInBufReal_F[i] = (FIXP_DBL*)FDKaalloc( QMF_SYNTH_CHANNELS * sizeof(FIXP_DBL), ALIGNMENT_DEFAULT); hQmfTran->qmfInBufImag_F[i] = (FIXP_DBL*)FDKaalloc( QMF_SYNTH_CHANNELS * sizeof(FIXP_DBL), ALIGNMENT_DEFAULT); if (hQmfTran->qmfInBufReal_F[i] == NULL) { QmfTransposerClose(hQmfTran); return SBRDEC_MEM_ALLOC_FAILED; } if (hQmfTran->qmfInBufImag_F[i] == NULL) { QmfTransposerClose(hQmfTran); return SBRDEC_MEM_ALLOC_FAILED; } } hQmfTran->qmfHBEBufReal_F = (FIXP_DBL**)FDKcalloc(HBE_MAX_OUT_SLOTS, sizeof(FIXP_DBL*)); hQmfTran->qmfHBEBufImag_F = (FIXP_DBL**)FDKcalloc(HBE_MAX_OUT_SLOTS, sizeof(FIXP_DBL*)); if (hQmfTran->qmfHBEBufReal_F == NULL) { QmfTransposerClose(hQmfTran); return SBRDEC_MEM_ALLOC_FAILED; } if (hQmfTran->qmfHBEBufImag_F == NULL) { QmfTransposerClose(hQmfTran); return SBRDEC_MEM_ALLOC_FAILED; } for (i = 0; i < HBE_MAX_OUT_SLOTS; i++) { hQmfTran->qmfHBEBufReal_F[i] = (FIXP_DBL*)FDKcalloc(QMF_SYNTH_CHANNELS, sizeof(FIXP_DBL)); hQmfTran->qmfHBEBufImag_F[i] = (FIXP_DBL*)FDKcalloc(QMF_SYNTH_CHANNELS, sizeof(FIXP_DBL)); if (hQmfTran->qmfHBEBufReal_F[i] == NULL) { QmfTransposerClose(hQmfTran); return SBRDEC_MEM_ALLOC_FAILED; } if (hQmfTran->qmfHBEBufImag_F[i] == NULL) { QmfTransposerClose(hQmfTran); return SBRDEC_MEM_ALLOC_FAILED; } } hQmfTran->qmfBufferCodecTempSlot_F = (FIXP_DBL*)FDKcalloc(QMF_SYNTH_CHANNELS / 2, sizeof(FIXP_DBL)); if (hQmfTran->qmfBufferCodecTempSlot_F == NULL) { QmfTransposerClose(hQmfTran); return SBRDEC_MEM_ALLOC_FAILED; } hQmfTran->bSbr41 = bSbr41; hQmfTran->highband_exp[0] = 0; hQmfTran->highband_exp[1] = 0; hQmfTran->target_exp[0] = 0; hQmfTran->target_exp[1] = 0; *hQmfTransposer = hQmfTran; } return SBRDEC_OK; } SBR_ERROR QmfTransposerReInit(HANDLE_HBE_TRANSPOSER hQmfTransposer, UCHAR* FreqBandTable[2], UCHAR NSfb[2]) /* removed bSbr41 from parameterlist: don't know where to get this value from at call-side */ { int L, sfb, patch, stopPatch, qmfErr; if (hQmfTransposer != NULL) { const FIXP_QTW* tmp_t_cos; const FIXP_QTW* tmp_t_sin; hQmfTransposer->startBand = FreqBandTable[0][0]; FDK_ASSERT((!hQmfTransposer->bSbr41 && hQmfTransposer->startBand <= 32) || (hQmfTransposer->bSbr41 && hQmfTransposer->startBand <= 16)); /* is checked by resetFreqBandTables() */ hQmfTransposer->stopBand = FreqBandTable[0][NSfb[0]]; hQmfTransposer->synthSize = 4 * ((hQmfTransposer->startBand + 4) / 8 + 1); /* 8, 12, 16, 20 */ hQmfTransposer->kstart = startSubband2kL[hQmfTransposer->startBand]; /* don't know where to take this information from */ /* hQmfTransposer->bSbr41 = bSbr41; */ if (hQmfTransposer->bSbr41) { if (hQmfTransposer->kstart + hQmfTransposer->synthSize > 16) hQmfTransposer->kstart = 16 - hQmfTransposer->synthSize; } else if (hQmfTransposer->timeDomainWinLen == 768) { if (hQmfTransposer->kstart + hQmfTransposer->synthSize > 24) hQmfTransposer->kstart = 24 - hQmfTransposer->synthSize; } hQmfTransposer->synthesisQmfPreModCos_F = &preModCos[hQmfTransposer->kstart]; hQmfTransposer->synthesisQmfPreModSin_F = &preModSin[hQmfTransposer->kstart]; L = 2 * hQmfTransposer->synthSize; /* 8, 16, 24, 32, 40 */ /* Change analysis post twiddles */ switch (L) { case 8: tmp_t_cos = post_twiddle_cos_8; tmp_t_sin = post_twiddle_sin_8; break; case 16: tmp_t_cos = post_twiddle_cos_16; tmp_t_sin = post_twiddle_sin_16; break; case 24: tmp_t_cos = post_twiddle_cos_24; tmp_t_sin = post_twiddle_sin_24; break; case 32: tmp_t_cos = post_twiddle_cos_32; tmp_t_sin = post_twiddle_sin_32; break; case 40: tmp_t_cos = post_twiddle_cos_40; tmp_t_sin = post_twiddle_sin_40; break; default: return SBRDEC_UNSUPPORTED_CONFIG; } qmfErr = qmfInitSynthesisFilterBank( &hQmfTransposer->HBESynthesisQMF, hQmfTransposer->synQmfStates, hQmfTransposer->noCols, 0, hQmfTransposer->synthSize, hQmfTransposer->synthSize, 1); if (qmfErr != 0) { return SBRDEC_UNSUPPORTED_CONFIG; } qmfErr = qmfInitAnalysisFilterBank( &hQmfTransposer->HBEAnalysiscQMF, hQmfTransposer->anaQmfStates, hQmfTransposer->noCols / 2, 0, 2 * hQmfTransposer->synthSize, 2 * hQmfTransposer->synthSize, 0); if (qmfErr != 0) { return SBRDEC_UNSUPPORTED_CONFIG; } hQmfTransposer->HBEAnalysiscQMF.t_cos = tmp_t_cos; hQmfTransposer->HBEAnalysiscQMF.t_sin = tmp_t_sin; FDKmemset(hQmfTransposer->xOverQmf, 0, MAX_NUM_PATCHES * sizeof(int)); /* global */ sfb = 0; if (hQmfTransposer->bSbr41) { stopPatch = MAX_NUM_PATCHES; hQmfTransposer->maxStretch = MAX_STRETCH_HBE; } else { stopPatch = MAX_STRETCH_HBE; } for (patch = 1; patch <= stopPatch; patch++) { while (sfb <= NSfb[0] && FreqBandTable[0][sfb] <= patch * hQmfTransposer->startBand) sfb++; if (sfb <= NSfb[0]) { /* If the distance is larger than three QMF bands - try aligning to high * resolution frequency bands instead. */ if ((patch * hQmfTransposer->startBand - FreqBandTable[0][sfb - 1]) <= 3) { hQmfTransposer->xOverQmf[patch - 1] = FreqBandTable[0][sfb - 1]; } else { int sfb_tmp = 0; while (sfb_tmp <= NSfb[1] && FreqBandTable[1][sfb_tmp] <= patch * hQmfTransposer->startBand) sfb_tmp++; hQmfTransposer->xOverQmf[patch - 1] = FreqBandTable[1][sfb_tmp - 1]; } } else { hQmfTransposer->xOverQmf[patch - 1] = hQmfTransposer->stopBand; hQmfTransposer->maxStretch = fMin(patch, MAX_STRETCH_HBE); break; } } hQmfTransposer->highband_exp[0] = 0; hQmfTransposer->highband_exp[1] = 0; hQmfTransposer->target_exp[0] = 0; hQmfTransposer->target_exp[1] = 0; } return SBRDEC_OK; } void QmfTransposerClose(HANDLE_HBE_TRANSPOSER hQmfTransposer) { int i; if (hQmfTransposer != NULL) { if (hQmfTransposer->inBuf_F) FDKfree(hQmfTransposer->inBuf_F); if (hQmfTransposer->qmfInBufReal_F) { for (i = 0; i < hQmfTransposer->qmfInBufSize; i++) { FDKafree(hQmfTransposer->qmfInBufReal_F[i]); } FDKfree(hQmfTransposer->qmfInBufReal_F); } if (hQmfTransposer->qmfInBufImag_F) { for (i = 0; i < hQmfTransposer->qmfInBufSize; i++) { FDKafree(hQmfTransposer->qmfInBufImag_F[i]); } FDKfree(hQmfTransposer->qmfInBufImag_F); } if (hQmfTransposer->qmfHBEBufReal_F) { for (i = 0; i < HBE_MAX_OUT_SLOTS; i++) { FDKfree(hQmfTransposer->qmfHBEBufReal_F[i]); } FDKfree(hQmfTransposer->qmfHBEBufReal_F); } if (hQmfTransposer->qmfHBEBufImag_F) { for (i = 0; i < HBE_MAX_OUT_SLOTS; i++) { FDKfree(hQmfTransposer->qmfHBEBufImag_F[i]); } FDKfree(hQmfTransposer->qmfHBEBufImag_F); } FDKfree(hQmfTransposer->qmfBufferCodecTempSlot_F); FDKfree(hQmfTransposer); } } inline void scaleUp(FIXP_DBL* real_m, FIXP_DBL* imag_m, INT* _e) { INT reserve; /* shift gc_r and gc_i up if possible */ reserve = CntLeadingZeros((INT(*real_m) ^ INT((*real_m >> 31))) | (INT(*imag_m) ^ INT((*imag_m >> 31)))) - 1; reserve = fMax(reserve - 1, 0); /* Leave one bit headroom such that (real_m^2 + imag_m^2) does not overflow later if both are 0x80000000. */ reserve = fMin(reserve, *_e); FDK_ASSERT(reserve >= 0); *real_m <<= reserve; *imag_m <<= reserve; *_e -= reserve; } static void calculateCenterFIXP(FIXP_DBL gammaVecReal, FIXP_DBL gammaVecImag, FIXP_DBL* centerReal, FIXP_DBL* centerImag, INT* exponent, int stretch, int mult) { scaleUp(&gammaVecReal, &gammaVecImag, exponent); FIXP_DBL energy = fPow2Div2(gammaVecReal) + fPow2Div2(gammaVecImag); if (energy != FL2FXCONST_DBL(0.f)) { FIXP_DBL gc_r_m, gc_i_m, factor_m = (FIXP_DBL)0; INT factor_e, gc_e; factor_e = 2 * (*exponent) + 1; switch (stretch) { case 2: factor_m = invFourthRootNorm2(energy, &factor_e); break; case 3: factor_m = invCubeRootNorm2(energy, &factor_e); break; case 4: factor_m = inv3EigthRootNorm2(energy, &factor_e); break; } gc_r_m = fMultDiv2(gammaVecReal, factor_m); /* exponent = HBE_SCALE + factor_e + 1 */ gc_i_m = fMultDiv2(gammaVecImag, factor_m); /* exponent = HBE_SCALE + factor_e + 1*/ gc_e = *exponent + factor_e + 1; scaleUp(&gc_r_m, &gc_i_m, &gc_e); switch (mult) { case 0: *centerReal = gc_r_m; *centerImag = gc_i_m; break; case 1: *centerReal = fPow2Div2(gc_r_m) - fPow2Div2(gc_i_m); *centerImag = fMult(gc_r_m, gc_i_m); gc_e = 2 * gc_e + 1; break; case 2: FIXP_DBL tmp_r = gc_r_m; FIXP_DBL tmp_i = gc_i_m; gc_r_m = fPow2Div2(gc_r_m) - fPow2Div2(gc_i_m); gc_i_m = fMult(tmp_r, gc_i_m); gc_e = 3 * gc_e + 1 + 1; cplxMultDiv2(¢erReal[0], ¢erImag[0], gc_r_m, gc_i_m, tmp_r, tmp_i); break; } scaleUp(centerReal, centerImag, &gc_e); FDK_ASSERT(gc_e >= 0); *exponent = gc_e; } else { *centerReal = energy; /* energy = 0 */ *centerImag = energy; /* energy = 0 */ *exponent = (INT)energy; } } static int getHBEScaleFactorFrame(const int bSbr41, const int maxStretch, const int pitchInBins) { if (pitchInBins >= pmin * (1 + bSbr41)) { /* crossproducts enabled */ return 26; } else { return (maxStretch == 2) ? 24 : 25; } } static void addHighBandPart(FIXP_DBL g_r_m, FIXP_DBL g_i_m, INT g_e, FIXP_DBL mult, FIXP_DBL gammaCenterReal_m, FIXP_DBL gammaCenterImag_m, INT gammaCenter_e, INT stretch, INT scale_factor_hbe, FIXP_DBL* qmfHBEBufReal_F, FIXP_DBL* qmfHBEBufImag_F) { if ((g_r_m | g_i_m) != FL2FXCONST_DBL(0.f)) { FIXP_DBL factor_m = (FIXP_DBL)0; INT factor_e; INT add = (stretch == 4) ? 1 : 0; INT shift = (stretch == 4) ? 1 : 2; scaleUp(&g_r_m, &g_i_m, &g_e); FIXP_DBL energy = fPow2AddDiv2(fPow2Div2(g_r_m), g_i_m); factor_e = 2 * g_e + 1; switch (stretch) { case 2: factor_m = invFourthRootNorm2(energy, &factor_e); break; case 3: factor_m = invCubeRootNorm2(energy, &factor_e); break; case 4: factor_m = inv3EigthRootNorm2(energy, &factor_e); break; } factor_m = fMult(factor_m, mult); FIXP_DBL tmp_r, tmp_i; cplxMultDiv2(&tmp_r, &tmp_i, g_r_m, g_i_m, gammaCenterReal_m, gammaCenterImag_m); g_r_m = fMultDiv2(tmp_r, factor_m) << shift; g_i_m = fMultDiv2(tmp_i, factor_m) << shift; g_e = scale_factor_hbe - (g_e + factor_e + gammaCenter_e + add); g_e = fMax((INT)0, g_e); *qmfHBEBufReal_F += g_r_m >> g_e; *qmfHBEBufImag_F += g_i_m >> g_e; } } void QmfTransposerApply(HANDLE_HBE_TRANSPOSER hQmfTransposer, FIXP_DBL** qmfBufferCodecReal, FIXP_DBL** qmfBufferCodecImag, int nColsIn, FIXP_DBL** ppQmfBufferOutReal_F, FIXP_DBL** ppQmfBufferOutImag_F, FIXP_DBL lpcFilterStatesReal[2 + (3 * (4))][(64)], FIXP_DBL lpcFilterStatesImag[2 + (3 * (4))][(64)], int pitchInBins, int scale_lb, int scale_hbe, int* scale_hb, int timeStep, int firstSlotOffsset, int ov_len, KEEP_STATES_SYNCED_MODE keepStatesSyncedMode) { int i, j, stretch, band, sourceband, r, s; int qmfVocoderColsIn = hQmfTransposer->noCols / 2; int bSbr41 = hQmfTransposer->bSbr41; const int winLength[3] = {10, 8, 6}; const int slotOffset = 6; /* hQmfTransposer->winLen-6; */ int qmfOffset = 2 * hQmfTransposer->kstart; int scale_border = (nColsIn == 64) ? 32 : nColsIn; INT slot_stretch4[9] = {0, 0, 0, 0, 2, 4, 6, 8, 10}; INT slot_stretch2[11] = {0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10}; INT slot_stretch3[10] = {0, 0, 0, 1, 3, 4, 6, 7, 9, 10}; INT filt_stretch3[10] = {0, 0, 0, 1, 0, 1, 0, 1, 0, 1}; INT filt_dummy[11] = {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0}; INT* pSlotStretch; INT* pFilt; int offset = 0; /* where to take QmfTransposer data */ int signPreMod = (hQmfTransposer->synthesisQmfPreModCos_F[0] < FL2FXCONST_DBL(0.f)) ? 1 : -1; int scale_factor_hbe = getHBEScaleFactorFrame(bSbr41, hQmfTransposer->maxStretch, pitchInBins); if (keepStatesSyncedMode != KEEP_STATES_SYNCED_OFF) { offset = hQmfTransposer->noCols - ov_len - LPC_ORDER; } hQmfTransposer->highband_exp[0] = hQmfTransposer->highband_exp[1]; hQmfTransposer->target_exp[0] = hQmfTransposer->target_exp[1]; hQmfTransposer->highband_exp[1] = scale_factor_hbe; hQmfTransposer->target_exp[1] = fixMax(hQmfTransposer->highband_exp[1], hQmfTransposer->highband_exp[0]); scale_factor_hbe = hQmfTransposer->target_exp[1]; int shift_ov = hQmfTransposer->target_exp[0] - hQmfTransposer->target_exp[1]; if (shift_ov != 0) { for (i = 0; i < HBE_MAX_OUT_SLOTS; i++) { for (band = 0; band < QMF_SYNTH_CHANNELS; band++) { if (shift_ov >= 0) { hQmfTransposer->qmfHBEBufReal_F[i][band] <<= shift_ov; hQmfTransposer->qmfHBEBufImag_F[i][band] <<= shift_ov; } else { hQmfTransposer->qmfHBEBufReal_F[i][band] >>= (-shift_ov); hQmfTransposer->qmfHBEBufImag_F[i][band] >>= (-shift_ov); } } } } if ((keepStatesSyncedMode == KEEP_STATES_SYNCED_OFF) && shift_ov != 0) { for (i = timeStep * firstSlotOffsset; i < ov_len; i++) { for (band = hQmfTransposer->startBand; band < hQmfTransposer->stopBand; band++) { if (shift_ov >= 0) { ppQmfBufferOutReal_F[i][band] <<= shift_ov; ppQmfBufferOutImag_F[i][band] <<= shift_ov; } else { ppQmfBufferOutReal_F[i][band] >>= (-shift_ov); ppQmfBufferOutImag_F[i][band] >>= (-shift_ov); } } } /* shift lpc filterstates */ for (i = 0; i < timeStep * firstSlotOffsset + LPC_ORDER; i++) { for (band = 0; band < (64); band++) { if (shift_ov >= 0) { lpcFilterStatesReal[i][band] <<= shift_ov; lpcFilterStatesImag[i][band] <<= shift_ov; } else { lpcFilterStatesReal[i][band] >>= (-shift_ov); lpcFilterStatesImag[i][band] >>= (-shift_ov); } } } } FIXP_DBL twid_m_new[3][2]; /* [stretch][cos/sin] */ INT stepsize = 1 + !bSbr41, sine_offset = 24, mod = 96; INT mult[3] = {1, 2, 3}; for (s = 0; s <= MAX_STRETCH_HBE - 2; s++) { twid_m_new[s][0] = twiddle[(mult[s] * (stepsize * pitchInBins)) % mod]; twid_m_new[s][1] = twiddle[((mult[s] * (stepsize * pitchInBins)) + sine_offset) % mod]; } /* Time-stretch */ for (j = 0; j < qmfVocoderColsIn; j++) { int sign = -1, k, z, addrshift, codecTemp_e; /* update inbuf */ for (i = 0; i < hQmfTransposer->synthSize; i++) { hQmfTransposer->inBuf_F[i] = hQmfTransposer->inBuf_F[i + 2 * hQmfTransposer->synthSize]; } /* run synthesis for two sbr slots as transposer uses half slots double bands representation */ for (z = 0; z < 2; z++) { int scale_factor = ((nColsIn == 64) && ((2 * j + z) < scale_border)) ? scale_lb : scale_hbe; codecTemp_e = scale_factor - 1; /* -2 for Div2 and cos/sin scale of 1 */ for (k = 0; k < hQmfTransposer->synthSize; k++) { int ki = hQmfTransposer->kstart + k; hQmfTransposer->qmfBufferCodecTempSlot_F[k] = fMultDiv2(signPreMod * hQmfTransposer->synthesisQmfPreModCos_F[k], qmfBufferCodecReal[2 * j + z][ki]); hQmfTransposer->qmfBufferCodecTempSlot_F[k] += fMultDiv2(signPreMod * hQmfTransposer->synthesisQmfPreModSin_F[k], qmfBufferCodecImag[2 * j + z][ki]); } C_AALLOC_SCRATCH_START(pWorkBuffer, FIXP_DBL, (HBE_MAX_QMF_BANDS << 1)); qmfSynthesisFilteringSlot( &hQmfTransposer->HBESynthesisQMF, hQmfTransposer->qmfBufferCodecTempSlot_F, NULL, 0, -7 - hQmfTransposer->HBESynthesisQMF.filterScale - codecTemp_e + 1, hQmfTransposer->inBuf_F + hQmfTransposer->synthSize * (z + 1), 1, pWorkBuffer); C_AALLOC_SCRATCH_END(pWorkBuffer, FIXP_DBL, (HBE_MAX_QMF_BANDS << 1)); } C_AALLOC_SCRATCH_START(pWorkBuffer, FIXP_DBL, (HBE_MAX_QMF_BANDS << 1)); qmfAnalysisFilteringSlot(&hQmfTransposer->HBEAnalysiscQMF, hQmfTransposer->qmfInBufReal_F[QMF_WIN_LEN - 1], hQmfTransposer->qmfInBufImag_F[QMF_WIN_LEN - 1], hQmfTransposer->inBuf_F + 1, 1, pWorkBuffer); C_AALLOC_SCRATCH_END(pWorkBuffer, FIXP_DBL, (HBE_MAX_QMF_BANDS << 1)); if ((keepStatesSyncedMode == KEEP_STATES_SYNCED_NORMAL) && j <= qmfVocoderColsIn - ((LPC_ORDER + ov_len + QMF_WIN_LEN - 1) >> 1)) { /* update in buffer */ for (i = 0; i < QMF_WIN_LEN - 1; i++) { FDKmemcpy( hQmfTransposer->qmfInBufReal_F[i], hQmfTransposer->qmfInBufReal_F[i + 1], sizeof(FIXP_DBL) * hQmfTransposer->HBEAnalysiscQMF.no_channels); FDKmemcpy( hQmfTransposer->qmfInBufImag_F[i], hQmfTransposer->qmfInBufImag_F[i + 1], sizeof(FIXP_DBL) * hQmfTransposer->HBEAnalysiscQMF.no_channels); } continue; } for (stretch = 2; stretch <= hQmfTransposer->maxStretch; stretch++) { int start = slotOffset - winLength[stretch - 2] / 2; int stop = slotOffset + winLength[stretch - 2] / 2; FIXP_DBL factor = FL2FXCONST_DBL(1.f / 3.f); for (band = hQmfTransposer->xOverQmf[stretch - 2]; band < hQmfTransposer->xOverQmf[stretch - 1]; band++) { FIXP_DBL gammaCenterReal_m[2] = {(FIXP_DBL)0, (FIXP_DBL)0}, gammaCenterImag_m[2] = {(FIXP_DBL)0, (FIXP_DBL)0}; INT gammaCenter_e[2] = {0, 0}; FIXP_DBL gammaVecReal_m[2] = {(FIXP_DBL)0, (FIXP_DBL)0}, gammaVecImag_m[2] = {(FIXP_DBL)0, (FIXP_DBL)0}; INT gammaVec_e[2] = {0, 0}; FIXP_DBL wingain = (FIXP_DBL)0; gammaCenter_e[0] = SCALE2EXP(-hQmfTransposer->HBEAnalysiscQMF.outScalefactor); gammaCenter_e[1] = SCALE2EXP(-hQmfTransposer->HBEAnalysiscQMF.outScalefactor); /* interpolation filters for 3rd order */ sourceband = 2 * band / stretch - qmfOffset; FDK_ASSERT(sourceband >= 0); /* maximum gammaCenter_e == 20 */ calculateCenterFIXP( hQmfTransposer->qmfInBufReal_F[slotOffset][sourceband], hQmfTransposer->qmfInBufImag_F[slotOffset][sourceband], &gammaCenterReal_m[0], &gammaCenterImag_m[0], &gammaCenter_e[0], stretch, stretch - 2); if (stretch == 4) { r = band - 2 * (band / 2); sourceband += (r == 0) ? -1 : 1; pSlotStretch = slot_stretch4; factor = FL2FXCONST_DBL(2.f / 3.f); pFilt = filt_dummy; } else if (stretch == 2) { r = 0; sourceband = 2 * band / stretch - qmfOffset; pSlotStretch = slot_stretch2; factor = FL2FXCONST_DBL(1.f / 3.f); pFilt = filt_dummy; } else { r = 2 * band - 3 * (2 * band / 3); sourceband = 2 * band / stretch - qmfOffset; pSlotStretch = slot_stretch3; factor = FL2FXCONST_DBL(1.4142f / 3.0f); pFilt = filt_stretch3; } if (r == 2) { calculateCenterFIXP( hQmfTransposer->qmfInBufReal_F[slotOffset][sourceband + 1], hQmfTransposer->qmfInBufImag_F[slotOffset][sourceband + 1], &gammaCenterReal_m[1], &gammaCenterImag_m[1], &gammaCenter_e[1], stretch, stretch - 2); factor = FL2FXCONST_DBL(1.4142f / 6.0f); } if (r == 2) { for (k = start; k < stop; k++) { gammaVecReal_m[0] = hQmfTransposer->qmfInBufReal_F[pSlotStretch[k]][sourceband]; gammaVecReal_m[1] = hQmfTransposer->qmfInBufReal_F[pSlotStretch[k]][sourceband + 1]; gammaVecImag_m[0] = hQmfTransposer->qmfInBufImag_F[pSlotStretch[k]][sourceband]; gammaVecImag_m[1] = hQmfTransposer->qmfInBufImag_F[pSlotStretch[k]][sourceband + 1]; gammaVec_e[0] = gammaVec_e[1] = SCALE2EXP(-hQmfTransposer->HBEAnalysiscQMF.outScalefactor); if (pFilt[k] == 1) { FIXP_DBL tmpRealF = gammaVecReal_m[0], tmpImagF; gammaVecReal_m[0] = (fMult(gammaVecReal_m[0], hintReal_F[sourceband % 4][1]) - fMult(gammaVecImag_m[0], hintReal_F[(sourceband + 3) % 4][1])) >> 1; /* sum should be <= 1 because of sin/cos multiplication */ gammaVecImag_m[0] = (fMult(tmpRealF, hintReal_F[(sourceband + 3) % 4][1]) + fMult(gammaVecImag_m[0], hintReal_F[sourceband % 4][1])) >> 1; /* sum should be <= 1 because of sin/cos multiplication */ tmpRealF = hQmfTransposer ->qmfInBufReal_F[pSlotStretch[k] + 1][sourceband]; tmpImagF = hQmfTransposer ->qmfInBufImag_F[pSlotStretch[k] + 1][sourceband]; gammaVecReal_m[0] += (fMult(tmpRealF, hintReal_F[sourceband % 4][1]) - fMult(tmpImagF, hintReal_F[(sourceband + 1) % 4][1])) >> 1; /* sum should be <= 1 because of sin/cos multiplication */ gammaVecImag_m[0] += (fMult(tmpRealF, hintReal_F[(sourceband + 1) % 4][1]) + fMult(tmpImagF, hintReal_F[sourceband % 4][1])) >> 1; /* sum should be <= 1 because of sin/cos multiplication */ gammaVec_e[0]++; tmpRealF = gammaVecReal_m[1]; gammaVecReal_m[1] = (fMult(gammaVecReal_m[1], hintReal_F[sourceband % 4][2]) - fMult(gammaVecImag_m[1], hintReal_F[(sourceband + 3) % 4][2])) >> 1; gammaVecImag_m[1] = (fMult(tmpRealF, hintReal_F[(sourceband + 3) % 4][2]) + fMult(gammaVecImag_m[1], hintReal_F[sourceband % 4][2])) >> 1; tmpRealF = hQmfTransposer ->qmfInBufReal_F[pSlotStretch[k] + 1][sourceband + 1]; tmpImagF = hQmfTransposer ->qmfInBufImag_F[pSlotStretch[k] + 1][sourceband + 1]; gammaVecReal_m[1] += (fMult(tmpRealF, hintReal_F[sourceband % 4][2]) - fMult(tmpImagF, hintReal_F[(sourceband + 1) % 4][2])) >> 1; gammaVecImag_m[1] += (fMult(tmpRealF, hintReal_F[(sourceband + 1) % 4][2]) + fMult(tmpImagF, hintReal_F[sourceband % 4][2])) >> 1; gammaVec_e[1]++; } addHighBandPart(gammaVecReal_m[1], gammaVecImag_m[1], gammaVec_e[1], factor, gammaCenterReal_m[0], gammaCenterImag_m[0], gammaCenter_e[0], stretch, scale_factor_hbe, &hQmfTransposer->qmfHBEBufReal_F[k][band], &hQmfTransposer->qmfHBEBufImag_F[k][band]); addHighBandPart(gammaVecReal_m[0], gammaVecImag_m[0], gammaVec_e[0], factor, gammaCenterReal_m[1], gammaCenterImag_m[1], gammaCenter_e[1], stretch, scale_factor_hbe, &hQmfTransposer->qmfHBEBufReal_F[k][band], &hQmfTransposer->qmfHBEBufImag_F[k][band]); } } else { for (k = start; k < stop; k++) { gammaVecReal_m[0] = hQmfTransposer->qmfInBufReal_F[pSlotStretch[k]][sourceband]; gammaVecImag_m[0] = hQmfTransposer->qmfInBufImag_F[pSlotStretch[k]][sourceband]; gammaVec_e[0] = SCALE2EXP(-hQmfTransposer->HBEAnalysiscQMF.outScalefactor); if (pFilt[k] == 1) { FIXP_DBL tmpRealF = gammaVecReal_m[0], tmpImagF; gammaVecReal_m[0] = (fMult(gammaVecReal_m[0], hintReal_F[sourceband % 4][1]) - fMult(gammaVecImag_m[0], hintReal_F[(sourceband + 3) % 4][1])) >> 1; /* sum should be <= 1 because of sin/cos multiplication */ gammaVecImag_m[0] = (fMult(tmpRealF, hintReal_F[(sourceband + 3) % 4][1]) + fMult(gammaVecImag_m[0], hintReal_F[sourceband % 4][1])) >> 1; /* sum should be <= 1 because of sin/cos multiplication */ tmpRealF = hQmfTransposer ->qmfInBufReal_F[pSlotStretch[k] + 1][sourceband]; tmpImagF = hQmfTransposer ->qmfInBufImag_F[pSlotStretch[k] + 1][sourceband]; gammaVecReal_m[0] += (fMult(tmpRealF, hintReal_F[sourceband % 4][1]) - fMult(tmpImagF, hintReal_F[(sourceband + 1) % 4][1])) >> 1; /* sum should be <= 1 because of sin/cos multiplication */ gammaVecImag_m[0] += (fMult(tmpRealF, hintReal_F[(sourceband + 1) % 4][1]) + fMult(tmpImagF, hintReal_F[sourceband % 4][1])) >> 1; /* sum should be <= 1 because of sin/cos multiplication */ gammaVec_e[0]++; } addHighBandPart(gammaVecReal_m[0], gammaVecImag_m[0], gammaVec_e[0], factor, gammaCenterReal_m[0], gammaCenterImag_m[0], gammaCenter_e[0], stretch, scale_factor_hbe, &hQmfTransposer->qmfHBEBufReal_F[k][band], &hQmfTransposer->qmfHBEBufImag_F[k][band]); } } /* pitchInBins is given with the resolution of a 768 bins FFT and we * need 64 QMF units so factor 768/64 = 12 */ if (pitchInBins >= pmin * (1 + bSbr41)) { int tr, ti1, ti2, mTr = 0, ts1 = 0, ts2 = 0, mVal_e = 0, temp_e = 0; int sqmag0_e = SCALE2EXP(-hQmfTransposer->HBEAnalysiscQMF.outScalefactor); FIXP_DBL mVal_F = FL2FXCONST_DBL(0.f), sqmag0_F, sqmag1_F, sqmag2_F, temp_F, f1_F; /* all equal exponent */ sign = -1; sourceband = 2 * band / stretch - qmfOffset; /* consistent with the already computed for stretch = 3,4. */ FDK_ASSERT(sourceband >= 0); FIXP_DBL sqmag0R_F = hQmfTransposer->qmfInBufReal_F[slotOffset][sourceband]; FIXP_DBL sqmag0I_F = hQmfTransposer->qmfInBufImag_F[slotOffset][sourceband]; scaleUp(&sqmag0R_F, &sqmag0I_F, &sqmag0_e); sqmag0_F = fPow2Div2(sqmag0R_F); sqmag0_F += fPow2Div2(sqmag0I_F); sqmag0_e = 2 * sqmag0_e + 1; for (tr = 1; tr < stretch; tr++) { int sqmag1_e = SCALE2EXP(-hQmfTransposer->HBEAnalysiscQMF.outScalefactor); int sqmag2_e = SCALE2EXP(-hQmfTransposer->HBEAnalysiscQMF.outScalefactor); FIXP_DBL tmp_band = band_F[band]; FIXP_DBL tr_p = fMult(p_F[pitchInBins] >> bSbr41, tr_str[tr - 1]); /* scale 7 */ f1_F = fMult(tmp_band - tr_p, stretchfac[stretch - 2]); /* scale 7 */ ti1 = (INT)(f1_F >> (DFRACT_BITS - 1 - 7)) - qmfOffset; ti2 = (INT)(((f1_F) + ((p_F[pitchInBins] >> bSbr41) >> 2)) >> (DFRACT_BITS - 1 - 7)) - qmfOffset; if (ti1 >= 0 && ti2 < 2 * hQmfTransposer->synthSize) { FIXP_DBL sqmag1R_F = hQmfTransposer->qmfInBufReal_F[slotOffset][ti1]; FIXP_DBL sqmag1I_F = hQmfTransposer->qmfInBufImag_F[slotOffset][ti1]; scaleUp(&sqmag1R_F, &sqmag1I_F, &sqmag1_e); sqmag1_F = fPow2Div2(sqmag1R_F); sqmag1_F += fPow2Div2(sqmag1I_F); sqmag1_e = 2 * sqmag1_e + 1; FIXP_DBL sqmag2R_F = hQmfTransposer->qmfInBufReal_F[slotOffset][ti2]; FIXP_DBL sqmag2I_F = hQmfTransposer->qmfInBufImag_F[slotOffset][ti2]; scaleUp(&sqmag2R_F, &sqmag2I_F, &sqmag2_e); sqmag2_F = fPow2Div2(sqmag2R_F); sqmag2_F += fPow2Div2(sqmag2I_F); sqmag2_e = 2 * sqmag2_e + 1; int shift1 = fMin(fMax(sqmag1_e, sqmag2_e) - sqmag1_e, 31); int shift2 = fMin(fMax(sqmag1_e, sqmag2_e) - sqmag2_e, 31); temp_F = fMin((sqmag1_F >> shift1), (sqmag2_F >> shift2)); temp_e = fMax(sqmag1_e, sqmag2_e); int shift3 = fMin(fMax(temp_e, mVal_e) - temp_e, 31); int shift4 = fMin(fMax(temp_e, mVal_e) - mVal_e, 31); if ((temp_F >> shift3) > (mVal_F >> shift4)) { mVal_F = temp_F; mVal_e = temp_e; /* equals sqmag2_e + shift2 */ mTr = tr; ts1 = ti1; ts2 = ti2; } } } int shift1 = fMin(fMax(sqmag0_e, mVal_e) - sqmag0_e, 31); int shift2 = fMin(fMax(sqmag0_e, mVal_e) - mVal_e, 31); if ((mVal_F >> shift2) > (sqmag0_F >> shift1) && ts1 >= 0 && ts2 < 2 * hQmfTransposer->synthSize) { INT gammaOut_e[2]; FIXP_DBL gammaOutReal_m[2], gammaOutImag_m[2]; FIXP_DBL tmpReal_m = (FIXP_DBL)0, tmpImag_m = (FIXP_DBL)0; int Tcenter, Tvec; Tcenter = stretch - mTr; /* default phase power parameters */ Tvec = mTr; switch (stretch) /* 2 tap block creation design depends on stretch order */ { case 2: wingain = FL2FXCONST_DBL(5.f / 12.f); /* sum of taps divided by two */ if (hQmfTransposer->bXProducts[0]) { gammaCenterReal_m[0] = hQmfTransposer->qmfInBufReal_F[slotOffset][ts1]; gammaCenterImag_m[0] = hQmfTransposer->qmfInBufImag_F[slotOffset][ts1]; for (k = 0; k < 2; k++) { gammaVecReal_m[k] = hQmfTransposer->qmfInBufReal_F[slotOffset - 1 + k][ts2]; gammaVecImag_m[k] = hQmfTransposer->qmfInBufImag_F[slotOffset - 1 + k][ts2]; } gammaCenter_e[0] = SCALE2EXP( -hQmfTransposer->HBEAnalysiscQMF.outScalefactor); gammaVec_e[0] = gammaVec_e[1] = SCALE2EXP( -hQmfTransposer->HBEAnalysiscQMF.outScalefactor); } break; case 4: wingain = FL2FXCONST_DBL(6.f / 12.f); /* sum of taps divided by two */ if (hQmfTransposer->bXProducts[2]) { if (mTr == 1) { gammaCenterReal_m[0] = hQmfTransposer->qmfInBufReal_F[slotOffset][ts1]; gammaCenterImag_m[0] = hQmfTransposer->qmfInBufImag_F[slotOffset][ts1]; for (k = 0; k < 2; k++) { gammaVecReal_m[k] = hQmfTransposer ->qmfInBufReal_F[slotOffset + 2 * (k - 1)][ts2]; gammaVecImag_m[k] = hQmfTransposer ->qmfInBufImag_F[slotOffset + 2 * (k - 1)][ts2]; } } else if (mTr == 2) { gammaCenterReal_m[0] = hQmfTransposer->qmfInBufReal_F[slotOffset][ts1]; gammaCenterImag_m[0] = hQmfTransposer->qmfInBufImag_F[slotOffset][ts1]; for (k = 0; k < 2; k++) { gammaVecReal_m[k] = hQmfTransposer ->qmfInBufReal_F[slotOffset + (k - 1)][ts2]; gammaVecImag_m[k] = hQmfTransposer ->qmfInBufImag_F[slotOffset + (k - 1)][ts2]; } } else /* (mTr == 3) */ { sign = 1; Tcenter = mTr; /* opposite phase power parameters as ts2 is center */ Tvec = stretch - mTr; gammaCenterReal_m[0] = hQmfTransposer->qmfInBufReal_F[slotOffset][ts2]; gammaCenterImag_m[0] = hQmfTransposer->qmfInBufImag_F[slotOffset][ts2]; for (k = 0; k < 2; k++) { gammaVecReal_m[k] = hQmfTransposer ->qmfInBufReal_F[slotOffset + 2 * (k - 1)][ts1]; gammaVecImag_m[k] = hQmfTransposer ->qmfInBufImag_F[slotOffset + 2 * (k - 1)][ts1]; } } gammaCenter_e[0] = SCALE2EXP( -hQmfTransposer->HBEAnalysiscQMF.outScalefactor); gammaVec_e[0] = gammaVec_e[1] = SCALE2EXP( -hQmfTransposer->HBEAnalysiscQMF.outScalefactor); } break; case 3: wingain = FL2FXCONST_DBL(5.6568f / 12.f); /* sum of taps divided by two */ if (hQmfTransposer->bXProducts[1]) { FIXP_DBL tmpReal_F, tmpImag_F; if (mTr == 1) { gammaCenterReal_m[0] = hQmfTransposer->qmfInBufReal_F[slotOffset][ts1]; gammaCenterImag_m[0] = hQmfTransposer->qmfInBufImag_F[slotOffset][ts1]; gammaVecReal_m[1] = hQmfTransposer->qmfInBufReal_F[slotOffset][ts2]; gammaVecImag_m[1] = hQmfTransposer->qmfInBufImag_F[slotOffset][ts2]; addrshift = -2; tmpReal_F = hQmfTransposer ->qmfInBufReal_F[addrshift + slotOffset][ts2]; tmpImag_F = hQmfTransposer ->qmfInBufImag_F[addrshift + slotOffset][ts2]; gammaVecReal_m[0] = (fMult(factors[ts2 % 4], tmpReal_F) - fMult(factors[(ts2 + 3) % 4], tmpImag_F)) >> 1; gammaVecImag_m[0] = (fMult(factors[(ts2 + 3) % 4], tmpReal_F) + fMult(factors[ts2 % 4], tmpImag_F)) >> 1; tmpReal_F = hQmfTransposer ->qmfInBufReal_F[addrshift + 1 + slotOffset][ts2]; tmpImag_F = hQmfTransposer ->qmfInBufImag_F[addrshift + 1 + slotOffset][ts2]; gammaVecReal_m[0] += (fMult(factors[ts2 % 4], tmpReal_F) - fMult(factors[(ts2 + 1) % 4], tmpImag_F)) >> 1; gammaVecImag_m[0] += (fMult(factors[(ts2 + 1) % 4], tmpReal_F) + fMult(factors[ts2 % 4], tmpImag_F)) >> 1; } else /* (mTr == 2) */ { sign = 1; Tcenter = mTr; /* opposite phase power parameters as ts2 is center */ Tvec = stretch - mTr; gammaCenterReal_m[0] = hQmfTransposer->qmfInBufReal_F[slotOffset][ts2]; gammaCenterImag_m[0] = hQmfTransposer->qmfInBufImag_F[slotOffset][ts2]; gammaVecReal_m[1] = hQmfTransposer->qmfInBufReal_F[slotOffset][ts1]; gammaVecImag_m[1] = hQmfTransposer->qmfInBufImag_F[slotOffset][ts1]; addrshift = -2; tmpReal_F = hQmfTransposer ->qmfInBufReal_F[addrshift + slotOffset][ts1]; tmpImag_F = hQmfTransposer ->qmfInBufImag_F[addrshift + slotOffset][ts1]; gammaVecReal_m[0] = (fMult(factors[ts1 % 4], tmpReal_F) - fMult(factors[(ts1 + 3) % 4], tmpImag_F)) >> 1; gammaVecImag_m[0] = (fMult(factors[(ts1 + 3) % 4], tmpReal_F) + fMult(factors[ts1 % 4], tmpImag_F)) >> 1; tmpReal_F = hQmfTransposer ->qmfInBufReal_F[addrshift + 1 + slotOffset][ts1]; tmpImag_F = hQmfTransposer ->qmfInBufImag_F[addrshift + 1 + slotOffset][ts1]; gammaVecReal_m[0] += (fMult(factors[ts1 % 4], tmpReal_F) - fMult(factors[(ts1 + 1) % 4], tmpImag_F)) >> 1; gammaVecImag_m[0] += (fMult(factors[(ts1 + 1) % 4], tmpReal_F) + fMult(factors[ts1 % 4], tmpImag_F)) >> 1; } gammaCenter_e[0] = gammaVec_e[1] = SCALE2EXP( -hQmfTransposer->HBEAnalysiscQMF.outScalefactor); gammaVec_e[0] = SCALE2EXP( -hQmfTransposer->HBEAnalysiscQMF.outScalefactor) + 1; } break; default: FDK_ASSERT(0); break; } /* stretch cases */ /* parameter controlled phase modification parts */ /* maximum *_e == 20 */ calculateCenterFIXP(gammaCenterReal_m[0], gammaCenterImag_m[0], &gammaCenterReal_m[0], &gammaCenterImag_m[0], &gammaCenter_e[0], stretch, Tcenter - 1); calculateCenterFIXP(gammaVecReal_m[0], gammaVecImag_m[0], &gammaVecReal_m[0], &gammaVecImag_m[0], &gammaVec_e[0], stretch, Tvec - 1); calculateCenterFIXP(gammaVecReal_m[1], gammaVecImag_m[1], &gammaVecReal_m[1], &gammaVecImag_m[1], &gammaVec_e[1], stretch, Tvec - 1); /* Final multiplication of prepared parts */ for (k = 0; k < 2; k++) { gammaOutReal_m[k] = fMultDiv2(gammaVecReal_m[k], gammaCenterReal_m[0]) - fMultDiv2(gammaVecImag_m[k], gammaCenterImag_m[0]); gammaOutImag_m[k] = fMultDiv2(gammaVecReal_m[k], gammaCenterImag_m[0]) + fMultDiv2(gammaVecImag_m[k], gammaCenterReal_m[0]); gammaOut_e[k] = gammaCenter_e[0] + gammaVec_e[k] + 1; } scaleUp(&gammaOutReal_m[0], &gammaOutImag_m[0], &gammaOut_e[0]); scaleUp(&gammaOutReal_m[1], &gammaOutImag_m[1], &gammaOut_e[1]); FDK_ASSERT(gammaOut_e[0] >= 0); FDK_ASSERT(gammaOut_e[0] < 32); tmpReal_m = gammaOutReal_m[0]; tmpImag_m = gammaOutImag_m[0]; INT modstretch4 = ((stretch == 4) && (mTr == 2)); FIXP_DBL cos_twid = twid_m_new[stretch - 2 - modstretch4][0]; FIXP_DBL sin_twid = sign * twid_m_new[stretch - 2 - modstretch4][1]; gammaOutReal_m[0] = fMult(tmpReal_m, cos_twid) - fMult(tmpImag_m, sin_twid); /* sum should be <= 1 because of sin/cos multiplication */ gammaOutImag_m[0] = fMult(tmpImag_m, cos_twid) + fMult(tmpReal_m, sin_twid); /* sum should be <= 1 because of sin/cos multiplication */ /* wingain */ for (k = 0; k < 2; k++) { gammaOutReal_m[k] = (fMult(gammaOutReal_m[k], wingain) << 1); gammaOutImag_m[k] = (fMult(gammaOutImag_m[k], wingain) << 1); } gammaOutReal_m[1] >>= 1; gammaOutImag_m[1] >>= 1; gammaOut_e[0] += 2; gammaOut_e[1] += 2; /* OLA including window scaling by wingain/3 */ for (k = 0; k < 2; k++) /* need k=1 to correspond to grainModImag[slotOffset] -> out to j*2+(slotOffset-offset) */ { hQmfTransposer->qmfHBEBufReal_F[(k + slotOffset - 1)][band] += gammaOutReal_m[k] >> (scale_factor_hbe - gammaOut_e[k]); hQmfTransposer->qmfHBEBufImag_F[(k + slotOffset - 1)][band] += gammaOutImag_m[k] >> (scale_factor_hbe - gammaOut_e[k]); } } /* mVal > qThrQMF * qThrQMF * sqmag0 && ts1 > 0 && ts2 < 64 */ } /* p >= pmin */ } /* for band */ } /* for stretch */ for (i = 0; i < QMF_WIN_LEN - 1; i++) { FDKmemcpy(hQmfTransposer->qmfInBufReal_F[i], hQmfTransposer->qmfInBufReal_F[i + 1], sizeof(FIXP_DBL) * hQmfTransposer->HBEAnalysiscQMF.no_channels); FDKmemcpy(hQmfTransposer->qmfInBufImag_F[i], hQmfTransposer->qmfInBufImag_F[i + 1], sizeof(FIXP_DBL) * hQmfTransposer->HBEAnalysiscQMF.no_channels); } if (keepStatesSyncedMode != KEEP_STATES_SYNCED_NOOUT) { if (2 * j >= offset) { /* copy first two slots of internal buffer to output */ if (keepStatesSyncedMode == KEEP_STATES_SYNCED_OUTDIFF) { for (i = 0; i < 2; i++) { FDKmemcpy(&ppQmfBufferOutReal_F[2 * j - offset + i] [hQmfTransposer->xOverQmf[0]], &hQmfTransposer ->qmfHBEBufReal_F[i][hQmfTransposer->xOverQmf[0]], (QMF_SYNTH_CHANNELS - hQmfTransposer->xOverQmf[0]) * sizeof(FIXP_DBL)); FDKmemcpy(&ppQmfBufferOutImag_F[2 * j - offset + i] [hQmfTransposer->xOverQmf[0]], &hQmfTransposer ->qmfHBEBufImag_F[i][hQmfTransposer->xOverQmf[0]], (QMF_SYNTH_CHANNELS - hQmfTransposer->xOverQmf[0]) * sizeof(FIXP_DBL)); } } else { for (i = 0; i < 2; i++) { FDKmemcpy(&ppQmfBufferOutReal_F[2 * j + i + ov_len] [hQmfTransposer->xOverQmf[0]], &hQmfTransposer ->qmfHBEBufReal_F[i][hQmfTransposer->xOverQmf[0]], (QMF_SYNTH_CHANNELS - hQmfTransposer->xOverQmf[0]) * sizeof(FIXP_DBL)); FDKmemcpy(&ppQmfBufferOutImag_F[2 * j + i + ov_len] [hQmfTransposer->xOverQmf[0]], &hQmfTransposer ->qmfHBEBufImag_F[i][hQmfTransposer->xOverQmf[0]], (QMF_SYNTH_CHANNELS - hQmfTransposer->xOverQmf[0]) * sizeof(FIXP_DBL)); } } } } /* move slots up */ for (i = 0; i < HBE_MAX_OUT_SLOTS - 2; i++) { FDKmemcpy( &hQmfTransposer->qmfHBEBufReal_F[i][hQmfTransposer->xOverQmf[0]], &hQmfTransposer->qmfHBEBufReal_F[i + 2][hQmfTransposer->xOverQmf[0]], (QMF_SYNTH_CHANNELS - hQmfTransposer->xOverQmf[0]) * sizeof(FIXP_DBL)); FDKmemcpy( &hQmfTransposer->qmfHBEBufImag_F[i][hQmfTransposer->xOverQmf[0]], &hQmfTransposer->qmfHBEBufImag_F[i + 2][hQmfTransposer->xOverQmf[0]], (QMF_SYNTH_CHANNELS - hQmfTransposer->xOverQmf[0]) * sizeof(FIXP_DBL)); } /* finally set last two slot to zero */ for (i = 0; i < 2; i++) { FDKmemset(&hQmfTransposer->qmfHBEBufReal_F[HBE_MAX_OUT_SLOTS - 1 - i] [hQmfTransposer->xOverQmf[0]], 0, (QMF_SYNTH_CHANNELS - hQmfTransposer->xOverQmf[0]) * sizeof(FIXP_DBL)); FDKmemset(&hQmfTransposer->qmfHBEBufImag_F[HBE_MAX_OUT_SLOTS - 1 - i] [hQmfTransposer->xOverQmf[0]], 0, (QMF_SYNTH_CHANNELS - hQmfTransposer->xOverQmf[0]) * sizeof(FIXP_DBL)); } } /* qmfVocoderColsIn */ if (keepStatesSyncedMode != KEEP_STATES_SYNCED_NOOUT) { if (keepStatesSyncedMode == KEEP_STATES_SYNCED_OUTDIFF) { for (i = 0; i < ov_len + LPC_ORDER; i++) { for (band = hQmfTransposer->startBand; band < hQmfTransposer->stopBand; band++) { FIXP_DBL tmpR = ppQmfBufferOutReal_F[i][band]; FIXP_DBL tmpI = ppQmfBufferOutImag_F[i][band]; ppQmfBufferOutReal_F[i][band] = fMult(tmpR, cos_F[band]) - fMult(tmpI, (-cos_F[64 - band - 1])); /* sum should be <= 1 because of sin/cos multiplication */ ppQmfBufferOutImag_F[i][band] = fMult(tmpR, (-cos_F[64 - band - 1])) + fMult(tmpI, cos_F[band]); /* sum should by <= 1 because of sin/cos multiplication */ } } } else { for (i = offset; i < hQmfTransposer->noCols; i++) { for (band = hQmfTransposer->startBand; band < hQmfTransposer->stopBand; band++) { FIXP_DBL tmpR = ppQmfBufferOutReal_F[i + ov_len][band]; FIXP_DBL tmpI = ppQmfBufferOutImag_F[i + ov_len][band]; ppQmfBufferOutReal_F[i + ov_len][band] = fMult(tmpR, cos_F[band]) - fMult(tmpI, (-cos_F[64 - band - 1])); /* sum should be <= 1 because of sin/cos multiplication */ ppQmfBufferOutImag_F[i + ov_len][band] = fMult(tmpR, (-cos_F[64 - band - 1])) + fMult(tmpI, cos_F[band]); /* sum should by <= 1 because of sin/cos multiplication */ } } } } *scale_hb = EXP2SCALE(scale_factor_hbe); } int* GetxOverBandQmfTransposer(HANDLE_HBE_TRANSPOSER hQmfTransposer) { if (hQmfTransposer) return hQmfTransposer->xOverQmf; else return NULL; } int Get41SbrQmfTransposer(HANDLE_HBE_TRANSPOSER hQmfTransposer) { if (hQmfTransposer != NULL) return hQmfTransposer->bSbr41; else return 0; } fdk-aac-2.0.2/libSBRdec/src/hbe.h000066400000000000000000000174061404225310700162760ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Description: *******************************************************************************/ #ifndef HBE_H #define HBE_H #include "sbrdecoder.h" #ifndef QMF_SYNTH_CHANNELS #define QMF_SYNTH_CHANNELS (64) #endif #define HBE_QMF_FILTER_STATE_ANA_SIZE (400) #define HBE_QMF_FILTER_STATE_SYN_SIZE (200) #ifndef MAX_NUM_PATCHES_HBE #define MAX_NUM_PATCHES_HBE (6) #endif #define MAX_STRETCH_HBE (4) typedef enum { KEEP_STATES_SYNCED_OFF = 0, /*!< normal QMF transposer behaviour */ KEEP_STATES_SYNCED_NORMAL = 1, /*!< QMF transposer called for syncing of states the last 8/14 slots are calculated in case next frame is HBE */ KEEP_STATES_SYNCED_OUTDIFF = 2, /*!< QMF transposer behaviour as in normal case, but the calculated slots are directly written to overlap area of buffer; only used in resetSbrDec function */ KEEP_STATES_SYNCED_NOOUT = 3 /*!< QMF transposer is called for syncing of states only, not output is generated at all; only used in resetSbrDec function */ } KEEP_STATES_SYNCED_MODE; struct hbeTransposer { FIXP_DBL anaQmfStates[HBE_QMF_FILTER_STATE_ANA_SIZE]; FIXP_QSS synQmfStates[HBE_QMF_FILTER_STATE_SYN_SIZE]; int xOverQmf[MAX_NUM_PATCHES_HBE]; int maxStretch; int timeDomainWinLen; int qmfInBufSize; int qmfOutBufSize; int noCols; int noChannels; int startBand; int stopBand; int bSbr41; LONG *inBuf_F; FIXP_DBL **qmfInBufReal_F; FIXP_DBL **qmfInBufImag_F; FIXP_DBL *qmfBufferCodecTempSlot_F; QMF_FILTER_BANK HBEAnalysiscQMF; QMF_FILTER_BANK HBESynthesisQMF; FIXP_DBL const *synthesisQmfPreModCos_F; FIXP_DBL const *synthesisQmfPreModSin_F; FIXP_DBL **qmfHBEBufReal_F; FIXP_DBL **qmfHBEBufImag_F; int bXProducts[MAX_STRETCH_HBE]; int kstart; int synthSize; int highband_exp[2]; int target_exp[2]; }; typedef struct hbeTransposer *HANDLE_HBE_TRANSPOSER; SBR_ERROR QmfTransposerCreate(HANDLE_HBE_TRANSPOSER *hQmfTransposer, const int frameSize, int bDisableCrossProducts, int bSbr41); SBR_ERROR QmfTransposerReInit(HANDLE_HBE_TRANSPOSER hQmfTransposer, UCHAR *FreqBandTable[2], UCHAR NSfb[2]); void QmfTransposerClose(HANDLE_HBE_TRANSPOSER hQmfTransposer); void QmfTransposerApply(HANDLE_HBE_TRANSPOSER hQmfTransposer, FIXP_DBL **qmfBufferCodecReal, FIXP_DBL **qmfBufferCodecImag, int nColsIn, FIXP_DBL **ppQmfBufferOutReal_F, FIXP_DBL **ppQmfBufferOutImag_F, FIXP_DBL lpcFilterStatesReal[2 + (3 * (4))][(64)], FIXP_DBL lpcFilterStatesImag[2 + (3 * (4))][(64)], int pitchInBins, int scale_lb, int scale_hbe, int *scale_hb, int timeStep, int firstSlotOffsset, int ov_len, KEEP_STATES_SYNCED_MODE keepStatesSyncedMode); int *GetxOverBandQmfTransposer(HANDLE_HBE_TRANSPOSER hQmfTransposer); int Get41SbrQmfTransposer(HANDLE_HBE_TRANSPOSER hQmfTransposer); #endif /* HBE_H */ fdk-aac-2.0.2/libSBRdec/src/huff_dec.cpp000066400000000000000000000127621404225310700176360ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Huffman Decoder */ #include "huff_dec.h" /***************************************************************************/ /*! \brief Decodes one huffman code word Reads bits from the bitstream until a valid codeword is found. The table entries are interpreted either as index to the next entry or - if negative - as the codeword. \return decoded value \author ****************************************************************************/ int DecodeHuffmanCW(Huffman h, /*!< pointer to huffman codebook table */ HANDLE_FDK_BITSTREAM hBs) /*!< Handle to Bitbuffer */ { SCHAR index = 0; int value, bit; while (index >= 0) { bit = FDKreadBits(hBs, 1); index = h[index][bit]; } value = index + 64; /* Add offset */ return value; } fdk-aac-2.0.2/libSBRdec/src/huff_dec.h000066400000000000000000000116631404225310700173020ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Huffman Decoder */ #ifndef HUFF_DEC_H #define HUFF_DEC_H #include "sbrdecoder.h" #include "FDK_bitstream.h" typedef const SCHAR (*Huffman)[2]; int DecodeHuffmanCW(Huffman h, HANDLE_FDK_BITSTREAM hBitBuf); #endif fdk-aac-2.0.2/libSBRdec/src/lpp_tran.cpp000066400000000000000000001463731404225310700177200ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Low Power Profile Transposer This module provides the transposer. The main entry point is lppTransposer(). The function generates high frequency content by copying data from the low band (provided by core codec) into the high band. This process is also referred to as "patching". The function also implements spectral whitening by means of inverse filtering based on LPC coefficients. Together with the QMF filterbank the transposer can be tested using a supplied test program. See main_audio.cpp for details. This module does use fractional arithmetic and the accuracy of the computations has an impact on the overall sound quality. The module also needs to take into account the different scaling of spectral data. \sa lppTransposer(), main_audio.cpp, sbr_scale.h, \ref documentationOverview */ #ifdef __ANDROID__ #include "log/log.h" #endif #include "lpp_tran.h" #include "sbr_ram.h" #include "sbr_rom.h" #include "genericStds.h" #include "autocorr2nd.h" #include "HFgen_preFlat.h" #if defined(__arm__) #include "arm/lpp_tran_arm.cpp" #endif #define LPC_SCALE_FACTOR 2 /*! * * \brief Get bandwidth expansion factor from filtering level * * Returns a filter parameter (bandwidth expansion factor) depending on * the desired filtering level signalled in the bitstream. * When switching the filtering level from LOW to OFF, an additional * level is being inserted to achieve a smooth transition. */ static FIXP_DBL mapInvfMode(INVF_MODE mode, INVF_MODE prevMode, WHITENING_FACTORS whFactors) { switch (mode) { case INVF_LOW_LEVEL: if (prevMode == INVF_OFF) return whFactors.transitionLevel; else return whFactors.lowLevel; case INVF_MID_LEVEL: return whFactors.midLevel; case INVF_HIGH_LEVEL: return whFactors.highLevel; default: if (prevMode == INVF_LOW_LEVEL) return whFactors.transitionLevel; else return whFactors.off; } } /*! * * \brief Perform inverse filtering level emphasis * * Retrieve bandwidth expansion factor and apply smoothing for each filter band * */ static void inverseFilteringLevelEmphasis( HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transposer */ UCHAR nInvfBands, /*!< Number of bands for inverse filtering */ INVF_MODE *sbr_invf_mode, /*!< Current inverse filtering modes */ INVF_MODE *sbr_invf_mode_prev, /*!< Previous inverse filtering modes */ FIXP_DBL *bwVector /*!< Resulting filtering levels */ ) { for (int i = 0; i < nInvfBands; i++) { FIXP_DBL accu; FIXP_DBL bwTmp = mapInvfMode(sbr_invf_mode[i], sbr_invf_mode_prev[i], hLppTrans->pSettings->whFactors); if (bwTmp < hLppTrans->bwVectorOld[i]) { accu = fMultDiv2(FL2FXCONST_DBL(0.75f), bwTmp) + fMultDiv2(FL2FXCONST_DBL(0.25f), hLppTrans->bwVectorOld[i]); } else { accu = fMultDiv2(FL2FXCONST_DBL(0.90625f), bwTmp) + fMultDiv2(FL2FXCONST_DBL(0.09375f), hLppTrans->bwVectorOld[i]); } if (accu> 1) { bwVector[i] = FL2FXCONST_DBL(0.0f); } else { bwVector[i] = fixMin(accu << 1, FL2FXCONST_DBL(0.99609375f)); } } } /* Resulting autocorrelation determinant exponent */ #define ACDET_EXP \ (2 * (DFRACT_BITS + sbrScaleFactor->lb_scale + 10 - ac.det_scale)) #define AC_EXP (-sbrScaleFactor->lb_scale + LPC_SCALE_FACTOR) #define ALPHA_EXP (-sbrScaleFactor->lb_scale + LPC_SCALE_FACTOR + 1) /* Resulting transposed QMF values exponent 16 bit normalized samplebits * assumed. */ #define QMFOUT_EXP ((SAMPLE_BITS - 15) - sbrScaleFactor->lb_scale) static inline void calc_qmfBufferReal(FIXP_DBL **qmfBufferReal, const FIXP_DBL *const lowBandReal, const int startSample, const int stopSample, const UCHAR hiBand, const int dynamicScale, const int descale, const FIXP_SGL a0r, const FIXP_SGL a1r) { FIXP_DBL accu1, accu2; int i; for (i = 0; i < stopSample - startSample; i++) { accu1 = fMultDiv2(a1r, lowBandReal[i]); accu1 = (fMultDiv2(a0r, lowBandReal[i + 1]) + accu1); accu1 = accu1 >> dynamicScale; accu1 <<= 1; accu2 = (lowBandReal[i + 2] >> descale); qmfBufferReal[i + startSample][hiBand] = accu1 + accu2; } } /*! * * \brief Perform transposition by patching of subband samples. * This function serves as the main entry point into the module. The function * determines the areas for the patching process (these are the source range as * well as the target range) and implements spectral whitening by means of * inverse filtering. The function autoCorrelation2nd() is an auxiliary function * for calculating the LPC coefficients for the filtering. The actual * calculation of the LPC coefficients and the implementation of the filtering * are done as part of lppTransposer(). * * Note that the filtering is done on all available QMF subsamples, whereas the * patching is only done on those QMF subsamples that will be used in the next * QMF synthesis. The filtering is also implemented before the patching includes * further dependencies on parameters from the SBR data. * */ void lppTransposer( HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transposer */ QMF_SCALE_FACTOR *sbrScaleFactor, /*!< Scaling factors */ FIXP_DBL **qmfBufferReal, /*!< Pointer to pointer to real part of subband samples (source) */ FIXP_DBL *degreeAlias, /*!< Vector for results of aliasing estimation */ FIXP_DBL **qmfBufferImag, /*!< Pointer to pointer to imaginary part of subband samples (source) */ const int useLP, const int fPreWhitening, const int v_k_master0, const int timeStep, /*!< Time step of envelope */ const int firstSlotOffs, /*!< Start position in time */ const int lastSlotOffs, /*!< Number of overlap-slots into next frame */ const int nInvfBands, /*!< Number of bands for inverse filtering */ INVF_MODE *sbr_invf_mode, /*!< Current inverse filtering modes */ INVF_MODE *sbr_invf_mode_prev /*!< Previous inverse filtering modes */ ) { INT bwIndex[MAX_NUM_PATCHES]; FIXP_DBL bwVector[MAX_NUM_PATCHES]; /*!< pole moving factors */ FIXP_DBL preWhiteningGains[(64) / 2]; int preWhiteningGains_exp[(64) / 2]; int i; int loBand, start, stop; TRANSPOSER_SETTINGS *pSettings = hLppTrans->pSettings; PATCH_PARAM *patchParam = pSettings->patchParam; int patch; FIXP_SGL alphar[LPC_ORDER], a0r, a1r; FIXP_SGL alphai[LPC_ORDER], a0i = 0, a1i = 0; FIXP_SGL bw = FL2FXCONST_SGL(0.0f); int autoCorrLength; FIXP_DBL k1, k1_below = 0, k1_below2 = 0; ACORR_COEFS ac; int startSample; int stopSample; int stopSampleClear; int comLowBandScale; int ovLowBandShift; int lowBandShift; /* int ovHighBandShift;*/ alphai[0] = FL2FXCONST_SGL(0.0f); alphai[1] = FL2FXCONST_SGL(0.0f); startSample = firstSlotOffs * timeStep; stopSample = pSettings->nCols + lastSlotOffs * timeStep; FDK_ASSERT((lastSlotOffs * timeStep) <= pSettings->overlap); inverseFilteringLevelEmphasis(hLppTrans, nInvfBands, sbr_invf_mode, sbr_invf_mode_prev, bwVector); stopSampleClear = stopSample; autoCorrLength = pSettings->nCols + pSettings->overlap; if (pSettings->noOfPatches > 0) { /* Set upper subbands to zero: This is required in case that the patches do not cover the complete highband (because the last patch would be too short). Possible optimization: Clearing bands up to usb would be sufficient here. */ int targetStopBand = patchParam[pSettings->noOfPatches - 1].targetStartBand + patchParam[pSettings->noOfPatches - 1].numBandsInPatch; int memSize = ((64) - targetStopBand) * sizeof(FIXP_DBL); if (!useLP) { for (i = startSample; i < stopSampleClear; i++) { FDKmemclear(&qmfBufferReal[i][targetStopBand], memSize); FDKmemclear(&qmfBufferImag[i][targetStopBand], memSize); } } else { for (i = startSample; i < stopSampleClear; i++) { FDKmemclear(&qmfBufferReal[i][targetStopBand], memSize); } } } #ifdef __ANDROID__ else { // Safetynet logging android_errorWriteLog(0x534e4554, "112160868"); } #endif /* init bwIndex for each patch */ FDKmemclear(bwIndex, sizeof(bwIndex)); /* Calc common low band scale factor */ comLowBandScale = fixMin(sbrScaleFactor->ov_lb_scale, sbrScaleFactor->lb_scale); ovLowBandShift = sbrScaleFactor->ov_lb_scale - comLowBandScale; lowBandShift = sbrScaleFactor->lb_scale - comLowBandScale; /* ovHighBandShift = firstSlotOffs == 0 ? ovLowBandShift:0;*/ if (fPreWhitening) { sbrDecoder_calculateGainVec( qmfBufferReal, qmfBufferImag, DFRACT_BITS - 1 - 16 - sbrScaleFactor->ov_lb_scale, /* convert scale to exponent */ DFRACT_BITS - 1 - 16 - sbrScaleFactor->lb_scale, /* convert scale to exponent */ pSettings->overlap, preWhiteningGains, preWhiteningGains_exp, v_k_master0, startSample, stopSample); } /* outer loop over bands to do analysis only once for each band */ if (!useLP) { start = pSettings->lbStartPatching; stop = pSettings->lbStopPatching; } else { start = fixMax(1, pSettings->lbStartPatching - 2); stop = patchParam[0].targetStartBand; } for (loBand = start; loBand < stop; loBand++) { FIXP_DBL lowBandReal[(((1024) / (32) * (4) / 2) + (3 * (4))) + LPC_ORDER]; FIXP_DBL *plowBandReal = lowBandReal; FIXP_DBL **pqmfBufferReal = qmfBufferReal + firstSlotOffs * timeStep /* + pSettings->overlap */; FIXP_DBL lowBandImag[(((1024) / (32) * (4) / 2) + (3 * (4))) + LPC_ORDER]; FIXP_DBL *plowBandImag = lowBandImag; FIXP_DBL **pqmfBufferImag = qmfBufferImag + firstSlotOffs * timeStep /* + pSettings->overlap */; int resetLPCCoeffs = 0; int dynamicScale = DFRACT_BITS - 1 - LPC_SCALE_FACTOR; int acDetScale = 0; /* scaling of autocorrelation determinant */ for (i = 0; i < LPC_ORDER + firstSlotOffs * timeStep /*+pSettings->overlap*/; i++) { *plowBandReal++ = hLppTrans->lpcFilterStatesRealLegSBR[i][loBand]; if (!useLP) *plowBandImag++ = hLppTrans->lpcFilterStatesImagLegSBR[i][loBand]; } /* Take old slope length qmf slot source values out of (overlap)qmf buffer */ if (!useLP) { for (i = 0; i < pSettings->nCols + pSettings->overlap - firstSlotOffs * timeStep; i++) { *plowBandReal++ = (*pqmfBufferReal++)[loBand]; *plowBandImag++ = (*pqmfBufferImag++)[loBand]; } } else { /* pSettings->overlap is always even */ FDK_ASSERT((pSettings->overlap & 1) == 0); for (i = 0; i < ((pSettings->nCols + pSettings->overlap - firstSlotOffs * timeStep) >> 1); i++) { *plowBandReal++ = (*pqmfBufferReal++)[loBand]; *plowBandReal++ = (*pqmfBufferReal++)[loBand]; } if (pSettings->nCols & 1) { *plowBandReal++ = (*pqmfBufferReal++)[loBand]; } } /* Determine dynamic scaling value. */ dynamicScale = fixMin(dynamicScale, getScalefactor(lowBandReal, LPC_ORDER + pSettings->overlap) + ovLowBandShift); dynamicScale = fixMin(dynamicScale, getScalefactor(&lowBandReal[LPC_ORDER + pSettings->overlap], pSettings->nCols) + lowBandShift); if (!useLP) { dynamicScale = fixMin(dynamicScale, getScalefactor(lowBandImag, LPC_ORDER + pSettings->overlap) + ovLowBandShift); dynamicScale = fixMin(dynamicScale, getScalefactor(&lowBandImag[LPC_ORDER + pSettings->overlap], pSettings->nCols) + lowBandShift); } if (dynamicScale == 0) { /* In this special case the available headroom bits as well as ovLowBandShift and lowBandShift are zero. The spectrum is limited to prevent -1.0, so negative values for dynamicScale can be avoided. */ for (i = 0; i < (LPC_ORDER + pSettings->overlap + pSettings->nCols); i++) { lowBandReal[i] = fixMax(lowBandReal[i], (FIXP_DBL)0x80000001); } if (!useLP) { for (i = 0; i < (LPC_ORDER + pSettings->overlap + pSettings->nCols); i++) { lowBandImag[i] = fixMax(lowBandImag[i], (FIXP_DBL)0x80000001); } } } else { dynamicScale = fixMax(0, dynamicScale - 1); /* one additional bit headroom to prevent -1.0 */ } /* Scale temporal QMF buffer. */ scaleValues(&lowBandReal[0], LPC_ORDER + pSettings->overlap, dynamicScale - ovLowBandShift); scaleValues(&lowBandReal[LPC_ORDER + pSettings->overlap], pSettings->nCols, dynamicScale - lowBandShift); if (!useLP) { scaleValues(&lowBandImag[0], LPC_ORDER + pSettings->overlap, dynamicScale - ovLowBandShift); scaleValues(&lowBandImag[LPC_ORDER + pSettings->overlap], pSettings->nCols, dynamicScale - lowBandShift); } if (!useLP) { acDetScale += autoCorr2nd_cplx(&ac, lowBandReal + LPC_ORDER, lowBandImag + LPC_ORDER, autoCorrLength); } else { acDetScale += autoCorr2nd_real(&ac, lowBandReal + LPC_ORDER, autoCorrLength); } /* Examine dynamic of determinant in autocorrelation. */ acDetScale += 2 * (comLowBandScale + dynamicScale); acDetScale *= 2; /* two times reflection coefficent scaling */ acDetScale += ac.det_scale; /* ac scaling of determinant */ /* In case of determinant < 10^-38, resetLPCCoeffs=1 has to be enforced. */ if (acDetScale > 126) { resetLPCCoeffs = 1; } alphar[1] = FL2FXCONST_SGL(0.0f); if (!useLP) alphai[1] = FL2FXCONST_SGL(0.0f); if (ac.det != FL2FXCONST_DBL(0.0f)) { FIXP_DBL tmp, absTmp, absDet; absDet = fixp_abs(ac.det); if (!useLP) { tmp = (fMultDiv2(ac.r01r, ac.r12r) >> (LPC_SCALE_FACTOR - 1)) - ((fMultDiv2(ac.r01i, ac.r12i) + fMultDiv2(ac.r02r, ac.r11r)) >> (LPC_SCALE_FACTOR - 1)); } else { tmp = (fMultDiv2(ac.r01r, ac.r12r) >> (LPC_SCALE_FACTOR - 1)) - (fMultDiv2(ac.r02r, ac.r11r) >> (LPC_SCALE_FACTOR - 1)); } absTmp = fixp_abs(tmp); /* Quick check: is first filter coeff >= 1(4) */ { INT scale; FIXP_DBL result = fDivNorm(absTmp, absDet, &scale); scale = scale + ac.det_scale; if ((scale > 0) && (result >= (FIXP_DBL)MAXVAL_DBL >> scale)) { resetLPCCoeffs = 1; } else { alphar[1] = FX_DBL2FX_SGL(scaleValue(result, scale)); if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) { alphar[1] = -alphar[1]; } } } if (!useLP) { tmp = (fMultDiv2(ac.r01i, ac.r12r) >> (LPC_SCALE_FACTOR - 1)) + ((fMultDiv2(ac.r01r, ac.r12i) - (FIXP_DBL)fMultDiv2(ac.r02i, ac.r11r)) >> (LPC_SCALE_FACTOR - 1)); absTmp = fixp_abs(tmp); /* Quick check: is second filter coeff >= 1(4) */ { INT scale; FIXP_DBL result = fDivNorm(absTmp, absDet, &scale); scale = scale + ac.det_scale; if ((scale > 0) && (result >= /*FL2FXCONST_DBL(1.f)*/ (FIXP_DBL)MAXVAL_DBL >> scale)) { resetLPCCoeffs = 1; } else { alphai[1] = FX_DBL2FX_SGL(scaleValue(result, scale)); if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) { alphai[1] = -alphai[1]; } } } } } alphar[0] = FL2FXCONST_SGL(0.0f); if (!useLP) alphai[0] = FL2FXCONST_SGL(0.0f); if (ac.r11r != FL2FXCONST_DBL(0.0f)) { /* ac.r11r is always >=0 */ FIXP_DBL tmp, absTmp; if (!useLP) { tmp = (ac.r01r >> (LPC_SCALE_FACTOR + 1)) + (fMultDiv2(alphar[1], ac.r12r) + fMultDiv2(alphai[1], ac.r12i)); } else { if (ac.r01r >= FL2FXCONST_DBL(0.0f)) tmp = (ac.r01r >> (LPC_SCALE_FACTOR + 1)) + fMultDiv2(alphar[1], ac.r12r); else tmp = -((-ac.r01r) >> (LPC_SCALE_FACTOR + 1)) + fMultDiv2(alphar[1], ac.r12r); } absTmp = fixp_abs(tmp); /* Quick check: is first filter coeff >= 1(4) */ if (absTmp >= (ac.r11r >> 1)) { resetLPCCoeffs = 1; } else { INT scale; FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale); alphar[0] = FX_DBL2FX_SGL(scaleValue(result, scale + 1)); if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f))) alphar[0] = -alphar[0]; } if (!useLP) { tmp = (ac.r01i >> (LPC_SCALE_FACTOR + 1)) + (fMultDiv2(alphai[1], ac.r12r) - fMultDiv2(alphar[1], ac.r12i)); absTmp = fixp_abs(tmp); /* Quick check: is second filter coeff >= 1(4) */ if (absTmp >= (ac.r11r >> 1)) { resetLPCCoeffs = 1; } else { INT scale; FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale); alphai[0] = FX_DBL2FX_SGL(scaleValue(result, scale + 1)); if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f))) alphai[0] = -alphai[0]; } } } if (!useLP) { /* Now check the quadratic criteria */ if ((fMultDiv2(alphar[0], alphar[0]) + fMultDiv2(alphai[0], alphai[0])) >= FL2FXCONST_DBL(0.5f)) resetLPCCoeffs = 1; if ((fMultDiv2(alphar[1], alphar[1]) + fMultDiv2(alphai[1], alphai[1])) >= FL2FXCONST_DBL(0.5f)) resetLPCCoeffs = 1; } if (resetLPCCoeffs) { alphar[0] = FL2FXCONST_SGL(0.0f); alphar[1] = FL2FXCONST_SGL(0.0f); if (!useLP) { alphai[0] = FL2FXCONST_SGL(0.0f); alphai[1] = FL2FXCONST_SGL(0.0f); } } if (useLP) { /* Aliasing detection */ if (ac.r11r == FL2FXCONST_DBL(0.0f)) { k1 = FL2FXCONST_DBL(0.0f); } else { if (fixp_abs(ac.r01r) >= fixp_abs(ac.r11r)) { if (fMultDiv2(ac.r01r, ac.r11r) < FL2FX_DBL(0.0f)) { k1 = (FIXP_DBL)MAXVAL_DBL /*FL2FXCONST_SGL(1.0f)*/; } else { /* Since this value is squared later, it must not ever become -1.0f. */ k1 = (FIXP_DBL)(MINVAL_DBL + 1) /*FL2FXCONST_SGL(-1.0f)*/; } } else { INT scale; FIXP_DBL result = fDivNorm(fixp_abs(ac.r01r), fixp_abs(ac.r11r), &scale); k1 = scaleValue(result, scale); if (!((ac.r01r < FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f)))) { k1 = -k1; } } } if ((loBand > 1) && (loBand < v_k_master0)) { /* Check if the gain should be locked */ FIXP_DBL deg = /*FL2FXCONST_DBL(1.0f)*/ (FIXP_DBL)MAXVAL_DBL - fPow2(k1_below); degreeAlias[loBand] = FL2FXCONST_DBL(0.0f); if (((loBand & 1) == 0) && (k1 < FL2FXCONST_DBL(0.0f))) { if (k1_below < FL2FXCONST_DBL(0.0f)) { /* 2-Ch Aliasing Detection */ degreeAlias[loBand] = (FIXP_DBL)MAXVAL_DBL /*FL2FXCONST_DBL(1.0f)*/; if (k1_below2 > FL2FXCONST_DBL(0.0f)) { /* 3-Ch Aliasing Detection */ degreeAlias[loBand - 1] = deg; } } else if (k1_below2 > FL2FXCONST_DBL(0.0f)) { /* 3-Ch Aliasing Detection */ degreeAlias[loBand] = deg; } } if (((loBand & 1) == 1) && (k1 > FL2FXCONST_DBL(0.0f))) { if (k1_below > FL2FXCONST_DBL(0.0f)) { /* 2-CH Aliasing Detection */ degreeAlias[loBand] = (FIXP_DBL)MAXVAL_DBL /*FL2FXCONST_DBL(1.0f)*/; if (k1_below2 < FL2FXCONST_DBL(0.0f)) { /* 3-CH Aliasing Detection */ degreeAlias[loBand - 1] = deg; } } else if (k1_below2 < FL2FXCONST_DBL(0.0f)) { /* 3-CH Aliasing Detection */ degreeAlias[loBand] = deg; } } } /* remember k1 values of the 2 QMF channels below the current channel */ k1_below2 = k1_below; k1_below = k1; } patch = 0; while (patch < pSettings->noOfPatches) { /* inner loop over every patch */ int hiBand = loBand + patchParam[patch].targetBandOffs; if (loBand < patchParam[patch].sourceStartBand || loBand >= patchParam[patch].sourceStopBand //|| hiBand >= hLppTrans->pSettings->noChannels ) { /* Lowband not in current patch - proceed */ patch++; continue; } FDK_ASSERT(hiBand < (64)); /* bwIndex[patch] is already initialized with value from previous band * inside this patch */ while (hiBand >= pSettings->bwBorders[bwIndex[patch]] && bwIndex[patch] < MAX_NUM_PATCHES - 1) { bwIndex[patch]++; } /* Filter Step 2: add the left slope with the current filter to the buffer pure source values are already in there */ bw = FX_DBL2FX_SGL(bwVector[bwIndex[patch]]); a0r = FX_DBL2FX_SGL( fMult(bw, alphar[0])); /* Apply current bandwidth expansion factor */ if (!useLP) a0i = FX_DBL2FX_SGL(fMult(bw, alphai[0])); bw = FX_DBL2FX_SGL(fPow2(bw)); a1r = FX_DBL2FX_SGL(fMult(bw, alphar[1])); if (!useLP) a1i = FX_DBL2FX_SGL(fMult(bw, alphai[1])); /* Filter Step 3: insert the middle part which won't be windowed */ if (bw <= FL2FXCONST_SGL(0.0f)) { if (!useLP) { int descale = fixMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale)); for (i = startSample; i < stopSample; i++) { FIXP_DBL accu1, accu2; accu1 = lowBandReal[LPC_ORDER + i] >> descale; accu2 = lowBandImag[LPC_ORDER + i] >> descale; if (fPreWhitening) { accu1 = scaleValueSaturate( fMultDiv2(accu1, preWhiteningGains[loBand]), preWhiteningGains_exp[loBand] + 1); accu2 = scaleValueSaturate( fMultDiv2(accu2, preWhiteningGains[loBand]), preWhiteningGains_exp[loBand] + 1); } qmfBufferReal[i][hiBand] = accu1; qmfBufferImag[i][hiBand] = accu2; } } else { int descale = fixMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale)); for (i = startSample; i < stopSample; i++) { qmfBufferReal[i][hiBand] = lowBandReal[LPC_ORDER + i] >> descale; } } } else { /* bw <= 0 */ if (!useLP) { int descale = fixMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale)); #ifdef FUNCTION_LPPTRANSPOSER_func1 lppTransposer_func1( lowBandReal + LPC_ORDER + startSample, lowBandImag + LPC_ORDER + startSample, qmfBufferReal + startSample, qmfBufferImag + startSample, stopSample - startSample, (int)hiBand, dynamicScale, descale, a0r, a0i, a1r, a1i, fPreWhitening, preWhiteningGains[loBand], preWhiteningGains_exp[loBand] + 1); #else for (i = startSample; i < stopSample; i++) { FIXP_DBL accu1, accu2; accu1 = (fMultDiv2(a0r, lowBandReal[LPC_ORDER + i - 1]) - fMultDiv2(a0i, lowBandImag[LPC_ORDER + i - 1]) + fMultDiv2(a1r, lowBandReal[LPC_ORDER + i - 2]) - fMultDiv2(a1i, lowBandImag[LPC_ORDER + i - 2])) >> dynamicScale; accu2 = (fMultDiv2(a0i, lowBandReal[LPC_ORDER + i - 1]) + fMultDiv2(a0r, lowBandImag[LPC_ORDER + i - 1]) + fMultDiv2(a1i, lowBandReal[LPC_ORDER + i - 2]) + fMultDiv2(a1r, lowBandImag[LPC_ORDER + i - 2])) >> dynamicScale; accu1 = (lowBandReal[LPC_ORDER + i] >> descale) + (accu1 << 1); accu2 = (lowBandImag[LPC_ORDER + i] >> descale) + (accu2 << 1); if (fPreWhitening) { accu1 = scaleValueSaturate( fMultDiv2(accu1, preWhiteningGains[loBand]), preWhiteningGains_exp[loBand] + 1); accu2 = scaleValueSaturate( fMultDiv2(accu2, preWhiteningGains[loBand]), preWhiteningGains_exp[loBand] + 1); } qmfBufferReal[i][hiBand] = accu1; qmfBufferImag[i][hiBand] = accu2; } #endif } else { FDK_ASSERT(dynamicScale >= 0); calc_qmfBufferReal( qmfBufferReal, &(lowBandReal[LPC_ORDER + startSample - 2]), startSample, stopSample, hiBand, dynamicScale, fMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale)), a0r, a1r); } } /* bw <= 0 */ patch++; } /* inner loop over patches */ /* * store the unmodified filter coefficients if there is * an overlapping envelope *****************************************************************/ } /* outer loop over bands (loBand) */ if (useLP) { for (loBand = pSettings->lbStartPatching; loBand < pSettings->lbStopPatching; loBand++) { patch = 0; while (patch < pSettings->noOfPatches) { UCHAR hiBand = loBand + patchParam[patch].targetBandOffs; if (loBand < patchParam[patch].sourceStartBand || loBand >= patchParam[patch].sourceStopBand || hiBand >= (64) /* Highband out of range (biterror) */ ) { /* Lowband not in current patch or highband out of range (might be * caused by biterrors)- proceed */ patch++; continue; } if (hiBand != patchParam[patch].targetStartBand) degreeAlias[hiBand] = degreeAlias[loBand]; patch++; } } /* end for loop */ } for (i = 0; i < nInvfBands; i++) { hLppTrans->bwVectorOld[i] = bwVector[i]; } /* set high band scale factor */ sbrScaleFactor->hb_scale = comLowBandScale - (LPC_SCALE_FACTOR); } void lppTransposerHBE( HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transposer */ HANDLE_HBE_TRANSPOSER hQmfTransposer, QMF_SCALE_FACTOR *sbrScaleFactor, /*!< Scaling factors */ FIXP_DBL **qmfBufferReal, /*!< Pointer to pointer to real part of subband samples (source) */ FIXP_DBL **qmfBufferImag, /*!< Pointer to pointer to imaginary part of subband samples (source) */ const int timeStep, /*!< Time step of envelope */ const int firstSlotOffs, /*!< Start position in time */ const int lastSlotOffs, /*!< Number of overlap-slots into next frame */ const int nInvfBands, /*!< Number of bands for inverse filtering */ INVF_MODE *sbr_invf_mode, /*!< Current inverse filtering modes */ INVF_MODE *sbr_invf_mode_prev /*!< Previous inverse filtering modes */ ) { INT bwIndex; FIXP_DBL bwVector[MAX_NUM_PATCHES_HBE]; /*!< pole moving factors */ int i; int loBand, start, stop; TRANSPOSER_SETTINGS *pSettings = hLppTrans->pSettings; PATCH_PARAM *patchParam = pSettings->patchParam; FIXP_SGL alphar[LPC_ORDER], a0r, a1r; FIXP_SGL alphai[LPC_ORDER], a0i = 0, a1i = 0; FIXP_SGL bw = FL2FXCONST_SGL(0.0f); int autoCorrLength; ACORR_COEFS ac; int startSample; int stopSample; int stopSampleClear; int comBandScale; int ovLowBandShift; int lowBandShift; /* int ovHighBandShift;*/ alphai[0] = FL2FXCONST_SGL(0.0f); alphai[1] = FL2FXCONST_SGL(0.0f); startSample = firstSlotOffs * timeStep; stopSample = pSettings->nCols + lastSlotOffs * timeStep; inverseFilteringLevelEmphasis(hLppTrans, nInvfBands, sbr_invf_mode, sbr_invf_mode_prev, bwVector); stopSampleClear = stopSample; autoCorrLength = pSettings->nCols + pSettings->overlap; if (pSettings->noOfPatches > 0) { /* Set upper subbands to zero: This is required in case that the patches do not cover the complete highband (because the last patch would be too short). Possible optimization: Clearing bands up to usb would be sufficient here. */ int targetStopBand = patchParam[pSettings->noOfPatches - 1].targetStartBand + patchParam[pSettings->noOfPatches - 1].numBandsInPatch; int memSize = ((64) - targetStopBand) * sizeof(FIXP_DBL); for (i = startSample; i < stopSampleClear; i++) { FDKmemclear(&qmfBufferReal[i][targetStopBand], memSize); FDKmemclear(&qmfBufferImag[i][targetStopBand], memSize); } } #ifdef __ANDROID__ else { // Safetynet logging android_errorWriteLog(0x534e4554, "112160868"); } #endif /* Calc common low band scale factor */ comBandScale = sbrScaleFactor->hb_scale; ovLowBandShift = sbrScaleFactor->hb_scale - comBandScale; lowBandShift = sbrScaleFactor->hb_scale - comBandScale; /* ovHighBandShift = firstSlotOffs == 0 ? ovLowBandShift:0;*/ /* outer loop over bands to do analysis only once for each band */ start = hQmfTransposer->startBand; stop = hQmfTransposer->stopBand; for (loBand = start; loBand < stop; loBand++) { bwIndex = 0; FIXP_DBL lowBandReal[(((1024) / (32) * (4) / 2) + (3 * (4))) + LPC_ORDER]; FIXP_DBL lowBandImag[(((1024) / (32) * (4) / 2) + (3 * (4))) + LPC_ORDER]; int resetLPCCoeffs = 0; int dynamicScale = DFRACT_BITS - 1 - LPC_SCALE_FACTOR; int acDetScale = 0; /* scaling of autocorrelation determinant */ for (i = 0; i < LPC_ORDER; i++) { lowBandReal[i] = hLppTrans->lpcFilterStatesRealHBE[i][loBand]; lowBandImag[i] = hLppTrans->lpcFilterStatesImagHBE[i][loBand]; } for (; i < LPC_ORDER + firstSlotOffs * timeStep; i++) { lowBandReal[i] = hLppTrans->lpcFilterStatesRealHBE[i][loBand]; lowBandImag[i] = hLppTrans->lpcFilterStatesImagHBE[i][loBand]; } /* Take old slope length qmf slot source values out of (overlap)qmf buffer */ for (i = firstSlotOffs * timeStep; i < pSettings->nCols + pSettings->overlap; i++) { lowBandReal[i + LPC_ORDER] = qmfBufferReal[i][loBand]; lowBandImag[i + LPC_ORDER] = qmfBufferImag[i][loBand]; } /* store unmodified values to buffer */ for (i = 0; i < LPC_ORDER + pSettings->overlap; i++) { hLppTrans->lpcFilterStatesRealHBE[i][loBand] = qmfBufferReal[pSettings->nCols - LPC_ORDER + i][loBand]; hLppTrans->lpcFilterStatesImagHBE[i][loBand] = qmfBufferImag[pSettings->nCols - LPC_ORDER + i][loBand]; } /* Determine dynamic scaling value. */ dynamicScale = fixMin(dynamicScale, getScalefactor(lowBandReal, LPC_ORDER + pSettings->overlap) + ovLowBandShift); dynamicScale = fixMin(dynamicScale, getScalefactor(&lowBandReal[LPC_ORDER + pSettings->overlap], pSettings->nCols) + lowBandShift); dynamicScale = fixMin(dynamicScale, getScalefactor(lowBandImag, LPC_ORDER + pSettings->overlap) + ovLowBandShift); dynamicScale = fixMin(dynamicScale, getScalefactor(&lowBandImag[LPC_ORDER + pSettings->overlap], pSettings->nCols) + lowBandShift); dynamicScale = dynamicScale - 1; /* one additional bit headroom to prevent -1.0 */ /* Scale temporal QMF buffer. */ scaleValues(&lowBandReal[0], LPC_ORDER + pSettings->overlap, dynamicScale - ovLowBandShift); scaleValues(&lowBandReal[LPC_ORDER + pSettings->overlap], pSettings->nCols, dynamicScale - lowBandShift); scaleValues(&lowBandImag[0], LPC_ORDER + pSettings->overlap, dynamicScale - ovLowBandShift); scaleValues(&lowBandImag[LPC_ORDER + pSettings->overlap], pSettings->nCols, dynamicScale - lowBandShift); acDetScale += autoCorr2nd_cplx(&ac, lowBandReal + LPC_ORDER, lowBandImag + LPC_ORDER, autoCorrLength); /* Examine dynamic of determinant in autocorrelation. */ acDetScale += 2 * (comBandScale + dynamicScale); acDetScale *= 2; /* two times reflection coefficent scaling */ acDetScale += ac.det_scale; /* ac scaling of determinant */ /* In case of determinant < 10^-38, resetLPCCoeffs=1 has to be enforced. */ if (acDetScale > 126) { resetLPCCoeffs = 1; } alphar[1] = FL2FXCONST_SGL(0.0f); alphai[1] = FL2FXCONST_SGL(0.0f); if (ac.det != FL2FXCONST_DBL(0.0f)) { FIXP_DBL tmp, absTmp, absDet; absDet = fixp_abs(ac.det); tmp = (fMultDiv2(ac.r01r, ac.r12r) >> (LPC_SCALE_FACTOR - 1)) - ((fMultDiv2(ac.r01i, ac.r12i) + fMultDiv2(ac.r02r, ac.r11r)) >> (LPC_SCALE_FACTOR - 1)); absTmp = fixp_abs(tmp); /* Quick check: is first filter coeff >= 1(4) */ { INT scale; FIXP_DBL result = fDivNorm(absTmp, absDet, &scale); scale = scale + ac.det_scale; if ((scale > 0) && (result >= (FIXP_DBL)MAXVAL_DBL >> scale)) { resetLPCCoeffs = 1; } else { alphar[1] = FX_DBL2FX_SGL(scaleValue(result, scale)); if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) { alphar[1] = -alphar[1]; } } } tmp = (fMultDiv2(ac.r01i, ac.r12r) >> (LPC_SCALE_FACTOR - 1)) + ((fMultDiv2(ac.r01r, ac.r12i) - (FIXP_DBL)fMultDiv2(ac.r02i, ac.r11r)) >> (LPC_SCALE_FACTOR - 1)); absTmp = fixp_abs(tmp); /* Quick check: is second filter coeff >= 1(4) */ { INT scale; FIXP_DBL result = fDivNorm(absTmp, absDet, &scale); scale = scale + ac.det_scale; if ((scale > 0) && (result >= /*FL2FXCONST_DBL(1.f)*/ (FIXP_DBL)MAXVAL_DBL >> scale)) { resetLPCCoeffs = 1; } else { alphai[1] = FX_DBL2FX_SGL(scaleValue(result, scale)); if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) { alphai[1] = -alphai[1]; } } } } alphar[0] = FL2FXCONST_SGL(0.0f); alphai[0] = FL2FXCONST_SGL(0.0f); if (ac.r11r != FL2FXCONST_DBL(0.0f)) { /* ac.r11r is always >=0 */ FIXP_DBL tmp, absTmp; tmp = (ac.r01r >> (LPC_SCALE_FACTOR + 1)) + (fMultDiv2(alphar[1], ac.r12r) + fMultDiv2(alphai[1], ac.r12i)); absTmp = fixp_abs(tmp); /* Quick check: is first filter coeff >= 1(4) */ if (absTmp >= (ac.r11r >> 1)) { resetLPCCoeffs = 1; } else { INT scale; FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale); alphar[0] = FX_DBL2FX_SGL(scaleValue(result, scale + 1)); if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f))) alphar[0] = -alphar[0]; } tmp = (ac.r01i >> (LPC_SCALE_FACTOR + 1)) + (fMultDiv2(alphai[1], ac.r12r) - fMultDiv2(alphar[1], ac.r12i)); absTmp = fixp_abs(tmp); /* Quick check: is second filter coeff >= 1(4) */ if (absTmp >= (ac.r11r >> 1)) { resetLPCCoeffs = 1; } else { INT scale; FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale); alphai[0] = FX_DBL2FX_SGL(scaleValue(result, scale + 1)); if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f))) { alphai[0] = -alphai[0]; } } } /* Now check the quadratic criteria */ if ((fMultDiv2(alphar[0], alphar[0]) + fMultDiv2(alphai[0], alphai[0])) >= FL2FXCONST_DBL(0.5f)) { resetLPCCoeffs = 1; } if ((fMultDiv2(alphar[1], alphar[1]) + fMultDiv2(alphai[1], alphai[1])) >= FL2FXCONST_DBL(0.5f)) { resetLPCCoeffs = 1; } if (resetLPCCoeffs) { alphar[0] = FL2FXCONST_SGL(0.0f); alphar[1] = FL2FXCONST_SGL(0.0f); alphai[0] = FL2FXCONST_SGL(0.0f); alphai[1] = FL2FXCONST_SGL(0.0f); } while (bwIndex < MAX_NUM_PATCHES - 1 && loBand >= pSettings->bwBorders[bwIndex]) { bwIndex++; } /* Filter Step 2: add the left slope with the current filter to the buffer pure source values are already in there */ bw = FX_DBL2FX_SGL(bwVector[bwIndex]); a0r = FX_DBL2FX_SGL( fMult(bw, alphar[0])); /* Apply current bandwidth expansion factor */ a0i = FX_DBL2FX_SGL(fMult(bw, alphai[0])); bw = FX_DBL2FX_SGL(fPow2(bw)); a1r = FX_DBL2FX_SGL(fMult(bw, alphar[1])); a1i = FX_DBL2FX_SGL(fMult(bw, alphai[1])); /* Filter Step 3: insert the middle part which won't be windowed */ if (bw <= FL2FXCONST_SGL(0.0f)) { int descale = fixMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale)); for (i = startSample; i < stopSample; i++) { qmfBufferReal[i][loBand] = lowBandReal[LPC_ORDER + i] >> descale; qmfBufferImag[i][loBand] = lowBandImag[LPC_ORDER + i] >> descale; } } else { /* bw <= 0 */ int descale = fixMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale)); dynamicScale += 1; /* prevent negativ scale factor due to 'one additional bit headroom' */ for (i = startSample; i < stopSample; i++) { FIXP_DBL accu1, accu2; accu1 = (fMultDiv2(a0r, lowBandReal[LPC_ORDER + i - 1]) - fMultDiv2(a0i, lowBandImag[LPC_ORDER + i - 1]) + fMultDiv2(a1r, lowBandReal[LPC_ORDER + i - 2]) - fMultDiv2(a1i, lowBandImag[LPC_ORDER + i - 2])) >> dynamicScale; accu2 = (fMultDiv2(a0i, lowBandReal[LPC_ORDER + i - 1]) + fMultDiv2(a0r, lowBandImag[LPC_ORDER + i - 1]) + fMultDiv2(a1i, lowBandReal[LPC_ORDER + i - 2]) + fMultDiv2(a1r, lowBandImag[LPC_ORDER + i - 2])) >> dynamicScale; qmfBufferReal[i][loBand] = (lowBandReal[LPC_ORDER + i] >> descale) + (accu1 << (1 + 1)); qmfBufferImag[i][loBand] = (lowBandImag[LPC_ORDER + i] >> descale) + (accu2 << (1 + 1)); } } /* bw <= 0 */ /* * store the unmodified filter coefficients if there is * an overlapping envelope *****************************************************************/ } /* outer loop over bands (loBand) */ for (i = 0; i < nInvfBands; i++) { hLppTrans->bwVectorOld[i] = bwVector[i]; } /* set high band scale factor */ sbrScaleFactor->hb_scale = comBandScale - (LPC_SCALE_FACTOR); } /*! * * \brief Initialize one low power transposer instance * * */ SBR_ERROR createLppTransposer( HANDLE_SBR_LPP_TRANS hs, /*!< Handle of low power transposer */ TRANSPOSER_SETTINGS *pSettings, /*!< Pointer to settings */ const int highBandStartSb, /*!< ? */ UCHAR *v_k_master, /*!< Master table */ const int numMaster, /*!< Valid entries in master table */ const int usb, /*!< Highband area stop subband */ const int timeSlots, /*!< Number of time slots */ const int nCols, /*!< Number of colums (codec qmf bank) */ UCHAR *noiseBandTable, /*!< Mapping of SBR noise bands to QMF bands */ const int noNoiseBands, /*!< Number of noise bands */ UINT fs, /*!< Sample Frequency */ const int chan, /*!< Channel number */ const int overlap) { /* FB inverse filtering settings */ hs->pSettings = pSettings; pSettings->nCols = nCols; pSettings->overlap = overlap; switch (timeSlots) { case 15: case 16: break; default: return SBRDEC_UNSUPPORTED_CONFIG; /* Unimplemented */ } if (chan == 0) { /* Init common data only once */ hs->pSettings->nCols = nCols; return resetLppTransposer(hs, highBandStartSb, v_k_master, numMaster, noiseBandTable, noNoiseBands, usb, fs); } return SBRDEC_OK; } static int findClosestEntry(UCHAR goalSb, UCHAR *v_k_master, UCHAR numMaster, UCHAR direction) { int index; if (goalSb <= v_k_master[0]) return v_k_master[0]; if (goalSb >= v_k_master[numMaster]) return v_k_master[numMaster]; if (direction) { index = 0; while (v_k_master[index] < goalSb) { index++; } } else { index = numMaster; while (v_k_master[index] > goalSb) { index--; } } return v_k_master[index]; } /*! * * \brief Reset memory for one lpp transposer instance * * \return SBRDEC_OK on success, SBRDEC_UNSUPPORTED_CONFIG on error */ SBR_ERROR resetLppTransposer( HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transposer */ UCHAR highBandStartSb, /*!< High band area: start subband */ UCHAR *v_k_master, /*!< Master table */ UCHAR numMaster, /*!< Valid entries in master table */ UCHAR *noiseBandTable, /*!< Mapping of SBR noise bands to QMF bands */ UCHAR noNoiseBands, /*!< Number of noise bands */ UCHAR usb, /*!< High band area: stop subband */ UINT fs /*!< SBR output sampling frequency */ ) { TRANSPOSER_SETTINGS *pSettings = hLppTrans->pSettings; PATCH_PARAM *patchParam = pSettings->patchParam; int i, patch; int targetStopBand; int sourceStartBand; int patchDistance; int numBandsInPatch; int lsb = v_k_master[0]; /* Start subband expressed in "non-critical" sampling terms*/ int xoverOffset = highBandStartSb - lsb; /* Calculate distance in QMF bands between k0 and kx */ int startFreqHz; int desiredBorder; usb = fixMin(usb, v_k_master[numMaster]); /* Avoid endless loops (compare with float code). */ /* * Plausibility check */ if (pSettings->nCols == 64) { if (lsb < 4) { /* 4:1 SBR Requirement k0 >= 4 missed! */ return SBRDEC_UNSUPPORTED_CONFIG; } } else if (lsb - SHIFT_START_SB < 4) { return SBRDEC_UNSUPPORTED_CONFIG; } /* * Initialize the patching parameter */ /* ISO/IEC 14496-3 (Figure 4.48): goalSb = round( 2.048e6 / fs ) */ desiredBorder = (((2048000 * 2) / fs) + 1) >> 1; desiredBorder = findClosestEntry(desiredBorder, v_k_master, numMaster, 1); /* Adapt region to master-table */ /* First patch */ sourceStartBand = SHIFT_START_SB + xoverOffset; targetStopBand = lsb + xoverOffset; /* upperBand */ /* Even (odd) numbered channel must be patched to even (odd) numbered channel */ patch = 0; while (targetStopBand < usb) { /* Too many patches? Allow MAX_NUM_PATCHES+1 patches here. we need to check later again, since patch might be the highest patch AND contain less than 3 bands => actual number of patches will be reduced by 1. */ if (patch > MAX_NUM_PATCHES) { return SBRDEC_UNSUPPORTED_CONFIG; } patchParam[patch].guardStartBand = targetStopBand; patchParam[patch].targetStartBand = targetStopBand; numBandsInPatch = desiredBorder - targetStopBand; /* Get the desired range of the patch */ if (numBandsInPatch >= lsb - sourceStartBand) { /* Desired number bands are not available -> patch whole source range */ patchDistance = targetStopBand - sourceStartBand; /* Get the targetOffset */ patchDistance = patchDistance & ~1; /* Rounding off odd numbers and make all even */ numBandsInPatch = lsb - (targetStopBand - patchDistance); /* Update number of bands to be patched */ numBandsInPatch = findClosestEntry(targetStopBand + numBandsInPatch, v_k_master, numMaster, 0) - targetStopBand; /* Adapt region to master-table */ } if (pSettings->nCols == 64) { if (numBandsInPatch == 0 && sourceStartBand == SHIFT_START_SB) { return SBRDEC_UNSUPPORTED_CONFIG; } } /* Desired number bands are available -> get the minimal even patching * distance */ patchDistance = numBandsInPatch + targetStopBand - lsb; /* Get minimal distance */ patchDistance = (patchDistance + 1) & ~1; /* Rounding up odd numbers and make all even */ if (numBandsInPatch > 0) { patchParam[patch].sourceStartBand = targetStopBand - patchDistance; patchParam[patch].targetBandOffs = patchDistance; patchParam[patch].numBandsInPatch = numBandsInPatch; patchParam[patch].sourceStopBand = patchParam[patch].sourceStartBand + numBandsInPatch; targetStopBand += patchParam[patch].numBandsInPatch; patch++; } /* All patches but first */ sourceStartBand = SHIFT_START_SB; /* Check if we are close to desiredBorder */ if (desiredBorder - targetStopBand < 3) /* MPEG doc */ { desiredBorder = usb; } } patch--; /* If highest patch contains less than three subband: skip it */ if ((patch > 0) && (patchParam[patch].numBandsInPatch < 3)) { patch--; targetStopBand = patchParam[patch].targetStartBand + patchParam[patch].numBandsInPatch; } /* now check if we don't have one too many */ if (patch >= MAX_NUM_PATCHES) { return SBRDEC_UNSUPPORTED_CONFIG; } pSettings->noOfPatches = patch + 1; /* Check lowest and highest source subband */ pSettings->lbStartPatching = targetStopBand; pSettings->lbStopPatching = 0; for (patch = 0; patch < pSettings->noOfPatches; patch++) { pSettings->lbStartPatching = fixMin(pSettings->lbStartPatching, patchParam[patch].sourceStartBand); pSettings->lbStopPatching = fixMax(pSettings->lbStopPatching, patchParam[patch].sourceStopBand); } for (i = 0; i < noNoiseBands; i++) { pSettings->bwBorders[i] = noiseBandTable[i + 1]; } for (; i < MAX_NUM_NOISE_VALUES; i++) { pSettings->bwBorders[i] = 255; } /* * Choose whitening factors */ startFreqHz = ((lsb + xoverOffset) * fs) >> 7; /* Shift does a division by 2*(64) */ for (i = 1; i < NUM_WHFACTOR_TABLE_ENTRIES; i++) { if (startFreqHz < FDK_sbrDecoder_sbr_whFactorsIndex[i]) break; } i--; pSettings->whFactors.off = FDK_sbrDecoder_sbr_whFactorsTable[i][0]; pSettings->whFactors.transitionLevel = FDK_sbrDecoder_sbr_whFactorsTable[i][1]; pSettings->whFactors.lowLevel = FDK_sbrDecoder_sbr_whFactorsTable[i][2]; pSettings->whFactors.midLevel = FDK_sbrDecoder_sbr_whFactorsTable[i][3]; pSettings->whFactors.highLevel = FDK_sbrDecoder_sbr_whFactorsTable[i][4]; return SBRDEC_OK; } fdk-aac-2.0.2/libSBRdec/src/lpp_tran.h000066400000000000000000000256221404225310700173560ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Low Power Profile Transposer */ #ifndef LPP_TRAN_H #define LPP_TRAN_H #include "sbrdecoder.h" #include "hbe.h" #include "qmf.h" /* Common */ #define QMF_OUT_SCALE 8 /* Frequency scales */ /* Env-Adjust */ #define MAX_NOISE_ENVELOPES 2 #define MAX_NOISE_COEFFS 5 #define MAX_NUM_NOISE_VALUES (MAX_NOISE_ENVELOPES * MAX_NOISE_COEFFS) #define MAX_NUM_LIMITERS 12 /* Set MAX_ENVELOPES to the largest value of all supported BSFORMATs by overriding MAX_ENVELOPES in the correct order: */ #define MAX_ENVELOPES_LEGACY 5 #define MAX_ENVELOPES_USAC 8 #define MAX_ENVELOPES MAX_ENVELOPES_USAC #define MAX_FREQ_COEFFS_DUAL_RATE 48 #define MAX_FREQ_COEFFS_QUAD_RATE 56 #define MAX_FREQ_COEFFS MAX_FREQ_COEFFS_QUAD_RATE #define MAX_FREQ_COEFFS_FS44100 35 #define MAX_FREQ_COEFFS_FS48000 32 #define MAX_NUM_ENVELOPE_VALUES (MAX_ENVELOPES * MAX_FREQ_COEFFS) #define MAX_GAIN_EXP 34 /* Maximum gain will be sqrt(0.5 * 2^MAX_GAIN_EXP) example: 34=99dB */ #define MAX_GAIN_CONCEAL_EXP 1 /* Maximum gain will be sqrt(0.5 * 2^MAX_GAIN_CONCEAL_EXP) in concealment case * (0dB) */ /* LPP Transposer */ #define LPC_ORDER 2 #define MAX_INVF_BANDS MAX_NOISE_COEFFS #define MAX_NUM_PATCHES 6 #define SHIFT_START_SB 1 /*!< lowest subband of source range */ typedef enum { INVF_OFF = 0, INVF_LOW_LEVEL, INVF_MID_LEVEL, INVF_HIGH_LEVEL, INVF_SWITCHED /* not a real choice but used here to control behaviour */ } INVF_MODE; /** parameter set for one single patch */ typedef struct { UCHAR sourceStartBand; /*!< first band in lowbands where to take the samples from */ UCHAR sourceStopBand; /*!< first band in lowbands which is not included in the patch anymore */ UCHAR guardStartBand; /*!< first band in highbands to be filled with zeros in order to reduce interferences between patches */ UCHAR targetStartBand; /*!< first band in highbands to be filled with whitened lowband signal */ UCHAR targetBandOffs; /*!< difference between 'startTargetBand' and 'startSourceBand' */ UCHAR numBandsInPatch; /*!< number of consecutive bands in this one patch */ } PATCH_PARAM; /** whitening factors for different levels of whitening need to be initialized corresponding to crossover frequency */ typedef struct { FIXP_DBL off; /*!< bw factor for signal OFF */ FIXP_DBL transitionLevel; FIXP_DBL lowLevel; /*!< bw factor for signal LOW_LEVEL */ FIXP_DBL midLevel; /*!< bw factor for signal MID_LEVEL */ FIXP_DBL highLevel; /*!< bw factor for signal HIGH_LEVEL */ } WHITENING_FACTORS; /*! The transposer settings are calculated on a header reset and are shared by * both channels. */ typedef struct { UCHAR nCols; /*!< number subsamples of a codec frame */ UCHAR noOfPatches; /*!< number of patches */ UCHAR lbStartPatching; /*!< first band of lowbands that will be patched */ UCHAR lbStopPatching; /*!< first band that won't be patched anymore*/ UCHAR bwBorders[MAX_NUM_NOISE_VALUES]; /*!< spectral bands with different inverse filtering levels */ PATCH_PARAM patchParam[MAX_NUM_PATCHES]; /*!< new parameter set for patching */ WHITENING_FACTORS whFactors; /*!< the pole moving factors for certain whitening levels as indicated in the bitstream depending on the crossover frequency */ UCHAR overlap; /*!< Overlap size */ } TRANSPOSER_SETTINGS; typedef struct { TRANSPOSER_SETTINGS *pSettings; /*!< Common settings for both channels */ FIXP_DBL bwVectorOld[MAX_NUM_PATCHES]; /*!< pole moving factors of past frame */ FIXP_DBL lpcFilterStatesRealLegSBR[LPC_ORDER + (3 * (4))][( 32)]; /*!< pointer array to save filter states */ FIXP_DBL lpcFilterStatesImagLegSBR[LPC_ORDER + (3 * (4))][( 32)]; /*!< pointer array to save filter states */ FIXP_DBL lpcFilterStatesRealHBE[LPC_ORDER + (3 * (4))][( 64)]; /*!< pointer array to save filter states */ FIXP_DBL lpcFilterStatesImagHBE[LPC_ORDER + (3 * (4))][( 64)]; /*!< pointer array to save filter states */ } SBR_LPP_TRANS; typedef SBR_LPP_TRANS *HANDLE_SBR_LPP_TRANS; void lppTransposer(HANDLE_SBR_LPP_TRANS hLppTrans, QMF_SCALE_FACTOR *sbrScaleFactor, FIXP_DBL **qmfBufferReal, FIXP_DBL *degreeAlias, FIXP_DBL **qmfBufferImag, const int useLP, const int fPreWhitening, const int v_k_master0, const int timeStep, const int firstSlotOffset, const int lastSlotOffset, const int nInvfBands, INVF_MODE *sbr_invf_mode, INVF_MODE *sbr_invf_mode_prev); void lppTransposerHBE( HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transposer */ HANDLE_HBE_TRANSPOSER hQmfTransposer, QMF_SCALE_FACTOR *sbrScaleFactor, /*!< Scaling factors */ FIXP_DBL **qmfBufferReal, /*!< Pointer to pointer to real part of subband samples (source) */ FIXP_DBL **qmfBufferImag, /*!< Pointer to pointer to imaginary part of subband samples (source) */ const int timeStep, /*!< Time step of envelope */ const int firstSlotOffs, /*!< Start position in time */ const int lastSlotOffs, /*!< Number of overlap-slots into next frame */ const int nInvfBands, /*!< Number of bands for inverse filtering */ INVF_MODE *sbr_invf_mode, /*!< Current inverse filtering modes */ INVF_MODE *sbr_invf_mode_prev /*!< Previous inverse filtering modes */ ); SBR_ERROR createLppTransposer(HANDLE_SBR_LPP_TRANS hLppTrans, TRANSPOSER_SETTINGS *pSettings, const int highBandStartSb, UCHAR *v_k_master, const int numMaster, const int usb, const int timeSlots, const int nCols, UCHAR *noiseBandTable, const int noNoiseBands, UINT fs, const int chan, const int overlap); SBR_ERROR resetLppTransposer(HANDLE_SBR_LPP_TRANS hLppTrans, UCHAR highBandStartSb, UCHAR *v_k_master, UCHAR numMaster, UCHAR *noiseBandTable, UCHAR noNoiseBands, UCHAR usb, UINT fs); #endif /* LPP_TRAN_H */ fdk-aac-2.0.2/libSBRdec/src/psbitdec.cpp000066400000000000000000000516701404225310700176710ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Description: *******************************************************************************/ #include "psbitdec.h" #include "sbr_rom.h" #include "huff_dec.h" /* PS dec privat functions */ SBR_ERROR ResetPsDec(HANDLE_PS_DEC h_ps_d); /***************************************************************************/ /*! \brief huffman decoding by codebook table \return index of huffman codebook table ****************************************************************************/ static SCHAR decode_huff_cw( Huffman h, /*!< pointer to huffman codebook table */ HANDLE_FDK_BITSTREAM hBitBuf, /*!< Handle to Bitbuffer */ int *length) /*!< length of huffman codeword (or NULL) */ { UCHAR bit = 0; SCHAR index = 0; UCHAR bitCount = 0; while (index >= 0) { bit = FDKreadBits(hBitBuf, 1); bitCount++; index = h[index][bit]; } if (length) { *length = bitCount; } return (index + 64); /* Add offset */ } /***************************************************************************/ /*! \brief helper function - limiting of value to min/max values \return limited value ****************************************************************************/ static SCHAR limitMinMax(SCHAR i, SCHAR min, SCHAR max) { if (i < min) return min; else if (i > max) return max; else return i; } /***************************************************************************/ /*! \brief Decodes delta values in-place and updates data buffers according to quantization classes. When delta coded in frequency the first element is deltacode from zero. aIndex buffer is decoded from delta values to actual values. \return none ****************************************************************************/ static void deltaDecodeArray( SCHAR enable, SCHAR *aIndex, /*!< ICC/IID parameters */ SCHAR *aPrevFrameIndex, /*!< ICC/IID parameters of previous frame */ SCHAR DtDf, UCHAR nrElements, /*!< as conveyed in bitstream */ /*!< output array size: nrElements*stride */ UCHAR stride, /*!< 1=dflt, 2=half freq. resolution */ SCHAR minIdx, SCHAR maxIdx) { int i; /* Delta decode */ if (enable == 1) { if (DtDf == 0) { /* Delta coded in freq */ aIndex[0] = 0 + aIndex[0]; aIndex[0] = limitMinMax(aIndex[0], minIdx, maxIdx); for (i = 1; i < nrElements; i++) { aIndex[i] = aIndex[i - 1] + aIndex[i]; aIndex[i] = limitMinMax(aIndex[i], minIdx, maxIdx); } } else { /* Delta time */ for (i = 0; i < nrElements; i++) { aIndex[i] = aPrevFrameIndex[i * stride] + aIndex[i]; aIndex[i] = limitMinMax(aIndex[i], minIdx, maxIdx); } } } else { /* No data is sent, set index to zero */ for (i = 0; i < nrElements; i++) { aIndex[i] = 0; } } if (stride == 2) { for (i = nrElements * stride - 1; i > 0; i--) { aIndex[i] = aIndex[i >> 1]; } } } /***************************************************************************/ /*! \brief Mapping of ICC/IID parameters to 20 stereo bands \return none ****************************************************************************/ static void map34IndexTo20(SCHAR *aIndex, /*!< decoded ICC/IID parameters */ UCHAR noBins) /*!< number of stereo bands */ { aIndex[0] = (2 * aIndex[0] + aIndex[1]) / 3; aIndex[1] = (aIndex[1] + 2 * aIndex[2]) / 3; aIndex[2] = (2 * aIndex[3] + aIndex[4]) / 3; aIndex[3] = (aIndex[4] + 2 * aIndex[5]) / 3; aIndex[4] = (aIndex[6] + aIndex[7]) / 2; aIndex[5] = (aIndex[8] + aIndex[9]) / 2; aIndex[6] = aIndex[10]; aIndex[7] = aIndex[11]; aIndex[8] = (aIndex[12] + aIndex[13]) / 2; aIndex[9] = (aIndex[14] + aIndex[15]) / 2; aIndex[10] = aIndex[16]; /* For IPD/OPD it stops here */ if (noBins == NO_HI_RES_BINS) { aIndex[11] = aIndex[17]; aIndex[12] = aIndex[18]; aIndex[13] = aIndex[19]; aIndex[14] = (aIndex[20] + aIndex[21]) / 2; aIndex[15] = (aIndex[22] + aIndex[23]) / 2; aIndex[16] = (aIndex[24] + aIndex[25]) / 2; aIndex[17] = (aIndex[26] + aIndex[27]) / 2; aIndex[18] = (aIndex[28] + aIndex[29] + aIndex[30] + aIndex[31]) / 4; aIndex[19] = (aIndex[32] + aIndex[33]) / 2; } } /***************************************************************************/ /*! \brief Decodes delta coded IID, ICC, IPD and OPD indices \return PS processing flag. If set to 1 ****************************************************************************/ int DecodePs(struct PS_DEC *h_ps_d, /*!< PS handle */ const UCHAR frameError, /*!< Flag telling that frame had errors */ PS_DEC_COEFFICIENTS *pScratch) { MPEG_PS_BS_DATA *pBsData; UCHAR gr, env; int bPsHeaderValid, bPsDataAvail; /* Assign Scratch */ h_ps_d->specificTo.mpeg.pCoef = pScratch; /* Shortcuts to avoid deferencing and keep the code readable */ pBsData = &h_ps_d->bsData[h_ps_d->processSlot].mpeg; bPsHeaderValid = pBsData->bPsHeaderValid; bPsDataAvail = (h_ps_d->bPsDataAvail[h_ps_d->processSlot] == ppt_mpeg) ? 1 : 0; /*************************************************************************************** * Decide whether to process or to conceal PS data or not. */ if ((h_ps_d->psDecodedPrv && !frameError && !bPsDataAvail) || (!h_ps_d->psDecodedPrv && (frameError || !bPsDataAvail || !bPsHeaderValid))) { /* Don't apply PS processing. * Declare current PS header and bitstream data invalid. */ pBsData->bPsHeaderValid = 0; h_ps_d->bPsDataAvail[h_ps_d->processSlot] = ppt_none; return (0); } if (frameError || !bPsHeaderValid) { /* no new PS data available (e.g. frame loss) */ /* => keep latest data constant (i.e. FIX with noEnv=0) */ pBsData->noEnv = 0; } /*************************************************************************************** * Decode bitstream payload or prepare parameter for concealment: */ for (env = 0; env < pBsData->noEnv; env++) { SCHAR *aPrevIidIndex; SCHAR *aPrevIccIndex; UCHAR noIidSteps = pBsData->bFineIidQ ? NO_IID_STEPS_FINE : NO_IID_STEPS; if (env == 0) { aPrevIidIndex = h_ps_d->specificTo.mpeg.aIidPrevFrameIndex; aPrevIccIndex = h_ps_d->specificTo.mpeg.aIccPrevFrameIndex; } else { aPrevIidIndex = pBsData->aaIidIndex[env - 1]; aPrevIccIndex = pBsData->aaIccIndex[env - 1]; } deltaDecodeArray(pBsData->bEnableIid, pBsData->aaIidIndex[env], aPrevIidIndex, pBsData->abIidDtFlag[env], FDK_sbrDecoder_aNoIidBins[pBsData->freqResIid], (pBsData->freqResIid) ? 1 : 2, -noIidSteps, noIidSteps); deltaDecodeArray(pBsData->bEnableIcc, pBsData->aaIccIndex[env], aPrevIccIndex, pBsData->abIccDtFlag[env], FDK_sbrDecoder_aNoIccBins[pBsData->freqResIcc], (pBsData->freqResIcc) ? 1 : 2, 0, NO_ICC_STEPS - 1); } /* for (env=0; envnoEnv; env++) */ /* handling of FIX noEnv=0 */ if (pBsData->noEnv == 0) { /* set noEnv=1, keep last parameters or force 0 if not enabled */ pBsData->noEnv = 1; if (pBsData->bEnableIid) { pBsData->bFineIidQ = h_ps_d->specificTo.mpeg.bPrevFrameFineIidQ; pBsData->freqResIid = h_ps_d->specificTo.mpeg.prevFreqResIid; for (gr = 0; gr < NO_HI_RES_IID_BINS; gr++) { pBsData->aaIidIndex[pBsData->noEnv - 1][gr] = h_ps_d->specificTo.mpeg.aIidPrevFrameIndex[gr]; } } else { for (gr = 0; gr < NO_HI_RES_IID_BINS; gr++) { pBsData->aaIidIndex[pBsData->noEnv - 1][gr] = 0; } } if (pBsData->bEnableIcc) { pBsData->freqResIcc = h_ps_d->specificTo.mpeg.prevFreqResIcc; for (gr = 0; gr < NO_HI_RES_ICC_BINS; gr++) { pBsData->aaIccIndex[pBsData->noEnv - 1][gr] = h_ps_d->specificTo.mpeg.aIccPrevFrameIndex[gr]; } } else { for (gr = 0; gr < NO_HI_RES_ICC_BINS; gr++) { pBsData->aaIccIndex[pBsData->noEnv - 1][gr] = 0; } } } /* Update previous frame Iid quantization */ h_ps_d->specificTo.mpeg.bPrevFrameFineIidQ = pBsData->bFineIidQ; /* Update previous frequency resolution for IID */ h_ps_d->specificTo.mpeg.prevFreqResIid = pBsData->freqResIid; /* Update previous frequency resolution for ICC */ h_ps_d->specificTo.mpeg.prevFreqResIcc = pBsData->freqResIcc; /* Update previous frame index buffers */ for (gr = 0; gr < NO_HI_RES_IID_BINS; gr++) { h_ps_d->specificTo.mpeg.aIidPrevFrameIndex[gr] = pBsData->aaIidIndex[pBsData->noEnv - 1][gr]; } for (gr = 0; gr < NO_HI_RES_ICC_BINS; gr++) { h_ps_d->specificTo.mpeg.aIccPrevFrameIndex[gr] = pBsData->aaIccIndex[pBsData->noEnv - 1][gr]; } /* PS data from bitstream (if avail) was decoded now */ h_ps_d->bPsDataAvail[h_ps_d->processSlot] = ppt_none; /* handling of env borders for FIX & VAR */ if (pBsData->bFrameClass == 0) { /* FIX_BORDERS NoEnv=0,1,2,4 */ pBsData->aEnvStartStop[0] = 0; for (env = 1; env < pBsData->noEnv; env++) { pBsData->aEnvStartStop[env] = (env * h_ps_d->noSubSamples) / pBsData->noEnv; } pBsData->aEnvStartStop[pBsData->noEnv] = h_ps_d->noSubSamples; /* 1024 (32 slots) env borders: 0, 8, 16, 24, 32 */ /* 960 (30 slots) env borders: 0, 7, 15, 22, 30 */ } else { /* if (h_ps_d->bFrameClass == 0) */ /* VAR_BORDERS NoEnv=1,2,3,4 */ pBsData->aEnvStartStop[0] = 0; /* handle case aEnvStartStop[noEnv]aEnvStartStop[pBsData->noEnv] < h_ps_d->noSubSamples) { for (gr = 0; gr < NO_HI_RES_IID_BINS; gr++) { pBsData->aaIidIndex[pBsData->noEnv][gr] = pBsData->aaIidIndex[pBsData->noEnv - 1][gr]; } for (gr = 0; gr < NO_HI_RES_ICC_BINS; gr++) { pBsData->aaIccIndex[pBsData->noEnv][gr] = pBsData->aaIccIndex[pBsData->noEnv - 1][gr]; } pBsData->noEnv++; pBsData->aEnvStartStop[pBsData->noEnv] = h_ps_d->noSubSamples; } /* enforce strictly monotonic increasing borders */ for (env = 1; env < pBsData->noEnv; env++) { UCHAR thr; thr = (UCHAR)h_ps_d->noSubSamples - (pBsData->noEnv - env); if (pBsData->aEnvStartStop[env] > thr) { pBsData->aEnvStartStop[env] = thr; } else { thr = pBsData->aEnvStartStop[env - 1] + 1; if (pBsData->aEnvStartStop[env] < thr) { pBsData->aEnvStartStop[env] = thr; } } } } /* if (h_ps_d->bFrameClass == 0) ... else */ /* copy data prior to possible 20<->34 in-place mapping */ for (env = 0; env < pBsData->noEnv; env++) { UCHAR i; for (i = 0; i < NO_HI_RES_IID_BINS; i++) { h_ps_d->specificTo.mpeg.pCoef->aaIidIndexMapped[env][i] = pBsData->aaIidIndex[env][i]; } for (i = 0; i < NO_HI_RES_ICC_BINS; i++) { h_ps_d->specificTo.mpeg.pCoef->aaIccIndexMapped[env][i] = pBsData->aaIccIndex[env][i]; } } /* MPEG baseline PS */ /* Baseline version of PS always uses the hybrid filter structure with 20 * stereo bands. */ /* If ICC/IID parameters for 34 stereo bands are decoded they have to be * mapped to 20 */ /* stereo bands. */ /* Additionaly the IPD/OPD parameters won't be used. */ for (env = 0; env < pBsData->noEnv; env++) { if (pBsData->freqResIid == 2) map34IndexTo20(h_ps_d->specificTo.mpeg.pCoef->aaIidIndexMapped[env], NO_HI_RES_IID_BINS); if (pBsData->freqResIcc == 2) map34IndexTo20(h_ps_d->specificTo.mpeg.pCoef->aaIccIndexMapped[env], NO_HI_RES_ICC_BINS); /* IPD/OPD is disabled in baseline version and thus was removed here */ } return (1); } /***************************************************************************/ /*! \brief Reads parametric stereo data from bitstream \return ****************************************************************************/ unsigned int ReadPsData( HANDLE_PS_DEC h_ps_d, /*!< handle to struct PS_DEC */ HANDLE_FDK_BITSTREAM hBitBuf, /*!< handle to struct BIT_BUF */ int nBitsLeft /*!< max number of bits available */ ) { MPEG_PS_BS_DATA *pBsData; UCHAR gr, env; SCHAR dtFlag; INT startbits; Huffman CurrentTable; SCHAR bEnableHeader; if (!h_ps_d) return 0; pBsData = &h_ps_d->bsData[h_ps_d->bsReadSlot].mpeg; if (h_ps_d->bsReadSlot != h_ps_d->bsLastSlot) { /* Copy last header data */ FDKmemcpy(pBsData, &h_ps_d->bsData[h_ps_d->bsLastSlot].mpeg, sizeof(MPEG_PS_BS_DATA)); } startbits = (INT)FDKgetValidBits(hBitBuf); bEnableHeader = (SCHAR)FDKreadBits(hBitBuf, 1); /* Read header */ if (bEnableHeader) { pBsData->bPsHeaderValid = 1; pBsData->bEnableIid = (UCHAR)FDKreadBits(hBitBuf, 1); if (pBsData->bEnableIid) { pBsData->modeIid = (UCHAR)FDKreadBits(hBitBuf, 3); } pBsData->bEnableIcc = (UCHAR)FDKreadBits(hBitBuf, 1); if (pBsData->bEnableIcc) { pBsData->modeIcc = (UCHAR)FDKreadBits(hBitBuf, 3); } pBsData->bEnableExt = (UCHAR)FDKreadBits(hBitBuf, 1); } pBsData->bFrameClass = (UCHAR)FDKreadBits(hBitBuf, 1); if (pBsData->bFrameClass == 0) { /* FIX_BORDERS NoEnv=0,1,2,4 */ pBsData->noEnv = FDK_sbrDecoder_aFixNoEnvDecode[(UCHAR)FDKreadBits(hBitBuf, 2)]; /* all additional handling of env borders is now in DecodePs() */ } else { /* VAR_BORDERS NoEnv=1,2,3,4 */ pBsData->noEnv = 1 + (UCHAR)FDKreadBits(hBitBuf, 2); for (env = 1; env < pBsData->noEnv + 1; env++) pBsData->aEnvStartStop[env] = ((UCHAR)FDKreadBits(hBitBuf, 5)) + 1; /* all additional handling of env borders is now in DecodePs() */ } /* verify that IID & ICC modes (quant grid, freq res) are supported */ if ((pBsData->modeIid > 5) || (pBsData->modeIcc > 5)) { /* no useful PS data could be read from bitstream */ h_ps_d->bPsDataAvail[h_ps_d->bsReadSlot] = ppt_none; /* discard all remaining bits */ nBitsLeft -= startbits - (INT)FDKgetValidBits(hBitBuf); while (nBitsLeft > 0) { int i = nBitsLeft; if (i > 8) { i = 8; } FDKreadBits(hBitBuf, i); nBitsLeft -= i; } return (UINT)(startbits - (INT)FDKgetValidBits(hBitBuf)); } if (pBsData->modeIid > 2) { pBsData->freqResIid = pBsData->modeIid - 3; pBsData->bFineIidQ = 1; } else { pBsData->freqResIid = pBsData->modeIid; pBsData->bFineIidQ = 0; } if (pBsData->modeIcc > 2) { pBsData->freqResIcc = pBsData->modeIcc - 3; } else { pBsData->freqResIcc = pBsData->modeIcc; } /* Extract IID data */ if (pBsData->bEnableIid) { for (env = 0; env < pBsData->noEnv; env++) { dtFlag = (SCHAR)FDKreadBits(hBitBuf, 1); if (!dtFlag) { if (pBsData->bFineIidQ) CurrentTable = (Huffman)&aBookPsIidFineFreqDecode; else CurrentTable = (Huffman)&aBookPsIidFreqDecode; } else { if (pBsData->bFineIidQ) CurrentTable = (Huffman)&aBookPsIidFineTimeDecode; else CurrentTable = (Huffman)&aBookPsIidTimeDecode; } for (gr = 0; gr < FDK_sbrDecoder_aNoIidBins[pBsData->freqResIid]; gr++) pBsData->aaIidIndex[env][gr] = decode_huff_cw(CurrentTable, hBitBuf, NULL); pBsData->abIidDtFlag[env] = dtFlag; } } /* Extract ICC data */ if (pBsData->bEnableIcc) { for (env = 0; env < pBsData->noEnv; env++) { dtFlag = (SCHAR)FDKreadBits(hBitBuf, 1); if (!dtFlag) CurrentTable = (Huffman)&aBookPsIccFreqDecode; else CurrentTable = (Huffman)&aBookPsIccTimeDecode; for (gr = 0; gr < FDK_sbrDecoder_aNoIccBins[pBsData->freqResIcc]; gr++) pBsData->aaIccIndex[env][gr] = decode_huff_cw(CurrentTable, hBitBuf, NULL); pBsData->abIccDtFlag[env] = dtFlag; } } if (pBsData->bEnableExt) { /*! Decoders that support only the baseline version of the PS tool are allowed to ignore the IPD/OPD data, but according header data has to be parsed. ISO/IEC 14496-3 Subpart 8 Annex 4 */ int cnt = FDKreadBits(hBitBuf, PS_EXTENSION_SIZE_BITS); if (cnt == (1 << PS_EXTENSION_SIZE_BITS) - 1) { cnt += FDKreadBits(hBitBuf, PS_EXTENSION_ESC_COUNT_BITS); } while (cnt--) FDKreadBits(hBitBuf, 8); } /* new PS data was read from bitstream */ h_ps_d->bPsDataAvail[h_ps_d->bsReadSlot] = ppt_mpeg; return (startbits - (INT)FDKgetValidBits(hBitBuf)); } fdk-aac-2.0.2/libSBRdec/src/psbitdec.h000066400000000000000000000120121404225310700173210ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Description: *******************************************************************************/ #ifndef PSBITDEC_H #define PSBITDEC_H #include "sbrdecoder.h" #include "psdec.h" unsigned int ReadPsData(struct PS_DEC *h_ps_d, HANDLE_FDK_BITSTREAM hBs, int nBitsLeft); int DecodePs(struct PS_DEC *h_ps_d, const UCHAR frameError, PS_DEC_COEFFICIENTS *pCoef); #endif /* PSBITDEC_H */ fdk-aac-2.0.2/libSBRdec/src/psdec.cpp000066400000000000000000000653531404225310700171750ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief parametric stereo decoder */ #include "psdec.h" #include "FDK_bitbuffer.h" #include "sbr_rom.h" #include "sbr_ram.h" #include "FDK_tools_rom.h" #include "genericStds.h" #include "FDK_trigFcts.h" /********************************************************************/ /* MLQUAL DEFINES */ /********************************************************************/ #define FRACT_ZERO FRACT_BITS - 1 /********************************************************************/ SBR_ERROR ResetPsDec(HANDLE_PS_DEC h_ps_d); /***** HELPERS *****/ /***************************************************************************/ /*! \brief Creates one instance of the PS_DEC struct \return Error info ****************************************************************************/ int CreatePsDec(HANDLE_PS_DEC *h_PS_DEC, /*!< pointer to the module state */ int aacSamplesPerFrame) { SBR_ERROR errorInfo = SBRDEC_OK; HANDLE_PS_DEC h_ps_d; int i; if (*h_PS_DEC == NULL) { /* Get ps dec ram */ h_ps_d = GetRam_ps_dec(); if (h_ps_d == NULL) { goto bail; } } else { /* Reset an open instance */ h_ps_d = *h_PS_DEC; } /* * Create Analysis Hybrid filterbank. */ FDKhybridAnalysisOpen(&h_ps_d->specificTo.mpeg.hybridAnalysis, h_ps_d->specificTo.mpeg.pHybridAnaStatesLFdmx, sizeof(h_ps_d->specificTo.mpeg.pHybridAnaStatesLFdmx), NULL, 0); /* initialisation */ switch (aacSamplesPerFrame) { case 960: h_ps_d->noSubSamples = 30; /* col */ break; case 1024: h_ps_d->noSubSamples = 32; /* col */ break; default: h_ps_d->noSubSamples = -1; break; } if (h_ps_d->noSubSamples > MAX_NUM_COL || h_ps_d->noSubSamples <= 0) { goto bail; } h_ps_d->noChannels = NO_QMF_CHANNELS; /* row */ h_ps_d->psDecodedPrv = 0; h_ps_d->procFrameBased = -1; for (i = 0; i < (1) + 1; i++) { h_ps_d->bPsDataAvail[i] = ppt_none; } { int error; error = FDKdecorrelateOpen(&(h_ps_d->specificTo.mpeg.apDecor), h_ps_d->specificTo.mpeg.decorrBufferCplx, (2 * ((825) + (373)))); if (error) goto bail; } for (i = 0; i < (1) + 1; i++) { FDKmemclear(&h_ps_d->bsData[i].mpeg, sizeof(MPEG_PS_BS_DATA)); } errorInfo = ResetPsDec(h_ps_d); if (errorInfo != SBRDEC_OK) goto bail; *h_PS_DEC = h_ps_d; return 0; bail: if (h_ps_d != NULL) { DeletePsDec(&h_ps_d); } return -1; } /*END CreatePsDec */ /***************************************************************************/ /*! \brief Delete one instance of the PS_DEC struct \return Error info ****************************************************************************/ int DeletePsDec(HANDLE_PS_DEC *h_PS_DEC) /*!< pointer to the module state */ { if (*h_PS_DEC == NULL) { return -1; } { HANDLE_PS_DEC h_ps_d = *h_PS_DEC; FDKdecorrelateClose(&(h_ps_d->specificTo.mpeg.apDecor)); } FreeRam_ps_dec(h_PS_DEC); return 0; } /*END DeletePsDec */ /***************************************************************************/ /*! \brief resets some values of the PS handle to default states \return ****************************************************************************/ SBR_ERROR ResetPsDec(HANDLE_PS_DEC h_ps_d) /*!< pointer to the module state */ { SBR_ERROR errorInfo = SBRDEC_OK; INT i; /* explicitly init state variables to safe values (until first ps header * arrives) */ h_ps_d->specificTo.mpeg.lastUsb = 0; /* * Initialize Analysis Hybrid filterbank. */ FDKhybridAnalysisInit(&h_ps_d->specificTo.mpeg.hybridAnalysis, THREE_TO_TEN, NO_QMF_BANDS_HYBRID20, NO_QMF_BANDS_HYBRID20, 1); /* * Initialize Synthesis Hybrid filterbank. */ for (i = 0; i < 2; i++) { FDKhybridSynthesisInit(&h_ps_d->specificTo.mpeg.hybridSynthesis[i], THREE_TO_TEN, NO_QMF_CHANNELS, NO_QMF_CHANNELS); } { INT error; error = FDKdecorrelateInit(&h_ps_d->specificTo.mpeg.apDecor, 71, DECORR_PS, DUCKER_AUTOMATIC, 0, 0, 0, 0, 1, /* isLegacyPS */ 1); if (error) return SBRDEC_NOT_INITIALIZED; } for (i = 0; i < NO_IID_GROUPS; i++) { h_ps_d->specificTo.mpeg.h11rPrev[i] = FL2FXCONST_DBL(0.5f); h_ps_d->specificTo.mpeg.h12rPrev[i] = FL2FXCONST_DBL(0.5f); } FDKmemclear(h_ps_d->specificTo.mpeg.h21rPrev, sizeof(h_ps_d->specificTo.mpeg.h21rPrev)); FDKmemclear(h_ps_d->specificTo.mpeg.h22rPrev, sizeof(h_ps_d->specificTo.mpeg.h22rPrev)); return errorInfo; } /***************************************************************************/ /*! \brief Feed delaylines when parametric stereo is switched on. \return ****************************************************************************/ void PreparePsProcessing(HANDLE_PS_DEC h_ps_d, const FIXP_DBL *const *const rIntBufferLeft, const FIXP_DBL *const *const iIntBufferLeft, const int scaleFactorLowBand) { if (h_ps_d->procFrameBased == 1) /* If we have switched from frame to slot based processing */ { /* fill hybrid delay buffer. */ int i, j; for (i = 0; i < HYBRID_FILTER_DELAY; i++) { FIXP_DBL qmfInputData[2][NO_QMF_BANDS_HYBRID20]; FIXP_DBL hybridOutputData[2][NO_SUB_QMF_CHANNELS]; for (j = 0; j < NO_QMF_BANDS_HYBRID20; j++) { qmfInputData[0][j] = scaleValue(rIntBufferLeft[i][j], scaleFactorLowBand); qmfInputData[1][j] = scaleValue(iIntBufferLeft[i][j], scaleFactorLowBand); } FDKhybridAnalysisApply(&h_ps_d->specificTo.mpeg.hybridAnalysis, qmfInputData[0], qmfInputData[1], hybridOutputData[0], hybridOutputData[1]); } h_ps_d->procFrameBased = 0; /* switch to slot based processing. */ } /* procFrameBased==1 */ } void initSlotBasedRotation( HANDLE_PS_DEC h_ps_d, /*!< pointer to the module state */ int env, int usb) { INT group = 0; INT bin = 0; INT noIidSteps; FIXP_SGL invL; FIXP_DBL ScaleL, ScaleR; FIXP_DBL Alpha, Beta; FIXP_DBL h11r, h12r, h21r, h22r; const FIXP_DBL *PScaleFactors; if (h_ps_d->bsData[h_ps_d->processSlot].mpeg.bFineIidQ) { PScaleFactors = ScaleFactorsFine; /* values are shiftet right by one */ noIidSteps = NO_IID_STEPS_FINE; } else { PScaleFactors = ScaleFactors; /* values are shiftet right by one */ noIidSteps = NO_IID_STEPS; } /* dequantize and decode */ for (group = 0; group < NO_IID_GROUPS; group++) { bin = bins2groupMap20[group]; /*!

type 'A' rotation

mixing procedure R_a, used in baseline version
Scale-factor vectors c1 and c2 are precalculated in initPsTables () and stored in scaleFactors[] and scaleFactorsFine[] = pScaleFactors []. From the linearized IID parameters (intensity differences), two scale factors are calculated. They are used to obtain the coefficients h11... h22. */ /* ScaleR and ScaleL are scaled by 1 shift right */ ScaleR = PScaleFactors[noIidSteps + h_ps_d->specificTo.mpeg.pCoef ->aaIidIndexMapped[env][bin]]; ScaleL = PScaleFactors[noIidSteps - h_ps_d->specificTo.mpeg.pCoef ->aaIidIndexMapped[env][bin]]; Beta = fMult( fMult(Alphas[h_ps_d->specificTo.mpeg.pCoef->aaIccIndexMapped[env][bin]], (ScaleR - ScaleL)), FIXP_SQRT05); Alpha = Alphas[h_ps_d->specificTo.mpeg.pCoef->aaIccIndexMapped[env][bin]] >> 1; /* Alpha and Beta are now both scaled by 2 shifts right */ /* calculate the coefficients h11... h22 from scale-factors and ICC * parameters */ /* h values are scaled by 1 shift right */ { FIXP_DBL trigData[4]; inline_fixp_cos_sin(Beta + Alpha, Beta - Alpha, 2, trigData); h11r = fMult(ScaleL, trigData[0]); h12r = fMult(ScaleR, trigData[2]); h21r = fMult(ScaleL, trigData[1]); h22r = fMult(ScaleR, trigData[3]); } /*****************************************************************************************/ /* Interpolation of the matrices H11... H22: */ /* */ /* H11(k,n) = H11(k,n[e]) + (n-n[e]) * (H11(k,n[e+1] - H11(k,n[e])) / * (n[e+1] - n[e]) */ /* ... */ /*****************************************************************************************/ /* invL = 1/(length of envelope) */ invL = FX_DBL2FX_SGL(GetInvInt( h_ps_d->bsData[h_ps_d->processSlot].mpeg.aEnvStartStop[env + 1] - h_ps_d->bsData[h_ps_d->processSlot].mpeg.aEnvStartStop[env])); h_ps_d->specificTo.mpeg.pCoef->H11r[group] = h_ps_d->specificTo.mpeg.h11rPrev[group]; h_ps_d->specificTo.mpeg.pCoef->H12r[group] = h_ps_d->specificTo.mpeg.h12rPrev[group]; h_ps_d->specificTo.mpeg.pCoef->H21r[group] = h_ps_d->specificTo.mpeg.h21rPrev[group]; h_ps_d->specificTo.mpeg.pCoef->H22r[group] = h_ps_d->specificTo.mpeg.h22rPrev[group]; h_ps_d->specificTo.mpeg.pCoef->DeltaH11r[group] = fMult(h11r - h_ps_d->specificTo.mpeg.pCoef->H11r[group], invL); h_ps_d->specificTo.mpeg.pCoef->DeltaH12r[group] = fMult(h12r - h_ps_d->specificTo.mpeg.pCoef->H12r[group], invL); h_ps_d->specificTo.mpeg.pCoef->DeltaH21r[group] = fMult(h21r - h_ps_d->specificTo.mpeg.pCoef->H21r[group], invL); h_ps_d->specificTo.mpeg.pCoef->DeltaH22r[group] = fMult(h22r - h_ps_d->specificTo.mpeg.pCoef->H22r[group], invL); /* update prev coefficients for interpolation in next envelope */ h_ps_d->specificTo.mpeg.h11rPrev[group] = h11r; h_ps_d->specificTo.mpeg.h12rPrev[group] = h12r; h_ps_d->specificTo.mpeg.h21rPrev[group] = h21r; h_ps_d->specificTo.mpeg.h22rPrev[group] = h22r; } /* group loop */ } static const UCHAR groupTable[NO_IID_GROUPS + 1] = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 18, 21, 25, 30, 42, 71}; static void applySlotBasedRotation( HANDLE_PS_DEC h_ps_d, /*!< pointer to the module state */ FIXP_DBL *mHybridRealLeft, /*!< hybrid values real left */ FIXP_DBL *mHybridImagLeft, /*!< hybrid values imag left */ FIXP_DBL *mHybridRealRight, /*!< hybrid values real right */ FIXP_DBL *mHybridImagRight /*!< hybrid values imag right */ ) { INT group; INT subband; /**********************************************************************************************/ /*!

Mapping

The number of stereo bands that is actually used depends on the number of availble parameters for IID and ICC: 
 nr. of IID para.| nr. of ICC para.
  | nr. of Stereo bands
   ----------------|------------------|-------------------
     10,20         |     10,20        |        20
     10,20         |     34           |        34
     34            |     10,20        |        34
     34            |     34           |        34
In the case the number of parameters for IIS and ICC differs from the number of stereo bands, a mapping from the lower number to the higher number of parameters is applied. Index mapping of IID and ICC parameters is already done in psbitdec.cpp. Further mapping is not needed here in baseline version. **********************************************************************************************/ /************************************************************************************************/ /*!

Mixing

To generate the QMF subband signals for the subband samples n = n[e]+1 ,,, n_[e+1] the parameters at position n[e] and n[e+1] are required as well as the subband domain signals s_k(n) and d_k(n) for n = n[e]+1... n_[e+1]. n[e] represents the start position for envelope e. The border positions n[e] are handled in DecodePS(). The stereo sub subband signals are constructed as: 
  l_k(n) = H11(k,n) s_k(n) + H21(k,n) d_k(n)
  r_k(n) = H21(k,n) s_k(n) + H22(k,n) d_k(n)
In order to obtain the matrices H11(k,n)... H22 (k,n), the vectors h11(b)... h22(b) need to be calculated first (b: parameter index). Depending on ICC mode either mixing procedure R_a or R_b is used for that. For both procedures, the parameters for parameter position n[e+1] is used. ************************************************************************************************/ /************************************************************************************************/ /*!

Phase parameters

With disabled phase parameters (which is the case in baseline version), the H-matrices are just calculated by: 
  H11(k,n[e+1] = h11(b(k))
  (...)
  b(k): parameter index according to mapping table

Processing of the samples in the sub subbands

this loop includes the interpolation of the coefficients Hxx ************************************************************************************************/ /******************************************************/ /* construct stereo sub subband signals according to: */ /* */ /* l_k(n) = H11(k,n) s_k(n) + H21(k,n) d_k(n) */ /* r_k(n) = H12(k,n) s_k(n) + H22(k,n) d_k(n) */ /******************************************************/ PS_DEC_COEFFICIENTS *pCoef = h_ps_d->specificTo.mpeg.pCoef; for (group = 0; group < NO_IID_GROUPS; group++) { pCoef->H11r[group] += pCoef->DeltaH11r[group]; pCoef->H12r[group] += pCoef->DeltaH12r[group]; pCoef->H21r[group] += pCoef->DeltaH21r[group]; pCoef->H22r[group] += pCoef->DeltaH22r[group]; const int start = groupTable[group]; const int stop = groupTable[group + 1]; for (subband = start; subband < stop; subband++) { FIXP_DBL tmpLeft = fMultAdd(fMultDiv2(pCoef->H11r[group], mHybridRealLeft[subband]), pCoef->H21r[group], mHybridRealRight[subband]); FIXP_DBL tmpRight = fMultAdd(fMultDiv2(pCoef->H12r[group], mHybridRealLeft[subband]), pCoef->H22r[group], mHybridRealRight[subband]); mHybridRealLeft[subband] = tmpLeft; mHybridRealRight[subband] = tmpRight; tmpLeft = fMultAdd(fMultDiv2(pCoef->H11r[group], mHybridImagLeft[subband]), pCoef->H21r[group], mHybridImagRight[subband]); tmpRight = fMultAdd(fMultDiv2(pCoef->H12r[group], mHybridImagLeft[subband]), pCoef->H22r[group], mHybridImagRight[subband]); mHybridImagLeft[subband] = tmpLeft; mHybridImagRight[subband] = tmpRight; } /* subband */ } } /***************************************************************************/ /*! \brief Applies IID, ICC, IPD and OPD parameters to the current frame. \return none ****************************************************************************/ void ApplyPsSlot( HANDLE_PS_DEC h_ps_d, /*!< handle PS_DEC*/ FIXP_DBL **rIntBufferLeft, /*!< real bands left qmf channel (38x64) */ FIXP_DBL **iIntBufferLeft, /*!< imag bands left qmf channel (38x64) */ FIXP_DBL *rIntBufferRight, /*!< real bands right qmf channel (38x64) */ FIXP_DBL *iIntBufferRight, /*!< imag bands right qmf channel (38x64) */ const int scaleFactorLowBand_no_ov, const int scaleFactorLowBand, const int scaleFactorHighBand, const int lsb, const int usb) { /*! The 64-band QMF representation of the monaural signal generated by the SBR tool is used as input of the PS tool. After the PS processing, the outputs of the left and right hybrid synthesis filterbanks are used to generate the stereo output signal. 

           -------------            ----------            -------------
          | Hybrid      | M_n[k,m] |          | L_n[k,m] | Hybrid      | l[n]
 m[n] --->| analysis    |--------->|          |--------->| synthesis   |----->
           -------------           | Stereo   |           -------------
                 |                 | recon-   |
                 |                 | stuction |
                \|/                |          |
           -------------           |          |
          | De-         | D_n[k,m] |          |
          | correlation |--------->|          |
           -------------           |          |           -------------
                                   |          | R_n[k,m] | Hybrid      | r[n]
                                   |          |--------->| synthesis   |----->
 IID, ICC ------------------------>|          |          | filter bank |
(IPD, OPD)                          ----------            -------------

m[n]:      QMF represantation of the mono input
M_n[k,m]:  (sub-)sub-band domain signals of the mono input
D_n[k,m]:  decorrelated (sub-)sub-band domain signals
L_n[k,m]:  (sub-)sub-band domain signals of the left output
R_n[k,m]:  (sub-)sub-band domain signals of the right output
l[n],r[n]: left/right output signals
*/ #define NO_HYBRID_DATA_BANDS (71) int i; FIXP_DBL qmfInputData[2][NO_QMF_BANDS_HYBRID20]; FIXP_DBL *hybridData[2][2]; C_ALLOC_SCRATCH_START(pHybridData, FIXP_DBL, 4 * NO_HYBRID_DATA_BANDS); hybridData[0][0] = pHybridData + 0 * NO_HYBRID_DATA_BANDS; /* left real hybrid data */ hybridData[0][1] = pHybridData + 1 * NO_HYBRID_DATA_BANDS; /* left imag hybrid data */ hybridData[1][0] = pHybridData + 2 * NO_HYBRID_DATA_BANDS; /* right real hybrid data */ hybridData[1][1] = pHybridData + 3 * NO_HYBRID_DATA_BANDS; /* right imag hybrid data */ /*! Hybrid analysis filterbank: The lower 3 (5) of the 64 QMF subbands are further split to provide better frequency resolution. for PS processing. For the 10 and 20 stereo bands configuration, the QMF band H_0(w) is split up into 8 (sub-) sub-bands and the QMF bands H_1(w) and H_2(w) are spit into 2 (sub-) 4th. (See figures 8.20 and 8.22 of ISO/IEC 14496-3:2001/FDAM 2:2004(E) ) */ /* * Hybrid analysis. */ /* Get qmf input data and apply descaling */ for (i = 0; i < NO_QMF_BANDS_HYBRID20; i++) { qmfInputData[0][i] = scaleValue(rIntBufferLeft[HYBRID_FILTER_DELAY][i], scaleFactorLowBand_no_ov); qmfInputData[1][i] = scaleValue(iIntBufferLeft[HYBRID_FILTER_DELAY][i], scaleFactorLowBand_no_ov); } /* LF - part */ FDKhybridAnalysisApply(&h_ps_d->specificTo.mpeg.hybridAnalysis, qmfInputData[0], qmfInputData[1], hybridData[0][0], hybridData[0][1]); /* HF - part */ /* bands up to lsb */ scaleValues(&hybridData[0][0][NO_SUB_QMF_CHANNELS - 2], &rIntBufferLeft[0][NO_QMF_BANDS_HYBRID20], lsb - NO_QMF_BANDS_HYBRID20, scaleFactorLowBand); scaleValues(&hybridData[0][1][NO_SUB_QMF_CHANNELS - 2], &iIntBufferLeft[0][NO_QMF_BANDS_HYBRID20], lsb - NO_QMF_BANDS_HYBRID20, scaleFactorLowBand); /* bands from lsb to usb */ scaleValues(&hybridData[0][0][lsb + (NO_SUB_QMF_CHANNELS - 2 - NO_QMF_BANDS_HYBRID20)], &rIntBufferLeft[0][lsb], usb - lsb, scaleFactorHighBand); scaleValues(&hybridData[0][1][lsb + (NO_SUB_QMF_CHANNELS - 2 - NO_QMF_BANDS_HYBRID20)], &iIntBufferLeft[0][lsb], usb - lsb, scaleFactorHighBand); /* bands from usb to NO_SUB_QMF_CHANNELS which should be zero for non-overlap slots but can be non-zero for overlap slots */ FDKmemcpy( &hybridData[0][0] [usb + (NO_SUB_QMF_CHANNELS - 2 - NO_QMF_BANDS_HYBRID20)], &rIntBufferLeft[0][usb], sizeof(FIXP_DBL) * (NO_QMF_CHANNELS - usb)); FDKmemcpy( &hybridData[0][1] [usb + (NO_SUB_QMF_CHANNELS - 2 - NO_QMF_BANDS_HYBRID20)], &iIntBufferLeft[0][usb], sizeof(FIXP_DBL) * (NO_QMF_CHANNELS - usb)); /*! Decorrelation: By means of all-pass filtering and delaying, the (sub-)sub-band samples s_k(n) are converted into de-correlated (sub-)sub-band samples d_k(n). - k: frequency in hybrid spectrum - n: time index */ FDKdecorrelateApply(&h_ps_d->specificTo.mpeg.apDecor, &hybridData[0][0][0], /* left real hybrid data */ &hybridData[0][1][0], /* left imag hybrid data */ &hybridData[1][0][0], /* right real hybrid data */ &hybridData[1][1][0], /* right imag hybrid data */ 0 /* startHybBand */ ); /*! Stereo Processing: The sets of (sub-)sub-band samples s_k(n) and d_k(n) are processed according to the stereo cues which are defined per stereo band. */ applySlotBasedRotation(h_ps_d, &hybridData[0][0][0], /* left real hybrid data */ &hybridData[0][1][0], /* left imag hybrid data */ &hybridData[1][0][0], /* right real hybrid data */ &hybridData[1][1][0] /* right imag hybrid data */ ); /*! Hybrid synthesis filterbank: The stereo processed hybrid subband signals l_k(n) and r_k(n) are fed into the hybrid synthesis filterbanks which are identical to the 64 complex synthesis filterbank of the SBR tool. The input to the filterbank are slots of 64 QMF samples. For each slot the filterbank outputs one block of 64 samples of one reconstructed stereo channel. The hybrid synthesis filterbank is computed seperatly for the left and right channel. */ /* * Hybrid synthesis. */ for (i = 0; i < 2; i++) { FDKhybridSynthesisApply( &h_ps_d->specificTo.mpeg.hybridSynthesis[i], hybridData[i][0], /* real hybrid data */ hybridData[i][1], /* imag hybrid data */ (i == 0) ? rIntBufferLeft[0] : rIntBufferRight, /* output real qmf buffer */ (i == 0) ? iIntBufferLeft[0] : iIntBufferRight /* output imag qmf buffer */ ); } /* free temporary hybrid qmf values of one timeslot */ C_ALLOC_SCRATCH_END(pHybridData, FIXP_DBL, 4 * NO_HYBRID_DATA_BANDS); } /* END ApplyPsSlot */ fdk-aac-2.0.2/libSBRdec/src/psdec.h000066400000000000000000000333141404225310700166320ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Sbr decoder */ #ifndef PSDEC_H #define PSDEC_H #include "sbrdecoder.h" #include "FDK_hybrid.h" #include "FDK_decorrelate.h" /* This PS decoder implements the baseline version. So it always uses the */ /* hybrid filter structure for 20 stereo bands and does not implemet IPD/OPD */ /* synthesis. The baseline version has to support the complete PS bitstream */ /* syntax. But IPD/OPD data is ignored and set to 0. If 34 stereo band config */ /* is used in the bitstream for IIS/ICC the decoded parameters are mapped to */ /* 20 stereo bands. */ #include "FDK_bitstream.h" #define SCAL_HEADROOM (2) #define PS_EXTENSION_SIZE_BITS (4) #define PS_EXTENSION_ESC_COUNT_BITS (8) #define NO_QMF_CHANNELS (64) #define MAX_NUM_COL (32) #define NO_QMF_BANDS_HYBRID20 (3) #define NO_SUB_QMF_CHANNELS (12) #define HYBRID_FILTER_DELAY (6) #define MAX_NO_PS_ENV (4 + 1) /* +1 needed for VAR_BORDER */ #define NO_HI_RES_BINS (34) #define NO_MID_RES_BINS (20) #define NO_LOW_RES_BINS (10) #define NO_HI_RES_IID_BINS (NO_HI_RES_BINS) #define NO_HI_RES_ICC_BINS (NO_HI_RES_BINS) #define NO_MID_RES_IID_BINS (NO_MID_RES_BINS) #define NO_MID_RES_ICC_BINS (NO_MID_RES_BINS) #define NO_LOW_RES_IID_BINS (NO_LOW_RES_BINS) #define NO_LOW_RES_ICC_BINS (NO_LOW_RES_BINS) #define SUBQMF_GROUPS (10) #define QMF_GROUPS (12) //#define SUBQMF_GROUPS_HI_RES ( 32 ) //#define QMF_GROUPS_HI_RES ( 18 ) #define NO_IID_GROUPS (SUBQMF_GROUPS + QMF_GROUPS) //#define NO_IID_GROUPS_HI_RES ( SUBQMF_GROUPS_HI_RES + // QMF_GROUPS_HI_RES ) #define NO_IID_STEPS (7) /* 1 .. + 7 */ #define NO_IID_STEPS_FINE (15) /* 1 .. +15 */ #define NO_ICC_STEPS (8) /* 0 .. + 7 */ #define NO_IID_LEVELS (2 * NO_IID_STEPS + 1) /* - 7 .. + 7 */ #define NO_IID_LEVELS_FINE (2 * NO_IID_STEPS_FINE + 1) /* -15 .. +15 */ #define NO_ICC_LEVELS (NO_ICC_STEPS) /* 0 .. + 7 */ #define FIXP_SQRT05 ((FIXP_DBL)0x5a827980) /* 1/SQRT2 */ struct PS_DEC_COEFFICIENTS { FIXP_DBL H11r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */ FIXP_DBL H12r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */ FIXP_DBL H21r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */ FIXP_DBL H22r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */ FIXP_DBL DeltaH11r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */ FIXP_DBL DeltaH12r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */ FIXP_DBL DeltaH21r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */ FIXP_DBL DeltaH22r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */ SCHAR aaIidIndexMapped[MAX_NO_PS_ENV] [NO_HI_RES_IID_BINS]; /*!< The mapped IID index for all envelopes and all IID bins */ SCHAR aaIccIndexMapped[MAX_NO_PS_ENV] [NO_HI_RES_ICC_BINS]; /*!< The mapped ICC index for all envelopes and all ICC bins */ }; typedef enum { ppt_none = 0, ppt_mpeg = 1, ppt_drm = 2 } PS_PAYLOAD_TYPE; typedef struct { UCHAR bPsHeaderValid; /*!< set if new header is available from bitstream */ UCHAR bEnableIid; /*!< One bit denoting the presence of IID parameters */ UCHAR bEnableIcc; /*!< One bit denoting the presence of ICC parameters */ UCHAR bEnableExt; /*!< The PS extension layer is enabled using the enable_ext bit. If it is set to %1 the IPD and OPD parameters are sent. If it is disabled, i.e. %0, the extension layer is skipped. */ UCHAR modeIid; /*!< The configuration of IID parameters (number of bands and quantisation grid, iid_quant) is determined by iid_mode. */ UCHAR modeIcc; /*!< The configuration of Inter-channel Coherence parameters (number of bands and quantisation grid) is determined by icc_mode. */ UCHAR freqResIid; /*!< 0=low, 1=mid or 2=high frequency resolution for iid */ UCHAR freqResIcc; /*!< 0=low, 1=mid or 2=high frequency resolution for icc */ UCHAR bFineIidQ; /*!< Use fine Iid quantisation. */ UCHAR bFrameClass; /*!< The frame_class bit determines whether the parameter positions of the current frame are uniformly spaced accross the frame or they are defined using the positions described by border_position. */ UCHAR noEnv; /*!< The number of envelopes per frame */ UCHAR aEnvStartStop[MAX_NO_PS_ENV + 1]; /*!< In case of variable parameter spacing the parameter positions are determined by border_position */ SCHAR abIidDtFlag[MAX_NO_PS_ENV]; /*!< Deltacoding time/freq flag for IID, 0 => freq */ SCHAR abIccDtFlag[MAX_NO_PS_ENV]; /*!< Deltacoding time/freq flag for ICC, 0 => freq */ SCHAR aaIidIndex[MAX_NO_PS_ENV] [NO_HI_RES_IID_BINS]; /*!< The IID index for all envelopes and all IID bins */ SCHAR aaIccIndex[MAX_NO_PS_ENV] [NO_HI_RES_ICC_BINS]; /*!< The ICC index for all envelopes and all ICC bins */ } MPEG_PS_BS_DATA; struct PS_DEC { SCHAR noSubSamples; SCHAR noChannels; SCHAR procFrameBased; /*!< Helper to detected switching from frame based to slot based processing */ PS_PAYLOAD_TYPE bPsDataAvail[(1) + 1]; /*!< set if new data available from bitstream */ UCHAR psDecodedPrv; /*!< set if PS has been processed in the last frame */ /* helpers for frame delay line */ UCHAR bsLastSlot; /*!< Index of last read slot. */ UCHAR bsReadSlot; /*!< Index of current read slot for additional delay. */ UCHAR processSlot; /*!< Index of current slot for processing (need for add. delay). */ union { /* Bitstream data */ MPEG_PS_BS_DATA mpeg; /*!< Struct containing all MPEG specific PS data from bitstream. */ } bsData[(1) + 1]; shouldBeUnion { /* Static data */ struct { SCHAR aIidPrevFrameIndex[NO_HI_RES_IID_BINS]; /*!< The IID index for previous frame */ SCHAR aIccPrevFrameIndex[NO_HI_RES_ICC_BINS]; /*!< The ICC index for previous frame */ UCHAR bPrevFrameFineIidQ; /*!< The IID quantization of the previous frame */ UCHAR prevFreqResIid; /*!< Frequency resolution for IID of the previous frame */ UCHAR prevFreqResIcc; /*!< Frequency resolution for ICC of the previous frame */ UCHAR lastUsb; /*!< uppermost WMF delay band of last frame */ FIXP_DBL pHybridAnaStatesLFdmx [2 * 13 * NO_QMF_BANDS_HYBRID20]; /*!< Memory used in hybrid analysis for filter states. */ FDK_ANA_HYB_FILTER hybridAnalysis; FDK_SYN_HYB_FILTER hybridSynthesis[2]; DECORR_DEC apDecor; /*!< Decorrelator instance. */ FIXP_DBL decorrBufferCplx[(2 * ((825) + (373)))]; FIXP_DBL h11rPrev[NO_IID_GROUPS]; /*!< previous calculated h(xy) coefficients */ FIXP_DBL h12rPrev[NO_IID_GROUPS]; /*!< previous calculated h(xy) coefficients */ FIXP_DBL h21rPrev[NO_IID_GROUPS]; /*!< previous calculated h(xy) coefficients */ FIXP_DBL h22rPrev[NO_IID_GROUPS]; /*!< previous calculated h(xy) coefficients */ PS_DEC_COEFFICIENTS *pCoef; /*!< temporal coefficients are on reusable scratch memory */ } mpeg; } specificTo; }; typedef struct PS_DEC *HANDLE_PS_DEC; int CreatePsDec(HANDLE_PS_DEC *h_PS_DEC, int aacSamplesPerFrame); int DeletePsDec(HANDLE_PS_DEC *h_PS_DEC); void PreparePsProcessing(HANDLE_PS_DEC h_ps_d, const FIXP_DBL *const *const rIntBufferLeft, const FIXP_DBL *const *const iIntBufferLeft, const int scaleFactorLowBand); void initSlotBasedRotation(HANDLE_PS_DEC h_ps_d, int env, int usb); void ApplyPsSlot( HANDLE_PS_DEC h_ps_d, /* parametric stereo decoder handle */ FIXP_DBL **rIntBufferLeft, /* real values of left qmf timeslot */ FIXP_DBL **iIntBufferLeft, /* imag values of left qmf timeslot */ FIXP_DBL *rIntBufferRight, /* real values of right qmf timeslot */ FIXP_DBL *iIntBufferRight, /* imag values of right qmf timeslot */ const int scaleFactorLowBand_no_ov, const int scaleFactorLowBand, const int scaleFactorHighBand, const int lsb, const int usb); #endif /* PSDEC_H */ fdk-aac-2.0.2/libSBRdec/src/psdec_drm.cpp000066400000000000000000000114521404225310700200260ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief parametric stereo decoder for Digital radio mondial */ #include "psdec_drm.h" fdk-aac-2.0.2/libSBRdec/src/psdec_drm.h000066400000000000000000000115551404225310700174770ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief parametric stereo decoder for digital radio mondial */ #ifndef PSDEC_DRM_H #define PSDEC_DRM_H #include "sbrdecoder.h" #endif /* PSDEC_DRM_H */ fdk-aac-2.0.2/libSBRdec/src/psdecrom_drm.cpp000066400000000000000000000114431404225310700205440ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief rom tables for Drm parametric stereo decoder */ #include "psdec_drm.h" fdk-aac-2.0.2/libSBRdec/src/pvc_dec.cpp000066400000000000000000000740241404225310700174750ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Matthias Hildenbrand Description: Decode Predictive Vector Coding Data *******************************************************************************/ #include "pvc_dec.h" /* PVC interal definitions */ #define PVC_DIVMODE_BITS 3 #define PVC_NSMODE_BITS 1 #define PVC_REUSEPVCID_BITS 1 #define PVC_PVCID_BITS 7 #define PVC_GRIDINFO_BITS 1 #define PVC_NQMFBAND 64 #define PVC_NBLOW 3 /* max. number of grouped QMF subbands below SBR range */ #define PVC_NTAB1 3 #define PVC_NTAB2 128 #define PVC_ID_NBIT 7 /* Exponent of pPvcStaticData->Esg and predictedEsg in dB domain. max(Esg) = 10*log10(2^15*2^15) = 90.30; min(Esg) = 10*log10(0.1) = -10 max of predicted Esg seems to be higher than 90dB but 7 Bit should be enough. */ #define PVC_ESG_EXP 7 #define LOG10FAC 0.752574989159953f /* == 10/log2(10) * 2^-2 */ #define LOG10FAC_INV 0.664385618977472f /* == log2(10)/10 * 2^1 */ RAM_ALIGN LNK_SECTION_CONSTDATA static const FIXP_SGL pvc_SC_16[] = { FX_DBL2FXCONST_SGL(0x14413695), FX_DBL2FXCONST_SGL(0x1434b6cb), FX_DBL2FXCONST_SGL(0x140f27c7), FX_DBL2FXCONST_SGL(0x13d0591d), FX_DBL2FXCONST_SGL(0x1377f502), FX_DBL2FXCONST_SGL(0x130577d6), FX_DBL2FXCONST_SGL(0x12782266), FX_DBL2FXCONST_SGL(0x11cee459), FX_DBL2FXCONST_SGL(0x11083a2a), FX_DBL2FXCONST_SGL(0x1021f5e9), FX_DBL2FXCONST_SGL(0x0f18e17c), FX_DBL2FXCONST_SGL(0x0de814ca), FX_DBL2FXCONST_SGL(0x0c87a568), FX_DBL2FXCONST_SGL(0x0ae9b167), FX_DBL2FXCONST_SGL(0x08f24226), FX_DBL2FXCONST_SGL(0x06575ed5), }; RAM_ALIGN LNK_SECTION_CONSTDATA static const FIXP_SGL pvc_SC_12[] = { FX_DBL2FXCONST_SGL(0x1aba6b3e), FX_DBL2FXCONST_SGL(0x1a9d164e), FX_DBL2FXCONST_SGL(0x1a44d56d), FX_DBL2FXCONST_SGL(0x19b0d742), FX_DBL2FXCONST_SGL(0x18df969a), FX_DBL2FXCONST_SGL(0x17ce91a0), FX_DBL2FXCONST_SGL(0x1679c3fa), FX_DBL2FXCONST_SGL(0x14daabfc), FX_DBL2FXCONST_SGL(0x12e65221), FX_DBL2FXCONST_SGL(0x1088d125), FX_DBL2FXCONST_SGL(0x0d9907b3), FX_DBL2FXCONST_SGL(0x09a80e9d), }; RAM_ALIGN LNK_SECTION_CONSTDATA static const FIXP_SGL pvc_SC_4[] = { FX_DBL2FXCONST_SGL(0x4ad6ab0f), FX_DBL2FXCONST_SGL(0x47ef0dbe), FX_DBL2FXCONST_SGL(0x3eee7496), FX_DBL2FXCONST_SGL(0x2e4bd29d), }; RAM_ALIGN LNK_SECTION_CONSTDATA static const FIXP_SGL pvc_SC_3[] = { FX_DBL2FXCONST_SGL(0x610dc761), FX_DBL2FXCONST_SGL(0x5a519a3d), FX_DBL2FXCONST_SGL(0x44a09e62), }; static const UCHAR g_3a_pvcTab1_mode1[PVC_NTAB1][PVC_NBLOW][PVC_NBHIGH_MODE1] = {{{0x4F, 0x5B, 0x57, 0x52, 0x4D, 0x65, 0x45, 0x57}, {0xF3, 0x0F, 0x18, 0x20, 0x19, 0x4F, 0x3D, 0x23}, {0x78, 0x57, 0x55, 0x50, 0x50, 0x20, 0x36, 0x37}}, {{0x4C, 0x5F, 0x53, 0x37, 0x1E, 0xFD, 0x15, 0x0A}, {0x05, 0x0E, 0x28, 0x41, 0x48, 0x6E, 0x54, 0x5B}, {0x59, 0x47, 0x40, 0x40, 0x3D, 0x33, 0x3F, 0x39}}, {{0x47, 0x5F, 0x57, 0x34, 0x3C, 0x2E, 0x2E, 0x31}, {0xFA, 0x13, 0x23, 0x4E, 0x44, 0x7C, 0x34, 0x38}, {0x63, 0x43, 0x41, 0x3D, 0x35, 0x19, 0x3D, 0x33}}}; static const UCHAR g_2a_pvcTab2_mode1[PVC_NTAB2][PVC_NBHIGH_MODE1] = { {0xCB, 0xD1, 0xCC, 0xD2, 0xE2, 0xEB, 0xE7, 0xE8}, {0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80}, {0x84, 0x8C, 0x88, 0x83, 0x90, 0x93, 0x86, 0x80}, {0xD7, 0xD8, 0xC0, 0xC7, 0xCF, 0xE5, 0xF1, 0xF6}, {0xA5, 0xA6, 0xAA, 0xA8, 0xB0, 0xB1, 0xB8, 0xB8}, {0xD7, 0xCB, 0xC1, 0xC3, 0xC5, 0xC9, 0xC9, 0xCE}, {0xCA, 0xB5, 0xB8, 0xB3, 0xAC, 0xB6, 0xBB, 0xB8}, {0xC1, 0xC4, 0xC3, 0xC5, 0xC6, 0xCA, 0xCA, 0xCB}, {0xE0, 0xE1, 0xD8, 0xCD, 0xCB, 0xCB, 0xCE, 0xCC}, {0xDB, 0xE1, 0xDF, 0xDB, 0xDC, 0xD9, 0xD9, 0xD6}, {0xE0, 0xDE, 0xDD, 0xDD, 0xE0, 0xE3, 0xE5, 0xE6}, {0xCA, 0xD2, 0xCD, 0xCE, 0xD5, 0xDB, 0xD9, 0xDB}, {0xD2, 0xE0, 0xDB, 0xD5, 0xDB, 0xDE, 0xE3, 0xE1}, {0xE5, 0xDB, 0xD0, 0xD2, 0xD8, 0xDD, 0xDB, 0xDD}, {0xC0, 0xB5, 0xBF, 0xDD, 0xE3, 0xDC, 0xDC, 0xE4}, {0xDB, 0xCE, 0xC6, 0xCF, 0xCF, 0xD1, 0xD3, 0xD4}, {0xC9, 0xD7, 0xDA, 0xE2, 0xE9, 0xE7, 0xDF, 0xDC}, {0x0A, 0x07, 0x0A, 0x08, 0x19, 0x24, 0x1F, 0x22}, {0x1E, 0x1F, 0x11, 0x0E, 0x22, 0x2D, 0x33, 0x32}, {0xF0, 0xDA, 0xDC, 0x18, 0x1F, 0x19, 0x0A, 0x1E}, {0x09, 0xF8, 0xE6, 0x05, 0x19, 0x11, 0x0E, 0x0B}, {0x09, 0x10, 0x0E, 0xE6, 0xF4, 0x20, 0x22, 0xFA}, {0xF2, 0xE5, 0xF8, 0x0E, 0x18, 0x15, 0x0D, 0x10}, {0x15, 0x13, 0x16, 0x0A, 0x0D, 0x1F, 0x1D, 0x1B}, {0xFA, 0xFF, 0xFE, 0xFF, 0x09, 0x11, 0x03, 0x0B}, {0xFE, 0xFA, 0xF2, 0xF8, 0x0C, 0x1E, 0x11, 0x12}, {0xFA, 0xF8, 0x0B, 0x17, 0x1D, 0x17, 0x0E, 0x16}, {0x00, 0xF3, 0xFD, 0x0A, 0x1C, 0x17, 0xFD, 0x08}, {0xEA, 0xEA, 0x03, 0x12, 0x1E, 0x14, 0x09, 0x04}, {0x02, 0xFE, 0x04, 0xFB, 0x0C, 0x0E, 0x07, 0x02}, {0xF6, 0x02, 0x07, 0x0B, 0x17, 0x17, 0x01, 0xFF}, {0xF5, 0xFB, 0xFE, 0x04, 0x12, 0x14, 0x0C, 0x0D}, {0x10, 0x10, 0x0E, 0x04, 0x07, 0x11, 0x0F, 0x13}, {0x0C, 0x0F, 0xFB, 0xF2, 0x0A, 0x12, 0x09, 0x0D}, {0x0D, 0x1D, 0xF1, 0xF4, 0x2A, 0x06, 0x3B, 0x32}, {0xFC, 0x08, 0x06, 0x02, 0x0E, 0x17, 0x08, 0x0E}, {0x07, 0x02, 0xEE, 0xEE, 0x2B, 0xF6, 0x23, 0x13}, {0x04, 0x02, 0x05, 0x08, 0x0B, 0x0E, 0xFB, 0xFB}, {0x00, 0x04, 0x10, 0x18, 0x22, 0x25, 0x1D, 0x1F}, {0xFB, 0x0D, 0x07, 0x00, 0x0C, 0x0F, 0xFC, 0x02}, {0x00, 0x00, 0x00, 0x01, 0x05, 0x07, 0x03, 0x05}, {0x04, 0x05, 0x08, 0x13, 0xFF, 0xEB, 0x0C, 0x06}, {0x05, 0x13, 0x0E, 0x0B, 0x12, 0x15, 0x09, 0x0A}, {0x09, 0x03, 0x09, 0x05, 0x12, 0x16, 0x11, 0x12}, {0x14, 0x1A, 0x06, 0x01, 0x10, 0x11, 0xFE, 0x02}, {0x01, 0x0B, 0x0B, 0x0C, 0x18, 0x21, 0x10, 0x13}, {0x12, 0x0D, 0x0A, 0x10, 0x1C, 0x1D, 0x0D, 0x10}, {0x03, 0x09, 0x14, 0x15, 0x1B, 0x1A, 0x01, 0xFF}, {0x08, 0x12, 0x13, 0x0E, 0x16, 0x1D, 0x14, 0x1B}, {0x07, 0x15, 0x1C, 0x1B, 0x20, 0x21, 0x11, 0x0E}, {0x12, 0x18, 0x19, 0x17, 0x20, 0x25, 0x1A, 0x1E}, {0x0C, 0x1A, 0x1D, 0x22, 0x2F, 0x33, 0x27, 0x28}, {0x0E, 0x1A, 0x17, 0x10, 0x0A, 0x0E, 0xFF, 0x06}, {0x1A, 0x1C, 0x18, 0x14, 0x1A, 0x16, 0x0A, 0x0E}, {0x1E, 0x27, 0x25, 0x26, 0x27, 0x2A, 0x21, 0x21}, {0xF1, 0x0A, 0x16, 0x1C, 0x28, 0x25, 0x15, 0x19}, {0x08, 0x12, 0x09, 0x08, 0x16, 0x17, 0xEF, 0xF6}, {0x0C, 0x0B, 0x00, 0xFC, 0x04, 0x09, 0xFC, 0x03}, {0xFB, 0xF1, 0xF8, 0x26, 0x24, 0x18, 0x1D, 0x20}, {0xF9, 0x01, 0x0C, 0x0F, 0x07, 0x08, 0x06, 0x07}, {0x07, 0x06, 0x08, 0x04, 0x07, 0x0D, 0x07, 0x09}, {0xFE, 0x01, 0x06, 0x05, 0x13, 0x1B, 0x14, 0x19}, {0x09, 0x0C, 0x0E, 0x01, 0x08, 0x05, 0xFB, 0xFD}, {0x07, 0x06, 0x03, 0x0A, 0x16, 0x12, 0x04, 0x07}, {0x04, 0x01, 0x00, 0x04, 0x1F, 0x20, 0x0E, 0x0A}, {0x03, 0xFF, 0xF6, 0xFB, 0x15, 0x1A, 0x00, 0x03}, {0xFC, 0x18, 0x0B, 0x2D, 0x35, 0x23, 0x12, 0x09}, {0x02, 0xFE, 0x01, 0xFF, 0x0C, 0x11, 0x0D, 0x0F}, {0xFA, 0xE9, 0xD9, 0xFF, 0x0D, 0x05, 0x0D, 0x10}, {0xF1, 0xE0, 0xF0, 0x01, 0x06, 0x06, 0x06, 0x10}, {0xE9, 0xD4, 0xD7, 0x0F, 0x14, 0x0B, 0x0D, 0x16}, {0x00, 0xFF, 0xEE, 0xE5, 0xFF, 0x08, 0x02, 0xF9}, {0xE0, 0xDA, 0xE5, 0xFE, 0x09, 0x02, 0xF9, 0x04}, {0xE0, 0xE2, 0xF4, 0x09, 0x13, 0x0C, 0x0D, 0x09}, {0xFC, 0x02, 0x04, 0xFF, 0x00, 0xFF, 0xF8, 0xF7}, {0xFE, 0xFB, 0xED, 0xF2, 0xFE, 0xFE, 0x08, 0x0C}, {0xF3, 0xEF, 0xD0, 0xE3, 0x05, 0x11, 0xFD, 0xFF}, {0xFA, 0xEF, 0xEA, 0xFE, 0x0D, 0x0E, 0xFE, 0x02}, {0xF7, 0xFB, 0xDB, 0xDF, 0x14, 0xDD, 0x07, 0xFE}, {0xFE, 0x08, 0x00, 0xDB, 0xE5, 0x1A, 0x13, 0xED}, {0xF9, 0xFE, 0xFF, 0xF4, 0xF3, 0x00, 0x05, 0x02}, {0xEF, 0xDE, 0xD8, 0xEB, 0xEA, 0xF5, 0x0E, 0x19}, {0xFB, 0xFC, 0xFA, 0xEC, 0xEB, 0xED, 0xEE, 0xE8}, {0xEE, 0xFC, 0xFD, 0x00, 0x04, 0xFC, 0xF0, 0xF5}, {0x00, 0xFA, 0xF4, 0xF1, 0xF5, 0xFA, 0xFB, 0xF9}, {0xEB, 0xF0, 0xDF, 0xE3, 0xEF, 0x07, 0x02, 0x05}, {0xF7, 0xF0, 0xE6, 0xE7, 0x06, 0x15, 0x06, 0x0C}, {0xF1, 0xE4, 0xD8, 0xEA, 0x06, 0xF2, 0x07, 0x09}, {0xFF, 0xFE, 0xFE, 0xF9, 0xFF, 0xFF, 0x02, 0xF9}, {0xDD, 0xF4, 0xF0, 0xF1, 0xFF, 0xFF, 0xEA, 0xF1}, {0xF0, 0xF1, 0xFD, 0x03, 0x03, 0xFE, 0x00, 0x05}, {0xF1, 0xF6, 0xE0, 0xDF, 0xF5, 0x01, 0xF4, 0xF8}, {0x02, 0x03, 0xE5, 0xDC, 0xE7, 0xFD, 0x02, 0x08}, {0xEC, 0xF1, 0xF5, 0xEC, 0xF2, 0xF8, 0xF6, 0xEE}, {0xF3, 0xF4, 0xF6, 0xF4, 0xF5, 0xF1, 0xE7, 0xEA}, {0xF7, 0xF3, 0xEC, 0xEA, 0xEF, 0xF0, 0xEE, 0xF1}, {0xEB, 0xF6, 0xFB, 0xFA, 0xEF, 0xF3, 0xF3, 0xF7}, {0x01, 0x03, 0xF1, 0xF6, 0x05, 0xF8, 0xE1, 0xEB}, {0xF5, 0xF6, 0xF6, 0xF4, 0xFB, 0xFB, 0xFF, 0x00}, {0xF8, 0x01, 0xFB, 0xFA, 0xFF, 0x03, 0xFE, 0x04}, {0x04, 0xFB, 0x03, 0xFD, 0xF5, 0xF7, 0xF6, 0xFB}, {0x06, 0x09, 0xFB, 0xF4, 0xF9, 0xFA, 0xFC, 0xFF}, {0xF5, 0xF6, 0xF1, 0xEE, 0xF5, 0xF8, 0xF5, 0xF9}, {0xF5, 0xF9, 0xFA, 0xFC, 0x07, 0x09, 0x01, 0xFB}, {0xD7, 0xE9, 0xE8, 0xEC, 0x00, 0x0C, 0xFE, 0xF1}, {0xEC, 0x04, 0xE9, 0xDF, 0x03, 0xE8, 0x00, 0xFA}, {0xE6, 0xE2, 0xFF, 0x0A, 0x13, 0x01, 0x00, 0xF7}, {0xF1, 0xFA, 0xF7, 0xF5, 0x01, 0x06, 0x05, 0x0A}, {0xF6, 0xF6, 0xFC, 0xF6, 0xE8, 0x11, 0xF2, 0xFE}, {0xFE, 0x08, 0x05, 0x12, 0xFD, 0xD0, 0x0E, 0x07}, {0xF1, 0xFE, 0xF7, 0xF2, 0xFB, 0x02, 0xFA, 0xF8}, {0xF4, 0xEA, 0xEC, 0xF3, 0xFE, 0x01, 0xF7, 0xF6}, {0xFF, 0xFA, 0xFB, 0xF9, 0xFF, 0x01, 0x04, 0x03}, {0x00, 0xF9, 0xF4, 0xFC, 0x05, 0xFC, 0xF7, 0xFB}, {0xF8, 0xFF, 0xEF, 0xEC, 0xFB, 0x04, 0xF8, 0x03}, {0xEB, 0xF1, 0xED, 0xF4, 0x02, 0x0E, 0x0B, 0x04}, {0xF7, 0x01, 0xF8, 0xF4, 0xF8, 0xEF, 0xF8, 0x04}, {0xEB, 0xF0, 0xF7, 0xFC, 0x10, 0x0D, 0xF8, 0xF8}, {0xE8, 0xFE, 0xEE, 0xE8, 0xED, 0xF7, 0xF5, 0xF8}, {0xED, 0xEB, 0xE9, 0xEA, 0xF2, 0xF5, 0xF4, 0xF9}, {0xEA, 0xF2, 0xEF, 0xEE, 0xF9, 0xFE, 0xFD, 0x02}, {0xFA, 0xFD, 0x02, 0x0D, 0xFA, 0xE4, 0x0F, 0x01}, {0xFF, 0x08, 0x05, 0xF6, 0xF7, 0xFB, 0xF1, 0xF1}, {0xF4, 0xEC, 0xEE, 0xF6, 0xEE, 0xEE, 0xF8, 0x06}, {0xE8, 0xFA, 0xF8, 0xE8, 0xF8, 0xE9, 0xEE, 0xF9}, {0xE5, 0xE9, 0xF0, 0x00, 0x00, 0xEF, 0xF3, 0xF8}, {0xF7, 0xFB, 0xFB, 0xF7, 0xF9, 0xF9, 0xF5, 0xF0}, {0xFD, 0xFF, 0xF2, 0xEE, 0xF2, 0xF5, 0xF1, 0xF3}}; static const UCHAR g_3a_pvcTab1_mode2[PVC_NTAB1][PVC_NBLOW][PVC_NBHIGH_MODE2] = {{{0x11, 0x27, 0x0F, 0xFD, 0x04, 0xFC}, {0x00, 0xBE, 0xE3, 0xF4, 0xDB, 0xF0}, {0x09, 0x1E, 0x18, 0x1A, 0x21, 0x1B}}, {{0x16, 0x28, 0x2B, 0x29, 0x25, 0x32}, {0xF2, 0xE9, 0xE4, 0xE5, 0xE2, 0xD4}, {0x0E, 0x0B, 0x0C, 0x0D, 0x0D, 0x0E}}, {{0x2E, 0x3C, 0x20, 0x16, 0x1B, 0x1A}, {0xE4, 0xC6, 0xE5, 0xF4, 0xDC, 0xDC}, {0x0F, 0x1B, 0x18, 0x14, 0x1E, 0x1A}}}; static const UCHAR g_2a_pvcTab2_mode2[PVC_NTAB2][PVC_NBHIGH_MODE2] = { {0x26, 0x25, 0x11, 0x0C, 0xFA, 0x15}, {0x1B, 0x18, 0x11, 0x0E, 0x0E, 0x0E}, {0x12, 0x10, 0x10, 0x10, 0x11, 0x10}, {0x1E, 0x24, 0x19, 0x15, 0x14, 0x12}, {0x24, 0x16, 0x12, 0x13, 0x15, 0x1C}, {0xEA, 0xED, 0xEB, 0xEA, 0xEC, 0xEB}, {0xFC, 0xFD, 0xFD, 0xFC, 0xFE, 0xFE}, {0x0F, 0x0C, 0x0B, 0x0A, 0x0B, 0x0B}, {0x22, 0x0B, 0x16, 0x18, 0x13, 0x19}, {0x1C, 0x14, 0x1D, 0x20, 0x19, 0x1A}, {0x10, 0x08, 0x00, 0xFF, 0x02, 0x05}, {0x06, 0x07, 0x05, 0x03, 0x05, 0x04}, {0x2A, 0x1F, 0x12, 0x12, 0x11, 0x18}, {0x19, 0x19, 0x02, 0x04, 0x00, 0x04}, {0x18, 0x17, 0x17, 0x15, 0x16, 0x15}, {0x21, 0x1E, 0x1B, 0x19, 0x1C, 0x1B}, {0x3C, 0x35, 0x20, 0x1D, 0x30, 0x34}, {0x3A, 0x1F, 0x37, 0x38, 0x33, 0x31}, {0x37, 0x34, 0x25, 0x27, 0x35, 0x34}, {0x34, 0x2E, 0x32, 0x31, 0x34, 0x31}, {0x36, 0x33, 0x2F, 0x2F, 0x32, 0x2F}, {0x35, 0x20, 0x2F, 0x32, 0x2F, 0x2C}, {0x2E, 0x2B, 0x2F, 0x34, 0x36, 0x30}, {0x3F, 0x39, 0x30, 0x28, 0x29, 0x29}, {0x3C, 0x30, 0x32, 0x37, 0x39, 0x36}, {0x37, 0x36, 0x30, 0x2B, 0x26, 0x24}, {0x44, 0x38, 0x2F, 0x2D, 0x2D, 0x2D}, {0x38, 0x2B, 0x2C, 0x2C, 0x30, 0x2D}, {0x37, 0x36, 0x2F, 0x23, 0x2D, 0x32}, {0x3C, 0x39, 0x29, 0x2E, 0x38, 0x37}, {0x3B, 0x3A, 0x35, 0x32, 0x31, 0x2D}, {0x32, 0x31, 0x2F, 0x2C, 0x2D, 0x28}, {0x2C, 0x31, 0x32, 0x30, 0x32, 0x2D}, {0x35, 0x34, 0x34, 0x34, 0x35, 0x33}, {0x34, 0x38, 0x3B, 0x3C, 0x3E, 0x3A}, {0x3E, 0x3C, 0x3B, 0x3A, 0x3C, 0x39}, {0x3D, 0x41, 0x46, 0x41, 0x3D, 0x38}, {0x44, 0x41, 0x40, 0x3E, 0x3F, 0x3A}, {0x47, 0x47, 0x47, 0x42, 0x44, 0x40}, {0x4C, 0x4A, 0x4A, 0x46, 0x49, 0x45}, {0x53, 0x52, 0x52, 0x4C, 0x4E, 0x49}, {0x41, 0x3D, 0x39, 0x2C, 0x2E, 0x2E}, {0x2D, 0x37, 0x36, 0x30, 0x28, 0x36}, {0x3B, 0x32, 0x2E, 0x2D, 0x2D, 0x29}, {0x40, 0x39, 0x36, 0x35, 0x36, 0x32}, {0x30, 0x2D, 0x2D, 0x2E, 0x31, 0x30}, {0x38, 0x3D, 0x3B, 0x37, 0x35, 0x34}, {0x44, 0x3D, 0x3C, 0x38, 0x37, 0x33}, {0x3A, 0x36, 0x37, 0x37, 0x39, 0x36}, {0x32, 0x36, 0x37, 0x30, 0x2E, 0x2A}, {0x3C, 0x33, 0x33, 0x31, 0x33, 0x30}, {0x30, 0x31, 0x36, 0x37, 0x38, 0x34}, {0x26, 0x27, 0x2E, 0x29, 0x1C, 0x16}, {0x14, 0x15, 0x1F, 0x17, 0x15, 0x1C}, {0x38, 0x2D, 0x18, 0x13, 0x1E, 0x2B}, {0x30, 0x22, 0x17, 0x1A, 0x26, 0x2B}, {0x24, 0x20, 0x1F, 0x10, 0x0C, 0x11}, {0x27, 0x1F, 0x13, 0x17, 0x24, 0x2A}, {0x2F, 0x13, 0x18, 0x13, 0x2A, 0x32}, {0x31, 0x1E, 0x1E, 0x1E, 0x21, 0x28}, {0x2A, 0x12, 0x19, 0x17, 0x16, 0x24}, {0x27, 0x0F, 0x16, 0x1D, 0x17, 0x1C}, {0x2F, 0x26, 0x25, 0x22, 0x20, 0x22}, {0x1E, 0x1B, 0x1E, 0x18, 0x1E, 0x24}, {0x31, 0x26, 0x0E, 0x15, 0x15, 0x25}, {0x2D, 0x22, 0x1E, 0x14, 0x10, 0x22}, {0x25, 0x1B, 0x18, 0x11, 0x13, 0x1F}, {0x2F, 0x1B, 0x13, 0x1B, 0x18, 0x22}, {0x21, 0x24, 0x1D, 0x1C, 0x1D, 0x1B}, {0x23, 0x1E, 0x28, 0x29, 0x27, 0x25}, {0x2E, 0x2A, 0x1D, 0x17, 0x26, 0x2D}, {0x31, 0x2C, 0x1A, 0x0E, 0x1A, 0x24}, {0x26, 0x16, 0x20, 0x1D, 0x14, 0x1E}, {0x29, 0x20, 0x1B, 0x1B, 0x17, 0x17}, {0x1D, 0x06, 0x1A, 0x1E, 0x1B, 0x1D}, {0x2B, 0x23, 0x1F, 0x1F, 0x1D, 0x1C}, {0x27, 0x1A, 0x0C, 0x0E, 0x0F, 0x1A}, {0x29, 0x1D, 0x1E, 0x22, 0x22, 0x24}, {0x20, 0x21, 0x1B, 0x18, 0x13, 0x21}, {0x27, 0x0E, 0x10, 0x14, 0x10, 0x1A}, {0x26, 0x24, 0x25, 0x25, 0x26, 0x28}, {0x1A, 0x24, 0x25, 0x29, 0x26, 0x24}, {0x1D, 0x1D, 0x15, 0x12, 0x0F, 0x18}, {0x1E, 0x14, 0x13, 0x12, 0x14, 0x18}, {0x16, 0x13, 0x13, 0x1A, 0x1B, 0x1D}, {0x20, 0x27, 0x22, 0x24, 0x1A, 0x19}, {0x1F, 0x17, 0x19, 0x18, 0x17, 0x18}, {0x20, 0x1B, 0x1C, 0x1C, 0x1B, 0x1A}, {0x23, 0x19, 0x1D, 0x1F, 0x1E, 0x21}, {0x26, 0x1F, 0x1D, 0x1B, 0x19, 0x1A}, {0x23, 0x1E, 0x1F, 0x20, 0x1F, 0x1E}, {0x29, 0x20, 0x22, 0x20, 0x20, 0x1F}, {0x26, 0x23, 0x21, 0x22, 0x23, 0x23}, {0x29, 0x1F, 0x24, 0x25, 0x26, 0x29}, {0x2B, 0x22, 0x25, 0x27, 0x23, 0x21}, {0x29, 0x21, 0x19, 0x0E, 0x22, 0x2D}, {0x32, 0x29, 0x1F, 0x1C, 0x1B, 0x21}, {0x1E, 0x1A, 0x1E, 0x24, 0x25, 0x25}, {0x24, 0x1D, 0x21, 0x22, 0x22, 0x25}, {0x2C, 0x25, 0x21, 0x22, 0x23, 0x25}, {0x24, 0x1E, 0x21, 0x26, 0x2B, 0x2C}, {0x28, 0x24, 0x1B, 0x1F, 0x28, 0x2D}, {0x23, 0x13, 0x16, 0x22, 0x22, 0x29}, {0x1B, 0x23, 0x1C, 0x20, 0x14, 0x0D}, {0x1E, 0x16, 0x1A, 0x1E, 0x1C, 0x1D}, {0x2B, 0x1C, 0x1D, 0x20, 0x1B, 0x1C}, {0x1C, 0x1B, 0x23, 0x1F, 0x19, 0x1E}, {0x21, 0x23, 0x26, 0x20, 0x20, 0x22}, {0x1D, 0x0B, 0x19, 0x1E, 0x11, 0x19}, {0x18, 0x17, 0x16, 0x17, 0x14, 0x16}, {0x16, 0x19, 0x1C, 0x20, 0x21, 0x22}, {0x30, 0x1E, 0x22, 0x24, 0x25, 0x26}, {0x1B, 0x1F, 0x17, 0x1D, 0x1E, 0x21}, {0x32, 0x2B, 0x27, 0x1F, 0x1B, 0x1A}, {0x28, 0x20, 0x1A, 0x1B, 0x1F, 0x23}, {0x32, 0x21, 0x20, 0x21, 0x1D, 0x1F}, {0x22, 0x18, 0x12, 0x15, 0x1B, 0x20}, {0x27, 0x27, 0x2A, 0x24, 0x21, 0x21}, {0x1E, 0x0F, 0x0D, 0x1A, 0x1D, 0x23}, {0x28, 0x25, 0x27, 0x21, 0x17, 0x25}, {0x2B, 0x27, 0x23, 0x19, 0x13, 0x14}, {0x25, 0x2B, 0x22, 0x22, 0x20, 0x21}, {0x27, 0x1B, 0x16, 0x17, 0x0F, 0x15}, {0x29, 0x26, 0x23, 0x15, 0x1E, 0x28}, {0x24, 0x1C, 0x19, 0x1A, 0x18, 0x19}, {0x2D, 0x15, 0x27, 0x2B, 0x24, 0x23}, {0x2C, 0x12, 0x1F, 0x23, 0x1F, 0x20}, {0x25, 0x0F, 0x22, 0x27, 0x1F, 0x21}}; static const UCHAR g_a_pvcTab1_dp_mode1[PVC_NTAB1 - 1] = {17, 68}; static const UCHAR g_a_pvcTab1_dp_mode2[PVC_NTAB1 - 1] = {16, 52}; /* fractional exponent which corresponds to Q representation value */ static const SCHAR g_a_scalingCoef_mode1[PVC_NBLOW + 1] = { -1, -1, 0, 6}; /* { 8, 8, 7, 1 }; Q scaling */ static const SCHAR g_a_scalingCoef_mode2[PVC_NBLOW + 1] = { 0, 0, 1, 7}; /* { 7, 7, 6, 0 }; Q scaling */ int pvcInitFrame(PVC_STATIC_DATA *pPvcStaticData, PVC_DYNAMIC_DATA *pPvcDynamicData, const UCHAR pvcMode, const UCHAR ns, const int RATE, const int kx, const int pvcBorder0, const UCHAR *pPvcID) { int lbw, hbw, i, temp; pPvcDynamicData->pvc_mode = pvcMode; pPvcDynamicData->kx = kx; pPvcDynamicData->RATE = RATE; switch (pvcMode) { case 0: /* legacy SBR, nothing to do */ return 0; case 1: pPvcDynamicData->nbHigh = 8; pPvcDynamicData->pPVCTab1 = (const UCHAR *)g_3a_pvcTab1_mode1; pPvcDynamicData->pPVCTab2 = (const UCHAR *)g_2a_pvcTab2_mode1; pPvcDynamicData->pPVCTab1_dp = g_a_pvcTab1_dp_mode1; pPvcDynamicData->pScalingCoef = g_a_scalingCoef_mode1; hbw = 8 / RATE; break; case 2: pPvcDynamicData->nbHigh = 6; pPvcDynamicData->pPVCTab1 = (const UCHAR *)g_3a_pvcTab1_mode2; pPvcDynamicData->pPVCTab2 = (const UCHAR *)g_2a_pvcTab2_mode2; pPvcDynamicData->pPVCTab1_dp = g_a_pvcTab1_dp_mode2; pPvcDynamicData->pScalingCoef = g_a_scalingCoef_mode2; hbw = 12 / RATE; break; default: /* invalid pvcMode */ return 1; } pPvcDynamicData->pvcBorder0 = pvcBorder0; UCHAR pvcBorder0_last = pPvcStaticData->pvcBorder0; pPvcStaticData->pvcBorder0 = pvcBorder0; pPvcDynamicData->pPvcID = pPvcID; pPvcDynamicData->ns = ns; switch (ns) { case 16: pPvcDynamicData->pSCcoeffs = pvc_SC_16; break; case 12: pPvcDynamicData->pSCcoeffs = pvc_SC_12; break; case 4: pPvcDynamicData->pSCcoeffs = pvc_SC_4; break; case 3: pPvcDynamicData->pSCcoeffs = pvc_SC_3; break; default: return 1; } /* in the lower part of Esg-array there are previous values of Esg (from last call to this function In case of an previous legay-SBR frame, or if there was a change in cross-over FQ the value of first PVC SBR timeslot is propagated to prev-values in order to have reasonable values for smooth-filtering */ if ((pPvcStaticData->pvc_mode_last == 0) || (pPvcStaticData->kx_last != kx)) { pPvcDynamicData->pastEsgSlotsAvail = 0; } else { pPvcDynamicData->pastEsgSlotsAvail = PVC_NS_MAX - pvcBorder0_last; } lbw = 8 / RATE; temp = kx; for (i = PVC_NBLOW; i >= 0; i--) { pPvcDynamicData->sg_offset_low[i] = temp; temp -= lbw; } temp = 0; for (i = 0; i <= pPvcDynamicData->nbHigh; i++) { pPvcDynamicData->sg_offset_high_kx[i] = temp; temp += hbw; } return 0; } /* call if pvcMode = 1,2 */ void pvcDecodeFrame(PVC_STATIC_DATA *pPvcStaticData, PVC_DYNAMIC_DATA *pPvcDynamicData, FIXP_DBL **qmfBufferReal, FIXP_DBL **qmfBufferImag, const int overlap, const int qmfExponentOverlap, const int qmfExponentCurrent) { int t; FIXP_DBL *predictedEsgSlot; int RATE = pPvcDynamicData->RATE; int pvcBorder0 = pPvcDynamicData->pvcBorder0; for (t = pvcBorder0; t < PVC_NTIMESLOT; t++) { int *pPredEsg_exp = &pPvcDynamicData->predEsg_exp[t]; predictedEsgSlot = pPvcDynamicData->predEsg[t]; pvcDecodeTimeSlot( pPvcStaticData, pPvcDynamicData, &qmfBufferReal[t * RATE], &qmfBufferImag[t * RATE], (t * RATE < overlap) ? qmfExponentOverlap : qmfExponentCurrent, pvcBorder0, t, predictedEsgSlot, pPredEsg_exp); } return; } void pvcDecodeTimeSlot(PVC_STATIC_DATA *pPvcStaticData, PVC_DYNAMIC_DATA *pPvcDynamicData, FIXP_DBL **qmfSlotReal, FIXP_DBL **qmfSlotImag, const int qmfExponent, const int pvcBorder0, const int timeSlotNumber, FIXP_DBL predictedEsgSlot[], int *predictedEsg_exp) { int i, band, ksg, ksg_start = 0; int RATE = pPvcDynamicData->RATE; int Esg_index = pPvcStaticData->Esg_slot_index; const SCHAR *sg_borders = pPvcDynamicData->sg_offset_low; FIXP_DBL *pEsg = pPvcStaticData->Esg[Esg_index]; FIXP_DBL E[PVC_NBLOW] = {0}; /* Subband grouping in QMF subbands below SBR range */ /* Within one timeslot ( i = [0...(RATE-1)] QMF subsamples) calculate energy E(ib,t) and group them to Esg(ksg,t). Then transfer values to logarithmical domain and store them for time domain smoothing. (7.5.6.3 Subband grouping in QMF subbands below SBR range) */ for (ksg = 0; sg_borders[ksg] < 0; ksg++) { pEsg[ksg] = FL2FXCONST_DBL(-10.0 / (1 << PVC_ESG_EXP)); /* 10*log10(0.1) */ ksg_start++; } for (i = 0; i < RATE; i++) { FIXP_DBL *qmfR, *qmfI; qmfR = qmfSlotReal[i]; qmfI = qmfSlotImag[i]; for (ksg = ksg_start; ksg < PVC_NBLOW; ksg++) { for (band = sg_borders[ksg]; band < sg_borders[ksg + 1]; band++) { /* The division by 8 == (RATE*lbw) is required algorithmically */ E[ksg] += ((fPow2Div2(qmfR[band]) >> 1) + (fPow2Div2(qmfI[band]) >> 1)) >> 3; } } } for (ksg = ksg_start; ksg < PVC_NBLOW; ksg++) { if (E[ksg] > (FIXP_DBL)0) { /* 10/log2(10) = 0.752574989159953 * 2^2 */ int exp_log; FIXP_DBL nrg = CalcLog2(E[ksg], 2 * qmfExponent + 2, &exp_log); nrg = fMult(nrg, FL2FXCONST_SGL(LOG10FAC)); nrg = scaleValue(nrg, exp_log - PVC_ESG_EXP + 2); pEsg[ksg] = fMax(nrg, FL2FXCONST_DBL(-10.0 / (1 << PVC_ESG_EXP))); } else { pEsg[ksg] = FL2FXCONST_DBL(-10.0 / (1 << PVC_ESG_EXP)); /* 10*log10(0.1) */ } } /* Time domain smoothing of subband-grouped energy */ { int idx = pPvcStaticData->Esg_slot_index; FIXP_DBL *pEsg_filt; FIXP_SGL SCcoeff; E[0] = E[1] = E[2] = (FIXP_DBL)0; for (i = 0; i < pPvcDynamicData->ns; i++) { SCcoeff = pPvcDynamicData->pSCcoeffs[i]; pEsg_filt = pPvcStaticData->Esg[idx]; /* Div2 is compensated by scaling of coeff table */ E[0] = fMultAddDiv2(E[0], pEsg_filt[0], SCcoeff); E[1] = fMultAddDiv2(E[1], pEsg_filt[1], SCcoeff); E[2] = fMultAddDiv2(E[2], pEsg_filt[2], SCcoeff); if (i >= pPvcDynamicData->pastEsgSlotsAvail) { /* if past Esg values are not available use the ones from the last valid * slot */ continue; } if (idx > 0) { idx--; } else { idx += PVC_NS_MAX - 1; } } } /* SBR envelope scalefactor prediction */ { int E_high_exp[PVC_NBHIGH_MAX]; int E_high_exp_max = 0; int pvcTab1ID; int pvcTab2ID = (int)pPvcDynamicData->pPvcID[timeSlotNumber]; const UCHAR *pTab1, *pTab2; if (pvcTab2ID < pPvcDynamicData->pPVCTab1_dp[0]) { pvcTab1ID = 0; } else if (pvcTab2ID < pPvcDynamicData->pPVCTab1_dp[1]) { pvcTab1ID = 1; } else { pvcTab1ID = 2; } pTab1 = &(pPvcDynamicData ->pPVCTab1[pvcTab1ID * PVC_NBLOW * pPvcDynamicData->nbHigh]); pTab2 = &(pPvcDynamicData->pPVCTab2[pvcTab2ID * pPvcDynamicData->nbHigh]); for (ksg = 0; ksg < pPvcDynamicData->nbHigh; ksg++) { FIXP_SGL predCoeff; FIXP_DBL accu; int predCoeff_exp, kb; E_high_exp[ksg] = 0; /* residual part */ accu = ((LONG)(SCHAR)*pTab2++) << (DFRACT_BITS - 8 - PVC_ESG_EXP - 2 + pPvcDynamicData->pScalingCoef[3]); /* linear combination of lower grouped energies part */ for (kb = 0; kb < PVC_NBLOW; kb++) { predCoeff = (FIXP_SGL)( (SHORT)(SCHAR)pTab1[kb * pPvcDynamicData->nbHigh + ksg] << 8); predCoeff_exp = -(pPvcDynamicData->pScalingCoef[kb] + 1 - 2); /* +1 to compensate for Div2; -2 for accu */ accu += fMultDiv2(E[kb], predCoeff) >> predCoeff_exp; } /* convert back to linear domain */ accu = fMult(accu, FL2FXCONST_SGL(LOG10FAC_INV)); accu = f2Pow(accu, PVC_ESG_EXP - 1 + 2, &predCoeff_exp); /* -1 compensates for exponent of LOG10FAC_INV; +2 for accu */ predictedEsgSlot[ksg] = accu; E_high_exp[ksg] = predCoeff_exp; if (predCoeff_exp > E_high_exp_max) { E_high_exp_max = predCoeff_exp; } } /* rescale output vector according to largest exponent */ for (ksg = 0; ksg < pPvcDynamicData->nbHigh; ksg++) { int scale = fMin(E_high_exp_max - E_high_exp[ksg], DFRACT_BITS - 1); predictedEsgSlot[ksg] = predictedEsgSlot[ksg] >> scale; } *predictedEsg_exp = E_high_exp_max; } pPvcStaticData->Esg_slot_index = (pPvcStaticData->Esg_slot_index + 1) & (PVC_NS_MAX - 1); pPvcDynamicData->pastEsgSlotsAvail = fMin(pPvcDynamicData->pastEsgSlotsAvail + 1, PVC_NS_MAX - 1); return; } /* call if pvcMode = 0,1,2 */ void pvcEndFrame(PVC_STATIC_DATA *pPvcStaticData, PVC_DYNAMIC_DATA *pPvcDynamicData) { pPvcStaticData->pvc_mode_last = pPvcDynamicData->pvc_mode; pPvcStaticData->kx_last = pPvcDynamicData->kx; if (pPvcDynamicData->pvc_mode == 0) return; { int t, max = -100; for (t = pPvcDynamicData->pvcBorder0; t < PVC_NTIMESLOT; t++) { if (pPvcDynamicData->predEsg_exp[t] > max) { max = pPvcDynamicData->predEsg_exp[t]; } } pPvcDynamicData->predEsg_expMax = max; } return; } void expandPredEsg(const PVC_DYNAMIC_DATA *pPvcDynamicData, const int timeSlot, const int lengthOutputVector, FIXP_DBL *pOutput, SCHAR *pOutput_exp) { int k = 0, ksg; const FIXP_DBL *predEsg = pPvcDynamicData->predEsg[timeSlot]; for (ksg = 0; ksg < pPvcDynamicData->nbHigh; ksg++) { for (; k < pPvcDynamicData->sg_offset_high_kx[ksg + 1]; k++) { pOutput[k] = predEsg[ksg]; pOutput_exp[k] = (SCHAR)pPvcDynamicData->predEsg_exp[timeSlot]; } } ksg--; for (; k < lengthOutputVector; k++) { pOutput[k] = predEsg[ksg]; pOutput_exp[k] = (SCHAR)pPvcDynamicData->predEsg_exp[timeSlot]; } return; } fdk-aac-2.0.2/libSBRdec/src/pvc_dec.h000066400000000000000000000261301404225310700171350ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Matthias Hildenbrand Description: Decode Predictive Vector Coding Data *******************************************************************************/ #ifndef PVC_DEC_H #define PVC_DEC_H #include "common_fix.h" #define PVC_DIVMODE_BITS 3 #define PVC_REUSEPVCID_BITS 1 #define PVC_PVCID_BITS 7 #define PVC_GRIDINFO_BITS 1 #define MAX_PVC_ENVELOPES 2 #define PVC_NTIMESLOT 16 #define PVC_NBLOW 3 /* max. number of grouped QMF subbands below SBR range */ #define PVC_NBHIGH_MODE1 8 #define PVC_NBHIGH_MODE2 6 #define PVC_NBHIGH_MAX (PVC_NBHIGH_MODE1) #define PVC_NS_MAX 16 /** Data for each PVC instance which needs to be persistent accross SBR frames */ typedef struct { UCHAR kx_last; /**< Xover frequency of last frame */ UCHAR pvc_mode_last; /**< PVC mode of last frame */ UCHAR Esg_slot_index; /**< Ring buffer index to current Esg time slot */ UCHAR pvcBorder0; /**< Start SBR time slot of PVC frame */ FIXP_DBL Esg[PVC_NS_MAX][PVC_NBLOW]; /**< Esg(ksg,t) of current and 15 previous time slots (ring buffer) in logarithmical domain */ } PVC_STATIC_DATA; /** Data for each PVC instance which is valid during one SBR frame */ typedef struct { UCHAR pvc_mode; /**< PVC mode 1 or 2, 0 means legacy SBR */ UCHAR pvcBorder0; /**< Start SBR time slot of PVC frame */ UCHAR kx; /**< Index of the first QMF subband in the SBR range */ UCHAR RATE; /**< Number of QMF subband samples per time slot (2 or 4) */ UCHAR ns; /**< Number of time slots for time-domain smoothing of Esg(ksg,t) */ const UCHAR *pPvcID; /**< Pointer to prediction coefficient matrix index table */ UCHAR pastEsgSlotsAvail; /**< Number of past Esg(ksg,t) which are available for smoothing filter */ const FIXP_SGL *pSCcoeffs; /**< Pointer to smoothing window table */ SCHAR sg_offset_low[PVC_NBLOW + 1]; /**< Offset table for PVC grouping of SBR subbands below SBR range */ SCHAR sg_offset_high_kx[PVC_NBHIGH_MAX + 1]; /**< Offset table for PVC grouping of SBR subbands in SBR range (relativ to kx) */ UCHAR nbHigh; /**< Number of grouped QMF subbands in the SBR range */ const SCHAR *pScalingCoef; /**< Pointer to scaling coeff table */ const UCHAR *pPVCTab1; /**< PVC mode 1 table */ const UCHAR *pPVCTab2; /**< PVC mode 2 table */ const UCHAR *pPVCTab1_dp; /**< Mapping of pvcID to PVC mode 1 table */ FIXP_DBL predEsg[PVC_NTIMESLOT] [PVC_NBHIGH_MAX]; /**< Predicted Energy in linear domain */ int predEsg_exp[PVC_NTIMESLOT]; /**< Exponent of predicted Energy in linear domain */ int predEsg_expMax; /**< Maximum of predEsg_exp[] */ } PVC_DYNAMIC_DATA; /** * \brief Initialize PVC data structures for current frame (call if pvcMode = * 0,1,2) * \param[in] pPvcStaticData Pointer to PVC persistent data * \param[out] pPvcDynamicData Pointer to PVC dynamic data * \param[in] pvcMode PVC mode 1 or 2, 0 means legacy SBR * \param[in] ns Number of time slots for time-domain smoothing of Esg(ksg,t) * \param[in] RATE Number of QMF subband samples per time slot (2 or 4) * \param[in] kx Index of the first QMF subband in the SBR range * \param[in] pvcBorder0 Start SBR time slot of PVC frame * \param[in] pPvcID Pointer to array of PvcIDs read from bitstream */ int pvcInitFrame(PVC_STATIC_DATA *pPvcStaticData, PVC_DYNAMIC_DATA *pPvcDynamicData, const UCHAR pvcMode, const UCHAR ns, const int RATE, const int kx, const int pvcBorder0, const UCHAR *pPvcID); /** * \brief Wrapper function for pvcDecodeTimeSlot() to decode PVC data of one * frame (call if pvcMode = 1,2) * \param[in,out] pPvcStaticData Pointer to PVC persistent data * \param[in,out] pPvcDynamicData Pointer to PVC dynamic data * \param[in] qmfBufferReal Pointer to array with real QMF subbands * \param[in] qmfBufferImag Pointer to array with imag QMF subbands * \param[in] overlap Number of QMF overlap slots * \param[in] qmfExponentOverlap Exponent of qmfBuffer (low part) of overlap * slots * \param[in] qmfExponentCurrent Exponent of qmfBuffer (low part) */ void pvcDecodeFrame(PVC_STATIC_DATA *pPvcStaticData, PVC_DYNAMIC_DATA *pPvcDynamicData, FIXP_DBL **qmfBufferReal, FIXP_DBL **qmfBufferImag, const int overlap, const int qmfExponentOverlap, const int qmfExponentCurrent); /** * \brief Decode PVC data for one SBR time slot (call if pvcMode = 1,2) * \param[in,out] pPvcStaticData Pointer to PVC persistent data * \param[in,out] pPvcDynamicData Pointer to PVC dynamic data * \param[in] qmfBufferReal Pointer to array with real QMF subbands * \param[in] qmfBufferImag Pointer to array with imag QMF subbands * \param[in] qmfExponent Exponent of qmfBuffer of current time slot * \param[in] pvcBorder0 Start SBR time slot of PVC frame * \param[in] timeSlotNumber Number of current SBR time slot (0..15) * \param[out] predictedEsgSlot Predicted Energy of current time slot * \param[out] predictedEsg_exp Exponent of predicted Energy of current time * slot */ void pvcDecodeTimeSlot(PVC_STATIC_DATA *pPvcStaticData, PVC_DYNAMIC_DATA *pPvcDynamicData, FIXP_DBL **qmfSlotReal, FIXP_DBL **qmfSlotImag, const int qmfExponent, const int pvcBorder0, const int timeSlotNumber, FIXP_DBL predictedEsgSlot[], int *predictedEsg_exp); /** * \brief Finish the current PVC frame (call if pvcMode = 0,1,2) * \param[in,out] pPvcStaticData Pointer to PVC persistent data * \param[in,out] pPvcDynamicData Pointer to PVC dynamic data */ void pvcEndFrame(PVC_STATIC_DATA *pPvcStaticData, PVC_DYNAMIC_DATA *pPvcDynamicData); /** * \brief Expand predicted PVC grouped energies to full QMF subband resolution * \param[in] pPvcDynamicData Pointer to PVC dynamic data * \param[in] timeSlot Number of current SBR time slot (0..15) * \param[in] lengthOutputVector Lenght of output vector * \param[out] pOutput Output array for predicted energies * \param[out] pOutput_exp Exponent of predicted energies */ void expandPredEsg(const PVC_DYNAMIC_DATA *pPvcDynamicData, const int timeSlot, const int lengthOutputVector, FIXP_DBL *pOutput, SCHAR *pOutput_exp); #endif /* PVC_DEC_H*/ fdk-aac-2.0.2/libSBRdec/src/sbr_deb.cpp000066400000000000000000000114221404225310700174630ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Print selected debug messages */ #include "sbr_deb.h" fdk-aac-2.0.2/libSBRdec/src/sbr_deb.h000066400000000000000000000114631404225310700171350ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Debugging aids */ #ifndef SBR_DEB_H #define SBR_DEB_H #include "sbrdecoder.h" #endif fdk-aac-2.0.2/libSBRdec/src/sbr_dec.cpp000066400000000000000000001627211404225310700174750ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Sbr decoder This module provides the actual decoder implementation. The SBR data (side information) is already decoded. Only three functions are provided: \li 1.) createSbrDec(): One time initialization \li 2.) resetSbrDec(): Called by sbr_Apply() when the information contained in an SBR_HEADER_ELEMENT requires a reset and recalculation of important SBR structures. \li 3.) sbr_dec(): The actual decoder. Calls the different tools such as filterbanks, lppTransposer(), and calculateSbrEnvelope() [the envelope adjuster]. \sa sbr_dec(), \ref documentationOverview */ #include "sbr_dec.h" #include "sbr_ram.h" #include "env_extr.h" #include "env_calc.h" #include "scale.h" #include "FDK_matrixCalloc.h" #include "hbe.h" #include "genericStds.h" #include "sbrdec_drc.h" static void copyHarmonicSpectrum(int *xOverQmf, FIXP_DBL **qmfReal, FIXP_DBL **qmfImag, int noCols, int overlap, KEEP_STATES_SYNCED_MODE keepStatesSynced) { int patchBands; int patch, band, col, target, sourceBands, i; int numPatches = 0; int slotOffset = 0; FIXP_DBL **ppqmfReal = qmfReal + overlap; FIXP_DBL **ppqmfImag = qmfImag + overlap; if (keepStatesSynced == KEEP_STATES_SYNCED_NORMAL) { slotOffset = noCols - overlap - LPC_ORDER; } if (keepStatesSynced == KEEP_STATES_SYNCED_OUTDIFF) { ppqmfReal = qmfReal; ppqmfImag = qmfImag; } for (i = 1; i < MAX_NUM_PATCHES; i++) { if (xOverQmf[i] != 0) { numPatches++; } } for (patch = (MAX_STRETCH_HBE - 1); patch < numPatches; patch++) { patchBands = xOverQmf[patch + 1] - xOverQmf[patch]; target = xOverQmf[patch]; sourceBands = xOverQmf[MAX_STRETCH_HBE - 1] - xOverQmf[MAX_STRETCH_HBE - 2]; while (patchBands > 0) { int numBands = sourceBands; int startBand = xOverQmf[MAX_STRETCH_HBE - 1] - 1; if (target + numBands >= xOverQmf[patch + 1]) { numBands = xOverQmf[patch + 1] - target; } if ((((target + numBands - 1) % 2) + ((xOverQmf[MAX_STRETCH_HBE - 1] - 1) % 2)) % 2) { if (numBands == sourceBands) { numBands--; } else { startBand--; } } if (keepStatesSynced == KEEP_STATES_SYNCED_OUTDIFF) { for (col = slotOffset; col < overlap + LPC_ORDER; col++) { i = 0; for (band = numBands; band > 0; band--) { if ((target + band - 1 < 64) && (target + band - 1 < xOverQmf[patch + 1])) { ppqmfReal[col][target + band - 1] = ppqmfReal[col][startBand - i]; ppqmfImag[col][target + band - 1] = ppqmfImag[col][startBand - i]; i++; } } } } else { for (col = slotOffset; col < noCols; col++) { i = 0; for (band = numBands; band > 0; band--) { if ((target + band - 1 < 64) && (target + band - 1 < xOverQmf[patch + 1])) { ppqmfReal[col][target + band - 1] = ppqmfReal[col][startBand - i]; ppqmfImag[col][target + band - 1] = ppqmfImag[col][startBand - i]; i++; } } } } target += numBands; patchBands -= numBands; } } } /*! \brief SBR decoder core function for one channel \image html BufferMgmtDetailed-1632.png Besides the filter states of the QMF filter bank and the LPC-states of the LPP-Transposer, processing is mainly based on four buffers: #timeIn, #timeOut, #WorkBuffer2 and #OverlapBuffer. The #WorkBuffer2 is reused for all channels and might be used by the core decoder, a static overlap buffer is required for each channel. Due to in-place processing, #timeIn and #timeOut point to identical locations. The spectral data is organized in so-called slots. Each slot contains 64 bands of complex data. The number of slots per frame depends on the frame size. For mp3PRO, there are 18 slots per frame and 6 slots per #OverlapBuffer. It is not necessary to have the slots in located consecutive address ranges. To optimize memory usage and to minimize the number of memory accesses, the memory management is organized as follows (slot numbers based on mp3PRO): 1.) Input time domain signal is located in #timeIn. The last slots (0..5) of the spectral data of the previous frame are located in the #OverlapBuffer. In addition, #frameData of the current frame resides in the upper part of #timeIn. 2.) During the cplxAnalysisQmfFiltering(), 32 samples from #timeIn are transformed into a slot of up to 32 complex spectral low band values at a time. The first spectral slot -- nr. 6 -- is written at slot number zero of #WorkBuffer2. #WorkBuffer2 will be completely filled with spectral data. 3.) LPP-Transposition in lppTransposer() is processed on 24 slots. During the transposition, the high band part of the spectral data is replicated based on the low band data. Envelope Adjustment is processed on the high band part of the spectral data only by calculateSbrEnvelope(). 4.) The cplxSynthesisQmfFiltering() creates 64 time domain samples out of a slot of 64 complex spectral values at a time. The first 6 slots in #timeOut are filled from the results of spectral slots 0..5 in the #OverlapBuffer. The consecutive slots in timeOut are now filled with the results of spectral slots 6..17. 5.) The preprocessed slots 18..23 have to be stored in the #OverlapBuffer. */ void sbr_dec( HANDLE_SBR_DEC hSbrDec, /*!< handle to Decoder channel */ LONG *timeIn, /*!< pointer to input time signal */ LONG *timeOut, /*!< pointer to output time signal */ HANDLE_SBR_DEC hSbrDecRight, /*!< handle to Decoder channel right */ LONG *timeOutRight, /*!< pointer to output time signal */ const int strideOut, /*!< Time data traversal strideOut */ HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ HANDLE_SBR_FRAME_DATA hFrameData, /*!< Control data of current frame */ HANDLE_SBR_PREV_FRAME_DATA hPrevFrameData, /*!< Some control data of last frame */ const int applyProcessing, /*!< Flag for SBR operation */ HANDLE_PS_DEC h_ps_d, const UINT flags, const int codecFrameSize, const INT sbrInDataHeadroom) { int i, slot, reserve; int saveLbScale; int lastSlotOffs; FIXP_DBL maxVal; /* temporary pointer / variable for QMF; required as we want to use temporary buffer creating one frame delay for HBE in LP mode */ LONG *pTimeInQmf = timeIn; /* Number of QMF timeslots in the overlap buffer: */ int ov_len = hSbrDec->LppTrans.pSettings->overlap; /* Number of QMF slots per frame */ int noCols = hHeaderData->numberTimeSlots * hHeaderData->timeStep; /* create pointer array for data to use for HBE and legacy sbr */ FIXP_DBL *pLowBandReal[(3 * 4) + 2 * ((1024) / (32) * (4) / 2)]; FIXP_DBL *pLowBandImag[(3 * 4) + 2 * ((1024) / (32) * (4) / 2)]; /* set pReal to where QMF analysis writes in case of legacy SBR */ FIXP_DBL **pReal = pLowBandReal + ov_len; FIXP_DBL **pImag = pLowBandImag + ov_len; /* map QMF buffer to pointer array (Overlap + Frame)*/ for (i = 0; i < noCols + ov_len; i++) { pLowBandReal[i] = hSbrDec->qmfDomainInCh->hQmfSlotsReal[i]; pLowBandImag[i] = hSbrDec->qmfDomainInCh->hQmfSlotsImag[i]; } if ((flags & SBRDEC_USAC_HARMONICSBR)) { /* in case of harmonic SBR and no HBE_LP map additional buffer for one more frame to pointer arry */ for (i = 0; i < noCols; i++) { pLowBandReal[i + noCols + ov_len] = hSbrDec->hQmfHBESlotsReal[i]; pLowBandImag[i + noCols + ov_len] = hSbrDec->hQmfHBESlotsImag[i]; } /* shift scale values according to buffer */ hSbrDec->scale_ov = hSbrDec->scale_lb; hSbrDec->scale_lb = hSbrDec->scale_hbe; /* set pReal to where QMF analysis writes in case of HBE */ pReal += noCols; pImag += noCols; if (flags & SBRDEC_SKIP_QMF_ANA) { /* stereoCfgIndex3 with HBE */ FDK_QmfDomain_QmfData2HBE(hSbrDec->qmfDomainInCh, hSbrDec->hQmfHBESlotsReal, hSbrDec->hQmfHBESlotsImag); } else { /* We have to move old hbe frame data to lb area of buffer */ for (i = 0; i < noCols; i++) { FDKmemcpy(pLowBandReal[ov_len + i], hSbrDec->hQmfHBESlotsReal[i], hHeaderData->numberOfAnalysisBands * sizeof(FIXP_DBL)); FDKmemcpy(pLowBandImag[ov_len + i], hSbrDec->hQmfHBESlotsImag[i], hHeaderData->numberOfAnalysisBands * sizeof(FIXP_DBL)); } } } /* low band codec signal subband filtering */ if (flags & SBRDEC_SKIP_QMF_ANA) { if (!(flags & SBRDEC_USAC_HARMONICSBR)) /* stereoCfgIndex3 w/o HBE */ FDK_QmfDomain_WorkBuffer2ProcChannel(hSbrDec->qmfDomainInCh); } else { C_AALLOC_SCRATCH_START(qmfTemp, FIXP_DBL, 2 * (64)); qmfAnalysisFiltering(&hSbrDec->qmfDomainInCh->fb, pReal, pImag, &hSbrDec->qmfDomainInCh->scaling, pTimeInQmf, 0 + sbrInDataHeadroom, 1, qmfTemp); C_AALLOC_SCRATCH_END(qmfTemp, FIXP_DBL, 2 * (64)); } /* Clear upper half of spectrum */ if (!((flags & SBRDEC_USAC_HARMONICSBR) && (hFrameData->sbrPatchingMode == 0))) { int nAnalysisBands = hHeaderData->numberOfAnalysisBands; if (!(flags & SBRDEC_LOW_POWER)) { for (slot = ov_len; slot < noCols + ov_len; slot++) { FDKmemclear(&pLowBandReal[slot][nAnalysisBands], ((64) - nAnalysisBands) * sizeof(FIXP_DBL)); FDKmemclear(&pLowBandImag[slot][nAnalysisBands], ((64) - nAnalysisBands) * sizeof(FIXP_DBL)); } } else { for (slot = ov_len; slot < noCols + ov_len; slot++) { FDKmemclear(&pLowBandReal[slot][nAnalysisBands], ((64) - nAnalysisBands) * sizeof(FIXP_DBL)); } } } /* Shift spectral data left to gain accuracy in transposer and adjustor */ /* Range was increased from lsb to no_channels because in some cases (e.g. USAC conf eSbr_4_Pvc.mp4 and some HBE cases) it could be observed that the signal between lsb and no_channels is used for the patching process. */ maxVal = maxSubbandSample(pReal, (flags & SBRDEC_LOW_POWER) ? NULL : pImag, 0, hSbrDec->qmfDomainInCh->fb.no_channels, 0, noCols); reserve = fixMax(0, CntLeadingZeros(maxVal) - 1); reserve = fixMin(reserve, DFRACT_BITS - 1 - hSbrDec->qmfDomainInCh->scaling.lb_scale); /* If all data is zero, lb_scale could become too large */ rescaleSubbandSamples(pReal, (flags & SBRDEC_LOW_POWER) ? NULL : pImag, 0, hSbrDec->qmfDomainInCh->fb.no_channels, 0, noCols, reserve); hSbrDec->qmfDomainInCh->scaling.lb_scale += reserve; if ((flags & SBRDEC_USAC_HARMONICSBR)) { /* actually this is our hbe_scale */ hSbrDec->scale_hbe = hSbrDec->qmfDomainInCh->scaling.lb_scale; /* the real lb_scale is stored in scale_lb from sbr */ hSbrDec->qmfDomainInCh->scaling.lb_scale = hSbrDec->scale_lb; } /* save low band scale, wavecoding or parametric stereo may modify it */ saveLbScale = hSbrDec->qmfDomainInCh->scaling.lb_scale; if (applyProcessing) { UCHAR *borders = hFrameData->frameInfo.borders; lastSlotOffs = borders[hFrameData->frameInfo.nEnvelopes] - hHeaderData->numberTimeSlots; FIXP_DBL degreeAlias[(64)]; PVC_DYNAMIC_DATA pvcDynamicData; pvcInitFrame( &hSbrDec->PvcStaticData, &pvcDynamicData, (hHeaderData->frameErrorFlag ? 0 : hHeaderData->bs_info.pvc_mode), hFrameData->ns, hHeaderData->timeStep, hHeaderData->freqBandData.lowSubband, hFrameData->frameInfo.pvcBorders[0], hFrameData->pvcID); if (!hHeaderData->frameErrorFlag && (hHeaderData->bs_info.pvc_mode > 0)) { pvcDecodeFrame(&hSbrDec->PvcStaticData, &pvcDynamicData, pLowBandReal, pLowBandImag, ov_len, SCALE2EXP(hSbrDec->qmfDomainInCh->scaling.ov_lb_scale), SCALE2EXP(hSbrDec->qmfDomainInCh->scaling.lb_scale)); } pvcEndFrame(&hSbrDec->PvcStaticData, &pvcDynamicData); /* The transposer will override most values in degreeAlias[]. The array needs to be cleared at least from lowSubband to highSubband before. */ if (flags & SBRDEC_LOW_POWER) FDKmemclear(°reeAlias[hHeaderData->freqBandData.lowSubband], (hHeaderData->freqBandData.highSubband - hHeaderData->freqBandData.lowSubband) * sizeof(FIXP_DBL)); /* Inverse filtering of lowband and transposition into the SBR-frequency range */ { KEEP_STATES_SYNCED_MODE keepStatesSyncedMode = ((flags & SBRDEC_USAC_HARMONICSBR) && (hFrameData->sbrPatchingMode != 0)) ? KEEP_STATES_SYNCED_NORMAL : KEEP_STATES_SYNCED_OFF; if (flags & SBRDEC_USAC_HARMONICSBR) { if (flags & SBRDEC_QUAD_RATE) { pReal -= 32; pImag -= 32; } if ((hSbrDec->savedStates == 0) && (hFrameData->sbrPatchingMode == 1)) { /* copy saved states from previous frame to legacy SBR lpc filterstate * buffer */ for (i = 0; i < LPC_ORDER + ov_len; i++) { FDKmemcpy( hSbrDec->LppTrans.lpcFilterStatesRealLegSBR[i], hSbrDec->codecQMFBufferReal[noCols - LPC_ORDER - ov_len + i], hSbrDec->hHBE->noChannels * sizeof(FIXP_DBL)); FDKmemcpy( hSbrDec->LppTrans.lpcFilterStatesImagLegSBR[i], hSbrDec->codecQMFBufferImag[noCols - LPC_ORDER - ov_len + i], hSbrDec->hHBE->noChannels * sizeof(FIXP_DBL)); } } /* saving unmodified QMF states in case we are switching from legacy SBR * to HBE */ for (i = 0; i < hSbrDec->hHBE->noCols; i++) { FDKmemcpy(hSbrDec->codecQMFBufferReal[i], pLowBandReal[ov_len + i], hSbrDec->hHBE->noChannels * sizeof(FIXP_DBL)); FDKmemcpy(hSbrDec->codecQMFBufferImag[i], pLowBandImag[ov_len + i], hSbrDec->hHBE->noChannels * sizeof(FIXP_DBL)); } QmfTransposerApply( hSbrDec->hHBE, pReal, pImag, noCols, pLowBandReal, pLowBandImag, hSbrDec->LppTrans.lpcFilterStatesRealHBE, hSbrDec->LppTrans.lpcFilterStatesImagHBE, hFrameData->sbrPitchInBins, hSbrDec->scale_lb, hSbrDec->scale_hbe, &hSbrDec->qmfDomainInCh->scaling.hb_scale, hHeaderData->timeStep, borders[0], ov_len, keepStatesSyncedMode); if (flags & SBRDEC_QUAD_RATE) { int *xOverQmf = GetxOverBandQmfTransposer(hSbrDec->hHBE); copyHarmonicSpectrum(xOverQmf, pLowBandReal, pLowBandImag, noCols, ov_len, keepStatesSyncedMode); } } } if ((flags & SBRDEC_USAC_HARMONICSBR) && (hFrameData->sbrPatchingMode == 0)) { hSbrDec->prev_frame_lSbr = 0; hSbrDec->prev_frame_hbeSbr = 1; lppTransposerHBE( &hSbrDec->LppTrans, hSbrDec->hHBE, &hSbrDec->qmfDomainInCh->scaling, pLowBandReal, pLowBandImag, hHeaderData->timeStep, borders[0], lastSlotOffs, hHeaderData->freqBandData.nInvfBands, hFrameData->sbr_invf_mode, hPrevFrameData->sbr_invf_mode); } else { if (flags & SBRDEC_USAC_HARMONICSBR) { for (i = 0; i < LPC_ORDER + hSbrDec->LppTrans.pSettings->overlap; i++) { /* Store the unmodified qmf Slots values for upper part of spectrum (required for LPC filtering) required if next frame is a HBE frame */ FDKmemcpy(hSbrDec->LppTrans.lpcFilterStatesRealHBE[i], hSbrDec->qmfDomainInCh ->hQmfSlotsReal[hSbrDec->hHBE->noCols - LPC_ORDER + i], (64) * sizeof(FIXP_DBL)); FDKmemcpy(hSbrDec->LppTrans.lpcFilterStatesImagHBE[i], hSbrDec->qmfDomainInCh ->hQmfSlotsImag[hSbrDec->hHBE->noCols - LPC_ORDER + i], (64) * sizeof(FIXP_DBL)); } } { hSbrDec->prev_frame_lSbr = 1; hSbrDec->prev_frame_hbeSbr = 0; } lppTransposer( &hSbrDec->LppTrans, &hSbrDec->qmfDomainInCh->scaling, pLowBandReal, degreeAlias, // only used if useLP = 1 pLowBandImag, flags & SBRDEC_LOW_POWER, hHeaderData->bs_info.sbr_preprocessing, hHeaderData->freqBandData.v_k_master[0], hHeaderData->timeStep, borders[0], lastSlotOffs, hHeaderData->freqBandData.nInvfBands, hFrameData->sbr_invf_mode, hPrevFrameData->sbr_invf_mode); } /* Adjust envelope of current frame. */ if ((hFrameData->sbrPatchingMode != hSbrDec->SbrCalculateEnvelope.sbrPatchingMode)) { ResetLimiterBands(hHeaderData->freqBandData.limiterBandTable, &hHeaderData->freqBandData.noLimiterBands, hHeaderData->freqBandData.freqBandTable[0], hHeaderData->freqBandData.nSfb[0], hSbrDec->LppTrans.pSettings->patchParam, hSbrDec->LppTrans.pSettings->noOfPatches, hHeaderData->bs_data.limiterBands, hFrameData->sbrPatchingMode, (flags & SBRDEC_USAC_HARMONICSBR) && (hFrameData->sbrPatchingMode == 0) ? GetxOverBandQmfTransposer(hSbrDec->hHBE) : NULL, Get41SbrQmfTransposer(hSbrDec->hHBE)); hSbrDec->SbrCalculateEnvelope.sbrPatchingMode = hFrameData->sbrPatchingMode; } calculateSbrEnvelope( &hSbrDec->qmfDomainInCh->scaling, &hSbrDec->SbrCalculateEnvelope, hHeaderData, hFrameData, &pvcDynamicData, pLowBandReal, pLowBandImag, flags & SBRDEC_LOW_POWER, degreeAlias, flags, (hHeaderData->frameErrorFlag || hPrevFrameData->frameErrorFlag)); #if (SBRDEC_MAX_HB_FADE_FRAMES > 0) /* Avoid hard onsets of high band */ if (hHeaderData->frameErrorFlag) { if (hSbrDec->highBandFadeCnt < SBRDEC_MAX_HB_FADE_FRAMES) { hSbrDec->highBandFadeCnt += 1; } } else { if (hSbrDec->highBandFadeCnt > 0) { /* Manipulate high band scale factor to get a smooth fade-in */ hSbrDec->qmfDomainInCh->scaling.hb_scale += hSbrDec->highBandFadeCnt; hSbrDec->qmfDomainInCh->scaling.hb_scale = fMin(hSbrDec->qmfDomainInCh->scaling.hb_scale, DFRACT_BITS - 1); hSbrDec->highBandFadeCnt -= 1; } } #endif /* Update hPrevFrameData (to be used in the next frame) */ for (i = 0; i < hHeaderData->freqBandData.nInvfBands; i++) { hPrevFrameData->sbr_invf_mode[i] = hFrameData->sbr_invf_mode[i]; } hPrevFrameData->coupling = hFrameData->coupling; hPrevFrameData->stopPos = borders[hFrameData->frameInfo.nEnvelopes]; hPrevFrameData->ampRes = hFrameData->ampResolutionCurrentFrame; hPrevFrameData->prevSbrPitchInBins = hFrameData->sbrPitchInBins; /* could be done in extractFrameInfo_pvc() but hPrevFrameData is not * available there */ FDKmemcpy(&hPrevFrameData->prevFrameInfo, &hFrameData->frameInfo, sizeof(FRAME_INFO)); } else { /* rescale from lsb to nAnalysisBands in order to compensate scaling with * hb_scale in this area, done by synthesisFiltering*/ int rescale; int lsb; int length; /* Reset hb_scale if no highband is present, because hb_scale is considered * in the QMF-synthesis */ hSbrDec->qmfDomainInCh->scaling.hb_scale = saveLbScale; rescale = hSbrDec->qmfDomainInCh->scaling.hb_scale - hSbrDec->qmfDomainInCh->scaling.ov_lb_scale; lsb = hSbrDec->qmfDomainOutCh->fb.lsb; length = (hSbrDec->qmfDomainInCh->fb.no_channels - lsb); if ((rescale < 0) && (length > 0)) { if (!(flags & SBRDEC_LOW_POWER)) { for (i = 0; i < ov_len; i++) { scaleValues(&pLowBandReal[i][lsb], length, rescale); scaleValues(&pLowBandImag[i][lsb], length, rescale); } } else { for (i = 0; i < ov_len; i++) { scaleValues(&pLowBandReal[i][lsb], length, rescale); } } } } if (!(flags & SBRDEC_USAC_HARMONICSBR)) { int length = hSbrDec->qmfDomainInCh->fb.lsb; if (flags & SBRDEC_SYNTAX_USAC) { length = hSbrDec->qmfDomainInCh->fb.no_channels; } /* in case of legacy sbr saving of filter states here */ for (i = 0; i < LPC_ORDER + ov_len; i++) { /* Store the unmodified qmf Slots values (required for LPC filtering) */ if (!(flags & SBRDEC_LOW_POWER)) { FDKmemcpy(hSbrDec->LppTrans.lpcFilterStatesRealLegSBR[i], pLowBandReal[noCols - LPC_ORDER + i], length * sizeof(FIXP_DBL)); FDKmemcpy(hSbrDec->LppTrans.lpcFilterStatesImagLegSBR[i], pLowBandImag[noCols - LPC_ORDER + i], length * sizeof(FIXP_DBL)); } else FDKmemcpy(hSbrDec->LppTrans.lpcFilterStatesRealLegSBR[i], pLowBandReal[noCols - LPC_ORDER + i], length * sizeof(FIXP_DBL)); } } /* Synthesis subband filtering. */ if (!(flags & SBRDEC_PS_DECODED)) { if (!(flags & SBRDEC_SKIP_QMF_SYN)) { int outScalefactor = -(8); if (h_ps_d != NULL) { h_ps_d->procFrameBased = 1; /* we here do frame based processing */ } sbrDecoder_drcApply(&hSbrDec->sbrDrcChannel, pLowBandReal, (flags & SBRDEC_LOW_POWER) ? NULL : pLowBandImag, hSbrDec->qmfDomainOutCh->fb.no_col, &outScalefactor); qmfChangeOutScalefactor(&hSbrDec->qmfDomainOutCh->fb, outScalefactor); { HANDLE_FREQ_BAND_DATA hFreq = &hHeaderData->freqBandData; int save_usb = hSbrDec->qmfDomainOutCh->fb.usb; #if (QMF_MAX_SYNTHESIS_BANDS <= 64) C_AALLOC_SCRATCH_START(qmfTemp, FIXP_DBL, 2 * QMF_MAX_SYNTHESIS_BANDS); #else C_AALLOC_STACK_START(qmfTemp, FIXP_DBL, 2 * QMF_MAX_SYNTHESIS_BANDS); #endif if (hSbrDec->qmfDomainOutCh->fb.usb < hFreq->ov_highSubband) { /* we need to patch usb for this frame as overlap may contain higher frequency range if headerchange occured; fb. usb is always limited to maximum fb.no_channels; In case of wrongly decoded headers it might be that ov_highSubband is higher than the number of synthesis channels (fb.no_channels), which is forbidden, therefore we need to limit ov_highSubband with fMin function to avoid not allowed usb in synthesis filterbank. */ hSbrDec->qmfDomainOutCh->fb.usb = fMin((UINT)hFreq->ov_highSubband, (UINT)hSbrDec->qmfDomainOutCh->fb.no_channels); } { qmfSynthesisFiltering( &hSbrDec->qmfDomainOutCh->fb, pLowBandReal, (flags & SBRDEC_LOW_POWER) ? NULL : pLowBandImag, &hSbrDec->qmfDomainInCh->scaling, hSbrDec->LppTrans.pSettings->overlap, timeOut, strideOut, qmfTemp); } /* restore saved value */ hSbrDec->qmfDomainOutCh->fb.usb = save_usb; hFreq->ov_highSubband = save_usb; #if (QMF_MAX_SYNTHESIS_BANDS <= 64) C_AALLOC_SCRATCH_END(qmfTemp, FIXP_DBL, 2 * QMF_MAX_SYNTHESIS_BANDS); #else C_AALLOC_STACK_END(qmfTemp, FIXP_DBL, 2 * QMF_MAX_SYNTHESIS_BANDS); #endif } } } else { /* (flags & SBRDEC_PS_DECODED) */ INT sdiff; INT scaleFactorHighBand, scaleFactorLowBand_ov, scaleFactorLowBand_no_ov; HANDLE_QMF_FILTER_BANK synQmf = &hSbrDec->qmfDomainOutCh->fb; HANDLE_QMF_FILTER_BANK synQmfRight = &hSbrDecRight->qmfDomainOutCh->fb; /* adapt scaling */ sdiff = hSbrDec->qmfDomainInCh->scaling.lb_scale - reserve; /* Scaling difference */ scaleFactorHighBand = sdiff - hSbrDec->qmfDomainInCh->scaling.hb_scale; scaleFactorLowBand_ov = sdiff - hSbrDec->qmfDomainInCh->scaling.ov_lb_scale; scaleFactorLowBand_no_ov = sdiff - hSbrDec->qmfDomainInCh->scaling.lb_scale; /* Scale of low band overlapping QMF data */ scaleFactorLowBand_ov = fMin(DFRACT_BITS - 1, fMax(-(DFRACT_BITS - 1), scaleFactorLowBand_ov)); /* Scale of low band current QMF data */ scaleFactorLowBand_no_ov = fMin( DFRACT_BITS - 1, fMax(-(DFRACT_BITS - 1), scaleFactorLowBand_no_ov)); /* Scale of current high band */ scaleFactorHighBand = fMin(DFRACT_BITS - 1, fMax(-(DFRACT_BITS - 1), scaleFactorHighBand)); if (h_ps_d->procFrameBased == 1) /* If we have switched from frame to slot based processing copy filter states */ { /* procFrameBased will be unset later */ /* copy filter states from left to right */ /* was ((640)-(64))*sizeof(FIXP_QSS) flexible amount of synthesis bands needed for QMF based resampling */ FDK_ASSERT(hSbrDec->qmfDomainInCh->pGlobalConf->nBandsSynthesis <= QMF_MAX_SYNTHESIS_BANDS); qmfChangeOutScalefactor(synQmfRight, -(8)); FDKmemcpy(synQmfRight->FilterStates, synQmf->FilterStates, 9 * hSbrDec->qmfDomainInCh->pGlobalConf->nBandsSynthesis * sizeof(FIXP_QSS)); } /* Feed delaylines when parametric stereo is switched on. */ PreparePsProcessing(h_ps_d, pLowBandReal, pLowBandImag, scaleFactorLowBand_ov); /* use the same synthese qmf values for left and right channel */ synQmfRight->no_col = synQmf->no_col; synQmfRight->lsb = synQmf->lsb; synQmfRight->usb = synQmf->usb; int env = 0; { #if (QMF_MAX_SYNTHESIS_BANDS <= 64) C_AALLOC_SCRATCH_START(pWorkBuffer, FIXP_DBL, 2 * QMF_MAX_SYNTHESIS_BANDS); #else C_AALLOC_STACK_START(pWorkBuffer, FIXP_DBL, 2 * QMF_MAX_SYNTHESIS_BANDS); #endif int maxShift = 0; if (hSbrDec->sbrDrcChannel.enable != 0) { if (hSbrDec->sbrDrcChannel.prevFact_exp > maxShift) { maxShift = hSbrDec->sbrDrcChannel.prevFact_exp; } if (hSbrDec->sbrDrcChannel.currFact_exp > maxShift) { maxShift = hSbrDec->sbrDrcChannel.currFact_exp; } if (hSbrDec->sbrDrcChannel.nextFact_exp > maxShift) { maxShift = hSbrDec->sbrDrcChannel.nextFact_exp; } } /* copy DRC data to right channel (with PS both channels use the same DRC * gains) */ FDKmemcpy(&hSbrDecRight->sbrDrcChannel, &hSbrDec->sbrDrcChannel, sizeof(SBRDEC_DRC_CHANNEL)); for (i = 0; i < synQmf->no_col; i++) { /* ----- no_col loop ----- */ INT outScalefactorR, outScalefactorL; /* qmf timeslot of right channel */ FIXP_DBL *rQmfReal = pWorkBuffer; FIXP_DBL *rQmfImag = pWorkBuffer + synQmf->no_channels; { if (i == h_ps_d->bsData[h_ps_d->processSlot].mpeg.aEnvStartStop[env]) { initSlotBasedRotation(h_ps_d, env, hHeaderData->freqBandData.highSubband); env++; } ApplyPsSlot( h_ps_d, /* parametric stereo decoder handle */ (pLowBandReal + i), /* one timeslot of left/mono channel */ (pLowBandImag + i), /* one timeslot of left/mono channel */ rQmfReal, /* one timeslot or right channel */ rQmfImag, /* one timeslot or right channel */ scaleFactorLowBand_no_ov, (i < hSbrDec->LppTrans.pSettings->overlap) ? scaleFactorLowBand_ov : scaleFactorLowBand_no_ov, scaleFactorHighBand, synQmf->lsb, synQmf->usb); outScalefactorL = outScalefactorR = 1 + sbrInDataHeadroom; /* psDiffScale! (MPEG-PS) */ } sbrDecoder_drcApplySlot(/* right channel */ &hSbrDecRight->sbrDrcChannel, rQmfReal, rQmfImag, i, synQmfRight->no_col, maxShift); outScalefactorR += maxShift; sbrDecoder_drcApplySlot(/* left channel */ &hSbrDec->sbrDrcChannel, *(pLowBandReal + i), *(pLowBandImag + i), i, synQmf->no_col, maxShift); outScalefactorL += maxShift; if (!(flags & SBRDEC_SKIP_QMF_SYN)) { qmfChangeOutScalefactor(synQmf, -(8)); qmfChangeOutScalefactor(synQmfRight, -(8)); qmfSynthesisFilteringSlot( synQmfRight, rQmfReal, /* QMF real buffer */ rQmfImag, /* QMF imag buffer */ outScalefactorL, outScalefactorL, timeOutRight + (i * synQmf->no_channels * strideOut), strideOut, pWorkBuffer); qmfSynthesisFilteringSlot( synQmf, *(pLowBandReal + i), /* QMF real buffer */ *(pLowBandImag + i), /* QMF imag buffer */ outScalefactorR, outScalefactorR, timeOut + (i * synQmf->no_channels * strideOut), strideOut, pWorkBuffer); } } /* no_col loop i */ #if (QMF_MAX_SYNTHESIS_BANDS <= 64) C_AALLOC_SCRATCH_END(pWorkBuffer, FIXP_DBL, 2 * QMF_MAX_SYNTHESIS_BANDS); #else C_AALLOC_STACK_END(pWorkBuffer, FIXP_DBL, 2 * QMF_MAX_SYNTHESIS_BANDS); #endif } } sbrDecoder_drcUpdateChannel(&hSbrDec->sbrDrcChannel); /* Update overlap buffer Even bands above usb are copied to avoid outdated spectral data in case the stop frequency raises. */ if (!(flags & SBRDEC_SKIP_QMF_SYN)) { { FDK_QmfDomain_SaveOverlap(hSbrDec->qmfDomainInCh, 0); FDK_ASSERT(hSbrDec->qmfDomainInCh->scaling.ov_lb_scale == saveLbScale); } } hSbrDec->savedStates = 0; /* Save current frame status */ hPrevFrameData->frameErrorFlag = hHeaderData->frameErrorFlag; hSbrDec->applySbrProc_old = applyProcessing; } /* sbr_dec() */ /*! \brief Creates sbr decoder structure \return errorCode, 0 if successful */ SBR_ERROR createSbrDec(SBR_CHANNEL *hSbrChannel, HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ TRANSPOSER_SETTINGS *pSettings, const int downsampleFac, /*!< Downsampling factor */ const UINT qmfFlags, /*!< flags -> 1: HQ/LP selector, 2: CLDFB */ const UINT flags, const int overlap, int chan, /*!< Channel for which to assign buffers etc. */ int codecFrameSize) { SBR_ERROR err = SBRDEC_OK; int timeSlots = hHeaderData->numberTimeSlots; /* Number of SBR slots per frame */ int noCols = timeSlots * hHeaderData->timeStep; /* Number of QMF slots per frame */ HANDLE_SBR_DEC hs = &(hSbrChannel->SbrDec); #if (SBRDEC_MAX_HB_FADE_FRAMES > 0) hs->highBandFadeCnt = SBRDEC_MAX_HB_FADE_FRAMES; #endif hs->scale_hbe = 15; hs->scale_lb = 15; hs->scale_ov = 15; hs->prev_frame_lSbr = 0; hs->prev_frame_hbeSbr = 0; hs->codecFrameSize = codecFrameSize; /* create envelope calculator */ err = createSbrEnvelopeCalc(&hs->SbrCalculateEnvelope, hHeaderData, chan, flags); if (err != SBRDEC_OK) { return err; } initSbrPrevFrameData(&hSbrChannel->prevFrameData, timeSlots); /* create transposer */ err = createLppTransposer( &hs->LppTrans, pSettings, hHeaderData->freqBandData.lowSubband, hHeaderData->freqBandData.v_k_master, hHeaderData->freqBandData.numMaster, hHeaderData->freqBandData.highSubband, timeSlots, noCols, hHeaderData->freqBandData.freqBandTableNoise, hHeaderData->freqBandData.nNfb, hHeaderData->sbrProcSmplRate, chan, overlap); if (err != SBRDEC_OK) { return err; } if (flags & SBRDEC_USAC_HARMONICSBR) { int noChannels, bSbr41 = flags & SBRDEC_QUAD_RATE ? 1 : 0; noChannels = QMF_SYNTH_CHANNELS / ((bSbr41 + 1) * 2); /* 32 for (32:64 and 24:64) and 16 for 16:64 */ /* shared memory between hbeLightTimeDelayBuffer and hQmfHBESlotsReal if * SBRDEC_HBE_ENABLE */ hSbrChannel->SbrDec.tmp_memory = (FIXP_DBL **)fdkCallocMatrix2D_aligned( noCols, noChannels, sizeof(FIXP_DBL)); if (hSbrChannel->SbrDec.tmp_memory == NULL) { return SBRDEC_MEM_ALLOC_FAILED; } hSbrChannel->SbrDec.hQmfHBESlotsReal = hSbrChannel->SbrDec.tmp_memory; hSbrChannel->SbrDec.hQmfHBESlotsImag = (FIXP_DBL **)fdkCallocMatrix2D_aligned(noCols, noChannels, sizeof(FIXP_DBL)); if (hSbrChannel->SbrDec.hQmfHBESlotsImag == NULL) { return SBRDEC_MEM_ALLOC_FAILED; } /* buffers containing unmodified qmf data; required when switching from * legacy SBR to HBE */ /* buffer can be used as LPCFilterstates buffer because legacy SBR needs * exactly these values for LPC filtering */ hSbrChannel->SbrDec.codecQMFBufferReal = (FIXP_DBL **)fdkCallocMatrix2D_aligned(noCols, noChannels, sizeof(FIXP_DBL)); if (hSbrChannel->SbrDec.codecQMFBufferReal == NULL) { return SBRDEC_MEM_ALLOC_FAILED; } hSbrChannel->SbrDec.codecQMFBufferImag = (FIXP_DBL **)fdkCallocMatrix2D_aligned(noCols, noChannels, sizeof(FIXP_DBL)); if (hSbrChannel->SbrDec.codecQMFBufferImag == NULL) { return SBRDEC_MEM_ALLOC_FAILED; } err = QmfTransposerCreate(&hs->hHBE, codecFrameSize, 0, bSbr41); if (err != SBRDEC_OK) { return err; } } return err; } /*! \brief Delete sbr decoder structure \return errorCode, 0 if successful */ int deleteSbrDec(SBR_CHANNEL *hSbrChannel) { HANDLE_SBR_DEC hs = &hSbrChannel->SbrDec; deleteSbrEnvelopeCalc(&hs->SbrCalculateEnvelope); if (hs->tmp_memory != NULL) { FDK_FREE_MEMORY_2D_ALIGNED(hs->tmp_memory); } /* modify here */ FDK_FREE_MEMORY_2D_ALIGNED(hs->hQmfHBESlotsImag); if (hs->hHBE != NULL) QmfTransposerClose(hs->hHBE); if (hs->codecQMFBufferReal != NULL) { FDK_FREE_MEMORY_2D_ALIGNED(hs->codecQMFBufferReal); } if (hs->codecQMFBufferImag != NULL) { FDK_FREE_MEMORY_2D_ALIGNED(hs->codecQMFBufferImag); } return 0; } /*! \brief resets sbr decoder structure \return errorCode, 0 if successful */ SBR_ERROR resetSbrDec(HANDLE_SBR_DEC hSbrDec, HANDLE_SBR_HEADER_DATA hHeaderData, HANDLE_SBR_PREV_FRAME_DATA hPrevFrameData, const int downsampleFac, const UINT flags, HANDLE_SBR_FRAME_DATA hFrameData) { SBR_ERROR sbrError = SBRDEC_OK; int i; FIXP_DBL *pLowBandReal[128]; FIXP_DBL *pLowBandImag[128]; int useLP = flags & SBRDEC_LOW_POWER; int old_lsb = hSbrDec->qmfDomainInCh->fb.lsb; int old_usb = hSbrDec->qmfDomainInCh->fb.usb; int new_lsb = hHeaderData->freqBandData.lowSubband; /* int new_usb = hHeaderData->freqBandData.highSubband; */ int l, startBand, stopBand, startSlot, size; FIXP_DBL **OverlapBufferReal = hSbrDec->qmfDomainInCh->hQmfSlotsReal; FIXP_DBL **OverlapBufferImag = hSbrDec->qmfDomainInCh->hQmfSlotsImag; /* in case the previous frame was not active in terms of SBR processing, the full band from 0 to no_channels was rescaled and not overwritten. Thats why the scaling factor lb_scale can be seen as assigned to all bands from 0 to no_channels in the previous frame. The same states for the current frame if the current frame is not active in terms of SBR processing */ int applySbrProc = (hHeaderData->syncState == SBR_ACTIVE || (hHeaderData->frameErrorFlag == 0 && hHeaderData->syncState == SBR_HEADER)); int applySbrProc_old = hSbrDec->applySbrProc_old; if (!applySbrProc) { new_lsb = (hSbrDec->qmfDomainInCh->fb).no_channels; } if (!applySbrProc_old) { old_lsb = (hSbrDec->qmfDomainInCh->fb).no_channels; old_usb = old_lsb; } resetSbrEnvelopeCalc(&hSbrDec->SbrCalculateEnvelope); /* Change lsb and usb */ /* Synthesis */ FDK_ASSERT(hSbrDec->qmfDomainOutCh != NULL); hSbrDec->qmfDomainOutCh->fb.lsb = fixMin((INT)hSbrDec->qmfDomainOutCh->fb.no_channels, (INT)hHeaderData->freqBandData.lowSubband); hSbrDec->qmfDomainOutCh->fb.usb = fixMin((INT)hSbrDec->qmfDomainOutCh->fb.no_channels, (INT)hHeaderData->freqBandData.highSubband); /* Analysis */ FDK_ASSERT(hSbrDec->qmfDomainInCh != NULL); hSbrDec->qmfDomainInCh->fb.lsb = hSbrDec->qmfDomainOutCh->fb.lsb; hSbrDec->qmfDomainInCh->fb.usb = hSbrDec->qmfDomainOutCh->fb.usb; /* The following initialization of spectral data in the overlap buffer is required for dynamic x-over or a change of the start-freq for 2 reasons: 1. If the lowband gets _wider_, unadjusted data would remain 2. If the lowband becomes _smaller_, the highest bands of the old lowband must be cleared because the whitening would be affected */ startBand = old_lsb; stopBand = new_lsb; startSlot = fMax(0, hHeaderData->timeStep * (hPrevFrameData->stopPos - hHeaderData->numberTimeSlots)); size = fMax(0, stopBand - startBand); /* in case of USAC we don't want to zero out the memory, as this can lead to holes in the spectrum; fix shall only be applied for USAC not for MPEG-4 SBR, in this case setting zero remains */ if (!(flags & SBRDEC_SYNTAX_USAC)) { /* keep already adjusted data in the x-over-area */ if (!useLP) { for (l = startSlot; l < hSbrDec->LppTrans.pSettings->overlap; l++) { FDKmemclear(&OverlapBufferReal[l][startBand], size * sizeof(FIXP_DBL)); FDKmemclear(&OverlapBufferImag[l][startBand], size * sizeof(FIXP_DBL)); } } else { for (l = startSlot; l < hSbrDec->LppTrans.pSettings->overlap; l++) { FDKmemclear(&OverlapBufferReal[l][startBand], size * sizeof(FIXP_DBL)); } } /* reset LPC filter states */ startBand = fixMin(old_lsb, new_lsb); stopBand = fixMax(old_lsb, new_lsb); size = fixMax(0, stopBand - startBand); FDKmemclear(&hSbrDec->LppTrans.lpcFilterStatesRealLegSBR[0][startBand], size * sizeof(FIXP_DBL)); FDKmemclear(&hSbrDec->LppTrans.lpcFilterStatesRealLegSBR[1][startBand], size * sizeof(FIXP_DBL)); if (!useLP) { FDKmemclear(&hSbrDec->LppTrans.lpcFilterStatesImagLegSBR[0][startBand], size * sizeof(FIXP_DBL)); FDKmemclear(&hSbrDec->LppTrans.lpcFilterStatesImagLegSBR[1][startBand], size * sizeof(FIXP_DBL)); } } if (startSlot != 0) { int source_exp, target_exp, delta_exp, target_lsb, target_usb, reserve; FIXP_DBL maxVal; /* Rescale already processed spectral data between old and new x-over frequency. This must be done because of the separate scalefactors for lowband and highband. */ /* We have four relevant transitions to cover: 1. old_usb is lower than new_lsb; old SBR area is completely below new SBR area. -> entire old area was highband and belongs to lowband now and has to be rescaled. 2. old_lsb is higher than new_usb; new SBR area is completely below old SBR area. -> old area between new_lsb and old_lsb was lowband and belongs to highband now and has to be rescaled to match new highband scale. 3. old_lsb is lower and old_usb is higher than new_lsb; old and new SBR areas overlap. -> old area between old_lsb and new_lsb was highband and belongs to lowband now and has to be rescaled to match new lowband scale. 4. new_lsb is lower and new_usb_is higher than old_lsb; old and new SBR areas overlap. -> old area between new_lsb and old_usb was lowband and belongs to highband now and has to be rescaled to match new highband scale. */ if (new_lsb > old_lsb) { /* case 1 and 3 */ source_exp = SCALE2EXP(hSbrDec->qmfDomainInCh->scaling.ov_hb_scale); target_exp = SCALE2EXP(hSbrDec->qmfDomainInCh->scaling.ov_lb_scale); startBand = old_lsb; if (new_lsb >= old_usb) { /* case 1 */ stopBand = old_usb; } else { /* case 3 */ stopBand = new_lsb; } target_lsb = 0; target_usb = old_lsb; } else { /* case 2 and 4 */ source_exp = SCALE2EXP(hSbrDec->qmfDomainInCh->scaling.ov_lb_scale); target_exp = SCALE2EXP(hSbrDec->qmfDomainInCh->scaling.ov_hb_scale); startBand = new_lsb; stopBand = old_lsb; target_lsb = old_lsb; target_usb = old_usb; } maxVal = maxSubbandSample(OverlapBufferReal, (useLP) ? NULL : OverlapBufferImag, startBand, stopBand, 0, startSlot); reserve = ((LONG)maxVal != 0 ? CntLeadingZeros(maxVal) - 1 : 0); reserve = fixMin( reserve, DFRACT_BITS - 1 - EXP2SCALE( source_exp)); /* what is this line for, why do we need it? */ /* process only if x-over-area is not dominant after rescale; otherwise I'm not sure if all buffers are scaled correctly; */ if (target_exp - (source_exp - reserve) >= 0) { rescaleSubbandSamples(OverlapBufferReal, (useLP) ? NULL : OverlapBufferImag, startBand, stopBand, 0, startSlot, reserve); source_exp -= reserve; } delta_exp = target_exp - source_exp; if (delta_exp < 0) { /* x-over-area is dominant */ startBand = target_lsb; stopBand = target_usb; delta_exp = -delta_exp; if (new_lsb > old_lsb) { /* The lowband has to be rescaled */ hSbrDec->qmfDomainInCh->scaling.ov_lb_scale = EXP2SCALE(source_exp); } else { /* The highband has to be rescaled */ hSbrDec->qmfDomainInCh->scaling.ov_hb_scale = EXP2SCALE(source_exp); } } FDK_ASSERT(startBand <= stopBand); if (!useLP) { for (l = 0; l < startSlot; l++) { scaleValues(OverlapBufferReal[l] + startBand, stopBand - startBand, -delta_exp); scaleValues(OverlapBufferImag[l] + startBand, stopBand - startBand, -delta_exp); } } else for (l = 0; l < startSlot; l++) { scaleValues(OverlapBufferReal[l] + startBand, stopBand - startBand, -delta_exp); } } /* startSlot != 0 */ /* Initialize transposer and limiter */ sbrError = resetLppTransposer( &hSbrDec->LppTrans, hHeaderData->freqBandData.lowSubband, hHeaderData->freqBandData.v_k_master, hHeaderData->freqBandData.numMaster, hHeaderData->freqBandData.freqBandTableNoise, hHeaderData->freqBandData.nNfb, hHeaderData->freqBandData.highSubband, hHeaderData->sbrProcSmplRate); if (sbrError != SBRDEC_OK) return sbrError; hSbrDec->savedStates = 0; if ((flags & SBRDEC_USAC_HARMONICSBR) && applySbrProc) { sbrError = QmfTransposerReInit(hSbrDec->hHBE, hHeaderData->freqBandData.freqBandTable, hHeaderData->freqBandData.nSfb); if (sbrError != SBRDEC_OK) return sbrError; /* copy saved states from previous frame to legacy SBR lpc filterstate * buffer */ for (i = 0; i < LPC_ORDER + hSbrDec->LppTrans.pSettings->overlap; i++) { FDKmemcpy( hSbrDec->LppTrans.lpcFilterStatesRealLegSBR[i], hSbrDec->codecQMFBufferReal[hSbrDec->hHBE->noCols - LPC_ORDER - hSbrDec->LppTrans.pSettings->overlap + i], hSbrDec->hHBE->noChannels * sizeof(FIXP_DBL)); FDKmemcpy( hSbrDec->LppTrans.lpcFilterStatesImagLegSBR[i], hSbrDec->codecQMFBufferImag[hSbrDec->hHBE->noCols - LPC_ORDER - hSbrDec->LppTrans.pSettings->overlap + i], hSbrDec->hHBE->noChannels * sizeof(FIXP_DBL)); } hSbrDec->savedStates = 1; { /* map QMF buffer to pointer array (Overlap + Frame)*/ for (i = 0; i < hSbrDec->LppTrans.pSettings->overlap + LPC_ORDER; i++) { pLowBandReal[i] = hSbrDec->LppTrans.lpcFilterStatesRealHBE[i]; pLowBandImag[i] = hSbrDec->LppTrans.lpcFilterStatesImagHBE[i]; } /* map QMF buffer to pointer array (Overlap + Frame)*/ for (i = 0; i < hSbrDec->hHBE->noCols; i++) { pLowBandReal[i + hSbrDec->LppTrans.pSettings->overlap + LPC_ORDER] = hSbrDec->codecQMFBufferReal[i]; pLowBandImag[i + hSbrDec->LppTrans.pSettings->overlap + LPC_ORDER] = hSbrDec->codecQMFBufferImag[i]; } if (flags & SBRDEC_QUAD_RATE) { if (hFrameData->sbrPatchingMode == 0) { int *xOverQmf = GetxOverBandQmfTransposer(hSbrDec->hHBE); /* in case of harmonic SBR and no HBE_LP map additional buffer for one more frame to pointer arry */ for (i = 0; i < hSbrDec->hHBE->noCols / 2; i++) { pLowBandReal[i + hSbrDec->hHBE->noCols + hSbrDec->LppTrans.pSettings->overlap + LPC_ORDER] = hSbrDec->hQmfHBESlotsReal[i]; pLowBandImag[i + hSbrDec->hHBE->noCols + hSbrDec->LppTrans.pSettings->overlap + LPC_ORDER] = hSbrDec->hQmfHBESlotsImag[i]; } QmfTransposerApply( hSbrDec->hHBE, pLowBandReal + hSbrDec->LppTrans.pSettings->overlap + hSbrDec->hHBE->noCols / 2 + LPC_ORDER, pLowBandImag + hSbrDec->LppTrans.pSettings->overlap + hSbrDec->hHBE->noCols / 2 + LPC_ORDER, hSbrDec->hHBE->noCols, pLowBandReal, pLowBandImag, hSbrDec->LppTrans.lpcFilterStatesRealHBE, hSbrDec->LppTrans.lpcFilterStatesImagHBE, hPrevFrameData->prevSbrPitchInBins, hSbrDec->scale_lb, hSbrDec->scale_hbe, &hSbrDec->qmfDomainInCh->scaling.hb_scale, hHeaderData->timeStep, hFrameData->frameInfo.borders[0], hSbrDec->LppTrans.pSettings->overlap, KEEP_STATES_SYNCED_OUTDIFF); copyHarmonicSpectrum( xOverQmf, pLowBandReal, pLowBandImag, hSbrDec->hHBE->noCols, hSbrDec->LppTrans.pSettings->overlap, KEEP_STATES_SYNCED_OUTDIFF); } } else { /* in case of harmonic SBR and no HBE_LP map additional buffer for one more frame to pointer arry */ for (i = 0; i < hSbrDec->hHBE->noCols; i++) { pLowBandReal[i + hSbrDec->hHBE->noCols + hSbrDec->LppTrans.pSettings->overlap + LPC_ORDER] = hSbrDec->hQmfHBESlotsReal[i]; pLowBandImag[i + hSbrDec->hHBE->noCols + hSbrDec->LppTrans.pSettings->overlap + LPC_ORDER] = hSbrDec->hQmfHBESlotsImag[i]; } if (hFrameData->sbrPatchingMode == 0) { QmfTransposerApply( hSbrDec->hHBE, pLowBandReal + hSbrDec->LppTrans.pSettings->overlap + LPC_ORDER, pLowBandImag + hSbrDec->LppTrans.pSettings->overlap + LPC_ORDER, hSbrDec->hHBE->noCols, pLowBandReal, pLowBandImag, hSbrDec->LppTrans.lpcFilterStatesRealHBE, hSbrDec->LppTrans.lpcFilterStatesImagHBE, 0 /* not required for keeping states updated in this frame*/, hSbrDec->scale_lb, hSbrDec->scale_lb, &hSbrDec->qmfDomainInCh->scaling.hb_scale, hHeaderData->timeStep, hFrameData->frameInfo.borders[0], hSbrDec->LppTrans.pSettings->overlap, KEEP_STATES_SYNCED_NOOUT); } QmfTransposerApply( hSbrDec->hHBE, pLowBandReal + hSbrDec->LppTrans.pSettings->overlap + hSbrDec->hHBE->noCols + LPC_ORDER, pLowBandImag + hSbrDec->LppTrans.pSettings->overlap + hSbrDec->hHBE->noCols + LPC_ORDER, hSbrDec->hHBE->noCols, pLowBandReal, pLowBandImag, hSbrDec->LppTrans.lpcFilterStatesRealHBE, hSbrDec->LppTrans.lpcFilterStatesImagHBE, hPrevFrameData->prevSbrPitchInBins, hSbrDec->scale_lb, hSbrDec->scale_hbe, &hSbrDec->qmfDomainInCh->scaling.hb_scale, hHeaderData->timeStep, hFrameData->frameInfo.borders[0], hSbrDec->LppTrans.pSettings->overlap, KEEP_STATES_SYNCED_OUTDIFF); } if (hFrameData->sbrPatchingMode == 0) { for (i = startSlot; i < hSbrDec->LppTrans.pSettings->overlap; i++) { /* Store the unmodified qmf Slots values for upper part of spectrum (required for LPC filtering) required if next frame is a HBE frame */ FDKmemcpy(hSbrDec->qmfDomainInCh->hQmfSlotsReal[i], hSbrDec->LppTrans.lpcFilterStatesRealHBE[i + LPC_ORDER], (64) * sizeof(FIXP_DBL)); FDKmemcpy(hSbrDec->qmfDomainInCh->hQmfSlotsImag[i], hSbrDec->LppTrans.lpcFilterStatesImagHBE[i + LPC_ORDER], (64) * sizeof(FIXP_DBL)); } for (i = startSlot; i < hSbrDec->LppTrans.pSettings->overlap; i++) { /* Store the unmodified qmf Slots values for upper part of spectrum (required for LPC filtering) required if next frame is a HBE frame */ FDKmemcpy( hSbrDec->qmfDomainInCh->hQmfSlotsReal[i], hSbrDec->codecQMFBufferReal[hSbrDec->hHBE->noCols - hSbrDec->LppTrans.pSettings->overlap + i], new_lsb * sizeof(FIXP_DBL)); FDKmemcpy( hSbrDec->qmfDomainInCh->hQmfSlotsImag[i], hSbrDec->codecQMFBufferImag[hSbrDec->hHBE->noCols - hSbrDec->LppTrans.pSettings->overlap + i], new_lsb * sizeof(FIXP_DBL)); } } } } { int adapt_lb = 0, diff = 0, new_scale = hSbrDec->qmfDomainInCh->scaling.ov_lb_scale; if ((hSbrDec->qmfDomainInCh->scaling.ov_lb_scale != hSbrDec->qmfDomainInCh->scaling.lb_scale) && startSlot != 0) { /* we need to adapt spectrum to have equal scale factor, always larger * than zero */ diff = SCALE2EXP(hSbrDec->qmfDomainInCh->scaling.ov_lb_scale) - SCALE2EXP(hSbrDec->qmfDomainInCh->scaling.lb_scale); if (diff > 0) { adapt_lb = 1; diff = -diff; new_scale = hSbrDec->qmfDomainInCh->scaling.ov_lb_scale; } stopBand = new_lsb; } if (hFrameData->sbrPatchingMode == 1) { /* scale states from LegSBR filterstates buffer */ for (i = 0; i < hSbrDec->LppTrans.pSettings->overlap + LPC_ORDER; i++) { scaleValues(hSbrDec->LppTrans.lpcFilterStatesRealLegSBR[i], new_lsb, diff); if (!useLP) { scaleValues(hSbrDec->LppTrans.lpcFilterStatesImagLegSBR[i], new_lsb, diff); } } if (flags & SBRDEC_SYNTAX_USAC) { /* get missing states between old and new x_over from LegSBR * filterstates buffer */ /* in case of legacy SBR we leave these values zeroed out */ for (i = startSlot; i < hSbrDec->LppTrans.pSettings->overlap; i++) { FDKmemcpy(&OverlapBufferReal[i][old_lsb], &hSbrDec->LppTrans .lpcFilterStatesRealLegSBR[LPC_ORDER + i][old_lsb], fMax(new_lsb - old_lsb, 0) * sizeof(FIXP_DBL)); if (!useLP) { FDKmemcpy(&OverlapBufferImag[i][old_lsb], &hSbrDec->LppTrans .lpcFilterStatesImagLegSBR[LPC_ORDER + i][old_lsb], fMax(new_lsb - old_lsb, 0) * sizeof(FIXP_DBL)); } } } if (new_lsb > old_lsb) { stopBand = old_lsb; } } if ((adapt_lb == 1) && (stopBand > startBand)) { for (l = startSlot; l < hSbrDec->LppTrans.pSettings->overlap; l++) { scaleValues(OverlapBufferReal[l] + startBand, stopBand - startBand, diff); if (!useLP) { scaleValues(OverlapBufferImag[l] + startBand, stopBand - startBand, diff); } } } hSbrDec->qmfDomainInCh->scaling.ov_lb_scale = new_scale; } sbrError = ResetLimiterBands(hHeaderData->freqBandData.limiterBandTable, &hHeaderData->freqBandData.noLimiterBands, hHeaderData->freqBandData.freqBandTable[0], hHeaderData->freqBandData.nSfb[0], hSbrDec->LppTrans.pSettings->patchParam, hSbrDec->LppTrans.pSettings->noOfPatches, hHeaderData->bs_data.limiterBands, hFrameData->sbrPatchingMode, GetxOverBandQmfTransposer(hSbrDec->hHBE), Get41SbrQmfTransposer(hSbrDec->hHBE)); hSbrDec->SbrCalculateEnvelope.sbrPatchingMode = hFrameData->sbrPatchingMode; return sbrError; } fdk-aac-2.0.2/libSBRdec/src/sbr_dec.h000066400000000000000000000170211404225310700171320ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Sbr decoder */ #ifndef SBR_DEC_H #define SBR_DEC_H #include "sbrdecoder.h" #include "lpp_tran.h" #include "qmf.h" #include "env_calc.h" #include "FDK_audio.h" #include "sbrdec_drc.h" #include "pvc_dec.h" #include "hbe.h" enum SBRDEC_QMF_SKIP { qmfSkipNothing = 0, qmfSkipAnalysis = 1 << 0, qmfSkipSynthesis = 1 << 1 }; typedef struct { SBR_CALCULATE_ENVELOPE SbrCalculateEnvelope; SBR_LPP_TRANS LppTrans; PVC_STATIC_DATA PvcStaticData; /* do scale handling in sbr an not in qmf */ SHORT scale_ov; SHORT scale_lb; SHORT scale_hbe; SHORT prev_frame_lSbr; SHORT prev_frame_hbeSbr; int codecFrameSize; HANDLE_HBE_TRANSPOSER hHBE; HANDLE_FDK_QMF_DOMAIN_IN qmfDomainInCh; HANDLE_FDK_QMF_DOMAIN_OUT qmfDomainOutCh; SBRDEC_DRC_CHANNEL sbrDrcChannel; #if (SBRDEC_MAX_HB_FADE_FRAMES > 0) INT highBandFadeCnt; /* counter for fading in high-band signal smoothly */ #endif FIXP_DBL **tmp_memory; /* shared memory between hbeLightTimeDelayBuffer and hQmfHBESlotsReal */ FIXP_DBL **hQmfHBESlotsReal; FIXP_DBL **hQmfHBESlotsImag; FIXP_DBL **codecQMFBufferReal; FIXP_DBL **codecQMFBufferImag; UCHAR savedStates; int applySbrProc_old; } SBR_DEC; typedef SBR_DEC *HANDLE_SBR_DEC; typedef struct { SBR_FRAME_DATA frameData[(1) + 1]; SBR_PREV_FRAME_DATA prevFrameData; SBR_DEC SbrDec; } SBR_CHANNEL; typedef SBR_CHANNEL *HANDLE_SBR_CHANNEL; void sbr_dec( HANDLE_SBR_DEC hSbrDec, /*!< handle to Decoder channel */ LONG *timeIn, /*!< pointer to input time signal */ LONG *timeOut, /*!< pointer to output time signal */ HANDLE_SBR_DEC hSbrDecRight, /*!< handle to Decoder channel right */ LONG *timeOutRight, /*!< pointer to output time signal */ INT strideOut, /*!< Time data traversal strideOut */ HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ HANDLE_SBR_FRAME_DATA hFrameData, /*!< Control data of current frame */ HANDLE_SBR_PREV_FRAME_DATA hPrevFrameData, /*!< Some control data of last frame */ const int applyProcessing, /*!< Flag for SBR operation */ HANDLE_PS_DEC h_ps_d, const UINT flags, const int codecFrameSize, const INT sbrInDataHeadroom); SBR_ERROR createSbrDec(SBR_CHANNEL *hSbrChannel, HANDLE_SBR_HEADER_DATA hHeaderData, TRANSPOSER_SETTINGS *pSettings, const int downsampleFac, const UINT qmfFlags, const UINT flags, const int overlap, int chan, int codecFrameSize); int deleteSbrDec(SBR_CHANNEL *hSbrChannel); SBR_ERROR resetSbrDec(HANDLE_SBR_DEC hSbrDec, HANDLE_SBR_HEADER_DATA hHeaderData, HANDLE_SBR_PREV_FRAME_DATA hPrevFrameData, const int downsampleFac, const UINT flags, HANDLE_SBR_FRAME_DATA hFrameData); #endif fdk-aac-2.0.2/libSBRdec/src/sbr_ram.cpp000066400000000000000000000164131404225310700175150ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Memory layout This module declares all static and dynamic memory spaces */ #include "sbr_ram.h" /*! \name StaticSbrData Static memory areas, must not be overwritten in other sections of the decoder */ /* @{ */ /*! SBR Decoder main structure */ C_ALLOC_MEM(Ram_SbrDecoder, struct SBR_DECODER_INSTANCE, 1) /*! SBR Decoder element data
Dimension: (8) */ C_ALLOC_MEM2(Ram_SbrDecElement, SBR_DECODER_ELEMENT, 1, (8)) /*! SBR Decoder individual channel data
Dimension: (8) */ C_ALLOC_MEM2(Ram_SbrDecChannel, SBR_CHANNEL, 1, (8) + 1) /*! Static Data of PS */ C_ALLOC_MEM(Ram_ps_dec, struct PS_DEC, 1) /* @} */ /*! \name DynamicSbrData Dynamic memory areas, might be reused in other algorithm sections, e.g. the core decoder
Depending on the mode set by DONT_USE_CORE_WORKBUFFER, workbuffers are defined additionally to the CoreWorkbuffer.
The size of WorkBuffers is ((1024) / (32) * (4) / 2)*(64) = 2048.
WorkBuffer2 is a pointer to the CoreWorkBuffer wich is reused here in the SBR part. In case of DONT_USE_CORE_WORKBUFFER, the CoreWorkbuffer is not used and the according Workbuffer2 is defined locally in this file.
WorkBuffer1 is reused in the AAC core (-> aacdecoder.cpp, aac_ram.cpp)
Use of WorkBuffers: 

    -------------------------------------------------------------
    AAC core:

      CoreWorkbuffer: spectral coefficients
      WorkBuffer1:    CAacDecoderChannelInfo, CAacDecoderDynamicData

    -------------------------------------------------------------
    SBR part:
      ----------------------------------------------
      Low Power Mode (useLP=1 or LOW_POWER_SBR_ONLY), see assignLcTimeSlots()

        SLOT_BASED_PROTOTYPE_SYN_FILTER

        WorkBuffer1                                WorkBuffer2(=CoreWorkbuffer)
         ________________                           ________________
        | RealLeft       |                         | RealRight      |
        |________________|                         |________________|

      ----------------------------------------------
      High Quality Mode (!LOW_POWER_SBR_ONLY and useLP=0), see
  assignHqTimeSlots()

         SLOTBASED_PS

         WorkBuffer1                                WorkBuffer2(=CoreWorkbuffer)
         ________________                           ________________
        | Real/Imag      |  interleaved            | Real/Imag      |
  interleaved
        |________________|  first half actual ch   |________________|  second
  half actual ch

    -------------------------------------------------------------
*/ fdk-aac-2.0.2/libSBRdec/src/sbr_ram.h000066400000000000000000000167441404225310700171710ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Memory layout */ #ifndef SBR_RAM_H #define SBR_RAM_H #include "sbrdecoder.h" #include "env_extr.h" #include "sbr_dec.h" #define SBRDEC_MAX_CH_PER_ELEMENT (2) #define FRAME_OK (0) #define FRAME_ERROR (1) #define FRAME_ERROR_ALLSLOTS (2) typedef struct { SBR_CHANNEL *pSbrChannel[SBRDEC_MAX_CH_PER_ELEMENT]; TRANSPOSER_SETTINGS transposerSettings; /* Common transport settings for each individual channel of an element */ HANDLE_FDK_BITSTREAM hBs; MP4_ELEMENT_ID elementID; /* Element ID set during initialization. Can be used for concealment */ int nChannels; /* Number of elements output channels (=2 in case of PS) */ UCHAR frameErrorFlag[(1) + 1]; /* Frame error status (for every slot in the delay line). Will be copied into header at the very beginning of decodeElement() routine. */ UCHAR useFrameSlot; /* Index which defines which slot will be decoded/filled next (used with additional delay) */ UCHAR useHeaderSlot[(1) + 1]; /* Index array that provides the link between header and frame data (important when processing with additional delay). */ } SBR_DECODER_ELEMENT; struct SBR_DECODER_INSTANCE { SBR_DECODER_ELEMENT *pSbrElement[(8)]; SBR_HEADER_DATA sbrHeader[( 8)][(1) + 1]; /* Sbr header for each individual channel of an element */ HANDLE_FDK_QMF_DOMAIN pQmfDomain; HANDLE_PS_DEC hParametricStereoDec; /* Global parameters */ AUDIO_OBJECT_TYPE coreCodec; /* AOT of core codec */ int numSbrElements; int numSbrChannels; INT sampleRateIn; /* SBR decoder input sampling rate; might be different than the transposer input sampling rate. */ INT sampleRateOut; /* Sampling rate of the SBR decoder output audio samples. */ USHORT codecFrameSize; UCHAR synDownsampleFac; INT downscaleFactor; UCHAR numDelayFrames; /* The current number of additional delay frames used for processing. */ UCHAR harmonicSBR; UCHAR numFlushedFrames; /* The variable counts the number of frames which are flushed consecutively. */ UINT flags; INT sbrInDataHeadroom; /* Headroom of the SBR input time signal to prevent clipping */ }; H_ALLOC_MEM(Ram_SbrDecElement, SBR_DECODER_ELEMENT) H_ALLOC_MEM(Ram_SbrDecChannel, SBR_CHANNEL) H_ALLOC_MEM(Ram_SbrDecoder, struct SBR_DECODER_INSTANCE) H_ALLOC_MEM(Ram_sbr_QmfStatesSynthesis, FIXP_QSS) H_ALLOC_MEM(Ram_sbr_OverlapBuffer, FIXP_DBL) H_ALLOC_MEM(Ram_sbr_HBEOverlapBuffer, FIXP_DBL) H_ALLOC_MEM(Ram_ps_dec, PS_DEC) #endif /* SBR_RAM_H */ fdk-aac-2.0.2/libSBRdec/src/sbr_rom.cpp000066400000000000000000002155611404225310700175400ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Definition of constant tables This module contains most of the constant data that can be stored in ROM. */ #include "sbr_rom.h" /*! \name StartStopBands \brief Start and stop subbands of the highband. k_o = startMin + offset[bs_start_freq]; startMin = {3000,4000,5000} * (128/FS_sbr) / FS_sbr < 32Khz, 32Khz <= FS_sbr < 64KHz, 64KHz <= FS_sbr The stop subband can also be calculated to save memory by defining #CALC_STOP_BAND. */ //@{ /* tables were created with ../addon/octave/sbr_start_freq_table.m */ const UCHAR FDK_sbrDecoder_sbr_start_freq_16[][16] = { {16, 17, 18, 19, 20, 21, 22, 23, 24, 25, 26, 27, 28, 29, 30, 31}, {4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19}}; const UCHAR FDK_sbrDecoder_sbr_start_freq_22[][16] = { {12, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23, 24, 26, 28, 30}, {4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 18, 20, 22}}; const UCHAR FDK_sbrDecoder_sbr_start_freq_24[][16] = { {11, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23, 25, 27, 29, 32}, {3, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 17, 19, 21, 24}}; const UCHAR FDK_sbrDecoder_sbr_start_freq_32[][16] = { {10, 12, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23, 25, 27, 29, 32}, {2, 4, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 17, 19, 21, 24}}; const UCHAR FDK_sbrDecoder_sbr_start_freq_40[][16] = { {12, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 24, 26, 28, 30, 32}, {5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 17, 19, 21, 23, 25}}; const UCHAR FDK_sbrDecoder_sbr_start_freq_44[][16] = { {8, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 21, 23, 25, 28, 32}, {2, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 15, 17, 19, 22, 26}}; const UCHAR FDK_sbrDecoder_sbr_start_freq_48[][16] = { {7, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 20, 22, 24, 27, 31}, {1, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 14, 16, 18, 21, 25}}; const UCHAR FDK_sbrDecoder_sbr_start_freq_64[][16] = { {6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 19, 21, 23, 26, 30}, {1, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 14, 16, 18, 21, 25}}; const UCHAR FDK_sbrDecoder_sbr_start_freq_88[][16] = { {5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 16, 18, 20, 23, 27, 31}, {2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 13, 15, 17, 20, 24, 28}}; const UCHAR FDK_sbrDecoder_sbr_start_freq_192[16] = { 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 12, 14, 16, 19, 23, 27}; const UCHAR FDK_sbrDecoder_sbr_start_freq_176[16] = { 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 13, 15, 17, 20, 24, 28}; const UCHAR FDK_sbrDecoder_sbr_start_freq_128[16] = { 1, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 14, 16, 18, 21, 25}; //@} /*! \name Whitening \brief Coefficients for spectral whitening in the transposer */ //@{ /*! Assignment of whitening tuning depending on the crossover frequency */ const USHORT FDK_sbrDecoder_sbr_whFactorsIndex[NUM_WHFACTOR_TABLE_ENTRIES] = { 0, 5000, 6000, 6500, 7000, 7500, 8000, 9000, 10000}; /*! \brief Whithening levels tuning table With the current tuning, there are some redundant entries: \li NUM_WHFACTOR_TABLE_ENTRIES can be reduced by 3, \li the first coloumn can be eliminated. */ const FIXP_DBL FDK_sbrDecoder_sbr_whFactorsTable[NUM_WHFACTOR_TABLE_ENTRIES][6] = { /* OFF_LEVEL, TRANSITION_LEVEL, LOW_LEVEL, MID_LEVEL, HIGH_LEVEL */ {FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* < 5000 */ {FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* 5000 < 6000 */ {FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* 6000 < 6500 */ {FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* 6500 < 7000 */ {FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* 7000 < 7500 */ {FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* 7500 < 8000 */ {FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* 8000 < 9000 */ {FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* 9000 < 10000 */ {FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* > 10000 */ }; //@} /*! \name EnvAdj \brief Constants and tables used for envelope adjustment */ //@{ /*! Mantissas of gain limits */ const FIXP_SGL FDK_sbrDecoder_sbr_limGains_m[4] = { FL2FXCONST_SGL(0.5011932025f), /*!< -3 dB. Gain limit when limiterGains in frameData is 0 */ FL2FXCONST_SGL( 0.5f), /*!< 0 dB. Gain limit when limiterGains in frameData is 1 */ FL2FXCONST_SGL(0.9976346258f), /*!< +3 dB. Gain limit when limiterGains in frameData is 2 */ FL2FXCONST_SGL(0.6776263578f) /*!< Inf. Gain limit when limiterGains in frameData is 3 */ }; /*! Exponents of gain limits */ const UCHAR FDK_sbrDecoder_sbr_limGains_e[4] = {0, 1, 1, 67}; /*! Constants for calculating the number of limiter bands */ const FIXP_SGL FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4[4] = { FL2FXCONST_SGL(1.0f / 4.0f), FL2FXCONST_SGL(1.2f / 4.0f), FL2FXCONST_SGL(2.0f / 4.0f), FL2FXCONST_SGL(3.0f / 4.0f)}; /*! Constants for calculating the number of limiter bands */ const FIXP_DBL FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4_DBL[4] = { FL2FXCONST_DBL(1.0f / 4.0f), FL2FXCONST_DBL(1.2f / 4.0f), FL2FXCONST_DBL(2.0f / 4.0f), FL2FXCONST_DBL(3.0f / 4.0f)}; /*! Ratio of old gains and noise levels for the first 4 timeslots of an envelope */ const FIXP_SGL FDK_sbrDecoder_sbr_smoothFilter[4] = { FL2FXCONST_SGL(0.66666666666666f), FL2FXCONST_SGL(0.36516383427084f), FL2FXCONST_SGL(0.14699433520835f), FL2FXCONST_SGL(0.03183050093751f)}; /*! Real and imaginary part of random noise which will be modulated to the desired level. An accuracy of 13 bits is sufficient for these random numbers. */ const FIXP_SGL FDK_sbrDecoder_sbr_randomPhase[SBR_NF_NO_RANDOM_VAL][2] = { {FL2FXCONST_SGL(-0.99948153278296f), FL2FXCONST_SGL(-0.59483417516607f)}, {FL2FXCONST_SGL(0.97113454393991f), FL2FXCONST_SGL(-0.67528515225647f)}, {FL2FXCONST_SGL(0.14130051758487f), FL2FXCONST_SGL(-0.95090983575689f)}, {FL2FXCONST_SGL(-0.47005496701697f), FL2FXCONST_SGL(-0.37340549728647f)}, {FL2FXCONST_SGL(0.80705063769351f), FL2FXCONST_SGL(0.29653668284408f)}, {FL2FXCONST_SGL(-0.38981478896926f), FL2FXCONST_SGL(0.89572605717087f)}, {FL2FXCONST_SGL(-0.01053049862020f), FL2FXCONST_SGL(-0.66959058036166f)}, {FL2FXCONST_SGL(-0.91266367957293f), FL2FXCONST_SGL(-0.11522938140034f)}, {FL2FXCONST_SGL(0.54840422910309f), FL2FXCONST_SGL(0.75221367176302f)}, {FL2FXCONST_SGL(0.40009252867955f), FL2FXCONST_SGL(-0.98929400334421f)}, {FL2FXCONST_SGL(-0.99867974711855f), FL2FXCONST_SGL(-0.88147068645358f)}, {FL2FXCONST_SGL(-0.95531076805040f), FL2FXCONST_SGL(0.90908757154593f)}, {FL2FXCONST_SGL(-0.45725933317144f), FL2FXCONST_SGL(-0.56716323646760f)}, {FL2FXCONST_SGL(-0.72929675029275f), FL2FXCONST_SGL(-0.98008272727324f)}, {FL2FXCONST_SGL(0.75622801399036f), FL2FXCONST_SGL(0.20950329995549f)}, {FL2FXCONST_SGL(0.07069442601050f), FL2FXCONST_SGL(-0.78247898470706f)}, {FL2FXCONST_SGL(0.74496252926055f), FL2FXCONST_SGL(-0.91169004445807f)}, {FL2FXCONST_SGL(-0.96440182703856f), FL2FXCONST_SGL(-0.94739918296622f)}, {FL2FXCONST_SGL(0.30424629369539f), FL2FXCONST_SGL(-0.49438267012479f)}, {FL2FXCONST_SGL(0.66565033746925f), FL2FXCONST_SGL(0.64652935542491f)}, {FL2FXCONST_SGL(0.91697008020594f), FL2FXCONST_SGL(0.17514097332009f)}, {FL2FXCONST_SGL(-0.70774918760427f), FL2FXCONST_SGL(0.52548653416543f)}, {FL2FXCONST_SGL(-0.70051415345560f), FL2FXCONST_SGL(-0.45340028808763f)}, {FL2FXCONST_SGL(-0.99496513054797f), FL2FXCONST_SGL(-0.90071908066973f)}, {FL2FXCONST_SGL(0.98164490790123f), FL2FXCONST_SGL(-0.77463155528697f)}, {FL2FXCONST_SGL(-0.54671580548181f), FL2FXCONST_SGL(-0.02570928536004f)}, {FL2FXCONST_SGL(-0.01689629065389f), FL2FXCONST_SGL(0.00287506445732f)}, {FL2FXCONST_SGL(-0.86110349531986f), FL2FXCONST_SGL(0.42548583726477f)}, {FL2FXCONST_SGL(-0.98892980586032f), FL2FXCONST_SGL(-0.87881132267556f)}, {FL2FXCONST_SGL(0.51756627678691f), FL2FXCONST_SGL(0.66926784710139f)}, {FL2FXCONST_SGL(-0.99635026409640f), FL2FXCONST_SGL(-0.58107730574765f)}, {FL2FXCONST_SGL(-0.99969370862163f), FL2FXCONST_SGL(0.98369989360250f)}, {FL2FXCONST_SGL(0.55266258627194f), FL2FXCONST_SGL(0.59449057465591f)}, {FL2FXCONST_SGL(0.34581177741673f), FL2FXCONST_SGL(0.94879421061866f)}, {FL2FXCONST_SGL(0.62664209577999f), FL2FXCONST_SGL(-0.74402970906471f)}, {FL2FXCONST_SGL(-0.77149701404973f), FL2FXCONST_SGL(-0.33883658042801f)}, {FL2FXCONST_SGL(-0.91592244254432f), FL2FXCONST_SGL(0.03687901376713f)}, {FL2FXCONST_SGL(-0.76285492357887f), FL2FXCONST_SGL(-0.91371867919124f)}, {FL2FXCONST_SGL(0.79788337195331f), FL2FXCONST_SGL(-0.93180971199849f)}, {FL2FXCONST_SGL(0.54473080610200f), FL2FXCONST_SGL(-0.11919206037186f)}, {FL2FXCONST_SGL(-0.85639281671058f), FL2FXCONST_SGL(0.42429854760451f)}, {FL2FXCONST_SGL(-0.92882402971423f), FL2FXCONST_SGL(0.27871809078609f)}, {FL2FXCONST_SGL(-0.11708371046774f), FL2FXCONST_SGL(-0.99800843444966f)}, {FL2FXCONST_SGL(0.21356749817493f), FL2FXCONST_SGL(-0.90716295627033f)}, {FL2FXCONST_SGL(-0.76191692573909f), FL2FXCONST_SGL(0.99768118356265f)}, {FL2FXCONST_SGL(0.98111043100884f), FL2FXCONST_SGL(-0.95854459734407f)}, {FL2FXCONST_SGL(-0.85913269895572f), FL2FXCONST_SGL(0.95766566168880f)}, {FL2FXCONST_SGL(-0.93307242253692f), FL2FXCONST_SGL(0.49431757696466f)}, {FL2FXCONST_SGL(0.30485754879632f), FL2FXCONST_SGL(-0.70540034357529f)}, {FL2FXCONST_SGL(0.85289650925190f), FL2FXCONST_SGL(0.46766131791044f)}, {FL2FXCONST_SGL(0.91328082618125f), FL2FXCONST_SGL(-0.99839597361769f)}, {FL2FXCONST_SGL(-0.05890199924154f), FL2FXCONST_SGL(0.70741827819497f)}, {FL2FXCONST_SGL(0.28398686150148f), FL2FXCONST_SGL(0.34633555702188f)}, {FL2FXCONST_SGL(0.95258164539612f), FL2FXCONST_SGL(-0.54893416026939f)}, {FL2FXCONST_SGL(-0.78566324168507f), FL2FXCONST_SGL(-0.75568541079691f)}, {FL2FXCONST_SGL(-0.95789495447877f), FL2FXCONST_SGL(-0.20423194696966f)}, {FL2FXCONST_SGL(0.82411158711197f), FL2FXCONST_SGL(0.96654618432562f)}, {FL2FXCONST_SGL(-0.65185446735885f), FL2FXCONST_SGL(-0.88734990773289f)}, {FL2FXCONST_SGL(-0.93643603134666f), FL2FXCONST_SGL(0.99870790442385f)}, {FL2FXCONST_SGL(0.91427159529618f), FL2FXCONST_SGL(-0.98290505544444f)}, {FL2FXCONST_SGL(-0.70395684036886f), FL2FXCONST_SGL(0.58796798221039f)}, {FL2FXCONST_SGL(0.00563771969365f), FL2FXCONST_SGL(0.61768196727244f)}, {FL2FXCONST_SGL(0.89065051931895f), FL2FXCONST_SGL(0.52783352697585f)}, {FL2FXCONST_SGL(-0.68683707712762f), FL2FXCONST_SGL(0.80806944710339f)}, {FL2FXCONST_SGL(0.72165342518718f), FL2FXCONST_SGL(-0.69259857349564f)}, {FL2FXCONST_SGL(-0.62928247730667f), FL2FXCONST_SGL(0.13627037407335f)}, {FL2FXCONST_SGL(0.29938434065514f), FL2FXCONST_SGL(-0.46051329682246f)}, {FL2FXCONST_SGL(-0.91781958879280f), FL2FXCONST_SGL(-0.74012716684186f)}, {FL2FXCONST_SGL(0.99298717043688f), FL2FXCONST_SGL(0.40816610075661f)}, {FL2FXCONST_SGL(0.82368298622748f), FL2FXCONST_SGL(-0.74036047190173f)}, {FL2FXCONST_SGL(-0.98512833386833f), FL2FXCONST_SGL(-0.99972330709594f)}, {FL2FXCONST_SGL(-0.95915368242257f), FL2FXCONST_SGL(-0.99237800466040f)}, {FL2FXCONST_SGL(-0.21411126572790f), FL2FXCONST_SGL(-0.93424819052545f)}, {FL2FXCONST_SGL(-0.68821476106884f), FL2FXCONST_SGL(-0.26892306315457f)}, {FL2FXCONST_SGL(0.91851997982317f), FL2FXCONST_SGL(0.09358228901785f)}, {FL2FXCONST_SGL(-0.96062769559127f), FL2FXCONST_SGL(0.36099095133739f)}, {FL2FXCONST_SGL(0.51646184922287f), FL2FXCONST_SGL(-0.71373332873917f)}, {FL2FXCONST_SGL(0.61130721139669f), FL2FXCONST_SGL(0.46950141175917f)}, {FL2FXCONST_SGL(0.47336129371299f), FL2FXCONST_SGL(-0.27333178296162f)}, {FL2FXCONST_SGL(0.90998308703519f), FL2FXCONST_SGL(0.96715662938132f)}, {FL2FXCONST_SGL(0.44844799194357f), FL2FXCONST_SGL(0.99211574628306f)}, {FL2FXCONST_SGL(0.66614891079092f), FL2FXCONST_SGL(0.96590176169121f)}, {FL2FXCONST_SGL(0.74922239129237f), FL2FXCONST_SGL(-0.89879858826087f)}, {FL2FXCONST_SGL(-0.99571588506485f), FL2FXCONST_SGL(0.52785521494349f)}, {FL2FXCONST_SGL(0.97401082477563f), FL2FXCONST_SGL(-0.16855870075190f)}, {FL2FXCONST_SGL(0.72683747733879f), FL2FXCONST_SGL(-0.48060774432251f)}, {FL2FXCONST_SGL(0.95432193457128f), FL2FXCONST_SGL(0.68849603408441f)}, {FL2FXCONST_SGL(-0.72962208425191f), FL2FXCONST_SGL(-0.76608443420917f)}, {FL2FXCONST_SGL(-0.85359479233537f), FL2FXCONST_SGL(0.88738125901579f)}, {FL2FXCONST_SGL(-0.81412430338535f), FL2FXCONST_SGL(-0.97480768049637f)}, {FL2FXCONST_SGL(-0.87930772356786f), FL2FXCONST_SGL(0.74748307690436f)}, {FL2FXCONST_SGL(-0.71573331064977f), FL2FXCONST_SGL(-0.98570608178923f)}, {FL2FXCONST_SGL(0.83524300028228f), FL2FXCONST_SGL(0.83702537075163f)}, {FL2FXCONST_SGL(-0.48086065601423f), FL2FXCONST_SGL(-0.98848504923531f)}, {FL2FXCONST_SGL(0.97139128574778f), FL2FXCONST_SGL(0.80093621198236f)}, {FL2FXCONST_SGL(0.51992825347895f), FL2FXCONST_SGL(0.80247631400510f)}, {FL2FXCONST_SGL(-0.00848591195325f), FL2FXCONST_SGL(-0.76670128000486f)}, {FL2FXCONST_SGL(-0.70294374303036f), FL2FXCONST_SGL(0.55359910445577f)}, {FL2FXCONST_SGL(-0.95894428168140f), FL2FXCONST_SGL(-0.43265504344783f)}, {FL2FXCONST_SGL(0.97079252950321f), FL2FXCONST_SGL(0.09325857238682f)}, {FL2FXCONST_SGL(-0.92404293670797f), FL2FXCONST_SGL(0.85507704027855f)}, {FL2FXCONST_SGL(-0.69506469500450f), FL2FXCONST_SGL(0.98633412625459f)}, {FL2FXCONST_SGL(0.26559203620024f), FL2FXCONST_SGL(0.73314307966524f)}, {FL2FXCONST_SGL(0.28038443336943f), FL2FXCONST_SGL(0.14537913654427f)}, {FL2FXCONST_SGL(-0.74138124825523f), FL2FXCONST_SGL(0.99310339807762f)}, {FL2FXCONST_SGL(-0.01752795995444f), FL2FXCONST_SGL(-0.82616635284178f)}, {FL2FXCONST_SGL(-0.55126773094930f), FL2FXCONST_SGL(-0.98898543862153f)}, {FL2FXCONST_SGL(0.97960898850996f), FL2FXCONST_SGL(-0.94021446752851f)}, {FL2FXCONST_SGL(-0.99196309146936f), FL2FXCONST_SGL(0.67019017358456f)}, {FL2FXCONST_SGL(-0.67684928085260f), FL2FXCONST_SGL(0.12631491649378f)}, {FL2FXCONST_SGL(0.09140039465500f), FL2FXCONST_SGL(-0.20537731453108f)}, {FL2FXCONST_SGL(-0.71658965751996f), FL2FXCONST_SGL(-0.97788200391224f)}, {FL2FXCONST_SGL(0.81014640078925f), FL2FXCONST_SGL(0.53722648362443f)}, {FL2FXCONST_SGL(0.40616991671205f), FL2FXCONST_SGL(-0.26469008598449f)}, {FL2FXCONST_SGL(-0.67680188682972f), FL2FXCONST_SGL(0.94502052337695f)}, {FL2FXCONST_SGL(0.86849774348749f), FL2FXCONST_SGL(-0.18333598647899f)}, {FL2FXCONST_SGL(-0.99500381284851f), FL2FXCONST_SGL(-0.02634122068550f)}, {FL2FXCONST_SGL(0.84329189340667f), FL2FXCONST_SGL(0.10406957462213f)}, {FL2FXCONST_SGL(-0.09215968531446f), FL2FXCONST_SGL(0.69540012101253f)}, {FL2FXCONST_SGL(0.99956173327206f), FL2FXCONST_SGL(-0.12358542001404f)}, {FL2FXCONST_SGL(-0.79732779473535f), FL2FXCONST_SGL(-0.91582524736159f)}, {FL2FXCONST_SGL(0.96349973642406f), FL2FXCONST_SGL(0.96640458041000f)}, {FL2FXCONST_SGL(-0.79942778496547f), FL2FXCONST_SGL(0.64323902822857f)}, {FL2FXCONST_SGL(-0.11566039853896f), FL2FXCONST_SGL(0.28587846253726f)}, {FL2FXCONST_SGL(-0.39922954514662f), FL2FXCONST_SGL(0.94129601616966f)}, {FL2FXCONST_SGL(0.99089197565987f), FL2FXCONST_SGL(-0.92062625581587f)}, {FL2FXCONST_SGL(0.28631285179909f), FL2FXCONST_SGL(-0.91035047143603f)}, {FL2FXCONST_SGL(-0.83302725605608f), FL2FXCONST_SGL(-0.67330410892084f)}, {FL2FXCONST_SGL(0.95404443402072f), FL2FXCONST_SGL(0.49162765398743f)}, {FL2FXCONST_SGL(-0.06449863579434f), FL2FXCONST_SGL(0.03250560813135f)}, {FL2FXCONST_SGL(-0.99575054486311f), FL2FXCONST_SGL(0.42389784469507f)}, {FL2FXCONST_SGL(-0.65501142790847f), FL2FXCONST_SGL(0.82546114655624f)}, {FL2FXCONST_SGL(-0.81254441908887f), FL2FXCONST_SGL(-0.51627234660629f)}, {FL2FXCONST_SGL(-0.99646369485481f), FL2FXCONST_SGL(0.84490533520752f)}, {FL2FXCONST_SGL(0.00287840603348f), FL2FXCONST_SGL(0.64768261158166f)}, {FL2FXCONST_SGL(0.70176989408455f), FL2FXCONST_SGL(-0.20453028573322f)}, {FL2FXCONST_SGL(0.96361882270190f), FL2FXCONST_SGL(0.40706967140989f)}, {FL2FXCONST_SGL(-0.68883758192426f), FL2FXCONST_SGL(0.91338958840772f)}, {FL2FXCONST_SGL(-0.34875585502238f), FL2FXCONST_SGL(0.71472290693300f)}, {FL2FXCONST_SGL(0.91980081243087f), FL2FXCONST_SGL(0.66507455644919f)}, {FL2FXCONST_SGL(-0.99009048343881f), FL2FXCONST_SGL(0.85868021604848f)}, {FL2FXCONST_SGL(0.68865791458395f), FL2FXCONST_SGL(0.55660316809678f)}, {FL2FXCONST_SGL(-0.99484402129368f), FL2FXCONST_SGL(-0.20052559254934f)}, {FL2FXCONST_SGL(0.94214511408023f), FL2FXCONST_SGL(-0.99696425367461f)}, {FL2FXCONST_SGL(-0.67414626793544f), FL2FXCONST_SGL(0.49548221180078f)}, {FL2FXCONST_SGL(-0.47339353684664f), FL2FXCONST_SGL(-0.85904328834047f)}, {FL2FXCONST_SGL(0.14323651387360f), FL2FXCONST_SGL(-0.94145598222488f)}, {FL2FXCONST_SGL(-0.29268293575672f), FL2FXCONST_SGL(0.05759224927952f)}, {FL2FXCONST_SGL(0.43793861458754f), FL2FXCONST_SGL(-0.78904969892724f)}, {FL2FXCONST_SGL(-0.36345126374441f), FL2FXCONST_SGL(0.64874435357162f)}, {FL2FXCONST_SGL(-0.08750604656825f), FL2FXCONST_SGL(0.97686944362527f)}, {FL2FXCONST_SGL(-0.96495267812511f), FL2FXCONST_SGL(-0.53960305946511f)}, {FL2FXCONST_SGL(0.55526940659947f), FL2FXCONST_SGL(0.78891523734774f)}, {FL2FXCONST_SGL(0.73538215752630f), FL2FXCONST_SGL(0.96452072373404f)}, {FL2FXCONST_SGL(-0.30889773919437f), FL2FXCONST_SGL(-0.80664389776860f)}, {FL2FXCONST_SGL(0.03574995626194f), FL2FXCONST_SGL(-0.97325616900959f)}, {FL2FXCONST_SGL(0.98720684660488f), FL2FXCONST_SGL(0.48409133691962f)}, {FL2FXCONST_SGL(-0.81689296271203f), FL2FXCONST_SGL(-0.90827703628298f)}, {FL2FXCONST_SGL(0.67866860118215f), FL2FXCONST_SGL(0.81284503870856f)}, {FL2FXCONST_SGL(-0.15808569732583f), FL2FXCONST_SGL(0.85279555024382f)}, {FL2FXCONST_SGL(0.80723395114371f), FL2FXCONST_SGL(-0.24717418514605f)}, {FL2FXCONST_SGL(0.47788757329038f), FL2FXCONST_SGL(-0.46333147839295f)}, {FL2FXCONST_SGL(0.96367554763201f), FL2FXCONST_SGL(0.38486749303242f)}, {FL2FXCONST_SGL(-0.99143875716818f), FL2FXCONST_SGL(-0.24945277239809f)}, {FL2FXCONST_SGL(0.83081876925833f), FL2FXCONST_SGL(-0.94780851414763f)}, {FL2FXCONST_SGL(-0.58753191905341f), FL2FXCONST_SGL(0.01290772389163f)}, {FL2FXCONST_SGL(0.95538108220960f), FL2FXCONST_SGL(-0.85557052096538f)}, {FL2FXCONST_SGL(-0.96490920476211f), FL2FXCONST_SGL(-0.64020970923102f)}, {FL2FXCONST_SGL(-0.97327101028521f), FL2FXCONST_SGL(0.12378128133110f)}, {FL2FXCONST_SGL(0.91400366022124f), FL2FXCONST_SGL(0.57972471346930f)}, {FL2FXCONST_SGL(-0.99925837363824f), FL2FXCONST_SGL(0.71084847864067f)}, {FL2FXCONST_SGL(-0.86875903507313f), FL2FXCONST_SGL(-0.20291699203564f)}, {FL2FXCONST_SGL(-0.26240034795124f), FL2FXCONST_SGL(-0.68264554369108f)}, {FL2FXCONST_SGL(-0.24664412953388f), FL2FXCONST_SGL(-0.87642273115183f)}, {FL2FXCONST_SGL(0.02416275806869f), FL2FXCONST_SGL(0.27192914288905f)}, {FL2FXCONST_SGL(0.82068619590515f), FL2FXCONST_SGL(-0.85087787994476f)}, {FL2FXCONST_SGL(0.88547373760759f), FL2FXCONST_SGL(-0.89636802901469f)}, {FL2FXCONST_SGL(-0.18173078152226f), FL2FXCONST_SGL(-0.26152145156800f)}, {FL2FXCONST_SGL(0.09355476558534f), FL2FXCONST_SGL(0.54845123045604f)}, {FL2FXCONST_SGL(-0.54668414224090f), FL2FXCONST_SGL(0.95980774020221f)}, {FL2FXCONST_SGL(0.37050990604091f), FL2FXCONST_SGL(-0.59910140383171f)}, {FL2FXCONST_SGL(-0.70373594262891f), FL2FXCONST_SGL(0.91227665827081f)}, {FL2FXCONST_SGL(-0.34600785879594f), FL2FXCONST_SGL(-0.99441426144200f)}, {FL2FXCONST_SGL(-0.68774481731008f), FL2FXCONST_SGL(-0.30238837956299f)}, {FL2FXCONST_SGL(-0.26843291251234f), FL2FXCONST_SGL(0.83115668004362f)}, {FL2FXCONST_SGL(0.49072334613242f), FL2FXCONST_SGL(-0.45359708737775f)}, {FL2FXCONST_SGL(0.38975993093975f), FL2FXCONST_SGL(0.95515358099121f)}, {FL2FXCONST_SGL(-0.97757125224150f), FL2FXCONST_SGL(0.05305894580606f)}, {FL2FXCONST_SGL(-0.17325552859616f), FL2FXCONST_SGL(-0.92770672250494f)}, {FL2FXCONST_SGL(0.99948035025744f), FL2FXCONST_SGL(0.58285545563426f)}, {FL2FXCONST_SGL(-0.64946246527458f), FL2FXCONST_SGL(0.68645507104960f)}, {FL2FXCONST_SGL(-0.12016920576437f), FL2FXCONST_SGL(-0.57147322153312f)}, {FL2FXCONST_SGL(-0.58947456517751f), FL2FXCONST_SGL(-0.34847132454388f)}, {FL2FXCONST_SGL(-0.41815140454465f), FL2FXCONST_SGL(0.16276422358861f)}, {FL2FXCONST_SGL(0.99885650204884f), FL2FXCONST_SGL(0.11136095490444f)}, {FL2FXCONST_SGL(-0.56649614128386f), FL2FXCONST_SGL(-0.90494866361587f)}, {FL2FXCONST_SGL(0.94138021032330f), FL2FXCONST_SGL(0.35281916733018f)}, {FL2FXCONST_SGL(-0.75725076534641f), FL2FXCONST_SGL(0.53650549640587f)}, {FL2FXCONST_SGL(0.20541973692630f), FL2FXCONST_SGL(-0.94435144369918f)}, {FL2FXCONST_SGL(0.99980371023351f), FL2FXCONST_SGL(0.79835913565599f)}, {FL2FXCONST_SGL(0.29078277605775f), FL2FXCONST_SGL(0.35393777921520f)}, {FL2FXCONST_SGL(-0.62858772103030f), FL2FXCONST_SGL(0.38765693387102f)}, {FL2FXCONST_SGL(0.43440904467688f), FL2FXCONST_SGL(-0.98546330463232f)}, {FL2FXCONST_SGL(-0.98298583762390f), FL2FXCONST_SGL(0.21021524625209f)}, {FL2FXCONST_SGL(0.19513029146934f), FL2FXCONST_SGL(-0.94239832251867f)}, {FL2FXCONST_SGL(-0.95476662400101f), FL2FXCONST_SGL(0.98364554179143f)}, {FL2FXCONST_SGL(0.93379635304810f), FL2FXCONST_SGL(-0.70881994583682f)}, {FL2FXCONST_SGL(-0.85235410573336f), FL2FXCONST_SGL(-0.08342347966410f)}, {FL2FXCONST_SGL(-0.86425093011245f), FL2FXCONST_SGL(-0.45795025029466f)}, {FL2FXCONST_SGL(0.38879779059045f), FL2FXCONST_SGL(0.97274429344593f)}, {FL2FXCONST_SGL(0.92045124735495f), FL2FXCONST_SGL(-0.62433652524220f)}, {FL2FXCONST_SGL(0.89162532251878f), FL2FXCONST_SGL(0.54950955570563f)}, {FL2FXCONST_SGL(-0.36834336949252f), FL2FXCONST_SGL(0.96458298020975f)}, {FL2FXCONST_SGL(0.93891760988045f), FL2FXCONST_SGL(-0.89968353740388f)}, {FL2FXCONST_SGL(0.99267657565094f), FL2FXCONST_SGL(-0.03757034316958f)}, {FL2FXCONST_SGL(-0.94063471614176f), FL2FXCONST_SGL(0.41332338538963f)}, {FL2FXCONST_SGL(0.99740224117019f), FL2FXCONST_SGL(-0.16830494996370f)}, {FL2FXCONST_SGL(-0.35899413170555f), FL2FXCONST_SGL(-0.46633226649613f)}, {FL2FXCONST_SGL(0.05237237274947f), FL2FXCONST_SGL(-0.25640361602661f)}, {FL2FXCONST_SGL(0.36703583957424f), FL2FXCONST_SGL(-0.38653265641875f)}, {FL2FXCONST_SGL(0.91653180367913f), FL2FXCONST_SGL(-0.30587628726597f)}, {FL2FXCONST_SGL(0.69000803499316f), FL2FXCONST_SGL(0.90952171386132f)}, {FL2FXCONST_SGL(-0.38658751133527f), FL2FXCONST_SGL(0.99501571208985f)}, {FL2FXCONST_SGL(-0.29250814029851f), FL2FXCONST_SGL(0.37444994344615f)}, {FL2FXCONST_SGL(-0.60182204677608f), FL2FXCONST_SGL(0.86779651036123f)}, {FL2FXCONST_SGL(-0.97418588163217f), FL2FXCONST_SGL(0.96468523666475f)}, {FL2FXCONST_SGL(0.88461574003963f), FL2FXCONST_SGL(0.57508405276414f)}, {FL2FXCONST_SGL(0.05198933055162f), FL2FXCONST_SGL(0.21269661669964f)}, {FL2FXCONST_SGL(-0.53499621979720f), FL2FXCONST_SGL(0.97241553731237f)}, {FL2FXCONST_SGL(-0.49429560226497f), FL2FXCONST_SGL(0.98183865291903f)}, {FL2FXCONST_SGL(-0.98935142339139f), FL2FXCONST_SGL(-0.40249159006933f)}, {FL2FXCONST_SGL(-0.98081380091130f), FL2FXCONST_SGL(-0.72856895534041f)}, {FL2FXCONST_SGL(-0.27338148835532f), FL2FXCONST_SGL(0.99950922447209f)}, {FL2FXCONST_SGL(0.06310802338302f), FL2FXCONST_SGL(-0.54539587529618f)}, {FL2FXCONST_SGL(-0.20461677199539f), FL2FXCONST_SGL(-0.14209977628489f)}, {FL2FXCONST_SGL(0.66223843141647f), FL2FXCONST_SGL(0.72528579940326f)}, {FL2FXCONST_SGL(-0.84764345483665f), FL2FXCONST_SGL(0.02372316801261f)}, {FL2FXCONST_SGL(-0.89039863483811f), FL2FXCONST_SGL(0.88866581484602f)}, {FL2FXCONST_SGL(0.95903308477986f), FL2FXCONST_SGL(0.76744927173873f)}, {FL2FXCONST_SGL(0.73504123909879f), FL2FXCONST_SGL(-0.03747203173192f)}, {FL2FXCONST_SGL(-0.31744434966056f), FL2FXCONST_SGL(-0.36834111883652f)}, {FL2FXCONST_SGL(-0.34110827591623f), FL2FXCONST_SGL(0.40211222807691f)}, {FL2FXCONST_SGL(0.47803883714199f), FL2FXCONST_SGL(-0.39423219786288f)}, {FL2FXCONST_SGL(0.98299195879514f), FL2FXCONST_SGL(0.01989791390047f)}, {FL2FXCONST_SGL(-0.30963073129751f), FL2FXCONST_SGL(-0.18076720599336f)}, {FL2FXCONST_SGL(0.99992588229018f), FL2FXCONST_SGL(-0.26281872094289f)}, {FL2FXCONST_SGL(-0.93149731080767f), FL2FXCONST_SGL(-0.98313162570490f)}, {FL2FXCONST_SGL(0.99923472302773f), FL2FXCONST_SGL(-0.80142993767554f)}, {FL2FXCONST_SGL(-0.26024169633417f), FL2FXCONST_SGL(-0.75999759855752f)}, {FL2FXCONST_SGL(-0.35712514743563f), FL2FXCONST_SGL(0.19298963768574f)}, {FL2FXCONST_SGL(-0.99899084509530f), FL2FXCONST_SGL(0.74645156992493f)}, {FL2FXCONST_SGL(0.86557171579452f), FL2FXCONST_SGL(0.55593866696299f)}, {FL2FXCONST_SGL(0.33408042438752f), FL2FXCONST_SGL(0.86185953874709f)}, {FL2FXCONST_SGL(0.99010736374716f), FL2FXCONST_SGL(0.04602397576623f)}, {FL2FXCONST_SGL(-0.66694269691195f), FL2FXCONST_SGL(-0.91643611810148f)}, {FL2FXCONST_SGL(0.64016792079480f), FL2FXCONST_SGL(0.15649530836856f)}, {FL2FXCONST_SGL(0.99570534804836f), FL2FXCONST_SGL(0.45844586038111f)}, {FL2FXCONST_SGL(-0.63431466947340f), FL2FXCONST_SGL(0.21079116459234f)}, {FL2FXCONST_SGL(-0.07706847005931f), FL2FXCONST_SGL(-0.89581437101329f)}, {FL2FXCONST_SGL(0.98590090577724f), FL2FXCONST_SGL(0.88241721133981f)}, {FL2FXCONST_SGL(0.80099335254678f), FL2FXCONST_SGL(-0.36851896710853f)}, {FL2FXCONST_SGL(0.78368131392666f), FL2FXCONST_SGL(0.45506999802597f)}, {FL2FXCONST_SGL(0.08707806671691f), FL2FXCONST_SGL(0.80938994918745f)}, {FL2FXCONST_SGL(-0.86811883080712f), FL2FXCONST_SGL(0.39347308654705f)}, {FL2FXCONST_SGL(-0.39466529740375f), FL2FXCONST_SGL(-0.66809432114456f)}, {FL2FXCONST_SGL(0.97875325649683f), FL2FXCONST_SGL(-0.72467840967746f)}, {FL2FXCONST_SGL(-0.95038560288864f), FL2FXCONST_SGL(0.89563219587625f)}, {FL2FXCONST_SGL(0.17005239424212f), FL2FXCONST_SGL(0.54683053962658f)}, {FL2FXCONST_SGL(-0.76910792026848f), FL2FXCONST_SGL(-0.96226617549298f)}, {FL2FXCONST_SGL(0.99743281016846f), FL2FXCONST_SGL(0.42697157037567f)}, {FL2FXCONST_SGL(0.95437383549973f), FL2FXCONST_SGL(0.97002324109952f)}, {FL2FXCONST_SGL(0.99578905365569f), FL2FXCONST_SGL(-0.54106826257356f)}, {FL2FXCONST_SGL(0.28058259829990f), FL2FXCONST_SGL(-0.85361420634036f)}, {FL2FXCONST_SGL(0.85256524470573f), FL2FXCONST_SGL(-0.64567607735589f)}, {FL2FXCONST_SGL(-0.50608540105128f), FL2FXCONST_SGL(-0.65846015480300f)}, {FL2FXCONST_SGL(-0.97210735183243f), FL2FXCONST_SGL(-0.23095213067791f)}, {FL2FXCONST_SGL(0.95424048234441f), FL2FXCONST_SGL(-0.99240147091219f)}, {FL2FXCONST_SGL(-0.96926570524023f), FL2FXCONST_SGL(0.73775654896574f)}, {FL2FXCONST_SGL(0.30872163214726f), FL2FXCONST_SGL(0.41514960556126f)}, {FL2FXCONST_SGL(-0.24523839572639f), FL2FXCONST_SGL(0.63206633394807f)}, {FL2FXCONST_SGL(-0.33813265086024f), FL2FXCONST_SGL(-0.38661779441897f)}, {FL2FXCONST_SGL(-0.05826828420146f), FL2FXCONST_SGL(-0.06940774188029f)}, {FL2FXCONST_SGL(-0.22898461455054f), FL2FXCONST_SGL(0.97054853316316f)}, {FL2FXCONST_SGL(-0.18509915019881f), FL2FXCONST_SGL(0.47565762892084f)}, {FL2FXCONST_SGL(-0.10488238045009f), FL2FXCONST_SGL(-0.87769947402394f)}, {FL2FXCONST_SGL(-0.71886586182037f), FL2FXCONST_SGL(0.78030982480538f)}, {FL2FXCONST_SGL(0.99793873738654f), FL2FXCONST_SGL(0.90041310491497f)}, {FL2FXCONST_SGL(0.57563307626120f), FL2FXCONST_SGL(-0.91034337352097f)}, {FL2FXCONST_SGL(0.28909646383717f), FL2FXCONST_SGL(0.96307783970534f)}, {FL2FXCONST_SGL(0.42188998312520f), FL2FXCONST_SGL(0.48148651230437f)}, {FL2FXCONST_SGL(0.93335049681047f), FL2FXCONST_SGL(-0.43537023883588f)}, {FL2FXCONST_SGL(-0.97087374418267f), FL2FXCONST_SGL(0.86636445711364f)}, {FL2FXCONST_SGL(0.36722871286923f), FL2FXCONST_SGL(0.65291654172961f)}, {FL2FXCONST_SGL(-0.81093025665696f), FL2FXCONST_SGL(0.08778370229363f)}, {FL2FXCONST_SGL(-0.26240603062237f), FL2FXCONST_SGL(-0.92774095379098f)}, {FL2FXCONST_SGL(0.83996497984604f), FL2FXCONST_SGL(0.55839849139647f)}, {FL2FXCONST_SGL(-0.99909615720225f), FL2FXCONST_SGL(-0.96024605713970f)}, {FL2FXCONST_SGL(0.74649464155061f), FL2FXCONST_SGL(0.12144893606462f)}, {FL2FXCONST_SGL(-0.74774595569805f), FL2FXCONST_SGL(-0.26898062008959f)}, {FL2FXCONST_SGL(0.95781667469567f), FL2FXCONST_SGL(-0.79047927052628f)}, {FL2FXCONST_SGL(0.95472308713099f), FL2FXCONST_SGL(-0.08588776019550f)}, {FL2FXCONST_SGL(0.48708332746299f), MAXVAL_SGL}, {FL2FXCONST_SGL(0.46332038247497f), FL2FXCONST_SGL(0.10964126185063f)}, {FL2FXCONST_SGL(-0.76497004940162f), FL2FXCONST_SGL(0.89210929242238f)}, {FL2FXCONST_SGL(0.57397389364339f), FL2FXCONST_SGL(0.35289703373760f)}, {FL2FXCONST_SGL(0.75374316974495f), FL2FXCONST_SGL(0.96705214651335f)}, {FL2FXCONST_SGL(-0.59174397685714f), FL2FXCONST_SGL(-0.89405370422752f)}, {FL2FXCONST_SGL(0.75087906691890f), FL2FXCONST_SGL(-0.29612672982396f)}, {FL2FXCONST_SGL(-0.98607857336230f), FL2FXCONST_SGL(0.25034911730023f)}, {FL2FXCONST_SGL(-0.40761056640505f), FL2FXCONST_SGL(-0.90045573444695f)}, {FL2FXCONST_SGL(0.66929266740477f), FL2FXCONST_SGL(0.98629493401748f)}, {FL2FXCONST_SGL(-0.97463695257310f), FL2FXCONST_SGL(-0.00190223301301f)}, {FL2FXCONST_SGL(0.90145509409859f), FL2FXCONST_SGL(0.99781390365446f)}, {FL2FXCONST_SGL(-0.87259289048043f), FL2FXCONST_SGL(0.99233587353666f)}, {FL2FXCONST_SGL(-0.91529461447692f), FL2FXCONST_SGL(-0.15698707534206f)}, {FL2FXCONST_SGL(-0.03305738840705f), FL2FXCONST_SGL(-0.37205262859764f)}, {FL2FXCONST_SGL(0.07223051368337f), FL2FXCONST_SGL(-0.88805001733626f)}, {FL2FXCONST_SGL(0.99498012188353f), FL2FXCONST_SGL(0.97094358113387f)}, {FL2FXCONST_SGL(-0.74904939500519f), FL2FXCONST_SGL(0.99985483641521f)}, {FL2FXCONST_SGL(0.04585228574211f), FL2FXCONST_SGL(0.99812337444082f)}, {FL2FXCONST_SGL(-0.89054954257993f), FL2FXCONST_SGL(-0.31791913188064f)}, {FL2FXCONST_SGL(-0.83782144651251f), FL2FXCONST_SGL(0.97637632547466f)}, {FL2FXCONST_SGL(0.33454804933804f), FL2FXCONST_SGL(-0.86231516800408f)}, {FL2FXCONST_SGL(-0.99707579362824f), FL2FXCONST_SGL(0.93237990079441f)}, {FL2FXCONST_SGL(-0.22827527843994f), FL2FXCONST_SGL(0.18874759397997f)}, {FL2FXCONST_SGL(0.67248046289143f), FL2FXCONST_SGL(-0.03646211390569f)}, {FL2FXCONST_SGL(-0.05146538187944f), FL2FXCONST_SGL(-0.92599700120679f)}, {FL2FXCONST_SGL(0.99947295749905f), FL2FXCONST_SGL(0.93625229707912f)}, {FL2FXCONST_SGL(0.66951124390363f), FL2FXCONST_SGL(0.98905825623893f)}, {FL2FXCONST_SGL(-0.99602956559179f), FL2FXCONST_SGL(-0.44654715757688f)}, {FL2FXCONST_SGL(0.82104905483590f), FL2FXCONST_SGL(0.99540741724928f)}, {FL2FXCONST_SGL(0.99186510988782f), FL2FXCONST_SGL(0.72023001312947f)}, {FL2FXCONST_SGL(-0.65284592392918f), FL2FXCONST_SGL(0.52186723253637f)}, {FL2FXCONST_SGL(0.93885443798188f), FL2FXCONST_SGL(-0.74895312615259f)}, {FL2FXCONST_SGL(0.96735248738388f), FL2FXCONST_SGL(0.90891816978629f)}, {FL2FXCONST_SGL(-0.22225968841114f), FL2FXCONST_SGL(0.57124029781228f)}, {FL2FXCONST_SGL(-0.44132783753414f), FL2FXCONST_SGL(-0.92688840659280f)}, {FL2FXCONST_SGL(-0.85694974219574f), FL2FXCONST_SGL(0.88844532719844f)}, {FL2FXCONST_SGL(0.91783042091762f), FL2FXCONST_SGL(-0.46356892383970f)}, {FL2FXCONST_SGL(0.72556974415690f), FL2FXCONST_SGL(-0.99899555770747f)}, {FL2FXCONST_SGL(-0.99711581834508f), FL2FXCONST_SGL(0.58211560180426f)}, {FL2FXCONST_SGL(0.77638976371966f), FL2FXCONST_SGL(0.94321834873819f)}, {FL2FXCONST_SGL(0.07717324253925f), FL2FXCONST_SGL(0.58638399856595f)}, {FL2FXCONST_SGL(-0.56049829194163f), FL2FXCONST_SGL(0.82522301569036f)}, {FL2FXCONST_SGL(0.98398893639988f), FL2FXCONST_SGL(0.39467440420569f)}, {FL2FXCONST_SGL(0.47546946844938f), FL2FXCONST_SGL(0.68613044836811f)}, {FL2FXCONST_SGL(0.65675089314631f), FL2FXCONST_SGL(0.18331637134880f)}, {FL2FXCONST_SGL(0.03273375457980f), FL2FXCONST_SGL(-0.74933109564108f)}, {FL2FXCONST_SGL(-0.38684144784738f), FL2FXCONST_SGL(0.51337349030406f)}, {FL2FXCONST_SGL(-0.97346267944545f), FL2FXCONST_SGL(-0.96549364384098f)}, {FL2FXCONST_SGL(-0.53282156061942f), FL2FXCONST_SGL(-0.91423265091354f)}, {FL2FXCONST_SGL(0.99817310731176f), FL2FXCONST_SGL(0.61133572482148f)}, {FL2FXCONST_SGL(-0.50254500772635f), FL2FXCONST_SGL(-0.88829338134294f)}, {FL2FXCONST_SGL(0.01995873238855f), FL2FXCONST_SGL(0.85223515096765f)}, {FL2FXCONST_SGL(0.99930381973804f), FL2FXCONST_SGL(0.94578896296649f)}, {FL2FXCONST_SGL(0.82907767600783f), FL2FXCONST_SGL(-0.06323442598128f)}, {FL2FXCONST_SGL(-0.58660709669728f), FL2FXCONST_SGL(0.96840773806582f)}, {FL2FXCONST_SGL(-0.17573736667267f), FL2FXCONST_SGL(-0.48166920859485f)}, {FL2FXCONST_SGL(0.83434292401346f), FL2FXCONST_SGL(-0.13023450646997f)}, {FL2FXCONST_SGL(0.05946491307025f), FL2FXCONST_SGL(0.20511047074866f)}, {FL2FXCONST_SGL(0.81505484574602f), FL2FXCONST_SGL(-0.94685947861369f)}, {FL2FXCONST_SGL(-0.44976380954860f), FL2FXCONST_SGL(0.40894572671545f)}, {FL2FXCONST_SGL(-0.89746474625671f), FL2FXCONST_SGL(0.99846578838537f)}, {FL2FXCONST_SGL(0.39677256130792f), FL2FXCONST_SGL(-0.74854668609359f)}, {FL2FXCONST_SGL(-0.07588948563079f), FL2FXCONST_SGL(0.74096214084170f)}, {FL2FXCONST_SGL(0.76343198951445f), FL2FXCONST_SGL(0.41746629422634f)}, {FL2FXCONST_SGL(-0.74490104699626f), FL2FXCONST_SGL(0.94725911744610f)}, {FL2FXCONST_SGL(0.64880119792759f), FL2FXCONST_SGL(0.41336660830571f)}, {FL2FXCONST_SGL(0.62319537462542f), FL2FXCONST_SGL(-0.93098313552599f)}, {FL2FXCONST_SGL(0.42215817594807f), FL2FXCONST_SGL(-0.07712787385208f)}, {FL2FXCONST_SGL(0.02704554141885f), FL2FXCONST_SGL(-0.05417518053666f)}, {FL2FXCONST_SGL(0.80001773566818f), FL2FXCONST_SGL(0.91542195141039f)}, {FL2FXCONST_SGL(-0.79351832348816f), FL2FXCONST_SGL(-0.36208897989136f)}, {FL2FXCONST_SGL(0.63872359151636f), FL2FXCONST_SGL(0.08128252493444f)}, {FL2FXCONST_SGL(0.52890520960295f), FL2FXCONST_SGL(0.60048872455592f)}, {FL2FXCONST_SGL(0.74238552914587f), FL2FXCONST_SGL(0.04491915291044f)}, {FL2FXCONST_SGL(0.99096131449250f), FL2FXCONST_SGL(-0.19451182854402f)}, {FL2FXCONST_SGL(-0.80412329643109f), FL2FXCONST_SGL(-0.88513818199457f)}, {FL2FXCONST_SGL(-0.64612616129736f), FL2FXCONST_SGL(0.72198674804544f)}, {FL2FXCONST_SGL(0.11657770663191f), FL2FXCONST_SGL(-0.83662833815041f)}, {FL2FXCONST_SGL(-0.95053182488101f), FL2FXCONST_SGL(-0.96939905138082f)}, {FL2FXCONST_SGL(-0.62228872928622f), FL2FXCONST_SGL(0.82767262846661f)}, {FL2FXCONST_SGL(0.03004475787316f), FL2FXCONST_SGL(-0.99738896333384f)}, {FL2FXCONST_SGL(-0.97987214341034f), FL2FXCONST_SGL(0.36526129686425f)}, {FL2FXCONST_SGL(-0.99986980746200f), FL2FXCONST_SGL(-0.36021610299715f)}, {FL2FXCONST_SGL(0.89110648599879f), FL2FXCONST_SGL(-0.97894250343044f)}, {FL2FXCONST_SGL(0.10407960510582f), FL2FXCONST_SGL(0.77357793811619f)}, {FL2FXCONST_SGL(0.95964737821728f), FL2FXCONST_SGL(-0.35435818285502f)}, {FL2FXCONST_SGL(0.50843233159162f), FL2FXCONST_SGL(0.96107691266205f)}, {FL2FXCONST_SGL(0.17006334670615f), FL2FXCONST_SGL(-0.76854025314829f)}, {FL2FXCONST_SGL(0.25872675063360f), FL2FXCONST_SGL(0.99893303933816f)}, {FL2FXCONST_SGL(-0.01115998681937f), FL2FXCONST_SGL(0.98496019742444f)}, {FL2FXCONST_SGL(-0.79598702973261f), FL2FXCONST_SGL(0.97138411318894f)}, {FL2FXCONST_SGL(-0.99264708948101f), FL2FXCONST_SGL(-0.99542822402536f)}, {FL2FXCONST_SGL(-0.99829663752818f), FL2FXCONST_SGL(0.01877138824311f)}, {FL2FXCONST_SGL(-0.70801016548184f), FL2FXCONST_SGL(0.33680685948117f)}, {FL2FXCONST_SGL(-0.70467057786826f), FL2FXCONST_SGL(0.93272777501857f)}, {FL2FXCONST_SGL(0.99846021905254f), FL2FXCONST_SGL(-0.98725746254433f)}, {FL2FXCONST_SGL(-0.63364968534650f), FL2FXCONST_SGL(-0.16473594423746f)}, {FL2FXCONST_SGL(-0.16258217500792f), FL2FXCONST_SGL(-0.95939125400802f)}, {FL2FXCONST_SGL(-0.43645594360633f), FL2FXCONST_SGL(-0.94805030113284f)}, {FL2FXCONST_SGL(-0.99848471702976f), FL2FXCONST_SGL(0.96245166923809f)}, {FL2FXCONST_SGL(-0.16796458968998f), FL2FXCONST_SGL(-0.98987511890470f)}, {FL2FXCONST_SGL(-0.87979225745213f), FL2FXCONST_SGL(-0.71725725041680f)}, {FL2FXCONST_SGL(0.44183099021786f), FL2FXCONST_SGL(-0.93568974498761f)}, {FL2FXCONST_SGL(0.93310180125532f), FL2FXCONST_SGL(-0.99913308068246f)}, {FL2FXCONST_SGL(-0.93941931782002f), FL2FXCONST_SGL(-0.56409379640356f)}, {FL2FXCONST_SGL(-0.88590003188677f), FL2FXCONST_SGL(0.47624600491382f)}, {FL2FXCONST_SGL(0.99971463703691f), FL2FXCONST_SGL(-0.83889954253462f)}, {FL2FXCONST_SGL(-0.75376385639978f), FL2FXCONST_SGL(0.00814643438625f)}, {FL2FXCONST_SGL(0.93887685615875f), FL2FXCONST_SGL(-0.11284528204636f)}, {FL2FXCONST_SGL(0.85126435782309f), FL2FXCONST_SGL(0.52349251543547f)}, {FL2FXCONST_SGL(0.39701421446381f), FL2FXCONST_SGL(0.81779634174316f)}, {FL2FXCONST_SGL(-0.37024464187437f), FL2FXCONST_SGL(-0.87071656222959f)}, {FL2FXCONST_SGL(-0.36024828242896f), FL2FXCONST_SGL(0.34655735648287f)}, {FL2FXCONST_SGL(-0.93388812549209f), FL2FXCONST_SGL(-0.84476541096429f)}, {FL2FXCONST_SGL(-0.65298804552119f), FL2FXCONST_SGL(-0.18439575450921f)}, {FL2FXCONST_SGL(0.11960319006843f), FL2FXCONST_SGL(0.99899346780168f)}, {FL2FXCONST_SGL(0.94292565553160f), FL2FXCONST_SGL(0.83163906518293f)}, {FL2FXCONST_SGL(0.75081145286948f), FL2FXCONST_SGL(-0.35533223142265f)}, {FL2FXCONST_SGL(0.56721979748394f), FL2FXCONST_SGL(-0.24076836414499f)}, {FL2FXCONST_SGL(0.46857766746029f), FL2FXCONST_SGL(-0.30140233457198f)}, {FL2FXCONST_SGL(0.97312313923635f), FL2FXCONST_SGL(-0.99548191630031f)}, {FL2FXCONST_SGL(-0.38299976567017f), FL2FXCONST_SGL(0.98516909715427f)}, {FL2FXCONST_SGL(0.41025800019463f), FL2FXCONST_SGL(0.02116736935734f)}, {FL2FXCONST_SGL(0.09638062008048f), FL2FXCONST_SGL(0.04411984381457f)}, {FL2FXCONST_SGL(-0.85283249275397f), FL2FXCONST_SGL(0.91475563922421f)}, {FL2FXCONST_SGL(0.88866808958124f), FL2FXCONST_SGL(-0.99735267083226f)}, {FL2FXCONST_SGL(-0.48202429536989f), FL2FXCONST_SGL(-0.96805608884164f)}, {FL2FXCONST_SGL(0.27572582416567f), FL2FXCONST_SGL(0.58634753335832f)}, {FL2FXCONST_SGL(-0.65889129659168f), FL2FXCONST_SGL(0.58835634138583f)}, {FL2FXCONST_SGL(0.98838086953732f), FL2FXCONST_SGL(0.99994349600236f)}, {FL2FXCONST_SGL(-0.20651349620689f), FL2FXCONST_SGL(0.54593044066355f)}, {FL2FXCONST_SGL(-0.62126416356920f), FL2FXCONST_SGL(-0.59893681700392f)}, {FL2FXCONST_SGL(0.20320105410437f), FL2FXCONST_SGL(-0.86879180355289f)}, {FL2FXCONST_SGL(-0.97790548600584f), FL2FXCONST_SGL(0.96290806999242f)}, {FL2FXCONST_SGL(0.11112534735126f), FL2FXCONST_SGL(0.21484763313301f)}, {FL2FXCONST_SGL(-0.41368337314182f), FL2FXCONST_SGL(0.28216837680365f)}, {FL2FXCONST_SGL(0.24133038992960f), FL2FXCONST_SGL(0.51294362630238f)}, {FL2FXCONST_SGL(-0.66393410674885f), FL2FXCONST_SGL(-0.08249679629081f)}, {FL2FXCONST_SGL(-0.53697829178752f), FL2FXCONST_SGL(-0.97649903936228f)}, {FL2FXCONST_SGL(-0.97224737889348f), FL2FXCONST_SGL(0.22081333579837f)}, {FL2FXCONST_SGL(0.87392477144549f), FL2FXCONST_SGL(-0.12796173740361f)}, {FL2FXCONST_SGL(0.19050361015753f), FL2FXCONST_SGL(0.01602615387195f)}, {FL2FXCONST_SGL(-0.46353441212724f), FL2FXCONST_SGL(-0.95249041539006f)}, {FL2FXCONST_SGL(-0.07064096339021f), FL2FXCONST_SGL(-0.94479803205886f)}, {FL2FXCONST_SGL(-0.92444085484466f), FL2FXCONST_SGL(-0.10457590187436f)}, {FL2FXCONST_SGL(-0.83822593578728f), FL2FXCONST_SGL(-0.01695043208885f)}, {FL2FXCONST_SGL(0.75214681811150f), FL2FXCONST_SGL(-0.99955681042665f)}, {FL2FXCONST_SGL(-0.42102998829339f), FL2FXCONST_SGL(0.99720941999394f)}, {FL2FXCONST_SGL(-0.72094786237696f), FL2FXCONST_SGL(-0.35008961934255f)}, {FL2FXCONST_SGL(0.78843311019251f), FL2FXCONST_SGL(0.52851398958271f)}, {FL2FXCONST_SGL(0.97394027897442f), FL2FXCONST_SGL(-0.26695944086561f)}, {FL2FXCONST_SGL(0.99206463477946f), FL2FXCONST_SGL(-0.57010120849429f)}, {FL2FXCONST_SGL(0.76789609461795f), FL2FXCONST_SGL(-0.76519356730966f)}, {FL2FXCONST_SGL(-0.82002421836409f), FL2FXCONST_SGL(-0.73530179553767f)}, {FL2FXCONST_SGL(0.81924990025724f), FL2FXCONST_SGL(0.99698425250579f)}, {FL2FXCONST_SGL(-0.26719850873357f), FL2FXCONST_SGL(0.68903369776193f)}, {FL2FXCONST_SGL(-0.43311260380975f), FL2FXCONST_SGL(0.85321815947490f)}, {FL2FXCONST_SGL(0.99194979673836f), FL2FXCONST_SGL(0.91876249766422f)}, {FL2FXCONST_SGL(-0.80692001248487f), FL2FXCONST_SGL(-0.32627540663214f)}, {FL2FXCONST_SGL(0.43080003649976f), FL2FXCONST_SGL(-0.21919095636638f)}, {FL2FXCONST_SGL(0.67709491937357f), FL2FXCONST_SGL(-0.95478075822906f)}, {FL2FXCONST_SGL(0.56151770568316f), FL2FXCONST_SGL(-0.70693811747778f)}, {FL2FXCONST_SGL(0.10831862810749f), FL2FXCONST_SGL(-0.08628837174592f)}, {FL2FXCONST_SGL(0.91229417540436f), FL2FXCONST_SGL(-0.65987351408410f)}, {FL2FXCONST_SGL(-0.48972893932274f), FL2FXCONST_SGL(0.56289246362686f)}, {FL2FXCONST_SGL(-0.89033658689697f), FL2FXCONST_SGL(-0.71656563987082f)}, {FL2FXCONST_SGL(0.65269447475094f), FL2FXCONST_SGL(0.65916004833932f)}, {FL2FXCONST_SGL(0.67439478141121f), FL2FXCONST_SGL(-0.81684380846796f)}, {FL2FXCONST_SGL(-0.47770832416973f), FL2FXCONST_SGL(-0.16789556203025f)}, {FL2FXCONST_SGL(-0.99715979260878f), FL2FXCONST_SGL(-0.93565784007648f)}, {FL2FXCONST_SGL(-0.90889593602546f), FL2FXCONST_SGL(0.62034397054380f)}, {FL2FXCONST_SGL(-0.06618622548177f), FL2FXCONST_SGL(-0.23812217221359f)}, {FL2FXCONST_SGL(0.99430266919728f), FL2FXCONST_SGL(0.18812555317553f)}, {FL2FXCONST_SGL(0.97686402381843f), FL2FXCONST_SGL(-0.28664534366620f)}, {FL2FXCONST_SGL(0.94813650221268f), FL2FXCONST_SGL(-0.97506640027128f)}, {FL2FXCONST_SGL(-0.95434497492853f), FL2FXCONST_SGL(-0.79607978501983f)}, {FL2FXCONST_SGL(-0.49104783137150f), FL2FXCONST_SGL(0.32895214359663f)}, {FL2FXCONST_SGL(0.99881175120751f), FL2FXCONST_SGL(0.88993983831354f)}, {FL2FXCONST_SGL(0.50449166760303f), FL2FXCONST_SGL(-0.85995072408434f)}, {FL2FXCONST_SGL(0.47162891065108f), FL2FXCONST_SGL(-0.18680204049569f)}, {FL2FXCONST_SGL(-0.62081581361840f), FL2FXCONST_SGL(0.75000676218956f)}, {FL2FXCONST_SGL(-0.43867015250812f), FL2FXCONST_SGL(0.99998069244322f)}, {FL2FXCONST_SGL(0.98630563232075f), FL2FXCONST_SGL(-0.53578899600662f)}, {FL2FXCONST_SGL(-0.61510362277374f), FL2FXCONST_SGL(-0.89515019899997f)}, {FL2FXCONST_SGL(-0.03841517601843f), FL2FXCONST_SGL(-0.69888815681179f)}, {FL2FXCONST_SGL(-0.30102157304644f), FL2FXCONST_SGL(-0.07667808922205f)}, {FL2FXCONST_SGL(0.41881284182683f), FL2FXCONST_SGL(0.02188098922282f)}, {FL2FXCONST_SGL(-0.86135454941237f), FL2FXCONST_SGL(0.98947480909359f)}, {FL2FXCONST_SGL(0.67226861393788f), FL2FXCONST_SGL(-0.13494389011014f)}, {FL2FXCONST_SGL(-0.70737398842068f), FL2FXCONST_SGL(-0.76547349325992f)}, {FL2FXCONST_SGL(0.94044946687963f), FL2FXCONST_SGL(0.09026201157416f)}, {FL2FXCONST_SGL(-0.82386352534327f), FL2FXCONST_SGL(0.08924768823676f)}, {FL2FXCONST_SGL(-0.32070666698656f), FL2FXCONST_SGL(0.50143421908753f)}, {FL2FXCONST_SGL(0.57593163224487f), FL2FXCONST_SGL(-0.98966422921509f)}, {FL2FXCONST_SGL(-0.36326018419965f), FL2FXCONST_SGL(0.07440243123228f)}, {FL2FXCONST_SGL(0.99979044674350f), FL2FXCONST_SGL(-0.14130287347405f)}, {FL2FXCONST_SGL(-0.92366023326932f), FL2FXCONST_SGL(-0.97979298068180f)}, {FL2FXCONST_SGL(-0.44607178518598f), FL2FXCONST_SGL(-0.54233252016394f)}, {FL2FXCONST_SGL(0.44226800932956f), FL2FXCONST_SGL(0.71326756742752f)}, {FL2FXCONST_SGL(0.03671907158312f), FL2FXCONST_SGL(0.63606389366675f)}, {FL2FXCONST_SGL(0.52175424682195f), FL2FXCONST_SGL(-0.85396826735705f)}, {FL2FXCONST_SGL(-0.94701139690956f), FL2FXCONST_SGL(-0.01826348194255f)}, {FL2FXCONST_SGL(-0.98759606946049f), FL2FXCONST_SGL(0.82288714303073f)}, {FL2FXCONST_SGL(0.87434794743625f), FL2FXCONST_SGL(0.89399495655433f)}, {FL2FXCONST_SGL(-0.93412041758744f), FL2FXCONST_SGL(0.41374052024363f)}, {FL2FXCONST_SGL(0.96063943315511f), FL2FXCONST_SGL(0.93116709541280f)}, {FL2FXCONST_SGL(0.97534253457837f), FL2FXCONST_SGL(0.86150930812689f)}, {FL2FXCONST_SGL(0.99642466504163f), FL2FXCONST_SGL(0.70190043427512f)}, {FL2FXCONST_SGL(-0.94705089665984f), FL2FXCONST_SGL(-0.29580042814306f)}, {FL2FXCONST_SGL(0.91599807087376f), FL2FXCONST_SGL(-0.98147830385781f)}}; //@} /* static const FIXP_SGL harmonicPhase [2][4] = { { 1.0, 0.0, -1.0, 0.0}, { 0.0, 1.0, 0.0, -1.0} }; */ /* tables for SBR and AAC LD */ /* table for 8 time slot index */ const int FDK_sbrDecoder_envelopeTable_8[8][5] = { /* transientIndex nEnv, tranIdx, shortEnv, border1, border2, ... */ /* borders from left to right side; -1 = not in use */ /*[|T-|------]*/ {2, 0, 0, 1, -1}, /*[|-T-|-----]*/ {2, 0, 0, 2, -1}, /*[--|T-|----]*/ {3, 1, 1, 2, 4}, /*[---|T-|---]*/ {3, 1, 1, 3, 5}, /*[----|T-|--]*/ {3, 1, 1, 4, 6}, /*[-----|T--|]*/ {2, 1, 1, 5, -1}, /*[------|T-|]*/ {2, 1, 1, 6, -1}, /*[-------|T|]*/ {2, 1, 1, 7, -1}, }; /* table for 15 time slot index */ const int FDK_sbrDecoder_envelopeTable_15[15][6] = { /* transientIndex nEnv, tranIdx, shortEnv, border1, border2, ... */ /* length from left to right side; -1 = not in use */ /*[|T---|------------]*/ {2, 0, 0, 4, -1, -1}, /*[|-T---|-----------]*/ {2, 0, 0, 5, -1, -1}, /*[|--|T---|---------]*/ {3, 1, 1, 2, 6, -1}, /*[|---|T---|--------]*/ {3, 1, 1, 3, 7, -1}, /*[|----|T---|-------]*/ {3, 1, 1, 4, 8, -1}, /*[|-----|T---|------]*/ {3, 1, 1, 5, 9, -1}, /*[|------|T---|-----]*/ {3, 1, 1, 6, 10, -1}, /*[|-------|T---|----]*/ {3, 1, 1, 7, 11, -1}, /*[|--------|T---|---]*/ {3, 1, 1, 8, 12, -1}, /*[|---------|T---|--]*/ {3, 1, 1, 9, 13, -1}, /*[|----------|T----|]*/ {2, 1, 1, 10, -1, -1}, /*[|-----------|T---|]*/ {2, 1, 1, 11, -1, -1}, /*[|------------|T--|]*/ {2, 1, 1, 12, -1, -1}, /*[|-------------|T-|]*/ {2, 1, 1, 13, -1, -1}, /*[|--------------|T|]*/ {2, 1, 1, 14, -1, -1}, }; /* table for 16 time slot index */ const int FDK_sbrDecoder_envelopeTable_16[16][6] = { /* transientIndex nEnv, tranIdx, shortEnv, border1, border2, ... */ /* length from left to right side; -1 = not in use */ /*[|T---|------------|]*/ {2, 0, 0, 4, -1, -1}, /*[|-T---|-----------|]*/ {2, 0, 0, 5, -1, -1}, /*[|--|T---|----------]*/ {3, 1, 1, 2, 6, -1}, /*[|---|T---|---------]*/ {3, 1, 1, 3, 7, -1}, /*[|----|T---|--------]*/ {3, 1, 1, 4, 8, -1}, /*[|-----|T---|-------]*/ {3, 1, 1, 5, 9, -1}, /*[|------|T---|------]*/ {3, 1, 1, 6, 10, -1}, /*[|-------|T---|-----]*/ {3, 1, 1, 7, 11, -1}, /*[|--------|T---|----]*/ {3, 1, 1, 8, 12, -1}, /*[|---------|T---|---]*/ {3, 1, 1, 9, 13, -1}, /*[|----------|T---|--]*/ {3, 1, 1, 10, 14, -1}, /*[|-----------|T----|]*/ {2, 1, 1, 11, -1, -1}, /*[|------------|T---|]*/ {2, 1, 1, 12, -1, -1}, /*[|-------------|T--|]*/ {2, 1, 1, 13, -1, -1}, /*[|--------------|T-|]*/ {2, 1, 1, 14, -1, -1}, /*[|---------------|T|]*/ {2, 1, 1, 15, -1, -1}, }; /*! \name FrameInfoDefaults Predefined envelope positions for the FIX-FIX case (static framing) */ //@{ const FRAME_INFO FDK_sbrDecoder_sbr_frame_info1_15 = { 0, 1, {0, 15, 0, 0, 0, 0}, {1, 0, 0, 0, 0}, -1, 1, {0, 15, 0}, {0, 0, 0}, 0, 0}; const FRAME_INFO FDK_sbrDecoder_sbr_frame_info2_15 = { 0, 2, {0, 8, 15, 0, 0, 0}, {1, 1, 0, 0, 0}, -1, 2, {0, 8, 15}, {0, 0, 0}, 0, 0}; const FRAME_INFO FDK_sbrDecoder_sbr_frame_info4_15 = { 0, 4, {0, 4, 8, 12, 15, 0}, {1, 1, 1, 1, 0}, -1, 2, {0, 8, 15}, {0, 0, 0}, 0, 0}; #if (MAX_ENVELOPES >= 8) const FRAME_INFO FDK_sbrDecoder_sbr_frame_info8_15 = { 0, 8, {0, 2, 4, 6, 8, 10, 12, 14, 15}, {1, 1, 1, 1, 1, 1, 1, 1}, -1, 2, {0, 8, 15}, {0, 0, 0}, 0, 0}; #endif const FRAME_INFO FDK_sbrDecoder_sbr_frame_info1_16 = { 0, 1, {0, 16, 0, 0, 0, 0}, {1, 0, 0, 0, 0}, -1, 1, {0, 16, 0}, {0, 0, 0}, 0, 0}; const FRAME_INFO FDK_sbrDecoder_sbr_frame_info2_16 = { 0, 2, {0, 8, 16, 0, 0, 0}, {1, 1, 0, 0, 0}, -1, 2, {0, 8, 16}, {0, 0, 0}, 0, 0}; const FRAME_INFO FDK_sbrDecoder_sbr_frame_info4_16 = { 0, 4, {0, 4, 8, 12, 16, 0}, {1, 1, 1, 1, 0}, -1, 2, {0, 8, 16}, {0, 0, 0}, 0, 0}; #if (MAX_ENVELOPES >= 8) const FRAME_INFO FDK_sbrDecoder_sbr_frame_info8_16 = { 0, 8, {0, 2, 4, 6, 8, 10, 12, 14, 16}, {1, 1, 1, 1, 1, 1, 1, 1}, -1, 2, {0, 8, 16}, {0, 0, 0}, 0, 0}; #endif //@} /*! \name SBR_HuffmanTables SBR Huffman Table Overview: \n \n o envelope level, 1.5 dB: \n 1) sbr_huffBook_EnvLevel10T[120][2] \n 2) sbr_huffBook_EnvLevel10F[120][2] \n \n o envelope balance, 1.5 dB: \n 3) sbr_huffBook_EnvBalance10T[48][2] \n 4) sbr_huffBook_EnvBalance10F[48][2] \n \n o envelope level, 3.0 dB: \n 5) sbr_huffBook_EnvLevel11T[62][2] \n 6) sbr_huffBook_EnvLevel11F[62][2] \n \n o envelope balance, 3.0 dB: \n 7) sbr_huffBook_EnvBalance11T[24][2] \n 8) sbr_huffBook_EnvBalance11F[24][2] \n \n o noise level, 3.0 dB: \n 9) sbr_huffBook_NoiseLevel11T[62][2] \n -) (sbr_huffBook_EnvLevel11F[62][2] is used for freq dir)\n \n o noise balance, 3.0 dB: \n 10) sbr_huffBook_NoiseBalance11T[24][2]\n -) (sbr_huffBook_EnvBalance11F[24][2] is used for freq dir)\n \n (1.5 dB is never used for noise) */ //@{ const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel10T[120][2] = { {1, 2}, {-64, -65}, {3, 4}, {-63, -66}, {5, 6}, {-62, -67}, {7, 8}, {-61, -68}, {9, 10}, {-60, -69}, {11, 12}, {-59, -70}, {13, 14}, {-58, -71}, {15, 16}, {-57, -72}, {17, 18}, {-73, -56}, {19, 21}, {-74, 20}, {-55, -75}, {22, 26}, {23, 24}, {-54, -76}, {-77, 25}, {-53, -78}, {27, 34}, {28, 29}, {-52, -79}, {30, 31}, {-80, -51}, {32, 33}, {-83, -82}, {-81, -50}, {35, 57}, {36, 40}, {37, 38}, {-88, -84}, {-48, 39}, {-90, -85}, {41, 46}, {42, 43}, {-49, -87}, {44, 45}, {-89, -86}, {-124, -123}, {47, 50}, {48, 49}, {-122, -121}, {-120, -119}, {51, 54}, {52, 53}, {-118, -117}, {-116, -115}, {55, 56}, {-114, -113}, {-112, -111}, {58, 89}, {59, 74}, {60, 67}, {61, 64}, {62, 63}, {-110, -109}, {-108, -107}, {65, 66}, {-106, -105}, {-104, -103}, {68, 71}, {69, 70}, {-102, -101}, {-100, -99}, {72, 73}, {-98, -97}, {-96, -95}, {75, 82}, {76, 79}, {77, 78}, {-94, -93}, {-92, -91}, {80, 81}, {-47, -46}, {-45, -44}, {83, 86}, {84, 85}, {-43, -42}, {-41, -40}, {87, 88}, {-39, -38}, {-37, -36}, {90, 105}, {91, 98}, {92, 95}, {93, 94}, {-35, -34}, {-33, -32}, {96, 97}, {-31, -30}, {-29, -28}, {99, 102}, {100, 101}, {-27, -26}, {-25, -24}, {103, 104}, {-23, -22}, {-21, -20}, {106, 113}, {107, 110}, {108, 109}, {-19, -18}, {-17, -16}, {111, 112}, {-15, -14}, {-13, -12}, {114, 117}, {115, 116}, {-11, -10}, {-9, -8}, {118, 119}, {-7, -6}, {-5, -4}}; const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel10F[120][2] = { {1, 2}, {-64, -65}, {3, 4}, {-63, -66}, {5, 6}, {-67, -62}, {7, 8}, {-68, -61}, {9, 10}, {-69, -60}, {11, 13}, {-70, 12}, {-59, -71}, {14, 16}, {-58, 15}, {-72, -57}, {17, 19}, {-73, 18}, {-56, -74}, {20, 23}, {21, 22}, {-55, -75}, {-54, -53}, {24, 27}, {25, 26}, {-76, -52}, {-77, -51}, {28, 31}, {29, 30}, {-50, -78}, {-79, -49}, {32, 36}, {33, 34}, {-48, -47}, {-80, 35}, {-81, -82}, {37, 47}, {38, 41}, {39, 40}, {-83, -46}, {-45, -84}, {42, 44}, {-85, 43}, {-44, -43}, {45, 46}, {-88, -87}, {-86, -90}, {48, 66}, {49, 56}, {50, 53}, {51, 52}, {-92, -42}, {-41, -39}, {54, 55}, {-105, -89}, {-38, -37}, {57, 60}, {58, 59}, {-94, -91}, {-40, -36}, {61, 63}, {-20, 62}, {-115, -110}, {64, 65}, {-108, -107}, {-101, -97}, {67, 89}, {68, 75}, {69, 72}, {70, 71}, {-95, -93}, {-34, -27}, {73, 74}, {-22, -17}, {-16, -124}, {76, 82}, {77, 79}, {-123, 78}, {-122, -121}, {80, 81}, {-120, -119}, {-118, -117}, {83, 86}, {84, 85}, {-116, -114}, {-113, -112}, {87, 88}, {-111, -109}, {-106, -104}, {90, 105}, {91, 98}, {92, 95}, {93, 94}, {-103, -102}, {-100, -99}, {96, 97}, {-98, -96}, {-35, -33}, {99, 102}, {100, 101}, {-32, -31}, {-30, -29}, {103, 104}, {-28, -26}, {-25, -24}, {106, 113}, {107, 110}, {108, 109}, {-23, -21}, {-19, -18}, {111, 112}, {-15, -14}, {-13, -12}, {114, 117}, {115, 116}, {-11, -10}, {-9, -8}, {118, 119}, {-7, -6}, {-5, -4}}; const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance10T[48][2] = { {-64, 1}, {-63, 2}, {-65, 3}, {-62, 4}, {-66, 5}, {-61, 6}, {-67, 7}, {-60, 8}, {-68, 9}, {10, 11}, {-69, -59}, {12, 13}, {-70, -58}, {14, 28}, {15, 21}, {16, 18}, {-57, 17}, {-71, -56}, {19, 20}, {-88, -87}, {-86, -85}, {22, 25}, {23, 24}, {-84, -83}, {-82, -81}, {26, 27}, {-80, -79}, {-78, -77}, {29, 36}, {30, 33}, {31, 32}, {-76, -75}, {-74, -73}, {34, 35}, {-72, -55}, {-54, -53}, {37, 41}, {38, 39}, {-52, -51}, {-50, 40}, {-49, -48}, {42, 45}, {43, 44}, {-47, -46}, {-45, -44}, {46, 47}, {-43, -42}, {-41, -40}}; const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance10F[48][2] = { {-64, 1}, {-65, 2}, {-63, 3}, {-66, 4}, {-62, 5}, {-61, 6}, {-67, 7}, {-68, 8}, {-60, 9}, {10, 11}, {-69, -59}, {-70, 12}, {-58, 13}, {14, 17}, {-71, 15}, {-57, 16}, {-56, -73}, {18, 32}, {19, 25}, {20, 22}, {-72, 21}, {-88, -87}, {23, 24}, {-86, -85}, {-84, -83}, {26, 29}, {27, 28}, {-82, -81}, {-80, -79}, {30, 31}, {-78, -77}, {-76, -75}, {33, 40}, {34, 37}, {35, 36}, {-74, -55}, {-54, -53}, {38, 39}, {-52, -51}, {-50, -49}, {41, 44}, {42, 43}, {-48, -47}, {-46, -45}, {45, 46}, {-44, -43}, {-42, 47}, {-41, -40}}; const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel11T[62][2] = { {-64, 1}, {-65, 2}, {-63, 3}, {-66, 4}, {-62, 5}, {-67, 6}, {-61, 7}, {-68, 8}, {-60, 9}, {10, 11}, {-69, -59}, {12, 14}, {-70, 13}, {-71, -58}, {15, 18}, {16, 17}, {-72, -57}, {-73, -74}, {19, 22}, {-56, 20}, {-55, 21}, {-54, -77}, {23, 31}, {24, 25}, {-75, -76}, {26, 27}, {-78, -53}, {28, 29}, {-52, -95}, {-94, 30}, {-93, -92}, {32, 47}, {33, 40}, {34, 37}, {35, 36}, {-91, -90}, {-89, -88}, {38, 39}, {-87, -86}, {-85, -84}, {41, 44}, {42, 43}, {-83, -82}, {-81, -80}, {45, 46}, {-79, -51}, {-50, -49}, {48, 55}, {49, 52}, {50, 51}, {-48, -47}, {-46, -45}, {53, 54}, {-44, -43}, {-42, -41}, {56, 59}, {57, 58}, {-40, -39}, {-38, -37}, {60, 61}, {-36, -35}, {-34, -33}}; const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel11F[62][2] = { {-64, 1}, {-65, 2}, {-63, 3}, {-66, 4}, {-62, 5}, {-67, 6}, {7, 8}, {-61, -68}, {9, 10}, {-60, -69}, {11, 12}, {-59, -70}, {13, 14}, {-58, -71}, {15, 16}, {-57, -72}, {17, 19}, {-56, 18}, {-55, -73}, {20, 24}, {21, 22}, {-74, -54}, {-53, 23}, {-75, -76}, {25, 30}, {26, 27}, {-52, -51}, {28, 29}, {-77, -79}, {-50, -49}, {31, 39}, {32, 35}, {33, 34}, {-78, -46}, {-82, -88}, {36, 37}, {-83, -48}, {-47, 38}, {-86, -85}, {40, 47}, {41, 44}, {42, 43}, {-80, -44}, {-43, -42}, {45, 46}, {-39, -87}, {-84, -40}, {48, 55}, {49, 52}, {50, 51}, {-95, -94}, {-93, -92}, {53, 54}, {-91, -90}, {-89, -81}, {56, 59}, {57, 58}, {-45, -41}, {-38, -37}, {60, 61}, {-36, -35}, {-34, -33}}; const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance11T[24][2] = { {-64, 1}, {-63, 2}, {-65, 3}, {-66, 4}, {-62, 5}, {-61, 6}, {-67, 7}, {-68, 8}, {-60, 9}, {10, 16}, {11, 13}, {-69, 12}, {-76, -75}, {14, 15}, {-74, -73}, {-72, -71}, {17, 20}, {18, 19}, {-70, -59}, {-58, -57}, {21, 22}, {-56, -55}, {-54, 23}, {-53, -52}}; const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance11F[24][2] = { {-64, 1}, {-65, 2}, {-63, 3}, {-66, 4}, {-62, 5}, {-61, 6}, {-67, 7}, {-68, 8}, {-60, 9}, {10, 13}, {-69, 11}, {-59, 12}, {-58, -76}, {14, 17}, {15, 16}, {-75, -74}, {-73, -72}, {18, 21}, {19, 20}, {-71, -70}, {-57, -56}, {22, 23}, {-55, -54}, {-53, -52}}; const SCHAR FDK_sbrDecoder_sbr_huffBook_NoiseLevel11T[62][2] = { {-64, 1}, {-63, 2}, {-65, 3}, {-66, 4}, {-62, 5}, {-67, 6}, {7, 8}, {-61, -68}, {9, 30}, {10, 15}, {-60, 11}, {-69, 12}, {13, 14}, {-59, -53}, {-95, -94}, {16, 23}, {17, 20}, {18, 19}, {-93, -92}, {-91, -90}, {21, 22}, {-89, -88}, {-87, -86}, {24, 27}, {25, 26}, {-85, -84}, {-83, -82}, {28, 29}, {-81, -80}, {-79, -78}, {31, 46}, {32, 39}, {33, 36}, {34, 35}, {-77, -76}, {-75, -74}, {37, 38}, {-73, -72}, {-71, -70}, {40, 43}, {41, 42}, {-58, -57}, {-56, -55}, {44, 45}, {-54, -52}, {-51, -50}, {47, 54}, {48, 51}, {49, 50}, {-49, -48}, {-47, -46}, {52, 53}, {-45, -44}, {-43, -42}, {55, 58}, {56, 57}, {-41, -40}, {-39, -38}, {59, 60}, {-37, -36}, {-35, 61}, {-34, -33}}; const SCHAR FDK_sbrDecoder_sbr_huffBook_NoiseBalance11T[24][2] = { {-64, 1}, {-65, 2}, {-63, 3}, {4, 9}, {-66, 5}, {-62, 6}, {7, 8}, {-76, -75}, {-74, -73}, {10, 17}, {11, 14}, {12, 13}, {-72, -71}, {-70, -69}, {15, 16}, {-68, -67}, {-61, -60}, {18, 21}, {19, 20}, {-59, -58}, {-57, -56}, {22, 23}, {-55, -54}, {-53, -52}}; //@} /*! \name parametric stereo \brief constants used by the parametric stereo part of the decoder */ /* constants used in psbitdec.cpp */ /* FIX_BORDER can have 0, 1, 2, 4 envelopes */ const UCHAR FDK_sbrDecoder_aFixNoEnvDecode[4] = {0, 1, 2, 4}; /* IID & ICC Huffman codebooks */ const SCHAR aBookPsIidTimeDecode[28][2] = { {-64, 1}, {-65, 2}, {-63, 3}, {-66, 4}, {-62, 5}, {-67, 6}, {-61, 7}, {-68, 8}, {-60, 9}, {-69, 10}, {-59, 11}, {-70, 12}, {-58, 13}, {-57, 14}, {-71, 15}, {16, 17}, {-56, -72}, {18, 21}, {19, 20}, {-55, -78}, {-77, -76}, {22, 25}, {23, 24}, {-75, -74}, {-73, -54}, {26, 27}, {-53, -52}, {-51, -50}}; const SCHAR aBookPsIidFreqDecode[28][2] = { {-64, 1}, {2, 3}, {-63, -65}, {4, 5}, {-62, -66}, {6, 7}, {-61, -67}, {8, 9}, {-68, -60}, {-59, 10}, {-69, 11}, {-58, 12}, {-70, 13}, {-71, 14}, {-57, 15}, {16, 17}, {-56, -72}, {18, 19}, {-55, -54}, {20, 21}, {-73, -53}, {22, 24}, {-74, 23}, {-75, -78}, {25, 26}, {-77, -76}, {-52, 27}, {-51, -50}}; const SCHAR aBookPsIccTimeDecode[14][2] = { {-64, 1}, {-63, 2}, {-65, 3}, {-62, 4}, {-66, 5}, {-61, 6}, {-67, 7}, {-60, 8}, {-68, 9}, {-59, 10}, {-69, 11}, {-58, 12}, {-70, 13}, {-71, -57}}; const SCHAR aBookPsIccFreqDecode[14][2] = { {-64, 1}, {-63, 2}, {-65, 3}, {-62, 4}, {-66, 5}, {-61, 6}, {-67, 7}, {-60, 8}, {-59, 9}, {-68, 10}, {-58, 11}, {-69, 12}, {-57, 13}, {-70, -71}}; /* IID-fine Huffman codebooks */ const SCHAR aBookPsIidFineTimeDecode[60][2] = { {1, -64}, {-63, 2}, {3, -65}, {4, 59}, {5, 7}, {6, -67}, {-68, -60}, {-61, 8}, {9, 11}, {-59, 10}, {-70, -58}, {12, 41}, {13, 20}, {14, -71}, {-55, 15}, {-53, 16}, {17, -77}, {18, 19}, {-85, -84}, {-46, -45}, {-57, 21}, {22, 40}, {23, 29}, {-51, 24}, {25, 26}, {-83, -82}, {27, 28}, {-90, -38}, {-92, -91}, {30, 37}, {31, 34}, {32, 33}, {-35, -34}, {-37, -36}, {35, 36}, {-94, -93}, {-89, -39}, {38, -79}, {39, -81}, {-88, -40}, {-74, -54}, {42, -69}, {43, 44}, {-72, -56}, {45, 52}, {46, 50}, {47, -76}, {-49, 48}, {-47, 49}, {-87, -41}, {-52, 51}, {-78, -50}, {53, -73}, {54, -75}, {55, 57}, {56, -80}, {-86, -42}, {-48, 58}, {-44, -43}, {-66, -62}}; const SCHAR aBookPsIidFineFreqDecode[60][2] = { {1, -64}, {2, 4}, {3, -65}, {-66, -62}, {-63, 5}, {6, 7}, {-67, -61}, {8, 9}, {-68, -60}, {10, 11}, {-69, -59}, {12, 13}, {-70, -58}, {14, 18}, {-57, 15}, {16, -72}, {-54, 17}, {-75, -53}, {19, 37}, {-56, 20}, {21, -73}, {22, 29}, {23, -76}, {24, -78}, {25, 28}, {26, 27}, {-85, -43}, {-83, -45}, {-81, -47}, {-52, 30}, {-50, 31}, {32, -79}, {33, 34}, {-82, -46}, {35, 36}, {-90, -89}, {-92, -91}, {38, -71}, {-55, 39}, {40, -74}, {41, 50}, {42, -77}, {-49, 43}, {44, 47}, {45, 46}, {-86, -42}, {-88, -87}, {48, 49}, {-39, -38}, {-41, -40}, {-51, 51}, {52, 59}, {53, 56}, {54, 55}, {-35, -34}, {-37, -36}, {57, 58}, {-94, -93}, {-84, -44}, {-80, -48}}; /* constants used in psdec.cpp */ /* the values of the following 3 tables are shiftet right by 1 ! */ const FIXP_DBL ScaleFactors[NO_IID_LEVELS] = { 0x5a5ded00, 0x59cd0400, 0x58c29680, 0x564c2e80, 0x52a3d480, 0x4c8be080, 0x46df3080, 0x40000000, 0x384ba5c0, 0x304c2980, 0x24e9f640, 0x1b4a2940, 0x11b5c0a0, 0x0b4e2540, 0x0514ea90}; const FIXP_DBL ScaleFactorsFine[NO_IID_LEVELS_FINE] = { 0x5a825c00, 0x5a821c00, 0x5a815100, 0x5a7ed000, 0x5a76e600, 0x5a5ded00, 0x5a39b880, 0x59f1fd00, 0x5964d680, 0x5852ca00, 0x564c2e80, 0x54174480, 0x50ea7500, 0x4c8be080, 0x46df3080, 0x40000000, 0x384ba5c0, 0x304c2980, 0x288dd240, 0x217a2900, 0x1b4a2940, 0x13c5ece0, 0x0e2b0090, 0x0a178ef0, 0x072ab798, 0x0514ea90, 0x02dc5944, 0x019bf87c, 0x00e7b173, 0x00824b8b, 0x00494568}; const FIXP_DBL Alphas[NO_ICC_LEVELS] = { 0x00000000, 0x0b6b5be0, 0x12485f80, 0x1da2fa40, 0x2637ebc0, 0x3243f6c0, 0x466b7480, 0x6487ed80}; const UCHAR bins2groupMap20[NO_IID_GROUPS] = { 0, 0, 1, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19}; const UCHAR FDK_sbrDecoder_aNoIidBins[3] = { NO_LOW_RES_IID_BINS, NO_MID_RES_IID_BINS, NO_HI_RES_IID_BINS}; const UCHAR FDK_sbrDecoder_aNoIccBins[3] = { NO_LOW_RES_ICC_BINS, NO_MID_RES_ICC_BINS, NO_HI_RES_ICC_BINS}; /************************************************************************/ /*! \brief Create lookup tables for some arithmetic functions The tables would normally be defined as const arrays, but initialization at run time allows to specify their accuracy. */ /************************************************************************/ /* 1/x-table: (example for INV_TABLE_BITS 8) The table covers an input range from 0.5 to 1.0 with a step size of 1/512, starting at 0.5 + 1/512. Each table entry corresponds to an input interval starting 1/1024 below the exact value and ending 1/1024 above it. The table is actually a 0.5/x-table, so that the output range is again 0.5...1.0 and the exponent of the result must be increased by 1. Input range Index in table result ------------------------------------------------------------------- 0.500000...0.500976 - 0.5 / 0.500000 = 1.000000 0.500976...0.502930 0 0.5 / 0.501953 = 0.996109 0.502930...0.500488 1 0.5 / 0.503906 = 0.992248 ... 0.999023...1.000000 255 0.5 / 1.000000 = 0.500000 for (i=0; iprevFact_mag[band] = FL2FXCONST_DBL(0.5f); } for (band = 0; band < SBRDEC_MAX_DRC_BANDS; band++) { hDrcData->currFact_mag[band] = FL2FXCONST_DBL(0.5f); hDrcData->nextFact_mag[band] = FL2FXCONST_DBL(0.5f); } hDrcData->prevFact_exp = 1; hDrcData->currFact_exp = 1; hDrcData->nextFact_exp = 1; hDrcData->numBandsCurr = 1; hDrcData->numBandsNext = 1; hDrcData->winSequenceCurr = 0; hDrcData->winSequenceNext = 0; hDrcData->drcInterpolationSchemeCurr = 0; hDrcData->drcInterpolationSchemeNext = 0; hDrcData->enable = 0; } /*! \brief Swap DRC QMF scaling factors after they have been applied. \hDrcData Handle to DRC channel data. \return none */ void sbrDecoder_drcUpdateChannel(HANDLE_SBR_DRC_CHANNEL hDrcData) { if (hDrcData == NULL) { return; } if (hDrcData->enable != 1) { return; } /* swap previous data */ FDKmemcpy(hDrcData->currFact_mag, hDrcData->nextFact_mag, SBRDEC_MAX_DRC_BANDS * sizeof(FIXP_DBL)); hDrcData->currFact_exp = hDrcData->nextFact_exp; hDrcData->numBandsCurr = hDrcData->numBandsNext; FDKmemcpy(hDrcData->bandTopCurr, hDrcData->bandTopNext, SBRDEC_MAX_DRC_BANDS * sizeof(USHORT)); hDrcData->drcInterpolationSchemeCurr = hDrcData->drcInterpolationSchemeNext; hDrcData->winSequenceCurr = hDrcData->winSequenceNext; } /*! \brief Apply DRC factors slot based. \hDrcData Handle to DRC channel data. \qmfRealSlot Pointer to real valued QMF data of one time slot. \qmfImagSlot Pointer to the imaginary QMF data of one time slot. \col Number of the time slot. \numQmfSubSamples Total number of time slots for one frame. \scaleFactor Pointer to the out scale factor of the time slot. \return None. */ void sbrDecoder_drcApplySlot(HANDLE_SBR_DRC_CHANNEL hDrcData, FIXP_DBL *qmfRealSlot, FIXP_DBL *qmfImagSlot, int col, int numQmfSubSamples, int maxShift) { const UCHAR *winBorderToColMap; int band, bottomMdct, topMdct, bin, useLP; int indx = numQmfSubSamples - (numQmfSubSamples >> 1) - 10; /* l_border */ int frameLenFlag = (numQmfSubSamples == 30) ? 1 : 0; int frameSize = (frameLenFlag == 1) ? 960 : 1024; const FIXP_DBL *fact_mag = NULL; INT fact_exp = 0; UINT numBands = 0; USHORT *bandTop = NULL; int shortDrc = 0; FIXP_DBL alphaValue = FL2FXCONST_DBL(0.0f); if (hDrcData == NULL) { return; } if (hDrcData->enable != 1) { return; } winBorderToColMap = winBorderToColMappingTab[frameLenFlag]; useLP = (qmfImagSlot == NULL) ? 1 : 0; col += indx; bottomMdct = 0; /* get respective data and calc interpolation factor */ if (col < (numQmfSubSamples >> 1)) { /* first half of current frame */ if (hDrcData->winSequenceCurr != 2) { /* long window */ int j = col + (numQmfSubSamples >> 1); if (hDrcData->drcInterpolationSchemeCurr == 0) { INT k = (frameLenFlag) ? 0x4444445 : 0x4000000; alphaValue = (FIXP_DBL)(j * k); } else { if (j >= (int)winBorderToColMap[hDrcData->drcInterpolationSchemeCurr]) { alphaValue = (FIXP_DBL)MAXVAL_DBL; } } } else { /* short windows */ shortDrc = 1; } fact_mag = hDrcData->currFact_mag; fact_exp = hDrcData->currFact_exp; numBands = hDrcData->numBandsCurr; bandTop = hDrcData->bandTopCurr; } else if (col < numQmfSubSamples) { /* second half of current frame */ if (hDrcData->winSequenceNext != 2) { /* next: long window */ int j = col - (numQmfSubSamples >> 1); if (hDrcData->drcInterpolationSchemeNext == 0) { INT k = (frameLenFlag) ? 0x4444445 : 0x4000000; alphaValue = (FIXP_DBL)(j * k); } else { if (j >= (int)winBorderToColMap[hDrcData->drcInterpolationSchemeNext]) { alphaValue = (FIXP_DBL)MAXVAL_DBL; } } fact_mag = hDrcData->nextFact_mag; fact_exp = hDrcData->nextFact_exp; numBands = hDrcData->numBandsNext; bandTop = hDrcData->bandTopNext; } else { /* next: short windows */ if (hDrcData->winSequenceCurr != 2) { /* current: long window */ alphaValue = (FIXP_DBL)0; fact_mag = hDrcData->nextFact_mag; fact_exp = hDrcData->nextFact_exp; numBands = hDrcData->numBandsNext; bandTop = hDrcData->bandTopNext; } else { /* current: short windows */ shortDrc = 1; fact_mag = hDrcData->currFact_mag; fact_exp = hDrcData->currFact_exp; numBands = hDrcData->numBandsCurr; bandTop = hDrcData->bandTopCurr; } } } else { /* first half of next frame */ if (hDrcData->winSequenceNext != 2) { /* long window */ int j = col - (numQmfSubSamples >> 1); if (hDrcData->drcInterpolationSchemeNext == 0) { INT k = (frameLenFlag) ? 0x4444445 : 0x4000000; alphaValue = (FIXP_DBL)(j * k); } else { if (j >= (int)winBorderToColMap[hDrcData->drcInterpolationSchemeNext]) { alphaValue = (FIXP_DBL)MAXVAL_DBL; } } } else { /* short windows */ shortDrc = 1; } fact_mag = hDrcData->nextFact_mag; fact_exp = hDrcData->nextFact_exp; numBands = hDrcData->numBandsNext; bandTop = hDrcData->bandTopNext; col -= numQmfSubSamples; } /* process bands */ for (band = 0; band < (int)numBands; band++) { int bottomQmf, topQmf; FIXP_DBL drcFact_mag = (FIXP_DBL)MAXVAL_DBL; topMdct = (bandTop[band] + 1) << 2; if (!shortDrc) { /* long window */ if (frameLenFlag) { /* 960 framing */ bottomQmf = fMultIfloor((FIXP_DBL)0x4444445, bottomMdct); topQmf = fMultIfloor((FIXP_DBL)0x4444445, topMdct); topMdct = 30 * topQmf; } else { /* 1024 framing */ topMdct &= ~0x1f; bottomQmf = bottomMdct >> 5; topQmf = topMdct >> 5; } if (band == ((int)numBands - 1)) { topQmf = (64); } for (bin = bottomQmf; bin < topQmf; bin++) { FIXP_DBL drcFact1_mag = hDrcData->prevFact_mag[bin]; FIXP_DBL drcFact2_mag = fact_mag[band]; /* normalize scale factors */ if (hDrcData->prevFact_exp < maxShift) { drcFact1_mag >>= maxShift - hDrcData->prevFact_exp; } if (fact_exp < maxShift) { drcFact2_mag >>= maxShift - fact_exp; } /* interpolate */ if (alphaValue == (FIXP_DBL)0) { drcFact_mag = drcFact1_mag; } else if (alphaValue == (FIXP_DBL)MAXVAL_DBL) { drcFact_mag = drcFact2_mag; } else { drcFact_mag = fMult(alphaValue, drcFact2_mag) + fMult(((FIXP_DBL)MAXVAL_DBL - alphaValue), drcFact1_mag); } /* apply scaling */ qmfRealSlot[bin] = fMult(qmfRealSlot[bin], drcFact_mag); if (!useLP) { qmfImagSlot[bin] = fMult(qmfImagSlot[bin], drcFact_mag); } /* save previous factors */ if (col == (numQmfSubSamples >> 1) - 1) { hDrcData->prevFact_mag[bin] = fact_mag[band]; } } } else { /* short windows */ unsigned startWinIdx, stopWinIdx; int startCol, stopCol; FIXP_DBL invFrameSizeDiv8 = (frameLenFlag) ? (FIXP_DBL)0x1111112 : (FIXP_DBL)0x1000000; /* limit top at the frame borders */ if (topMdct < 0) { topMdct = 0; } if (topMdct >= frameSize) { topMdct = frameSize - 1; } if (frameLenFlag) { /* 960 framing */ topMdct = fMultIfloor((FIXP_DBL)0x78000000, fMultIfloor((FIXP_DBL)0x22222223, topMdct) << 2); startWinIdx = fMultIfloor(invFrameSizeDiv8, bottomMdct) + 1; /* winBorderToColMap table has offset of 1 */ stopWinIdx = fMultIceil(invFrameSizeDiv8 - (FIXP_DBL)1, topMdct) + 1; } else { /* 1024 framing */ topMdct &= ~0x03; startWinIdx = fMultIfloor(invFrameSizeDiv8, bottomMdct) + 1; stopWinIdx = fMultIceil(invFrameSizeDiv8, topMdct) + 1; } /* startCol is truncated to the nearest corresponding start subsample in the QMF of the short window bottom is present in:*/ startCol = (int)winBorderToColMap[startWinIdx]; /* stopCol is rounded upwards to the nearest corresponding stop subsample in the QMF of the short window top is present in. */ stopCol = (int)winBorderToColMap[stopWinIdx]; bottomQmf = fMultIfloor(invFrameSizeDiv8, ((bottomMdct % (numQmfSubSamples << 2)) << 5)); topQmf = fMultIfloor(invFrameSizeDiv8, ((topMdct % (numQmfSubSamples << 2)) << 5)); /* extend last band */ if (band == ((int)numBands - 1)) { topQmf = (64); stopCol = numQmfSubSamples; stopWinIdx = 10; } if (topQmf == 0) { if (frameLenFlag) { FIXP_DBL rem = fMult(invFrameSizeDiv8, (FIXP_DBL)(topMdct << (DFRACT_BITS - 12))); if ((LONG)rem & (LONG)0x1F) { stopWinIdx -= 1; stopCol = (int)winBorderToColMap[stopWinIdx]; } } topQmf = (64); } /* save previous factors */ if (stopCol == numQmfSubSamples) { int tmpBottom = bottomQmf; if ((int)winBorderToColMap[8] > startCol) { tmpBottom = 0; /* band starts in previous short window */ } for (bin = tmpBottom; bin < topQmf; bin++) { hDrcData->prevFact_mag[bin] = fact_mag[band]; } } /* apply */ if ((col >= startCol) && (col < stopCol)) { if (col >= (int)winBorderToColMap[startWinIdx + 1]) { bottomQmf = 0; /* band starts in previous short window */ } if (col < (int)winBorderToColMap[stopWinIdx - 1]) { topQmf = (64); /* band ends in next short window */ } drcFact_mag = fact_mag[band]; /* normalize scale factor */ if (fact_exp < maxShift) { drcFact_mag >>= maxShift - fact_exp; } /* apply scaling */ for (bin = bottomQmf; bin < topQmf; bin++) { qmfRealSlot[bin] = fMult(qmfRealSlot[bin], drcFact_mag); if (!useLP) { qmfImagSlot[bin] = fMult(qmfImagSlot[bin], drcFact_mag); } } } } bottomMdct = topMdct; } /* end of bands loop */ if (col == (numQmfSubSamples >> 1) - 1) { hDrcData->prevFact_exp = fact_exp; } } /*! \brief Apply DRC factors frame based. \hDrcData Handle to DRC channel data. \qmfRealSlot Pointer to real valued QMF data of the whole frame. \qmfImagSlot Pointer to the imaginary QMF data of the whole frame. \numQmfSubSamples Total number of time slots for one frame. \scaleFactor Pointer to the out scale factor of the frame. \return None. */ void sbrDecoder_drcApply(HANDLE_SBR_DRC_CHANNEL hDrcData, FIXP_DBL **QmfBufferReal, FIXP_DBL **QmfBufferImag, int numQmfSubSamples, int *scaleFactor) { int col; int maxShift = 0; if (hDrcData == NULL) { return; } if (hDrcData->enable == 0) { return; /* Avoid changing the scaleFactor even though the processing is disabled. */ } /* get max scale factor */ if (hDrcData->prevFact_exp > maxShift) { maxShift = hDrcData->prevFact_exp; } if (hDrcData->currFact_exp > maxShift) { maxShift = hDrcData->currFact_exp; } if (hDrcData->nextFact_exp > maxShift) { maxShift = hDrcData->nextFact_exp; } for (col = 0; col < numQmfSubSamples; col++) { FIXP_DBL *qmfSlotReal = QmfBufferReal[col]; FIXP_DBL *qmfSlotImag = (QmfBufferImag == NULL) ? NULL : QmfBufferImag[col]; sbrDecoder_drcApplySlot(hDrcData, qmfSlotReal, qmfSlotImag, col, numQmfSubSamples, maxShift); } *scaleFactor += maxShift; } fdk-aac-2.0.2/libSBRdec/src/sbrdec_drc.h000066400000000000000000000137741404225310700176360ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Christian Griebel Description: Dynamic range control (DRC) decoder tool for SBR *******************************************************************************/ #ifndef SBRDEC_DRC_H #define SBRDEC_DRC_H #include "sbrdecoder.h" #define SBRDEC_MAX_DRC_CHANNELS (8) #define SBRDEC_MAX_DRC_BANDS (16) typedef struct { FIXP_DBL prevFact_mag[(64)]; INT prevFact_exp; FIXP_DBL currFact_mag[SBRDEC_MAX_DRC_BANDS]; FIXP_DBL nextFact_mag[SBRDEC_MAX_DRC_BANDS]; INT currFact_exp; INT nextFact_exp; UINT numBandsCurr; UINT numBandsNext; USHORT bandTopCurr[SBRDEC_MAX_DRC_BANDS]; USHORT bandTopNext[SBRDEC_MAX_DRC_BANDS]; SHORT drcInterpolationSchemeCurr; SHORT drcInterpolationSchemeNext; SHORT enable; UCHAR winSequenceCurr; UCHAR winSequenceNext; } SBRDEC_DRC_CHANNEL; typedef SBRDEC_DRC_CHANNEL *HANDLE_SBR_DRC_CHANNEL; void sbrDecoder_drcInitChannel(HANDLE_SBR_DRC_CHANNEL hDrcData); void sbrDecoder_drcUpdateChannel(HANDLE_SBR_DRC_CHANNEL hDrcData); void sbrDecoder_drcApplySlot(HANDLE_SBR_DRC_CHANNEL hDrcData, FIXP_DBL *qmfRealSlot, FIXP_DBL *qmfImagSlot, int col, int numQmfSubSamples, int maxShift); void sbrDecoder_drcApply(HANDLE_SBR_DRC_CHANNEL hDrcData, FIXP_DBL **QmfBufferReal, FIXP_DBL **QmfBufferImag, int numQmfSubSamples, int *scaleFactor); #endif /* SBRDEC_DRC_H */ fdk-aac-2.0.2/libSBRdec/src/sbrdec_freq_sca.cpp000066400000000000000000000625161404225310700212020ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Frequency scale calculation */ #include "sbrdec_freq_sca.h" #include "transcendent.h" #include "sbr_rom.h" #include "env_extr.h" #include "genericStds.h" /* need log() for debug-code only */ #define MAX_OCTAVE 29 #define MAX_SECOND_REGION 50 static int numberOfBands(FIXP_SGL bpo_div16, int start, int stop, int warpFlag); static void CalcBands(UCHAR *diff, UCHAR start, UCHAR stop, UCHAR num_bands); static SBR_ERROR modifyBands(UCHAR max_band, UCHAR *diff, UCHAR length); static void cumSum(UCHAR start_value, UCHAR *diff, UCHAR length, UCHAR *start_adress); /*! \brief Retrieve QMF-band where the SBR range starts Convert startFreq which was read from the bitstream into a QMF-channel number. \return Number of start band */ static UCHAR getStartBand( UINT fs, /*!< Output sampling frequency */ UCHAR startFreq, /*!< Index to table of possible start bands */ UINT headerDataFlags) /*!< Info to SBR mode */ { INT band; UINT fsMapped = fs; SBR_RATE rate = DUAL; if (headerDataFlags & (SBRDEC_SYNTAX_USAC | SBRDEC_SYNTAX_RSVD50)) { if (headerDataFlags & SBRDEC_QUAD_RATE) { rate = QUAD; } fsMapped = sbrdec_mapToStdSampleRate(fs, 1); } FDK_ASSERT(2 * (rate + 1) <= (4)); switch (fsMapped) { case 192000: band = FDK_sbrDecoder_sbr_start_freq_192[startFreq]; break; case 176400: band = FDK_sbrDecoder_sbr_start_freq_176[startFreq]; break; case 128000: band = FDK_sbrDecoder_sbr_start_freq_128[startFreq]; break; case 96000: case 88200: band = FDK_sbrDecoder_sbr_start_freq_88[rate][startFreq]; break; case 64000: band = FDK_sbrDecoder_sbr_start_freq_64[rate][startFreq]; break; case 48000: band = FDK_sbrDecoder_sbr_start_freq_48[rate][startFreq]; break; case 44100: band = FDK_sbrDecoder_sbr_start_freq_44[rate][startFreq]; break; case 40000: band = FDK_sbrDecoder_sbr_start_freq_40[rate][startFreq]; break; case 32000: band = FDK_sbrDecoder_sbr_start_freq_32[rate][startFreq]; break; case 24000: band = FDK_sbrDecoder_sbr_start_freq_24[rate][startFreq]; break; case 22050: band = FDK_sbrDecoder_sbr_start_freq_22[rate][startFreq]; break; case 16000: band = FDK_sbrDecoder_sbr_start_freq_16[rate][startFreq]; break; default: band = 255; } return band; } /*! \brief Retrieve QMF-band where the SBR range starts Convert startFreq which was read from the bitstream into a QMF-channel number. \return Number of start band */ static UCHAR getStopBand( UINT fs, /*!< Output sampling frequency */ UCHAR stopFreq, /*!< Index to table of possible start bands */ UINT headerDataFlags, /*!< Info to SBR mode */ UCHAR k0) /*!< Start freq index */ { UCHAR k2; if (stopFreq < 14) { INT stopMin; INT num = 2 * (64); UCHAR diff_tot[MAX_OCTAVE + MAX_SECOND_REGION]; UCHAR *diff0 = diff_tot; UCHAR *diff1 = diff_tot + MAX_OCTAVE; if (headerDataFlags & SBRDEC_QUAD_RATE) { num >>= 1; } if (fs < 32000) { stopMin = (((2 * 6000 * num) / fs) + 1) >> 1; } else { if (fs < 64000) { stopMin = (((2 * 8000 * num) / fs) + 1) >> 1; } else { stopMin = (((2 * 10000 * num) / fs) + 1) >> 1; } } stopMin = fMin(stopMin, 64); /* Choose a stop band between k1 and 64 depending on stopFreq (0..13), based on a logarithmic scale. The vectors diff0 and diff1 are used temporarily here. */ CalcBands(diff0, stopMin, 64, 13); shellsort(diff0, 13); cumSum(stopMin, diff0, 13, diff1); k2 = diff1[stopFreq]; } else if (stopFreq == 14) k2 = 2 * k0; else k2 = 3 * k0; /* Limit to Nyquist */ if (k2 > (64)) k2 = (64); /* Range checks */ /* 1 <= difference <= 48; 1 <= fs <= 96000 */ { UCHAR max_freq_coeffs = (headerDataFlags & SBRDEC_QUAD_RATE) ? MAX_FREQ_COEFFS_QUAD_RATE : MAX_FREQ_COEFFS; if (((k2 - k0) > max_freq_coeffs) || (k2 <= k0)) { return 255; } } if (headerDataFlags & SBRDEC_QUAD_RATE) { return k2; /* skip other checks: (k2 - k0) must be <= MAX_FREQ_COEFFS_QUAD_RATE for all fs */ } if (headerDataFlags & (SBRDEC_SYNTAX_USAC | SBRDEC_SYNTAX_RSVD50)) { /* 1 <= difference <= 35; 42000 <= fs <= 96000 */ if ((fs >= 42000) && ((k2 - k0) > MAX_FREQ_COEFFS_FS44100)) { return 255; } /* 1 <= difference <= 32; 46009 <= fs <= 96000 */ if ((fs >= 46009) && ((k2 - k0) > MAX_FREQ_COEFFS_FS48000)) { return 255; } } else { /* 1 <= difference <= 35; fs == 44100 */ if ((fs == 44100) && ((k2 - k0) > MAX_FREQ_COEFFS_FS44100)) { return 255; } /* 1 <= difference <= 32; 48000 <= fs <= 96000 */ if ((fs >= 48000) && ((k2 - k0) > MAX_FREQ_COEFFS_FS48000)) { return 255; } } return k2; } /*! \brief Generates master frequency tables Frequency tables are calculated according to the selected domain (linear/logarithmic) and granularity. IEC 14496-3 4.6.18.3.2.1 \return errorCode, 0 if successful */ SBR_ERROR sbrdecUpdateFreqScale( UCHAR *v_k_master, /*!< Master table to be created */ UCHAR *numMaster, /*!< Number of entries in master table */ UINT fs, /*!< SBR working sampling rate */ HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Control data from bitstream */ UINT flags) { FIXP_SGL bpo_div16; /* bands_per_octave divided by 16 */ INT dk = 0; /* Internal variables */ UCHAR k0, k2, i; UCHAR num_bands0 = 0; UCHAR num_bands1 = 0; UCHAR diff_tot[MAX_OCTAVE + MAX_SECOND_REGION]; UCHAR *diff0 = diff_tot; UCHAR *diff1 = diff_tot + MAX_OCTAVE; INT k2_achived; INT k2_diff; INT incr = 0; /* Determine start band */ if (flags & SBRDEC_QUAD_RATE) { fs >>= 1; } k0 = getStartBand(fs, hHeaderData->bs_data.startFreq, flags); if (k0 == 255) { return SBRDEC_UNSUPPORTED_CONFIG; } /* Determine stop band */ k2 = getStopBand(fs, hHeaderData->bs_data.stopFreq, flags, k0); if (k2 == 255) { return SBRDEC_UNSUPPORTED_CONFIG; } if (hHeaderData->bs_data.freqScale > 0) { /* Bark */ INT k1; if (hHeaderData->bs_data.freqScale == 1) { bpo_div16 = FL2FXCONST_SGL(12.0f / 16.0f); } else if (hHeaderData->bs_data.freqScale == 2) { bpo_div16 = FL2FXCONST_SGL(10.0f / 16.0f); } else { bpo_div16 = FL2FXCONST_SGL(8.0f / 16.0f); } /* Ref: ISO/IEC 23003-3, Figure 12 - Flowchart calculation of fMaster for * 4:1 system when bs_freq_scale > 0 */ if (flags & SBRDEC_QUAD_RATE) { if ((SHORT)k0 < (SHORT)(bpo_div16 >> ((FRACT_BITS - 1) - 4))) { bpo_div16 = (FIXP_SGL)(k0 & (UCHAR)0xfe) << ((FRACT_BITS - 1) - 4); /* bpo_div16 = floor(k0/2)*2 */ } } if (1000 * k2 > 2245 * k0) { /* Two or more regions */ k1 = 2 * k0; num_bands0 = numberOfBands(bpo_div16, k0, k1, 0); num_bands1 = numberOfBands(bpo_div16, k1, k2, hHeaderData->bs_data.alterScale); if (num_bands0 < 1) { return SBRDEC_UNSUPPORTED_CONFIG; } if (num_bands1 < 1) { return SBRDEC_UNSUPPORTED_CONFIG; } CalcBands(diff0, k0, k1, num_bands0); shellsort(diff0, num_bands0); if (diff0[0] == 0) { return SBRDEC_UNSUPPORTED_CONFIG; } cumSum(k0, diff0, num_bands0, v_k_master); CalcBands(diff1, k1, k2, num_bands1); shellsort(diff1, num_bands1); if (diff0[num_bands0 - 1] > diff1[0]) { SBR_ERROR err; err = modifyBands(diff0[num_bands0 - 1], diff1, num_bands1); if (err) return SBRDEC_UNSUPPORTED_CONFIG; } /* Add 2nd region */ cumSum(k1, diff1, num_bands1, &v_k_master[num_bands0]); *numMaster = num_bands0 + num_bands1; /* Output nr of bands */ } else { /* Only one region */ k1 = k2; num_bands0 = numberOfBands(bpo_div16, k0, k1, 0); if (num_bands0 < 1) { return SBRDEC_UNSUPPORTED_CONFIG; } CalcBands(diff0, k0, k1, num_bands0); shellsort(diff0, num_bands0); if (diff0[0] == 0) { return SBRDEC_UNSUPPORTED_CONFIG; } cumSum(k0, diff0, num_bands0, v_k_master); *numMaster = num_bands0; /* Output nr of bands */ } } else { /* Linear mode */ if (hHeaderData->bs_data.alterScale == 0) { dk = 1; /* FLOOR to get to few number of bands (next lower even number) */ num_bands0 = (k2 - k0) & 254; } else { dk = 2; num_bands0 = (((k2 - k0) >> 1) + 1) & 254; /* ROUND to the closest fit */ } if (num_bands0 < 1) { return SBRDEC_UNSUPPORTED_CONFIG; /* We must return already here because 'i' can become negative below. */ } k2_achived = k0 + num_bands0 * dk; k2_diff = k2 - k2_achived; for (i = 0; i < num_bands0; i++) diff_tot[i] = dk; /* If linear scale wasn't achieved */ /* and we got too wide SBR area */ if (k2_diff < 0) { incr = 1; i = 0; } /* If linear scale wasn't achieved */ /* and we got too small SBR area */ if (k2_diff > 0) { incr = -1; i = num_bands0 - 1; } /* Adjust diff vector to get sepc. SBR range */ while (k2_diff != 0) { diff_tot[i] = diff_tot[i] - incr; i = i + incr; k2_diff = k2_diff + incr; } cumSum(k0, diff_tot, num_bands0, v_k_master); /* cumsum */ *numMaster = num_bands0; /* Output nr of bands */ } if (*numMaster < 1) { return SBRDEC_UNSUPPORTED_CONFIG; } /* Ref: ISO/IEC 23003-3 Cor.3, "In 7.5.5.2, add to the requirements:"*/ if (flags & SBRDEC_QUAD_RATE) { int k; for (k = 1; k < *numMaster; k++) { if (!(v_k_master[k] - v_k_master[k - 1] <= k0 - 2)) { return SBRDEC_UNSUPPORTED_CONFIG; } } } /* Print out the calculated table */ return SBRDEC_OK; } /*! \brief Calculate frequency ratio of one SBR band All SBR bands should span a constant frequency range in the logarithmic domain. This function calculates the ratio of any SBR band's upper and lower frequency. \return num_band-th root of k_start/k_stop */ static FIXP_SGL calcFactorPerBand(int k_start, int k_stop, int num_bands) { /* Scaled bandfactor and step 1 bit right to avoid overflow * use double data type */ FIXP_DBL bandfactor = FL2FXCONST_DBL(0.25f); /* Start value */ FIXP_DBL step = FL2FXCONST_DBL(0.125f); /* Initial increment for factor */ int direction = 1; /* Because saturation can't be done in INT IIS, * changed start and stop data type from FIXP_SGL to FIXP_DBL */ FIXP_DBL start = k_start << (DFRACT_BITS - 8); FIXP_DBL stop = k_stop << (DFRACT_BITS - 8); FIXP_DBL temp; int j, i = 0; while (step > FL2FXCONST_DBL(0.0f)) { i++; temp = stop; /* Calculate temp^num_bands: */ for (j = 0; j < num_bands; j++) // temp = fMult(temp,bandfactor); temp = fMultDiv2(temp, bandfactor) << 2; if (temp < start) { /* Factor too strong, make it weaker */ if (direction == 0) /* Halfen step. Right shift is not done as fract because otherwise the lowest bit cannot be cleared due to rounding */ step = (FIXP_DBL)((LONG)step >> 1); direction = 1; bandfactor = bandfactor + step; } else { /* Factor is too weak: make it stronger */ if (direction == 1) step = (FIXP_DBL)((LONG)step >> 1); direction = 0; bandfactor = bandfactor - step; } if (i > 100) { step = FL2FXCONST_DBL(0.0f); } } return (bandfactor >= FL2FXCONST_DBL(0.5)) ? (FIXP_SGL)MAXVAL_SGL : FX_DBL2FX_SGL(bandfactor << 1); } /*! \brief Calculate number of SBR bands between start and stop band Given the number of bands per octave, this function calculates how many bands fit in the given frequency range. When the warpFlag is set, the 'band density' is decreased by a factor of 1/1.3 \return number of bands */ static int numberOfBands( FIXP_SGL bpo_div16, /*!< Input: number of bands per octave divided by 16 */ int start, /*!< First QMF band of SBR frequency range */ int stop, /*!< Last QMF band of SBR frequency range + 1 */ int warpFlag) /*!< Stretching flag */ { FIXP_SGL num_bands_div128; int num_bands; num_bands_div128 = FX_DBL2FX_SGL(fMult(FDK_getNumOctavesDiv8(start, stop), bpo_div16)); if (warpFlag) { /* Apply the warp factor of 1.3 to get wider bands. We use a value of 32768/25200 instead of the exact value to avoid critical cases of rounding. */ num_bands_div128 = FX_DBL2FX_SGL( fMult(num_bands_div128, FL2FXCONST_SGL(25200.0 / 32768.0))); } /* add scaled 1 for rounding to even numbers: */ num_bands_div128 = num_bands_div128 + FL2FXCONST_SGL(1.0f / 128.0f); /* scale back to right aligned integer and double the value: */ num_bands = 2 * ((LONG)num_bands_div128 >> (FRACT_BITS - 7)); return (num_bands); } /*! \brief Calculate width of SBR bands Given the desired number of bands within the SBR frequency range, this function calculates the width of each SBR band in QMF channels. The bands get wider from start to stop (bark scale). */ static void CalcBands(UCHAR *diff, /*!< Vector of widths to be calculated */ UCHAR start, /*!< Lower end of subband range */ UCHAR stop, /*!< Upper end of subband range */ UCHAR num_bands) /*!< Desired number of bands */ { int i; int previous; int current; FIXP_SGL exact, temp; FIXP_SGL bandfactor = calcFactorPerBand(start, stop, num_bands); previous = stop; /* Start with highest QMF channel */ exact = (FIXP_SGL)( stop << (FRACT_BITS - 8)); /* Shift left to gain some accuracy */ for (i = num_bands - 1; i >= 0; i--) { /* Calculate border of next lower sbr band */ exact = FX_DBL2FX_SGL(fMult(exact, bandfactor)); /* Add scaled 0.5 for rounding: We use a value 128/256 instead of 0.5 to avoid some critical cases of rounding. */ temp = exact + FL2FXCONST_SGL(128.0 / 32768.0); /* scale back to right alinged integer: */ current = (LONG)temp >> (FRACT_BITS - 8); /* Save width of band i */ diff[i] = previous - current; previous = current; } } /*! \brief Calculate cumulated sum vector from delta vector */ static void cumSum(UCHAR start_value, UCHAR *diff, UCHAR length, UCHAR *start_adress) { int i; start_adress[0] = start_value; for (i = 1; i <= length; i++) start_adress[i] = start_adress[i - 1] + diff[i - 1]; } /*! \brief Adapt width of frequency bands in the second region If SBR spans more than 2 octaves, the upper part of a bark-frequency-scale is calculated separately. This function tries to avoid that the second region starts with a band smaller than the highest band of the first region. */ static SBR_ERROR modifyBands(UCHAR max_band_previous, UCHAR *diff, UCHAR length) { int change = max_band_previous - diff[0]; /* Limit the change so that the last band cannot get narrower than the first * one */ if (change > (diff[length - 1] - diff[0]) >> 1) change = (diff[length - 1] - diff[0]) >> 1; diff[0] += change; diff[length - 1] -= change; shellsort(diff, length); return SBRDEC_OK; } /*! \brief Update high resolution frequency band table */ static void sbrdecUpdateHiRes(UCHAR *h_hires, UCHAR *num_hires, UCHAR *v_k_master, UCHAR num_bands, UCHAR xover_band) { UCHAR i; *num_hires = num_bands - xover_band; for (i = xover_band; i <= num_bands; i++) { h_hires[i - xover_band] = v_k_master[i]; } } /*! \brief Build low resolution table out of high resolution table */ static void sbrdecUpdateLoRes(UCHAR *h_lores, UCHAR *num_lores, UCHAR *h_hires, UCHAR num_hires) { UCHAR i; if ((num_hires & 1) == 0) { /* If even number of hires bands */ *num_lores = num_hires >> 1; /* Use every second lores=hires[0,2,4...] */ for (i = 0; i <= *num_lores; i++) h_lores[i] = h_hires[i * 2]; } else { /* Odd number of hires, which means xover is odd */ *num_lores = (num_hires + 1) >> 1; /* Use lores=hires[0,1,3,5 ...] */ h_lores[0] = h_hires[0]; for (i = 1; i <= *num_lores; i++) { h_lores[i] = h_hires[i * 2 - 1]; } } } /*! \brief Derive a low-resolution frequency-table from the master frequency table */ void sbrdecDownSampleLoRes(UCHAR *v_result, UCHAR num_result, UCHAR *freqBandTableRef, UCHAR num_Ref) { int step; int i, j; int org_length, result_length; int v_index[MAX_FREQ_COEFFS >> 1]; /* init */ org_length = num_Ref; result_length = num_result; v_index[0] = 0; /* Always use left border */ i = 0; while (org_length > 0) { /* Create downsample vector */ i++; step = org_length / result_length; org_length = org_length - step; result_length--; v_index[i] = v_index[i - 1] + step; } for (j = 0; j <= i; j++) { /* Use downsample vector to index LoResolution vector */ v_result[j] = freqBandTableRef[v_index[j]]; } } /*! \brief Sorting routine */ void shellsort(UCHAR *in, UCHAR n) { int i, j, v, w; int inc = 1; do inc = 3 * inc + 1; while (inc <= n); do { inc = inc / 3; for (i = inc; i < n; i++) { v = in[i]; j = i; while ((w = in[j - inc]) > v) { in[j] = w; j -= inc; if (j < inc) break; } in[j] = v; } } while (inc > 1); } /*! \brief Reset frequency band tables \return errorCode, 0 if successful */ SBR_ERROR resetFreqBandTables(HANDLE_SBR_HEADER_DATA hHeaderData, const UINT flags) { SBR_ERROR err = SBRDEC_OK; int k2, kx, lsb, usb; int intTemp; UCHAR nBandsLo, nBandsHi; HANDLE_FREQ_BAND_DATA hFreq = &hHeaderData->freqBandData; /* Calculate master frequency function */ err = sbrdecUpdateFreqScale(hFreq->v_k_master, &hFreq->numMaster, hHeaderData->sbrProcSmplRate, hHeaderData, flags); if (err || (hHeaderData->bs_info.xover_band > hFreq->numMaster)) { return SBRDEC_UNSUPPORTED_CONFIG; } /* Derive Hiresolution from master frequency function */ sbrdecUpdateHiRes(hFreq->freqBandTable[1], &nBandsHi, hFreq->v_k_master, hFreq->numMaster, hHeaderData->bs_info.xover_band); /* Derive Loresolution from Hiresolution */ sbrdecUpdateLoRes(hFreq->freqBandTable[0], &nBandsLo, hFreq->freqBandTable[1], nBandsHi); hFreq->nSfb[0] = nBandsLo; hFreq->nSfb[1] = nBandsHi; /* Check index to freqBandTable[0] */ if (!(nBandsLo > 0) || (nBandsLo > (((hHeaderData->numberOfAnalysisBands == 16) ? MAX_FREQ_COEFFS_QUAD_RATE : MAX_FREQ_COEFFS_DUAL_RATE) >> 1))) { return SBRDEC_UNSUPPORTED_CONFIG; } lsb = hFreq->freqBandTable[0][0]; usb = hFreq->freqBandTable[0][nBandsLo]; /* Check for start frequency border k_x: - ISO/IEC 14496-3 4.6.18.3.6 Requirements - ISO/IEC 23003-3 7.5.5.2 Modifications and additions to the MPEG-4 SBR tool */ /* Note that lsb > as hHeaderData->numberOfAnalysisBands is a valid SBR config * for 24 band QMF analysis. */ if ((lsb > ((flags & SBRDEC_QUAD_RATE) ? 16 : (32))) || (lsb >= usb)) { return SBRDEC_UNSUPPORTED_CONFIG; } /* Calculate number of noise bands */ k2 = hFreq->freqBandTable[1][nBandsHi]; kx = hFreq->freqBandTable[1][0]; if (hHeaderData->bs_data.noise_bands == 0) { hFreq->nNfb = 1; } else /* Calculate no of noise bands 1,2 or 3 bands/octave */ { /* Fetch number of octaves divided by 32 */ intTemp = (LONG)FDK_getNumOctavesDiv8(kx, k2) >> 2; /* Integer-Multiplication with number of bands: */ intTemp = intTemp * hHeaderData->bs_data.noise_bands; /* Add scaled 0.5 for rounding: */ intTemp = intTemp + (LONG)FL2FXCONST_SGL(0.5f / 32.0f); /* Convert to right-aligned integer: */ intTemp = intTemp >> (FRACT_BITS - 1 /*sign*/ - 5 /* rescale */); if (intTemp == 0) intTemp = 1; hFreq->nNfb = intTemp; } hFreq->nInvfBands = hFreq->nNfb; if (hFreq->nNfb > MAX_NOISE_COEFFS) { return SBRDEC_UNSUPPORTED_CONFIG; } /* Get noise bands */ sbrdecDownSampleLoRes(hFreq->freqBandTableNoise, hFreq->nNfb, hFreq->freqBandTable[0], nBandsLo); /* save old highband; required for overlap in usac when headerchange occurs at XVAR and VARX frame; */ hFreq->ov_highSubband = hFreq->highSubband; hFreq->lowSubband = lsb; hFreq->highSubband = usb; return SBRDEC_OK; } fdk-aac-2.0.2/libSBRdec/src/sbrdec_freq_sca.h000066400000000000000000000124371404225310700206440ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Frequency scale prototypes */ #ifndef SBRDEC_FREQ_SCA_H #define SBRDEC_FREQ_SCA_H #include "sbrdecoder.h" #include "env_extr.h" typedef enum { DUAL, QUAD } SBR_RATE; SBR_ERROR sbrdecUpdateFreqScale(UCHAR *v_k_master, UCHAR *numMaster, UINT fs, HANDLE_SBR_HEADER_DATA headerData, UINT flags); void sbrdecDownSampleLoRes(UCHAR *v_result, UCHAR num_result, UCHAR *freqBandTableRef, UCHAR num_Ref); void shellsort(UCHAR *in, UCHAR n); SBR_ERROR resetFreqBandTables(HANDLE_SBR_HEADER_DATA hHeaderData, const UINT flags); #endif fdk-aac-2.0.2/libSBRdec/src/sbrdecoder.cpp000066400000000000000000002002031404225310700201740ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief SBR decoder frontend This module provides a frontend to the SBR decoder. The function openSBR() is called for initialization. The function sbrDecoder_Apply() is called for each frame. sbr_Apply() will call the required functions to decode the raw SBR data (provided by env_extr.cpp), to decode the envelope data and noise floor levels [decodeSbrData()], and to finally apply SBR to the current frame [sbr_dec()]. \sa sbrDecoder_Apply(), \ref documentationOverview */ /*! \page documentationOverview Overview of important information resources and source code documentation As part of this documentation you can find more extensive descriptions about key concepts and algorithms at the following locations:

Programming

\li Buffer management: sbrDecoder_Apply() and sbr_dec() \li Internal scale factors to maximize SNR on fixed point processors: #QMF_SCALE_FACTOR \li Special mantissa-exponent format: Created in requantizeEnvelopeData() and used in calculateSbrEnvelope()

Algorithmic details

\li About the SBR data format: \ref SBR_HEADER_ELEMENT and \ref SBR_STANDARD_ELEMENT \li Details about the bitstream decoder: env_extr.cpp \li Details about the QMF filterbank and the provided polyphase implementation: qmf_dec.cpp \li Details about the transposer: lpp_tran.cpp \li Details about the envelope adjuster: env_calc.cpp */ #include "sbrdecoder.h" #include "FDK_bitstream.h" #include "sbrdec_freq_sca.h" #include "env_extr.h" #include "sbr_dec.h" #include "env_dec.h" #include "FDK_crc.h" #include "sbr_ram.h" #include "sbr_rom.h" #include "lpp_tran.h" #include "transcendent.h" #include "sbrdec_drc.h" #include "psbitdec.h" /* Decoder library info */ #define SBRDECODER_LIB_VL0 3 #define SBRDECODER_LIB_VL1 1 #define SBRDECODER_LIB_VL2 0 #define SBRDECODER_LIB_TITLE "SBR Decoder" #ifdef SUPPRESS_BUILD_DATE_INFO #define SBRDECODER_LIB_BUILD_DATE "" #define SBRDECODER_LIB_BUILD_TIME "" #else #define SBRDECODER_LIB_BUILD_DATE __DATE__ #define SBRDECODER_LIB_BUILD_TIME __TIME__ #endif static void setFrameErrorFlag(SBR_DECODER_ELEMENT *pSbrElement, UCHAR value) { if (pSbrElement != NULL) { switch (value) { case FRAME_ERROR_ALLSLOTS: FDKmemset(pSbrElement->frameErrorFlag, FRAME_ERROR, sizeof(pSbrElement->frameErrorFlag)); break; default: pSbrElement->frameErrorFlag[pSbrElement->useFrameSlot] = value; } } } static UCHAR getHeaderSlot(UCHAR currentSlot, UCHAR hdrSlotUsage[(1) + 1]) { UINT occupied = 0; int s; UCHAR slot = hdrSlotUsage[currentSlot]; FDK_ASSERT((1) + 1 < 32); for (s = 0; s < (1) + 1; s++) { if ((hdrSlotUsage[s] == slot) && (s != slot)) { occupied = 1; break; } } if (occupied) { occupied = 0; for (s = 0; s < (1) + 1; s++) { occupied |= 1 << hdrSlotUsage[s]; } for (s = 0; s < (1) + 1; s++) { if (!(occupied & 0x1)) { slot = s; break; } occupied >>= 1; } } return slot; } static void copySbrHeader(HANDLE_SBR_HEADER_DATA hDst, const HANDLE_SBR_HEADER_DATA hSrc) { /* copy the whole header memory (including pointers) */ FDKmemcpy(hDst, hSrc, sizeof(SBR_HEADER_DATA)); /* update pointers */ hDst->freqBandData.freqBandTable[0] = hDst->freqBandData.freqBandTableLo; hDst->freqBandData.freqBandTable[1] = hDst->freqBandData.freqBandTableHi; } static int compareSbrHeader(const HANDLE_SBR_HEADER_DATA hHdr1, const HANDLE_SBR_HEADER_DATA hHdr2) { int result = 0; /* compare basic data */ result |= (hHdr1->syncState != hHdr2->syncState) ? 1 : 0; result |= (hHdr1->status != hHdr2->status) ? 1 : 0; result |= (hHdr1->frameErrorFlag != hHdr2->frameErrorFlag) ? 1 : 0; result |= (hHdr1->numberTimeSlots != hHdr2->numberTimeSlots) ? 1 : 0; result |= (hHdr1->numberOfAnalysisBands != hHdr2->numberOfAnalysisBands) ? 1 : 0; result |= (hHdr1->timeStep != hHdr2->timeStep) ? 1 : 0; result |= (hHdr1->sbrProcSmplRate != hHdr2->sbrProcSmplRate) ? 1 : 0; /* compare bitstream data */ result |= FDKmemcmp(&hHdr1->bs_data, &hHdr2->bs_data, sizeof(SBR_HEADER_DATA_BS)); result |= FDKmemcmp(&hHdr1->bs_dflt, &hHdr2->bs_dflt, sizeof(SBR_HEADER_DATA_BS)); result |= FDKmemcmp(&hHdr1->bs_info, &hHdr2->bs_info, sizeof(SBR_HEADER_DATA_BS_INFO)); /* compare frequency band data */ result |= FDKmemcmp(&hHdr1->freqBandData, &hHdr2->freqBandData, (8 + MAX_NUM_LIMITERS + 1) * sizeof(UCHAR)); result |= FDKmemcmp(hHdr1->freqBandData.freqBandTableLo, hHdr2->freqBandData.freqBandTableLo, (MAX_FREQ_COEFFS / 2 + 1) * sizeof(UCHAR)); result |= FDKmemcmp(hHdr1->freqBandData.freqBandTableHi, hHdr2->freqBandData.freqBandTableHi, (MAX_FREQ_COEFFS + 1) * sizeof(UCHAR)); result |= FDKmemcmp(hHdr1->freqBandData.freqBandTableNoise, hHdr2->freqBandData.freqBandTableNoise, (MAX_NOISE_COEFFS + 1) * sizeof(UCHAR)); result |= FDKmemcmp(hHdr1->freqBandData.v_k_master, hHdr2->freqBandData.v_k_master, (MAX_FREQ_COEFFS + 1) * sizeof(UCHAR)); return result; } /*! \brief Reset SBR decoder. Reset should only be called if SBR has been sucessfully detected by an appropriate checkForPayload() function. \return Error code. */ static SBR_ERROR sbrDecoder_ResetElement(HANDLE_SBRDECODER self, int sampleRateIn, int sampleRateOut, int samplesPerFrame, const MP4_ELEMENT_ID elementID, const int elementIndex, const int overlap) { SBR_ERROR sbrError = SBRDEC_OK; HANDLE_SBR_HEADER_DATA hSbrHeader; UINT qmfFlags = 0; int i, synDownsampleFac; /* USAC: assuming theoretical case 8 kHz output sample rate with 4:1 SBR */ const int sbr_min_sample_rate_in = IS_USAC(self->coreCodec) ? 2000 : 6400; /* Check in/out samplerates */ if (sampleRateIn < sbr_min_sample_rate_in || sampleRateIn > (96000)) { sbrError = SBRDEC_UNSUPPORTED_CONFIG; goto bail; } if (sampleRateOut > (96000)) { sbrError = SBRDEC_UNSUPPORTED_CONFIG; goto bail; } /* Set QMF mode flags */ if (self->flags & SBRDEC_LOW_POWER) qmfFlags |= QMF_FLAG_LP; if (self->coreCodec == AOT_ER_AAC_ELD) { if (self->flags & SBRDEC_LD_MPS_QMF) { qmfFlags |= QMF_FLAG_MPSLDFB; } else { qmfFlags |= QMF_FLAG_CLDFB; } } /* Set downsampling factor for synthesis filter bank */ if (sampleRateOut == 0) { /* no single rate mode */ sampleRateOut = sampleRateIn << 1; /* In case of implicit signalling, assume dual rate SBR */ } if (sampleRateIn == sampleRateOut) { synDownsampleFac = 2; self->flags |= SBRDEC_DOWNSAMPLE; } else { synDownsampleFac = 1; self->flags &= ~SBRDEC_DOWNSAMPLE; } self->synDownsampleFac = synDownsampleFac; self->sampleRateOut = sampleRateOut; { for (i = 0; i < (1) + 1; i++) { int setDflt; hSbrHeader = &(self->sbrHeader[elementIndex][i]); setDflt = ((hSbrHeader->syncState == SBR_NOT_INITIALIZED) || (self->flags & SBRDEC_FORCE_RESET)) ? 1 : 0; /* init a default header such that we can at least do upsampling later */ sbrError = initHeaderData(hSbrHeader, sampleRateIn, sampleRateOut, self->downscaleFactor, samplesPerFrame, self->flags, setDflt); /* Set synchState to UPSAMPLING in case it already is initialized */ hSbrHeader->syncState = hSbrHeader->syncState > UPSAMPLING ? UPSAMPLING : hSbrHeader->syncState; } } if (sbrError != SBRDEC_OK) { goto bail; } if (!self->pQmfDomain->globalConf.qmfDomainExplicitConfig) { self->pQmfDomain->globalConf.flags_requested |= qmfFlags; self->pQmfDomain->globalConf.nBandsAnalysis_requested = self->sbrHeader[elementIndex][0].numberOfAnalysisBands; self->pQmfDomain->globalConf.nBandsSynthesis_requested = (synDownsampleFac == 1) ? 64 : 32; /* may be overwritten by MPS */ self->pQmfDomain->globalConf.nBandsSynthesis_requested /= self->downscaleFactor; self->pQmfDomain->globalConf.nQmfTimeSlots_requested = self->sbrHeader[elementIndex][0].numberTimeSlots * self->sbrHeader[elementIndex][0].timeStep; self->pQmfDomain->globalConf.nQmfOvTimeSlots_requested = overlap; self->pQmfDomain->globalConf.nQmfProcBands_requested = 64; /* always 64 */ self->pQmfDomain->globalConf.nQmfProcChannels_requested = 1; /* may be overwritten by MPS */ } /* Init SBR channels going to be assigned to a SBR element */ { int ch; for (ch = 0; ch < self->pSbrElement[elementIndex]->nChannels; ch++) { int headerIndex = getHeaderSlot(self->pSbrElement[elementIndex]->useFrameSlot, self->pSbrElement[elementIndex]->useHeaderSlot); /* and create sbrDec */ sbrError = createSbrDec(self->pSbrElement[elementIndex]->pSbrChannel[ch], &self->sbrHeader[elementIndex][headerIndex], &self->pSbrElement[elementIndex]->transposerSettings, synDownsampleFac, qmfFlags, self->flags, overlap, ch, self->codecFrameSize); if (sbrError != SBRDEC_OK) { goto bail; } } } // FDKmemclear(sbr_OverlapBuffer, sizeof(sbr_OverlapBuffer)); if (self->numSbrElements == 1) { switch (self->coreCodec) { case AOT_AAC_LC: case AOT_SBR: case AOT_PS: case AOT_ER_AAC_SCAL: case AOT_DRM_AAC: case AOT_DRM_SURROUND: if (CreatePsDec(&self->hParametricStereoDec, samplesPerFrame)) { sbrError = SBRDEC_CREATE_ERROR; goto bail; } break; default: break; } } /* Init frame delay slot handling */ self->pSbrElement[elementIndex]->useFrameSlot = 0; for (i = 0; i < ((1) + 1); i++) { self->pSbrElement[elementIndex]->useHeaderSlot[i] = i; } bail: return sbrError; } /*! \brief Assign QMF domain provided QMF channels to SBR channels. \return void */ static void sbrDecoder_AssignQmfChannels2SbrChannels(HANDLE_SBRDECODER self) { int ch, el, absCh_offset = 0; for (el = 0; el < self->numSbrElements; el++) { if (self->pSbrElement[el] != NULL) { for (ch = 0; ch < self->pSbrElement[el]->nChannels; ch++) { FDK_ASSERT(((absCh_offset + ch) < ((8) + (1))) && ((absCh_offset + ch) < ((8) + (1)))); self->pSbrElement[el]->pSbrChannel[ch]->SbrDec.qmfDomainInCh = &self->pQmfDomain->QmfDomainIn[absCh_offset + ch]; self->pSbrElement[el]->pSbrChannel[ch]->SbrDec.qmfDomainOutCh = &self->pQmfDomain->QmfDomainOut[absCh_offset + ch]; } absCh_offset += self->pSbrElement[el]->nChannels; } } } SBR_ERROR sbrDecoder_Open(HANDLE_SBRDECODER *pSelf, HANDLE_FDK_QMF_DOMAIN pQmfDomain) { HANDLE_SBRDECODER self = NULL; SBR_ERROR sbrError = SBRDEC_OK; int elIdx; if ((pSelf == NULL) || (pQmfDomain == NULL)) { return SBRDEC_INVALID_ARGUMENT; } /* Get memory for this instance */ self = GetRam_SbrDecoder(); if (self == NULL) { sbrError = SBRDEC_MEM_ALLOC_FAILED; goto bail; } self->pQmfDomain = pQmfDomain; /* Already zero because of calloc self->numSbrElements = 0; self->numSbrChannels = 0; self->codecFrameSize = 0; */ self->numDelayFrames = (1); /* set to the max value by default */ /* Initialize header sync state */ for (elIdx = 0; elIdx < (8); elIdx += 1) { int i; for (i = 0; i < (1) + 1; i += 1) { self->sbrHeader[elIdx][i].syncState = SBR_NOT_INITIALIZED; } } *pSelf = self; bail: return sbrError; } /** * \brief determine if the given core codec AOT can be processed or not. * \param coreCodec core codec audio object type. * \return 1 if SBR can be processed, 0 if SBR cannot be processed/applied. */ static int sbrDecoder_isCoreCodecValid(AUDIO_OBJECT_TYPE coreCodec) { switch (coreCodec) { case AOT_AAC_LC: case AOT_SBR: case AOT_PS: case AOT_ER_AAC_SCAL: case AOT_ER_AAC_ELD: case AOT_DRM_AAC: case AOT_DRM_SURROUND: case AOT_USAC: return 1; default: return 0; } } static void sbrDecoder_DestroyElement(HANDLE_SBRDECODER self, const int elementIndex) { if (self->pSbrElement[elementIndex] != NULL) { int ch; for (ch = 0; ch < SBRDEC_MAX_CH_PER_ELEMENT; ch++) { if (self->pSbrElement[elementIndex]->pSbrChannel[ch] != NULL) { deleteSbrDec(self->pSbrElement[elementIndex]->pSbrChannel[ch]); FreeRam_SbrDecChannel( &self->pSbrElement[elementIndex]->pSbrChannel[ch]); self->numSbrChannels -= 1; } } FreeRam_SbrDecElement(&self->pSbrElement[elementIndex]); self->numSbrElements -= 1; } } SBR_ERROR sbrDecoder_InitElement( HANDLE_SBRDECODER self, const int sampleRateIn, const int sampleRateOut, const int samplesPerFrame, const AUDIO_OBJECT_TYPE coreCodec, const MP4_ELEMENT_ID elementID, const int elementIndex, const UCHAR harmonicSBR, const UCHAR stereoConfigIndex, const UCHAR configMode, UCHAR *configChanged, const INT downscaleFactor) { SBR_ERROR sbrError = SBRDEC_OK; int chCnt = 0; int nSbrElementsStart; int nSbrChannelsStart; if (self == NULL) { return SBRDEC_INVALID_ARGUMENT; } nSbrElementsStart = self->numSbrElements; nSbrChannelsStart = self->numSbrChannels; /* Check core codec AOT */ if (!sbrDecoder_isCoreCodecValid(coreCodec) || elementIndex >= (8)) { sbrError = SBRDEC_UNSUPPORTED_CONFIG; goto bail; } if (elementID != ID_SCE && elementID != ID_CPE && elementID != ID_LFE) { sbrError = SBRDEC_UNSUPPORTED_CONFIG; goto bail; } if (self->sampleRateIn == sampleRateIn && self->codecFrameSize == samplesPerFrame && self->coreCodec == coreCodec && self->pSbrElement[elementIndex] != NULL && self->pSbrElement[elementIndex]->elementID == elementID && !(self->flags & SBRDEC_FORCE_RESET) && ((sampleRateOut == 0) ? 1 : (self->sampleRateOut == sampleRateOut)) && ((harmonicSBR == 2) ? 1 : (self->harmonicSBR == harmonicSBR)) /* The value 2 signalizes that harmonicSBR shall be ignored in the config change detection */ ) { /* Nothing to do */ return SBRDEC_OK; } else { if (configMode & AC_CM_DET_CFG_CHANGE) { *configChanged = 1; } } /* reaching this point the SBR-decoder gets (re-)configured */ /* The flags field is used for all elements! */ self->flags &= (SBRDEC_FORCE_RESET | SBRDEC_FLUSH); /* Keep the global flags. They will be reset after decoding. */ self->flags |= (downscaleFactor > 1) ? SBRDEC_ELD_DOWNSCALE : 0; self->flags |= (coreCodec == AOT_ER_AAC_ELD) ? SBRDEC_ELD_GRID : 0; self->flags |= (coreCodec == AOT_ER_AAC_SCAL) ? SBRDEC_SYNTAX_SCAL : 0; self->flags |= (coreCodec == AOT_DRM_AAC) ? SBRDEC_SYNTAX_SCAL | SBRDEC_SYNTAX_DRM : 0; self->flags |= (coreCodec == AOT_DRM_SURROUND) ? SBRDEC_SYNTAX_SCAL | SBRDEC_SYNTAX_DRM : 0; self->flags |= (coreCodec == AOT_USAC) ? SBRDEC_SYNTAX_USAC : 0; /* Robustness: Take integer division rounding into consideration. E.g. 22050 * Hz with 4:1 SBR => 5512 Hz core sampling rate. */ self->flags |= (sampleRateIn == sampleRateOut / 4) ? SBRDEC_QUAD_RATE : 0; self->flags |= (harmonicSBR == 1) ? SBRDEC_USAC_HARMONICSBR : 0; if (configMode & AC_CM_DET_CFG_CHANGE) { return SBRDEC_OK; } self->sampleRateIn = sampleRateIn; self->codecFrameSize = samplesPerFrame; self->coreCodec = coreCodec; self->harmonicSBR = harmonicSBR; self->downscaleFactor = downscaleFactor; /* Init SBR elements */ { int elChannels, ch; if (self->pSbrElement[elementIndex] == NULL) { self->pSbrElement[elementIndex] = GetRam_SbrDecElement(elementIndex); if (self->pSbrElement[elementIndex] == NULL) { sbrError = SBRDEC_MEM_ALLOC_FAILED; goto bail; } self->numSbrElements++; } else { self->numSbrChannels -= self->pSbrElement[elementIndex]->nChannels; } /* Determine amount of channels for this element */ switch (elementID) { case ID_NONE: case ID_CPE: elChannels = 2; break; case ID_LFE: case ID_SCE: elChannels = 1; break; default: elChannels = 0; break; } /* Handle case of Parametric Stereo */ if (elementIndex == 0 && elementID == ID_SCE) { switch (coreCodec) { case AOT_AAC_LC: case AOT_SBR: case AOT_PS: case AOT_ER_AAC_SCAL: case AOT_DRM_AAC: case AOT_DRM_SURROUND: elChannels = 2; break; default: break; } } /* Sanity check to avoid memory leaks */ if (elChannels < self->pSbrElement[elementIndex]->nChannels || (self->numSbrChannels + elChannels) > (8) + (1)) { self->numSbrChannels += self->pSbrElement[elementIndex]->nChannels; sbrError = SBRDEC_PARSE_ERROR; goto bail; } /* Save element ID for sanity checks and to have a fallback for concealment. */ self->pSbrElement[elementIndex]->elementID = elementID; self->pSbrElement[elementIndex]->nChannels = elChannels; for (ch = 0; ch < elChannels; ch++) { if (self->pSbrElement[elementIndex]->pSbrChannel[ch] == NULL) { self->pSbrElement[elementIndex]->pSbrChannel[ch] = GetRam_SbrDecChannel(chCnt); if (self->pSbrElement[elementIndex]->pSbrChannel[ch] == NULL) { sbrError = SBRDEC_MEM_ALLOC_FAILED; goto bail; } } self->numSbrChannels++; sbrDecoder_drcInitChannel(&self->pSbrElement[elementIndex] ->pSbrChannel[ch] ->SbrDec.sbrDrcChannel); chCnt++; } } if (!self->pQmfDomain->globalConf.qmfDomainExplicitConfig) { self->pQmfDomain->globalConf.nInputChannels_requested = self->numSbrChannels; self->pQmfDomain->globalConf.nOutputChannels_requested = fMax((INT)self->numSbrChannels, (INT)self->pQmfDomain->globalConf.nOutputChannels_requested); } /* Make sure each SBR channel has one QMF channel assigned even if * numSbrChannels or element set-up has changed. */ sbrDecoder_AssignQmfChannels2SbrChannels(self); /* clear error flags for all delay slots */ FDKmemclear(self->pSbrElement[elementIndex]->frameErrorFlag, ((1) + 1) * sizeof(UCHAR)); { int overlap; if (coreCodec == AOT_ER_AAC_ELD) { overlap = 0; } else if (self->flags & SBRDEC_QUAD_RATE) { overlap = (3 * 4); } else { overlap = (3 * 2); } /* Initialize this instance */ sbrError = sbrDecoder_ResetElement(self, sampleRateIn, sampleRateOut, samplesPerFrame, elementID, elementIndex, overlap); } bail: if (sbrError != SBRDEC_OK) { if ((nSbrElementsStart < self->numSbrElements) || (nSbrChannelsStart < self->numSbrChannels)) { /* Free the memory allocated for this element */ sbrDecoder_DestroyElement(self, elementIndex); } else if ((elementIndex < (8)) && (self->pSbrElement[elementIndex] != NULL)) { /* Set error flag to trigger concealment */ setFrameErrorFlag(self->pSbrElement[elementIndex], FRAME_ERROR); } } return sbrError; } /** * \brief Free config dependent SBR memory. * \param self SBR decoder instance handle */ SBR_ERROR sbrDecoder_FreeMem(HANDLE_SBRDECODER *self) { int i; int elIdx; if (self != NULL && *self != NULL) { for (i = 0; i < (8); i++) { sbrDecoder_DestroyElement(*self, i); } for (elIdx = 0; elIdx < (8); elIdx += 1) { for (i = 0; i < (1) + 1; i += 1) { (*self)->sbrHeader[elIdx][i].syncState = SBR_NOT_INITIALIZED; } } } return SBRDEC_OK; } /** * \brief Apply decoded SBR header for one element. * \param self SBR decoder instance handle * \param hSbrHeader SBR header handle to be processed. * \param hSbrChannel pointer array to the SBR element channels corresponding to * the SBR header. * \param headerStatus header status value returned from SBR header parser. * \param numElementChannels amount of channels for the SBR element whos header * is to be processed. */ static SBR_ERROR sbrDecoder_HeaderUpdate(HANDLE_SBRDECODER self, HANDLE_SBR_HEADER_DATA hSbrHeader, SBR_HEADER_STATUS headerStatus, HANDLE_SBR_CHANNEL hSbrChannel[], const int numElementChannels) { SBR_ERROR errorStatus = SBRDEC_OK; /* change of control data, reset decoder */ errorStatus = resetFreqBandTables(hSbrHeader, self->flags); if (errorStatus == SBRDEC_OK) { if (hSbrHeader->syncState == UPSAMPLING && headerStatus != HEADER_RESET) { #if (SBRDEC_MAX_HB_FADE_FRAMES > 0) int ch; for (ch = 0; ch < numElementChannels; ch += 1) { hSbrChannel[ch]->SbrDec.highBandFadeCnt = SBRDEC_MAX_HB_FADE_FRAMES; } #endif /* As the default header would limit the frequency range, lowSubband and highSubband must be patched. */ hSbrHeader->freqBandData.lowSubband = hSbrHeader->numberOfAnalysisBands; hSbrHeader->freqBandData.highSubband = hSbrHeader->numberOfAnalysisBands; } /* Trigger a reset before processing this slot */ hSbrHeader->status |= SBRDEC_HDR_STAT_RESET; } return errorStatus; } INT sbrDecoder_Header(HANDLE_SBRDECODER self, HANDLE_FDK_BITSTREAM hBs, const INT sampleRateIn, const INT sampleRateOut, const INT samplesPerFrame, const AUDIO_OBJECT_TYPE coreCodec, const MP4_ELEMENT_ID elementID, const INT elementIndex, const UCHAR harmonicSBR, const UCHAR stereoConfigIndex, const UCHAR configMode, UCHAR *configChanged, const INT downscaleFactor) { SBR_HEADER_STATUS headerStatus; HANDLE_SBR_HEADER_DATA hSbrHeader; SBR_ERROR sbrError = SBRDEC_OK; int headerIndex; UINT flagsSaved = 0; /* flags should not be changed in AC_CM_DET_CFG_CHANGE - mode after parsing */ if (self == NULL || elementIndex >= (8)) { return SBRDEC_UNSUPPORTED_CONFIG; } if (!sbrDecoder_isCoreCodecValid(coreCodec)) { return SBRDEC_UNSUPPORTED_CONFIG; } if (configMode & AC_CM_DET_CFG_CHANGE) { flagsSaved = self->flags; /* store */ } sbrError = sbrDecoder_InitElement( self, sampleRateIn, sampleRateOut, samplesPerFrame, coreCodec, elementID, elementIndex, harmonicSBR, stereoConfigIndex, configMode, configChanged, downscaleFactor); if ((sbrError != SBRDEC_OK) || (elementID == ID_LFE)) { goto bail; } if (configMode & AC_CM_DET_CFG_CHANGE) { hSbrHeader = NULL; } else { headerIndex = getHeaderSlot(self->pSbrElement[elementIndex]->useFrameSlot, self->pSbrElement[elementIndex]->useHeaderSlot); hSbrHeader = &(self->sbrHeader[elementIndex][headerIndex]); } headerStatus = sbrGetHeaderData(hSbrHeader, hBs, self->flags, 0, configMode); if (coreCodec == AOT_USAC) { if (configMode & AC_CM_DET_CFG_CHANGE) { self->flags = flagsSaved; /* restore */ } return sbrError; } if (configMode & AC_CM_ALLOC_MEM) { SBR_DECODER_ELEMENT *pSbrElement; pSbrElement = self->pSbrElement[elementIndex]; /* Sanity check */ if (pSbrElement != NULL) { if ((elementID == ID_CPE && pSbrElement->nChannels != 2) || (elementID != ID_CPE && pSbrElement->nChannels != 1)) { return SBRDEC_UNSUPPORTED_CONFIG; } if (headerStatus == HEADER_RESET) { sbrError = sbrDecoder_HeaderUpdate(self, hSbrHeader, headerStatus, pSbrElement->pSbrChannel, pSbrElement->nChannels); if (sbrError == SBRDEC_OK) { hSbrHeader->syncState = SBR_HEADER; hSbrHeader->status |= SBRDEC_HDR_STAT_UPDATE; } else { hSbrHeader->syncState = SBR_NOT_INITIALIZED; hSbrHeader->status = HEADER_ERROR; } } } } bail: if (configMode & AC_CM_DET_CFG_CHANGE) { self->flags = flagsSaved; /* restore */ } return sbrError; } SBR_ERROR sbrDecoder_SetParam(HANDLE_SBRDECODER self, const SBRDEC_PARAM param, const INT value) { SBR_ERROR errorStatus = SBRDEC_OK; /* configure the subsystems */ switch (param) { case SBR_SYSTEM_BITSTREAM_DELAY: if (value < 0 || value > (1)) { errorStatus = SBRDEC_SET_PARAM_FAIL; break; } if (self == NULL) { errorStatus = SBRDEC_NOT_INITIALIZED; } else { self->numDelayFrames = (UCHAR)value; } break; case SBR_QMF_MODE: if (self == NULL) { errorStatus = SBRDEC_NOT_INITIALIZED; } else { if (value == 1) { self->flags |= SBRDEC_LOW_POWER; } else { self->flags &= ~SBRDEC_LOW_POWER; } } break; case SBR_LD_QMF_TIME_ALIGN: if (self == NULL) { errorStatus = SBRDEC_NOT_INITIALIZED; } else { if (value == 1) { self->flags |= SBRDEC_LD_MPS_QMF; } else { self->flags &= ~SBRDEC_LD_MPS_QMF; } } break; case SBR_FLUSH_DATA: if (value != 0) { if (self == NULL) { errorStatus = SBRDEC_NOT_INITIALIZED; } else { self->flags |= SBRDEC_FLUSH; } } break; case SBR_CLEAR_HISTORY: if (value != 0) { if (self == NULL) { errorStatus = SBRDEC_NOT_INITIALIZED; } else { self->flags |= SBRDEC_FORCE_RESET; } } break; case SBR_BS_INTERRUPTION: { int elementIndex; if (self == NULL) { errorStatus = SBRDEC_NOT_INITIALIZED; break; } /* Loop over SBR elements */ for (elementIndex = 0; elementIndex < self->numSbrElements; elementIndex++) { if (self->pSbrElement[elementIndex] != NULL) { HANDLE_SBR_HEADER_DATA hSbrHeader; int headerIndex = getHeaderSlot(self->pSbrElement[elementIndex]->useFrameSlot, self->pSbrElement[elementIndex]->useHeaderSlot); hSbrHeader = &(self->sbrHeader[elementIndex][headerIndex]); /* Set sync state UPSAMPLING for the corresponding slot. This switches off bitstream parsing until a new header arrives. */ hSbrHeader->syncState = UPSAMPLING; hSbrHeader->status |= SBRDEC_HDR_STAT_UPDATE; } } } break; case SBR_SKIP_QMF: if (self == NULL) { errorStatus = SBRDEC_NOT_INITIALIZED; } else { if (value == 1) { self->flags |= SBRDEC_SKIP_QMF_ANA; } else { self->flags &= ~SBRDEC_SKIP_QMF_ANA; } if (value == 2) { self->flags |= SBRDEC_SKIP_QMF_SYN; } else { self->flags &= ~SBRDEC_SKIP_QMF_SYN; } } break; default: errorStatus = SBRDEC_SET_PARAM_FAIL; break; } /* switch(param) */ return (errorStatus); } static SBRDEC_DRC_CHANNEL *sbrDecoder_drcGetChannel( const HANDLE_SBRDECODER self, const INT channel) { SBRDEC_DRC_CHANNEL *pSbrDrcChannelData = NULL; int elementIndex, elChanIdx = 0, numCh = 0; for (elementIndex = 0; (elementIndex < (8)) && (numCh <= channel); elementIndex++) { SBR_DECODER_ELEMENT *pSbrElement = self->pSbrElement[elementIndex]; int c, elChannels; elChanIdx = 0; if (pSbrElement == NULL) break; /* Determine amount of channels for this element */ switch (pSbrElement->elementID) { case ID_CPE: elChannels = 2; break; case ID_LFE: case ID_SCE: elChannels = 1; break; case ID_NONE: default: elChannels = 0; break; } /* Limit with actual allocated element channels */ elChannels = fMin(elChannels, pSbrElement->nChannels); for (c = 0; (c < elChannels) && (numCh <= channel); c++) { if (pSbrElement->pSbrChannel[elChanIdx] != NULL) { numCh++; elChanIdx++; } } } elementIndex -= 1; elChanIdx -= 1; if (elChanIdx < 0 || elementIndex < 0) { return NULL; } if (self->pSbrElement[elementIndex] != NULL) { if (self->pSbrElement[elementIndex]->pSbrChannel[elChanIdx] != NULL) { pSbrDrcChannelData = &self->pSbrElement[elementIndex] ->pSbrChannel[elChanIdx] ->SbrDec.sbrDrcChannel; } } return (pSbrDrcChannelData); } SBR_ERROR sbrDecoder_drcFeedChannel(HANDLE_SBRDECODER self, INT ch, UINT numBands, FIXP_DBL *pNextFact_mag, INT nextFact_exp, SHORT drcInterpolationScheme, UCHAR winSequence, USHORT *pBandTop) { SBRDEC_DRC_CHANNEL *pSbrDrcChannelData = NULL; int band, isValidData = 0; if (self == NULL) { return SBRDEC_NOT_INITIALIZED; } if (ch > (8) || pNextFact_mag == NULL) { return SBRDEC_SET_PARAM_FAIL; } /* Search for gain values different to 1.0f */ for (band = 0; band < (int)numBands; band += 1) { if (!((pNextFact_mag[band] == FL2FXCONST_DBL(0.5)) && (nextFact_exp == 1)) && !((pNextFact_mag[band] == (FIXP_DBL)MAXVAL_DBL) && (nextFact_exp == 0))) { isValidData = 1; break; } } /* Find the right SBR channel */ pSbrDrcChannelData = sbrDecoder_drcGetChannel(self, ch); if (pSbrDrcChannelData != NULL) { if (pSbrDrcChannelData->enable || isValidData) { /* Activate processing only with real and valid data */ int i; pSbrDrcChannelData->enable = 1; pSbrDrcChannelData->numBandsNext = numBands; pSbrDrcChannelData->winSequenceNext = winSequence; pSbrDrcChannelData->drcInterpolationSchemeNext = drcInterpolationScheme; pSbrDrcChannelData->nextFact_exp = nextFact_exp; for (i = 0; i < (int)numBands; i++) { pSbrDrcChannelData->bandTopNext[i] = pBandTop[i]; pSbrDrcChannelData->nextFact_mag[i] = pNextFact_mag[i]; } } } return SBRDEC_OK; } void sbrDecoder_drcDisable(HANDLE_SBRDECODER self, INT ch) { SBRDEC_DRC_CHANNEL *pSbrDrcChannelData = NULL; if ((self == NULL) || (ch > (8)) || (self->numSbrElements == 0) || (self->numSbrChannels == 0)) { return; } /* Find the right SBR channel */ pSbrDrcChannelData = sbrDecoder_drcGetChannel(self, ch); if (pSbrDrcChannelData != NULL) { sbrDecoder_drcInitChannel(pSbrDrcChannelData); } } SBR_ERROR sbrDecoder_Parse(HANDLE_SBRDECODER self, HANDLE_FDK_BITSTREAM hBs, UCHAR *pDrmBsBuffer, USHORT drmBsBufferSize, int *count, int bsPayLen, int crcFlag, MP4_ELEMENT_ID prevElement, int elementIndex, UINT acFlags, UINT acElFlags[]) { SBR_DECODER_ELEMENT *hSbrElement = NULL; HANDLE_SBR_HEADER_DATA hSbrHeader = NULL; HANDLE_SBR_CHANNEL *pSbrChannel; SBR_FRAME_DATA *hFrameDataLeft = NULL; SBR_FRAME_DATA *hFrameDataRight = NULL; SBR_FRAME_DATA frameDataLeftCopy; SBR_FRAME_DATA frameDataRightCopy; SBR_ERROR errorStatus = SBRDEC_OK; SBR_HEADER_STATUS headerStatus = HEADER_NOT_PRESENT; INT startPos = FDKgetValidBits(hBs); FDK_CRCINFO crcInfo; INT crcReg = 0; USHORT sbrCrc = 0; UINT crcPoly; UINT crcStartValue = 0; UINT crcLen; HANDLE_FDK_BITSTREAM hBsOriginal = hBs; FDK_BITSTREAM bsBwd; const int fGlobalIndependencyFlag = acFlags & AC_INDEP; const int bs_pvc = acElFlags[elementIndex] & AC_EL_USAC_PVC; const int bs_interTes = acElFlags[elementIndex] & AC_EL_USAC_ITES; int stereo; int fDoDecodeSbrData = 1; int alignBits = 0; int lastSlot, lastHdrSlot = 0, thisHdrSlot = 0; if (*count <= 0) { setFrameErrorFlag(self->pSbrElement[elementIndex], FRAME_ERROR); return SBRDEC_OK; } /* SBR sanity checks */ if (self == NULL) { errorStatus = SBRDEC_NOT_INITIALIZED; goto bail; } /* Reverse bits of DRM SBR payload */ if ((self->flags & SBRDEC_SYNTAX_DRM) && *count > 0) { int dataBytes, dataBits; FDK_ASSERT(drmBsBufferSize >= (512)); dataBits = *count; if (dataBits > ((512) * 8)) { /* do not flip more data than needed */ dataBits = (512) * 8; } dataBytes = (dataBits + 7) >> 3; int j; if ((j = (int)FDKgetValidBits(hBs)) != 8) { FDKpushBiDirectional(hBs, (j - 8)); } j = 0; for (; dataBytes > 0; dataBytes--) { int i; UCHAR tmpByte; UCHAR buffer = 0x00; tmpByte = (UCHAR)FDKreadBits(hBs, 8); for (i = 0; i < 4; i++) { int shift = 2 * i + 1; buffer |= (tmpByte & (0x08 >> i)) << shift; buffer |= (tmpByte & (0x10 << i)) >> shift; } pDrmBsBuffer[j++] = buffer; FDKpushBack(hBs, 16); } FDKinitBitStream(&bsBwd, pDrmBsBuffer, (512), dataBits, BS_READER); /* Use reversed data */ hBs = &bsBwd; bsPayLen = *count; } /* Remember start position of SBR element */ startPos = FDKgetValidBits(hBs); /* SBR sanity checks */ if (self->pSbrElement[elementIndex] == NULL) { errorStatus = SBRDEC_NOT_INITIALIZED; goto bail; } hSbrElement = self->pSbrElement[elementIndex]; lastSlot = (hSbrElement->useFrameSlot > 0) ? hSbrElement->useFrameSlot - 1 : self->numDelayFrames; lastHdrSlot = hSbrElement->useHeaderSlot[lastSlot]; thisHdrSlot = getHeaderSlot( hSbrElement->useFrameSlot, hSbrElement->useHeaderSlot); /* Get a free header slot not used by frames not processed yet. */ /* Assign the free slot to store a new header if there is one. */ hSbrHeader = &self->sbrHeader[elementIndex][thisHdrSlot]; pSbrChannel = hSbrElement->pSbrChannel; stereo = (hSbrElement->elementID == ID_CPE) ? 1 : 0; hFrameDataLeft = &self->pSbrElement[elementIndex] ->pSbrChannel[0] ->frameData[hSbrElement->useFrameSlot]; if (stereo) { hFrameDataRight = &self->pSbrElement[elementIndex] ->pSbrChannel[1] ->frameData[hSbrElement->useFrameSlot]; } /* store frameData; new parsed frameData possibly corrupted */ FDKmemcpy(&frameDataLeftCopy, hFrameDataLeft, sizeof(SBR_FRAME_DATA)); if (stereo) { FDKmemcpy(&frameDataRightCopy, hFrameDataRight, sizeof(SBR_FRAME_DATA)); } /* reset PS flag; will be set after PS was found */ self->flags &= ~SBRDEC_PS_DECODED; if (hSbrHeader->status & SBRDEC_HDR_STAT_UPDATE) { /* Got a new header from extern (e.g. from an ASC) */ headerStatus = HEADER_OK; hSbrHeader->status &= ~SBRDEC_HDR_STAT_UPDATE; } else if (thisHdrSlot != lastHdrSlot) { /* Copy the last header into this slot otherwise the header compare will trigger more HEADER_RESETs than needed. */ copySbrHeader(hSbrHeader, &self->sbrHeader[elementIndex][lastHdrSlot]); } /* Check if bit stream data is valid and matches the element context */ if (((prevElement != ID_SCE) && (prevElement != ID_CPE)) || prevElement != hSbrElement->elementID) { /* In case of LFE we also land here, since there is no LFE SBR element (do * upsampling only) */ fDoDecodeSbrData = 0; } if (fDoDecodeSbrData) { if ((INT)FDKgetValidBits(hBs) <= 0) { fDoDecodeSbrData = 0; } } /* SBR CRC-check */ if (fDoDecodeSbrData) { if (crcFlag) { switch (self->coreCodec) { case AOT_DRM_AAC: case AOT_DRM_SURROUND: crcPoly = 0x001d; crcLen = 8; crcStartValue = 0x000000ff; break; default: crcPoly = 0x0633; crcLen = 10; crcStartValue = 0x00000000; break; } sbrCrc = (USHORT)FDKreadBits(hBs, crcLen); /* Setup CRC decoder */ FDKcrcInit(&crcInfo, crcPoly, crcStartValue, crcLen); /* Start CRC region */ crcReg = FDKcrcStartReg(&crcInfo, hBs, 0); } } /* if (fDoDecodeSbrData) */ /* Read in the header data and issue a reset if change occured */ if (fDoDecodeSbrData) { int sbrHeaderPresent; if (self->flags & (SBRDEC_SYNTAX_RSVD50 | SBRDEC_SYNTAX_USAC)) { SBR_HEADER_DATA_BS_INFO newSbrInfo; int sbrInfoPresent; if (bs_interTes) { self->flags |= SBRDEC_USAC_ITES; } else { self->flags &= ~SBRDEC_USAC_ITES; } if (fGlobalIndependencyFlag) { self->flags |= SBRDEC_USAC_INDEP; sbrInfoPresent = 1; sbrHeaderPresent = 1; } else { self->flags &= ~SBRDEC_USAC_INDEP; sbrInfoPresent = FDKreadBit(hBs); if (sbrInfoPresent) { sbrHeaderPresent = FDKreadBit(hBs); } else { sbrHeaderPresent = 0; } } if (sbrInfoPresent) { newSbrInfo.ampResolution = FDKreadBit(hBs); newSbrInfo.xover_band = FDKreadBits(hBs, 4); newSbrInfo.sbr_preprocessing = FDKreadBit(hBs); if (bs_pvc) { newSbrInfo.pvc_mode = FDKreadBits(hBs, 2); /* bs_pvc_mode: 0 -> no PVC, 1 -> PVC mode 1, 2 -> PVC mode 2, 3 -> * reserved */ if (newSbrInfo.pvc_mode > 2) { headerStatus = HEADER_ERROR; } if (stereo && newSbrInfo.pvc_mode > 0) { /* bs_pvc is always transmitted but pvc_mode is set to zero in case * of stereo SBR. The config might be wrong but we cannot tell for * sure. */ newSbrInfo.pvc_mode = 0; } } else { newSbrInfo.pvc_mode = 0; } if (headerStatus != HEADER_ERROR) { if (FDKmemcmp(&hSbrHeader->bs_info, &newSbrInfo, sizeof(SBR_HEADER_DATA_BS_INFO))) { /* in case of ampResolution and preprocessing change no full reset * required */ /* HEADER reset would trigger HBE transposer reset which breaks * eSbr_3_Eaa.mp4 */ if ((hSbrHeader->bs_info.pvc_mode != newSbrInfo.pvc_mode) || (hSbrHeader->bs_info.xover_band != newSbrInfo.xover_band)) { headerStatus = HEADER_RESET; } else { headerStatus = HEADER_OK; } hSbrHeader->bs_info = newSbrInfo; } else { headerStatus = HEADER_OK; } } } if (headerStatus == HEADER_ERROR) { /* Corrupt SBR info data, do not decode and switch to UPSAMPLING */ hSbrHeader->syncState = UPSAMPLING; fDoDecodeSbrData = 0; sbrHeaderPresent = 0; } if (sbrHeaderPresent && fDoDecodeSbrData) { int useDfltHeader; useDfltHeader = FDKreadBit(hBs); if (useDfltHeader) { sbrHeaderPresent = 0; if (FDKmemcmp(&hSbrHeader->bs_data, &hSbrHeader->bs_dflt, sizeof(SBR_HEADER_DATA_BS)) || hSbrHeader->syncState != SBR_ACTIVE) { hSbrHeader->bs_data = hSbrHeader->bs_dflt; headerStatus = HEADER_RESET; } } } } else { sbrHeaderPresent = FDKreadBit(hBs); } if (sbrHeaderPresent) { headerStatus = sbrGetHeaderData(hSbrHeader, hBs, self->flags, 1, 0); } if (headerStatus == HEADER_RESET) { errorStatus = sbrDecoder_HeaderUpdate( self, hSbrHeader, headerStatus, pSbrChannel, hSbrElement->nChannels); if (errorStatus == SBRDEC_OK) { hSbrHeader->syncState = SBR_HEADER; } else { hSbrHeader->syncState = SBR_NOT_INITIALIZED; headerStatus = HEADER_ERROR; } } if (errorStatus != SBRDEC_OK) { fDoDecodeSbrData = 0; } } /* if (fDoDecodeSbrData) */ /* Print debugging output only if state has changed */ /* read frame data */ if ((hSbrHeader->syncState >= SBR_HEADER) && fDoDecodeSbrData) { int sbrFrameOk; /* read the SBR element data */ if (!stereo && (self->hParametricStereoDec != NULL)) { /* update slot index for PS bitstream parsing */ self->hParametricStereoDec->bsLastSlot = self->hParametricStereoDec->bsReadSlot; self->hParametricStereoDec->bsReadSlot = hSbrElement->useFrameSlot; } sbrFrameOk = sbrGetChannelElement( hSbrHeader, hFrameDataLeft, (stereo) ? hFrameDataRight : NULL, &pSbrChannel[0]->prevFrameData, pSbrChannel[0]->SbrDec.PvcStaticData.pvc_mode_last, hBs, (stereo) ? NULL : self->hParametricStereoDec, self->flags, self->pSbrElement[elementIndex]->transposerSettings.overlap); if (!sbrFrameOk) { fDoDecodeSbrData = 0; } else { INT valBits; if (bsPayLen > 0) { valBits = bsPayLen - ((INT)startPos - (INT)FDKgetValidBits(hBs)); } else { valBits = (INT)FDKgetValidBits(hBs); } /* sanity check of remaining bits */ if (valBits < 0) { fDoDecodeSbrData = 0; } else { switch (self->coreCodec) { case AOT_SBR: case AOT_PS: case AOT_AAC_LC: { /* This sanity check is only meaningful with General Audio * bitstreams */ alignBits = valBits & 0x7; if (valBits > alignBits) { fDoDecodeSbrData = 0; } } break; default: /* No sanity check available */ break; } } } } else { /* The returned bit count will not be the actual payload size since we did not parse the frame data. Return an error so that the caller can react respectively. */ errorStatus = SBRDEC_PARSE_ERROR; } if (crcFlag && (hSbrHeader->syncState >= SBR_HEADER) && fDoDecodeSbrData) { FDKpushFor(hBs, alignBits); FDKcrcEndReg(&crcInfo, hBs, crcReg); /* End CRC region */ FDKpushBack(hBs, alignBits); /* Check CRC */ if ((FDKcrcGetCRC(&crcInfo) ^ crcStartValue) != sbrCrc) { fDoDecodeSbrData = 0; if (headerStatus != HEADER_NOT_PRESENT) { headerStatus = HEADER_ERROR; hSbrHeader->syncState = SBR_NOT_INITIALIZED; } } } if (!fDoDecodeSbrData) { /* Set error flag for this slot to trigger concealment */ setFrameErrorFlag(self->pSbrElement[elementIndex], FRAME_ERROR); /* restore old frameData for concealment */ FDKmemcpy(hFrameDataLeft, &frameDataLeftCopy, sizeof(SBR_FRAME_DATA)); if (stereo) { FDKmemcpy(hFrameDataRight, &frameDataRightCopy, sizeof(SBR_FRAME_DATA)); } errorStatus = SBRDEC_PARSE_ERROR; } else { /* Everything seems to be ok so clear the error flag */ setFrameErrorFlag(self->pSbrElement[elementIndex], FRAME_OK); } if (!stereo) { /* Turn coupling off explicitely to avoid access to absent right frame data that might occur with corrupt bitstreams. */ hFrameDataLeft->coupling = COUPLING_OFF; } bail: if (self != NULL) { if (self->flags & SBRDEC_SYNTAX_DRM) { hBs = hBsOriginal; } if (errorStatus != SBRDEC_NOT_INITIALIZED) { int useOldHdr = ((headerStatus == HEADER_NOT_PRESENT) || (headerStatus == HEADER_ERROR) || (headerStatus == HEADER_RESET && errorStatus == SBRDEC_PARSE_ERROR)) ? 1 : 0; if (!useOldHdr && (thisHdrSlot != lastHdrSlot) && (hSbrHeader != NULL)) { useOldHdr |= (compareSbrHeader(hSbrHeader, &self->sbrHeader[elementIndex][lastHdrSlot]) == 0) ? 1 : 0; } if (hSbrElement != NULL) { if (useOldHdr != 0) { /* Use the old header for this frame */ hSbrElement->useHeaderSlot[hSbrElement->useFrameSlot] = lastHdrSlot; } else { /* Use the new header for this frame */ hSbrElement->useHeaderSlot[hSbrElement->useFrameSlot] = thisHdrSlot; } /* Move frame pointer to the next slot which is up to be decoded/applied * next */ hSbrElement->useFrameSlot = (hSbrElement->useFrameSlot + 1) % (self->numDelayFrames + 1); } } } *count -= startPos - (INT)FDKgetValidBits(hBs); return errorStatus; } /** * \brief Render one SBR element into time domain signal. * \param self SBR decoder handle * \param timeData pointer to output buffer * \param channelMapping pointer to UCHAR array where next 2 channel offsets are * stored. * \param elementIndex enumerating index of the SBR element to render. * \param numInChannels number of channels from core coder. * \param numOutChannels pointer to a location to return number of output * channels. * \param psPossible flag indicating if PS is possible or not. * \return SBRDEC_OK if successfull, else error code */ static SBR_ERROR sbrDecoder_DecodeElement( HANDLE_SBRDECODER self, LONG *input, LONG *timeData, const int timeDataSize, const FDK_channelMapDescr *const mapDescr, const int mapIdx, int channelIndex, const int elementIndex, const int numInChannels, int *numOutChannels, const int psPossible) { SBR_DECODER_ELEMENT *hSbrElement = self->pSbrElement[elementIndex]; HANDLE_SBR_CHANNEL *pSbrChannel = self->pSbrElement[elementIndex]->pSbrChannel; HANDLE_SBR_HEADER_DATA hSbrHeader = &self->sbrHeader[elementIndex] [hSbrElement->useHeaderSlot[hSbrElement->useFrameSlot]]; HANDLE_PS_DEC h_ps_d = self->hParametricStereoDec; /* get memory for frame data from scratch */ SBR_FRAME_DATA *hFrameDataLeft = NULL; SBR_FRAME_DATA *hFrameDataRight = NULL; SBR_ERROR errorStatus = SBRDEC_OK; INT strideOut, offset0 = 255, offset0_block = 0, offset1 = 255, offset1_block = 0; INT codecFrameSize = self->codecFrameSize; int stereo = (hSbrElement->elementID == ID_CPE) ? 1 : 0; int numElementChannels = hSbrElement ->nChannels; /* Number of channels of the current SBR element */ hFrameDataLeft = &hSbrElement->pSbrChannel[0]->frameData[hSbrElement->useFrameSlot]; if (stereo) { hFrameDataRight = &hSbrElement->pSbrChannel[1]->frameData[hSbrElement->useFrameSlot]; } if (self->flags & SBRDEC_FLUSH) { if (self->numFlushedFrames > self->numDelayFrames) { int hdrIdx; /* No valid SBR payload available, hence switch to upsampling (in all * headers) */ for (hdrIdx = 0; hdrIdx < ((1) + 1); hdrIdx += 1) { self->sbrHeader[elementIndex][hdrIdx].syncState = UPSAMPLING; } } else { /* Move frame pointer to the next slot which is up to be decoded/applied * next */ hSbrElement->useFrameSlot = (hSbrElement->useFrameSlot + 1) % (self->numDelayFrames + 1); /* Update header and frame data pointer because they have already been set */ hSbrHeader = &self->sbrHeader[elementIndex] [hSbrElement ->useHeaderSlot[hSbrElement->useFrameSlot]]; hFrameDataLeft = &hSbrElement->pSbrChannel[0]->frameData[hSbrElement->useFrameSlot]; if (stereo) { hFrameDataRight = &hSbrElement->pSbrChannel[1]->frameData[hSbrElement->useFrameSlot]; } } } /* Update the header error flag */ hSbrHeader->frameErrorFlag = hSbrElement->frameErrorFlag[hSbrElement->useFrameSlot]; /* Prepare filterbank for upsampling if no valid bit stream data is available. */ if (hSbrHeader->syncState == SBR_NOT_INITIALIZED) { errorStatus = initHeaderData(hSbrHeader, self->sampleRateIn, self->sampleRateOut, self->downscaleFactor, codecFrameSize, self->flags, 1 /* SET_DEFAULT_HDR */ ); if (errorStatus != SBRDEC_OK) { return errorStatus; } hSbrHeader->syncState = UPSAMPLING; errorStatus = sbrDecoder_HeaderUpdate(self, hSbrHeader, HEADER_NOT_PRESENT, pSbrChannel, hSbrElement->nChannels); if (errorStatus != SBRDEC_OK) { hSbrHeader->syncState = SBR_NOT_INITIALIZED; return errorStatus; } } /* reset */ if (hSbrHeader->status & SBRDEC_HDR_STAT_RESET) { int ch; int applySbrProc = (hSbrHeader->syncState == SBR_ACTIVE || (hSbrHeader->frameErrorFlag == 0 && hSbrHeader->syncState == SBR_HEADER)); for (ch = 0; ch < numElementChannels; ch++) { SBR_ERROR errorStatusTmp = SBRDEC_OK; errorStatusTmp = resetSbrDec( &pSbrChannel[ch]->SbrDec, hSbrHeader, &pSbrChannel[ch]->prevFrameData, self->synDownsampleFac, self->flags, pSbrChannel[ch]->frameData); if (errorStatusTmp != SBRDEC_OK) { hSbrHeader->syncState = UPSAMPLING; } } if (applySbrProc) { hSbrHeader->status &= ~SBRDEC_HDR_STAT_RESET; } } /* decoding */ if ((hSbrHeader->syncState == SBR_ACTIVE) || ((hSbrHeader->syncState == SBR_HEADER) && (hSbrHeader->frameErrorFlag == 0))) { errorStatus = SBRDEC_OK; decodeSbrData(hSbrHeader, hFrameDataLeft, &pSbrChannel[0]->prevFrameData, (stereo) ? hFrameDataRight : NULL, (stereo) ? &pSbrChannel[1]->prevFrameData : NULL); /* Now we have a full parameter set and can do parameter based concealment instead of plain upsampling. */ hSbrHeader->syncState = SBR_ACTIVE; } if (timeDataSize < hSbrHeader->numberTimeSlots * hSbrHeader->timeStep * self->pQmfDomain->globalConf.nBandsSynthesis * (psPossible ? fMax(2, numInChannels) : numInChannels)) { return SBRDEC_OUTPUT_BUFFER_TOO_SMALL; } { self->flags &= ~SBRDEC_PS_DECODED; C_ALLOC_SCRATCH_START(pPsScratch, struct PS_DEC_COEFFICIENTS, 1) /* decode PS data if available */ if (h_ps_d != NULL && psPossible && (hSbrHeader->syncState == SBR_ACTIVE)) { int applyPs = 1; /* define which frame delay line slot to process */ h_ps_d->processSlot = hSbrElement->useFrameSlot; applyPs = DecodePs(h_ps_d, hSbrHeader->frameErrorFlag, pPsScratch); self->flags |= (applyPs) ? SBRDEC_PS_DECODED : 0; } offset0 = FDK_chMapDescr_getMapValue(mapDescr, channelIndex, mapIdx); offset0_block = offset0 * codecFrameSize; if (stereo || psPossible) { /* the value of offset1 only matters if the condition is true, however if it is not true channelIndex+1 may exceed the channel map resutling in an error, though the value of offset1 is actually meaningless. This is prevented here. */ offset1 = FDK_chMapDescr_getMapValue(mapDescr, channelIndex + 1, mapIdx); offset1_block = offset1 * codecFrameSize; } /* Set strides for reading and writing */ if (psPossible) strideOut = (numInChannels < 2) ? 2 : numInChannels; else strideOut = numInChannels; /* use same buffers for left and right channel and apply PS per timeslot */ /* Process left channel */ sbr_dec(&pSbrChannel[0]->SbrDec, input + offset0_block, timeData + offset0, (self->flags & SBRDEC_PS_DECODED) ? &pSbrChannel[1]->SbrDec : NULL, timeData + offset1, strideOut, hSbrHeader, hFrameDataLeft, &pSbrChannel[0]->prevFrameData, (hSbrHeader->syncState == SBR_ACTIVE), h_ps_d, self->flags, codecFrameSize, self->sbrInDataHeadroom); if (stereo) { /* Process right channel */ sbr_dec(&pSbrChannel[1]->SbrDec, input + offset1_block, timeData + offset1, NULL, NULL, strideOut, hSbrHeader, hFrameDataRight, &pSbrChannel[1]->prevFrameData, (hSbrHeader->syncState == SBR_ACTIVE), NULL, self->flags, codecFrameSize, self->sbrInDataHeadroom); } C_ALLOC_SCRATCH_END(pPsScratch, struct PS_DEC_COEFFICIENTS, 1) } if (h_ps_d != NULL) { /* save PS status for next run */ h_ps_d->psDecodedPrv = (self->flags & SBRDEC_PS_DECODED) ? 1 : 0; } if (psPossible && !(self->flags & SBRDEC_SKIP_QMF_SYN)) { FDK_ASSERT(strideOut > 1); if (!(self->flags & SBRDEC_PS_DECODED)) { /* A decoder which is able to decode PS has to produce a stereo output * even if no PS data is available. */ /* So copy left channel to right channel. */ int copyFrameSize = codecFrameSize * self->pQmfDomain->QmfDomainOut->fb.no_channels; copyFrameSize /= self->pQmfDomain->QmfDomainIn->fb.no_channels; LONG *ptr; INT i; FDK_ASSERT(strideOut == 2); ptr = timeData; for (i = copyFrameSize >> 1; i--;) { LONG tmp; /* This temporal variable is required because some compilers can't do *ptr++ = *ptr++ correctly. */ tmp = *ptr++; *ptr++ = tmp; tmp = *ptr++; *ptr++ = tmp; } } *numOutChannels = 2; /* Output minimum two channels when PS is enabled. */ } return errorStatus; } SBR_ERROR sbrDecoder_Apply(HANDLE_SBRDECODER self, LONG *input, LONG *timeData, const int timeDataSize, int *numChannels, int *sampleRate, const FDK_channelMapDescr *const mapDescr, const int mapIdx, const int coreDecodedOk, UCHAR *psDecoded, const INT inDataHeadroom, INT *outDataHeadroom) { SBR_ERROR errorStatus = SBRDEC_OK; int psPossible; int sbrElementNum; int numCoreChannels; int numSbrChannels = 0; if ((self == NULL) || (timeData == NULL) || (numChannels == NULL) || (sampleRate == NULL) || (psDecoded == NULL) || !FDK_chMapDescr_isValid(mapDescr)) { return SBRDEC_INVALID_ARGUMENT; } psPossible = *psDecoded; numCoreChannels = *numChannels; if (numCoreChannels <= 0) { return SBRDEC_INVALID_ARGUMENT; } if (self->numSbrElements < 1) { /* exit immediately to avoid access violations */ return SBRDEC_NOT_INITIALIZED; } /* Sanity check of allocated SBR elements. */ for (sbrElementNum = 0; sbrElementNum < self->numSbrElements; sbrElementNum++) { if (self->pSbrElement[sbrElementNum] == NULL) { return SBRDEC_NOT_INITIALIZED; } } if (self->numSbrElements != 1 || self->pSbrElement[0]->elementID != ID_SCE) { psPossible = 0; } self->sbrInDataHeadroom = inDataHeadroom; *outDataHeadroom = (INT)(8); /* Make sure that even if no SBR data was found/parsed *psDecoded is returned * 1 if psPossible was 0. */ if (psPossible == 0) { self->flags &= ~SBRDEC_PS_DECODED; } /* replaces channel based reset inside sbr_dec() */ if (((self->flags & SBRDEC_LOW_POWER) ? 1 : 0) != ((self->pQmfDomain->globalConf.flags & QMF_FLAG_LP) ? 1 : 0)) { if (self->flags & SBRDEC_LOW_POWER) { self->pQmfDomain->globalConf.flags |= QMF_FLAG_LP; self->pQmfDomain->globalConf.flags_requested |= QMF_FLAG_LP; } else { self->pQmfDomain->globalConf.flags &= ~QMF_FLAG_LP; self->pQmfDomain->globalConf.flags_requested &= ~QMF_FLAG_LP; } if (FDK_QmfDomain_InitFilterBank(self->pQmfDomain, QMF_FLAG_KEEP_STATES)) { return SBRDEC_UNSUPPORTED_CONFIG; } } if (self->numSbrChannels > self->pQmfDomain->globalConf.nInputChannels) { return SBRDEC_UNSUPPORTED_CONFIG; } if (self->flags & SBRDEC_FLUSH) { /* flushing is signalized, hence increment the flush frame counter */ self->numFlushedFrames++; } else { /* no flushing is signalized, hence reset the flush frame counter */ self->numFlushedFrames = 0; } /* Loop over SBR elements */ for (sbrElementNum = 0; sbrElementNum < self->numSbrElements; sbrElementNum++) { int numElementChan; if (psPossible && self->pSbrElement[sbrElementNum]->pSbrChannel[1] == NULL) { /* Disable PS and try decoding SBR mono. */ psPossible = 0; } numElementChan = (self->pSbrElement[sbrElementNum]->elementID == ID_CPE) ? 2 : 1; /* If core signal is bad then force upsampling */ if (!coreDecodedOk) { setFrameErrorFlag(self->pSbrElement[sbrElementNum], FRAME_ERROR_ALLSLOTS); } errorStatus = sbrDecoder_DecodeElement( self, input, timeData, timeDataSize, mapDescr, mapIdx, numSbrChannels, sbrElementNum, numCoreChannels, /* is correct even for USC SCI==2 case */ &numElementChan, psPossible); if (errorStatus != SBRDEC_OK) { goto bail; } numSbrChannels += numElementChan; if (numSbrChannels >= numCoreChannels) { break; } } /* Update numChannels and samplerate */ /* Do not mess with output channels in case of USAC. numSbrChannels != * numChannels for stereoConfigIndex == 2 */ if (!(self->flags & SBRDEC_SYNTAX_USAC)) { *numChannels = numSbrChannels; } *sampleRate = self->sampleRateOut; *psDecoded = (self->flags & SBRDEC_PS_DECODED) ? 1 : 0; /* Clear reset and flush flag because everything seems to be done * successfully. */ self->flags &= ~SBRDEC_FORCE_RESET; self->flags &= ~SBRDEC_FLUSH; bail: return errorStatus; } SBR_ERROR sbrDecoder_Close(HANDLE_SBRDECODER *pSelf) { HANDLE_SBRDECODER self = *pSelf; int i; if (self != NULL) { if (self->hParametricStereoDec != NULL) { DeletePsDec(&self->hParametricStereoDec); } for (i = 0; i < (8); i++) { sbrDecoder_DestroyElement(self, i); } FreeRam_SbrDecoder(pSelf); } return SBRDEC_OK; } INT sbrDecoder_GetLibInfo(LIB_INFO *info) { int i; if (info == NULL) { return -1; } /* search for next free tab */ for (i = 0; i < FDK_MODULE_LAST; i++) { if (info[i].module_id == FDK_NONE) break; } if (i == FDK_MODULE_LAST) return -1; info += i; info->module_id = FDK_SBRDEC; info->version = LIB_VERSION(SBRDECODER_LIB_VL0, SBRDECODER_LIB_VL1, SBRDECODER_LIB_VL2); LIB_VERSION_STRING(info); info->build_date = SBRDECODER_LIB_BUILD_DATE; info->build_time = SBRDECODER_LIB_BUILD_TIME; info->title = SBRDECODER_LIB_TITLE; /* Set flags */ info->flags = 0 | CAPF_SBR_HQ | CAPF_SBR_LP | CAPF_SBR_PS_MPEG | CAPF_SBR_DRM_BS | CAPF_SBR_CONCEALMENT | CAPF_SBR_DRC | CAPF_SBR_ELD_DOWNSCALE | CAPF_SBR_HBEHQ; /* End of flags */ return 0; } UINT sbrDecoder_GetDelay(const HANDLE_SBRDECODER self) { UINT outputDelay = 0; if (self != NULL) { UINT flags = self->flags; /* See chapter 1.6.7.2 of ISO/IEC 14496-3 for the GA-SBR figures below. */ /* Are we initialized? */ if ((self->numSbrChannels > 0) && (self->numSbrElements > 0)) { /* Add QMF synthesis delay */ if ((flags & SBRDEC_ELD_GRID) && IS_LOWDELAY(self->coreCodec)) { /* Low delay SBR: */ if (!(flags & SBRDEC_SKIP_QMF_SYN)) { outputDelay += (flags & SBRDEC_DOWNSAMPLE) ? 32 : 64; /* QMF synthesis */ if (flags & SBRDEC_LD_MPS_QMF) { outputDelay += 32; } } } else if (!IS_USAC(self->coreCodec)) { /* By the method of elimination this is the GA (AAC-LC, HE-AAC, ...) * branch: */ outputDelay += (flags & SBRDEC_DOWNSAMPLE) ? 481 : 962; if (flags & SBRDEC_SKIP_QMF_SYN) { outputDelay -= 257; /* QMF synthesis */ } } } } return (outputDelay); } fdk-aac-2.0.2/libSBRdec/src/transcendent.h000066400000000000000000000326061404225310700202270ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR decoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief FDK Fixed Point Arithmetic Library Interface */ #ifndef TRANSCENDENT_H #define TRANSCENDENT_H #include "sbrdecoder.h" #include "sbr_rom.h" /************************************************************************/ /*! \brief Get number of octaves between frequencies a and b The Result is scaled with 1/8. The valid range for a and b is 1 to LOG_DUALIS_TABLE_SIZE. \return ld(a/b) / 8 */ /************************************************************************/ static inline FIXP_SGL FDK_getNumOctavesDiv8(INT a, /*!< lower band */ INT b) /*!< upper band */ { return ((SHORT)((LONG)(CalcLdInt(b) - CalcLdInt(a)) >> (FRACT_BITS - 3))); } /************************************************************************/ /*! \brief Add two values given by mantissa and exponent. Mantissas are in fract format with values between 0 and 1.
The base for exponents is 2. Example: \f$ a = a\_m * 2^{a\_e} \f$
*/ /************************************************************************/ inline void FDK_add_MantExp(FIXP_SGL a_m, /*!< Mantissa of 1st operand a */ SCHAR a_e, /*!< Exponent of 1st operand a */ FIXP_SGL b_m, /*!< Mantissa of 2nd operand b */ SCHAR b_e, /*!< Exponent of 2nd operand b */ FIXP_SGL *ptrSum_m, /*!< Mantissa of result */ SCHAR *ptrSum_e) /*!< Exponent of result */ { FIXP_DBL accu; int shift; int shiftAbs; FIXP_DBL shiftedMantissa; FIXP_DBL otherMantissa; /* Equalize exponents of the summands. For the smaller summand, the exponent is adapted and for compensation, the mantissa is shifted right. */ shift = (int)(a_e - b_e); shiftAbs = (shift > 0) ? shift : -shift; shiftAbs = (shiftAbs < DFRACT_BITS - 1) ? shiftAbs : DFRACT_BITS - 1; shiftedMantissa = (shift > 0) ? (FX_SGL2FX_DBL(b_m) >> shiftAbs) : (FX_SGL2FX_DBL(a_m) >> shiftAbs); otherMantissa = (shift > 0) ? FX_SGL2FX_DBL(a_m) : FX_SGL2FX_DBL(b_m); *ptrSum_e = (shift > 0) ? a_e : b_e; accu = (shiftedMantissa >> 1) + (otherMantissa >> 1); /* shift by 1 bit to avoid overflow */ if ((accu >= (FL2FXCONST_DBL(0.5f) - (FIXP_DBL)1)) || (accu <= FL2FXCONST_DBL(-0.5f))) *ptrSum_e += 1; else accu = (shiftedMantissa + otherMantissa); *ptrSum_m = FX_DBL2FX_SGL(accu); } inline void FDK_add_MantExp(FIXP_DBL a, /*!< Mantissa of 1st operand a */ SCHAR a_e, /*!< Exponent of 1st operand a */ FIXP_DBL b, /*!< Mantissa of 2nd operand b */ SCHAR b_e, /*!< Exponent of 2nd operand b */ FIXP_DBL *ptrSum, /*!< Mantissa of result */ SCHAR *ptrSum_e) /*!< Exponent of result */ { FIXP_DBL accu; int shift; int shiftAbs; FIXP_DBL shiftedMantissa; FIXP_DBL otherMantissa; /* Equalize exponents of the summands. For the smaller summand, the exponent is adapted and for compensation, the mantissa is shifted right. */ shift = (int)(a_e - b_e); shiftAbs = (shift > 0) ? shift : -shift; shiftAbs = (shiftAbs < DFRACT_BITS - 1) ? shiftAbs : DFRACT_BITS - 1; shiftedMantissa = (shift > 0) ? (b >> shiftAbs) : (a >> shiftAbs); otherMantissa = (shift > 0) ? a : b; *ptrSum_e = (shift > 0) ? a_e : b_e; accu = (shiftedMantissa >> 1) + (otherMantissa >> 1); /* shift by 1 bit to avoid overflow */ if ((accu >= (FL2FXCONST_DBL(0.5f) - (FIXP_DBL)1)) || (accu <= FL2FXCONST_DBL(-0.5f))) *ptrSum_e += 1; else accu = (shiftedMantissa + otherMantissa); *ptrSum = accu; } /************************************************************************/ /*! \brief Divide two values given by mantissa and exponent. Mantissas are in fract format with values between 0 and 1.
The base for exponents is 2. Example: \f$ a = a\_m * 2^{a\_e} \f$
For performance reasons, the division is based on a table lookup which limits accuracy. */ /************************************************************************/ static inline void FDK_divide_MantExp( FIXP_SGL a_m, /*!< Mantissa of dividend a */ SCHAR a_e, /*!< Exponent of dividend a */ FIXP_SGL b_m, /*!< Mantissa of divisor b */ SCHAR b_e, /*!< Exponent of divisor b */ FIXP_SGL *ptrResult_m, /*!< Mantissa of quotient a/b */ SCHAR *ptrResult_e) /*!< Exponent of quotient a/b */ { int preShift, postShift, index, shift; FIXP_DBL ratio_m; FIXP_SGL bInv_m = FL2FXCONST_SGL(0.0f); preShift = CntLeadingZeros(FX_SGL2FX_DBL(b_m)); /* Shift b into the range from 0..INV_TABLE_SIZE-1, E.g. 10 bits must be skipped for INV_TABLE_BITS 8: - leave 8 bits as index for table - skip sign bit, - skip first bit of mantissa, because this is always the same (>0.5) We are dealing with energies, so we need not care about negative numbers */ /* The first interval has half width so the lowest bit of the index is needed for a doubled resolution. */ shift = (FRACT_BITS - 2 - INV_TABLE_BITS - preShift); index = (shift < 0) ? (LONG)b_m << (-shift) : (LONG)b_m >> shift; /* The index has INV_TABLE_BITS +1 valid bits here. Clear the other bits. */ index &= (1 << (INV_TABLE_BITS + 1)) - 1; /* Remove offset of half an interval */ index--; /* Now the lowest bit is shifted out */ index = index >> 1; /* Fetch inversed mantissa from table: */ bInv_m = (index < 0) ? bInv_m : FDK_sbrDecoder_invTable[index]; /* Multiply a with the inverse of b: */ ratio_m = (index < 0) ? FX_SGL2FX_DBL(a_m >> 1) : fMultDiv2(bInv_m, a_m); postShift = CntLeadingZeros(ratio_m) - 1; *ptrResult_m = FX_DBL2FX_SGL(ratio_m << postShift); *ptrResult_e = a_e - b_e + 1 + preShift - postShift; } static inline void FDK_divide_MantExp( FIXP_DBL a_m, /*!< Mantissa of dividend a */ SCHAR a_e, /*!< Exponent of dividend a */ FIXP_DBL b_m, /*!< Mantissa of divisor b */ SCHAR b_e, /*!< Exponent of divisor b */ FIXP_DBL *ptrResult_m, /*!< Mantissa of quotient a/b */ SCHAR *ptrResult_e) /*!< Exponent of quotient a/b */ { int preShift, postShift, index, shift; FIXP_DBL ratio_m; FIXP_SGL bInv_m = FL2FXCONST_SGL(0.0f); preShift = CntLeadingZeros(b_m); /* Shift b into the range from 0..INV_TABLE_SIZE-1, E.g. 10 bits must be skipped for INV_TABLE_BITS 8: - leave 8 bits as index for table - skip sign bit, - skip first bit of mantissa, because this is always the same (>0.5) We are dealing with energies, so we need not care about negative numbers */ /* The first interval has half width so the lowest bit of the index is needed for a doubled resolution. */ shift = (DFRACT_BITS - 2 - INV_TABLE_BITS - preShift); index = (shift < 0) ? (LONG)b_m << (-shift) : (LONG)b_m >> shift; /* The index has INV_TABLE_BITS +1 valid bits here. Clear the other bits. */ index &= (1 << (INV_TABLE_BITS + 1)) - 1; /* Remove offset of half an interval */ index--; /* Now the lowest bit is shifted out */ index = index >> 1; /* Fetch inversed mantissa from table: */ bInv_m = (index < 0) ? bInv_m : FDK_sbrDecoder_invTable[index]; /* Multiply a with the inverse of b: */ ratio_m = (index < 0) ? (a_m >> 1) : fMultDiv2(bInv_m, a_m); postShift = CntLeadingZeros(ratio_m) - 1; *ptrResult_m = ratio_m << postShift; *ptrResult_e = a_e - b_e + 1 + preShift - postShift; } /*! \brief Calculate the squareroot of a number given by mantissa and exponent Mantissa is in fract format with values between 0 and 1.
The base for the exponent is 2. Example: \f$ a = a\_m * 2^{a\_e} \f$
The operand is addressed via pointers and will be overwritten with the result. For performance reasons, the square root is based on a table lookup which limits accuracy. */ static inline void FDK_sqrt_MantExp( FIXP_DBL *mantissa, /*!< Pointer to mantissa */ SCHAR *exponent, const SCHAR *destScale) { FIXP_DBL input_m = *mantissa; int input_e = (int)*exponent; FIXP_DBL result = FL2FXCONST_DBL(0.0f); int result_e = -FRACT_BITS; /* Call lookup square root, which does internally normalization. */ result = sqrtFixp_lookup(input_m, &input_e); result_e = input_e; /* Write result */ if (exponent == destScale) { *mantissa = result; *exponent = result_e; } else { int shift = result_e - *destScale; *mantissa = (shift >= 0) ? result << (INT)fixMin(DFRACT_BITS - 1, shift) : result >> (INT)fixMin(DFRACT_BITS - 1, -shift); *exponent = *destScale; } } #endif fdk-aac-2.0.2/libSBRenc/000077500000000000000000000000001404225310700146025ustar00rootroot00000000000000fdk-aac-2.0.2/libSBRenc/include/000077500000000000000000000000001404225310700162255ustar00rootroot00000000000000fdk-aac-2.0.2/libSBRenc/include/sbr_encoder.h000066400000000000000000000454061404225310700206740ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Description: SBR encoder top level processing prototype *******************************************************************************/ #ifndef SBR_ENCODER_H #define SBR_ENCODER_H #include "common_fix.h" #include "FDK_audio.h" #include "FDK_bitstream.h" /* core coder helpers */ #define MAX_TRANS_FAC 8 #define MAX_CODEC_FRAME_RATIO 2 #define MAX_PAYLOAD_SIZE 256 typedef enum codecType { CODEC_AAC = 0, CODEC_AACLD = 1, CODEC_UNSPECIFIED = 99 } CODEC_TYPE; typedef struct { INT bitRate; INT nChannels; INT sampleFreq; INT transFac; INT standardBitrate; } CODEC_PARAM; typedef enum { SBR_MONO, SBR_LEFT_RIGHT, SBR_COUPLING, SBR_SWITCH_LRC } SBR_STEREO_MODE; /* bitstream syntax flags */ enum { SBR_SYNTAX_LOW_DELAY = 0x0001, SBR_SYNTAX_SCALABLE = 0x0002, SBR_SYNTAX_CRC = 0x0004, SBR_SYNTAX_DRM_CRC = 0x0008, SBR_SYNTAX_ELD_REDUCED_DELAY = 0x0010 }; typedef enum { FREQ_RES_LOW = 0, FREQ_RES_HIGH } FREQ_RES; typedef struct { CODEC_TYPE coreCoder; /*!< LC or ELD */ UINT bitrateFrom; /*!< inclusive */ UINT bitrateTo; /*!< exclusive */ UINT sampleRate; /*!< */ UCHAR numChannels; /*!< */ UCHAR startFreq; /*!< bs_start_freq */ UCHAR startFreqSpeech; /*!< bs_start_freq for speech config flag */ UCHAR stopFreq; /*!< bs_stop_freq */ UCHAR stopFreqSpeech; /*!< bs_stop_freq for speech config flag */ UCHAR numNoiseBands; /*!< */ UCHAR noiseFloorOffset; /*!< */ SCHAR noiseMaxLevel; /*!< */ SBR_STEREO_MODE stereoMode; /*!< */ UCHAR freqScale; /*!< */ } sbrTuningTable_t; typedef struct sbrConfiguration { /* core coder dependent configurations */ CODEC_PARAM codecSettings; /*!< Core coder settings. To be set from core coder. */ INT SendHeaderDataTime; /*!< SBR header send update frequency in ms. */ INT useWaveCoding; /*!< Flag: usage of wavecoding tool. */ INT crcSbr; /*!< Flag: usage of SBR-CRC. */ INT dynBwSupported; /*!< Flag: support for dynamic bandwidth in this combination. */ INT parametricCoding; /*!< Flag: usage of parametric coding tool. */ INT downSampleFactor; /*!< Sampling rate relation between the SBR and the core encoder. */ FREQ_RES freq_res_fixfix[2]; /*!< Frequency resolution of envelopes in frame class FIXFIX, for non-split case and split case */ UCHAR fResTransIsLow; /*!< Frequency resolution of envelopes in transient frames: low (0) or variable (1) */ /* core coder dependent tuning parameters */ INT tran_thr; /*!< SBR transient detector threshold (* 100). */ INT noiseFloorOffset; /*!< Noise floor offset. */ UINT useSpeechConfig; /*!< Flag: adapt tuning parameters according to speech. */ /* core coder independent configurations */ INT sbrFrameSize; /*!< SBR frame size in samples. Will be calculated from core coder settings. */ INT sbr_data_extra; /*!< Flag usage of data extra. */ INT amp_res; /*!< Amplitude resolution. */ INT ana_max_level; /*!< Noise insertion maximum level. */ INT tran_fc; /*!< Transient detector start frequency. */ INT tran_det_mode; /*!< Transient detector mode. */ INT spread; /*!< Flag: usage of SBR spread. */ INT stat; /*!< Flag: usage of static framing. */ INT e; /*!< Number of envelopes when static framing is chosen. */ SBR_STEREO_MODE stereoMode; /*!< SBR stereo mode. */ INT deltaTAcrossFrames; /*!< Flag: allow time-delta coding. */ FIXP_DBL dF_edge_1stEnv; /*!< Extra fraction delta-F coding is allowed to be more expensive. */ FIXP_DBL dF_edge_incr; /*!< Increment dF_edge_1stEnv this much if dT-coding was used this frame. */ INT sbr_invf_mode; /*!< Inverse filtering mode. */ INT sbr_xpos_mode; /*!< Transposer mode. */ INT sbr_xpos_ctrl; /*!< Transposer control. */ INT sbr_xpos_level; /*!< Transposer 3rd order level. */ INT startFreq; /*!< The start frequency table index. */ INT stopFreq; /*!< The stop frequency table index. */ INT useSaPan; /*!< Flag: usage of SAPAN stereo. */ INT dynBwEnabled; /*!< Flag: usage of dynamic bandwidth. */ INT bParametricStereo; /*!< Flag: usage of parametric stereo coding tool. */ /* header_extra1 configuration */ UCHAR freqScale; /*!< Frequency grouping. */ INT alterScale; /*!< Scale resolution. */ INT sbr_noise_bands; /*!< Number of noise bands. */ /* header_extra2 configuration */ INT sbr_limiter_bands; /*!< Number of limiter bands. */ INT sbr_limiter_gains; /*!< Gain of limiter. */ INT sbr_interpol_freq; /*!< Flag: use interpolation in freq. direction. */ INT sbr_smoothing_length; /*!< Flag: choose length 4 or 0 (=on, off). */ UCHAR init_amp_res_FF; FIXP_DBL threshold_AmpRes_FF_m; SCHAR threshold_AmpRes_FF_e; } sbrConfiguration, *sbrConfigurationPtr; typedef struct SBR_CONFIG_DATA { UINT sbrSyntaxFlags; /**< SBR syntax flags derived from AOT. */ INT nChannels; /**< Number of channels. */ INT nSfb[2]; /**< Number of SBR scalefactor bands for LO_RES and HI_RES (?) */ INT num_Master; /**< Number of elements in v_k_master. */ INT sampleFreq; /**< SBR sampling frequency. */ INT frameSize; INT xOverFreq; /**< The SBR start frequency. */ INT dynXOverFreq; /**< Used crossover frequency when dynamic bandwidth is enabled. */ INT noQmfBands; /**< Number of QMF frequency bands. */ INT noQmfSlots; /**< Number of QMF slots. */ UCHAR *freqBandTable[2]; /**< Frequency table for low and hires, only MAX_FREQ_COEFFS/2 +1 coeffs actually needed for lowres. */ UCHAR *v_k_master; /**< Master BandTable where freqBandTable is derived from. */ SBR_STEREO_MODE stereoMode; INT noEnvChannels; /**< Number of envelope channels. */ INT useWaveCoding; /**< Flag indicates whether to use wave coding at all. */ INT useParametricCoding; /**< Flag indicates whether to use para coding at all. */ INT xposCtrlSwitch; /**< Flag indicates whether to switch xpos ctrl on the fly. */ INT switchTransposers; /**< Flag indicates whether to switch xpos on the fly . */ UCHAR initAmpResFF; FIXP_DBL thresholdAmpResFF_m; SCHAR thresholdAmpResFF_e; } SBR_CONFIG_DATA, *HANDLE_SBR_CONFIG_DATA; typedef struct { MP4_ELEMENT_ID elType; INT bitRate; int instanceTag; UCHAR fParametricStereo; UCHAR fDualMono; /**< This flags allows to disable coupling in sbr channel pair element */ UCHAR nChannelsInEl; UCHAR ChannelIndex[2]; } SBR_ELEMENT_INFO; #ifdef __cplusplus extern "C" { #endif typedef struct SBR_ENCODER *HANDLE_SBR_ENCODER; /** * \brief Get the max required input buffer size including delay balancing * space for N audio channels. * \param noChannels Number of audio channels. * \return Max required input buffer size in bytes. */ INT sbrEncoder_GetInBufferSize(int noChannels); INT sbrEncoder_Open(HANDLE_SBR_ENCODER *phSbrEncoder, INT nElements, INT nChannels, INT supportPS); /** * \brief Get closest working bitrate to specified desired * bitrate for a single SBR element. * \param bitRate The desired target bit rate * \param numChannels The amount of audio channels * \param coreSampleRate The sample rate of the core coder * \param aot The current Audio Object Type * \return Closest working bit rate to bitRate value */ UINT sbrEncoder_LimitBitRate(UINT bitRate, UINT numChannels, UINT coreSampleRate, AUDIO_OBJECT_TYPE aot); /** * \brief Check whether downsampled SBR single rate is possible * with given audio object type. * \param aot The Audio object type. * \return 0 when downsampled SBR is not possible, * 1 when downsampled SBR is possible. */ UINT sbrEncoder_IsSingleRatePossible(AUDIO_OBJECT_TYPE aot); /** * \brief Initialize SBR Encoder instance. * \param phSbrEncoder Pointer to a SBR Encoder instance. * \param elInfo Structure that describes the element/channel * arrangement. * \param noElements Amount of elements described in elInfo. * \param inputBuffer Pointer to the encoder audio buffer * \param inputBufferBufSize Buffer offset of one channel (frameSize + delay) * \param bandwidth Returns the core audio encoder bandwidth (output) * \param bufferOffset Returns the offset for the audio input data in order * to do delay balancing. * \param numChannels Input: Encoder input channels. output: core encoder * channels. * \param sampleRate Input: Encoder samplerate. output core encoder * samplerate. * \param downSampleFactor Input: Relation between SBR and core coder sampling * rate; * \param frameLength Input: Encoder frameLength. output core encoder * frameLength. * \param aot Input: AOT.. * \param delay Input: core encoder delay. Output: total delay * because of SBR. * \param transformFactor The core encoder transform factor (blockswitching). * \param headerPeriod Repetition rate of the SBR header: * - (-1) means intern configuration. * - (1-10) corresponds to header repetition rate in * frames. * \return 0 on success, and non-zero if failed. */ INT sbrEncoder_Init(HANDLE_SBR_ENCODER hSbrEncoder, SBR_ELEMENT_INFO elInfo[(8)], int noElements, INT_PCM *inputBuffer, UINT inputBufferBufSize, INT *coreBandwidth, INT *inputBufferOffset, INT *numChannels, const UINT syntaxFlags, INT *sampleRate, UINT *downSampleFactor, INT *frameLength, AUDIO_OBJECT_TYPE aot, int *delay, int transformFactor, const int headerPeriod, ULONG statesInitFlag); /** * \brief Do delay line buffers housekeeping. To be called after * each encoded audio frame. * \param hEnvEnc SBR Encoder handle. * \param timeBuffer Pointer to the encoder audio buffer. * \param timeBufferBufSIze buffer size for one channel * \return 0 on success, and non-zero if failed. */ INT sbrEncoder_UpdateBuffers(HANDLE_SBR_ENCODER hEnvEnc, INT_PCM *timeBuffer, UINT timeBufferBufSIze); /** * \brief Close SBR encoder instance. * \param phEbrEncoder Handle of SBR encoder instance to be closed. * \return void */ void sbrEncoder_Close(HANDLE_SBR_ENCODER *phEbrEncoder); /** * \brief Encode SBR data of one complete audio frame. * \param hEnvEncoder Handle of SBR encoder instance. * \param samples Time samples, not interleaved. * \param timeInStride Channel offset of samples buffer. * \param sbrDataBits Size of SBR payload in bits. * \param sbrData SBR payload. * \return 0 on success, and non-zero if failed. */ INT sbrEncoder_EncodeFrame(HANDLE_SBR_ENCODER hEnvEncoder, INT_PCM *samples, UINT samplesBufSize, UINT sbrDataBits[(8)], UCHAR sbrData[(8)][MAX_PAYLOAD_SIZE]); /** * \brief Write SBR headers of one SBR element. * \param sbrEncoder Handle of the SBR encoder instance. * \param hBs Handle of bit stream handle to write SBR header to. * \param element_index Index of the SBR element which header should be written. * \param fSendHeaders Flag indicating that the SBR encoder should send more * headers in the SBR payload or not. * \return void */ void sbrEncoder_GetHeader(HANDLE_SBR_ENCODER sbrEncoder, HANDLE_FDK_BITSTREAM hBs, INT element_index, int fSendHeaders); /** * \brief Request to write SBR header. * \param hSbrEncoder SBR encoder handle. * \return 0 on success, and non-zero if failed. */ INT sbrEncoder_SendHeader(HANDLE_SBR_ENCODER hSbrEncoder); /** * \brief Request if last sbr payload contains an SBR header. * \param hSbrEncoder SBR encoder handle. * \return 1 contains sbr header, 0 without sbr header. */ INT sbrEncoder_ContainsHeader(HANDLE_SBR_ENCODER hSbrEncoder); /** * \brief SBR header delay in frames. * \param hSbrEncoder SBR encoder handle. * \return Delay in frames, -1 on failure. */ INT sbrEncoder_GetHeaderDelay(HANDLE_SBR_ENCODER hSbrEncoder); /** * \brief Bitstrem delay in SBR frames. * \param hSbrEncoder SBR encoder handle. * \return Delay in frames, -1 on failure. */ INT sbrEncoder_GetBsDelay(HANDLE_SBR_ENCODER hSbrEncoder); /** * \brief Prepare SBR payload for SAP. * \param hSbrEncoder SBR encoder handle. * \return 0 on success, and non-zero if failed. */ INT sbrEncoder_SAPPrepare(HANDLE_SBR_ENCODER hSbrEncoder); /** * \brief SBR encoder bitrate estimation. * \param hSbrEncoder SBR encoder handle. * \return Estimated bitrate. */ INT sbrEncoder_GetEstimateBitrate(HANDLE_SBR_ENCODER hSbrEncoder); /** * \brief Delay between input data and downsampled output data. * \param hSbrEncoder SBR encoder handle. * \return Delay. */ INT sbrEncoder_GetInputDataDelay(HANDLE_SBR_ENCODER hSbrEncoder); /** * \brief Delay caused by the SBR decoder. * \param hSbrEncoder SBR encoder handle. * \return Delay. */ INT sbrEncoder_GetSbrDecDelay(HANDLE_SBR_ENCODER hSbrEncoder); /** * \brief Get decoder library version info. * \param info Pointer to an allocated LIB_INFO struct, where library info is * written to. * \return 0 on sucess. */ INT sbrEncoder_GetLibInfo(LIB_INFO *info); void sbrPrintRAM(void); void sbrPrintROM(void); #ifdef __cplusplus } #endif #endif /* ifndef __SBR_MAIN_H */ fdk-aac-2.0.2/libSBRenc/src/000077500000000000000000000000001404225310700153715ustar00rootroot00000000000000fdk-aac-2.0.2/libSBRenc/src/bit_sbr.cpp000066400000000000000000001173011404225310700175240ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief SBR bit writing routines $Revision: 93300 $ */ #include "bit_sbr.h" #include "code_env.h" #include "cmondata.h" #include "sbr.h" #include "ps_main.h" typedef enum { SBR_ID_SCE = 1, SBR_ID_CPE } SBR_ELEMENT_TYPE; static INT encodeSbrData(HANDLE_SBR_ENV_DATA sbrEnvDataLeft, HANDLE_SBR_ENV_DATA sbrEnvDataRight, HANDLE_PARAMETRIC_STEREO hParametricStereo, HANDLE_COMMON_DATA cmonData, SBR_ELEMENT_TYPE sbrElem, INT coupling, UINT sbrSyntaxFlags); static INT encodeSbrHeader(HANDLE_SBR_HEADER_DATA sbrHeaderData, HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, HANDLE_COMMON_DATA cmonData); static INT encodeSbrHeaderData(HANDLE_SBR_HEADER_DATA sbrHeaderData, HANDLE_FDK_BITSTREAM hBitStream); static INT encodeSbrSingleChannelElement( HANDLE_SBR_ENV_DATA sbrEnvData, HANDLE_FDK_BITSTREAM hBitStream, HANDLE_PARAMETRIC_STEREO hParametricStereo, const UINT sbrSyntaxFlags); static INT encodeSbrChannelPairElement( HANDLE_SBR_ENV_DATA sbrEnvDataLeft, HANDLE_SBR_ENV_DATA sbrEnvDataRight, HANDLE_PARAMETRIC_STEREO hParametricStereo, HANDLE_FDK_BITSTREAM hBitStream, const INT coupling, const UINT sbrSyntaxFlags); static INT encodeSbrGrid(HANDLE_SBR_ENV_DATA sbrEnvData, HANDLE_FDK_BITSTREAM hBitStream); static int encodeLowDelaySbrGrid(HANDLE_SBR_ENV_DATA sbrEnvData, HANDLE_FDK_BITSTREAM hBitStream, const int transmitFreqs, const UINT sbrSyntaxFlags); static INT encodeSbrDtdf(HANDLE_SBR_ENV_DATA sbrEnvData, HANDLE_FDK_BITSTREAM hBitStream); static INT writeNoiseLevelData(HANDLE_SBR_ENV_DATA sbrEnvData, HANDLE_FDK_BITSTREAM hBitStream, INT coupling); static INT writeEnvelopeData(HANDLE_SBR_ENV_DATA sbrEnvData, HANDLE_FDK_BITSTREAM hBitStream, INT coupling); static INT writeSyntheticCodingData(HANDLE_SBR_ENV_DATA sbrEnvData, HANDLE_FDK_BITSTREAM hBitStream); static INT encodeExtendedData(HANDLE_PARAMETRIC_STEREO hParametricStereo, HANDLE_FDK_BITSTREAM hBitStream); static INT getSbrExtendedDataSize(HANDLE_PARAMETRIC_STEREO hParametricStereo); /***************************************************************************** functionname: FDKsbrEnc_WriteEnvSingleChannelElement description: writes pure SBR single channel data element returns: number of bits written input: output: *****************************************************************************/ INT FDKsbrEnc_WriteEnvSingleChannelElement( HANDLE_SBR_HEADER_DATA sbrHeaderData, HANDLE_PARAMETRIC_STEREO hParametricStereo, HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, HANDLE_SBR_ENV_DATA sbrEnvData, HANDLE_COMMON_DATA cmonData, UINT sbrSyntaxFlags) { INT payloadBits = 0; cmonData->sbrHdrBits = 0; cmonData->sbrDataBits = 0; /* write pure sbr data */ if (sbrEnvData != NULL) { /* write header */ payloadBits += encodeSbrHeader(sbrHeaderData, sbrBitstreamData, cmonData); /* write data */ payloadBits += encodeSbrData(sbrEnvData, NULL, hParametricStereo, cmonData, SBR_ID_SCE, 0, sbrSyntaxFlags); } return payloadBits; } /***************************************************************************** functionname: FDKsbrEnc_WriteEnvChannelPairElement description: writes pure SBR channel pair data element returns: number of bits written input: output: *****************************************************************************/ INT FDKsbrEnc_WriteEnvChannelPairElement( HANDLE_SBR_HEADER_DATA sbrHeaderData, HANDLE_PARAMETRIC_STEREO hParametricStereo, HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, HANDLE_SBR_ENV_DATA sbrEnvDataLeft, HANDLE_SBR_ENV_DATA sbrEnvDataRight, HANDLE_COMMON_DATA cmonData, UINT sbrSyntaxFlags) { INT payloadBits = 0; cmonData->sbrHdrBits = 0; cmonData->sbrDataBits = 0; /* write pure sbr data */ if ((sbrEnvDataLeft != NULL) && (sbrEnvDataRight != NULL)) { /* write header */ payloadBits += encodeSbrHeader(sbrHeaderData, sbrBitstreamData, cmonData); /* write data */ payloadBits += encodeSbrData(sbrEnvDataLeft, sbrEnvDataRight, hParametricStereo, cmonData, SBR_ID_CPE, sbrHeaderData->coupling, sbrSyntaxFlags); } return payloadBits; } INT FDKsbrEnc_CountSbrChannelPairElement( HANDLE_SBR_HEADER_DATA sbrHeaderData, HANDLE_PARAMETRIC_STEREO hParametricStereo, HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, HANDLE_SBR_ENV_DATA sbrEnvDataLeft, HANDLE_SBR_ENV_DATA sbrEnvDataRight, HANDLE_COMMON_DATA cmonData, UINT sbrSyntaxFlags) { INT payloadBits; INT bitPos = FDKgetValidBits(&cmonData->sbrBitbuf); payloadBits = FDKsbrEnc_WriteEnvChannelPairElement( sbrHeaderData, hParametricStereo, sbrBitstreamData, sbrEnvDataLeft, sbrEnvDataRight, cmonData, sbrSyntaxFlags); FDKpushBack(&cmonData->sbrBitbuf, (FDKgetValidBits(&cmonData->sbrBitbuf) - bitPos)); return payloadBits; } void sbrEncoder_GetHeader(SBR_ENCODER *sbrEncoder, HANDLE_FDK_BITSTREAM hBs, INT element_index, int fSendHeaders) { encodeSbrHeaderData(&sbrEncoder->sbrElement[element_index]->sbrHeaderData, hBs); if (fSendHeaders == 0) { /* Prevent header being embedded into the SBR payload. */ sbrEncoder->sbrElement[element_index]->sbrBitstreamData.NrSendHeaderData = -1; sbrEncoder->sbrElement[element_index]->sbrBitstreamData.HeaderActive = 0; sbrEncoder->sbrElement[element_index] ->sbrBitstreamData.CountSendHeaderData = -1; } } /***************************************************************************** functionname: encodeSbrHeader description: encodes SBR Header information returns: number of bits written input: output: *****************************************************************************/ static INT encodeSbrHeader(HANDLE_SBR_HEADER_DATA sbrHeaderData, HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, HANDLE_COMMON_DATA cmonData) { INT payloadBits = 0; if (sbrBitstreamData->HeaderActive) { payloadBits += FDKwriteBits(&cmonData->sbrBitbuf, 1, 1); payloadBits += encodeSbrHeaderData(sbrHeaderData, &cmonData->sbrBitbuf); } else { payloadBits += FDKwriteBits(&cmonData->sbrBitbuf, 0, 1); } cmonData->sbrHdrBits = payloadBits; return payloadBits; } /***************************************************************************** functionname: encodeSbrHeaderData description: writes sbr_header() bs_protocol_version through bs_header_extra_2 returns: number of bits written input: output: *****************************************************************************/ static INT encodeSbrHeaderData(HANDLE_SBR_HEADER_DATA sbrHeaderData, HANDLE_FDK_BITSTREAM hBitStream) { INT payloadBits = 0; if (sbrHeaderData != NULL) { payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->sbr_amp_res, SI_SBR_AMP_RES_BITS); payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->sbr_start_frequency, SI_SBR_START_FREQ_BITS); payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->sbr_stop_frequency, SI_SBR_STOP_FREQ_BITS); payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->sbr_xover_band, SI_SBR_XOVER_BAND_BITS); payloadBits += FDKwriteBits(hBitStream, 0, SI_SBR_RESERVED_BITS); payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->header_extra_1, SI_SBR_HEADER_EXTRA_1_BITS); payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->header_extra_2, SI_SBR_HEADER_EXTRA_2_BITS); if (sbrHeaderData->header_extra_1) { payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->freqScale, SI_SBR_FREQ_SCALE_BITS); payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->alterScale, SI_SBR_ALTER_SCALE_BITS); payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->sbr_noise_bands, SI_SBR_NOISE_BANDS_BITS); } /* sbrHeaderData->header_extra_1 */ if (sbrHeaderData->header_extra_2) { payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->sbr_limiter_bands, SI_SBR_LIMITER_BANDS_BITS); payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->sbr_limiter_gains, SI_SBR_LIMITER_GAINS_BITS); payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->sbr_interpol_freq, SI_SBR_INTERPOL_FREQ_BITS); payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->sbr_smoothing_length, SI_SBR_SMOOTHING_LENGTH_BITS); } /* sbrHeaderData->header_extra_2 */ } /* sbrHeaderData != NULL */ return payloadBits; } /***************************************************************************** functionname: encodeSbrData description: encodes sbr Data information returns: number of bits written input: output: *****************************************************************************/ static INT encodeSbrData(HANDLE_SBR_ENV_DATA sbrEnvDataLeft, HANDLE_SBR_ENV_DATA sbrEnvDataRight, HANDLE_PARAMETRIC_STEREO hParametricStereo, HANDLE_COMMON_DATA cmonData, SBR_ELEMENT_TYPE sbrElem, INT coupling, UINT sbrSyntaxFlags) { INT payloadBits = 0; switch (sbrElem) { case SBR_ID_SCE: payloadBits += encodeSbrSingleChannelElement(sbrEnvDataLeft, &cmonData->sbrBitbuf, hParametricStereo, sbrSyntaxFlags); break; case SBR_ID_CPE: payloadBits += encodeSbrChannelPairElement( sbrEnvDataLeft, sbrEnvDataRight, hParametricStereo, &cmonData->sbrBitbuf, coupling, sbrSyntaxFlags); break; default: /* we never should apply SBR to any other element type */ FDK_ASSERT(0); } cmonData->sbrDataBits = payloadBits; return payloadBits; } #define MODE_FREQ_TANS 1 #define MODE_NO_FREQ_TRAN 0 #define LD_TRANSMISSION MODE_FREQ_TANS static int encodeFreqs(int mode) { return ((mode & MODE_FREQ_TANS) ? 1 : 0); } /***************************************************************************** functionname: encodeSbrSingleChannelElement description: encodes sbr SCE information returns: number of bits written input: output: *****************************************************************************/ static INT encodeSbrSingleChannelElement( HANDLE_SBR_ENV_DATA sbrEnvData, HANDLE_FDK_BITSTREAM hBitStream, HANDLE_PARAMETRIC_STEREO hParametricStereo, const UINT sbrSyntaxFlags) { INT i, payloadBits = 0; payloadBits += FDKwriteBits(hBitStream, 0, SI_SBR_DATA_EXTRA_BITS); /* no reserved bits */ if (sbrEnvData->ldGrid) { if (sbrEnvData->hSbrBSGrid->frameClass != FIXFIXonly) { /* encode normal SbrGrid */ payloadBits += encodeSbrGrid(sbrEnvData, hBitStream); } else { /* use FIXFIXonly frame Grid */ payloadBits += encodeLowDelaySbrGrid( sbrEnvData, hBitStream, encodeFreqs(LD_TRANSMISSION), sbrSyntaxFlags); } } else { if (sbrSyntaxFlags & SBR_SYNTAX_SCALABLE) { payloadBits += FDKwriteBits(hBitStream, 1, SI_SBR_COUPLING_BITS); } payloadBits += encodeSbrGrid(sbrEnvData, hBitStream); } payloadBits += encodeSbrDtdf(sbrEnvData, hBitStream); for (i = 0; i < sbrEnvData->noOfnoisebands; i++) { payloadBits += FDKwriteBits(hBitStream, sbrEnvData->sbr_invf_mode_vec[i], SI_SBR_INVF_MODE_BITS); } payloadBits += writeEnvelopeData(sbrEnvData, hBitStream, 0); payloadBits += writeNoiseLevelData(sbrEnvData, hBitStream, 0); payloadBits += writeSyntheticCodingData(sbrEnvData, hBitStream); payloadBits += encodeExtendedData(hParametricStereo, hBitStream); return payloadBits; } /***************************************************************************** functionname: encodeSbrChannelPairElement description: encodes sbr CPE information returns: input: output: *****************************************************************************/ static INT encodeSbrChannelPairElement( HANDLE_SBR_ENV_DATA sbrEnvDataLeft, HANDLE_SBR_ENV_DATA sbrEnvDataRight, HANDLE_PARAMETRIC_STEREO hParametricStereo, HANDLE_FDK_BITSTREAM hBitStream, const INT coupling, const UINT sbrSyntaxFlags) { INT payloadBits = 0; INT i = 0; payloadBits += FDKwriteBits(hBitStream, 0, SI_SBR_DATA_EXTRA_BITS); /* no reserved bits */ payloadBits += FDKwriteBits(hBitStream, coupling, SI_SBR_COUPLING_BITS); if (coupling) { if (sbrEnvDataLeft->ldGrid) { if (sbrEnvDataLeft->hSbrBSGrid->frameClass != FIXFIXonly) { /* normal SbrGrid */ payloadBits += encodeSbrGrid(sbrEnvDataLeft, hBitStream); } else { /* FIXFIXonly frame Grid */ payloadBits += encodeLowDelaySbrGrid(sbrEnvDataLeft, hBitStream, encodeFreqs(LD_TRANSMISSION), sbrSyntaxFlags); } } else payloadBits += encodeSbrGrid(sbrEnvDataLeft, hBitStream); payloadBits += encodeSbrDtdf(sbrEnvDataLeft, hBitStream); payloadBits += encodeSbrDtdf(sbrEnvDataRight, hBitStream); for (i = 0; i < sbrEnvDataLeft->noOfnoisebands; i++) { payloadBits += FDKwriteBits(hBitStream, sbrEnvDataLeft->sbr_invf_mode_vec[i], SI_SBR_INVF_MODE_BITS); } payloadBits += writeEnvelopeData(sbrEnvDataLeft, hBitStream, 1); payloadBits += writeNoiseLevelData(sbrEnvDataLeft, hBitStream, 1); payloadBits += writeEnvelopeData(sbrEnvDataRight, hBitStream, 1); payloadBits += writeNoiseLevelData(sbrEnvDataRight, hBitStream, 1); payloadBits += writeSyntheticCodingData(sbrEnvDataLeft, hBitStream); payloadBits += writeSyntheticCodingData(sbrEnvDataRight, hBitStream); } else { /* no coupling */ FDK_ASSERT(sbrEnvDataLeft->ldGrid == sbrEnvDataRight->ldGrid); if (sbrEnvDataLeft->ldGrid || sbrEnvDataRight->ldGrid) { /* sbrEnvDataLeft (left channel) */ if (sbrEnvDataLeft->hSbrBSGrid->frameClass != FIXFIXonly) { /* no FIXFIXonly Frame so we dont need encodeLowDelaySbrGrid */ /* normal SbrGrid */ payloadBits += encodeSbrGrid(sbrEnvDataLeft, hBitStream); } else { /* FIXFIXonly frame Grid */ payloadBits += encodeLowDelaySbrGrid(sbrEnvDataLeft, hBitStream, encodeFreqs(LD_TRANSMISSION), sbrSyntaxFlags); } /* sbrEnvDataRight (right channel) */ if (sbrEnvDataRight->hSbrBSGrid->frameClass != FIXFIXonly) { /* no FIXFIXonly Frame so we dont need encodeLowDelaySbrGrid */ /* normal SbrGrid */ payloadBits += encodeSbrGrid(sbrEnvDataRight, hBitStream); } else { /* FIXFIXonly frame Grid */ payloadBits += encodeLowDelaySbrGrid(sbrEnvDataRight, hBitStream, encodeFreqs(LD_TRANSMISSION), sbrSyntaxFlags); } } else { payloadBits += encodeSbrGrid(sbrEnvDataLeft, hBitStream); payloadBits += encodeSbrGrid(sbrEnvDataRight, hBitStream); } payloadBits += encodeSbrDtdf(sbrEnvDataLeft, hBitStream); payloadBits += encodeSbrDtdf(sbrEnvDataRight, hBitStream); for (i = 0; i < sbrEnvDataLeft->noOfnoisebands; i++) { payloadBits += FDKwriteBits(hBitStream, sbrEnvDataLeft->sbr_invf_mode_vec[i], SI_SBR_INVF_MODE_BITS); } for (i = 0; i < sbrEnvDataRight->noOfnoisebands; i++) { payloadBits += FDKwriteBits(hBitStream, sbrEnvDataRight->sbr_invf_mode_vec[i], SI_SBR_INVF_MODE_BITS); } payloadBits += writeEnvelopeData(sbrEnvDataLeft, hBitStream, 0); payloadBits += writeEnvelopeData(sbrEnvDataRight, hBitStream, 0); payloadBits += writeNoiseLevelData(sbrEnvDataLeft, hBitStream, 0); payloadBits += writeNoiseLevelData(sbrEnvDataRight, hBitStream, 0); payloadBits += writeSyntheticCodingData(sbrEnvDataLeft, hBitStream); payloadBits += writeSyntheticCodingData(sbrEnvDataRight, hBitStream); } /* coupling */ payloadBits += encodeExtendedData(hParametricStereo, hBitStream); return payloadBits; } static INT ceil_ln2(INT x) { INT tmp = -1; while ((1 << ++tmp) < x) ; return (tmp); } /***************************************************************************** functionname: encodeSbrGrid description: if hBitStream != NULL writes bits that describes the time/frequency grouping of a frame; else counts them only returns: number of bits written or counted input: output: *****************************************************************************/ static INT encodeSbrGrid(HANDLE_SBR_ENV_DATA sbrEnvData, HANDLE_FDK_BITSTREAM hBitStream) { INT payloadBits = 0; INT i, temp; INT bufferFrameStart = sbrEnvData->hSbrBSGrid->bufferFrameStart; INT numberTimeSlots = sbrEnvData->hSbrBSGrid->numberTimeSlots; if (sbrEnvData->ldGrid) payloadBits += FDKwriteBits(hBitStream, sbrEnvData->hSbrBSGrid->frameClass, SBR_CLA_BITS_LD); else payloadBits += FDKwriteBits(hBitStream, sbrEnvData->hSbrBSGrid->frameClass, SBR_CLA_BITS); switch (sbrEnvData->hSbrBSGrid->frameClass) { case FIXFIXonly: FDK_ASSERT(0 /* Fatal error in encodeSbrGrid! */); break; case FIXFIX: temp = ceil_ln2(sbrEnvData->hSbrBSGrid->bs_num_env); payloadBits += FDKwriteBits(hBitStream, temp, SBR_ENV_BITS); if ((sbrEnvData->ldGrid) && (sbrEnvData->hSbrBSGrid->bs_num_env == 1)) payloadBits += FDKwriteBits(hBitStream, sbrEnvData->currentAmpResFF, SI_SBR_AMP_RES_BITS); payloadBits += FDKwriteBits(hBitStream, sbrEnvData->hSbrBSGrid->v_f[0], SBR_RES_BITS); break; case FIXVAR: case VARFIX: if (sbrEnvData->hSbrBSGrid->frameClass == FIXVAR) temp = sbrEnvData->hSbrBSGrid->bs_abs_bord - (bufferFrameStart + numberTimeSlots); else temp = sbrEnvData->hSbrBSGrid->bs_abs_bord - bufferFrameStart; payloadBits += FDKwriteBits(hBitStream, temp, SBR_ABS_BITS); payloadBits += FDKwriteBits(hBitStream, sbrEnvData->hSbrBSGrid->n, SBR_NUM_BITS); for (i = 0; i < sbrEnvData->hSbrBSGrid->n; i++) { temp = (sbrEnvData->hSbrBSGrid->bs_rel_bord[i] - 2) >> 1; payloadBits += FDKwriteBits(hBitStream, temp, SBR_REL_BITS); } temp = ceil_ln2(sbrEnvData->hSbrBSGrid->n + 2); payloadBits += FDKwriteBits(hBitStream, sbrEnvData->hSbrBSGrid->p, temp); for (i = 0; i < sbrEnvData->hSbrBSGrid->n + 1; i++) { payloadBits += FDKwriteBits(hBitStream, sbrEnvData->hSbrBSGrid->v_f[i], SBR_RES_BITS); } break; case VARVAR: temp = sbrEnvData->hSbrBSGrid->bs_abs_bord_0 - bufferFrameStart; payloadBits += FDKwriteBits(hBitStream, temp, SBR_ABS_BITS); temp = sbrEnvData->hSbrBSGrid->bs_abs_bord_1 - (bufferFrameStart + numberTimeSlots); payloadBits += FDKwriteBits(hBitStream, temp, SBR_ABS_BITS); payloadBits += FDKwriteBits( hBitStream, sbrEnvData->hSbrBSGrid->bs_num_rel_0, SBR_NUM_BITS); payloadBits += FDKwriteBits( hBitStream, sbrEnvData->hSbrBSGrid->bs_num_rel_1, SBR_NUM_BITS); for (i = 0; i < sbrEnvData->hSbrBSGrid->bs_num_rel_0; i++) { temp = (sbrEnvData->hSbrBSGrid->bs_rel_bord_0[i] - 2) >> 1; payloadBits += FDKwriteBits(hBitStream, temp, SBR_REL_BITS); } for (i = 0; i < sbrEnvData->hSbrBSGrid->bs_num_rel_1; i++) { temp = (sbrEnvData->hSbrBSGrid->bs_rel_bord_1[i] - 2) >> 1; payloadBits += FDKwriteBits(hBitStream, temp, SBR_REL_BITS); } temp = ceil_ln2(sbrEnvData->hSbrBSGrid->bs_num_rel_0 + sbrEnvData->hSbrBSGrid->bs_num_rel_1 + 2); payloadBits += FDKwriteBits(hBitStream, sbrEnvData->hSbrBSGrid->p, temp); temp = sbrEnvData->hSbrBSGrid->bs_num_rel_0 + sbrEnvData->hSbrBSGrid->bs_num_rel_1 + 1; for (i = 0; i < temp; i++) { payloadBits += FDKwriteBits( hBitStream, sbrEnvData->hSbrBSGrid->v_fLR[i], SBR_RES_BITS); } break; } return payloadBits; } #define SBR_CLA_BITS_LD 1 /***************************************************************************** functionname: encodeLowDelaySbrGrid description: if hBitStream != NULL writes bits that describes the time/frequency grouping of a frame; else counts them only (this function only write the FIXFIXonly Bitstream data) returns: number of bits written or counted input: output: *****************************************************************************/ static int encodeLowDelaySbrGrid(HANDLE_SBR_ENV_DATA sbrEnvData, HANDLE_FDK_BITSTREAM hBitStream, const int transmitFreqs, const UINT sbrSyntaxFlags) { int payloadBits = 0; int i; /* write FIXFIXonly Grid */ /* write frameClass [1 bit] for FIXFIXonly Grid */ payloadBits += FDKwriteBits(hBitStream, 1, SBR_CLA_BITS_LD); /* absolute Borders are fix: 0,X,X,X,nTimeSlots; so we dont have to transmit * them */ /* only transmit the transient position! */ /* with this info (b1) we can reconstruct the Frame on Decoder side : */ /* border[0] = 0; border[1] = b1; border[2]=b1+2; border[3] = nrTimeSlots */ /* use 3 or 4bits for transient border (border) */ if (sbrEnvData->hSbrBSGrid->numberTimeSlots == 8) payloadBits += FDKwriteBits(hBitStream, sbrEnvData->hSbrBSGrid->bs_abs_bord, 3); else payloadBits += FDKwriteBits(hBitStream, sbrEnvData->hSbrBSGrid->bs_abs_bord, 4); if (transmitFreqs) { /* write FreqRes grid */ for (i = 0; i < sbrEnvData->hSbrBSGrid->bs_num_env; i++) { payloadBits += FDKwriteBits(hBitStream, sbrEnvData->hSbrBSGrid->v_f[i], SBR_RES_BITS); } } return payloadBits; } /***************************************************************************** functionname: encodeSbrDtdf description: writes bits that describes the direction of the envelopes of a frame returns: number of bits written input: output: *****************************************************************************/ static INT encodeSbrDtdf(HANDLE_SBR_ENV_DATA sbrEnvData, HANDLE_FDK_BITSTREAM hBitStream) { INT i, payloadBits = 0, noOfNoiseEnvelopes; noOfNoiseEnvelopes = sbrEnvData->noOfEnvelopes > 1 ? 2 : 1; for (i = 0; i < sbrEnvData->noOfEnvelopes; ++i) { payloadBits += FDKwriteBits(hBitStream, sbrEnvData->domain_vec[i], SBR_DIR_BITS); } for (i = 0; i < noOfNoiseEnvelopes; ++i) { payloadBits += FDKwriteBits(hBitStream, sbrEnvData->domain_vec_noise[i], SBR_DIR_BITS); } return payloadBits; } /***************************************************************************** functionname: writeNoiseLevelData description: writes bits corresponding to the noise-floor-level returns: number of bits written input: output: *****************************************************************************/ static INT writeNoiseLevelData(HANDLE_SBR_ENV_DATA sbrEnvData, HANDLE_FDK_BITSTREAM hBitStream, INT coupling) { INT j, i, payloadBits = 0; INT nNoiseEnvelopes = sbrEnvData->noOfEnvelopes > 1 ? 2 : 1; for (i = 0; i < nNoiseEnvelopes; i++) { switch (sbrEnvData->domain_vec_noise[i]) { case FREQ: if (coupling && sbrEnvData->balance) { payloadBits += FDKwriteBits( hBitStream, sbrEnvData->sbr_noise_levels[i * sbrEnvData->noOfnoisebands], sbrEnvData->si_sbr_start_noise_bits_balance); } else { payloadBits += FDKwriteBits( hBitStream, sbrEnvData->sbr_noise_levels[i * sbrEnvData->noOfnoisebands], sbrEnvData->si_sbr_start_noise_bits); } for (j = 1 + i * sbrEnvData->noOfnoisebands; j < (sbrEnvData->noOfnoisebands * (1 + i)); j++) { if (coupling) { if (sbrEnvData->balance) { /* coupling && balance */ payloadBits += FDKwriteBits(hBitStream, sbrEnvData->hufftableNoiseBalanceFreqC [sbrEnvData->sbr_noise_levels[j] + CODE_BOOK_SCF_LAV_BALANCE11], sbrEnvData->hufftableNoiseBalanceFreqL [sbrEnvData->sbr_noise_levels[j] + CODE_BOOK_SCF_LAV_BALANCE11]); } else { /* coupling && !balance */ payloadBits += FDKwriteBits( hBitStream, sbrEnvData->hufftableNoiseLevelFreqC [sbrEnvData->sbr_noise_levels[j] + CODE_BOOK_SCF_LAV11], sbrEnvData->hufftableNoiseLevelFreqL [sbrEnvData->sbr_noise_levels[j] + CODE_BOOK_SCF_LAV11]); } } else { /* !coupling */ payloadBits += FDKwriteBits( hBitStream, sbrEnvData ->hufftableNoiseFreqC[sbrEnvData->sbr_noise_levels[j] + CODE_BOOK_SCF_LAV11], sbrEnvData ->hufftableNoiseFreqL[sbrEnvData->sbr_noise_levels[j] + CODE_BOOK_SCF_LAV11]); } } break; case TIME: for (j = i * sbrEnvData->noOfnoisebands; j < (sbrEnvData->noOfnoisebands * (1 + i)); j++) { if (coupling) { if (sbrEnvData->balance) { /* coupling && balance */ payloadBits += FDKwriteBits(hBitStream, sbrEnvData->hufftableNoiseBalanceTimeC [sbrEnvData->sbr_noise_levels[j] + CODE_BOOK_SCF_LAV_BALANCE11], sbrEnvData->hufftableNoiseBalanceTimeL [sbrEnvData->sbr_noise_levels[j] + CODE_BOOK_SCF_LAV_BALANCE11]); } else { /* coupling && !balance */ payloadBits += FDKwriteBits( hBitStream, sbrEnvData->hufftableNoiseLevelTimeC [sbrEnvData->sbr_noise_levels[j] + CODE_BOOK_SCF_LAV11], sbrEnvData->hufftableNoiseLevelTimeL [sbrEnvData->sbr_noise_levels[j] + CODE_BOOK_SCF_LAV11]); } } else { /* !coupling */ payloadBits += FDKwriteBits( hBitStream, sbrEnvData ->hufftableNoiseLevelTimeC[sbrEnvData->sbr_noise_levels[j] + CODE_BOOK_SCF_LAV11], sbrEnvData ->hufftableNoiseLevelTimeL[sbrEnvData->sbr_noise_levels[j] + CODE_BOOK_SCF_LAV11]); } } break; } } return payloadBits; } /***************************************************************************** functionname: writeEnvelopeData description: writes bits corresponding to the envelope returns: number of bits written input: output: *****************************************************************************/ static INT writeEnvelopeData(HANDLE_SBR_ENV_DATA sbrEnvData, HANDLE_FDK_BITSTREAM hBitStream, INT coupling) { INT payloadBits = 0, j, i, delta; for (j = 0; j < sbrEnvData->noOfEnvelopes; j++) { /* loop over all envelopes */ if (sbrEnvData->domain_vec[j] == FREQ) { if (coupling && sbrEnvData->balance) { payloadBits += FDKwriteBits(hBitStream, sbrEnvData->ienvelope[j][0], sbrEnvData->si_sbr_start_env_bits_balance); } else { payloadBits += FDKwriteBits(hBitStream, sbrEnvData->ienvelope[j][0], sbrEnvData->si_sbr_start_env_bits); } } for (i = 1 - sbrEnvData->domain_vec[j]; i < sbrEnvData->noScfBands[j]; i++) { delta = sbrEnvData->ienvelope[j][i]; if (coupling && sbrEnvData->balance) { FDK_ASSERT(fixp_abs(delta) <= sbrEnvData->codeBookScfLavBalance); } else { FDK_ASSERT(fixp_abs(delta) <= sbrEnvData->codeBookScfLav); } if (coupling) { if (sbrEnvData->balance) { if (sbrEnvData->domain_vec[j]) { /* coupling && balance && TIME */ payloadBits += FDKwriteBits( hBitStream, sbrEnvData ->hufftableBalanceTimeC[delta + sbrEnvData->codeBookScfLavBalance], sbrEnvData ->hufftableBalanceTimeL[delta + sbrEnvData->codeBookScfLavBalance]); } else { /* coupling && balance && FREQ */ payloadBits += FDKwriteBits( hBitStream, sbrEnvData ->hufftableBalanceFreqC[delta + sbrEnvData->codeBookScfLavBalance], sbrEnvData ->hufftableBalanceFreqL[delta + sbrEnvData->codeBookScfLavBalance]); } } else { if (sbrEnvData->domain_vec[j]) { /* coupling && !balance && TIME */ payloadBits += FDKwriteBits( hBitStream, sbrEnvData ->hufftableLevelTimeC[delta + sbrEnvData->codeBookScfLav], sbrEnvData ->hufftableLevelTimeL[delta + sbrEnvData->codeBookScfLav]); } else { /* coupling && !balance && FREQ */ payloadBits += FDKwriteBits( hBitStream, sbrEnvData ->hufftableLevelFreqC[delta + sbrEnvData->codeBookScfLav], sbrEnvData ->hufftableLevelFreqL[delta + sbrEnvData->codeBookScfLav]); } } } else { if (sbrEnvData->domain_vec[j]) { /* !coupling && TIME */ payloadBits += FDKwriteBits( hBitStream, sbrEnvData->hufftableTimeC[delta + sbrEnvData->codeBookScfLav], sbrEnvData->hufftableTimeL[delta + sbrEnvData->codeBookScfLav]); } else { /* !coupling && FREQ */ payloadBits += FDKwriteBits( hBitStream, sbrEnvData->hufftableFreqC[delta + sbrEnvData->codeBookScfLav], sbrEnvData->hufftableFreqL[delta + sbrEnvData->codeBookScfLav]); } } } } return payloadBits; } /***************************************************************************** functionname: encodeExtendedData description: writes bits corresponding to the extended data returns: number of bits written input: output: *****************************************************************************/ static INT encodeExtendedData(HANDLE_PARAMETRIC_STEREO hParametricStereo, HANDLE_FDK_BITSTREAM hBitStream) { INT extDataSize; INT payloadBits = 0; extDataSize = getSbrExtendedDataSize(hParametricStereo); if (extDataSize != 0) { INT maxExtSize = (1 << SI_SBR_EXTENSION_SIZE_BITS) - 1; INT writtenNoBits = 0; /* needed to byte align the extended data */ payloadBits += FDKwriteBits(hBitStream, 1, SI_SBR_EXTENDED_DATA_BITS); FDK_ASSERT(extDataSize <= SBR_EXTENDED_DATA_MAX_CNT); if (extDataSize < maxExtSize) { payloadBits += FDKwriteBits(hBitStream, extDataSize, SI_SBR_EXTENSION_SIZE_BITS); } else { payloadBits += FDKwriteBits(hBitStream, maxExtSize, SI_SBR_EXTENSION_SIZE_BITS); payloadBits += FDKwriteBits(hBitStream, extDataSize - maxExtSize, SI_SBR_EXTENSION_ESC_COUNT_BITS); } /* parametric coding signalled here? */ if (hParametricStereo) { writtenNoBits += FDKwriteBits(hBitStream, EXTENSION_ID_PS_CODING, SI_SBR_EXTENSION_ID_BITS); writtenNoBits += FDKsbrEnc_PSEnc_WritePSData(hParametricStereo, hBitStream); } payloadBits += writtenNoBits; /* byte alignment */ writtenNoBits = writtenNoBits % 8; if (writtenNoBits) payloadBits += FDKwriteBits(hBitStream, 0, (8 - writtenNoBits)); } else { payloadBits += FDKwriteBits(hBitStream, 0, SI_SBR_EXTENDED_DATA_BITS); } return payloadBits; } /***************************************************************************** functionname: writeSyntheticCodingData description: writes bits corresponding to the "synthetic-coding"-extension returns: number of bits written input: output: *****************************************************************************/ static INT writeSyntheticCodingData(HANDLE_SBR_ENV_DATA sbrEnvData, HANDLE_FDK_BITSTREAM hBitStream) { INT i; INT payloadBits = 0; payloadBits += FDKwriteBits(hBitStream, sbrEnvData->addHarmonicFlag, 1); if (sbrEnvData->addHarmonicFlag) { for (i = 0; i < sbrEnvData->noHarmonics; i++) { payloadBits += FDKwriteBits(hBitStream, sbrEnvData->addHarmonic[i], 1); } } return payloadBits; } /***************************************************************************** functionname: getSbrExtendedDataSize description: counts the number of bits needed for encoding the extended data (including extension id) returns: number of bits needed for the extended data input: output: *****************************************************************************/ static INT getSbrExtendedDataSize(HANDLE_PARAMETRIC_STEREO hParametricStereo) { INT extDataBits = 0; /* add your new extended data counting methods here */ /* no extended data */ if (hParametricStereo) { /* PS extended data */ extDataBits += SI_SBR_EXTENSION_ID_BITS; extDataBits += FDKsbrEnc_PSEnc_WritePSData(hParametricStereo, NULL); } return (extDataBits + 7) >> 3; } fdk-aac-2.0.2/libSBRenc/src/bit_sbr.h000066400000000000000000000221401404225310700171650ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief SBR bit writing $Revision: 92790 $ */ #ifndef BIT_SBR_H #define BIT_SBR_H #include "sbr_def.h" #include "cmondata.h" #include "fram_gen.h" struct SBR_ENV_DATA; struct SBR_BITSTREAM_DATA { INT TotalBits; INT PayloadBits; INT FillBits; INT HeaderActive; INT HeaderActiveDelay; /**< sbr payload and its header is delayed depending on encoder configuration*/ INT NrSendHeaderData; /**< input from commandline */ INT CountSendHeaderData; /**< modulo count. If < 0 then no counting is done (no SBR headers) */ INT rightBorderFIX; /**< force VARFIX or FIXFIX frames */ }; typedef struct SBR_BITSTREAM_DATA *HANDLE_SBR_BITSTREAM_DATA; struct SBR_HEADER_DATA { AMP_RES sbr_amp_res; INT sbr_start_frequency; INT sbr_stop_frequency; INT sbr_xover_band; INT sbr_noise_bands; INT sbr_data_extra; INT header_extra_1; INT header_extra_2; INT sbr_lc_stereo_mode; INT sbr_limiter_bands; INT sbr_limiter_gains; INT sbr_interpol_freq; INT sbr_smoothing_length; INT alterScale; INT freqScale; /* element of channelpairelement */ INT coupling; INT prev_coupling; /* element of singlechannelelement */ }; typedef struct SBR_HEADER_DATA *HANDLE_SBR_HEADER_DATA; struct SBR_ENV_DATA { INT sbr_xpos_ctrl; FREQ_RES freq_res_fixfix[2]; UCHAR fResTransIsLow; INVF_MODE sbr_invf_mode; INVF_MODE sbr_invf_mode_vec[MAX_NUM_NOISE_VALUES]; XPOS_MODE sbr_xpos_mode; INT ienvelope[MAX_ENVELOPES][MAX_FREQ_COEFFS]; INT codeBookScfLavBalance; INT codeBookScfLav; const INT *hufftableTimeC; const INT *hufftableFreqC; const UCHAR *hufftableTimeL; const UCHAR *hufftableFreqL; const INT *hufftableLevelTimeC; const INT *hufftableBalanceTimeC; const INT *hufftableLevelFreqC; const INT *hufftableBalanceFreqC; const UCHAR *hufftableLevelTimeL; const UCHAR *hufftableBalanceTimeL; const UCHAR *hufftableLevelFreqL; const UCHAR *hufftableBalanceFreqL; const UCHAR *hufftableNoiseTimeL; const INT *hufftableNoiseTimeC; const UCHAR *hufftableNoiseFreqL; const INT *hufftableNoiseFreqC; const UCHAR *hufftableNoiseLevelTimeL; const INT *hufftableNoiseLevelTimeC; const UCHAR *hufftableNoiseBalanceTimeL; const INT *hufftableNoiseBalanceTimeC; const UCHAR *hufftableNoiseLevelFreqL; const INT *hufftableNoiseLevelFreqC; const UCHAR *hufftableNoiseBalanceFreqL; const INT *hufftableNoiseBalanceFreqC; HANDLE_SBR_GRID hSbrBSGrid; INT noHarmonics; INT addHarmonicFlag; UCHAR addHarmonic[MAX_FREQ_COEFFS]; /* calculated helper vars */ INT si_sbr_start_env_bits_balance; INT si_sbr_start_env_bits; INT si_sbr_start_noise_bits_balance; INT si_sbr_start_noise_bits; INT noOfEnvelopes; INT noScfBands[MAX_ENVELOPES]; INT domain_vec[MAX_ENVELOPES]; INT domain_vec_noise[MAX_ENVELOPES]; SCHAR sbr_noise_levels[MAX_FREQ_COEFFS]; INT noOfnoisebands; INT balance; AMP_RES init_sbr_amp_res; AMP_RES currentAmpResFF; FIXP_DBL ton_HF[SBR_GLOBAL_TONALITY_VALUES]; /* tonality is scaled by 2^19/0.524288f (fract part of RELAXATION) */ FIXP_DBL global_tonality; /* extended data */ INT extended_data; INT extension_size; INT extension_id; UCHAR extended_data_buffer[SBR_EXTENDED_DATA_MAX_CNT]; UCHAR ldGrid; }; typedef struct SBR_ENV_DATA *HANDLE_SBR_ENV_DATA; INT FDKsbrEnc_WriteEnvSingleChannelElement( struct SBR_HEADER_DATA *sbrHeaderData, struct T_PARAMETRIC_STEREO *hParametricStereo, struct SBR_BITSTREAM_DATA *sbrBitstreamData, struct SBR_ENV_DATA *sbrEnvData, struct COMMON_DATA *cmonData, UINT sbrSyntaxFlags); INT FDKsbrEnc_WriteEnvChannelPairElement( struct SBR_HEADER_DATA *sbrHeaderData, struct T_PARAMETRIC_STEREO *hParametricStereo, struct SBR_BITSTREAM_DATA *sbrBitstreamData, struct SBR_ENV_DATA *sbrEnvDataLeft, struct SBR_ENV_DATA *sbrEnvDataRight, struct COMMON_DATA *cmonData, UINT sbrSyntaxFlags); INT FDKsbrEnc_CountSbrChannelPairElement( struct SBR_HEADER_DATA *sbrHeaderData, struct T_PARAMETRIC_STEREO *hParametricStereo, struct SBR_BITSTREAM_DATA *sbrBitstreamData, struct SBR_ENV_DATA *sbrEnvDataLeft, struct SBR_ENV_DATA *sbrEnvDataRight, struct COMMON_DATA *cmonData, UINT sbrSyntaxFlags); /* debugging and tuning functions */ /*#define SBR_ENV_STATISTICS */ /*#define SBR_PAYLOAD_MONITOR*/ #endif fdk-aac-2.0.2/libSBRenc/src/cmondata.h000066400000000000000000000130401404225310700173260ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Core Coder's and SBR's shared data structure definition $Revision: 92790 $ */ #ifndef CMONDATA_H #define CMONDATA_H #include "FDK_bitstream.h" struct COMMON_DATA { INT sbrHdrBits; /**< number of SBR header bits */ INT sbrDataBits; /**< number of SBR data bits */ INT sbrFillBits; /**< number of SBR fill bits */ FDK_BITSTREAM sbrBitbuf; /**< the SBR data bitbuffer */ FDK_BITSTREAM tmpWriteBitbuf; /**< helper var for writing header*/ INT xOverFreq; /**< the SBR crossover frequency */ INT dynBwEnabled; /**< indicates if dynamic bandwidth is enabled */ INT sbrNumChannels; /**< number of channels (meaning mono or stereo) */ INT dynXOverFreqEnc; /**< encoder dynamic crossover frequency */ }; typedef struct COMMON_DATA *HANDLE_COMMON_DATA; #endif fdk-aac-2.0.2/libSBRenc/src/code_env.cpp000066400000000000000000000555401404225310700176700ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Description: *******************************************************************************/ #include "code_env.h" #include "sbrenc_rom.h" /***************************************************************************** functionname: FDKsbrEnc_InitSbrHuffmanTables description: initializes Huffman Tables dependent on chosen amp_res returns: error handle input: output: *****************************************************************************/ INT FDKsbrEnc_InitSbrHuffmanTables(HANDLE_SBR_ENV_DATA sbrEnvData, HANDLE_SBR_CODE_ENVELOPE henv, HANDLE_SBR_CODE_ENVELOPE hnoise, AMP_RES amp_res) { if ((!henv) || (!hnoise) || (!sbrEnvData)) return (1); /* not init. */ sbrEnvData->init_sbr_amp_res = amp_res; switch (amp_res) { case SBR_AMP_RES_3_0: /*envelope data*/ /*Level/Pan - coding */ sbrEnvData->hufftableLevelTimeC = v_Huff_envelopeLevelC11T; sbrEnvData->hufftableLevelTimeL = v_Huff_envelopeLevelL11T; sbrEnvData->hufftableBalanceTimeC = bookSbrEnvBalanceC11T; sbrEnvData->hufftableBalanceTimeL = bookSbrEnvBalanceL11T; sbrEnvData->hufftableLevelFreqC = v_Huff_envelopeLevelC11F; sbrEnvData->hufftableLevelFreqL = v_Huff_envelopeLevelL11F; sbrEnvData->hufftableBalanceFreqC = bookSbrEnvBalanceC11F; sbrEnvData->hufftableBalanceFreqL = bookSbrEnvBalanceL11F; /*Right/Left - coding */ sbrEnvData->hufftableTimeC = v_Huff_envelopeLevelC11T; sbrEnvData->hufftableTimeL = v_Huff_envelopeLevelL11T; sbrEnvData->hufftableFreqC = v_Huff_envelopeLevelC11F; sbrEnvData->hufftableFreqL = v_Huff_envelopeLevelL11F; sbrEnvData->codeBookScfLavBalance = CODE_BOOK_SCF_LAV_BALANCE11; sbrEnvData->codeBookScfLav = CODE_BOOK_SCF_LAV11; sbrEnvData->si_sbr_start_env_bits = SI_SBR_START_ENV_BITS_AMP_RES_3_0; sbrEnvData->si_sbr_start_env_bits_balance = SI_SBR_START_ENV_BITS_BALANCE_AMP_RES_3_0; break; case SBR_AMP_RES_1_5: /*envelope data*/ /*Level/Pan - coding */ sbrEnvData->hufftableLevelTimeC = v_Huff_envelopeLevelC10T; sbrEnvData->hufftableLevelTimeL = v_Huff_envelopeLevelL10T; sbrEnvData->hufftableBalanceTimeC = bookSbrEnvBalanceC10T; sbrEnvData->hufftableBalanceTimeL = bookSbrEnvBalanceL10T; sbrEnvData->hufftableLevelFreqC = v_Huff_envelopeLevelC10F; sbrEnvData->hufftableLevelFreqL = v_Huff_envelopeLevelL10F; sbrEnvData->hufftableBalanceFreqC = bookSbrEnvBalanceC10F; sbrEnvData->hufftableBalanceFreqL = bookSbrEnvBalanceL10F; /*Right/Left - coding */ sbrEnvData->hufftableTimeC = v_Huff_envelopeLevelC10T; sbrEnvData->hufftableTimeL = v_Huff_envelopeLevelL10T; sbrEnvData->hufftableFreqC = v_Huff_envelopeLevelC10F; sbrEnvData->hufftableFreqL = v_Huff_envelopeLevelL10F; sbrEnvData->codeBookScfLavBalance = CODE_BOOK_SCF_LAV_BALANCE10; sbrEnvData->codeBookScfLav = CODE_BOOK_SCF_LAV10; sbrEnvData->si_sbr_start_env_bits = SI_SBR_START_ENV_BITS_AMP_RES_1_5; sbrEnvData->si_sbr_start_env_bits_balance = SI_SBR_START_ENV_BITS_BALANCE_AMP_RES_1_5; break; default: return (1); /* undefined amp_res mode */ } /* these are common to both amp_res values */ /*Noise data*/ /*Level/Pan - coding */ sbrEnvData->hufftableNoiseLevelTimeC = v_Huff_NoiseLevelC11T; sbrEnvData->hufftableNoiseLevelTimeL = v_Huff_NoiseLevelL11T; sbrEnvData->hufftableNoiseBalanceTimeC = bookSbrNoiseBalanceC11T; sbrEnvData->hufftableNoiseBalanceTimeL = bookSbrNoiseBalanceL11T; sbrEnvData->hufftableNoiseLevelFreqC = v_Huff_envelopeLevelC11F; sbrEnvData->hufftableNoiseLevelFreqL = v_Huff_envelopeLevelL11F; sbrEnvData->hufftableNoiseBalanceFreqC = bookSbrEnvBalanceC11F; sbrEnvData->hufftableNoiseBalanceFreqL = bookSbrEnvBalanceL11F; /*Right/Left - coding */ sbrEnvData->hufftableNoiseTimeC = v_Huff_NoiseLevelC11T; sbrEnvData->hufftableNoiseTimeL = v_Huff_NoiseLevelL11T; sbrEnvData->hufftableNoiseFreqC = v_Huff_envelopeLevelC11F; sbrEnvData->hufftableNoiseFreqL = v_Huff_envelopeLevelL11F; sbrEnvData->si_sbr_start_noise_bits = SI_SBR_START_NOISE_BITS_AMP_RES_3_0; sbrEnvData->si_sbr_start_noise_bits_balance = SI_SBR_START_NOISE_BITS_BALANCE_AMP_RES_3_0; /* init envelope tables and codebooks */ henv->codeBookScfLavBalanceTime = sbrEnvData->codeBookScfLavBalance; henv->codeBookScfLavBalanceFreq = sbrEnvData->codeBookScfLavBalance; henv->codeBookScfLavLevelTime = sbrEnvData->codeBookScfLav; henv->codeBookScfLavLevelFreq = sbrEnvData->codeBookScfLav; henv->codeBookScfLavTime = sbrEnvData->codeBookScfLav; henv->codeBookScfLavFreq = sbrEnvData->codeBookScfLav; henv->hufftableLevelTimeL = sbrEnvData->hufftableLevelTimeL; henv->hufftableBalanceTimeL = sbrEnvData->hufftableBalanceTimeL; henv->hufftableTimeL = sbrEnvData->hufftableTimeL; henv->hufftableLevelFreqL = sbrEnvData->hufftableLevelFreqL; henv->hufftableBalanceFreqL = sbrEnvData->hufftableBalanceFreqL; henv->hufftableFreqL = sbrEnvData->hufftableFreqL; henv->codeBookScfLavFreq = sbrEnvData->codeBookScfLav; henv->codeBookScfLavTime = sbrEnvData->codeBookScfLav; henv->start_bits = sbrEnvData->si_sbr_start_env_bits; henv->start_bits_balance = sbrEnvData->si_sbr_start_env_bits_balance; /* init noise tables and codebooks */ hnoise->codeBookScfLavBalanceTime = CODE_BOOK_SCF_LAV_BALANCE11; hnoise->codeBookScfLavBalanceFreq = CODE_BOOK_SCF_LAV_BALANCE11; hnoise->codeBookScfLavLevelTime = CODE_BOOK_SCF_LAV11; hnoise->codeBookScfLavLevelFreq = CODE_BOOK_SCF_LAV11; hnoise->codeBookScfLavTime = CODE_BOOK_SCF_LAV11; hnoise->codeBookScfLavFreq = CODE_BOOK_SCF_LAV11; hnoise->hufftableLevelTimeL = sbrEnvData->hufftableNoiseLevelTimeL; hnoise->hufftableBalanceTimeL = sbrEnvData->hufftableNoiseBalanceTimeL; hnoise->hufftableTimeL = sbrEnvData->hufftableNoiseTimeL; hnoise->hufftableLevelFreqL = sbrEnvData->hufftableNoiseLevelFreqL; hnoise->hufftableBalanceFreqL = sbrEnvData->hufftableNoiseBalanceFreqL; hnoise->hufftableFreqL = sbrEnvData->hufftableNoiseFreqL; hnoise->start_bits = sbrEnvData->si_sbr_start_noise_bits; hnoise->start_bits_balance = sbrEnvData->si_sbr_start_noise_bits_balance; /* No delta coding in time from the previous frame due to 1.5dB FIx-FIX rule */ henv->upDate = 0; hnoise->upDate = 0; return (0); } /******************************************************************************* Functionname: indexLow2High ******************************************************************************* Description: Nice small patch-functions in order to cope with non-factor-2 ratios between high-res and low-res Arguments: INT offset, INT index, FREQ_RES res Return: INT *******************************************************************************/ static INT indexLow2High(INT offset, INT index, FREQ_RES res) { if (res == FREQ_RES_LOW) { if (offset >= 0) { if (index < offset) return (index); else return (2 * index - offset); } else { offset = -offset; if (index < offset) return (2 * index + index); else return (2 * index + offset); } } else return (index); } /******************************************************************************* Functionname: mapLowResEnergyVal ******************************************************************************* Description: Arguments: INT currVal,INT* prevData, INT offset, INT index, FREQ_RES res Return: none *******************************************************************************/ static void mapLowResEnergyVal(SCHAR currVal, SCHAR *prevData, INT offset, INT index, FREQ_RES res) { if (res == FREQ_RES_LOW) { if (offset >= 0) { if (index < offset) prevData[index] = currVal; else { prevData[2 * index - offset] = currVal; prevData[2 * index + 1 - offset] = currVal; } } else { offset = -offset; if (index < offset) { prevData[3 * index] = currVal; prevData[3 * index + 1] = currVal; prevData[3 * index + 2] = currVal; } else { prevData[2 * index + offset] = currVal; prevData[2 * index + 1 + offset] = currVal; } } } else prevData[index] = currVal; } /******************************************************************************* Functionname: computeBits ******************************************************************************* Description: Arguments: INT delta, INT codeBookScfLavLevel, INT codeBookScfLavBalance, const UCHAR * hufftableLevel, const UCHAR * hufftableBalance, INT coupling, INT channel) Return: INT *******************************************************************************/ static INT computeBits(SCHAR *delta, INT codeBookScfLavLevel, INT codeBookScfLavBalance, const UCHAR *hufftableLevel, const UCHAR *hufftableBalance, INT coupling, INT channel) { INT index; INT delta_bits = 0; if (coupling) { if (channel == 1) { if (*delta < 0) index = fixMax(*delta, -codeBookScfLavBalance); else index = fixMin(*delta, codeBookScfLavBalance); if (index != *delta) { *delta = index; return (10000); } delta_bits = hufftableBalance[index + codeBookScfLavBalance]; } else { if (*delta < 0) index = fixMax(*delta, -codeBookScfLavLevel); else index = fixMin(*delta, codeBookScfLavLevel); if (index != *delta) { *delta = index; return (10000); } delta_bits = hufftableLevel[index + codeBookScfLavLevel]; } } else { if (*delta < 0) index = fixMax(*delta, -codeBookScfLavLevel); else index = fixMin(*delta, codeBookScfLavLevel); if (index != *delta) { *delta = index; return (10000); } delta_bits = hufftableLevel[index + codeBookScfLavLevel]; } return (delta_bits); } /******************************************************************************* Functionname: FDKsbrEnc_codeEnvelope ******************************************************************************* Description: Arguments: INT *sfb_nrg, const FREQ_RES *freq_res, SBR_CODE_ENVELOPE * h_sbrCodeEnvelope, INT *directionVec, INT scalable, INT nEnvelopes, INT channel, INT headerActive) Return: none h_sbrCodeEnvelope->sfb_nrg_prev is modified ! sfb_nrg is modified h_sbrCodeEnvelope->update is modfied ! *directionVec is modified *******************************************************************************/ void FDKsbrEnc_codeEnvelope(SCHAR *sfb_nrg, const FREQ_RES *freq_res, SBR_CODE_ENVELOPE *h_sbrCodeEnvelope, INT *directionVec, INT coupling, INT nEnvelopes, INT channel, INT headerActive) { INT i, no_of_bands, band; FIXP_DBL tmp1, tmp2, tmp3, dF_edge_1stEnv; SCHAR *ptr_nrg; INT codeBookScfLavLevelTime; INT codeBookScfLavLevelFreq; INT codeBookScfLavBalanceTime; INT codeBookScfLavBalanceFreq; const UCHAR *hufftableLevelTimeL; const UCHAR *hufftableBalanceTimeL; const UCHAR *hufftableLevelFreqL; const UCHAR *hufftableBalanceFreqL; INT offset = h_sbrCodeEnvelope->offset; INT envDataTableCompFactor; INT delta_F_bits = 0, delta_T_bits = 0; INT use_dT; SCHAR delta_F[MAX_FREQ_COEFFS]; SCHAR delta_T[MAX_FREQ_COEFFS]; SCHAR last_nrg, curr_nrg; tmp1 = FL2FXCONST_DBL(0.5f) >> (DFRACT_BITS - 16 - 1); tmp2 = h_sbrCodeEnvelope->dF_edge_1stEnv >> (DFRACT_BITS - 16); tmp3 = (FIXP_DBL)fMult(h_sbrCodeEnvelope->dF_edge_incr, ((FIXP_DBL)h_sbrCodeEnvelope->dF_edge_incr_fac) << 15); dF_edge_1stEnv = tmp1 + tmp2 + tmp3; if (coupling) { codeBookScfLavLevelTime = h_sbrCodeEnvelope->codeBookScfLavLevelTime; codeBookScfLavLevelFreq = h_sbrCodeEnvelope->codeBookScfLavLevelFreq; codeBookScfLavBalanceTime = h_sbrCodeEnvelope->codeBookScfLavBalanceTime; codeBookScfLavBalanceFreq = h_sbrCodeEnvelope->codeBookScfLavBalanceFreq; hufftableLevelTimeL = h_sbrCodeEnvelope->hufftableLevelTimeL; hufftableBalanceTimeL = h_sbrCodeEnvelope->hufftableBalanceTimeL; hufftableLevelFreqL = h_sbrCodeEnvelope->hufftableLevelFreqL; hufftableBalanceFreqL = h_sbrCodeEnvelope->hufftableBalanceFreqL; } else { codeBookScfLavLevelTime = h_sbrCodeEnvelope->codeBookScfLavTime; codeBookScfLavLevelFreq = h_sbrCodeEnvelope->codeBookScfLavFreq; codeBookScfLavBalanceTime = h_sbrCodeEnvelope->codeBookScfLavTime; codeBookScfLavBalanceFreq = h_sbrCodeEnvelope->codeBookScfLavFreq; hufftableLevelTimeL = h_sbrCodeEnvelope->hufftableTimeL; hufftableBalanceTimeL = h_sbrCodeEnvelope->hufftableTimeL; hufftableLevelFreqL = h_sbrCodeEnvelope->hufftableFreqL; hufftableBalanceFreqL = h_sbrCodeEnvelope->hufftableFreqL; } if (coupling == 1 && channel == 1) envDataTableCompFactor = 1; /*should be one when the new huffman-tables are ready*/ else envDataTableCompFactor = 0; if (h_sbrCodeEnvelope->deltaTAcrossFrames == 0) h_sbrCodeEnvelope->upDate = 0; /* no delta coding in time in case of a header */ if (headerActive) h_sbrCodeEnvelope->upDate = 0; for (i = 0; i < nEnvelopes; i++) { if (freq_res[i] == FREQ_RES_HIGH) no_of_bands = h_sbrCodeEnvelope->nSfb[FREQ_RES_HIGH]; else no_of_bands = h_sbrCodeEnvelope->nSfb[FREQ_RES_LOW]; ptr_nrg = sfb_nrg; curr_nrg = *ptr_nrg; delta_F[0] = curr_nrg >> envDataTableCompFactor; if (coupling && channel == 1) delta_F_bits = h_sbrCodeEnvelope->start_bits_balance; else delta_F_bits = h_sbrCodeEnvelope->start_bits; if (h_sbrCodeEnvelope->upDate != 0) { delta_T[0] = (curr_nrg - h_sbrCodeEnvelope->sfb_nrg_prev[0]) >> envDataTableCompFactor; delta_T_bits = computeBits(&delta_T[0], codeBookScfLavLevelTime, codeBookScfLavBalanceTime, hufftableLevelTimeL, hufftableBalanceTimeL, coupling, channel); } mapLowResEnergyVal(curr_nrg, h_sbrCodeEnvelope->sfb_nrg_prev, offset, 0, freq_res[i]); /* ensure that nrg difference is not higher than codeBookScfLavXXXFreq */ if (coupling && channel == 1) { for (band = no_of_bands - 1; band > 0; band--) { if (ptr_nrg[band] - ptr_nrg[band - 1] > codeBookScfLavBalanceFreq) { ptr_nrg[band - 1] = ptr_nrg[band] - codeBookScfLavBalanceFreq; } } for (band = 1; band < no_of_bands; band++) { if (ptr_nrg[band - 1] - ptr_nrg[band] > codeBookScfLavBalanceFreq) { ptr_nrg[band] = ptr_nrg[band - 1] - codeBookScfLavBalanceFreq; } } } else { for (band = no_of_bands - 1; band > 0; band--) { if (ptr_nrg[band] - ptr_nrg[band - 1] > codeBookScfLavLevelFreq) { ptr_nrg[band - 1] = ptr_nrg[band] - codeBookScfLavLevelFreq; } } for (band = 1; band < no_of_bands; band++) { if (ptr_nrg[band - 1] - ptr_nrg[band] > codeBookScfLavLevelFreq) { ptr_nrg[band] = ptr_nrg[band - 1] - codeBookScfLavLevelFreq; } } } /* Coding loop*/ for (band = 1; band < no_of_bands; band++) { last_nrg = (*ptr_nrg); ptr_nrg++; curr_nrg = (*ptr_nrg); delta_F[band] = (curr_nrg - last_nrg) >> envDataTableCompFactor; delta_F_bits += computeBits( &delta_F[band], codeBookScfLavLevelFreq, codeBookScfLavBalanceFreq, hufftableLevelFreqL, hufftableBalanceFreqL, coupling, channel); if (h_sbrCodeEnvelope->upDate != 0) { delta_T[band] = curr_nrg - h_sbrCodeEnvelope ->sfb_nrg_prev[indexLow2High(offset, band, freq_res[i])]; delta_T[band] = delta_T[band] >> envDataTableCompFactor; } mapLowResEnergyVal(curr_nrg, h_sbrCodeEnvelope->sfb_nrg_prev, offset, band, freq_res[i]); if (h_sbrCodeEnvelope->upDate != 0) { delta_T_bits += computeBits( &delta_T[band], codeBookScfLavLevelTime, codeBookScfLavBalanceTime, hufftableLevelTimeL, hufftableBalanceTimeL, coupling, channel); } } /* Replace sfb_nrg with deltacoded samples and set flag */ if (i == 0) { INT tmp_bits; tmp_bits = (((delta_T_bits * dF_edge_1stEnv) >> (DFRACT_BITS - 18)) + (FIXP_DBL)1) >> 1; use_dT = (h_sbrCodeEnvelope->upDate != 0 && (delta_F_bits > tmp_bits)); } else use_dT = (delta_T_bits < delta_F_bits && h_sbrCodeEnvelope->upDate != 0); if (use_dT) { directionVec[i] = TIME; FDKmemcpy(sfb_nrg, delta_T, no_of_bands * sizeof(SCHAR)); } else { h_sbrCodeEnvelope->upDate = 0; directionVec[i] = FREQ; FDKmemcpy(sfb_nrg, delta_F, no_of_bands * sizeof(SCHAR)); } sfb_nrg += no_of_bands; h_sbrCodeEnvelope->upDate = 1; } } /******************************************************************************* Functionname: FDKsbrEnc_InitSbrCodeEnvelope ******************************************************************************* Description: Arguments: Return: *******************************************************************************/ INT FDKsbrEnc_InitSbrCodeEnvelope(HANDLE_SBR_CODE_ENVELOPE h_sbrCodeEnvelope, INT *nSfb, INT deltaTAcrossFrames, FIXP_DBL dF_edge_1stEnv, FIXP_DBL dF_edge_incr) { FDKmemclear(h_sbrCodeEnvelope, sizeof(SBR_CODE_ENVELOPE)); h_sbrCodeEnvelope->deltaTAcrossFrames = deltaTAcrossFrames; h_sbrCodeEnvelope->dF_edge_1stEnv = dF_edge_1stEnv; h_sbrCodeEnvelope->dF_edge_incr = dF_edge_incr; h_sbrCodeEnvelope->dF_edge_incr_fac = 0; h_sbrCodeEnvelope->upDate = 0; h_sbrCodeEnvelope->nSfb[FREQ_RES_LOW] = nSfb[FREQ_RES_LOW]; h_sbrCodeEnvelope->nSfb[FREQ_RES_HIGH] = nSfb[FREQ_RES_HIGH]; h_sbrCodeEnvelope->offset = 2 * h_sbrCodeEnvelope->nSfb[FREQ_RES_LOW] - h_sbrCodeEnvelope->nSfb[FREQ_RES_HIGH]; return (0); } fdk-aac-2.0.2/libSBRenc/src/code_env.h000066400000000000000000000145341404225310700173330ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief DPCM Envelope coding $Revision: 92790 $ */ #ifndef CODE_ENV_H #define CODE_ENV_H #include "sbr_def.h" #include "bit_sbr.h" #include "fram_gen.h" typedef struct { INT offset; INT upDate; INT nSfb[2]; SCHAR sfb_nrg_prev[MAX_FREQ_COEFFS]; INT deltaTAcrossFrames; FIXP_DBL dF_edge_1stEnv; FIXP_DBL dF_edge_incr; INT dF_edge_incr_fac; INT codeBookScfLavTime; INT codeBookScfLavFreq; INT codeBookScfLavLevelTime; INT codeBookScfLavLevelFreq; INT codeBookScfLavBalanceTime; INT codeBookScfLavBalanceFreq; INT start_bits; INT start_bits_balance; const UCHAR *hufftableTimeL; const UCHAR *hufftableFreqL; const UCHAR *hufftableLevelTimeL; const UCHAR *hufftableBalanceTimeL; const UCHAR *hufftableLevelFreqL; const UCHAR *hufftableBalanceFreqL; } SBR_CODE_ENVELOPE; typedef SBR_CODE_ENVELOPE *HANDLE_SBR_CODE_ENVELOPE; void FDKsbrEnc_codeEnvelope(SCHAR *sfb_nrg, const FREQ_RES *freq_res, SBR_CODE_ENVELOPE *h_sbrCodeEnvelope, INT *directionVec, INT coupling, INT nEnvelopes, INT channel, INT headerActive); INT FDKsbrEnc_InitSbrCodeEnvelope(HANDLE_SBR_CODE_ENVELOPE h_sbrCodeEnvelope, INT *nSfb, INT deltaTAcrossFrames, FIXP_DBL dF_edge_1stEnv, FIXP_DBL dF_edge_incr); INT FDKsbrEnc_InitSbrHuffmanTables(struct SBR_ENV_DATA *sbrEnvData, HANDLE_SBR_CODE_ENVELOPE henv, HANDLE_SBR_CODE_ENVELOPE hnoise, AMP_RES amp_res); #endif fdk-aac-2.0.2/libSBRenc/src/env_bit.cpp000066400000000000000000000213371404225310700175310ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Remaining SBR Bit Writing Routines */ #include "env_bit.h" #include "cmondata.h" #ifndef min #define min(a, b) (a < b ? a : b) #endif #ifndef max #define max(a, b) (a > b ? a : b) #endif /* ***************************** crcAdvance **********************************/ /** * @fn * @brief updates crc data register * @return none * * This function updates the crc register * */ static void crcAdvance(USHORT crcPoly, USHORT crcMask, USHORT *crc, ULONG bValue, INT bBits) { INT i; USHORT flag; for (i = bBits - 1; i >= 0; i--) { flag = ((*crc) & crcMask) ? (1) : (0); flag ^= (bValue & (1 << i)) ? (1) : (0); (*crc) <<= 1; if (flag) (*crc) ^= crcPoly; } } /* ***************************** FDKsbrEnc_InitSbrBitstream * **********************************/ /** * @fn * @brief Inittialisation of sbr bitstream, write of dummy header and CRC * @return none * * * */ INT FDKsbrEnc_InitSbrBitstream( HANDLE_COMMON_DATA hCmonData, UCHAR *memoryBase, /*!< Pointer to bitstream buffer */ INT memorySize, /*!< Length of bitstream buffer in bytes */ HANDLE_FDK_CRCINFO hCrcInfo, UINT sbrSyntaxFlags) /*!< SBR syntax flags */ { INT crcRegion = 0; /* reset bit buffer */ FDKresetBitbuffer(&hCmonData->sbrBitbuf, BS_WRITER); FDKinitBitStream(&hCmonData->tmpWriteBitbuf, memoryBase, memorySize, 0, BS_WRITER); if (sbrSyntaxFlags & SBR_SYNTAX_CRC) { if (sbrSyntaxFlags & SBR_SYNTAX_DRM_CRC) { /* Init and start CRC region */ FDKwriteBits(&hCmonData->sbrBitbuf, 0x0, SI_SBR_DRM_CRC_BITS); FDKcrcInit(hCrcInfo, 0x001d, 0xFFFF, SI_SBR_DRM_CRC_BITS); crcRegion = FDKcrcStartReg(hCrcInfo, &hCmonData->sbrBitbuf, 0); } else { FDKwriteBits(&hCmonData->sbrBitbuf, 0x0, SI_SBR_CRC_BITS); } } return (crcRegion); } /* ************************** FDKsbrEnc_AssembleSbrBitstream * *******************************/ /** * @fn * @brief Formats the SBR payload * @return nothing * * Also the CRC will be calculated here. * */ void FDKsbrEnc_AssembleSbrBitstream(HANDLE_COMMON_DATA hCmonData, HANDLE_FDK_CRCINFO hCrcInfo, INT crcRegion, UINT sbrSyntaxFlags) { USHORT crcReg = SBR_CRCINIT; INT numCrcBits, i; /* check if SBR is present */ if (hCmonData == NULL) return; hCmonData->sbrFillBits = 0; /* Fill bits are written only for GA streams */ if (sbrSyntaxFlags & SBR_SYNTAX_DRM_CRC) { /* * Calculate and write DRM CRC */ FDKcrcEndReg(hCrcInfo, &hCmonData->sbrBitbuf, crcRegion); FDKwriteBits(&hCmonData->tmpWriteBitbuf, FDKcrcGetCRC(hCrcInfo) ^ 0xFF, SI_SBR_DRM_CRC_BITS); } else { if (!(sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY)) { /* Do alignment here, because its defined as part of the * sbr_extension_data */ int sbrLoad = hCmonData->sbrHdrBits + hCmonData->sbrDataBits; if (sbrSyntaxFlags & SBR_SYNTAX_CRC) { sbrLoad += SI_SBR_CRC_BITS; } sbrLoad += 4; /* Do byte Align with 4 bit offset. ISO/IEC 14496-3:2005(E) page 39. */ hCmonData->sbrFillBits = (8 - (sbrLoad % 8)) % 8; /* append fill bits */ FDKwriteBits(&hCmonData->sbrBitbuf, 0, hCmonData->sbrFillBits); FDK_ASSERT(FDKgetValidBits(&hCmonData->sbrBitbuf) % 8 == 4); } /* calculate crc */ if (sbrSyntaxFlags & SBR_SYNTAX_CRC) { FDK_BITSTREAM tmpCRCBuf = hCmonData->sbrBitbuf; FDKresetBitbuffer(&tmpCRCBuf, BS_READER); numCrcBits = hCmonData->sbrHdrBits + hCmonData->sbrDataBits + hCmonData->sbrFillBits; for (i = 0; i < numCrcBits; i++) { INT bit; bit = FDKreadBits(&tmpCRCBuf, 1); crcAdvance(SBR_CRC_POLY, SBR_CRC_MASK, &crcReg, bit, 1); } crcReg &= (SBR_CRC_RANGE); /* * Write CRC data. */ FDKwriteBits(&hCmonData->tmpWriteBitbuf, crcReg, SI_SBR_CRC_BITS); } } FDKsyncCache(&hCmonData->tmpWriteBitbuf); } fdk-aac-2.0.2/libSBRenc/src/env_bit.h000066400000000000000000000131021404225310700171650ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Remaining SBR Bit Writing Routines */ #ifndef ENV_BIT_H #define ENV_BIT_H #include "sbr_encoder.h" #include "FDK_crc.h" /* G(x) = x^10 + x^9 + x^5 + x^4 + x + 1 */ #define SBR_CRC_POLY (0x0233) #define SBR_CRC_MASK (0x0200) #define SBR_CRC_RANGE (0x03FF) #define SBR_CRC_MAXREGS 1 #define SBR_CRCINIT (0x0) #define SI_SBR_CRC_ENABLE_BITS 0 #define SI_SBR_CRC_BITS 10 #define SI_SBR_DRM_CRC_BITS 8 struct COMMON_DATA; INT FDKsbrEnc_InitSbrBitstream(struct COMMON_DATA *hCmonData, UCHAR *memoryBase, INT memorySize, HANDLE_FDK_CRCINFO hCrcInfo, UINT sbrSyntaxFlags); void FDKsbrEnc_AssembleSbrBitstream(struct COMMON_DATA *hCmonData, HANDLE_FDK_CRCINFO hCrcInfo, INT crcRegion, UINT sbrSyntaxFlags); #endif /* #ifndef ENV_BIT_H */ fdk-aac-2.0.2/libSBRenc/src/env_est.cpp000066400000000000000000002203061404225310700175430ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Description: *******************************************************************************/ #include "env_est.h" #include "tran_det.h" #include "qmf.h" #include "fram_gen.h" #include "bit_sbr.h" #include "cmondata.h" #include "sbrenc_ram.h" #include "genericStds.h" #define QUANT_ERROR_THRES 200 #define Y_NRG_SCALE 5 /* noCols = 32 -> shift(5) */ #define MAX_NRG_SLOTS_LD 16 static const UCHAR panTable[2][10] = {{0, 2, 4, 6, 8, 12, 16, 20, 24}, {0, 2, 4, 8, 12, 0, 0, 0, 0}}; static const UCHAR maxIndex[2] = {9, 5}; /****************************************************************************** Functionname: FDKsbrEnc_GetTonality ******************************************************************************/ /***************************************************************************/ /*! \brief Calculates complete energy per band from the energy values of the QMF subsamples. \brief quotaMatrix - calculated in FDKsbrEnc_CalculateTonalityQuotas() \brief noEstPerFrame - number of estimations per frame \brief startIndex - start index for the quota matrix \brief Energies - energy matrix \brief startBand - start band \brief stopBand - number of QMF bands \brief numberCols - number of QMF subsamples \return mean tonality of the 5 bands with the highest energy scaled by 2^(RELAXATION_SHIFT+2)*RELAXATION_FRACT ****************************************************************************/ static FIXP_DBL FDKsbrEnc_GetTonality(const FIXP_DBL *const *quotaMatrix, const INT noEstPerFrame, const INT startIndex, const FIXP_DBL *const *Energies, const UCHAR startBand, const INT stopBand, const INT numberCols) { UCHAR b, e, k; INT no_enMaxBand[SBR_MAX_ENERGY_VALUES] = {-1, -1, -1, -1, -1}; FIXP_DBL energyMax[SBR_MAX_ENERGY_VALUES] = { FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f)}; FIXP_DBL energyMaxMin = MAXVAL_DBL; /* min. energy in energyMax array */ UCHAR posEnergyMaxMin = 0; /* min. energy in energyMax array position */ FIXP_DBL tonalityBand[SBR_MAX_ENERGY_VALUES] = { FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f)}; FIXP_DBL globalTonality = FL2FXCONST_DBL(0.0f); FIXP_DBL energyBand[64]; INT maxNEnergyValues; /* max. number of max. energy values */ /*** Sum up energies for each band ***/ FDK_ASSERT(numberCols == 15 || numberCols == 16); /* numberCols is always 15 or 16 for ELD. In case of 16 bands, the energyBands are initialized with the [15]th column. The rest of the column energies are added in the next step. */ if (numberCols == 15) { for (b = startBand; b < stopBand; b++) { energyBand[b] = FL2FXCONST_DBL(0.0f); } } else { for (b = startBand; b < stopBand; b++) { energyBand[b] = Energies[15][b] >> 4; } } for (k = 0; k < 15; k++) { for (b = startBand; b < stopBand; b++) { energyBand[b] += Energies[k][b] >> 4; } } /*** Determine 5 highest band-energies ***/ maxNEnergyValues = fMin(SBR_MAX_ENERGY_VALUES, stopBand - startBand); /* Get min. value in energyMax array */ energyMaxMin = energyMax[0] = energyBand[startBand]; no_enMaxBand[0] = startBand; posEnergyMaxMin = 0; for (k = 1; k < maxNEnergyValues; k++) { energyMax[k] = energyBand[startBand + k]; no_enMaxBand[k] = startBand + k; if (energyMaxMin > energyMax[k]) { energyMaxMin = energyMax[k]; posEnergyMaxMin = k; } } for (b = startBand + maxNEnergyValues; b < stopBand; b++) { if (energyBand[b] > energyMaxMin) { energyMax[posEnergyMaxMin] = energyBand[b]; no_enMaxBand[posEnergyMaxMin] = b; /* Again, get min. value in energyMax array */ energyMaxMin = energyMax[0]; posEnergyMaxMin = 0; for (k = 1; k < maxNEnergyValues; k++) { if (energyMaxMin > energyMax[k]) { energyMaxMin = energyMax[k]; posEnergyMaxMin = k; } } } } /*** End determine 5 highest band-energies ***/ /* Get tonality values for 5 highest energies */ for (e = 0; e < maxNEnergyValues; e++) { tonalityBand[e] = FL2FXCONST_DBL(0.0f); for (k = 0; k < noEstPerFrame; k++) { tonalityBand[e] += quotaMatrix[startIndex + k][no_enMaxBand[e]] >> 1; } globalTonality += tonalityBand[e] >> 2; /* headroom of 2+1 (max. 5 additions) */ } return globalTonality; } /***************************************************************************/ /*! \brief Calculates energy form real and imaginary part of the QMF subsamples \return none ****************************************************************************/ LNK_SECTION_CODE_L1 static void FDKsbrEnc_getEnergyFromCplxQmfData( FIXP_DBL **RESTRICT energyValues, /*!< the result of the operation */ FIXP_DBL **RESTRICT realValues, /*!< the real part of the QMF subsamples */ FIXP_DBL **RESTRICT imagValues, /*!< the imaginary part of the QMF subsamples */ INT numberBands, /*!< number of QMF bands */ INT numberCols, /*!< number of QMF subsamples */ INT *qmfScale, /*!< sclefactor of QMF subsamples */ INT *energyScale) /*!< scalefactor of energies */ { int j, k; int scale; FIXP_DBL max_val = FL2FXCONST_DBL(0.0f); /* Get Scratch buffer */ C_ALLOC_SCRATCH_START(tmpNrg, FIXP_DBL, 32 * 64 / 2) /* Get max possible scaling of QMF data */ scale = DFRACT_BITS; for (k = 0; k < numberCols; k++) { scale = fixMin(scale, fixMin(getScalefactor(realValues[k], numberBands), getScalefactor(imagValues[k], numberBands))); } /* Tweak scaling stability for zero signal to non-zero signal transitions */ if (scale >= DFRACT_BITS - 1) { scale = (FRACT_BITS - 1 - *qmfScale); } /* prevent scaling of QMF values to -1.f */ scale = fixMax(0, scale - 1); /* Update QMF scale */ *qmfScale += scale; /* Calculate energy of each time slot pair, max energy and shift QMF values as far as possible to the left. */ { FIXP_DBL *nrgValues = tmpNrg; for (k = 0; k < numberCols; k += 2) { /* Load band vector addresses of 2 consecutive timeslots */ FIXP_DBL *RESTRICT r0 = realValues[k]; FIXP_DBL *RESTRICT i0 = imagValues[k]; FIXP_DBL *RESTRICT r1 = realValues[k + 1]; FIXP_DBL *RESTRICT i1 = imagValues[k + 1]; for (j = 0; j < numberBands; j++) { FIXP_DBL energy; FIXP_DBL tr0, tr1, ti0, ti1; /* Read QMF values of 2 timeslots */ tr0 = r0[j]; tr1 = r1[j]; ti0 = i0[j]; ti1 = i1[j]; /* Scale QMF Values and Calc Energy average of both timeslots */ tr0 <<= scale; ti0 <<= scale; energy = fPow2AddDiv2(fPow2Div2(tr0), ti0) >> 1; tr1 <<= scale; ti1 <<= scale; energy += fPow2AddDiv2(fPow2Div2(tr1), ti1) >> 1; /* Write timeslot pair energy to scratch */ *nrgValues++ = energy; max_val = fixMax(max_val, energy); /* Write back scaled QMF values */ r0[j] = tr0; r1[j] = tr1; i0[j] = ti0; i1[j] = ti1; } } } /* energyScale: scalefactor energies of current frame */ *energyScale = 2 * (*qmfScale) - 1; /* if qmfScale > 0: nr of right shifts otherwise nr of left shifts */ /* Scale timeslot pair energies and write to output buffer */ scale = CountLeadingBits(max_val); { FIXP_DBL *nrgValues = tmpNrg; for (k = 0; k> 1; k++) { scaleValues(energyValues[k], nrgValues, numberBands, scale); nrgValues += numberBands; } *energyScale += scale; } /* Free Scratch buffer */ C_ALLOC_SCRATCH_END(tmpNrg, FIXP_DBL, 32 * 64 / 2) } LNK_SECTION_CODE_L1 static void FDKsbrEnc_getEnergyFromCplxQmfDataFull( FIXP_DBL **RESTRICT energyValues, /*!< the result of the operation */ FIXP_DBL **RESTRICT realValues, /*!< the real part of the QMF subsamples */ FIXP_DBL **RESTRICT imagValues, /*!< the imaginary part of the QMF subsamples */ int numberBands, /*!< number of QMF bands */ int numberCols, /*!< number of QMF subsamples */ int *qmfScale, /*!< scalefactor of QMF subsamples */ int *energyScale) /*!< scalefactor of energies */ { int j, k; int scale; FIXP_DBL max_val = FL2FXCONST_DBL(0.0f); /* Get Scratch buffer */ C_ALLOC_SCRATCH_START(tmpNrg, FIXP_DBL, MAX_NRG_SLOTS_LD * 64) FDK_ASSERT(numberCols <= MAX_NRG_SLOTS_LD); FDK_ASSERT(numberBands <= 64); /* Get max possible scaling of QMF data */ scale = DFRACT_BITS; for (k = 0; k < numberCols; k++) { scale = fixMin(scale, fixMin(getScalefactor(realValues[k], numberBands), getScalefactor(imagValues[k], numberBands))); } /* Tweak scaling stability for zero signal to non-zero signal transitions */ if (scale >= DFRACT_BITS - 1) { scale = (FRACT_BITS - 1 - *qmfScale); } /* prevent scaling of QFM values to -1.f */ scale = fixMax(0, scale - 1); /* Update QMF scale */ *qmfScale += scale; /* Calculate energy of each time slot pair, max energy and shift QMF values as far as possible to the left. */ { FIXP_DBL *nrgValues = tmpNrg; for (k = 0; k < numberCols; k++) { /* Load band vector addresses of 1 timeslot */ FIXP_DBL *RESTRICT r0 = realValues[k]; FIXP_DBL *RESTRICT i0 = imagValues[k]; for (j = 0; j < numberBands; j++) { FIXP_DBL energy; FIXP_DBL tr0, ti0; /* Read QMF values of 1 timeslot */ tr0 = r0[j]; ti0 = i0[j]; /* Scale QMF Values and Calc Energy */ tr0 <<= scale; ti0 <<= scale; energy = fPow2AddDiv2(fPow2Div2(tr0), ti0); *nrgValues++ = energy; max_val = fixMax(max_val, energy); /* Write back scaled QMF values */ r0[j] = tr0; i0[j] = ti0; } } } /* energyScale: scalefactor energies of current frame */ *energyScale = 2 * (*qmfScale) - 1; /* if qmfScale > 0: nr of right shifts otherwise nr of left shifts */ /* Scale timeslot pair energies and write to output buffer */ scale = CountLeadingBits(max_val); { FIXP_DBL *nrgValues = tmpNrg; for (k = 0; k < numberCols; k++) { scaleValues(energyValues[k], nrgValues, numberBands, scale); nrgValues += numberBands; } *energyScale += scale; } /* Free Scratch buffer */ C_ALLOC_SCRATCH_END(tmpNrg, FIXP_DBL, MAX_NRG_SLOTS_LD * 64) } /***************************************************************************/ /*! \brief Quantisation of the panorama value (balance) \return the quantized pan value ****************************************************************************/ static INT mapPanorama(INT nrgVal, /*! integer value of the energy */ INT ampRes, /*! amplitude resolution [1.5/3dB] */ INT *quantError /*! quantization error of energy val*/ ) { int i; INT min_val, val; UCHAR panIndex; INT sign; sign = nrgVal > 0 ? 1 : -1; nrgVal *= sign; min_val = FDK_INT_MAX; panIndex = 0; for (i = 0; i < maxIndex[ampRes]; i++) { val = fixp_abs((nrgVal - (INT)panTable[ampRes][i])); if (val < min_val) { min_val = val; panIndex = i; } } *quantError = min_val; return panTable[ampRes][maxIndex[ampRes] - 1] + sign * panTable[ampRes][panIndex]; } /***************************************************************************/ /*! \brief Quantisation of the noise floor levels \return void ****************************************************************************/ static void sbrNoiseFloorLevelsQuantisation( SCHAR *RESTRICT iNoiseLevels, /*! quantized noise levels */ FIXP_DBL *RESTRICT NoiseLevels, /*! the noise levels. Exponent = LD_DATA_SHIFT */ INT coupling /*! the coupling flag */ ) { INT i; INT tmp, dummy; /* Quantisation, similar to sfb quant... */ for (i = 0; i < MAX_NUM_NOISE_VALUES; i++) { /* tmp = NoiseLevels[i] > (PFLOAT)30.0f ? 30: (INT) (NoiseLevels[i] + * (PFLOAT)0.5); */ /* 30>>LD_DATA_SHIFT = 0.46875 */ if ((FIXP_DBL)NoiseLevels[i] > FL2FXCONST_DBL(0.46875f)) { tmp = 30; } else { /* tmp = (INT)((FIXP_DBL)NoiseLevels[i] + (FL2FXCONST_DBL(0.5f)>>(*/ /* FRACT_BITS+ */ /* 6-1)));*/ /* tmp = tmp >> (DFRACT_BITS-1-LD_DATA_SHIFT); */ /* conversion to integer happens here */ /* rounding is done by shifting one bit less than necessary to the right, * adding '1' and then shifting the final bit */ tmp = ((((INT)NoiseLevels[i]) >> (DFRACT_BITS - 1 - LD_DATA_SHIFT))); /* conversion to integer */ if (tmp != 0) tmp += 1; } if (coupling) { tmp = tmp < -30 ? -30 : tmp; tmp = mapPanorama(tmp, 1, &dummy); } iNoiseLevels[i] = tmp; } } /***************************************************************************/ /*! \brief Calculation of noise floor for coupling \return void ****************************************************************************/ static void coupleNoiseFloor( FIXP_DBL *RESTRICT noise_level_left, /*! noise level left (modified)*/ FIXP_DBL *RESTRICT noise_level_right /*! noise level right (modified)*/ ) { FIXP_DBL cmpValLeft, cmpValRight; INT i; FIXP_DBL temp1, temp2; for (i = 0; i < MAX_NUM_NOISE_VALUES; i++) { /* Calculation of the power function using ld64: z = x^y; z' = CalcLd64(z) = y*CalcLd64(x)/64; z = CalcInvLd64(z'); */ cmpValLeft = NOISE_FLOOR_OFFSET_64 - noise_level_left[i]; cmpValRight = NOISE_FLOOR_OFFSET_64 - noise_level_right[i]; if (cmpValRight < FL2FXCONST_DBL(0.0f)) { temp1 = CalcInvLdData(NOISE_FLOOR_OFFSET_64 - noise_level_right[i]); } else { temp1 = CalcInvLdData(NOISE_FLOOR_OFFSET_64 - noise_level_right[i]); temp1 = temp1 << (DFRACT_BITS - 1 - LD_DATA_SHIFT - 1); /* INT to fract conversion of result, if input of CalcInvLdData is positiv */ } if (cmpValLeft < FL2FXCONST_DBL(0.0f)) { temp2 = CalcInvLdData(NOISE_FLOOR_OFFSET_64 - noise_level_left[i]); } else { temp2 = CalcInvLdData(NOISE_FLOOR_OFFSET_64 - noise_level_left[i]); temp2 = temp2 << (DFRACT_BITS - 1 - LD_DATA_SHIFT - 1); /* INT to fract conversion of result, if input of CalcInvLdData is positiv */ } if ((cmpValLeft < FL2FXCONST_DBL(0.0f)) && (cmpValRight < FL2FXCONST_DBL(0.0f))) { noise_level_left[i] = NOISE_FLOOR_OFFSET_64 - (CalcLdData( ((temp1 >> 1) + (temp2 >> 1)))); /* no scaling needed! both values are dfract */ noise_level_right[i] = CalcLdData(temp2) - CalcLdData(temp1); } if ((cmpValLeft >= FL2FXCONST_DBL(0.0f)) && (cmpValRight >= FL2FXCONST_DBL(0.0f))) { noise_level_left[i] = NOISE_FLOOR_OFFSET_64 - (CalcLdData(((temp1 >> 1) + (temp2 >> 1))) + FL2FXCONST_DBL(0.109375f)); /* scaled with 7/64 */ noise_level_right[i] = CalcLdData(temp2) - CalcLdData(temp1); } if ((cmpValLeft >= FL2FXCONST_DBL(0.0f)) && (cmpValRight < FL2FXCONST_DBL(0.0f))) { noise_level_left[i] = NOISE_FLOOR_OFFSET_64 - (CalcLdData(((temp1 >> (7 + 1)) + (temp2 >> 1))) + FL2FXCONST_DBL(0.109375f)); /* scaled with 7/64 */ noise_level_right[i] = (CalcLdData(temp2) + FL2FXCONST_DBL(0.109375f)) - CalcLdData(temp1); } if ((cmpValLeft < FL2FXCONST_DBL(0.0f)) && (cmpValRight >= FL2FXCONST_DBL(0.0f))) { noise_level_left[i] = NOISE_FLOOR_OFFSET_64 - (CalcLdData(((temp1 >> 1) + (temp2 >> (7 + 1)))) + FL2FXCONST_DBL(0.109375f)); /* scaled with 7/64 */ noise_level_right[i] = CalcLdData(temp2) - (CalcLdData(temp1) + FL2FXCONST_DBL(0.109375f)); /* scaled with 7/64 */ } } } /***************************************************************************/ /*! \brief Calculation of energy starting in lower band (li) up to upper band (ui) over slots (start_pos) to (stop_pos) \return void ****************************************************************************/ static FIXP_DBL getEnvSfbEnergy( INT li, /*! lower band */ INT ui, /*! upper band */ INT start_pos, /*! start slot */ INT stop_pos, /*! stop slot */ INT border_pos, /*! slots scaling border */ FIXP_DBL **YBuffer, /*! sfb energy buffer */ INT YBufferSzShift, /*! Energy buffer index scale */ INT scaleNrg0, /*! scaling of lower slots */ INT scaleNrg1) /*! scaling of upper slots */ { /* use dynamic scaling for outer energy loop; energies are critical and every bit is important */ int sc0, sc1, k, l; FIXP_DBL nrgSum, nrg1, nrg2, accu1, accu2; INT dynScale, dynScale1, dynScale2; if (ui - li == 0) dynScale = DFRACT_BITS - 1; else dynScale = CalcLdInt(ui - li) >> (DFRACT_BITS - 1 - LD_DATA_SHIFT); sc0 = fixMin(scaleNrg0, Y_NRG_SCALE); sc1 = fixMin(scaleNrg1, Y_NRG_SCALE); /* dynScale{1,2} is set such that the right shift below is positive */ dynScale1 = fixMin((scaleNrg0 - sc0), dynScale); dynScale2 = fixMin((scaleNrg1 - sc1), dynScale); nrgSum = accu1 = accu2 = (FIXP_DBL)0; for (k = li; k < ui; k++) { nrg1 = nrg2 = (FIXP_DBL)0; for (l = start_pos; l < border_pos; l++) { nrg1 += YBuffer[l >> YBufferSzShift][k] >> sc0; } for (; l < stop_pos; l++) { nrg2 += YBuffer[l >> YBufferSzShift][k] >> sc1; } accu1 += (nrg1 >> dynScale1); accu2 += (nrg2 >> dynScale2); } /* This shift factor is always positive. See comment above. */ nrgSum += (accu1 >> fixMin((scaleNrg0 - sc0 - dynScale1), (DFRACT_BITS - 1))) + (accu2 >> fixMin((scaleNrg1 - sc1 - dynScale2), (DFRACT_BITS - 1))); return nrgSum; } /***************************************************************************/ /*! \brief Energy compensation in missing harmonic mode \return void ****************************************************************************/ static FIXP_DBL mhLoweringEnergy(FIXP_DBL nrg, INT M) { /* Compensating for the fact that we in the decoder map the "average energy to every QMF band, and use this when we calculate the boost-factor. Since the mapped energy isn't the average energy but the maximum energy in case of missing harmonic creation, we will in the boost function calculate that too much limiting has been applied and hence we will boost the signal although it isn't called for. Hence we need to compensate for this by lowering the transmitted energy values for the sines so they will get the correct level after the boost is applied. */ if (M > 2) { INT tmpScale; tmpScale = CountLeadingBits(nrg); nrg <<= tmpScale; nrg = fMult(nrg, FL2FXCONST_DBL(0.398107267f)); /* The maximum boost is 1.584893, so the maximum attenuation should be square(1/1.584893) = 0.398107267 */ nrg >>= tmpScale; } else { if (M > 1) { nrg >>= 1; } } return nrg; } /***************************************************************************/ /*! \brief Energy compensation in none missing harmonic mode \return void ****************************************************************************/ static FIXP_DBL nmhLoweringEnergy(FIXP_DBL nrg, const FIXP_DBL nrgSum, const INT nrgSum_scale, const INT M) { if (nrg > FL2FXCONST_DBL(0)) { int sc = 0; /* gain = nrgSum / (nrg*(M+1)) */ FIXP_DBL gain = fMult(fDivNorm(nrgSum, nrg, &sc), GetInvInt(M + 1)); sc += nrgSum_scale; /* reduce nrg if gain smaller 1.f */ if (!((sc >= 0) && (gain > ((FIXP_DBL)MAXVAL_DBL >> sc)))) { nrg = fMult(scaleValue(gain, sc), nrg); } } return nrg; } /***************************************************************************/ /*! \brief calculates the envelope values from the energies, depending on framing and stereo mode \return void ****************************************************************************/ static void calculateSbrEnvelope( FIXP_DBL **RESTRICT YBufferLeft, /*! energy buffer left */ FIXP_DBL **RESTRICT YBufferRight, /*! energy buffer right */ int *RESTRICT YBufferScaleLeft, /*! scale energy buffer left */ int *RESTRICT YBufferScaleRight, /*! scale energy buffer right */ const SBR_FRAME_INFO *frame_info, /*! frame info vector */ SCHAR *RESTRICT sfb_nrgLeft, /*! sfb energy buffer left */ SCHAR *RESTRICT sfb_nrgRight, /*! sfb energy buffer right */ HANDLE_SBR_CONFIG_DATA h_con, /*! handle to config data */ HANDLE_ENV_CHANNEL h_sbr, /*! envelope channel handle */ SBR_STEREO_MODE stereoMode, /*! stereo coding mode */ INT *maxQuantError, /*! maximum quantization error, for panorama. */ int YBufferSzShift) /*! Energy buffer index scale */ { int env, j, m = 0; INT no_of_bands, start_pos, stop_pos, li, ui; FREQ_RES freq_res; INT ca = 2 - h_sbr->encEnvData.init_sbr_amp_res; INT oneBitLess = 0; if (ca == 2) oneBitLess = 1; /* LD_DATA_SHIFT => ld64 scaling; one bit less for rounding */ INT quantError; INT nEnvelopes = frame_info->nEnvelopes; INT short_env = frame_info->shortEnv - 1; INT timeStep = h_sbr->sbrExtractEnvelope.time_step; INT commonScale, scaleLeft0, scaleLeft1; INT scaleRight0 = 0, scaleRight1 = 0; commonScale = fixMin(YBufferScaleLeft[0], YBufferScaleLeft[1]); if (stereoMode == SBR_COUPLING) { commonScale = fixMin(commonScale, YBufferScaleRight[0]); commonScale = fixMin(commonScale, YBufferScaleRight[1]); } commonScale = commonScale - 7; scaleLeft0 = YBufferScaleLeft[0] - commonScale; scaleLeft1 = YBufferScaleLeft[1] - commonScale; FDK_ASSERT((scaleLeft0 >= 0) && (scaleLeft1 >= 0)); if (stereoMode == SBR_COUPLING) { scaleRight0 = YBufferScaleRight[0] - commonScale; scaleRight1 = YBufferScaleRight[1] - commonScale; FDK_ASSERT((scaleRight0 >= 0) && (scaleRight1 >= 0)); *maxQuantError = 0; } for (env = 0; env < nEnvelopes; env++) { FIXP_DBL pNrgLeft[32]; FIXP_DBL pNrgRight[32]; int envNrg_scale; FIXP_DBL envNrgLeft = FL2FXCONST_DBL(0.0f); FIXP_DBL envNrgRight = FL2FXCONST_DBL(0.0f); int missingHarmonic[32]; int count[32]; start_pos = timeStep * frame_info->borders[env]; stop_pos = timeStep * frame_info->borders[env + 1]; freq_res = frame_info->freqRes[env]; no_of_bands = h_con->nSfb[freq_res]; envNrg_scale = DFRACT_BITS - fNormz((FIXP_DBL)no_of_bands); if (env == short_env) { j = fMax(2, timeStep); /* consider at least 2 QMF slots less for short envelopes (envelopes just before transients) */ if ((stop_pos - start_pos - j) > 0) { stop_pos = stop_pos - j; } } for (j = 0; j < no_of_bands; j++) { FIXP_DBL nrgLeft = FL2FXCONST_DBL(0.0f); FIXP_DBL nrgRight = FL2FXCONST_DBL(0.0f); li = h_con->freqBandTable[freq_res][j]; ui = h_con->freqBandTable[freq_res][j + 1]; if (freq_res == FREQ_RES_HIGH) { if (j == 0 && ui - li > 1) { li++; } } else { if (j == 0 && ui - li > 2) { li++; } } /* Find out whether a sine will be missing in the scale-factor band that we're currently processing. */ missingHarmonic[j] = 0; if (h_sbr->encEnvData.addHarmonicFlag) { if (freq_res == FREQ_RES_HIGH) { if (h_sbr->encEnvData .addHarmonic[j]) { /*A missing sine in the current band*/ missingHarmonic[j] = 1; } } else { INT i; INT startBandHigh = 0; INT stopBandHigh = 0; while (h_con->freqBandTable[FREQ_RES_HIGH][startBandHigh] < h_con->freqBandTable[FREQ_RES_LOW][j]) startBandHigh++; while (h_con->freqBandTable[FREQ_RES_HIGH][stopBandHigh] < h_con->freqBandTable[FREQ_RES_LOW][j + 1]) stopBandHigh++; for (i = startBandHigh; i < stopBandHigh; i++) { if (h_sbr->encEnvData.addHarmonic[i]) { missingHarmonic[j] = 1; } } } } /* If a sine is missing in a scalefactorband, with more than one qmf channel use the nrg from the channel with the largest nrg rather than the mean. Compensate for the boost calculation in the decdoder. */ int border_pos = fixMin(stop_pos, h_sbr->sbrExtractEnvelope.YBufferWriteOffset << YBufferSzShift); if (missingHarmonic[j]) { int k; count[j] = stop_pos - start_pos; nrgLeft = FL2FXCONST_DBL(0.0f); for (k = li; k < ui; k++) { FIXP_DBL tmpNrg; tmpNrg = getEnvSfbEnergy(k, k + 1, start_pos, stop_pos, border_pos, YBufferLeft, YBufferSzShift, scaleLeft0, scaleLeft1); nrgLeft = fixMax(nrgLeft, tmpNrg); } /* Energy lowering compensation */ nrgLeft = mhLoweringEnergy(nrgLeft, ui - li); if (stereoMode == SBR_COUPLING) { nrgRight = FL2FXCONST_DBL(0.0f); for (k = li; k < ui; k++) { FIXP_DBL tmpNrg; tmpNrg = getEnvSfbEnergy(k, k + 1, start_pos, stop_pos, border_pos, YBufferRight, YBufferSzShift, scaleRight0, scaleRight1); nrgRight = fixMax(nrgRight, tmpNrg); } /* Energy lowering compensation */ nrgRight = mhLoweringEnergy(nrgRight, ui - li); } } /* end missingHarmonic */ else { count[j] = (stop_pos - start_pos) * (ui - li); nrgLeft = getEnvSfbEnergy(li, ui, start_pos, stop_pos, border_pos, YBufferLeft, YBufferSzShift, scaleLeft0, scaleLeft1); if (stereoMode == SBR_COUPLING) { nrgRight = getEnvSfbEnergy(li, ui, start_pos, stop_pos, border_pos, YBufferRight, YBufferSzShift, scaleRight0, scaleRight1); } } /* !missingHarmonic */ /* save energies */ pNrgLeft[j] = nrgLeft; pNrgRight[j] = nrgRight; envNrgLeft += (nrgLeft >> envNrg_scale); envNrgRight += (nrgRight >> envNrg_scale); } /* j */ for (j = 0; j < no_of_bands; j++) { FIXP_DBL nrgLeft2 = FL2FXCONST_DBL(0.0f); FIXP_DBL nrgLeft = pNrgLeft[j]; FIXP_DBL nrgRight = pNrgRight[j]; /* None missing harmonic Energy lowering compensation */ if (!missingHarmonic[j] && h_sbr->fLevelProtect) { /* in case of missing energy in base band, reduce reference energy to prevent overflows in decoder output */ nrgLeft = nmhLoweringEnergy(nrgLeft, envNrgLeft, envNrg_scale, no_of_bands); if (stereoMode == SBR_COUPLING) { nrgRight = nmhLoweringEnergy(nrgRight, envNrgRight, envNrg_scale, no_of_bands); } } if (stereoMode == SBR_COUPLING) { /* calc operation later with log */ nrgLeft2 = nrgLeft; nrgLeft = (nrgRight + nrgLeft) >> 1; } /* nrgLeft = f20_log2(nrgLeft / (PFLOAT)(count * 64))+(PFLOAT)44; */ /* If nrgLeft == 0 then the Log calculations below do fail. */ if (nrgLeft > FL2FXCONST_DBL(0.0f)) { FIXP_DBL tmp0, tmp1, tmp2, tmp3; INT tmpScale; tmpScale = CountLeadingBits(nrgLeft); nrgLeft = nrgLeft << tmpScale; tmp0 = CalcLdData(nrgLeft); /* scaled by 1/64 */ tmp1 = ((FIXP_DBL)(commonScale + tmpScale)) << (DFRACT_BITS - 1 - LD_DATA_SHIFT - 1); /* scaled by 1/64 */ tmp2 = ((FIXP_DBL)(count[j] * 64)) << (DFRACT_BITS - 1 - 14 - 1); tmp2 = CalcLdData(tmp2); /* scaled by 1/64 */ tmp3 = FL2FXCONST_DBL(0.6875f - 0.21875f - 0.015625f) >> 1; /* scaled by 1/64 */ nrgLeft = ((tmp0 - tmp2) >> 1) + (tmp3 - tmp1); } else { nrgLeft = FL2FXCONST_DBL(-1.0f); } /* ld64 to integer conversion */ nrgLeft = fixMin(fixMax(nrgLeft, FL2FXCONST_DBL(0.0f)), (FL2FXCONST_DBL(0.5f) >> oneBitLess)); nrgLeft = (FIXP_DBL)(LONG)nrgLeft >> (DFRACT_BITS - 1 - LD_DATA_SHIFT - 1 - oneBitLess - 1); sfb_nrgLeft[m] = ((INT)nrgLeft + 1) >> 1; /* rounding */ if (stereoMode == SBR_COUPLING) { FIXP_DBL scaleFract; int sc0, sc1; nrgLeft2 = fixMax((FIXP_DBL)0x1, nrgLeft2); nrgRight = fixMax((FIXP_DBL)0x1, nrgRight); sc0 = CountLeadingBits(nrgLeft2); sc1 = CountLeadingBits(nrgRight); scaleFract = ((FIXP_DBL)(sc0 - sc1)) << (DFRACT_BITS - 1 - LD_DATA_SHIFT); /* scale value in ld64 representation */ nrgRight = CalcLdData(nrgLeft2 << sc0) - CalcLdData(nrgRight << sc1) - scaleFract; /* ld64 to integer conversion */ nrgRight = (FIXP_DBL)(LONG)(nrgRight) >> (DFRACT_BITS - 1 - LD_DATA_SHIFT - 1 - oneBitLess); nrgRight = (nrgRight + (FIXP_DBL)1) >> 1; /* rounding */ sfb_nrgRight[m] = mapPanorama( nrgRight, h_sbr->encEnvData.init_sbr_amp_res, &quantError); *maxQuantError = fixMax(quantError, *maxQuantError); } m++; } /* j */ /* Do energy compensation for sines that are present in two QMF-bands in the original, but will only occur in one band in the decoder due to the synthetic sine coding.*/ if (h_con->useParametricCoding) { m -= no_of_bands; for (j = 0; j < no_of_bands; j++) { if (freq_res == FREQ_RES_HIGH && h_sbr->sbrExtractEnvelope.envelopeCompensation[j]) { sfb_nrgLeft[m] -= (ca * fixp_abs( (INT)h_sbr->sbrExtractEnvelope.envelopeCompensation[j])); } sfb_nrgLeft[m] = fixMax(0, sfb_nrgLeft[m]); m++; } } /* useParametricCoding */ } /* env loop */ } /***************************************************************************/ /*! \brief calculates the noise floor and the envelope values from the energies, depending on framing and stereo mode FDKsbrEnc_extractSbrEnvelope is the main function for encoding and writing the envelope and the noise floor. The function includes the following processes: -Analysis subband filtering. -Encoding SA and pan parameters (if enabled). -Transient detection. ****************************************************************************/ LNK_SECTION_CODE_L1 void FDKsbrEnc_extractSbrEnvelope1( HANDLE_SBR_CONFIG_DATA h_con, /*! handle to config data */ HANDLE_SBR_HEADER_DATA sbrHeaderData, HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, HANDLE_ENV_CHANNEL hEnvChan, HANDLE_COMMON_DATA hCmonData, SBR_ENV_TEMP_DATA *eData, SBR_FRAME_TEMP_DATA *fData) { HANDLE_SBR_EXTRACT_ENVELOPE sbrExtrEnv = &hEnvChan->sbrExtractEnvelope; if (sbrExtrEnv->YBufferSzShift == 0) FDKsbrEnc_getEnergyFromCplxQmfDataFull( &sbrExtrEnv->YBuffer[sbrExtrEnv->YBufferWriteOffset], sbrExtrEnv->rBuffer + sbrExtrEnv->rBufferReadOffset, sbrExtrEnv->iBuffer + sbrExtrEnv->rBufferReadOffset, h_con->noQmfBands, sbrExtrEnv->no_cols, &hEnvChan->qmfScale, &sbrExtrEnv->YBufferScale[1]); else FDKsbrEnc_getEnergyFromCplxQmfData( &sbrExtrEnv->YBuffer[sbrExtrEnv->YBufferWriteOffset], sbrExtrEnv->rBuffer + sbrExtrEnv->rBufferReadOffset, sbrExtrEnv->iBuffer + sbrExtrEnv->rBufferReadOffset, h_con->noQmfBands, sbrExtrEnv->no_cols, &hEnvChan->qmfScale, &sbrExtrEnv->YBufferScale[1]); /* Energie values = * sbrExtrEnv->YBuffer[sbrExtrEnv->YBufferWriteOffset][x].floatVal * * (1<<2*7-sbrExtrEnv->YBufferScale[1]) */ /* Precalculation of Tonality Quotas COEFF Transform OK */ FDKsbrEnc_CalculateTonalityQuotas( &hEnvChan->TonCorr, sbrExtrEnv->rBuffer, sbrExtrEnv->iBuffer, h_con->freqBandTable[HI][h_con->nSfb[HI]], hEnvChan->qmfScale); if (h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { FIXP_DBL tonality = FDKsbrEnc_GetTonality( hEnvChan->TonCorr.quotaMatrix, hEnvChan->TonCorr.numberOfEstimatesPerFrame, hEnvChan->TonCorr.startIndexMatrix, sbrExtrEnv->YBuffer + sbrExtrEnv->YBufferWriteOffset, h_con->freqBandTable[HI][0] + 1, h_con->noQmfBands, sbrExtrEnv->no_cols); hEnvChan->encEnvData.ton_HF[1] = hEnvChan->encEnvData.ton_HF[0]; hEnvChan->encEnvData.ton_HF[0] = tonality; /* tonality is scaled by 2^19/0.524288f (fract part of RELAXATION) */ hEnvChan->encEnvData.global_tonality = (hEnvChan->encEnvData.ton_HF[0] >> 1) + (hEnvChan->encEnvData.ton_HF[1] >> 1); } /* Transient detection COEFF Transform OK */ if (h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { FDKsbrEnc_fastTransientDetect(&hEnvChan->sbrFastTransientDetector, sbrExtrEnv->YBuffer, sbrExtrEnv->YBufferScale, sbrExtrEnv->YBufferWriteOffset, eData->transient_info); } else { FDKsbrEnc_transientDetect( &hEnvChan->sbrTransientDetector, sbrExtrEnv->YBuffer, sbrExtrEnv->YBufferScale, eData->transient_info, sbrExtrEnv->YBufferWriteOffset, sbrExtrEnv->YBufferSzShift, sbrExtrEnv->time_step, hEnvChan->SbrEnvFrame.frameMiddleSlot); } /* Generate flags for 2 env in a FIXFIX-frame. Remove this function to get always 1 env per FIXFIX-frame. */ /* frame Splitter COEFF Transform OK */ FDKsbrEnc_frameSplitter( sbrExtrEnv->YBuffer, sbrExtrEnv->YBufferScale, &hEnvChan->sbrTransientDetector, h_con->freqBandTable[1], eData->transient_info, sbrExtrEnv->YBufferWriteOffset, sbrExtrEnv->YBufferSzShift, h_con->nSfb[1], sbrExtrEnv->time_step, sbrExtrEnv->no_cols, &hEnvChan->encEnvData.global_tonality); } /***************************************************************************/ /*! \brief calculates the noise floor and the envelope values from the energies, depending on framing and stereo mode FDKsbrEnc_extractSbrEnvelope is the main function for encoding and writing the envelope and the noise floor. The function includes the following processes: -Determine time/frequency division of current granule. -Sending transient info to bitstream. -Set amp_res to 1.5 dB if the current frame contains only one envelope. -Lock dynamic bandwidth frequency change if the next envelope not starts on a frame boundary. -MDCT transposer (needed to detect where harmonics will be missing). -Spectrum Estimation (used for pulse train and missing harmonics detection). -Pulse train detection. -Inverse Filtering detection. -Waveform Coding. -Missing Harmonics detection. -Extract envelope of current frame. -Noise floor estimation. -Noise floor quantisation and coding. -Encode envelope of current frame. -Send the encoded data to the bitstream. -Write to bitstream. ****************************************************************************/ LNK_SECTION_CODE_L1 void FDKsbrEnc_extractSbrEnvelope2( HANDLE_SBR_CONFIG_DATA h_con, /*! handle to config data */ HANDLE_SBR_HEADER_DATA sbrHeaderData, HANDLE_PARAMETRIC_STEREO hParametricStereo, HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, HANDLE_ENV_CHANNEL h_envChan0, HANDLE_ENV_CHANNEL h_envChan1, HANDLE_COMMON_DATA hCmonData, SBR_ENV_TEMP_DATA *eData, SBR_FRAME_TEMP_DATA *fData, int clearOutput) { HANDLE_ENV_CHANNEL h_envChan[MAX_NUM_CHANNELS] = {h_envChan0, h_envChan1}; int ch, i, j, c, YSzShift = h_envChan[0]->sbrExtractEnvelope.YBufferSzShift; SBR_STEREO_MODE stereoMode = h_con->stereoMode; int nChannels = h_con->nChannels; const int *v_tuning; static const int v_tuningHEAAC[6] = {0, 2, 4, 0, 0, 0}; static const int v_tuningELD[6] = {0, 2, 3, 0, 0, 0}; if (h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) v_tuning = v_tuningELD; else v_tuning = v_tuningHEAAC; /* Select stereo mode. */ if (stereoMode == SBR_COUPLING) { if (eData[0].transient_info[1] && eData[1].transient_info[1]) { eData[0].transient_info[0] = fixMin(eData[1].transient_info[0], eData[0].transient_info[0]); eData[1].transient_info[0] = eData[0].transient_info[0]; } else { if (eData[0].transient_info[1] && !eData[1].transient_info[1]) { eData[1].transient_info[0] = eData[0].transient_info[0]; } else { if (!eData[0].transient_info[1] && eData[1].transient_info[1]) eData[0].transient_info[0] = eData[1].transient_info[0]; else { eData[0].transient_info[0] = fixMax(eData[1].transient_info[0], eData[0].transient_info[0]); eData[1].transient_info[0] = eData[0].transient_info[0]; } } } } /* Determine time/frequency division of current granule */ eData[0].frame_info = FDKsbrEnc_frameInfoGenerator( &h_envChan[0]->SbrEnvFrame, eData[0].transient_info, sbrBitstreamData->rightBorderFIX, h_envChan[0]->sbrExtractEnvelope.pre_transient_info, h_envChan[0]->encEnvData.ldGrid, v_tuning); h_envChan[0]->encEnvData.hSbrBSGrid = &h_envChan[0]->SbrEnvFrame.SbrGrid; /* AAC LD patch for transient prediction */ if (h_envChan[0]->encEnvData.ldGrid && eData[0].transient_info[2]) { /* if next frame will start with transient, set shortEnv to * numEnvelopes(shortend Envelope = shortEnv-1)*/ h_envChan[0]->SbrEnvFrame.SbrFrameInfo.shortEnv = h_envChan[0]->SbrEnvFrame.SbrFrameInfo.nEnvelopes; } switch (stereoMode) { case SBR_LEFT_RIGHT: case SBR_SWITCH_LRC: eData[1].frame_info = FDKsbrEnc_frameInfoGenerator( &h_envChan[1]->SbrEnvFrame, eData[1].transient_info, sbrBitstreamData->rightBorderFIX, h_envChan[1]->sbrExtractEnvelope.pre_transient_info, h_envChan[1]->encEnvData.ldGrid, v_tuning); h_envChan[1]->encEnvData.hSbrBSGrid = &h_envChan[1]->SbrEnvFrame.SbrGrid; if (h_envChan[1]->encEnvData.ldGrid && eData[1].transient_info[2]) { /* if next frame will start with transient, set shortEnv to * numEnvelopes(shortend Envelope = shortEnv-1)*/ h_envChan[1]->SbrEnvFrame.SbrFrameInfo.shortEnv = h_envChan[1]->SbrEnvFrame.SbrFrameInfo.nEnvelopes; } /* compare left and right frame_infos */ if (eData[0].frame_info->nEnvelopes != eData[1].frame_info->nEnvelopes) { stereoMode = SBR_LEFT_RIGHT; } else { for (i = 0; i < eData[0].frame_info->nEnvelopes + 1; i++) { if (eData[0].frame_info->borders[i] != eData[1].frame_info->borders[i]) { stereoMode = SBR_LEFT_RIGHT; break; } } for (i = 0; i < eData[0].frame_info->nEnvelopes; i++) { if (eData[0].frame_info->freqRes[i] != eData[1].frame_info->freqRes[i]) { stereoMode = SBR_LEFT_RIGHT; break; } } if (eData[0].frame_info->shortEnv != eData[1].frame_info->shortEnv) { stereoMode = SBR_LEFT_RIGHT; } } break; case SBR_COUPLING: eData[1].frame_info = eData[0].frame_info; h_envChan[1]->encEnvData.hSbrBSGrid = &h_envChan[0]->SbrEnvFrame.SbrGrid; break; case SBR_MONO: /* nothing to do */ break; default: FDK_ASSERT(0); } for (ch = 0; ch < nChannels; ch++) { HANDLE_ENV_CHANNEL hEnvChan = h_envChan[ch]; HANDLE_SBR_EXTRACT_ENVELOPE sbrExtrEnv = &hEnvChan->sbrExtractEnvelope; SBR_ENV_TEMP_DATA *ed = &eData[ch]; /* Send transient info to bitstream and store for next call */ sbrExtrEnv->pre_transient_info[0] = ed->transient_info[0]; /* tran_pos */ sbrExtrEnv->pre_transient_info[1] = ed->transient_info[1]; /* tran_flag */ hEnvChan->encEnvData.noOfEnvelopes = ed->nEnvelopes = ed->frame_info->nEnvelopes; /* number of envelopes of current frame */ /* Check if the current frame is divided into one envelope only. If so, set the amplitude resolution to 1.5 dB, otherwise may set back to chosen value */ if ((hEnvChan->encEnvData.hSbrBSGrid->frameClass == FIXFIX) && (ed->nEnvelopes == 1)) { if (h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { /* Note: global_tonaliy_float_value == ((float)hEnvChan->encEnvData.global_tonality/((INT64)(1)<<(31-(19+2)))/0.524288*(2.0/3.0))); threshold_float_value == ((float)h_con->thresholdAmpResFF_m/((INT64)(1)<<(31-(h_con->thresholdAmpResFF_e)))/0.524288*(2.0/3.0))); */ /* decision of SBR_AMP_RES */ if (fIsLessThan(/* global_tonality > threshold ? */ h_con->thresholdAmpResFF_m, h_con->thresholdAmpResFF_e, hEnvChan->encEnvData.global_tonality, RELAXATION_SHIFT + 2)) { hEnvChan->encEnvData.currentAmpResFF = SBR_AMP_RES_1_5; } else { hEnvChan->encEnvData.currentAmpResFF = SBR_AMP_RES_3_0; } } else hEnvChan->encEnvData.currentAmpResFF = SBR_AMP_RES_1_5; if (hEnvChan->encEnvData.currentAmpResFF != hEnvChan->encEnvData.init_sbr_amp_res) { FDKsbrEnc_InitSbrHuffmanTables( &hEnvChan->encEnvData, &hEnvChan->sbrCodeEnvelope, &hEnvChan->sbrCodeNoiseFloor, hEnvChan->encEnvData.currentAmpResFF); } } else { if (sbrHeaderData->sbr_amp_res != hEnvChan->encEnvData.init_sbr_amp_res) { FDKsbrEnc_InitSbrHuffmanTables( &hEnvChan->encEnvData, &hEnvChan->sbrCodeEnvelope, &hEnvChan->sbrCodeNoiseFloor, sbrHeaderData->sbr_amp_res); } } if (!clearOutput) { /* Tonality correction parameter extraction (inverse filtering level, noise floor additional sines). */ FDKsbrEnc_TonCorrParamExtr( &hEnvChan->TonCorr, hEnvChan->encEnvData.sbr_invf_mode_vec, ed->noiseFloor, &hEnvChan->encEnvData.addHarmonicFlag, hEnvChan->encEnvData.addHarmonic, sbrExtrEnv->envelopeCompensation, ed->frame_info, ed->transient_info, h_con->freqBandTable[HI], h_con->nSfb[HI], hEnvChan->encEnvData.sbr_xpos_mode, h_con->sbrSyntaxFlags); } /* Low energy in low band fix */ if (hEnvChan->sbrTransientDetector.prevLowBandEnergy < hEnvChan->sbrTransientDetector.prevHighBandEnergy && hEnvChan->sbrTransientDetector.prevHighBandEnergy > FL2FX_DBL(0.03) /* The fix needs the non-fast transient detector running. It sets prevLowBandEnergy and prevHighBandEnergy. */ && !(h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY)) { hEnvChan->fLevelProtect = 1; for (i = 0; i < MAX_NUM_NOISE_VALUES; i++) hEnvChan->encEnvData.sbr_invf_mode_vec[i] = INVF_HIGH_LEVEL; } else { hEnvChan->fLevelProtect = 0; } hEnvChan->encEnvData.sbr_invf_mode = hEnvChan->encEnvData.sbr_invf_mode_vec[0]; hEnvChan->encEnvData.noOfnoisebands = hEnvChan->TonCorr.sbrNoiseFloorEstimate.noNoiseBands; } /* ch */ /* Save number of scf bands per envelope */ for (ch = 0; ch < nChannels; ch++) { for (i = 0; i < eData[ch].nEnvelopes; i++) { h_envChan[ch]->encEnvData.noScfBands[i] = (eData[ch].frame_info->freqRes[i] == FREQ_RES_HIGH ? h_con->nSfb[FREQ_RES_HIGH] : h_con->nSfb[FREQ_RES_LOW]); } } /* Extract envelope of current frame. */ switch (stereoMode) { case SBR_MONO: calculateSbrEnvelope(h_envChan[0]->sbrExtractEnvelope.YBuffer, NULL, h_envChan[0]->sbrExtractEnvelope.YBufferScale, NULL, eData[0].frame_info, eData[0].sfb_nrg, NULL, h_con, h_envChan[0], SBR_MONO, NULL, YSzShift); break; case SBR_LEFT_RIGHT: calculateSbrEnvelope(h_envChan[0]->sbrExtractEnvelope.YBuffer, NULL, h_envChan[0]->sbrExtractEnvelope.YBufferScale, NULL, eData[0].frame_info, eData[0].sfb_nrg, NULL, h_con, h_envChan[0], SBR_MONO, NULL, YSzShift); calculateSbrEnvelope(h_envChan[1]->sbrExtractEnvelope.YBuffer, NULL, h_envChan[1]->sbrExtractEnvelope.YBufferScale, NULL, eData[1].frame_info, eData[1].sfb_nrg, NULL, h_con, h_envChan[1], SBR_MONO, NULL, YSzShift); break; case SBR_COUPLING: calculateSbrEnvelope(h_envChan[0]->sbrExtractEnvelope.YBuffer, h_envChan[1]->sbrExtractEnvelope.YBuffer, h_envChan[0]->sbrExtractEnvelope.YBufferScale, h_envChan[1]->sbrExtractEnvelope.YBufferScale, eData[0].frame_info, eData[0].sfb_nrg, eData[1].sfb_nrg, h_con, h_envChan[0], SBR_COUPLING, &fData->maxQuantError, YSzShift); break; case SBR_SWITCH_LRC: calculateSbrEnvelope(h_envChan[0]->sbrExtractEnvelope.YBuffer, NULL, h_envChan[0]->sbrExtractEnvelope.YBufferScale, NULL, eData[0].frame_info, eData[0].sfb_nrg, NULL, h_con, h_envChan[0], SBR_MONO, NULL, YSzShift); calculateSbrEnvelope(h_envChan[1]->sbrExtractEnvelope.YBuffer, NULL, h_envChan[1]->sbrExtractEnvelope.YBufferScale, NULL, eData[1].frame_info, eData[1].sfb_nrg, NULL, h_con, h_envChan[1], SBR_MONO, NULL, YSzShift); calculateSbrEnvelope(h_envChan[0]->sbrExtractEnvelope.YBuffer, h_envChan[1]->sbrExtractEnvelope.YBuffer, h_envChan[0]->sbrExtractEnvelope.YBufferScale, h_envChan[1]->sbrExtractEnvelope.YBufferScale, eData[0].frame_info, eData[0].sfb_nrg_coupling, eData[1].sfb_nrg_coupling, h_con, h_envChan[0], SBR_COUPLING, &fData->maxQuantError, YSzShift); break; } /* Noise floor quantisation and coding. */ switch (stereoMode) { case SBR_MONO: sbrNoiseFloorLevelsQuantisation(eData[0].noise_level, eData[0].noiseFloor, 0); FDKsbrEnc_codeEnvelope(eData[0].noise_level, fData->res, &h_envChan[0]->sbrCodeNoiseFloor, h_envChan[0]->encEnvData.domain_vec_noise, 0, (eData[0].frame_info->nEnvelopes > 1 ? 2 : 1), 0, sbrBitstreamData->HeaderActive); break; case SBR_LEFT_RIGHT: sbrNoiseFloorLevelsQuantisation(eData[0].noise_level, eData[0].noiseFloor, 0); FDKsbrEnc_codeEnvelope(eData[0].noise_level, fData->res, &h_envChan[0]->sbrCodeNoiseFloor, h_envChan[0]->encEnvData.domain_vec_noise, 0, (eData[0].frame_info->nEnvelopes > 1 ? 2 : 1), 0, sbrBitstreamData->HeaderActive); sbrNoiseFloorLevelsQuantisation(eData[1].noise_level, eData[1].noiseFloor, 0); FDKsbrEnc_codeEnvelope(eData[1].noise_level, fData->res, &h_envChan[1]->sbrCodeNoiseFloor, h_envChan[1]->encEnvData.domain_vec_noise, 0, (eData[1].frame_info->nEnvelopes > 1 ? 2 : 1), 0, sbrBitstreamData->HeaderActive); break; case SBR_COUPLING: coupleNoiseFloor(eData[0].noiseFloor, eData[1].noiseFloor); sbrNoiseFloorLevelsQuantisation(eData[0].noise_level, eData[0].noiseFloor, 0); FDKsbrEnc_codeEnvelope(eData[0].noise_level, fData->res, &h_envChan[0]->sbrCodeNoiseFloor, h_envChan[0]->encEnvData.domain_vec_noise, 1, (eData[0].frame_info->nEnvelopes > 1 ? 2 : 1), 0, sbrBitstreamData->HeaderActive); sbrNoiseFloorLevelsQuantisation(eData[1].noise_level, eData[1].noiseFloor, 1); FDKsbrEnc_codeEnvelope(eData[1].noise_level, fData->res, &h_envChan[1]->sbrCodeNoiseFloor, h_envChan[1]->encEnvData.domain_vec_noise, 1, (eData[1].frame_info->nEnvelopes > 1 ? 2 : 1), 1, sbrBitstreamData->HeaderActive); break; case SBR_SWITCH_LRC: sbrNoiseFloorLevelsQuantisation(eData[0].noise_level, eData[0].noiseFloor, 0); sbrNoiseFloorLevelsQuantisation(eData[1].noise_level, eData[1].noiseFloor, 0); coupleNoiseFloor(eData[0].noiseFloor, eData[1].noiseFloor); sbrNoiseFloorLevelsQuantisation(eData[0].noise_level_coupling, eData[0].noiseFloor, 0); sbrNoiseFloorLevelsQuantisation(eData[1].noise_level_coupling, eData[1].noiseFloor, 1); break; } /* Encode envelope of current frame. */ switch (stereoMode) { case SBR_MONO: sbrHeaderData->coupling = 0; h_envChan[0]->encEnvData.balance = 0; FDKsbrEnc_codeEnvelope( eData[0].sfb_nrg, eData[0].frame_info->freqRes, &h_envChan[0]->sbrCodeEnvelope, h_envChan[0]->encEnvData.domain_vec, sbrHeaderData->coupling, eData[0].frame_info->nEnvelopes, 0, sbrBitstreamData->HeaderActive); break; case SBR_LEFT_RIGHT: sbrHeaderData->coupling = 0; h_envChan[0]->encEnvData.balance = 0; h_envChan[1]->encEnvData.balance = 0; FDKsbrEnc_codeEnvelope( eData[0].sfb_nrg, eData[0].frame_info->freqRes, &h_envChan[0]->sbrCodeEnvelope, h_envChan[0]->encEnvData.domain_vec, sbrHeaderData->coupling, eData[0].frame_info->nEnvelopes, 0, sbrBitstreamData->HeaderActive); FDKsbrEnc_codeEnvelope( eData[1].sfb_nrg, eData[1].frame_info->freqRes, &h_envChan[1]->sbrCodeEnvelope, h_envChan[1]->encEnvData.domain_vec, sbrHeaderData->coupling, eData[1].frame_info->nEnvelopes, 0, sbrBitstreamData->HeaderActive); break; case SBR_COUPLING: sbrHeaderData->coupling = 1; h_envChan[0]->encEnvData.balance = 0; h_envChan[1]->encEnvData.balance = 1; FDKsbrEnc_codeEnvelope( eData[0].sfb_nrg, eData[0].frame_info->freqRes, &h_envChan[0]->sbrCodeEnvelope, h_envChan[0]->encEnvData.domain_vec, sbrHeaderData->coupling, eData[0].frame_info->nEnvelopes, 0, sbrBitstreamData->HeaderActive); FDKsbrEnc_codeEnvelope( eData[1].sfb_nrg, eData[1].frame_info->freqRes, &h_envChan[1]->sbrCodeEnvelope, h_envChan[1]->encEnvData.domain_vec, sbrHeaderData->coupling, eData[1].frame_info->nEnvelopes, 1, sbrBitstreamData->HeaderActive); break; case SBR_SWITCH_LRC: { INT payloadbitsLR; INT payloadbitsCOUPLING; SCHAR sfbNrgPrevTemp[MAX_NUM_CHANNELS][MAX_FREQ_COEFFS]; SCHAR noisePrevTemp[MAX_NUM_CHANNELS][MAX_NUM_NOISE_COEFFS]; INT upDateNrgTemp[MAX_NUM_CHANNELS]; INT upDateNoiseTemp[MAX_NUM_CHANNELS]; INT domainVecTemp[MAX_NUM_CHANNELS][MAX_ENVELOPES]; INT domainVecNoiseTemp[MAX_NUM_CHANNELS][MAX_ENVELOPES]; INT tempFlagRight = 0; INT tempFlagLeft = 0; /* Store previous values, in order to be able to "undo" what is being done. */ for (ch = 0; ch < nChannels; ch++) { FDKmemcpy(sfbNrgPrevTemp[ch], h_envChan[ch]->sbrCodeEnvelope.sfb_nrg_prev, MAX_FREQ_COEFFS * sizeof(SCHAR)); FDKmemcpy(noisePrevTemp[ch], h_envChan[ch]->sbrCodeNoiseFloor.sfb_nrg_prev, MAX_NUM_NOISE_COEFFS * sizeof(SCHAR)); upDateNrgTemp[ch] = h_envChan[ch]->sbrCodeEnvelope.upDate; upDateNoiseTemp[ch] = h_envChan[ch]->sbrCodeNoiseFloor.upDate; /* forbid time coding in the first envelope in case of a different previous stereomode */ if (sbrHeaderData->prev_coupling) { h_envChan[ch]->sbrCodeEnvelope.upDate = 0; h_envChan[ch]->sbrCodeNoiseFloor.upDate = 0; } } /* ch */ /* Code ordinary Left/Right stereo */ FDKsbrEnc_codeEnvelope(eData[0].sfb_nrg, eData[0].frame_info->freqRes, &h_envChan[0]->sbrCodeEnvelope, h_envChan[0]->encEnvData.domain_vec, 0, eData[0].frame_info->nEnvelopes, 0, sbrBitstreamData->HeaderActive); FDKsbrEnc_codeEnvelope(eData[1].sfb_nrg, eData[1].frame_info->freqRes, &h_envChan[1]->sbrCodeEnvelope, h_envChan[1]->encEnvData.domain_vec, 0, eData[1].frame_info->nEnvelopes, 0, sbrBitstreamData->HeaderActive); c = 0; for (i = 0; i < eData[0].nEnvelopes; i++) { for (j = 0; j < h_envChan[0]->encEnvData.noScfBands[i]; j++) { h_envChan[0]->encEnvData.ienvelope[i][j] = eData[0].sfb_nrg[c]; h_envChan[1]->encEnvData.ienvelope[i][j] = eData[1].sfb_nrg[c]; c++; } } FDKsbrEnc_codeEnvelope(eData[0].noise_level, fData->res, &h_envChan[0]->sbrCodeNoiseFloor, h_envChan[0]->encEnvData.domain_vec_noise, 0, (eData[0].frame_info->nEnvelopes > 1 ? 2 : 1), 0, sbrBitstreamData->HeaderActive); for (i = 0; i < MAX_NUM_NOISE_VALUES; i++) h_envChan[0]->encEnvData.sbr_noise_levels[i] = eData[0].noise_level[i]; FDKsbrEnc_codeEnvelope(eData[1].noise_level, fData->res, &h_envChan[1]->sbrCodeNoiseFloor, h_envChan[1]->encEnvData.domain_vec_noise, 0, (eData[1].frame_info->nEnvelopes > 1 ? 2 : 1), 0, sbrBitstreamData->HeaderActive); for (i = 0; i < MAX_NUM_NOISE_VALUES; i++) h_envChan[1]->encEnvData.sbr_noise_levels[i] = eData[1].noise_level[i]; sbrHeaderData->coupling = 0; h_envChan[0]->encEnvData.balance = 0; h_envChan[1]->encEnvData.balance = 0; payloadbitsLR = FDKsbrEnc_CountSbrChannelPairElement( sbrHeaderData, hParametricStereo, sbrBitstreamData, &h_envChan[0]->encEnvData, &h_envChan[1]->encEnvData, hCmonData, h_con->sbrSyntaxFlags); /* swap saved stored with current values */ for (ch = 0; ch < nChannels; ch++) { INT itmp; for (i = 0; i < MAX_FREQ_COEFFS; i++) { /* swap sfb energies */ itmp = h_envChan[ch]->sbrCodeEnvelope.sfb_nrg_prev[i]; h_envChan[ch]->sbrCodeEnvelope.sfb_nrg_prev[i] = sfbNrgPrevTemp[ch][i]; sfbNrgPrevTemp[ch][i] = itmp; } for (i = 0; i < MAX_NUM_NOISE_COEFFS; i++) { /* swap noise energies */ itmp = h_envChan[ch]->sbrCodeNoiseFloor.sfb_nrg_prev[i]; h_envChan[ch]->sbrCodeNoiseFloor.sfb_nrg_prev[i] = noisePrevTemp[ch][i]; noisePrevTemp[ch][i] = itmp; } /* swap update flags */ itmp = h_envChan[ch]->sbrCodeEnvelope.upDate; h_envChan[ch]->sbrCodeEnvelope.upDate = upDateNrgTemp[ch]; upDateNrgTemp[ch] = itmp; itmp = h_envChan[ch]->sbrCodeNoiseFloor.upDate; h_envChan[ch]->sbrCodeNoiseFloor.upDate = upDateNoiseTemp[ch]; upDateNoiseTemp[ch] = itmp; /* save domain vecs */ FDKmemcpy(domainVecTemp[ch], h_envChan[ch]->encEnvData.domain_vec, sizeof(INT) * MAX_ENVELOPES); FDKmemcpy(domainVecNoiseTemp[ch], h_envChan[ch]->encEnvData.domain_vec_noise, sizeof(INT) * MAX_ENVELOPES); /* forbid time coding in the first envelope in case of a different previous stereomode */ if (!sbrHeaderData->prev_coupling) { h_envChan[ch]->sbrCodeEnvelope.upDate = 0; h_envChan[ch]->sbrCodeNoiseFloor.upDate = 0; } } /* ch */ /* Coupling */ FDKsbrEnc_codeEnvelope( eData[0].sfb_nrg_coupling, eData[0].frame_info->freqRes, &h_envChan[0]->sbrCodeEnvelope, h_envChan[0]->encEnvData.domain_vec, 1, eData[0].frame_info->nEnvelopes, 0, sbrBitstreamData->HeaderActive); FDKsbrEnc_codeEnvelope( eData[1].sfb_nrg_coupling, eData[1].frame_info->freqRes, &h_envChan[1]->sbrCodeEnvelope, h_envChan[1]->encEnvData.domain_vec, 1, eData[1].frame_info->nEnvelopes, 1, sbrBitstreamData->HeaderActive); c = 0; for (i = 0; i < eData[0].nEnvelopes; i++) { for (j = 0; j < h_envChan[0]->encEnvData.noScfBands[i]; j++) { h_envChan[0]->encEnvData.ienvelope[i][j] = eData[0].sfb_nrg_coupling[c]; h_envChan[1]->encEnvData.ienvelope[i][j] = eData[1].sfb_nrg_coupling[c]; c++; } } FDKsbrEnc_codeEnvelope(eData[0].noise_level_coupling, fData->res, &h_envChan[0]->sbrCodeNoiseFloor, h_envChan[0]->encEnvData.domain_vec_noise, 1, (eData[0].frame_info->nEnvelopes > 1 ? 2 : 1), 0, sbrBitstreamData->HeaderActive); for (i = 0; i < MAX_NUM_NOISE_VALUES; i++) h_envChan[0]->encEnvData.sbr_noise_levels[i] = eData[0].noise_level_coupling[i]; FDKsbrEnc_codeEnvelope(eData[1].noise_level_coupling, fData->res, &h_envChan[1]->sbrCodeNoiseFloor, h_envChan[1]->encEnvData.domain_vec_noise, 1, (eData[1].frame_info->nEnvelopes > 1 ? 2 : 1), 1, sbrBitstreamData->HeaderActive); for (i = 0; i < MAX_NUM_NOISE_VALUES; i++) h_envChan[1]->encEnvData.sbr_noise_levels[i] = eData[1].noise_level_coupling[i]; sbrHeaderData->coupling = 1; h_envChan[0]->encEnvData.balance = 0; h_envChan[1]->encEnvData.balance = 1; tempFlagLeft = h_envChan[0]->encEnvData.addHarmonicFlag; tempFlagRight = h_envChan[1]->encEnvData.addHarmonicFlag; payloadbitsCOUPLING = FDKsbrEnc_CountSbrChannelPairElement( sbrHeaderData, hParametricStereo, sbrBitstreamData, &h_envChan[0]->encEnvData, &h_envChan[1]->encEnvData, hCmonData, h_con->sbrSyntaxFlags); h_envChan[0]->encEnvData.addHarmonicFlag = tempFlagLeft; h_envChan[1]->encEnvData.addHarmonicFlag = tempFlagRight; if (payloadbitsCOUPLING < payloadbitsLR) { /* copy coded coupling envelope and noise data to l/r */ for (ch = 0; ch < nChannels; ch++) { SBR_ENV_TEMP_DATA *ed = &eData[ch]; FDKmemcpy(ed->sfb_nrg, ed->sfb_nrg_coupling, MAX_NUM_ENVELOPE_VALUES * sizeof(SCHAR)); FDKmemcpy(ed->noise_level, ed->noise_level_coupling, MAX_NUM_NOISE_VALUES * sizeof(SCHAR)); } sbrHeaderData->coupling = 1; h_envChan[0]->encEnvData.balance = 0; h_envChan[1]->encEnvData.balance = 1; } else { /* restore saved l/r items */ for (ch = 0; ch < nChannels; ch++) { FDKmemcpy(h_envChan[ch]->sbrCodeEnvelope.sfb_nrg_prev, sfbNrgPrevTemp[ch], MAX_FREQ_COEFFS * sizeof(SCHAR)); h_envChan[ch]->sbrCodeEnvelope.upDate = upDateNrgTemp[ch]; FDKmemcpy(h_envChan[ch]->sbrCodeNoiseFloor.sfb_nrg_prev, noisePrevTemp[ch], MAX_NUM_NOISE_COEFFS * sizeof(SCHAR)); FDKmemcpy(h_envChan[ch]->encEnvData.domain_vec, domainVecTemp[ch], sizeof(INT) * MAX_ENVELOPES); FDKmemcpy(h_envChan[ch]->encEnvData.domain_vec_noise, domainVecNoiseTemp[ch], sizeof(INT) * MAX_ENVELOPES); h_envChan[ch]->sbrCodeNoiseFloor.upDate = upDateNoiseTemp[ch]; } sbrHeaderData->coupling = 0; h_envChan[0]->encEnvData.balance = 0; h_envChan[1]->encEnvData.balance = 0; } } break; } /* switch */ /* tell the envelope encoders how long it has been, since we last sent a frame starting with a dF-coded envelope */ if (stereoMode == SBR_MONO) { if (h_envChan[0]->encEnvData.domain_vec[0] == TIME) h_envChan[0]->sbrCodeEnvelope.dF_edge_incr_fac++; else h_envChan[0]->sbrCodeEnvelope.dF_edge_incr_fac = 0; } else { if (h_envChan[0]->encEnvData.domain_vec[0] == TIME || h_envChan[1]->encEnvData.domain_vec[0] == TIME) { h_envChan[0]->sbrCodeEnvelope.dF_edge_incr_fac++; h_envChan[1]->sbrCodeEnvelope.dF_edge_incr_fac++; } else { h_envChan[0]->sbrCodeEnvelope.dF_edge_incr_fac = 0; h_envChan[1]->sbrCodeEnvelope.dF_edge_incr_fac = 0; } } /* Send the encoded data to the bitstream */ for (ch = 0; ch < nChannels; ch++) { SBR_ENV_TEMP_DATA *ed = &eData[ch]; c = 0; for (i = 0; i < ed->nEnvelopes; i++) { for (j = 0; j < h_envChan[ch]->encEnvData.noScfBands[i]; j++) { h_envChan[ch]->encEnvData.ienvelope[i][j] = ed->sfb_nrg[c]; c++; } } for (i = 0; i < MAX_NUM_NOISE_VALUES; i++) { h_envChan[ch]->encEnvData.sbr_noise_levels[i] = ed->noise_level[i]; } } /* ch */ /* Write bitstream */ if (nChannels == 2) { FDKsbrEnc_WriteEnvChannelPairElement( sbrHeaderData, hParametricStereo, sbrBitstreamData, &h_envChan[0]->encEnvData, &h_envChan[1]->encEnvData, hCmonData, h_con->sbrSyntaxFlags); } else { FDKsbrEnc_WriteEnvSingleChannelElement( sbrHeaderData, hParametricStereo, sbrBitstreamData, &h_envChan[0]->encEnvData, hCmonData, h_con->sbrSyntaxFlags); } /* * Update buffers. */ for (ch = 0; ch < nChannels; ch++) { int YBufferLength = h_envChan[ch]->sbrExtractEnvelope.no_cols >> h_envChan[ch]->sbrExtractEnvelope.YBufferSzShift; for (i = 0; i < h_envChan[ch]->sbrExtractEnvelope.YBufferWriteOffset; i++) { FDKmemcpy(h_envChan[ch]->sbrExtractEnvelope.YBuffer[i], h_envChan[ch]->sbrExtractEnvelope.YBuffer[i + YBufferLength], sizeof(FIXP_DBL) * 64); } h_envChan[ch]->sbrExtractEnvelope.YBufferScale[0] = h_envChan[ch]->sbrExtractEnvelope.YBufferScale[1]; } sbrHeaderData->prev_coupling = sbrHeaderData->coupling; } /***************************************************************************/ /*! \brief creates an envelope extractor handle \return error status ****************************************************************************/ INT FDKsbrEnc_CreateExtractSbrEnvelope(HANDLE_SBR_EXTRACT_ENVELOPE hSbrCut, INT channel, INT chInEl, UCHAR *dynamic_RAM) { INT i; FIXP_DBL *rBuffer, *iBuffer; INT n; FIXP_DBL *YBufferDyn; FDKmemclear(hSbrCut, sizeof(SBR_EXTRACT_ENVELOPE)); if (NULL == (hSbrCut->p_YBuffer = GetRam_Sbr_envYBuffer(channel))) { goto bail; } for (i = 0; i < (32 >> 1); i++) { hSbrCut->YBuffer[i] = hSbrCut->p_YBuffer + (i * 64); } YBufferDyn = GetRam_Sbr_envYBuffer(chInEl, dynamic_RAM); for (n = 0; i < 32; i++, n++) { hSbrCut->YBuffer[i] = YBufferDyn + (n * 64); } rBuffer = GetRam_Sbr_envRBuffer(0, dynamic_RAM); iBuffer = GetRam_Sbr_envIBuffer(0, dynamic_RAM); for (i = 0; i < 32; i++) { hSbrCut->rBuffer[i] = rBuffer + (i * 64); hSbrCut->iBuffer[i] = iBuffer + (i * 64); } return 0; bail: FDKsbrEnc_deleteExtractSbrEnvelope(hSbrCut); return -1; } /***************************************************************************/ /*! \brief Initialize an envelope extractor instance. \return error status ****************************************************************************/ INT FDKsbrEnc_InitExtractSbrEnvelope(HANDLE_SBR_EXTRACT_ENVELOPE hSbrCut, int no_cols, int no_rows, int start_index, int time_slots, int time_step, int tran_off, ULONG statesInitFlag, int chInEl, UCHAR *dynamic_RAM, UINT sbrSyntaxFlags) { int YBufferLength, rBufferLength; int i; if (sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { int off = TRANSIENT_OFFSET_LD; hSbrCut->YBufferWriteOffset = (no_cols >> 1) + off * time_step; } else { hSbrCut->YBufferWriteOffset = tran_off * time_step; } hSbrCut->rBufferReadOffset = 0; YBufferLength = hSbrCut->YBufferWriteOffset + no_cols; rBufferLength = no_cols; hSbrCut->pre_transient_info[0] = 0; hSbrCut->pre_transient_info[1] = 0; hSbrCut->no_cols = no_cols; hSbrCut->no_rows = no_rows; hSbrCut->start_index = start_index; hSbrCut->time_slots = time_slots; hSbrCut->time_step = time_step; FDK_ASSERT(no_rows <= 64); /* Use half the Energy values if time step is 2 or greater */ if (time_step >= 2) hSbrCut->YBufferSzShift = 1; else hSbrCut->YBufferSzShift = 0; YBufferLength >>= hSbrCut->YBufferSzShift; hSbrCut->YBufferWriteOffset >>= hSbrCut->YBufferSzShift; FDK_ASSERT(YBufferLength <= 32); FIXP_DBL *YBufferDyn = GetRam_Sbr_envYBuffer(chInEl, dynamic_RAM); INT n = 0; for (i = (32 >> 1); i < 32; i++, n++) { hSbrCut->YBuffer[i] = YBufferDyn + (n * 64); } if (statesInitFlag) { for (i = 0; i < YBufferLength; i++) { FDKmemclear(hSbrCut->YBuffer[i], 64 * sizeof(FIXP_DBL)); } } for (i = 0; i < rBufferLength; i++) { FDKmemclear(hSbrCut->rBuffer[i], 64 * sizeof(FIXP_DBL)); FDKmemclear(hSbrCut->iBuffer[i], 64 * sizeof(FIXP_DBL)); } FDKmemclear(hSbrCut->envelopeCompensation, sizeof(UCHAR) * MAX_FREQ_COEFFS); if (statesInitFlag) { hSbrCut->YBufferScale[0] = hSbrCut->YBufferScale[1] = FRACT_BITS - 1; } return (0); } /***************************************************************************/ /*! \brief deinitializes an envelope extractor handle \return void ****************************************************************************/ void FDKsbrEnc_deleteExtractSbrEnvelope(HANDLE_SBR_EXTRACT_ENVELOPE hSbrCut) { if (hSbrCut) { FreeRam_Sbr_envYBuffer(&hSbrCut->p_YBuffer); } } INT FDKsbrEnc_GetEnvEstDelay(HANDLE_SBR_EXTRACT_ENVELOPE hSbr) { return hSbr->no_rows * ((hSbr->YBufferWriteOffset) * 2 /* mult 2 because nrg's are grouped half */ - hSbr->rBufferReadOffset); /* in reference hold half spec and calc nrg's on overlapped spec */ } fdk-aac-2.0.2/libSBRenc/src/env_est.h000066400000000000000000000210421404225310700172040ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Envelope estimation structs and prototypes $Revision: 92790 $ */ #ifndef ENV_EST_H #define ENV_EST_H #include "sbr_def.h" #include "sbr_encoder.h" /* SBR econfig structs */ #include "ps_main.h" #include "bit_sbr.h" #include "fram_gen.h" #include "tran_det.h" #include "code_env.h" #include "ton_corr.h" typedef struct { FIXP_DBL *rBuffer[32]; FIXP_DBL *iBuffer[32]; FIXP_DBL *p_YBuffer; FIXP_DBL *YBuffer[32]; int YBufferScale[2]; UCHAR envelopeCompensation[MAX_FREQ_COEFFS]; UCHAR pre_transient_info[2]; int YBufferWriteOffset; int YBufferSzShift; int rBufferReadOffset; int no_cols; int no_rows; int start_index; int time_slots; int time_step; } SBR_EXTRACT_ENVELOPE; typedef SBR_EXTRACT_ENVELOPE *HANDLE_SBR_EXTRACT_ENVELOPE; struct ENV_CHANNEL { FAST_TRAN_DETECTOR sbrFastTransientDetector; SBR_TRANSIENT_DETECTOR sbrTransientDetector; SBR_CODE_ENVELOPE sbrCodeEnvelope; SBR_CODE_ENVELOPE sbrCodeNoiseFloor; SBR_EXTRACT_ENVELOPE sbrExtractEnvelope; SBR_ENVELOPE_FRAME SbrEnvFrame; SBR_TON_CORR_EST TonCorr; struct SBR_ENV_DATA encEnvData; int qmfScale; UCHAR fLevelProtect; }; typedef struct ENV_CHANNEL *HANDLE_ENV_CHANNEL; /************ Function Declarations ***************/ INT FDKsbrEnc_CreateExtractSbrEnvelope(HANDLE_SBR_EXTRACT_ENVELOPE hSbrCut, INT channel, INT chInEl, UCHAR *dynamic_RAM); INT FDKsbrEnc_InitExtractSbrEnvelope(HANDLE_SBR_EXTRACT_ENVELOPE hSbr, int no_cols, int no_rows, int start_index, int time_slots, int time_step, int tran_off, ULONG statesInitFlag, int chInEl, UCHAR *dynamic_RAM, UINT sbrSyntaxFlags); void FDKsbrEnc_deleteExtractSbrEnvelope(HANDLE_SBR_EXTRACT_ENVELOPE hSbrCut); typedef struct { FREQ_RES res[MAX_NUM_NOISE_VALUES]; int maxQuantError; } SBR_FRAME_TEMP_DATA; typedef struct { const SBR_FRAME_INFO *frame_info; FIXP_DBL noiseFloor[MAX_NUM_NOISE_VALUES]; SCHAR sfb_nrg_coupling [MAX_NUM_ENVELOPE_VALUES]; /* only used if stereomode = SWITCH_L_R_C */ SCHAR sfb_nrg[MAX_NUM_ENVELOPE_VALUES]; SCHAR noise_level_coupling [MAX_NUM_NOISE_VALUES]; /* only used if stereomode = SWITCH_L_R_C */ SCHAR noise_level[MAX_NUM_NOISE_VALUES]; UCHAR transient_info[3]; UCHAR nEnvelopes; } SBR_ENV_TEMP_DATA; /* * Extract features from QMF data. Afterwards, the QMF data is not required * anymore. */ void FDKsbrEnc_extractSbrEnvelope1(HANDLE_SBR_CONFIG_DATA h_con, HANDLE_SBR_HEADER_DATA sbrHeaderData, HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, HANDLE_ENV_CHANNEL h_envChan, HANDLE_COMMON_DATA cmonData, SBR_ENV_TEMP_DATA *eData, SBR_FRAME_TEMP_DATA *fData); /* * Process the previously features extracted by FDKsbrEnc_extractSbrEnvelope1 * and create/encode SBR envelopes. */ void FDKsbrEnc_extractSbrEnvelope2(HANDLE_SBR_CONFIG_DATA h_con, HANDLE_SBR_HEADER_DATA sbrHeaderData, HANDLE_PARAMETRIC_STEREO hParametricStereo, HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, HANDLE_ENV_CHANNEL sbrEnvChannel0, HANDLE_ENV_CHANNEL sbrEnvChannel1, HANDLE_COMMON_DATA cmonData, SBR_ENV_TEMP_DATA *eData, SBR_FRAME_TEMP_DATA *fData, int clearOutput); INT FDKsbrEnc_GetEnvEstDelay(HANDLE_SBR_EXTRACT_ENVELOPE hSbr); #endif fdk-aac-2.0.2/libSBRenc/src/fram_gen.cpp000066400000000000000000002102331404225310700176540ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Description: *******************************************************************************/ #include "fram_gen.h" #include "sbr_misc.h" #include "genericStds.h" static const SBR_FRAME_INFO frameInfo1_2048 = {1, {0, 16}, {FREQ_RES_HIGH}, 0, 1, {0, 16}}; static const SBR_FRAME_INFO frameInfo2_2048 = { 2, {0, 8, 16}, {FREQ_RES_HIGH, FREQ_RES_HIGH}, 0, 2, {0, 8, 16}}; static const SBR_FRAME_INFO frameInfo4_2048 = { 4, {0, 4, 8, 12, 16}, {FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH}, 0, 2, {0, 8, 16}}; static const SBR_FRAME_INFO frameInfo1_2304 = {1, {0, 18}, {FREQ_RES_HIGH}, 0, 1, {0, 18}}; static const SBR_FRAME_INFO frameInfo2_2304 = { 2, {0, 9, 18}, {FREQ_RES_HIGH, FREQ_RES_HIGH}, 0, 2, {0, 9, 18}}; static const SBR_FRAME_INFO frameInfo4_2304 = { 4, {0, 5, 9, 14, 18}, {FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH}, 0, 2, {0, 9, 18}}; static const SBR_FRAME_INFO frameInfo1_1920 = {1, {0, 15}, {FREQ_RES_HIGH}, 0, 1, {0, 15}}; static const SBR_FRAME_INFO frameInfo2_1920 = { 2, {0, 8, 15}, {FREQ_RES_HIGH, FREQ_RES_HIGH}, 0, 2, {0, 8, 15}}; static const SBR_FRAME_INFO frameInfo4_1920 = { 4, {0, 4, 8, 12, 15}, {FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH}, 0, 2, {0, 8, 15}}; static const SBR_FRAME_INFO frameInfo1_1152 = {1, {0, 9}, {FREQ_RES_HIGH}, 0, 1, {0, 9}}; static const SBR_FRAME_INFO frameInfo2_1152 = { 2, {0, 5, 9}, {FREQ_RES_HIGH, FREQ_RES_HIGH}, 0, 2, {0, 5, 9}}; static const SBR_FRAME_INFO frameInfo4_1152 = { 4, {0, 2, 5, 7, 9}, {FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH}, 0, 2, {0, 5, 9}}; /* AACLD frame info */ static const SBR_FRAME_INFO frameInfo1_512LD = {1, {0, 8}, {FREQ_RES_HIGH}, 0, 1, {0, 8}}; static const SBR_FRAME_INFO frameInfo2_512LD = { 2, {0, 4, 8}, {FREQ_RES_HIGH, FREQ_RES_HIGH}, 0, 2, {0, 4, 8}}; static const SBR_FRAME_INFO frameInfo4_512LD = { 4, {0, 2, 4, 6, 8}, {FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH}, 0, 2, {0, 4, 8}}; static int calcFillLengthMax( int tranPos, /*!< input : transient position (ref: tran det) */ int numberTimeSlots /*!< input : number of timeslots */ ); static void fillFrameTran( const int *v_tuningSegm, /*!< tuning: desired segment lengths */ const int *v_tuningFreq, /*!< tuning: desired frequency resolutions */ int tran, /*!< input : position of transient */ int *v_bord, /*!< memNew: borders */ int *length_v_bord, /*!< memNew: # borders */ int *v_freq, /*!< memNew: frequency resolutions */ int *length_v_freq, /*!< memNew: # frequency resolutions */ int *bmin, /*!< hlpNew: first mandatory border */ int *bmax /*!< hlpNew: last mandatory border */ ); static void fillFramePre(INT dmax, INT *v_bord, INT *length_v_bord, INT *v_freq, INT *length_v_freq, INT bmin, INT rest); static void fillFramePost(INT *parts, INT *d, INT dmax, INT *v_bord, INT *length_v_bord, INT *v_freq, INT *length_v_freq, INT bmax, INT bufferFrameStart, INT numberTimeSlots, INT fmax); static void fillFrameInter(INT *nL, const int *v_tuningSegm, INT *v_bord, INT *length_v_bord, INT bmin, INT *v_freq, INT *length_v_freq, INT *v_bordFollow, INT *length_v_bordFollow, INT *v_freqFollow, INT *length_v_freqFollow, INT i_fillFollow, INT dmin, INT dmax, INT numberTimeSlots); static void calcFrameClass(FRAME_CLASS *frameClass, FRAME_CLASS *frameClassOld, INT tranFlag, INT *spreadFlag); static void specialCase(INT *spreadFlag, INT allowSpread, INT *v_bord, INT *length_v_bord, INT *v_freq, INT *length_v_freq, INT *parts, INT d); static void calcCmonBorder(INT *i_cmon, INT *i_tran, INT *v_bord, INT *length_v_bord, INT tran, INT bufferFrameStart, INT numberTimeSlots); static void keepForFollowUp(INT *v_bordFollow, INT *length_v_bordFollow, INT *v_freqFollow, INT *length_v_freqFollow, INT *i_tranFollow, INT *i_fillFollow, INT *v_bord, INT *length_v_bord, INT *v_freq, INT i_cmon, INT i_tran, INT parts, INT numberTimeSlots); static void calcCtrlSignal(HANDLE_SBR_GRID hSbrGrid, FRAME_CLASS frameClass, INT *v_bord, INT length_v_bord, INT *v_freq, INT length_v_freq, INT i_cmon, INT i_tran, INT spreadFlag, INT nL); static void ctrlSignal2FrameInfo(HANDLE_SBR_GRID hSbrGrid, HANDLE_SBR_FRAME_INFO hFrameInfo, FREQ_RES *freq_res_fixfix); /* table for 8 time slot index */ static const int envelopeTable_8[8][5] = { /* transientIndex nEnv, tranIdx, shortEnv, border1, border2, ... */ /* borders from left to right side; -1 = not in use */ /*[|T-|------]*/ {2, 0, 0, 1, -1}, /*[|-T-|-----]*/ {2, 0, 0, 2, -1}, /*[--|T-|----]*/ {3, 1, 1, 2, 4}, /*[---|T-|---]*/ {3, 1, 1, 3, 5}, /*[----|T-|--]*/ {3, 1, 1, 4, 6}, /*[-----|T--|]*/ {2, 1, 1, 5, -1}, /*[------|T-|]*/ {2, 1, 1, 6, -1}, /*[-------|T|]*/ {2, 1, 1, 7, -1}, }; /* table for 16 time slot index */ static const int envelopeTable_16[16][6] = { /* transientIndex nEnv, tranIdx, shortEnv, border1, border2, ... */ /* length from left to right side; -1 = not in use */ /*[|T---|------------|]*/ {2, 0, 0, 4, -1, -1}, /*[|-T---|-----------|]*/ {2, 0, 0, 5, -1, -1}, /*[|--|T---|----------]*/ {3, 1, 1, 2, 6, -1}, /*[|---|T---|---------]*/ {3, 1, 1, 3, 7, -1}, /*[|----|T---|--------]*/ {3, 1, 1, 4, 8, -1}, /*[|-----|T---|-------]*/ {3, 1, 1, 5, 9, -1}, /*[|------|T---|------]*/ {3, 1, 1, 6, 10, -1}, /*[|-------|T---|-----]*/ {3, 1, 1, 7, 11, -1}, /*[|--------|T---|----]*/ {3, 1, 1, 8, 12, -1}, /*[|---------|T---|---]*/ {3, 1, 1, 9, 13, -1}, /*[|----------|T---|--]*/ {3, 1, 1, 10, 14, -1}, /*[|-----------|T----|]*/ {2, 1, 1, 11, -1, -1}, /*[|------------|T---|]*/ {2, 1, 1, 12, -1, -1}, /*[|-------------|T--|]*/ {2, 1, 1, 13, -1, -1}, /*[|--------------|T-|]*/ {2, 1, 1, 14, -1, -1}, /*[|---------------|T|]*/ {2, 1, 1, 15, -1, -1}, }; /* table for 15 time slot index */ static const int envelopeTable_15[15][6] = { /* transientIndex nEnv, tranIdx, shortEnv, border1, border2, ... */ /* length from left to right side; -1 = not in use */ /*[|T---|------------]*/ {2, 0, 0, 4, -1, -1}, /*[|-T---|-----------]*/ {2, 0, 0, 5, -1, -1}, /*[|--|T---|---------]*/ {3, 1, 1, 2, 6, -1}, /*[|---|T---|--------]*/ {3, 1, 1, 3, 7, -1}, /*[|----|T---|-------]*/ {3, 1, 1, 4, 8, -1}, /*[|-----|T---|------]*/ {3, 1, 1, 5, 9, -1}, /*[|------|T---|-----]*/ {3, 1, 1, 6, 10, -1}, /*[|-------|T---|----]*/ {3, 1, 1, 7, 11, -1}, /*[|--------|T---|---]*/ {3, 1, 1, 8, 12, -1}, /*[|---------|T---|--]*/ {3, 1, 1, 9, 13, -1}, /*[|----------|T----|]*/ {2, 1, 1, 10, -1, -1}, /*[|-----------|T---|]*/ {2, 1, 1, 11, -1, -1}, /*[|------------|T--|]*/ {2, 1, 1, 12, -1, -1}, /*[|-------------|T-|]*/ {2, 1, 1, 13, -1, -1}, /*[|--------------|T|]*/ {2, 1, 1, 14, -1, -1}, }; static const int minFrameTranDistance = 4; static const FREQ_RES freqRes_table_8[] = { FREQ_RES_LOW, FREQ_RES_LOW, FREQ_RES_LOW, FREQ_RES_LOW, FREQ_RES_LOW, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH}; static const FREQ_RES freqRes_table_16[16] = { /* size of envelope */ /* 0-4 */ FREQ_RES_LOW, FREQ_RES_LOW, FREQ_RES_LOW, FREQ_RES_LOW, FREQ_RES_LOW, /* 5-9 */ FREQ_RES_LOW, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, /* 10-16 */ FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH}; static void generateFixFixOnly(HANDLE_SBR_FRAME_INFO hSbrFrameInfo, HANDLE_SBR_GRID hSbrGrid, int tranPosInternal, int numberTimeSlots, UCHAR fResTransIsLow); /*! Functionname: FDKsbrEnc_frameInfoGenerator Description: produces the FRAME_INFO struct for the current frame Arguments: hSbrEnvFrame - pointer to sbr envelope handle v_pre_transient_info - pointer to transient info vector v_transient_info - pointer to previous transient info vector v_tuning - pointer to tuning vector Return: frame_info - pointer to SBR_FRAME_INFO struct *******************************************************************************/ HANDLE_SBR_FRAME_INFO FDKsbrEnc_frameInfoGenerator(HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame, UCHAR *v_transient_info, const INT rightBorderFIX, UCHAR *v_transient_info_pre, int ldGrid, const int *v_tuning) { INT numEnv, tranPosInternal = 0, bmin = 0, bmax = 0, parts, d, i_cmon = 0, i_tran = 0, nL; INT fmax = 0; INT *v_bord = hSbrEnvFrame->v_bord; INT *v_freq = hSbrEnvFrame->v_freq; INT *v_bordFollow = hSbrEnvFrame->v_bordFollow; INT *v_freqFollow = hSbrEnvFrame->v_freqFollow; INT *length_v_bordFollow = &hSbrEnvFrame->length_v_bordFollow; INT *length_v_freqFollow = &hSbrEnvFrame->length_v_freqFollow; INT *length_v_bord = &hSbrEnvFrame->length_v_bord; INT *length_v_freq = &hSbrEnvFrame->length_v_freq; INT *spreadFlag = &hSbrEnvFrame->spreadFlag; INT *i_tranFollow = &hSbrEnvFrame->i_tranFollow; INT *i_fillFollow = &hSbrEnvFrame->i_fillFollow; FRAME_CLASS *frameClassOld = &hSbrEnvFrame->frameClassOld; FRAME_CLASS frameClass = FIXFIX; INT allowSpread = hSbrEnvFrame->allowSpread; INT numEnvStatic = hSbrEnvFrame->numEnvStatic; INT staticFraming = hSbrEnvFrame->staticFraming; INT dmin = hSbrEnvFrame->dmin; INT dmax = hSbrEnvFrame->dmax; INT bufferFrameStart = hSbrEnvFrame->SbrGrid.bufferFrameStart; INT numberTimeSlots = hSbrEnvFrame->SbrGrid.numberTimeSlots; INT frameMiddleSlot = hSbrEnvFrame->frameMiddleSlot; INT tranPos = v_transient_info[0]; INT tranFlag = v_transient_info[1]; const int *v_tuningSegm = v_tuning; const int *v_tuningFreq = v_tuning + 3; hSbrEnvFrame->v_tuningSegm = v_tuningSegm; if (ldGrid) { /* in case there was a transient at the very end of the previous frame, * start with a transient envelope */ if (!tranFlag && v_transient_info_pre[1] && (numberTimeSlots - v_transient_info_pre[0] < minFrameTranDistance)) { tranFlag = 1; tranPos = 0; } } /* * Synopsis: * * The frame generator creates the time-/frequency-grid for one SBR frame. * Input signals are provided by the transient detector and the frame * splitter (transientDetectNew() & FrameSplitter() in tran_det.c). The * framing is controlled by adjusting tuning parameters stored in * FRAME_GEN_TUNING. The parameter values are dependent on frame lengths * and bitrates, and may in the future be signal dependent. * * The envelope borders are stored for frame generator internal use in * aBorders. The contents of aBorders represent positions along the time * axis given in the figures in fram_gen.h (the "frame-generator" rows). * The unit is "time slot". The figures in fram_gen.h also define the * detection ranges for the transient detector. For every border in * aBorders, there is a corresponding entry in aFreqRes, which defines the * frequency resolution of the envelope following (delimited by) the * border. * * When no transients are present, FIXFIX class frames are used. The * frame splitter decides whether to use one or two envelopes in the * FIXFIX frame. "Sparse transients" (separated by a few frames without * transients) are handeled by [FIXVAR, VARFIX] pairs or (depending on * tuning and transient position relative the nominal frame boundaries) * by [FIXVAR, VARVAR, VARFIX] triples. "Tight transients" (in * consecutive frames) are handeled by [..., VARVAR, VARVAR, ...] * sequences. * * The generator assumes that transients are "sparse", and designs * borders for [FIXVAR, VARFIX] pairs right away, where the first frame * corresponds to the present frame. At the next call of the generator * it is known whether the transient actually is "sparse" or not. If * 'yes', the already calculated VARFIX borders are used. If 'no', new * borders, meeting the requirements of the "tight" transient, are * calculated. * * The generator produces two outputs: A "clear-text bitstream" stored in * SBR_GRID, and a straight-forward representation of the grid stored in * SBR_FRAME_INFO. The former is subsequently converted to the actual * bitstream sbr_grid() (encodeSbrGrid() in bit_sbr.c). The latter is * used by other encoder functions, such as the envelope estimator * (calculateSbrEnvelope() in env_est.c) and the noise floor and missing * harmonics detector (TonCorrParamExtr() in nf_est.c). */ if (staticFraming) { /*-------------------------------------------------------------------------- Ignore transient detector ---------------------------------------------------------------------------*/ frameClass = FIXFIX; numEnv = numEnvStatic; /* {1,2,4,8} */ *frameClassOld = FIXFIX; /* for change to dyn */ hSbrEnvFrame->SbrGrid.bs_num_env = numEnv; hSbrEnvFrame->SbrGrid.frameClass = frameClass; } else { /*-------------------------------------------------------------------------- Calculate frame class to use ---------------------------------------------------------------------------*/ if (rightBorderFIX) { tranFlag = 0; *spreadFlag = 0; } calcFrameClass(&frameClass, frameClassOld, tranFlag, spreadFlag); /* patch for new frame class FIXFIXonly for AAC LD */ if (tranFlag && ldGrid) { frameClass = FIXFIXonly; *frameClassOld = FIXFIX; } /* * every transient is processed below by inserting * * - one border at the onset of the transient * - one or more "decay borders" (after the onset of the transient) * - optionally one "attack border" (before the onset of the transient) * * those borders are referred to as "mandatory borders" and are * defined by the 'segmentLength' array in FRAME_GEN_TUNING * * the frequency resolutions of the corresponding envelopes are * defined by the 'segmentRes' array in FRAME_GEN_TUNING */ /*-------------------------------------------------------------------------- Design frame (or follow-up old design) ---------------------------------------------------------------------------*/ if (tranFlag) { /* Always for FixVar, often but not always for VarVar */ /*-------------------------------------------------------------------------- Design part of T/F-grid around the new transient ---------------------------------------------------------------------------*/ tranPosInternal = frameMiddleSlot + tranPos + bufferFrameStart; /* FH 00-06-26 */ /* add mandatory borders around transient */ fillFrameTran(v_tuningSegm, v_tuningFreq, tranPosInternal, v_bord, length_v_bord, v_freq, length_v_freq, &bmin, &bmax); /* make sure we stay within the maximum SBR frame overlap */ fmax = calcFillLengthMax(tranPos, numberTimeSlots); } switch (frameClass) { case FIXFIXonly: FDK_ASSERT(ldGrid); tranPosInternal = tranPos; generateFixFixOnly(&(hSbrEnvFrame->SbrFrameInfo), &(hSbrEnvFrame->SbrGrid), tranPosInternal, numberTimeSlots, hSbrEnvFrame->fResTransIsLow); return &(hSbrEnvFrame->SbrFrameInfo); case FIXVAR: /*-------------------------------------------------------------------------- Design remaining parts of T/F-grid (assuming next frame is VarFix) ---------------------------------------------------------------------------*/ /*-------------------------------------------------------------------------- Fill region before new transient: ---------------------------------------------------------------------------*/ fillFramePre(dmax, v_bord, length_v_bord, v_freq, length_v_freq, bmin, bmin - bufferFrameStart); /* FH 00-06-26 */ /*-------------------------------------------------------------------------- Fill region after new transient: ---------------------------------------------------------------------------*/ fillFramePost(&parts, &d, dmax, v_bord, length_v_bord, v_freq, length_v_freq, bmax, bufferFrameStart, numberTimeSlots, fmax); /*-------------------------------------------------------------------------- Take care of special case: ---------------------------------------------------------------------------*/ if (parts == 1 && d < dmin) /* no fill, short last envelope */ specialCase(spreadFlag, allowSpread, v_bord, length_v_bord, v_freq, length_v_freq, &parts, d); /*-------------------------------------------------------------------------- Calculate common border (split-point) ---------------------------------------------------------------------------*/ calcCmonBorder(&i_cmon, &i_tran, v_bord, length_v_bord, tranPosInternal, bufferFrameStart, numberTimeSlots); /* FH 00-06-26 */ /*-------------------------------------------------------------------------- Extract data for proper follow-up in next frame ---------------------------------------------------------------------------*/ keepForFollowUp(v_bordFollow, length_v_bordFollow, v_freqFollow, length_v_freqFollow, i_tranFollow, i_fillFollow, v_bord, length_v_bord, v_freq, i_cmon, i_tran, parts, numberTimeSlots); /* FH 00-06-26 */ /*-------------------------------------------------------------------------- Calculate control signal ---------------------------------------------------------------------------*/ calcCtrlSignal(&hSbrEnvFrame->SbrGrid, frameClass, v_bord, *length_v_bord, v_freq, *length_v_freq, i_cmon, i_tran, *spreadFlag, DC); break; case VARFIX: /*-------------------------------------------------------------------------- Follow-up old transient - calculate control signal ---------------------------------------------------------------------------*/ calcCtrlSignal(&hSbrEnvFrame->SbrGrid, frameClass, v_bordFollow, *length_v_bordFollow, v_freqFollow, *length_v_freqFollow, DC, *i_tranFollow, *spreadFlag, DC); break; case VARVAR: if (*spreadFlag) { /* spread across three frames */ /*-------------------------------------------------------------------------- Follow-up old transient - calculate control signal ---------------------------------------------------------------------------*/ calcCtrlSignal(&hSbrEnvFrame->SbrGrid, frameClass, v_bordFollow, *length_v_bordFollow, v_freqFollow, *length_v_freqFollow, DC, *i_tranFollow, *spreadFlag, DC); *spreadFlag = 0; /*-------------------------------------------------------------------------- Extract data for proper follow-up in next frame ---------------------------------------------------------------------------*/ v_bordFollow[0] = hSbrEnvFrame->SbrGrid.bs_abs_bord_1 - numberTimeSlots; /* FH 00-06-26 */ v_freqFollow[0] = 1; *length_v_bordFollow = 1; *length_v_freqFollow = 1; *i_tranFollow = -DC; *i_fillFollow = -DC; } else { /*-------------------------------------------------------------------------- Design remaining parts of T/F-grid (assuming next frame is VarFix) adapt or fill region before new transient: ---------------------------------------------------------------------------*/ fillFrameInter(&nL, v_tuningSegm, v_bord, length_v_bord, bmin, v_freq, length_v_freq, v_bordFollow, length_v_bordFollow, v_freqFollow, length_v_freqFollow, *i_fillFollow, dmin, dmax, numberTimeSlots); /*-------------------------------------------------------------------------- Fill after transient: ---------------------------------------------------------------------------*/ fillFramePost(&parts, &d, dmax, v_bord, length_v_bord, v_freq, length_v_freq, bmax, bufferFrameStart, numberTimeSlots, fmax); /*-------------------------------------------------------------------------- Take care of special case: ---------------------------------------------------------------------------*/ if (parts == 1 && d < dmin) /*% no fill, short last envelope */ specialCase(spreadFlag, allowSpread, v_bord, length_v_bord, v_freq, length_v_freq, &parts, d); /*-------------------------------------------------------------------------- Calculate common border (split-point) ---------------------------------------------------------------------------*/ calcCmonBorder(&i_cmon, &i_tran, v_bord, length_v_bord, tranPosInternal, bufferFrameStart, numberTimeSlots); /*-------------------------------------------------------------------------- Extract data for proper follow-up in next frame ---------------------------------------------------------------------------*/ keepForFollowUp(v_bordFollow, length_v_bordFollow, v_freqFollow, length_v_freqFollow, i_tranFollow, i_fillFollow, v_bord, length_v_bord, v_freq, i_cmon, i_tran, parts, numberTimeSlots); /*-------------------------------------------------------------------------- Calculate control signal ---------------------------------------------------------------------------*/ calcCtrlSignal(&hSbrEnvFrame->SbrGrid, frameClass, v_bord, *length_v_bord, v_freq, *length_v_freq, i_cmon, i_tran, 0, nL); } break; case FIXFIX: if (tranPos == 0) numEnv = 1; else numEnv = 2; hSbrEnvFrame->SbrGrid.bs_num_env = numEnv; hSbrEnvFrame->SbrGrid.frameClass = frameClass; break; default: FDK_ASSERT(0); } } /*------------------------------------------------------------------------- Convert control signal to frame info struct ---------------------------------------------------------------------------*/ ctrlSignal2FrameInfo(&hSbrEnvFrame->SbrGrid, &hSbrEnvFrame->SbrFrameInfo, hSbrEnvFrame->freq_res_fixfix); return &hSbrEnvFrame->SbrFrameInfo; } /***************************************************************************/ /*! \brief Gnerates frame info for FIXFIXonly frame class used for low delay version \return nothing ****************************************************************************/ static void generateFixFixOnly(HANDLE_SBR_FRAME_INFO hSbrFrameInfo, HANDLE_SBR_GRID hSbrGrid, int tranPosInternal, int numberTimeSlots, UCHAR fResTransIsLow) { int nEnv, i, k = 0, tranIdx; const int *pTable = NULL; const FREQ_RES *freqResTable = NULL; switch (numberTimeSlots) { case 8: { pTable = envelopeTable_8[tranPosInternal]; } freqResTable = freqRes_table_8; break; case 15: pTable = envelopeTable_15[tranPosInternal]; freqResTable = freqRes_table_16; break; case 16: pTable = envelopeTable_16[tranPosInternal]; freqResTable = freqRes_table_16; break; } /* look number of envolpes in table */ nEnv = pTable[0]; /* look up envolpe distribution in table */ for (i = 1; i < nEnv; i++) hSbrFrameInfo->borders[i] = pTable[i + 2]; /* open and close frame border */ hSbrFrameInfo->borders[0] = 0; hSbrFrameInfo->borders[nEnv] = numberTimeSlots; /* adjust segment-frequency-resolution according to the segment-length */ for (i = 0; i < nEnv; i++) { k = hSbrFrameInfo->borders[i + 1] - hSbrFrameInfo->borders[i]; if (!fResTransIsLow) hSbrFrameInfo->freqRes[i] = freqResTable[k]; else hSbrFrameInfo->freqRes[i] = FREQ_RES_LOW; hSbrGrid->v_f[i] = hSbrFrameInfo->freqRes[i]; } hSbrFrameInfo->nEnvelopes = nEnv; hSbrFrameInfo->shortEnv = pTable[2]; /* transient idx */ tranIdx = pTable[1]; /* add noise floors */ hSbrFrameInfo->bordersNoise[0] = 0; hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[tranIdx ? tranIdx : 1]; hSbrFrameInfo->bordersNoise[2] = numberTimeSlots; hSbrFrameInfo->nNoiseEnvelopes = 2; hSbrGrid->frameClass = FIXFIXonly; hSbrGrid->bs_abs_bord = tranPosInternal; hSbrGrid->bs_num_env = nEnv; } /******************************************************************************* Functionname: FDKsbrEnc_initFrameInfoGenerator ******************************************************************************* Description: Arguments: hSbrEnvFrame - pointer to sbr envelope handle allowSpread - commandline parameter numEnvStatic - commandline parameter staticFraming - commandline parameter Return: none *******************************************************************************/ void FDKsbrEnc_initFrameInfoGenerator(HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame, INT allowSpread, INT numEnvStatic, INT staticFraming, INT timeSlots, const FREQ_RES *freq_res_fixfix, UCHAR fResTransIsLow, INT ldGrid) { /* FH 00-06-26 */ FDKmemclear(hSbrEnvFrame, sizeof(SBR_ENVELOPE_FRAME)); /* Initialisation */ hSbrEnvFrame->frameClassOld = FIXFIX; hSbrEnvFrame->spreadFlag = 0; hSbrEnvFrame->allowSpread = allowSpread; hSbrEnvFrame->numEnvStatic = numEnvStatic; hSbrEnvFrame->staticFraming = staticFraming; hSbrEnvFrame->freq_res_fixfix[0] = freq_res_fixfix[0]; hSbrEnvFrame->freq_res_fixfix[1] = freq_res_fixfix[1]; hSbrEnvFrame->fResTransIsLow = fResTransIsLow; hSbrEnvFrame->length_v_bord = 0; hSbrEnvFrame->length_v_bordFollow = 0; hSbrEnvFrame->length_v_freq = 0; hSbrEnvFrame->length_v_freqFollow = 0; hSbrEnvFrame->i_tranFollow = 0; hSbrEnvFrame->i_fillFollow = 0; hSbrEnvFrame->SbrGrid.numberTimeSlots = timeSlots; if (ldGrid) { /*case CODEC_AACLD:*/ hSbrEnvFrame->dmin = 2; hSbrEnvFrame->dmax = 16; hSbrEnvFrame->frameMiddleSlot = FRAME_MIDDLE_SLOT_512LD; hSbrEnvFrame->SbrGrid.bufferFrameStart = 0; } else switch (timeSlots) { case NUMBER_TIME_SLOTS_1920: hSbrEnvFrame->dmin = 4; hSbrEnvFrame->dmax = 12; hSbrEnvFrame->SbrGrid.bufferFrameStart = 0; hSbrEnvFrame->frameMiddleSlot = FRAME_MIDDLE_SLOT_1920; break; case NUMBER_TIME_SLOTS_2048: hSbrEnvFrame->dmin = 4; hSbrEnvFrame->dmax = 12; hSbrEnvFrame->SbrGrid.bufferFrameStart = 0; hSbrEnvFrame->frameMiddleSlot = FRAME_MIDDLE_SLOT_2048; break; case NUMBER_TIME_SLOTS_1152: hSbrEnvFrame->dmin = 2; hSbrEnvFrame->dmax = 8; hSbrEnvFrame->SbrGrid.bufferFrameStart = 0; hSbrEnvFrame->frameMiddleSlot = FRAME_MIDDLE_SLOT_1152; break; case NUMBER_TIME_SLOTS_2304: hSbrEnvFrame->dmin = 4; hSbrEnvFrame->dmax = 15; hSbrEnvFrame->SbrGrid.bufferFrameStart = 0; hSbrEnvFrame->frameMiddleSlot = FRAME_MIDDLE_SLOT_2304; break; default: FDK_ASSERT(0); } } /******************************************************************************* Functionname: fillFrameTran ******************************************************************************* Description: Add mandatory borders, as described by the tuning vector and the current transient position Arguments: modified: v_bord - int pointer to v_bord vector length_v_bord - length of v_bord vector v_freq - int pointer to v_freq vector length_v_freq - length of v_freq vector bmin - int pointer to bmin (call by reference) bmax - int pointer to bmax (call by reference) not modified: tran - position of transient v_tuningSegm - int pointer to v_tuningSegm vector v_tuningFreq - int pointer to v_tuningFreq vector Return: none *******************************************************************************/ static void fillFrameTran( const int *v_tuningSegm, /*!< tuning: desired segment lengths */ const int *v_tuningFreq, /*!< tuning: desired frequency resolutions */ int tran, /*!< input : position of transient */ int *v_bord, /*!< memNew: borders */ int *length_v_bord, /*!< memNew: # borders */ int *v_freq, /*!< memNew: frequency resolutions */ int *length_v_freq, /*!< memNew: # frequency resolutions */ int *bmin, /*!< hlpNew: first mandatory border */ int *bmax /*!< hlpNew: last mandatory border */ ) { int bord, i; *length_v_bord = 0; *length_v_freq = 0; /* add attack env leading border (optional) */ if (v_tuningSegm[0]) { /* v_bord = [(Ba)] start of attack env */ FDKsbrEnc_AddRight(v_bord, length_v_bord, (tran - v_tuningSegm[0])); /* v_freq = [(Fa)] res of attack env */ FDKsbrEnc_AddRight(v_freq, length_v_freq, v_tuningFreq[0]); } /* add attack env trailing border/first decay env leading border */ bord = tran; FDKsbrEnc_AddRight(v_bord, length_v_bord, tran); /* v_bord = [(Ba),Bd1] */ /* add first decay env trailing border/2:nd decay env leading border */ if (v_tuningSegm[1]) { bord += v_tuningSegm[1]; /* v_bord = [(Ba),Bd1,Bd2] */ FDKsbrEnc_AddRight(v_bord, length_v_bord, bord); /* v_freq = [(Fa),Fd1] */ FDKsbrEnc_AddRight(v_freq, length_v_freq, v_tuningFreq[1]); } /* add 2:nd decay env trailing border (optional) */ if (v_tuningSegm[2] != 0) { bord += v_tuningSegm[2]; /* v_bord = [(Ba),Bd1, Bd2,(Bd3)] */ FDKsbrEnc_AddRight(v_bord, length_v_bord, bord); /* v_freq = [(Fa),Fd1,(Fd2)] */ FDKsbrEnc_AddRight(v_freq, length_v_freq, v_tuningFreq[2]); } /* v_freq = [(Fa),Fd1,(Fd2),1] */ FDKsbrEnc_AddRight(v_freq, length_v_freq, 1); /* calc min and max values of mandatory borders */ *bmin = v_bord[0]; for (i = 0; i < *length_v_bord; i++) if (v_bord[i] < *bmin) *bmin = v_bord[i]; *bmax = v_bord[0]; for (i = 0; i < *length_v_bord; i++) if (v_bord[i] > *bmax) *bmax = v_bord[i]; } /******************************************************************************* Functionname: fillFramePre ******************************************************************************* Description: Add borders before mandatory borders, if needed Arguments: modified: v_bord - int pointer to v_bord vector length_v_bord - length of v_bord vector v_freq - int pointer to v_freq vector length_v_freq - length of v_freq vector not modified: dmax - int value bmin - int value rest - int value Return: none *******************************************************************************/ static void fillFramePre(INT dmax, INT *v_bord, INT *length_v_bord, INT *v_freq, INT *length_v_freq, INT bmin, INT rest) { /* input state: v_bord = [(Ba),Bd1, Bd2 ,(Bd3)] v_freq = [(Fa),Fd1,(Fd2),1 ] */ INT parts, d, j, S, s = 0, segm, bord; /* start with one envelope */ parts = 1; d = rest; /* calc # of additional envelopes and corresponding lengths */ while (d > dmax) { parts++; segm = rest / parts; S = (segm - 2) >> 1; s = fixMin(8, 2 * S + 2); d = rest - (parts - 1) * s; } /* add borders before mandatory borders */ bord = bmin; for (j = 0; j <= parts - 2; j++) { bord = bord - s; /* v_bord = [...,(Bf),(Ba),Bd1, Bd2 ,(Bd3)] */ FDKsbrEnc_AddLeft(v_bord, length_v_bord, bord); /* v_freq = [...,(1 ),(Fa),Fd1,(Fd2), 1 ] */ FDKsbrEnc_AddLeft(v_freq, length_v_freq, 1); } } /***************************************************************************/ /*! \brief Overlap control Calculate max length of trailing fill segments, such that we always get a border within the frame overlap region \return void ****************************************************************************/ static int calcFillLengthMax( int tranPos, /*!< input : transient position (ref: tran det) */ int numberTimeSlots /*!< input : number of timeslots */ ) { int fmax; /* calculate transient position within envelope buffer */ switch (numberTimeSlots) { case NUMBER_TIME_SLOTS_2048: if (tranPos < 4) fmax = 6; else if (tranPos == 4 || tranPos == 5) fmax = 4; else fmax = 8; break; case NUMBER_TIME_SLOTS_1920: if (tranPos < 4) fmax = 5; else if (tranPos == 4 || tranPos == 5) fmax = 3; else fmax = 7; break; default: fmax = 8; break; } return fmax; } /******************************************************************************* Functionname: fillFramePost ******************************************************************************* Description: -Add borders after mandatory borders, if needed Make a preliminary design of next frame, assuming no transient is present there Arguments: modified: parts - int pointer to parts (call by reference) d - int pointer to d (call by reference) v_bord - int pointer to v_bord vector length_v_bord - length of v_bord vector v_freq - int pointer to v_freq vector length_v_freq - length of v_freq vector not modified: bmax - int value dmax - int value Return: none *******************************************************************************/ static void fillFramePost(INT *parts, INT *d, INT dmax, INT *v_bord, INT *length_v_bord, INT *v_freq, INT *length_v_freq, INT bmax, INT bufferFrameStart, INT numberTimeSlots, INT fmax) { INT j, rest, segm, S, s = 0, bord; /* input state: v_bord = [...,(Bf),(Ba),Bd1, Bd2 ,(Bd3)] v_freq = [...,(1 ),(Fa),Fd1,(Fd2),1 ] */ rest = bufferFrameStart + 2 * numberTimeSlots - bmax; *d = rest; if (*d > 0) { *parts = 1; /* start with one envelope */ /* calc # of additional envelopes and corresponding lengths */ while (*d > dmax) { *parts = *parts + 1; segm = rest / (*parts); S = (segm - 2) >> 1; s = fixMin(fmax, 2 * S + 2); *d = rest - (*parts - 1) * s; } /* add borders after mandatory borders */ bord = bmax; for (j = 0; j <= *parts - 2; j++) { bord += s; /* v_bord = [...,(Bf),(Ba),Bd1, Bd2 ,(Bd3),(Bf)] */ FDKsbrEnc_AddRight(v_bord, length_v_bord, bord); /* v_freq = [...,(1 ),(Fa),Fd1,(Fd2), 1 , 1! ,1] */ FDKsbrEnc_AddRight(v_freq, length_v_freq, 1); } } else { *parts = 1; /* remove last element from v_bord and v_freq */ *length_v_bord = *length_v_bord - 1; *length_v_freq = *length_v_freq - 1; } } /******************************************************************************* Functionname: fillFrameInter ******************************************************************************* Description: Arguments: nL - v_tuningSegm - v_bord - length_v_bord - bmin - v_freq - length_v_freq - v_bordFollow - length_v_bordFollow - v_freqFollow - length_v_freqFollow - i_fillFollow - dmin - dmax - Return: none *******************************************************************************/ static void fillFrameInter(INT *nL, const int *v_tuningSegm, INT *v_bord, INT *length_v_bord, INT bmin, INT *v_freq, INT *length_v_freq, INT *v_bordFollow, INT *length_v_bordFollow, INT *v_freqFollow, INT *length_v_freqFollow, INT i_fillFollow, INT dmin, INT dmax, INT numberTimeSlots) { INT middle, b_new, numBordFollow, bordMaxFollow, i; if (numberTimeSlots != NUMBER_TIME_SLOTS_1152) { /* % remove fill borders: */ if (i_fillFollow >= 1) { *length_v_bordFollow = i_fillFollow; *length_v_freqFollow = i_fillFollow; } numBordFollow = *length_v_bordFollow; bordMaxFollow = v_bordFollow[numBordFollow - 1]; /* remove even more borders if needed */ middle = bmin - bordMaxFollow; while (middle < 0) { numBordFollow--; bordMaxFollow = v_bordFollow[numBordFollow - 1]; middle = bmin - bordMaxFollow; } *length_v_bordFollow = numBordFollow; *length_v_freqFollow = numBordFollow; *nL = numBordFollow - 1; b_new = *length_v_bord; if (middle <= dmax) { if (middle >= dmin) { /* concatenate */ FDKsbrEnc_AddVecLeft(v_bord, length_v_bord, v_bordFollow, *length_v_bordFollow); FDKsbrEnc_AddVecLeft(v_freq, length_v_freq, v_freqFollow, *length_v_freqFollow); } else { if (v_tuningSegm[0] != 0) { /* remove one new border and concatenate */ *length_v_bord = b_new - 1; FDKsbrEnc_AddVecLeft(v_bord, length_v_bord, v_bordFollow, *length_v_bordFollow); *length_v_freq = b_new - 1; FDKsbrEnc_AddVecLeft(v_freq + 1, length_v_freq, v_freqFollow, *length_v_freqFollow); } else { if (*length_v_bordFollow > 1) { /* remove one old border and concatenate */ FDKsbrEnc_AddVecLeft(v_bord, length_v_bord, v_bordFollow, *length_v_bordFollow - 1); FDKsbrEnc_AddVecLeft(v_freq, length_v_freq, v_freqFollow, *length_v_bordFollow - 1); *nL = *nL - 1; } else { /* remove new "transient" border and concatenate */ for (i = 0; i < *length_v_bord - 1; i++) v_bord[i] = v_bord[i + 1]; for (i = 0; i < *length_v_freq - 1; i++) v_freq[i] = v_freq[i + 1]; *length_v_bord = b_new - 1; *length_v_freq = b_new - 1; FDKsbrEnc_AddVecLeft(v_bord, length_v_bord, v_bordFollow, *length_v_bordFollow); FDKsbrEnc_AddVecLeft(v_freq, length_v_freq, v_freqFollow, *length_v_freqFollow); } } } } else { /* middle > dmax */ fillFramePre(dmax, v_bord, length_v_bord, v_freq, length_v_freq, bmin, middle); FDKsbrEnc_AddVecLeft(v_bord, length_v_bord, v_bordFollow, *length_v_bordFollow); FDKsbrEnc_AddVecLeft(v_freq, length_v_freq, v_freqFollow, *length_v_freqFollow); } } else { /* numberTimeSlots==NUMBER_TIME_SLOTS_1152 */ INT l, m; /*------------------------------------------------------------------------ remove fill borders ------------------------------------------------------------------------*/ if (i_fillFollow >= 1) { *length_v_bordFollow = i_fillFollow; *length_v_freqFollow = i_fillFollow; } numBordFollow = *length_v_bordFollow; bordMaxFollow = v_bordFollow[numBordFollow - 1]; /*------------------------------------------------------------------------ remove more borders if necessary to eliminate overlap ------------------------------------------------------------------------*/ /* check for overlap */ middle = bmin - bordMaxFollow; /* intervals: i) middle < 0 : overlap, must remove borders ii) 0 <= middle < dmin : no overlap but too tight, must remove borders iii) dmin <= middle <= dmax : ok, just concatenate iv) dmax <= middle : too wide, must add borders */ /* first remove old non-fill-borders... */ while (middle < 0) { /* ...but don't remove all of them */ if (numBordFollow == 1) break; numBordFollow--; bordMaxFollow = v_bordFollow[numBordFollow - 1]; middle = bmin - bordMaxFollow; } /* if this isn't enough, remove new non-fill borders */ if (middle < 0) { for (l = 0, m = 0; l < *length_v_bord; l++) { if (v_bord[l] > bordMaxFollow) { v_bord[m] = v_bord[l]; v_freq[m] = v_freq[l]; m++; } } *length_v_bord = l; *length_v_freq = l; bmin = v_bord[0]; } /*------------------------------------------------------------------------ update modified follow-up data ------------------------------------------------------------------------*/ *length_v_bordFollow = numBordFollow; *length_v_freqFollow = numBordFollow; /* left relative borders correspond to follow-up */ *nL = numBordFollow - 1; /*------------------------------------------------------------------------ take care of intervals ii through iv ------------------------------------------------------------------------*/ /* now middle should be >= 0 */ middle = bmin - bordMaxFollow; if (middle <= dmin) /* (ii) */ { b_new = *length_v_bord; if (v_tuningSegm[0] != 0) { /* remove new "luxury" border and concatenate */ *length_v_bord = b_new - 1; FDKsbrEnc_AddVecLeft(v_bord, length_v_bord, v_bordFollow, *length_v_bordFollow); *length_v_freq = b_new - 1; FDKsbrEnc_AddVecLeft(v_freq + 1, length_v_freq, v_freqFollow, *length_v_freqFollow); } else if (*length_v_bordFollow > 1) { /* remove old border and concatenate */ FDKsbrEnc_AddVecLeft(v_bord, length_v_bord, v_bordFollow, *length_v_bordFollow - 1); FDKsbrEnc_AddVecLeft(v_freq, length_v_freq, v_freqFollow, *length_v_bordFollow - 1); *nL = *nL - 1; } else { /* remove new border and concatenate */ for (i = 0; i < *length_v_bord - 1; i++) v_bord[i] = v_bord[i + 1]; for (i = 0; i < *length_v_freq - 1; i++) v_freq[i] = v_freq[i + 1]; *length_v_bord = b_new - 1; *length_v_freq = b_new - 1; FDKsbrEnc_AddVecLeft(v_bord, length_v_bord, v_bordFollow, *length_v_bordFollow); FDKsbrEnc_AddVecLeft(v_freq, length_v_freq, v_freqFollow, *length_v_freqFollow); } } else if ((middle >= dmin) && (middle <= dmax)) /* (iii) */ { /* concatenate */ FDKsbrEnc_AddVecLeft(v_bord, length_v_bord, v_bordFollow, *length_v_bordFollow); FDKsbrEnc_AddVecLeft(v_freq, length_v_freq, v_freqFollow, *length_v_freqFollow); } else /* (iv) */ { fillFramePre(dmax, v_bord, length_v_bord, v_freq, length_v_freq, bmin, middle); FDKsbrEnc_AddVecLeft(v_bord, length_v_bord, v_bordFollow, *length_v_bordFollow); FDKsbrEnc_AddVecLeft(v_freq, length_v_freq, v_freqFollow, *length_v_freqFollow); } } } /******************************************************************************* Functionname: calcFrameClass ******************************************************************************* Description: Arguments: INT* frameClass, INT* frameClassOld, INT tranFlag, INT* spreadFlag) Return: none *******************************************************************************/ static void calcFrameClass(FRAME_CLASS *frameClass, FRAME_CLASS *frameClassOld, INT tranFlag, INT *spreadFlag) { switch (*frameClassOld) { case FIXFIXonly: case FIXFIX: if (tranFlag) *frameClass = FIXVAR; else *frameClass = FIXFIX; break; case FIXVAR: if (tranFlag) { *frameClass = VARVAR; *spreadFlag = 0; } else { if (*spreadFlag) *frameClass = VARVAR; else *frameClass = VARFIX; } break; case VARFIX: if (tranFlag) *frameClass = FIXVAR; else *frameClass = FIXFIX; break; case VARVAR: if (tranFlag) { *frameClass = VARVAR; *spreadFlag = 0; } else { if (*spreadFlag) *frameClass = VARVAR; else *frameClass = VARFIX; } break; }; *frameClassOld = *frameClass; } /******************************************************************************* Functionname: specialCase ******************************************************************************* Description: Arguments: spreadFlag allowSpread v_bord length_v_bord v_freq length_v_freq parts d Return: none *******************************************************************************/ static void specialCase(INT *spreadFlag, INT allowSpread, INT *v_bord, INT *length_v_bord, INT *v_freq, INT *length_v_freq, INT *parts, INT d) { INT L; L = *length_v_bord; if (allowSpread) { /* add one "step 8" */ *spreadFlag = 1; FDKsbrEnc_AddRight(v_bord, length_v_bord, v_bord[L - 1] + 8); FDKsbrEnc_AddRight(v_freq, length_v_freq, 1); (*parts)++; } else { if (d == 1) { /* stretch one slot */ *length_v_bord = L - 1; *length_v_freq = L - 1; } else { if ((v_bord[L - 1] - v_bord[L - 2]) > 2) { /* compress one quant step */ v_bord[L - 1] = v_bord[L - 1] - 2; v_freq[*length_v_freq - 1] = 0; /* use low res for short segment */ } } } } /******************************************************************************* Functionname: calcCmonBorder ******************************************************************************* Description: Arguments: i_cmon i_tran v_bord length_v_bord tran Return: none *******************************************************************************/ static void calcCmonBorder(INT *i_cmon, INT *i_tran, INT *v_bord, INT *length_v_bord, INT tran, INT bufferFrameStart, INT numberTimeSlots) { /* FH 00-06-26 */ INT i; for (i = 0; i < *length_v_bord; i++) if (v_bord[i] >= bufferFrameStart + numberTimeSlots) { /* FH 00-06-26 */ *i_cmon = i; break; } /* keep track of transient: */ for (i = 0; i < *length_v_bord; i++) if (v_bord[i] >= tran) { *i_tran = i; break; } else *i_tran = EMPTY; } /******************************************************************************* Functionname: keepForFollowUp ******************************************************************************* Description: Arguments: v_bordFollow length_v_bordFollow v_freqFollow length_v_freqFollow i_tranFollow i_fillFollow v_bord length_v_bord v_freq i_cmon i_tran parts) Return: none *******************************************************************************/ static void keepForFollowUp(INT *v_bordFollow, INT *length_v_bordFollow, INT *v_freqFollow, INT *length_v_freqFollow, INT *i_tranFollow, INT *i_fillFollow, INT *v_bord, INT *length_v_bord, INT *v_freq, INT i_cmon, INT i_tran, INT parts, INT numberTimeSlots) { /* FH 00-06-26 */ INT L, i, j; L = *length_v_bord; (*length_v_bordFollow) = 0; (*length_v_freqFollow) = 0; for (j = 0, i = i_cmon; i < L; i++, j++) { v_bordFollow[j] = v_bord[i] - numberTimeSlots; /* FH 00-06-26 */ v_freqFollow[j] = v_freq[i]; (*length_v_bordFollow)++; (*length_v_freqFollow)++; } if (i_tran != EMPTY) *i_tranFollow = i_tran - i_cmon; else *i_tranFollow = EMPTY; *i_fillFollow = L - (parts - 1) - i_cmon; } /******************************************************************************* Functionname: calcCtrlSignal ******************************************************************************* Description: Arguments: hSbrGrid frameClass v_bord length_v_bord v_freq length_v_freq i_cmon i_tran spreadFlag nL Return: none *******************************************************************************/ static void calcCtrlSignal(HANDLE_SBR_GRID hSbrGrid, FRAME_CLASS frameClass, INT *v_bord, INT length_v_bord, INT *v_freq, INT length_v_freq, INT i_cmon, INT i_tran, INT spreadFlag, INT nL) { INT i, r, a, n, p, b, aL, aR, ntot, nmax, nR; INT *v_f = hSbrGrid->v_f; INT *v_fLR = hSbrGrid->v_fLR; INT *v_r = hSbrGrid->bs_rel_bord; INT *v_rL = hSbrGrid->bs_rel_bord_0; INT *v_rR = hSbrGrid->bs_rel_bord_1; INT length_v_r = 0; INT length_v_rR = 0; INT length_v_rL = 0; switch (frameClass) { case FIXVAR: /* absolute border: */ a = v_bord[i_cmon]; /* relative borders: */ length_v_r = 0; i = i_cmon; while (i >= 1) { r = v_bord[i] - v_bord[i - 1]; FDKsbrEnc_AddRight(v_r, &length_v_r, r); i--; } /* number of relative borders: */ n = length_v_r; /* freq res: */ for (i = 0; i < i_cmon; i++) v_f[i] = v_freq[i_cmon - 1 - i]; v_f[i_cmon] = 1; /* pointer: */ p = (i_cmon >= i_tran && i_tran != EMPTY) ? (i_cmon - i_tran + 1) : (0); hSbrGrid->frameClass = frameClass; hSbrGrid->bs_abs_bord = a; hSbrGrid->n = n; hSbrGrid->p = p; break; case VARFIX: /* absolute border: */ a = v_bord[0]; /* relative borders: */ length_v_r = 0; for (i = 1; i < length_v_bord; i++) { r = v_bord[i] - v_bord[i - 1]; FDKsbrEnc_AddRight(v_r, &length_v_r, r); } /* number of relative borders: */ n = length_v_r; /* freq res: */ FDKmemcpy(v_f, v_freq, length_v_freq * sizeof(INT)); /* pointer: */ p = (i_tran >= 0 && i_tran != EMPTY) ? (i_tran + 1) : (0); hSbrGrid->frameClass = frameClass; hSbrGrid->bs_abs_bord = a; hSbrGrid->n = n; hSbrGrid->p = p; break; case VARVAR: if (spreadFlag) { /* absolute borders: */ b = length_v_bord; aL = v_bord[0]; aR = v_bord[b - 1]; /* number of relative borders: */ ntot = b - 2; nmax = 2; /* n: {0,1,2} */ if (ntot > nmax) { nL = nmax; nR = ntot - nmax; } else { nL = ntot; nR = 0; } /* relative borders: */ length_v_rL = 0; for (i = 1; i <= nL; i++) { r = v_bord[i] - v_bord[i - 1]; FDKsbrEnc_AddRight(v_rL, &length_v_rL, r); } length_v_rR = 0; i = b - 1; while (i >= b - nR) { r = v_bord[i] - v_bord[i - 1]; FDKsbrEnc_AddRight(v_rR, &length_v_rR, r); i--; } /* pointer (only one due to constraint in frame info): */ p = (i_tran > 0 && i_tran != EMPTY) ? (b - i_tran) : (0); /* freq res: */ for (i = 0; i < b - 1; i++) v_fLR[i] = v_freq[i]; } else { length_v_bord = i_cmon + 1; /* absolute borders: */ b = length_v_bord; aL = v_bord[0]; aR = v_bord[b - 1]; /* number of relative borders: */ ntot = b - 2; nR = ntot - nL; /* relative borders: */ length_v_rL = 0; for (i = 1; i <= nL; i++) { r = v_bord[i] - v_bord[i - 1]; FDKsbrEnc_AddRight(v_rL, &length_v_rL, r); } length_v_rR = 0; i = b - 1; while (i >= b - nR) { r = v_bord[i] - v_bord[i - 1]; FDKsbrEnc_AddRight(v_rR, &length_v_rR, r); i--; } /* pointer (only one due to constraint in frame info): */ p = (i_cmon >= i_tran && i_tran != EMPTY) ? (i_cmon - i_tran + 1) : (0); /* freq res: */ for (i = 0; i < b - 1; i++) v_fLR[i] = v_freq[i]; } hSbrGrid->frameClass = frameClass; hSbrGrid->bs_abs_bord_0 = aL; hSbrGrid->bs_abs_bord_1 = aR; hSbrGrid->bs_num_rel_0 = nL; hSbrGrid->bs_num_rel_1 = nR; hSbrGrid->p = p; break; default: /* do nothing */ break; } } /******************************************************************************* Functionname: createDefFrameInfo ******************************************************************************* Description: Copies the default (static) frameInfo structs to the frameInfo passed by reference; only used for FIXFIX frames Arguments: hFrameInfo - HANLDE_SBR_FRAME_INFO nEnv - INT nTimeSlots - INT Return: none; hSbrFrameInfo contains a copy of the default frameInfo Written: Andreas Schneider Revised: *******************************************************************************/ static void createDefFrameInfo(HANDLE_SBR_FRAME_INFO hSbrFrameInfo, INT nEnv, INT nTimeSlots) { switch (nEnv) { case 1: switch (nTimeSlots) { case NUMBER_TIME_SLOTS_1920: FDKmemcpy(hSbrFrameInfo, &frameInfo1_1920, sizeof(SBR_FRAME_INFO)); break; case NUMBER_TIME_SLOTS_2048: FDKmemcpy(hSbrFrameInfo, &frameInfo1_2048, sizeof(SBR_FRAME_INFO)); break; case NUMBER_TIME_SLOTS_1152: FDKmemcpy(hSbrFrameInfo, &frameInfo1_1152, sizeof(SBR_FRAME_INFO)); break; case NUMBER_TIME_SLOTS_2304: FDKmemcpy(hSbrFrameInfo, &frameInfo1_2304, sizeof(SBR_FRAME_INFO)); break; case NUMBER_TIME_SLOTS_512LD: FDKmemcpy(hSbrFrameInfo, &frameInfo1_512LD, sizeof(SBR_FRAME_INFO)); break; default: FDK_ASSERT(0); } break; case 2: switch (nTimeSlots) { case NUMBER_TIME_SLOTS_1920: FDKmemcpy(hSbrFrameInfo, &frameInfo2_1920, sizeof(SBR_FRAME_INFO)); break; case NUMBER_TIME_SLOTS_2048: FDKmemcpy(hSbrFrameInfo, &frameInfo2_2048, sizeof(SBR_FRAME_INFO)); break; case NUMBER_TIME_SLOTS_1152: FDKmemcpy(hSbrFrameInfo, &frameInfo2_1152, sizeof(SBR_FRAME_INFO)); break; case NUMBER_TIME_SLOTS_2304: FDKmemcpy(hSbrFrameInfo, &frameInfo2_2304, sizeof(SBR_FRAME_INFO)); break; case NUMBER_TIME_SLOTS_512LD: FDKmemcpy(hSbrFrameInfo, &frameInfo2_512LD, sizeof(SBR_FRAME_INFO)); break; default: FDK_ASSERT(0); } break; case 4: switch (nTimeSlots) { case NUMBER_TIME_SLOTS_1920: FDKmemcpy(hSbrFrameInfo, &frameInfo4_1920, sizeof(SBR_FRAME_INFO)); break; case NUMBER_TIME_SLOTS_2048: FDKmemcpy(hSbrFrameInfo, &frameInfo4_2048, sizeof(SBR_FRAME_INFO)); break; case NUMBER_TIME_SLOTS_1152: FDKmemcpy(hSbrFrameInfo, &frameInfo4_1152, sizeof(SBR_FRAME_INFO)); break; case NUMBER_TIME_SLOTS_2304: FDKmemcpy(hSbrFrameInfo, &frameInfo4_2304, sizeof(SBR_FRAME_INFO)); break; case NUMBER_TIME_SLOTS_512LD: FDKmemcpy(hSbrFrameInfo, &frameInfo4_512LD, sizeof(SBR_FRAME_INFO)); break; default: FDK_ASSERT(0); } break; default: FDK_ASSERT(0); } } /******************************************************************************* Functionname: ctrlSignal2FrameInfo ******************************************************************************* Description: Convert "clear-text" sbr_grid() to "frame info" used by the envelope and noise floor estimators. This is basically (except for "low level" calculations) the bitstream decoder defined in the MPEG-4 standard, sub clause 4.6.18.3.3, Time / Frequency Grid. See inline comments for explanation of the shorten and noise border algorithms. Arguments: hSbrGrid - source hSbrFrameInfo - destination freq_res_fixfix - frequency resolution for FIXFIX frames Return: void; hSbrFrameInfo contains the updated FRAME_INFO struct *******************************************************************************/ static void ctrlSignal2FrameInfo( HANDLE_SBR_GRID hSbrGrid, /* input : the grid handle */ HANDLE_SBR_FRAME_INFO hSbrFrameInfo, /* output: the frame info handle */ FREQ_RES *freq_res_fixfix /* in/out: frequency resolution for FIXFIX frames */ ) { INT frameSplit = 0; INT nEnv = 0, border = 0, i, k, p /*?*/; INT *v_r = hSbrGrid->bs_rel_bord; INT *v_f = hSbrGrid->v_f; FRAME_CLASS frameClass = hSbrGrid->frameClass; INT bufferFrameStart = hSbrGrid->bufferFrameStart; INT numberTimeSlots = hSbrGrid->numberTimeSlots; switch (frameClass) { case FIXFIX: createDefFrameInfo(hSbrFrameInfo, hSbrGrid->bs_num_env, numberTimeSlots); frameSplit = (hSbrFrameInfo->nEnvelopes > 1); for (i = 0; i < hSbrFrameInfo->nEnvelopes; i++) { hSbrGrid->v_f[i] = hSbrFrameInfo->freqRes[i] = freq_res_fixfix[frameSplit]; } break; case FIXVAR: case VARFIX: nEnv = hSbrGrid->n + 1; /* read n [SBR_NUM_BITS bits] */ /*? snd*/ FDK_ASSERT(nEnv <= MAX_ENVELOPES_FIXVAR_VARFIX); hSbrFrameInfo->nEnvelopes = nEnv; border = hSbrGrid->bs_abs_bord; /* read the absolute border */ if (nEnv == 1) hSbrFrameInfo->nNoiseEnvelopes = 1; else hSbrFrameInfo->nNoiseEnvelopes = 2; break; default: /* do nothing */ break; } switch (frameClass) { case FIXVAR: hSbrFrameInfo->borders[0] = bufferFrameStart; /* start-position of 1st envelope */ hSbrFrameInfo->borders[nEnv] = border; for (k = 0, i = nEnv - 1; k < nEnv - 1; k++, i--) { border -= v_r[k]; hSbrFrameInfo->borders[i] = border; } /* make either envelope nr. nEnv + 1 - p short; or don't shorten if p == 0 */ p = hSbrGrid->p; if (p == 0) { hSbrFrameInfo->shortEnv = 0; } else { hSbrFrameInfo->shortEnv = nEnv + 1 - p; } for (k = 0, i = nEnv - 1; k < nEnv; k++, i--) { hSbrFrameInfo->freqRes[i] = (FREQ_RES)v_f[k]; } /* if either there is no short envelope or the last envelope is short... */ if (p == 0 || p == 1) { hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[nEnv - 1]; } else { hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[hSbrFrameInfo->shortEnv]; } break; case VARFIX: /* in this case 'border' indicates the start of the 1st envelope */ hSbrFrameInfo->borders[0] = border; for (k = 0; k < nEnv - 1; k++) { border += v_r[k]; hSbrFrameInfo->borders[k + 1] = border; } hSbrFrameInfo->borders[nEnv] = bufferFrameStart + numberTimeSlots; p = hSbrGrid->p; if (p == 0 || p == 1) { hSbrFrameInfo->shortEnv = 0; } else { hSbrFrameInfo->shortEnv = p - 1; } for (k = 0; k < nEnv; k++) { hSbrFrameInfo->freqRes[k] = (FREQ_RES)v_f[k]; } switch (p) { case 0: hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[1]; break; case 1: hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[nEnv - 1]; break; default: hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[hSbrFrameInfo->shortEnv]; break; } break; case VARVAR: nEnv = hSbrGrid->bs_num_rel_0 + hSbrGrid->bs_num_rel_1 + 1; FDK_ASSERT(nEnv <= MAX_ENVELOPES_VARVAR); /* just to be sure */ hSbrFrameInfo->nEnvelopes = nEnv; hSbrFrameInfo->borders[0] = border = hSbrGrid->bs_abs_bord_0; for (k = 0, i = 1; k < hSbrGrid->bs_num_rel_0; k++, i++) { border += hSbrGrid->bs_rel_bord_0[k]; hSbrFrameInfo->borders[i] = border; } border = hSbrGrid->bs_abs_bord_1; hSbrFrameInfo->borders[nEnv] = border; for (k = 0, i = nEnv - 1; k < hSbrGrid->bs_num_rel_1; k++, i--) { border -= hSbrGrid->bs_rel_bord_1[k]; hSbrFrameInfo->borders[i] = border; } p = hSbrGrid->p; if (p == 0) { hSbrFrameInfo->shortEnv = 0; } else { hSbrFrameInfo->shortEnv = nEnv + 1 - p; } for (k = 0; k < nEnv; k++) { hSbrFrameInfo->freqRes[k] = (FREQ_RES)hSbrGrid->v_fLR[k]; } if (nEnv == 1) { hSbrFrameInfo->nNoiseEnvelopes = 1; hSbrFrameInfo->bordersNoise[0] = hSbrGrid->bs_abs_bord_0; hSbrFrameInfo->bordersNoise[1] = hSbrGrid->bs_abs_bord_1; } else { hSbrFrameInfo->nNoiseEnvelopes = 2; hSbrFrameInfo->bordersNoise[0] = hSbrGrid->bs_abs_bord_0; if (p == 0 || p == 1) { hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[nEnv - 1]; } else { hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[hSbrFrameInfo->shortEnv]; } hSbrFrameInfo->bordersNoise[2] = hSbrGrid->bs_abs_bord_1; } break; default: /* do nothing */ break; } if (frameClass == VARFIX || frameClass == FIXVAR) { hSbrFrameInfo->bordersNoise[0] = hSbrFrameInfo->borders[0]; if (nEnv == 1) { hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[nEnv]; } else { hSbrFrameInfo->bordersNoise[2] = hSbrFrameInfo->borders[nEnv]; } } } fdk-aac-2.0.2/libSBRenc/src/fram_gen.h000066400000000000000000000360751404225310700173330ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Framing generator prototypes and structs $Revision: 92790 $ */ #ifndef FRAM_GEN_H #define FRAM_GEN_H #include "sbr_def.h" /* for MAX_ENVELOPES and MAX_NOISE_ENVELOPES in struct FRAME_INFO and CODEC_TYPE */ #include "sbr_encoder.h" /* for FREQ_RES */ #define MAX_ENVELOPES_VARVAR \ MAX_ENVELOPES /*!< worst case number of envelopes in a VARVAR frame */ #define MAX_ENVELOPES_FIXVAR_VARFIX \ 4 /*!< worst case number of envelopes in VARFIX and FIXVAR frames */ #define MAX_NUM_REL \ 3 /*!< maximum number of relative borders in any VAR frame */ /* SBR frame class definitions */ typedef enum { FIXFIX = 0, /*!< bs_frame_class: leading and trailing frame borders are fixed */ FIXVAR, /*!< bs_frame_class: leading frame border is fixed, trailing frame border is variable */ VARFIX, /*!< bs_frame_class: leading frame border is variable, trailing frame border is fixed */ VARVAR /*!< bs_frame_class: leading and trailing frame borders are variable */ , FIXFIXonly /*!< bs_frame_class: leading border fixed (0), trailing border fixed (nrTimeSlots) and encased borders are dynamically derived from the tranPos */ } FRAME_CLASS; /* helper constants */ #define DC 4711 /*!< helper constant: don't care */ #define EMPTY (-99) /*!< helper constant: empty */ /* system constants: AAC+SBR, DRM Frame-Length */ #define FRAME_MIDDLE_SLOT_1920 4 #define NUMBER_TIME_SLOTS_1920 15 #define LD_PRETRAN_OFF 3 #define FRAME_MIDDLE_SLOT_512LD 4 #define NUMBER_TIME_SLOTS_512LD 8 #define TRANSIENT_OFFSET_LD 0 /* system constants: AAC+SBR or aacPRO (hybrid format), Standard Frame-Length, Multi-Rate --------------------------------------------------------------------------- Number of slots (numberTimeSlots): 16 (NUMBER_TIME_SLOTS_2048) Detector-offset (frameMiddleSlot): 4 Overlap : 3 Buffer-offset : 8 (BUFFER_FRAME_START_2048 = 0) |<------------tranPos---------->| |c|d|e|f|0|1|2|3|4|5|6|7|8|9|a|b|c|d|e|f| FixFix | | FixVar | :<- ->: VarFix :<- ->: | VarVar :<- ->: :<- ->: 0 1 2 3 4 5 6 7 8 9 a b c d e f 0 1 2 3 ................................................................................ |-|-|-|-|-|-|-|-B-|-|-|-|-|-|-|-|-|-|-|-|-|-|-|-B-|-|-|-|-|-|-|-|-|-|-|-|-|-|-|-| frame-generator:0 16 24 32 analysis-buffer:8 24 32 40 */ #define FRAME_MIDDLE_SLOT_2048 4 #define NUMBER_TIME_SLOTS_2048 16 /* system constants: mp3PRO, Multi-Rate & Single-Rate -------------------------------------------------- Number of slots (numberTimeSlots): 9 (NUMBER_TIME_SLOTS_1152) Detector-offset (frameMiddleSlot): 4 (FRAME_MIDDLE_SLOT_1152) Overlap : 3 Buffer-offset : 4.5 (BUFFER_FRAME_START_1152 = 0) |<----tranPos---->| |5|6|7|8|0|1|2|3|4|5|6|7|8| FixFix | | FixVar | :<- ->: VarFix :<- ->: | VarVar :<- ->: :<- ->: 0 1 2 3 4 5 6 7 8 0 1 2 3 ............................................. -|-|-|-|-B-|-|-|-|-|-|-|-|-B-|-|-|-|-|-|-|-|-| frame-generator: 0 9 13 18 analysis-buffer: 4.5 13.5 22.5 */ #define FRAME_MIDDLE_SLOT_1152 4 #define NUMBER_TIME_SLOTS_1152 9 /* system constants: Layer2+SBR */ #define FRAME_MIDDLE_SLOT_2304 8 #define NUMBER_TIME_SLOTS_2304 18 /*! \struct SBR_GRID \brief sbr_grid() signals to be converted to bitstream elements The variables hold the signals (e.g. lengths and numbers) in "clear text" */ typedef struct { /* system constants */ INT bufferFrameStart; /*!< frame generator vs analysis buffer time alignment (currently set to 0, offset added elsewhere) */ INT numberTimeSlots; /*!< number of SBR timeslots per frame */ /* will be adjusted for every frame */ FRAME_CLASS frameClass; /*!< SBR frame class */ INT bs_num_env; /*!< bs_num_env, number of envelopes for FIXFIX */ INT bs_abs_bord; /*!< bs_abs_bord, absolute border for VARFIX and FIXVAR */ INT n; /*!< number of relative borders for VARFIX and FIXVAR */ INT p; /*!< pointer-to-transient-border */ INT bs_rel_bord[MAX_NUM_REL]; /*!< bs_rel_bord, relative borders for all VAR */ INT v_f[MAX_ENVELOPES_FIXVAR_VARFIX]; /*!< envelope frequency resolutions for FIXVAR and VARFIX */ INT bs_abs_bord_0; /*!< bs_abs_bord_0, leading absolute border for VARVAR */ INT bs_abs_bord_1; /*!< bs_abs_bord_1, trailing absolute border for VARVAR */ INT bs_num_rel_0; /*!< bs_num_rel_0, number of relative borders associated with leading absolute border for VARVAR */ INT bs_num_rel_1; /*!< bs_num_rel_1, number of relative borders associated with trailing absolute border for VARVAR */ INT bs_rel_bord_0[MAX_NUM_REL]; /*!< bs_rel_bord_0, relative borders associated with leading absolute border for VARVAR */ INT bs_rel_bord_1[MAX_NUM_REL]; /*!< bs_rel_bord_1, relative borders associated with trailing absolute border for VARVAR */ INT v_fLR[MAX_ENVELOPES_VARVAR]; /*!< envelope frequency resolutions for VARVAR */ } SBR_GRID; typedef SBR_GRID *HANDLE_SBR_GRID; /*! \struct SBR_FRAME_INFO \brief time/frequency grid description for one frame */ typedef struct { INT nEnvelopes; /*!< number of envelopes */ INT borders[MAX_ENVELOPES + 1]; /*!< envelope borders in SBR timeslots */ FREQ_RES freqRes[MAX_ENVELOPES]; /*!< frequency resolution of each envelope */ INT shortEnv; /*!< number of an envelope to be shortened (starting at 1) or 0 for no shortened envelope */ INT nNoiseEnvelopes; /*!< number of noise floors */ INT bordersNoise[MAX_NOISE_ENVELOPES + 1]; /*!< noise floor borders in SBR timeslots */ } SBR_FRAME_INFO; /* WARNING: When rearranging the elements of this struct keep in mind that the * static initializations in the corresponding C-file have to be rearranged as * well! snd 2002/01/23 */ typedef SBR_FRAME_INFO *HANDLE_SBR_FRAME_INFO; /*! \struct SBR_ENVELOPE_FRAME \brief frame generator main struct Contains tuning parameters, time/frequency grid description, sbr_grid() bitstream elements, and generator internal signals */ typedef struct { /* system constants */ INT frameMiddleSlot; /*!< transient detector offset in SBR timeslots */ /* basic tuning parameters */ INT staticFraming; /*!< 1: run static framing in time, i.e. exclusive use of bs_frame_class = FIXFIX */ INT numEnvStatic; /*!< number of envelopes per frame for static framing */ FREQ_RES freq_res_fixfix[2]; /*!< envelope frequency resolution to use for bs_frame_class = FIXFIX; single env and split */ UCHAR fResTransIsLow; /*!< frequency resolution for transient frames - always low (0) or according to table (1) */ /* expert tuning parameters */ const int *v_tuningSegm; /*!< segment lengths to use around transient */ const int *v_tuningFreq; /*!< frequency resolutions to use around transient */ INT dmin; /*!< minimum length of dependent segments */ INT dmax; /*!< maximum length of dependent segments */ INT allowSpread; /*!< 1: allow isolated transient to influence grid of 3 consecutive frames */ /* internally used signals */ FRAME_CLASS frameClassOld; /*!< frame class used for previous frame */ INT spreadFlag; /*!< 1: use VARVAR instead of VARFIX to follow up old transient */ INT v_bord[2 * MAX_ENVELOPES_VARVAR + 1]; /*!< borders for current frame and preliminary borders for next frame (fixed borders excluded) */ INT length_v_bord; /*!< helper variable: length of v_bord */ INT v_freq[2 * MAX_ENVELOPES_VARVAR + 1]; /*!< frequency resolutions for current frame and preliminary resolutions for next frame */ INT length_v_freq; /*!< helper variable: length of v_freq */ INT v_bordFollow[MAX_ENVELOPES_VARVAR]; /*!< preliminary borders for current frame (calculated during previous frame) */ INT length_v_bordFollow; /*!< helper variable: length of v_bordFollow */ INT i_tranFollow; /*!< points to transient border in v_bordFollow (may be negative, see keepForFollowUp()) */ INT i_fillFollow; /*!< points to first fill border in v_bordFollow */ INT v_freqFollow[MAX_ENVELOPES_VARVAR]; /*!< preliminary frequency resolutions for current frame (calculated during previous frame) */ INT length_v_freqFollow; /*!< helper variable: length of v_freqFollow */ /* externally needed signals */ SBR_GRID SbrGrid; /*!< sbr_grid() signals to be converted to bitstream elements */ SBR_FRAME_INFO SbrFrameInfo; /*!< time/frequency grid description for one frame */ } SBR_ENVELOPE_FRAME; typedef SBR_ENVELOPE_FRAME *HANDLE_SBR_ENVELOPE_FRAME; void FDKsbrEnc_initFrameInfoGenerator(HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame, INT allowSpread, INT numEnvStatic, INT staticFraming, INT timeSlots, const FREQ_RES *freq_res_fixfix, UCHAR fResTransIsLow, INT ldGrid); HANDLE_SBR_FRAME_INFO FDKsbrEnc_frameInfoGenerator(HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame, UCHAR *v_transient_info, const INT rightBorderFIX, UCHAR *v_transient_info_pre, int ldGrid, const int *v_tuning); #endif fdk-aac-2.0.2/libSBRenc/src/invf_est.cpp000066400000000000000000000562711404225310700177250ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Description: *******************************************************************************/ #include "invf_est.h" #include "sbr_misc.h" #include "genericStds.h" #define MAX_NUM_REGIONS 10 #define SCALE_FAC_QUO 512.0f #define SCALE_FAC_NRG 256.0f #ifndef min #define min(a, b) (a < b ? a : b) #endif #ifndef max #define max(a, b) (a > b ? a : b) #endif static const FIXP_DBL quantStepsSbr[4] = { 0x00400000, 0x02800000, 0x03800000, 0x04c00000}; /* table scaled with SCALE_FAC_QUO */ static const FIXP_DBL quantStepsOrig[4] = { 0x00000000, 0x00c00000, 0x01c00000, 0x02800000}; /* table scaled with SCALE_FAC_QUO */ static const FIXP_DBL nrgBorders[4] = { 0x0c800000, 0x0f000000, 0x11800000, 0x14000000}; /* table scaled with SCALE_FAC_NRG */ static const DETECTOR_PARAMETERS detectorParamsAAC = { quantStepsSbr, quantStepsOrig, nrgBorders, 4, /* Number of borders SBR. */ 4, /* Number of borders orig. */ 4, /* Number of borders Nrg. */ { /* Region space. */ {INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF, INVF_OFF, INVF_OFF}, /* | */ {INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF, INVF_OFF, INVF_OFF}, /* | */ {INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF, INVF_OFF}, /* regionSbr */ {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF, INVF_OFF}, /* | */ {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF, INVF_OFF} /* | */ }, /*------------------------ regionOrig ---------------------------------*/ { /* Region space transient. */ {INVF_LOW_LEVEL, INVF_LOW_LEVEL, INVF_LOW_LEVEL, INVF_OFF, INVF_OFF}, /* | */ {INVF_LOW_LEVEL, INVF_LOW_LEVEL, INVF_LOW_LEVEL, INVF_OFF, INVF_OFF}, /* | */ {INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_OFF, INVF_OFF}, /* regionSbr */ {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF, INVF_OFF}, /* | */ {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF, INVF_OFF} /* | */ }, /*------------------------ regionOrig ---------------------------------*/ {-4, -3, -2, -1, 0} /* Reduction factor of the inverse filtering for low energies.*/ }; static const FIXP_DBL hysteresis = 0x00400000; /* Delta value for hysteresis. scaled with SCALE_FAC_QUO */ /* * AAC+SBR PARAMETERS for Speech *********************************/ static const DETECTOR_PARAMETERS detectorParamsAACSpeech = { quantStepsSbr, quantStepsOrig, nrgBorders, 4, /* Number of borders SBR. */ 4, /* Number of borders orig. */ 4, /* Number of borders Nrg. */ { /* Region space. */ {INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF, INVF_OFF}, /* | */ {INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF, INVF_OFF}, /* | */ {INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_OFF, INVF_OFF}, /* regionSbr */ {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF, INVF_OFF}, /* | */ {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF, INVF_OFF} /* | */ }, /*------------------------ regionOrig ---------------------------------*/ { /* Region space transient. */ {INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF, INVF_OFF}, /* | */ {INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF, INVF_OFF}, /* | */ {INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_OFF, INVF_OFF}, /* regionSbr */ {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF, INVF_OFF}, /* | */ {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF, INVF_OFF} /* | */ }, /*------------------------ regionOrig ---------------------------------*/ {-4, -3, -2, -1, 0} /* Reduction factor of the inverse filtering for low energies.*/ }; /* * Smoothing filters. ************************/ typedef const FIXP_DBL FIR_FILTER[5]; static const FIR_FILTER fir_0 = {0x7fffffff, 0x00000000, 0x00000000, 0x00000000, 0x00000000}; static const FIR_FILTER fir_1 = {0x2aaaaa80, 0x555554ff, 0x00000000, 0x00000000, 0x00000000}; static const FIR_FILTER fir_2 = {0x10000000, 0x30000000, 0x40000000, 0x00000000, 0x00000000}; static const FIR_FILTER fir_3 = {0x077f80e8, 0x199999a0, 0x2bb3b240, 0x33333340, 0x00000000}; static const FIR_FILTER fir_4 = {0x04130598, 0x0ebdb000, 0x1becfa60, 0x2697a4c0, 0x2aaaaa80}; static const FIR_FILTER *const fir_table[5] = {&fir_0, &fir_1, &fir_2, &fir_3, &fir_4}; /**************************************************************************/ /*! \brief Calculates the values used for the detector. \return none */ /**************************************************************************/ static void calculateDetectorValues( FIXP_DBL **quotaMatrixOrig, /*!< Matrix holding the tonality values of the original. */ SCHAR *indexVector, /*!< Index vector to obtain the patched data. */ FIXP_DBL *nrgVector, /*!< Energy vector. */ DETECTOR_VALUES *detectorValues, /*!< pointer to DETECTOR_VALUES struct. */ INT startChannel, /*!< Start channel. */ INT stopChannel, /*!< Stop channel. */ INT startIndex, /*!< Start index. */ INT stopIndex, /*!< Stop index. */ INT numberOfStrongest /*!< The number of sorted tonal components to be considered. */ ) { INT i, temp, j; const FIXP_DBL *filter = *fir_table[INVF_SMOOTHING_LENGTH]; FIXP_DBL origQuotaMeanStrongest, sbrQuotaMeanStrongest; FIXP_DBL origQuota, sbrQuota; FIXP_DBL invIndex, invChannel, invTemp; FIXP_DBL quotaVecOrig[64], quotaVecSbr[64]; FDKmemclear(quotaVecOrig, 64 * sizeof(FIXP_DBL)); FDKmemclear(quotaVecSbr, 64 * sizeof(FIXP_DBL)); invIndex = GetInvInt(stopIndex - startIndex); invChannel = GetInvInt(stopChannel - startChannel); /* Calculate the mean value, over the current time segment, for the original, the HFR and the difference, over all channels in the current frequency range. NOTE: the averaging is done on the values quota/(1 - quota + RELAXATION). */ /* The original, the sbr signal and the total energy */ detectorValues->avgNrg = FL2FXCONST_DBL(0.0f); for (j = startIndex; j < stopIndex; j++) { for (i = startChannel; i < stopChannel; i++) { quotaVecOrig[i] += fMult(quotaMatrixOrig[j][i], invIndex); if (indexVector[i] != -1) quotaVecSbr[i] += fMult(quotaMatrixOrig[j][indexVector[i]], invIndex); } detectorValues->avgNrg += fMult(nrgVector[j], invIndex); } /* Calculate the mean value, over the current frequency range, for the original, the HFR and the difference. Also calculate the same mean values for the three vectors, but only includeing the x strongest copmponents. */ origQuota = FL2FXCONST_DBL(0.0f); sbrQuota = FL2FXCONST_DBL(0.0f); for (i = startChannel; i < stopChannel; i++) { origQuota += fMultDiv2(quotaVecOrig[i], invChannel); sbrQuota += fMultDiv2(quotaVecSbr[i], invChannel); } /* Calculate the mean value for the x strongest components */ FDKsbrEnc_Shellsort_fract(quotaVecOrig + startChannel, stopChannel - startChannel); FDKsbrEnc_Shellsort_fract(quotaVecSbr + startChannel, stopChannel - startChannel); origQuotaMeanStrongest = FL2FXCONST_DBL(0.0f); sbrQuotaMeanStrongest = FL2FXCONST_DBL(0.0f); temp = min(stopChannel - startChannel, numberOfStrongest); invTemp = GetInvInt(temp); for (i = 0; i < temp; i++) { origQuotaMeanStrongest += fMultDiv2(quotaVecOrig[i + stopChannel - temp], invTemp); sbrQuotaMeanStrongest += fMultDiv2(quotaVecSbr[i + stopChannel - temp], invTemp); } /* The value for the strongest component */ detectorValues->origQuotaMax = quotaVecOrig[stopChannel - 1]; detectorValues->sbrQuotaMax = quotaVecSbr[stopChannel - 1]; /* Buffer values */ FDKmemmove(detectorValues->origQuotaMean, detectorValues->origQuotaMean + 1, INVF_SMOOTHING_LENGTH * sizeof(FIXP_DBL)); FDKmemmove(detectorValues->sbrQuotaMean, detectorValues->sbrQuotaMean + 1, INVF_SMOOTHING_LENGTH * sizeof(FIXP_DBL)); FDKmemmove(detectorValues->origQuotaMeanStrongest, detectorValues->origQuotaMeanStrongest + 1, INVF_SMOOTHING_LENGTH * sizeof(FIXP_DBL)); FDKmemmove(detectorValues->sbrQuotaMeanStrongest, detectorValues->sbrQuotaMeanStrongest + 1, INVF_SMOOTHING_LENGTH * sizeof(FIXP_DBL)); detectorValues->origQuotaMean[INVF_SMOOTHING_LENGTH] = origQuota << 1; detectorValues->sbrQuotaMean[INVF_SMOOTHING_LENGTH] = sbrQuota << 1; detectorValues->origQuotaMeanStrongest[INVF_SMOOTHING_LENGTH] = origQuotaMeanStrongest << 1; detectorValues->sbrQuotaMeanStrongest[INVF_SMOOTHING_LENGTH] = sbrQuotaMeanStrongest << 1; /* Filter values */ detectorValues->origQuotaMeanFilt = FL2FXCONST_DBL(0.0f); detectorValues->sbrQuotaMeanFilt = FL2FXCONST_DBL(0.0f); detectorValues->origQuotaMeanStrongestFilt = FL2FXCONST_DBL(0.0f); detectorValues->sbrQuotaMeanStrongestFilt = FL2FXCONST_DBL(0.0f); for (i = 0; i < INVF_SMOOTHING_LENGTH + 1; i++) { detectorValues->origQuotaMeanFilt += fMult(detectorValues->origQuotaMean[i], filter[i]); detectorValues->sbrQuotaMeanFilt += fMult(detectorValues->sbrQuotaMean[i], filter[i]); detectorValues->origQuotaMeanStrongestFilt += fMult(detectorValues->origQuotaMeanStrongest[i], filter[i]); detectorValues->sbrQuotaMeanStrongestFilt += fMult(detectorValues->sbrQuotaMeanStrongest[i], filter[i]); } } /**************************************************************************/ /*! \brief Returns the region in which the input value belongs. \return region. */ /**************************************************************************/ static INT findRegion( FIXP_DBL currVal, /*!< The current value. */ const FIXP_DBL *borders, /*!< The border of the regions. */ const INT numBorders /*!< The number of borders. */ ) { INT i; if (currVal < borders[0]) { return 0; } for (i = 1; i < numBorders; i++) { if (currVal >= borders[i - 1] && currVal < borders[i]) { return i; } } if (currVal >= borders[numBorders - 1]) { return numBorders; } return 0; /* We never get here, it's just to avoid compiler warnings.*/ } /**************************************************************************/ /*! \brief Makes a clever decision based on the quota vector. \return decision on which invf mode to use */ /**************************************************************************/ static INVF_MODE decisionAlgorithm( const DETECTOR_PARAMETERS *detectorParams, /*!< Struct with the detector parameters. */ DETECTOR_VALUES *detectorValues, /*!< Struct with the detector values. */ INT transientFlag, /*!< Flag indicating if there is a transient present.*/ INT *prevRegionSbr, /*!< The previous region in which the Sbr value was. */ INT *prevRegionOrig /*!< The previous region in which the Orig value was. */ ) { INT invFiltLevel, regionSbr, regionOrig, regionNrg; /* Current thresholds. */ const INT numRegionsSbr = detectorParams->numRegionsSbr; const INT numRegionsOrig = detectorParams->numRegionsOrig; const INT numRegionsNrg = detectorParams->numRegionsNrg; FIXP_DBL quantStepsSbrTmp[MAX_NUM_REGIONS]; FIXP_DBL quantStepsOrigTmp[MAX_NUM_REGIONS]; /* Current detector values. */ FIXP_DBL origQuotaMeanFilt; FIXP_DBL sbrQuotaMeanFilt; FIXP_DBL nrg; /* 0.375 = 3.0 / 8.0; 0.31143075889 = log2(RELAXATION)/64.0; 0.625 = * log(16)/64.0; 0.6875 = 44/64.0 */ origQuotaMeanFilt = (fMultDiv2(FL2FXCONST_DBL(2.f * 0.375f), (FIXP_DBL)(CalcLdData(max(detectorValues->origQuotaMeanFilt, (FIXP_DBL)1)) + FL2FXCONST_DBL(0.31143075889f)))) << 0; /* scaled by 1/2^9 */ sbrQuotaMeanFilt = (fMultDiv2(FL2FXCONST_DBL(2.f * 0.375f), (FIXP_DBL)(CalcLdData(max(detectorValues->sbrQuotaMeanFilt, (FIXP_DBL)1)) + FL2FXCONST_DBL(0.31143075889f)))) << 0; /* scaled by 1/2^9 */ /* If energy is zero then we will get different results for different word * lengths. */ nrg = (fMultDiv2(FL2FXCONST_DBL(2.f * 0.375f), (FIXP_DBL)(CalcLdData(detectorValues->avgNrg + (FIXP_DBL)1) + FL2FXCONST_DBL(0.0625f) + FL2FXCONST_DBL(0.6875f)))) << 0; /* scaled by 1/2^8; 2^44 -> qmf energy scale */ FDKmemcpy(quantStepsSbrTmp, detectorParams->quantStepsSbr, numRegionsSbr * sizeof(FIXP_DBL)); FDKmemcpy(quantStepsOrigTmp, detectorParams->quantStepsOrig, numRegionsOrig * sizeof(FIXP_DBL)); if (*prevRegionSbr < numRegionsSbr) quantStepsSbrTmp[*prevRegionSbr] = detectorParams->quantStepsSbr[*prevRegionSbr] + hysteresis; if (*prevRegionSbr > 0) quantStepsSbrTmp[*prevRegionSbr - 1] = detectorParams->quantStepsSbr[*prevRegionSbr - 1] - hysteresis; if (*prevRegionOrig < numRegionsOrig) quantStepsOrigTmp[*prevRegionOrig] = detectorParams->quantStepsOrig[*prevRegionOrig] + hysteresis; if (*prevRegionOrig > 0) quantStepsOrigTmp[*prevRegionOrig - 1] = detectorParams->quantStepsOrig[*prevRegionOrig - 1] - hysteresis; regionSbr = findRegion(sbrQuotaMeanFilt, quantStepsSbrTmp, numRegionsSbr); regionOrig = findRegion(origQuotaMeanFilt, quantStepsOrigTmp, numRegionsOrig); regionNrg = findRegion(nrg, detectorParams->nrgBorders, numRegionsNrg); *prevRegionSbr = regionSbr; *prevRegionOrig = regionOrig; /* Use different settings if a transient is present*/ invFiltLevel = (transientFlag == 1) ? detectorParams->regionSpaceTransient[regionSbr][regionOrig] : detectorParams->regionSpace[regionSbr][regionOrig]; /* Compensate for low energy.*/ invFiltLevel = max(invFiltLevel + detectorParams->EnergyCompFactor[regionNrg], 0); return (INVF_MODE)(invFiltLevel); } /**************************************************************************/ /*! \brief Estiamtion of the inverse filtering level required in the decoder. A second order LPC is calculated for every filterbank channel, using the covariance method. THe ratio between the energy of the predicted signal and the energy of the non-predictable signal is calcualted. \return none. */ /**************************************************************************/ void FDKsbrEnc_qmfInverseFilteringDetector( HANDLE_SBR_INV_FILT_EST hInvFilt, /*!< Handle to the SBR_INV_FILT_EST struct. */ FIXP_DBL **quotaMatrix, /*!< The matrix holding the tonality values of the original. */ FIXP_DBL *nrgVector, /*!< The energy vector. */ SCHAR *indexVector, /*!< Index vector to obtain the patched data. */ INT startIndex, /*!< Start index. */ INT stopIndex, /*!< Stop index. */ INT transientFlag, /*!< Flag indicating if a transient is present or not.*/ INVF_MODE *infVec /*!< Vector holding the inverse filtering levels. */ ) { INT band; /* * Do the inverse filtering level estimation. *****************************************************/ for (band = 0; band < hInvFilt->noDetectorBands; band++) { INT startChannel = hInvFilt->freqBandTableInvFilt[band]; INT stopChannel = hInvFilt->freqBandTableInvFilt[band + 1]; calculateDetectorValues(quotaMatrix, indexVector, nrgVector, &hInvFilt->detectorValues[band], startChannel, stopChannel, startIndex, stopIndex, hInvFilt->numberOfStrongest); infVec[band] = decisionAlgorithm( hInvFilt->detectorParams, &hInvFilt->detectorValues[band], transientFlag, &hInvFilt->prevRegionSbr[band], &hInvFilt->prevRegionOrig[band]); } } /**************************************************************************/ /*! \brief Initialize an instance of the inverse filtering level estimator. \return errorCode, noError if successful. */ /**************************************************************************/ INT FDKsbrEnc_initInvFiltDetector( HANDLE_SBR_INV_FILT_EST hInvFilt, /*!< Pointer to a handle to the SBR_INV_FILT_EST struct. */ INT *freqBandTableDetector, /*!< Frequency band table for the inverse filtering. */ INT numDetectorBands, /*!< Number of inverse filtering bands. */ UINT useSpeechConfig /*!< Flag: adapt tuning parameters according to speech*/ ) { INT i; FDKmemclear(hInvFilt, sizeof(SBR_INV_FILT_EST)); hInvFilt->detectorParams = (useSpeechConfig) ? &detectorParamsAACSpeech : &detectorParamsAAC; hInvFilt->noDetectorBandsMax = numDetectorBands; /* Memory initialisation */ for (i = 0; i < hInvFilt->noDetectorBandsMax; i++) { FDKmemclear(&hInvFilt->detectorValues[i], sizeof(DETECTOR_VALUES)); hInvFilt->prevInvfMode[i] = INVF_OFF; hInvFilt->prevRegionOrig[i] = 0; hInvFilt->prevRegionSbr[i] = 0; } /* Reset the inverse fltering detector. */ FDKsbrEnc_resetInvFiltDetector(hInvFilt, freqBandTableDetector, hInvFilt->noDetectorBandsMax); return (0); } /**************************************************************************/ /*! \brief resets sbr inverse filtering structure. \return errorCode, noError if successful. */ /**************************************************************************/ INT FDKsbrEnc_resetInvFiltDetector( HANDLE_SBR_INV_FILT_EST hInvFilt, /*!< Handle to the SBR_INV_FILT_EST struct. */ INT *freqBandTableDetector, /*!< Frequency band table for the inverse filtering. */ INT numDetectorBands) /*!< Number of inverse filtering bands. */ { hInvFilt->numberOfStrongest = 1; FDKmemcpy(hInvFilt->freqBandTableInvFilt, freqBandTableDetector, (numDetectorBands + 1) * sizeof(INT)); hInvFilt->noDetectorBands = numDetectorBands; return (0); } fdk-aac-2.0.2/libSBRenc/src/invf_est.h000066400000000000000000000157601404225310700173700ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Inverse Filtering detection prototypes $Revision: 92790 $ */ #ifndef INVF_EST_H #define INVF_EST_H #include "sbr_encoder.h" #include "sbr_def.h" #define INVF_SMOOTHING_LENGTH 2 typedef struct { const FIXP_DBL *quantStepsSbr; const FIXP_DBL *quantStepsOrig; const FIXP_DBL *nrgBorders; INT numRegionsSbr; INT numRegionsOrig; INT numRegionsNrg; INVF_MODE regionSpace[5][5]; INVF_MODE regionSpaceTransient[5][5]; INT EnergyCompFactor[5]; } DETECTOR_PARAMETERS; typedef struct { FIXP_DBL origQuotaMean[INVF_SMOOTHING_LENGTH + 1]; FIXP_DBL sbrQuotaMean[INVF_SMOOTHING_LENGTH + 1]; FIXP_DBL origQuotaMeanStrongest[INVF_SMOOTHING_LENGTH + 1]; FIXP_DBL sbrQuotaMeanStrongest[INVF_SMOOTHING_LENGTH + 1]; FIXP_DBL origQuotaMeanFilt; FIXP_DBL sbrQuotaMeanFilt; FIXP_DBL origQuotaMeanStrongestFilt; FIXP_DBL sbrQuotaMeanStrongestFilt; FIXP_DBL origQuotaMax; FIXP_DBL sbrQuotaMax; FIXP_DBL avgNrg; } DETECTOR_VALUES; typedef struct { INT numberOfStrongest; INT prevRegionSbr[MAX_NUM_NOISE_VALUES]; INT prevRegionOrig[MAX_NUM_NOISE_VALUES]; INT freqBandTableInvFilt[MAX_NUM_NOISE_VALUES]; INT noDetectorBands; INT noDetectorBandsMax; const DETECTOR_PARAMETERS *detectorParams; INVF_MODE prevInvfMode[MAX_NUM_NOISE_VALUES]; DETECTOR_VALUES detectorValues[MAX_NUM_NOISE_VALUES]; FIXP_DBL nrgAvg; FIXP_DBL wmQmf[MAX_NUM_NOISE_VALUES]; } SBR_INV_FILT_EST; typedef SBR_INV_FILT_EST *HANDLE_SBR_INV_FILT_EST; void FDKsbrEnc_qmfInverseFilteringDetector(HANDLE_SBR_INV_FILT_EST hInvFilt, FIXP_DBL **quotaMatrix, FIXP_DBL *nrgVector, SCHAR *indexVector, INT startIndex, INT stopIndex, INT transientFlag, INVF_MODE *infVec); INT FDKsbrEnc_initInvFiltDetector(HANDLE_SBR_INV_FILT_EST hInvFilt, INT *freqBandTableDetector, INT numDetectorBands, UINT useSpeechConfig); INT FDKsbrEnc_resetInvFiltDetector(HANDLE_SBR_INV_FILT_EST hInvFilt, INT *freqBandTableDetector, INT numDetectorBands); #endif /* _QMF_INV_FILT_H */ fdk-aac-2.0.2/libSBRenc/src/mh_det.cpp000066400000000000000000001435521404225310700173470ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Description: *******************************************************************************/ #include "mh_det.h" #include "sbrenc_ram.h" #include "sbr_misc.h" #include "genericStds.h" #define SFM_SHIFT 2 /* Attention: SFM_SCALE depends on SFM_SHIFT */ #define SFM_SCALE (MAXVAL_DBL >> SFM_SHIFT) /* 1.0 >> SFM_SHIFT */ /*!< Detector Parameters for AAC core codec. */ static const DETECTOR_PARAMETERS_MH paramsAac = { 9, /*!< deltaTime */ { FL2FXCONST_DBL(20.0f * RELAXATION_FLOAT), /*!< thresHoldDiff */ FL2FXCONST_DBL(1.26f * RELAXATION_FLOAT), /*!< thresHoldDiffGuide */ FL2FXCONST_DBL(15.0f * RELAXATION_FLOAT), /*!< thresHoldTone */ FL2FXCONST_DBL((1.0f / 15.0f) * RELAXATION_FLOAT), /*!< invThresHoldTone */ FL2FXCONST_DBL(1.26f * RELAXATION_FLOAT), /*!< thresHoldToneGuide */ FL2FXCONST_DBL(0.3f) >> SFM_SHIFT, /*!< sfmThresSbr */ FL2FXCONST_DBL(0.1f) >> SFM_SHIFT, /*!< sfmThresOrig */ FL2FXCONST_DBL(0.3f), /*!< decayGuideOrig */ FL2FXCONST_DBL(0.5f), /*!< decayGuideDiff */ FL2FXCONST_DBL(-0.000112993269), /* LD64(FL2FXCONST_DBL(0.995f)) */ /*!< derivThresMaxLD64 */ FL2FXCONST_DBL(-0.000112993269), /* LD64(FL2FXCONST_DBL(0.995f)) */ /*!< derivThresBelowLD64 */ FL2FXCONST_DBL( -0.005030126483f) /* LD64(FL2FXCONST_DBL(0.8f)) */ /*!< derivThresAboveLD64 */ }, 50 /*!< maxComp */ }; /*!< Detector Parameters for AAC LD core codec. */ static const DETECTOR_PARAMETERS_MH paramsAacLd = { 16, /*!< Delta time. */ { FL2FXCONST_DBL(25.0f * RELAXATION_FLOAT), /*!< thresHoldDiff */ FL2FXCONST_DBL(1.26f * RELAXATION_FLOAT), /*!< tresHoldDiffGuide */ FL2FXCONST_DBL(15.0f * RELAXATION_FLOAT), /*!< thresHoldTone */ FL2FXCONST_DBL((1.0f / 15.0f) * RELAXATION_FLOAT), /*!< invThresHoldTone */ FL2FXCONST_DBL(1.26f * RELAXATION_FLOAT), /*!< thresHoldToneGuide */ FL2FXCONST_DBL(0.3f) >> SFM_SHIFT, /*!< sfmThresSbr */ FL2FXCONST_DBL(0.1f) >> SFM_SHIFT, /*!< sfmThresOrig */ FL2FXCONST_DBL(0.3f), /*!< decayGuideOrig */ FL2FXCONST_DBL(0.2f), /*!< decayGuideDiff */ FL2FXCONST_DBL(-0.000112993269), /* LD64(FL2FXCONST_DBL(0.995f)) */ /*!< derivThresMaxLD64 */ FL2FXCONST_DBL(-0.000112993269), /* LD64(FL2FXCONST_DBL(0.995f)) */ /*!< derivThresBelowLD64 */ FL2FXCONST_DBL( -0.005030126483f) /* LD64(FL2FXCONST_DBL(0.8f)) */ /*!< derivThresAboveLD64 */ }, 50 /*!< maxComp */ }; /**************************************************************************/ /*! \brief Calculates the difference in tonality between original and SBR for a given time and frequency region. The values for pDiffMapped2Scfb are scaled by RELAXATION \return none. */ /**************************************************************************/ static void diff(FIXP_DBL *RESTRICT pTonalityOrig, FIXP_DBL *pDiffMapped2Scfb, const UCHAR *RESTRICT pFreqBandTable, INT nScfb, SCHAR *indexVector) { UCHAR i, ll, lu, k; FIXP_DBL maxValOrig, maxValSbr, tmp; INT scale; for (i = 0; i < nScfb; i++) { ll = pFreqBandTable[i]; lu = pFreqBandTable[i + 1]; maxValOrig = FL2FXCONST_DBL(0.0f); maxValSbr = FL2FXCONST_DBL(0.0f); for (k = ll; k < lu; k++) { maxValOrig = fixMax(maxValOrig, pTonalityOrig[k]); maxValSbr = fixMax(maxValSbr, pTonalityOrig[indexVector[k]]); } if ((maxValSbr >= RELAXATION)) { tmp = fDivNorm(maxValOrig, maxValSbr, &scale); pDiffMapped2Scfb[i] = scaleValue(fMult(tmp, RELAXATION_FRACT), fixMax(-(DFRACT_BITS - 1), (scale - RELAXATION_SHIFT))); } else { pDiffMapped2Scfb[i] = maxValOrig; } } } /**************************************************************************/ /*! \brief Calculates a flatness measure of the tonality measures. Calculation of the power function and using scalefactor for basis: Using log2: z = (2^k * x)^y; z' = CalcLd(z) = y*CalcLd(x) + y*k; z = CalcInvLd(z'); Using ld64: z = (2^k * x)^y; z' = CalcLd64(z) = y*CalcLd64(x)/64 + y*k/64; z = CalcInvLd64(z'); The values pSfmOrigVec and pSfmSbrVec are scaled by the factor 1/4.0 \return none. */ /**************************************************************************/ static void calculateFlatnessMeasure(FIXP_DBL *pQuotaBuffer, SCHAR *indexVector, FIXP_DBL *pSfmOrigVec, FIXP_DBL *pSfmSbrVec, const UCHAR *pFreqBandTable, INT nSfb) { INT i, j; FIXP_DBL invBands, tmp1, tmp2; INT shiftFac0, shiftFacSum0; INT shiftFac1, shiftFacSum1; FIXP_DBL accu; for (i = 0; i < nSfb; i++) { INT ll = pFreqBandTable[i]; INT lu = pFreqBandTable[i + 1]; pSfmOrigVec[i] = (FIXP_DBL)(MAXVAL_DBL >> 2); pSfmSbrVec[i] = (FIXP_DBL)(MAXVAL_DBL >> 2); if (lu - ll > 1) { FIXP_DBL amOrig, amTransp, gmOrig, gmTransp, sfmOrig, sfmTransp; invBands = GetInvInt(lu - ll); shiftFacSum0 = 0; shiftFacSum1 = 0; amOrig = amTransp = FL2FXCONST_DBL(0.0f); gmOrig = gmTransp = (FIXP_DBL)MAXVAL_DBL; for (j = ll; j < lu; j++) { sfmOrig = pQuotaBuffer[j]; sfmTransp = pQuotaBuffer[indexVector[j]]; amOrig += fMult(sfmOrig, invBands); amTransp += fMult(sfmTransp, invBands); shiftFac0 = CountLeadingBits(sfmOrig); shiftFac1 = CountLeadingBits(sfmTransp); gmOrig = fMult(gmOrig, sfmOrig << shiftFac0); gmTransp = fMult(gmTransp, sfmTransp << shiftFac1); shiftFacSum0 += shiftFac0; shiftFacSum1 += shiftFac1; } if (gmOrig > FL2FXCONST_DBL(0.0f)) { tmp1 = CalcLdData(gmOrig); /* CalcLd64(x)/64 */ tmp1 = fMult(invBands, tmp1); /* y*CalcLd64(x)/64 */ /* y*k/64 */ accu = (FIXP_DBL)-shiftFacSum0 << (DFRACT_BITS - 1 - 8); tmp2 = fMultDiv2(invBands, accu) << (2 + 1); tmp2 = tmp1 + tmp2; /* y*CalcLd64(x)/64 + y*k/64 */ gmOrig = CalcInvLdData(tmp2); /* CalcInvLd64(z'); */ } else { gmOrig = FL2FXCONST_DBL(0.0f); } if (gmTransp > FL2FXCONST_DBL(0.0f)) { tmp1 = CalcLdData(gmTransp); /* CalcLd64(x)/64 */ tmp1 = fMult(invBands, tmp1); /* y*CalcLd64(x)/64 */ /* y*k/64 */ accu = (FIXP_DBL)-shiftFacSum1 << (DFRACT_BITS - 1 - 8); tmp2 = fMultDiv2(invBands, accu) << (2 + 1); tmp2 = tmp1 + tmp2; /* y*CalcLd64(x)/64 + y*k/64 */ gmTransp = CalcInvLdData(tmp2); /* CalcInvLd64(z'); */ } else { gmTransp = FL2FXCONST_DBL(0.0f); } if (amOrig != FL2FXCONST_DBL(0.0f)) pSfmOrigVec[i] = FDKsbrEnc_LSI_divide_scale_fract(gmOrig, amOrig, SFM_SCALE); if (amTransp != FL2FXCONST_DBL(0.0f)) pSfmSbrVec[i] = FDKsbrEnc_LSI_divide_scale_fract(gmTransp, amTransp, SFM_SCALE); } } } /**************************************************************************/ /*! \brief Calculates the input to the missing harmonics detection. \return none. */ /**************************************************************************/ static void calculateDetectorInput( FIXP_DBL **RESTRICT pQuotaBuffer, /*!< Pointer to tonality matrix. */ SCHAR *RESTRICT indexVector, FIXP_DBL **RESTRICT tonalityDiff, FIXP_DBL **RESTRICT pSfmOrig, FIXP_DBL **RESTRICT pSfmSbr, const UCHAR *freqBandTable, INT nSfb, INT noEstPerFrame, INT move) { INT est; /* New estimate. */ for (est = 0; est < noEstPerFrame; est++) { diff(pQuotaBuffer[est + move], tonalityDiff[est + move], freqBandTable, nSfb, indexVector); calculateFlatnessMeasure(pQuotaBuffer[est + move], indexVector, pSfmOrig[est + move], pSfmSbr[est + move], freqBandTable, nSfb); } } /**************************************************************************/ /*! \brief Checks that the detection is not due to a LP filter This function determines if a newly detected missing harmonics is not in fact just a low-pass filtere input signal. If so, the detection is removed. \return none. */ /**************************************************************************/ static void removeLowPassDetection(UCHAR *RESTRICT pAddHarmSfb, UCHAR **RESTRICT pDetectionVectors, INT start, INT stop, INT nSfb, const UCHAR *RESTRICT pFreqBandTable, FIXP_DBL *RESTRICT pNrgVector, THRES_HOLDS mhThresh) { INT i, est; INT maxDerivPos = pFreqBandTable[nSfb]; INT numBands = pFreqBandTable[nSfb]; FIXP_DBL nrgLow, nrgHigh; FIXP_DBL nrgLD64, nrgLowLD64, nrgHighLD64, nrgDiffLD64; FIXP_DBL valLD64, maxValLD64, maxValAboveLD64; INT bLPsignal = 0; maxValLD64 = FL2FXCONST_DBL(-1.0f); for (i = numBands - 1 - 2; i > pFreqBandTable[0]; i--) { nrgLow = pNrgVector[i]; nrgHigh = pNrgVector[i + 2]; if (nrgLow != FL2FXCONST_DBL(0.0f) && nrgLow > nrgHigh) { nrgLowLD64 = CalcLdData(nrgLow >> 1); nrgDiffLD64 = CalcLdData((nrgLow >> 1) - (nrgHigh >> 1)); valLD64 = nrgDiffLD64 - nrgLowLD64; if (valLD64 > maxValLD64) { maxDerivPos = i; maxValLD64 = valLD64; } if (maxValLD64 > mhThresh.derivThresMaxLD64) { break; } } } /* Find the largest "gradient" above. (should be relatively flat, hence we expect a low value if the signal is LP.*/ maxValAboveLD64 = FL2FXCONST_DBL(-1.0f); for (i = numBands - 1 - 2; i > maxDerivPos + 2; i--) { nrgLow = pNrgVector[i]; nrgHigh = pNrgVector[i + 2]; if (nrgLow != FL2FXCONST_DBL(0.0f) && nrgLow > nrgHigh) { nrgLowLD64 = CalcLdData(nrgLow >> 1); nrgDiffLD64 = CalcLdData((nrgLow >> 1) - (nrgHigh >> 1)); valLD64 = nrgDiffLD64 - nrgLowLD64; if (valLD64 > maxValAboveLD64) { maxValAboveLD64 = valLD64; } } else { if (nrgHigh != FL2FXCONST_DBL(0.0f) && nrgHigh > nrgLow) { nrgHighLD64 = CalcLdData(nrgHigh >> 1); nrgDiffLD64 = CalcLdData((nrgHigh >> 1) - (nrgLow >> 1)); valLD64 = nrgDiffLD64 - nrgHighLD64; if (valLD64 > maxValAboveLD64) { maxValAboveLD64 = valLD64; } } } } if (maxValLD64 > mhThresh.derivThresMaxLD64 && maxValAboveLD64 < mhThresh.derivThresAboveLD64) { bLPsignal = 1; for (i = maxDerivPos - 1; i > maxDerivPos - 5 && i >= 0; i--) { if (pNrgVector[i] != FL2FXCONST_DBL(0.0f) && pNrgVector[i] > pNrgVector[maxDerivPos + 2]) { nrgDiffLD64 = CalcLdData((pNrgVector[i] >> 1) - (pNrgVector[maxDerivPos + 2] >> 1)); nrgLD64 = CalcLdData(pNrgVector[i] >> 1); valLD64 = nrgDiffLD64 - nrgLD64; if (valLD64 < mhThresh.derivThresBelowLD64) { bLPsignal = 0; break; } } else { bLPsignal = 0; break; } } } if (bLPsignal) { for (i = 0; i < nSfb; i++) { if (maxDerivPos >= pFreqBandTable[i] && maxDerivPos < pFreqBandTable[i + 1]) break; } if (pAddHarmSfb[i]) { pAddHarmSfb[i] = 0; for (est = start; est < stop; est++) { pDetectionVectors[est][i] = 0; } } } } /**************************************************************************/ /*! \brief Checks if it is allowed to detect a missing tone, that wasn't detected previously. \return newDetectionAllowed flag. */ /**************************************************************************/ static INT isDetectionOfNewToneAllowed( const SBR_FRAME_INFO *pFrameInfo, INT *pDetectionStartPos, INT noEstPerFrame, INT prevTransientFrame, INT prevTransientPos, INT prevTransientFlag, INT transientPosOffset, INT transientFlag, INT transientPos, INT deltaTime, HANDLE_SBR_MISSING_HARMONICS_DETECTOR h_sbrMissingHarmonicsDetector) { INT transientFrame, newDetectionAllowed; /* Determine if this is a frame where a transient starts... * If the transient flag was set the previous frame but not the * transient frame flag, the transient frame flag is set in the current frame. *****************************************************************************/ transientFrame = 0; if (transientFlag) { if (transientPos + transientPosOffset < pFrameInfo->borders[pFrameInfo->nEnvelopes]) { transientFrame = 1; if (noEstPerFrame > 1) { if (transientPos + transientPosOffset > h_sbrMissingHarmonicsDetector->timeSlots >> 1) { *pDetectionStartPos = noEstPerFrame; } else { *pDetectionStartPos = noEstPerFrame >> 1; } } else { *pDetectionStartPos = noEstPerFrame; } } } else { if (prevTransientFlag && !prevTransientFrame) { transientFrame = 1; *pDetectionStartPos = 0; } } /* * Determine if detection of new missing harmonics are allowed. * If the frame contains a transient it's ok. If the previous * frame contained a transient it needs to be sufficiently close * to the start of the current frame. ****************************************************************/ newDetectionAllowed = 0; if (transientFrame) { newDetectionAllowed = 1; } else { if (prevTransientFrame && fixp_abs(pFrameInfo->borders[0] - (prevTransientPos + transientPosOffset - h_sbrMissingHarmonicsDetector->timeSlots)) < deltaTime) { newDetectionAllowed = 1; *pDetectionStartPos = 0; } } h_sbrMissingHarmonicsDetector->previousTransientFlag = transientFlag; h_sbrMissingHarmonicsDetector->previousTransientFrame = transientFrame; h_sbrMissingHarmonicsDetector->previousTransientPos = transientPos; return (newDetectionAllowed); } /**************************************************************************/ /*! \brief Cleans up the detection after a transient. \return none. */ /**************************************************************************/ static void transientCleanUp(FIXP_DBL **quotaBuffer, INT nSfb, UCHAR **detectionVectors, UCHAR *pAddHarmSfb, UCHAR *pPrevAddHarmSfb, INT **signBuffer, const UCHAR *pFreqBandTable, INT start, INT stop, INT newDetectionAllowed, FIXP_DBL *pNrgVector, THRES_HOLDS mhThresh) { INT i, j, est; for (est = start; est < stop; est++) { for (i = 0; i < nSfb; i++) { pAddHarmSfb[i] = pAddHarmSfb[i] || detectionVectors[est][i]; } } if (newDetectionAllowed == 1) { /* * Check for duplication of sines located * on the border of two scf-bands. *************************************************/ for (i = 0; i < nSfb - 1; i++) { /* detection in adjacent channels.*/ if (pAddHarmSfb[i] && pAddHarmSfb[i + 1]) { FIXP_DBL maxVal1, maxVal2; INT maxPos1, maxPos2, maxPosTime1, maxPosTime2; INT li = pFreqBandTable[i]; INT ui = pFreqBandTable[i + 1]; /* Find maximum tonality in the the two scf bands.*/ maxPosTime1 = start; maxPos1 = li; maxVal1 = quotaBuffer[start][li]; for (est = start; est < stop; est++) { for (j = li; j < ui; j++) { if (quotaBuffer[est][j] > maxVal1) { maxVal1 = quotaBuffer[est][j]; maxPos1 = j; maxPosTime1 = est; } } } li = pFreqBandTable[i + 1]; ui = pFreqBandTable[i + 2]; /* Find maximum tonality in the the two scf bands.*/ maxPosTime2 = start; maxPos2 = li; maxVal2 = quotaBuffer[start][li]; for (est = start; est < stop; est++) { for (j = li; j < ui; j++) { if (quotaBuffer[est][j] > maxVal2) { maxVal2 = quotaBuffer[est][j]; maxPos2 = j; maxPosTime2 = est; } } } /* If the maximum values are in adjacent QMF-channels, we need to remove the lowest of the two.*/ if (maxPos2 - maxPos1 < 2) { if (pPrevAddHarmSfb[i] == 1 && pPrevAddHarmSfb[i + 1] == 0) { /* Keep the lower, remove the upper.*/ pAddHarmSfb[i + 1] = 0; for (est = start; est < stop; est++) { detectionVectors[est][i + 1] = 0; } } else { if (pPrevAddHarmSfb[i] == 0 && pPrevAddHarmSfb[i + 1] == 1) { /* Keep the upper, remove the lower.*/ pAddHarmSfb[i] = 0; for (est = start; est < stop; est++) { detectionVectors[est][i] = 0; } } else { /* If the maximum values are in adjacent QMF-channels, and if the signs indicate that it is the same sine, we need to remove the lowest of the two.*/ if (maxVal1 > maxVal2) { if (signBuffer[maxPosTime1][maxPos2] < 0 && signBuffer[maxPosTime1][maxPos1] > 0) { /* Keep the lower, remove the upper.*/ pAddHarmSfb[i + 1] = 0; for (est = start; est < stop; est++) { detectionVectors[est][i + 1] = 0; } } } else { if (signBuffer[maxPosTime2][maxPos2] < 0 && signBuffer[maxPosTime2][maxPos1] > 0) { /* Keep the upper, remove the lower.*/ pAddHarmSfb[i] = 0; for (est = start; est < stop; est++) { detectionVectors[est][i] = 0; } } } } } } } } /* Make sure that the detection is not the cut-off of a low pass filter. */ removeLowPassDetection(pAddHarmSfb, detectionVectors, start, stop, nSfb, pFreqBandTable, pNrgVector, mhThresh); } else { /* * If a missing harmonic wasn't missing the previous frame * the transient-flag needs to be set in order to be allowed to detect it. *************************************************************************/ for (i = 0; i < nSfb; i++) { if (pAddHarmSfb[i] - pPrevAddHarmSfb[i] > 0) pAddHarmSfb[i] = 0; } } } /*****************************************************************************/ /*! \brief Detection for one tonality estimate. This is the actual missing harmonics detection, using information from the previous detection. If a missing harmonic was detected (in a previous frame) due to too high tonality differences, but there was not enough tonality difference in the current frame, the detection algorithm still continues to trace the strongest tone in the scalefactor band (assuming that this is the tone that is going to be replaced in the decoder). This is done to avoid abrupt endings of sines fading out (e.g. in the glockenspiel). The function also tries to estimate where one sine is going to be replaced with multiple sines (due to the patching). This is done by comparing the tonality flatness measure of the original and the SBR signal. The function also tries to estimate (for the scalefactor bands only containing one qmf subband) when a strong tone in the original will be replaced by a strong tone in the adjacent QMF subband. \return none. */ /**************************************************************************/ static void detection(FIXP_DBL *quotaBuffer, FIXP_DBL *pDiffVecScfb, INT nSfb, UCHAR *pHarmVec, const UCHAR *pFreqBandTable, FIXP_DBL *sfmOrig, FIXP_DBL *sfmSbr, GUIDE_VECTORS guideVectors, GUIDE_VECTORS newGuideVectors, THRES_HOLDS mhThresh) { INT i, j, ll, lu; FIXP_DBL thresTemp, thresOrig; /* * Do detection on the difference vector, i.e. the difference between * the original and the transposed. *********************************************************************/ for (i = 0; i < nSfb; i++) { thresTemp = (guideVectors.guideVectorDiff[i] != FL2FXCONST_DBL(0.0f)) ? fMax(fMult(mhThresh.decayGuideDiff, guideVectors.guideVectorDiff[i]), mhThresh.thresHoldDiffGuide) : mhThresh.thresHoldDiff; thresTemp = fMin(thresTemp, mhThresh.thresHoldDiff); if (pDiffVecScfb[i] > thresTemp) { pHarmVec[i] = 1; newGuideVectors.guideVectorDiff[i] = pDiffVecScfb[i]; } else { /* If the guide wasn't zero, but the current level is to low, start tracking the decay on the tone in the original rather than the difference.*/ if (guideVectors.guideVectorDiff[i] != FL2FXCONST_DBL(0.0f)) { guideVectors.guideVectorOrig[i] = mhThresh.thresHoldToneGuide; } } } /* * Trace tones in the original signal that at one point * have been detected because they will be replaced by * multiple tones in the sbr signal. ****************************************************/ for (i = 0; i < nSfb; i++) { ll = pFreqBandTable[i]; lu = pFreqBandTable[i + 1]; thresOrig = fixMax(fMult(guideVectors.guideVectorOrig[i], mhThresh.decayGuideOrig), mhThresh.thresHoldToneGuide); thresOrig = fixMin(thresOrig, mhThresh.thresHoldTone); if (guideVectors.guideVectorOrig[i] != FL2FXCONST_DBL(0.0f)) { for (j = ll; j < lu; j++) { if (quotaBuffer[j] > thresOrig) { pHarmVec[i] = 1; newGuideVectors.guideVectorOrig[i] = quotaBuffer[j]; } } } } /* * Check for multiple sines in the transposed signal, * where there is only one in the original. ****************************************************/ thresOrig = mhThresh.thresHoldTone; for (i = 0; i < nSfb; i++) { ll = pFreqBandTable[i]; lu = pFreqBandTable[i + 1]; if (pHarmVec[i] == 0) { if (lu - ll > 1) { for (j = ll; j < lu; j++) { if (quotaBuffer[j] > thresOrig && (sfmSbr[i] > mhThresh.sfmThresSbr && sfmOrig[i] < mhThresh.sfmThresOrig)) { pHarmVec[i] = 1; newGuideVectors.guideVectorOrig[i] = quotaBuffer[j]; } } } else { if (i < nSfb - 1) { ll = pFreqBandTable[i]; if (i > 0) { if (quotaBuffer[ll] > mhThresh.thresHoldTone && (pDiffVecScfb[i + 1] < mhThresh.invThresHoldTone || pDiffVecScfb[i - 1] < mhThresh.invThresHoldTone)) { pHarmVec[i] = 1; newGuideVectors.guideVectorOrig[i] = quotaBuffer[ll]; } } else { if (quotaBuffer[ll] > mhThresh.thresHoldTone && pDiffVecScfb[i + 1] < mhThresh.invThresHoldTone) { pHarmVec[i] = 1; newGuideVectors.guideVectorOrig[i] = quotaBuffer[ll]; } } } } } } } /**************************************************************************/ /*! \brief Do detection for every tonality estimate, using forward prediction. \return none. */ /**************************************************************************/ static void detectionWithPrediction( FIXP_DBL **quotaBuffer, FIXP_DBL **pDiffVecScfb, INT **signBuffer, INT nSfb, const UCHAR *pFreqBandTable, FIXP_DBL **sfmOrig, FIXP_DBL **sfmSbr, UCHAR **detectionVectors, UCHAR *pPrevAddHarmSfb, GUIDE_VECTORS *guideVectors, INT noEstPerFrame, INT detectionStart, INT totNoEst, INT newDetectionAllowed, INT *pAddHarmFlag, UCHAR *pAddHarmSfb, FIXP_DBL *pNrgVector, const DETECTOR_PARAMETERS_MH *mhParams) { INT est = 0, i; INT start; FDKmemclear(pAddHarmSfb, nSfb * sizeof(UCHAR)); if (newDetectionAllowed) { /* Since we don't want to use the transient region for detection (since the tonality values tend to be a bit unreliable for this region) the guide-values are copied to the current starting point. */ if (totNoEst > 1) { start = detectionStart + 1; if (start != 0) { FDKmemcpy(guideVectors[start].guideVectorDiff, guideVectors[0].guideVectorDiff, nSfb * sizeof(FIXP_DBL)); FDKmemcpy(guideVectors[start].guideVectorOrig, guideVectors[0].guideVectorOrig, nSfb * sizeof(FIXP_DBL)); FDKmemclear(guideVectors[start - 1].guideVectorDetected, nSfb * sizeof(UCHAR)); } } else { start = 0; } } else { start = 0; } for (est = start; est < totNoEst; est++) { /* * Do detection on the current frame using * guide-info from the previous. *******************************************/ if (est > 0) { FDKmemcpy(guideVectors[est].guideVectorDetected, detectionVectors[est - 1], nSfb * sizeof(UCHAR)); } FDKmemclear(detectionVectors[est], nSfb * sizeof(UCHAR)); if (est < totNoEst - 1) { FDKmemclear(guideVectors[est + 1].guideVectorDiff, nSfb * sizeof(FIXP_DBL)); FDKmemclear(guideVectors[est + 1].guideVectorOrig, nSfb * sizeof(FIXP_DBL)); FDKmemclear(guideVectors[est + 1].guideVectorDetected, nSfb * sizeof(UCHAR)); detection(quotaBuffer[est], pDiffVecScfb[est], nSfb, detectionVectors[est], pFreqBandTable, sfmOrig[est], sfmSbr[est], guideVectors[est], guideVectors[est + 1], mhParams->thresHolds); } else { FDKmemclear(guideVectors[est].guideVectorDiff, nSfb * sizeof(FIXP_DBL)); FDKmemclear(guideVectors[est].guideVectorOrig, nSfb * sizeof(FIXP_DBL)); FDKmemclear(guideVectors[est].guideVectorDetected, nSfb * sizeof(UCHAR)); detection(quotaBuffer[est], pDiffVecScfb[est], nSfb, detectionVectors[est], pFreqBandTable, sfmOrig[est], sfmSbr[est], guideVectors[est], guideVectors[est], mhParams->thresHolds); } } /* Clean up the detection.*/ transientCleanUp(quotaBuffer, nSfb, detectionVectors, pAddHarmSfb, pPrevAddHarmSfb, signBuffer, pFreqBandTable, start, totNoEst, newDetectionAllowed, pNrgVector, mhParams->thresHolds); /* Set flag... */ *pAddHarmFlag = 0; for (i = 0; i < nSfb; i++) { if (pAddHarmSfb[i]) { *pAddHarmFlag = 1; break; } } FDKmemcpy(pPrevAddHarmSfb, pAddHarmSfb, nSfb * sizeof(UCHAR)); FDKmemcpy(guideVectors[0].guideVectorDetected, pAddHarmSfb, nSfb * sizeof(INT)); for (i = 0; i < nSfb; i++) { guideVectors[0].guideVectorDiff[i] = FL2FXCONST_DBL(0.0f); guideVectors[0].guideVectorOrig[i] = FL2FXCONST_DBL(0.0f); if (pAddHarmSfb[i] == 1) { /* If we had a detection use the guide-value in the next frame from the last estimate were the detection was done.*/ for (est = start; est < totNoEst; est++) { if (guideVectors[est].guideVectorDiff[i] != FL2FXCONST_DBL(0.0f)) { guideVectors[0].guideVectorDiff[i] = guideVectors[est].guideVectorDiff[i]; } if (guideVectors[est].guideVectorOrig[i] != FL2FXCONST_DBL(0.0f)) { guideVectors[0].guideVectorOrig[i] = guideVectors[est].guideVectorOrig[i]; } } } } } /**************************************************************************/ /*! \brief Calculates a compensation vector for the energy data. This function calculates a compensation vector for the energy data (i.e. envelope data) that is calculated elsewhere. This is since, one sine on the border of two scalefactor bands, will be replace by one sine in the middle of either scalefactor band. However, since the sine that is replaced will influence the energy estimate in both scalefactor bands (in the envelops calculation function) a compensation value is required in order to avoid noise substitution in the decoder next to the synthetic sine. \return none. */ /**************************************************************************/ static void calculateCompVector(UCHAR *pAddHarmSfb, FIXP_DBL **pTonalityMatrix, INT **pSignMatrix, UCHAR *pEnvComp, INT nSfb, const UCHAR *freqBandTable, INT totNoEst, INT maxComp, UCHAR *pPrevEnvComp, INT newDetectionAllowed) { INT scfBand, est, l, ll, lu, maxPosF, maxPosT; FIXP_DBL maxVal; INT compValue; FIXP_DBL tmp; FDKmemclear(pEnvComp, nSfb * sizeof(UCHAR)); for (scfBand = 0; scfBand < nSfb; scfBand++) { if (pAddHarmSfb[scfBand]) { /* A missing sine was detected */ ll = freqBandTable[scfBand]; lu = freqBandTable[scfBand + 1]; maxPosF = 0; /* First find the maximum*/ maxPosT = 0; maxVal = FL2FXCONST_DBL(0.0f); for (est = 0; est < totNoEst; est++) { for (l = ll; l < lu; l++) { if (pTonalityMatrix[est][l] > maxVal) { maxVal = pTonalityMatrix[est][l]; maxPosF = l; maxPosT = est; } } } /* * If the maximum tonality is at the lower border of the * scalefactor band, we check the sign of the adjacent channels * to see if this sine is shared by the lower channel. If so, the * energy of the single sine will be present in two scalefactor bands * in the SBR data, which will cause problems in the decoder, when we * add a sine to just one of the channels. *********************************************************************/ if (maxPosF == ll && scfBand) { if (!pAddHarmSfb[scfBand - 1]) { /* No detection below*/ if (pSignMatrix[maxPosT][maxPosF - 1] > 0 && pSignMatrix[maxPosT][maxPosF] < 0) { /* The comp value is calulated as the tonallity value, i.e we want to reduce the envelope data for this channel with as much as the tonality that is spread from the channel above. (ld64(RELAXATION) = 0.31143075889) */ tmp = fixp_abs( (FIXP_DBL)CalcLdData(pTonalityMatrix[maxPosT][maxPosF - 1]) + RELAXATION_LD64); tmp = (tmp >> (DFRACT_BITS - 1 - LD_DATA_SHIFT - 1)) + (FIXP_DBL)1; /* shift one bit less for rounding */ compValue = ((INT)(LONG)tmp) >> 1; /* limit the comp-value*/ if (compValue > maxComp) compValue = maxComp; pEnvComp[scfBand - 1] = compValue; } } } /* * Same as above, but for the upper end of the scalefactor-band. ***************************************************************/ if (maxPosF == lu - 1 && scfBand + 1 < nSfb) { /* Upper border*/ if (!pAddHarmSfb[scfBand + 1]) { if (pSignMatrix[maxPosT][maxPosF] > 0 && pSignMatrix[maxPosT][maxPosF + 1] < 0) { tmp = fixp_abs( (FIXP_DBL)CalcLdData(pTonalityMatrix[maxPosT][maxPosF + 1]) + RELAXATION_LD64); tmp = (tmp >> (DFRACT_BITS - 1 - LD_DATA_SHIFT - 1)) + (FIXP_DBL)1; /* shift one bit less for rounding */ compValue = ((INT)(LONG)tmp) >> 1; if (compValue > maxComp) compValue = maxComp; pEnvComp[scfBand + 1] = compValue; } } } } } if (newDetectionAllowed == 0) { for (scfBand = 0; scfBand < nSfb; scfBand++) { if (pEnvComp[scfBand] != 0 && pPrevEnvComp[scfBand] == 0) pEnvComp[scfBand] = 0; } } /* remember the value for the next frame.*/ FDKmemcpy(pPrevEnvComp, pEnvComp, nSfb * sizeof(UCHAR)); } /**************************************************************************/ /*! \brief Detects where strong tonal components will be missing after HFR in the decoder. \return none. */ /**************************************************************************/ void FDKsbrEnc_SbrMissingHarmonicsDetectorQmf( HANDLE_SBR_MISSING_HARMONICS_DETECTOR h_sbrMHDet, FIXP_DBL **pQuotaBuffer, INT **pSignBuffer, SCHAR *indexVector, const SBR_FRAME_INFO *pFrameInfo, const UCHAR *pTranInfo, INT *pAddHarmonicsFlag, UCHAR *pAddHarmonicsScaleFactorBands, const UCHAR *freqBandTable, INT nSfb, UCHAR *envelopeCompensation, FIXP_DBL *pNrgVector) { INT transientFlag = pTranInfo[1]; INT transientPos = pTranInfo[0]; INT newDetectionAllowed; INT transientDetStart = 0; UCHAR **detectionVectors = h_sbrMHDet->detectionVectors; INT move = h_sbrMHDet->move; INT noEstPerFrame = h_sbrMHDet->noEstPerFrame; INT totNoEst = h_sbrMHDet->totNoEst; INT prevTransientFlag = h_sbrMHDet->previousTransientFlag; INT prevTransientFrame = h_sbrMHDet->previousTransientFrame; INT transientPosOffset = h_sbrMHDet->transientPosOffset; INT prevTransientPos = h_sbrMHDet->previousTransientPos; GUIDE_VECTORS *guideVectors = h_sbrMHDet->guideVectors; INT deltaTime = h_sbrMHDet->mhParams->deltaTime; INT maxComp = h_sbrMHDet->mhParams->maxComp; int est; /* Buffer values. */ FDK_ASSERT(move <= (MAX_NO_OF_ESTIMATES >> 1)); FDK_ASSERT(noEstPerFrame <= (MAX_NO_OF_ESTIMATES >> 1)); FIXP_DBL *sfmSbr[MAX_NO_OF_ESTIMATES]; FIXP_DBL *sfmOrig[MAX_NO_OF_ESTIMATES]; FIXP_DBL *tonalityDiff[MAX_NO_OF_ESTIMATES]; for (est = 0; est < MAX_NO_OF_ESTIMATES / 2; est++) { sfmSbr[est] = h_sbrMHDet->sfmSbr[est]; sfmOrig[est] = h_sbrMHDet->sfmOrig[est]; tonalityDiff[est] = h_sbrMHDet->tonalityDiff[est]; } C_ALLOC_SCRATCH_START(_scratch, FIXP_DBL, 3 * MAX_NO_OF_ESTIMATES / 2 * MAX_FREQ_COEFFS) FIXP_DBL *scratch = _scratch; for (; est < MAX_NO_OF_ESTIMATES; est++) { sfmSbr[est] = scratch; scratch += MAX_FREQ_COEFFS; sfmOrig[est] = scratch; scratch += MAX_FREQ_COEFFS; tonalityDiff[est] = scratch; scratch += MAX_FREQ_COEFFS; } /* Determine if we're allowed to detect "missing harmonics" that wasn't detected before. In order to be allowed to do new detection, there must be a transient in the current frame, or a transient in the previous frame sufficiently close to the current frame. */ newDetectionAllowed = isDetectionOfNewToneAllowed( pFrameInfo, &transientDetStart, noEstPerFrame, prevTransientFrame, prevTransientPos, prevTransientFlag, transientPosOffset, transientFlag, transientPos, deltaTime, h_sbrMHDet); /* Calulate the variables that will be used subsequently for the actual * detection */ calculateDetectorInput(pQuotaBuffer, indexVector, tonalityDiff, sfmOrig, sfmSbr, freqBandTable, nSfb, noEstPerFrame, move); /* Do the actual detection using information from previous detections */ detectionWithPrediction(pQuotaBuffer, tonalityDiff, pSignBuffer, nSfb, freqBandTable, sfmOrig, sfmSbr, detectionVectors, h_sbrMHDet->guideScfb, guideVectors, noEstPerFrame, transientDetStart, totNoEst, newDetectionAllowed, pAddHarmonicsFlag, pAddHarmonicsScaleFactorBands, pNrgVector, h_sbrMHDet->mhParams); /* Calculate the comp vector, so that the energy can be compensated for a sine between two QMF-bands. */ calculateCompVector(pAddHarmonicsScaleFactorBands, pQuotaBuffer, pSignBuffer, envelopeCompensation, nSfb, freqBandTable, totNoEst, maxComp, h_sbrMHDet->prevEnvelopeCompensation, newDetectionAllowed); for (est = 0; est < move; est++) { FDKmemcpy(tonalityDiff[est], tonalityDiff[est + noEstPerFrame], sizeof(FIXP_DBL) * MAX_FREQ_COEFFS); FDKmemcpy(sfmOrig[est], sfmOrig[est + noEstPerFrame], sizeof(FIXP_DBL) * MAX_FREQ_COEFFS); FDKmemcpy(sfmSbr[est], sfmSbr[est + noEstPerFrame], sizeof(FIXP_DBL) * MAX_FREQ_COEFFS); } C_ALLOC_SCRATCH_END(_scratch, FIXP_DBL, 3 * MAX_NO_OF_ESTIMATES / 2 * MAX_FREQ_COEFFS) } /**************************************************************************/ /*! \brief Initialize an instance of the missing harmonics detector. \return errorCode, noError if OK. */ /**************************************************************************/ INT FDKsbrEnc_CreateSbrMissingHarmonicsDetector( HANDLE_SBR_MISSING_HARMONICS_DETECTOR hSbrMHDet, INT chan) { HANDLE_SBR_MISSING_HARMONICS_DETECTOR hs = hSbrMHDet; INT i; UCHAR *detectionVectors = GetRam_Sbr_detectionVectors(chan); UCHAR *guideVectorDetected = GetRam_Sbr_guideVectorDetected(chan); FIXP_DBL *guideVectorDiff = GetRam_Sbr_guideVectorDiff(chan); FIXP_DBL *guideVectorOrig = GetRam_Sbr_guideVectorOrig(chan); FDKmemclear(hs, sizeof(SBR_MISSING_HARMONICS_DETECTOR)); hs->prevEnvelopeCompensation = GetRam_Sbr_prevEnvelopeCompensation(chan); hs->guideScfb = GetRam_Sbr_guideScfb(chan); if ((NULL == detectionVectors) || (NULL == guideVectorDetected) || (NULL == guideVectorDiff) || (NULL == guideVectorOrig) || (NULL == hs->prevEnvelopeCompensation) || (NULL == hs->guideScfb)) { goto bail; } for (i = 0; i < MAX_NO_OF_ESTIMATES; i++) { hs->guideVectors[i].guideVectorDiff = guideVectorDiff + (i * MAX_FREQ_COEFFS); hs->guideVectors[i].guideVectorOrig = guideVectorOrig + (i * MAX_FREQ_COEFFS); hs->detectionVectors[i] = detectionVectors + (i * MAX_FREQ_COEFFS); hs->guideVectors[i].guideVectorDetected = guideVectorDetected + (i * MAX_FREQ_COEFFS); } return 0; bail: hs->guideVectors[0].guideVectorDiff = guideVectorDiff; hs->guideVectors[0].guideVectorOrig = guideVectorOrig; hs->detectionVectors[0] = detectionVectors; hs->guideVectors[0].guideVectorDetected = guideVectorDetected; FDKsbrEnc_DeleteSbrMissingHarmonicsDetector(hs); return -1; } /**************************************************************************/ /*! \brief Initialize an instance of the missing harmonics detector. \return errorCode, noError if OK. */ /**************************************************************************/ INT FDKsbrEnc_InitSbrMissingHarmonicsDetector( HANDLE_SBR_MISSING_HARMONICS_DETECTOR hSbrMHDet, INT sampleFreq, INT frameSize, INT nSfb, INT qmfNoChannels, INT totNoEst, INT move, INT noEstPerFrame, UINT sbrSyntaxFlags) { HANDLE_SBR_MISSING_HARMONICS_DETECTOR hs = hSbrMHDet; int i; FDK_ASSERT(totNoEst <= MAX_NO_OF_ESTIMATES); if (sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { switch (frameSize) { case 1024: case 512: hs->transientPosOffset = FRAME_MIDDLE_SLOT_512LD; hs->timeSlots = 16; break; case 960: case 480: hs->transientPosOffset = FRAME_MIDDLE_SLOT_512LD; hs->timeSlots = 15; break; default: return -1; } } else { switch (frameSize) { case 2048: case 1024: hs->transientPosOffset = FRAME_MIDDLE_SLOT_2048; hs->timeSlots = NUMBER_TIME_SLOTS_2048; break; case 1920: case 960: hs->transientPosOffset = FRAME_MIDDLE_SLOT_1920; hs->timeSlots = NUMBER_TIME_SLOTS_1920; break; default: return -1; } } if (sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { hs->mhParams = ¶msAacLd; } else hs->mhParams = ¶msAac; hs->qmfNoChannels = qmfNoChannels; hs->sampleFreq = sampleFreq; hs->nSfb = nSfb; hs->totNoEst = totNoEst; hs->move = move; hs->noEstPerFrame = noEstPerFrame; for (i = 0; i < totNoEst; i++) { FDKmemclear(hs->guideVectors[i].guideVectorDiff, sizeof(FIXP_DBL) * MAX_FREQ_COEFFS); FDKmemclear(hs->guideVectors[i].guideVectorOrig, sizeof(FIXP_DBL) * MAX_FREQ_COEFFS); FDKmemclear(hs->detectionVectors[i], sizeof(UCHAR) * MAX_FREQ_COEFFS); FDKmemclear(hs->guideVectors[i].guideVectorDetected, sizeof(UCHAR) * MAX_FREQ_COEFFS); } // for(i=0; itonalityDiff[i], sizeof(FIXP_DBL) * MAX_FREQ_COEFFS); FDKmemclear(hs->sfmOrig[i], sizeof(FIXP_DBL) * MAX_FREQ_COEFFS); FDKmemclear(hs->sfmSbr[i], sizeof(FIXP_DBL) * MAX_FREQ_COEFFS); } FDKmemclear(hs->prevEnvelopeCompensation, sizeof(UCHAR) * MAX_FREQ_COEFFS); FDKmemclear(hs->guideScfb, sizeof(UCHAR) * MAX_FREQ_COEFFS); hs->previousTransientFlag = 0; hs->previousTransientFrame = 0; hs->previousTransientPos = 0; return (0); } /**************************************************************************/ /*! \brief Deletes an instance of the missing harmonics detector. \return none. */ /**************************************************************************/ void FDKsbrEnc_DeleteSbrMissingHarmonicsDetector( HANDLE_SBR_MISSING_HARMONICS_DETECTOR hSbrMHDet) { if (hSbrMHDet) { HANDLE_SBR_MISSING_HARMONICS_DETECTOR hs = hSbrMHDet; FreeRam_Sbr_detectionVectors(&hs->detectionVectors[0]); FreeRam_Sbr_guideVectorDetected(&hs->guideVectors[0].guideVectorDetected); FreeRam_Sbr_guideVectorDiff(&hs->guideVectors[0].guideVectorDiff); FreeRam_Sbr_guideVectorOrig(&hs->guideVectors[0].guideVectorOrig); FreeRam_Sbr_prevEnvelopeCompensation(&hs->prevEnvelopeCompensation); FreeRam_Sbr_guideScfb(&hs->guideScfb); } } /**************************************************************************/ /*! \brief Resets an instance of the missing harmonics detector. \return error code, noError if OK. */ /**************************************************************************/ INT FDKsbrEnc_ResetSbrMissingHarmonicsDetector( HANDLE_SBR_MISSING_HARMONICS_DETECTOR hSbrMissingHarmonicsDetector, INT nSfb) { int i; FIXP_DBL tempGuide[MAX_FREQ_COEFFS]; UCHAR tempGuideInt[MAX_FREQ_COEFFS]; INT nSfbPrev; nSfbPrev = hSbrMissingHarmonicsDetector->nSfb; hSbrMissingHarmonicsDetector->nSfb = nSfb; FDKmemcpy(tempGuideInt, hSbrMissingHarmonicsDetector->guideScfb, nSfbPrev * sizeof(UCHAR)); if (nSfb > nSfbPrev) { for (i = 0; i < (nSfb - nSfbPrev); i++) { hSbrMissingHarmonicsDetector->guideScfb[i] = 0; } for (i = 0; i < nSfbPrev; i++) { hSbrMissingHarmonicsDetector->guideScfb[i + (nSfb - nSfbPrev)] = tempGuideInt[i]; } } else { for (i = 0; i < nSfb; i++) { hSbrMissingHarmonicsDetector->guideScfb[i] = tempGuideInt[i + (nSfbPrev - nSfb)]; } } FDKmemcpy(tempGuide, hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorDiff, nSfbPrev * sizeof(FIXP_DBL)); if (nSfb > nSfbPrev) { for (i = 0; i < (nSfb - nSfbPrev); i++) { hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorDiff[i] = FL2FXCONST_DBL(0.0f); } for (i = 0; i < nSfbPrev; i++) { hSbrMissingHarmonicsDetector->guideVectors[0] .guideVectorDiff[i + (nSfb - nSfbPrev)] = tempGuide[i]; } } else { for (i = 0; i < nSfb; i++) { hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorDiff[i] = tempGuide[i + (nSfbPrev - nSfb)]; } } FDKmemcpy(tempGuide, hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorOrig, nSfbPrev * sizeof(FIXP_DBL)); if (nSfb > nSfbPrev) { for (i = 0; i < (nSfb - nSfbPrev); i++) { hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorOrig[i] = FL2FXCONST_DBL(0.0f); } for (i = 0; i < nSfbPrev; i++) { hSbrMissingHarmonicsDetector->guideVectors[0] .guideVectorOrig[i + (nSfb - nSfbPrev)] = tempGuide[i]; } } else { for (i = 0; i < nSfb; i++) { hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorOrig[i] = tempGuide[i + (nSfbPrev - nSfb)]; } } FDKmemcpy(tempGuideInt, hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorDetected, nSfbPrev * sizeof(UCHAR)); if (nSfb > nSfbPrev) { for (i = 0; i < (nSfb - nSfbPrev); i++) { hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorDetected[i] = 0; } for (i = 0; i < nSfbPrev; i++) { hSbrMissingHarmonicsDetector->guideVectors[0] .guideVectorDetected[i + (nSfb - nSfbPrev)] = tempGuideInt[i]; } } else { for (i = 0; i < nSfb; i++) { hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorDetected[i] = tempGuideInt[i + (nSfbPrev - nSfb)]; } } FDKmemcpy(tempGuideInt, hSbrMissingHarmonicsDetector->prevEnvelopeCompensation, nSfbPrev * sizeof(UCHAR)); if (nSfb > nSfbPrev) { for (i = 0; i < (nSfb - nSfbPrev); i++) { hSbrMissingHarmonicsDetector->prevEnvelopeCompensation[i] = 0; } for (i = 0; i < nSfbPrev; i++) { hSbrMissingHarmonicsDetector ->prevEnvelopeCompensation[i + (nSfb - nSfbPrev)] = tempGuideInt[i]; } } else { for (i = 0; i < nSfb; i++) { hSbrMissingHarmonicsDetector->prevEnvelopeCompensation[i] = tempGuideInt[i + (nSfbPrev - nSfb)]; } } return 0; } fdk-aac-2.0.2/libSBRenc/src/mh_det.h000066400000000000000000000207331404225310700170070ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief missing harmonics detection header file $Revision: 92790 $ */ #ifndef MH_DET_H #define MH_DET_H #include "sbr_encoder.h" #include "fram_gen.h" typedef struct { FIXP_DBL thresHoldDiff; /*!< threshold for tonality difference */ FIXP_DBL thresHoldDiffGuide; /*!< threshold for tonality difference for the guide */ FIXP_DBL thresHoldTone; /*!< threshold for tonality for a sine */ FIXP_DBL invThresHoldTone; FIXP_DBL thresHoldToneGuide; /*!< threshold for tonality for a sine for the guide */ FIXP_DBL sfmThresSbr; /*!< tonality flatness measure threshold for the SBR signal.*/ FIXP_DBL sfmThresOrig; /*!< tonality flatness measure threshold for the original signal.*/ FIXP_DBL decayGuideOrig; /*!< decay value of the tonality value of the guide for the tone. */ FIXP_DBL decayGuideDiff; /*!< decay value of the tonality value of the guide for the tonality difference. */ FIXP_DBL derivThresMaxLD64; /*!< threshold for detecting LP character in a signal. */ FIXP_DBL derivThresBelowLD64; /*!< threshold for detecting LP character in a signal. */ FIXP_DBL derivThresAboveLD64; /*!< threshold for detecting LP character in a signal. */ } THRES_HOLDS; typedef struct { INT deltaTime; /*!< maximum allowed transient distance (from frame border in number of qmf subband sample) for a frame to be considered a transient frame.*/ THRES_HOLDS thresHolds; /*!< the thresholds used for detection. */ INT maxComp; /*!< maximum alllowed compensation factor for the envelope data. */ } DETECTOR_PARAMETERS_MH; typedef struct { FIXP_DBL *guideVectorDiff; FIXP_DBL *guideVectorOrig; UCHAR *guideVectorDetected; } GUIDE_VECTORS; typedef struct { INT qmfNoChannels; INT nSfb; INT sampleFreq; INT previousTransientFlag; INT previousTransientFrame; INT previousTransientPos; INT noVecPerFrame; INT transientPosOffset; INT move; INT totNoEst; INT noEstPerFrame; INT timeSlots; UCHAR *guideScfb; UCHAR *prevEnvelopeCompensation; UCHAR *detectionVectors[MAX_NO_OF_ESTIMATES]; FIXP_DBL tonalityDiff[MAX_NO_OF_ESTIMATES / 2][MAX_FREQ_COEFFS]; FIXP_DBL sfmOrig[MAX_NO_OF_ESTIMATES / 2][MAX_FREQ_COEFFS]; FIXP_DBL sfmSbr[MAX_NO_OF_ESTIMATES / 2][MAX_FREQ_COEFFS]; const DETECTOR_PARAMETERS_MH *mhParams; GUIDE_VECTORS guideVectors[MAX_NO_OF_ESTIMATES]; } SBR_MISSING_HARMONICS_DETECTOR; typedef SBR_MISSING_HARMONICS_DETECTOR *HANDLE_SBR_MISSING_HARMONICS_DETECTOR; void FDKsbrEnc_SbrMissingHarmonicsDetectorQmf( HANDLE_SBR_MISSING_HARMONICS_DETECTOR h_sbrMissingHarmonicsDetector, FIXP_DBL **pQuotaBuffer, INT **pSignBuffer, SCHAR *indexVector, const SBR_FRAME_INFO *pFrameInfo, const UCHAR *pTranInfo, INT *pAddHarmonicsFlag, UCHAR *pAddHarmonicsScaleFactorBands, const UCHAR *freqBandTable, INT nSfb, UCHAR *envelopeCompensation, FIXP_DBL *pNrgVector); INT FDKsbrEnc_CreateSbrMissingHarmonicsDetector( HANDLE_SBR_MISSING_HARMONICS_DETECTOR hSbrMHDet, INT chan); INT FDKsbrEnc_InitSbrMissingHarmonicsDetector( HANDLE_SBR_MISSING_HARMONICS_DETECTOR h_sbrMissingHarmonicsDetector, INT sampleFreq, INT frameSize, INT nSfb, INT qmfNoChannels, INT totNoEst, INT move, INT noEstPerFrame, UINT sbrSyntaxFlags); void FDKsbrEnc_DeleteSbrMissingHarmonicsDetector( HANDLE_SBR_MISSING_HARMONICS_DETECTOR h_sbrMissingHarmonicsDetector); INT FDKsbrEnc_ResetSbrMissingHarmonicsDetector( HANDLE_SBR_MISSING_HARMONICS_DETECTOR hSbrMissingHarmonicsDetector, INT nSfb); #endif fdk-aac-2.0.2/libSBRenc/src/nf_est.cpp000066400000000000000000000544111404225310700173600ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Description: *******************************************************************************/ #include "nf_est.h" #include "sbr_misc.h" #include "genericStds.h" /* smoothFilter[4] = {0.05857864376269f, 0.2f, 0.34142135623731f, 0.4f}; */ static const FIXP_DBL smoothFilter[4] = {0x077f813d, 0x19999995, 0x2bb3b1f5, 0x33333335}; /* static const INT smoothFilterLength = 4; */ static const FIXP_DBL QuantOffset = (INT)0xfc000000; /* ld64(0.25) */ #ifndef min #define min(a, b) (a < b ? a : b) #endif #ifndef max #define max(a, b) (a > b ? a : b) #endif #define NOISE_FLOOR_OFFSET_SCALING (4) /**************************************************************************/ /*! \brief The function applies smoothing to the noise levels. \return none */ /**************************************************************************/ static void smoothingOfNoiseLevels( FIXP_DBL *NoiseLevels, /*!< pointer to noise-floor levels.*/ INT nEnvelopes, /*!< Number of noise floor envelopes.*/ INT noNoiseBands, /*!< Number of noise bands for every noise floor envelope. */ FIXP_DBL prevNoiseLevels[NF_SMOOTHING_LENGTH] [MAX_NUM_NOISE_VALUES], /*!< Previous noise floor envelopes. */ const FIXP_DBL * pSmoothFilter, /*!< filter used for smoothing the noise floor levels. */ INT transientFlag) /*!< flag indicating if a transient is present*/ { INT i, band, env; FIXP_DBL accu; for (env = 0; env < nEnvelopes; env++) { if (transientFlag) { for (i = 0; i < NF_SMOOTHING_LENGTH; i++) { FDKmemcpy(prevNoiseLevels[i], NoiseLevels + env * noNoiseBands, noNoiseBands * sizeof(FIXP_DBL)); } } else { for (i = 1; i < NF_SMOOTHING_LENGTH; i++) { FDKmemcpy(prevNoiseLevels[i - 1], prevNoiseLevels[i], noNoiseBands * sizeof(FIXP_DBL)); } FDKmemcpy(prevNoiseLevels[NF_SMOOTHING_LENGTH - 1], NoiseLevels + env * noNoiseBands, noNoiseBands * sizeof(FIXP_DBL)); } for (band = 0; band < noNoiseBands; band++) { accu = FL2FXCONST_DBL(0.0f); for (i = 0; i < NF_SMOOTHING_LENGTH; i++) { accu += fMultDiv2(pSmoothFilter[i], prevNoiseLevels[i][band]); } FDK_ASSERT((band + env * noNoiseBands) < MAX_NUM_NOISE_VALUES); NoiseLevels[band + env * noNoiseBands] = accu << 1; } } } /**************************************************************************/ /*! \brief Does the noise floor level estiamtion. The noiseLevel samples are scaled by the factor 0.25 \return none */ /**************************************************************************/ static void qmfBasedNoiseFloorDetection( FIXP_DBL *noiseLevel, /*!< Pointer to vector to store the noise levels in.*/ FIXP_DBL **quotaMatrixOrig, /*!< Matrix holding the quota values of the original. */ SCHAR *indexVector, /*!< Index vector to obtain the patched data. */ INT startIndex, /*!< Start index. */ INT stopIndex, /*!< Stop index. */ INT startChannel, /*!< Start channel of the current noise floor band.*/ INT stopChannel, /*!< Stop channel of the current noise floor band. */ FIXP_DBL ana_max_level, /*!< Maximum level of the adaptive noise.*/ FIXP_DBL noiseFloorOffset, /*!< Noise floor offset. */ INT missingHarmonicFlag, /*!< Flag indicating if a strong tonal component is missing.*/ FIXP_DBL weightFac, /*!< Weightening factor for the difference between orig and sbr. */ INVF_MODE diffThres, /*!< Threshold value to control the inverse filtering decision.*/ INVF_MODE inverseFilteringLevel) /*!< Inverse filtering level of the current band.*/ { INT scale, l, k; FIXP_DBL meanOrig = FL2FXCONST_DBL(0.0f), meanSbr = FL2FXCONST_DBL(0.0f), diff; FIXP_DBL invIndex = GetInvInt(stopIndex - startIndex); FIXP_DBL invChannel = GetInvInt(stopChannel - startChannel); FIXP_DBL accu; /* Calculate the mean value, over the current time segment, for the original, the HFR and the difference, over all channels in the current frequency range. */ if (missingHarmonicFlag == 1) { for (l = startChannel; l < stopChannel; l++) { /* tonalityOrig */ accu = FL2FXCONST_DBL(0.0f); for (k = startIndex; k < stopIndex; k++) { accu += fMultDiv2(quotaMatrixOrig[k][l], invIndex); } meanOrig = fixMax(meanOrig, (accu << 1)); /* tonalitySbr */ accu = FL2FXCONST_DBL(0.0f); for (k = startIndex; k < stopIndex; k++) { accu += fMultDiv2(quotaMatrixOrig[k][indexVector[l]], invIndex); } meanSbr = fixMax(meanSbr, (accu << 1)); } } else { for (l = startChannel; l < stopChannel; l++) { /* tonalityOrig */ accu = FL2FXCONST_DBL(0.0f); for (k = startIndex; k < stopIndex; k++) { accu += fMultDiv2(quotaMatrixOrig[k][l], invIndex); } meanOrig += fMult((accu << 1), invChannel); /* tonalitySbr */ accu = FL2FXCONST_DBL(0.0f); for (k = startIndex; k < stopIndex; k++) { accu += fMultDiv2(quotaMatrixOrig[k][indexVector[l]], invIndex); } meanSbr += fMult((accu << 1), invChannel); } } /* Small fix to avoid noise during silent passages.*/ if (meanOrig <= FL2FXCONST_DBL(0.000976562f * RELAXATION_FLOAT) && meanSbr <= FL2FXCONST_DBL(0.000976562f * RELAXATION_FLOAT)) { meanOrig = FL2FXCONST_DBL(101.5936673f * RELAXATION_FLOAT); meanSbr = FL2FXCONST_DBL(101.5936673f * RELAXATION_FLOAT); } meanOrig = fixMax(meanOrig, RELAXATION); meanSbr = fixMax(meanSbr, RELAXATION); if (missingHarmonicFlag == 1 || inverseFilteringLevel == INVF_MID_LEVEL || inverseFilteringLevel == INVF_LOW_LEVEL || inverseFilteringLevel == INVF_OFF || inverseFilteringLevel <= diffThres) { diff = RELAXATION; } else { accu = fDivNorm(meanSbr, meanOrig, &scale); diff = fixMax(RELAXATION, fMult(RELAXATION_FRACT, fMult(weightFac, accu)) >> (RELAXATION_SHIFT - scale)); } /* * noise Level is now a positive value, i.e. * the more harmonic the signal is the higher noise level, * this makes no sense so we change the sign. *********************************************************/ accu = fDivNorm(diff, meanOrig, &scale); scale -= 2; if ((scale > 0) && (accu > ((FIXP_DBL)MAXVAL_DBL) >> scale)) { *noiseLevel = (FIXP_DBL)MAXVAL_DBL; } else { *noiseLevel = scaleValue(accu, scale); } /* * Add a noise floor offset to compensate for bias in the detector *****************************************************************/ if (!missingHarmonicFlag) { *noiseLevel = fixMin(fMult(*noiseLevel, noiseFloorOffset), (FIXP_DBL)MAXVAL_DBL >> NOISE_FLOOR_OFFSET_SCALING) << NOISE_FLOOR_OFFSET_SCALING; } /* * check to see that we don't exceed the maximum allowed level **************************************************************/ *noiseLevel = fixMin(*noiseLevel, ana_max_level); /* ana_max_level is scaled with factor 0.25 */ } /**************************************************************************/ /*! \brief Does the noise floor level estiamtion. The function calls the Noisefloor estimation function for the time segments decided based upon the transient information. The block is always divided into one or two segments. \return none */ /**************************************************************************/ void FDKsbrEnc_sbrNoiseFloorEstimateQmf( HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */ const SBR_FRAME_INFO *frame_info, /*!< Time frequency grid of the current frame. */ FIXP_DBL *noiseLevels, /*!< Pointer to vector to store the noise levels in.*/ FIXP_DBL **quotaMatrixOrig, /*!< Matrix holding the quota values of the original. */ SCHAR *indexVector, /*!< Index vector to obtain the patched data. */ INT missingHarmonicsFlag, /*!< Flag indicating if a strong tonal component will be missing. */ INT startIndex, /*!< Start index. */ UINT numberOfEstimatesPerFrame, /*!< The number of tonality estimates per frame. */ int transientFrame, /*!< A flag indicating if a transient is present. */ INVF_MODE *pInvFiltLevels, /*!< Pointer to the vector holding the inverse filtering levels. */ UINT sbrSyntaxFlags) { INT nNoiseEnvelopes, startPos[2], stopPos[2], env, band; INT noNoiseBands = h_sbrNoiseFloorEstimate->noNoiseBands; INT *freqBandTable = h_sbrNoiseFloorEstimate->freqBandTableQmf; nNoiseEnvelopes = frame_info->nNoiseEnvelopes; startPos[0] = startIndex; if (nNoiseEnvelopes == 1) { stopPos[0] = startIndex + min(numberOfEstimatesPerFrame, 2); } else { stopPos[0] = startIndex + 1; startPos[1] = startIndex + 1; stopPos[1] = startIndex + min(numberOfEstimatesPerFrame, 2); } /* * Estimate the noise floor. **************************************/ for (env = 0; env < nNoiseEnvelopes; env++) { for (band = 0; band < noNoiseBands; band++) { FDK_ASSERT((band + env * noNoiseBands) < MAX_NUM_NOISE_VALUES); qmfBasedNoiseFloorDetection( &noiseLevels[band + env * noNoiseBands], quotaMatrixOrig, indexVector, startPos[env], stopPos[env], freqBandTable[band], freqBandTable[band + 1], h_sbrNoiseFloorEstimate->ana_max_level, h_sbrNoiseFloorEstimate->noiseFloorOffset[band], missingHarmonicsFlag, h_sbrNoiseFloorEstimate->weightFac, h_sbrNoiseFloorEstimate->diffThres, pInvFiltLevels[band]); } } /* * Smoothing of the values. **************************/ smoothingOfNoiseLevels(noiseLevels, nNoiseEnvelopes, h_sbrNoiseFloorEstimate->noNoiseBands, h_sbrNoiseFloorEstimate->prevNoiseLevels, h_sbrNoiseFloorEstimate->smoothFilter, transientFrame); /* quantisation*/ for (env = 0; env < nNoiseEnvelopes; env++) { for (band = 0; band < noNoiseBands; band++) { FDK_ASSERT((band + env * noNoiseBands) < MAX_NUM_NOISE_VALUES); noiseLevels[band + env * noNoiseBands] = (FIXP_DBL)NOISE_FLOOR_OFFSET_64 - (FIXP_DBL)CalcLdData(noiseLevels[band + env * noNoiseBands] + (FIXP_DBL)1) + QuantOffset; } } } /**************************************************************************/ /*! \brief \return errorCode, noError if successful */ /**************************************************************************/ static INT downSampleLoRes(INT *v_result, /*!< */ INT num_result, /*!< */ const UCHAR *freqBandTableRef, /*!< */ INT num_Ref) /*!< */ { INT step; INT i, j; INT org_length, result_length; INT v_index[MAX_FREQ_COEFFS / 2]; /* init */ org_length = num_Ref; result_length = num_result; v_index[0] = 0; /* Always use left border */ i = 0; while (org_length > 0) /* Create downsample vector */ { i++; step = org_length / result_length; /* floor; */ org_length = org_length - step; result_length--; v_index[i] = v_index[i - 1] + step; } if (i != num_result) /* Should never happen */ return (1); /* error downsampling */ for (j = 0; j <= i; j++) /* Use downsample vector to index LoResolution vector. */ { v_result[j] = freqBandTableRef[v_index[j]]; } return (0); } /**************************************************************************/ /*! \brief Initialize an instance of the noise floor level estimation module. \return errorCode, noError if successful */ /**************************************************************************/ INT FDKsbrEnc_InitSbrNoiseFloorEstimate( HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */ INT ana_max_level, /*!< Maximum level of the adaptive noise. */ const UCHAR *freqBandTable, /*!< Frequency band table. */ INT nSfb, /*!< Number of frequency bands. */ INT noiseBands, /*!< Number of noise bands per octave. */ INT noiseFloorOffset, /*!< Noise floor offset. */ INT timeSlots, /*!< Number of time slots in a frame. */ UINT useSpeechConfig /*!< Flag: adapt tuning parameters according to speech */ ) { INT i, qexp, qtmp; FIXP_DBL tmp, exp; FDKmemclear(h_sbrNoiseFloorEstimate, sizeof(SBR_NOISE_FLOOR_ESTIMATE)); h_sbrNoiseFloorEstimate->smoothFilter = smoothFilter; if (useSpeechConfig) { h_sbrNoiseFloorEstimate->weightFac = (FIXP_DBL)MAXVAL_DBL; h_sbrNoiseFloorEstimate->diffThres = INVF_LOW_LEVEL; } else { h_sbrNoiseFloorEstimate->weightFac = FL2FXCONST_DBL(0.25f); h_sbrNoiseFloorEstimate->diffThres = INVF_MID_LEVEL; } h_sbrNoiseFloorEstimate->timeSlots = timeSlots; h_sbrNoiseFloorEstimate->noiseBands = noiseBands; /* h_sbrNoiseFloorEstimate->ana_max_level is scaled by 0.25 */ switch (ana_max_level) { case 6: h_sbrNoiseFloorEstimate->ana_max_level = (FIXP_DBL)MAXVAL_DBL; break; case 3: h_sbrNoiseFloorEstimate->ana_max_level = FL2FXCONST_DBL(0.5); break; case -3: h_sbrNoiseFloorEstimate->ana_max_level = FL2FXCONST_DBL(0.125); break; default: /* Should not enter here */ h_sbrNoiseFloorEstimate->ana_max_level = (FIXP_DBL)MAXVAL_DBL; break; } /* calculate number of noise bands and allocate */ if (FDKsbrEnc_resetSbrNoiseFloorEstimate(h_sbrNoiseFloorEstimate, freqBandTable, nSfb)) return (1); if (noiseFloorOffset == 0) { tmp = ((FIXP_DBL)MAXVAL_DBL) >> NOISE_FLOOR_OFFSET_SCALING; } else { /* noiseFloorOffset has to be smaller than 12, because the result of the calculation below must be smaller than 1: (2^(noiseFloorOffset/3))*2^4<1 */ FDK_ASSERT(noiseFloorOffset < 12); /* Assumes the noise floor offset in tuning table are in q31 */ /* Change the qformat here when non-zero values would be filled */ exp = fDivNorm((FIXP_DBL)noiseFloorOffset, 3, &qexp); tmp = fPow(2, DFRACT_BITS - 1, exp, qexp, &qtmp); tmp = scaleValue(tmp, qtmp - NOISE_FLOOR_OFFSET_SCALING); } for (i = 0; i < h_sbrNoiseFloorEstimate->noNoiseBands; i++) { h_sbrNoiseFloorEstimate->noiseFloorOffset[i] = tmp; } return (0); } /**************************************************************************/ /*! \brief Resets the current instance of the noise floor estiamtion module. \return errorCode, noError if successful */ /**************************************************************************/ INT FDKsbrEnc_resetSbrNoiseFloorEstimate( HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */ const UCHAR *freqBandTable, /*!< Frequency band table. */ INT nSfb /*!< Number of bands in the frequency band table. */ ) { INT k2, kx; /* * Calculate number of noise bands ***********************************/ k2 = freqBandTable[nSfb]; kx = freqBandTable[0]; if (h_sbrNoiseFloorEstimate->noiseBands == 0) { h_sbrNoiseFloorEstimate->noNoiseBands = 1; } else { /* * Calculate number of noise bands 1,2 or 3 bands/octave ********************************************************/ FIXP_DBL tmp, ratio, lg2; INT ratio_e, qlg2, nNoiseBands; ratio = fDivNorm(k2, kx, &ratio_e); lg2 = fLog2(ratio, ratio_e, &qlg2); tmp = fMult((FIXP_DBL)(h_sbrNoiseFloorEstimate->noiseBands << 24), lg2); tmp = scaleValue(tmp, qlg2 - 23); nNoiseBands = (INT)((tmp + (FIXP_DBL)1) >> 1); if (nNoiseBands > MAX_NUM_NOISE_COEFFS) { nNoiseBands = MAX_NUM_NOISE_COEFFS; } if (nNoiseBands == 0) { nNoiseBands = 1; } h_sbrNoiseFloorEstimate->noNoiseBands = nNoiseBands; } return (downSampleLoRes(h_sbrNoiseFloorEstimate->freqBandTableQmf, h_sbrNoiseFloorEstimate->noNoiseBands, freqBandTable, nSfb)); } /**************************************************************************/ /*! \brief Deletes the current instancce of the noise floor level estimation module. \return none */ /**************************************************************************/ void FDKsbrEnc_deleteSbrNoiseFloorEstimate( HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate) /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */ { if (h_sbrNoiseFloorEstimate) { /* nothing to do */ } } fdk-aac-2.0.2/libSBRenc/src/nf_est.h000066400000000000000000000205201404225310700170170ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Noise floor estimation structs and prototypes $Revision: 92790 $ */ #ifndef NF_EST_H #define NF_EST_H #include "sbr_encoder.h" #include "fram_gen.h" #define NF_SMOOTHING_LENGTH 4 /*!< Smoothing length of the noise floors. */ typedef struct { FIXP_DBL prevNoiseLevels[NF_SMOOTHING_LENGTH] [MAX_NUM_NOISE_VALUES]; /*!< The previous noise levels. */ FIXP_DBL noiseFloorOffset [MAX_NUM_NOISE_VALUES]; /*!< Noise floor offset, scaled with NOISE_FLOOR_OFFSET_SCALING */ const FIXP_DBL *smoothFilter; /*!< Smoothing filter to use. */ FIXP_DBL ana_max_level; /*!< Max level allowed. */ FIXP_DBL weightFac; /*!< Weightening factor for the difference between orig and sbr. */ INT freqBandTableQmf[MAX_NUM_NOISE_VALUES + 1]; /*!< Frequncy band table for the noise floor bands.*/ INT noNoiseBands; /*!< Number of noisebands. */ INT noiseBands; /*!< NoiseBands switch 4 bit.*/ INT timeSlots; /*!< Number of timeslots in a frame. */ INVF_MODE diffThres; /*!< Threshold value to control the inverse filtering decision */ } SBR_NOISE_FLOOR_ESTIMATE; typedef SBR_NOISE_FLOOR_ESTIMATE *HANDLE_SBR_NOISE_FLOOR_ESTIMATE; void FDKsbrEnc_sbrNoiseFloorEstimateQmf( HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */ const SBR_FRAME_INFO *frame_info, /*!< Time frequency grid of the current frame. */ FIXP_DBL *noiseLevels, /*!< Pointer to vector to store the noise levels in.*/ FIXP_DBL **quotaMatrixOrig, /*!< Matrix holding the quota values of the original. */ SCHAR *indexVector, /*!< Index vector to obtain the patched data. */ INT missingHarmonicsFlag, /*!< Flag indicating if a strong tonal component will be missing. */ INT startIndex, /*!< Start index. */ UINT numberOfEstimatesPerFrame, /*!< The number of tonality estimates per frame. */ INT transientFrame, /*!< A flag indicating if a transient is present. */ INVF_MODE *pInvFiltLevels, /*!< Pointer to the vector holding the inverse filtering levels. */ UINT sbrSyntaxFlags); INT FDKsbrEnc_InitSbrNoiseFloorEstimate( HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */ INT ana_max_level, /*!< Maximum level of the adaptive noise. */ const UCHAR *freqBandTable, /*!< Frequany band table. */ INT nSfb, /*!< Number of frequency bands. */ INT noiseBands, /*!< Number of noise bands per octave. */ INT noiseFloorOffset, /*!< Noise floor offset. */ INT timeSlots, /*!< Number of time slots in a frame. */ UINT useSpeechConfig /*!< Flag: adapt tuning parameters according to speech */ ); INT FDKsbrEnc_resetSbrNoiseFloorEstimate( HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */ const UCHAR *freqBandTable, /*!< Frequany band table. */ INT nSfb); /*!< Number of bands in the frequency band table. */ void FDKsbrEnc_deleteSbrNoiseFloorEstimate( HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate); /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */ #endif fdk-aac-2.0.2/libSBRenc/src/ps_bitenc.cpp000066400000000000000000000544201404225310700200500ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): N. Rettelbach Description: Parametric Stereo bitstream encoder *******************************************************************************/ #include "ps_bitenc.h" #include "ps_main.h" static inline UCHAR FDKsbrEnc_WriteBits_ps(HANDLE_FDK_BITSTREAM hBitStream, UINT value, const UINT numberOfBits) { /* hBitStream == NULL happens here intentionally */ if (hBitStream != NULL) { FDKwriteBits(hBitStream, value, numberOfBits); } return numberOfBits; } #define SI_SBR_EXTENSION_SIZE_BITS 4 #define SI_SBR_EXTENSION_ESC_COUNT_BITS 8 #define SI_SBR_EXTENSION_ID_BITS 2 #define EXTENSION_ID_PS_CODING 2 #define PS_EXT_ID_V0 0 static const INT iidDeltaCoarse_Offset = 14; static const INT iidDeltaCoarse_MaxVal = 28; static const INT iidDeltaFine_Offset = 30; static const INT iidDeltaFine_MaxVal = 60; /* PS Stereo Huffmantable: iidDeltaFreqCoarse */ static const UINT iidDeltaFreqCoarse_Length[] = { 17, 17, 17, 17, 16, 15, 13, 10, 9, 7, 6, 5, 4, 3, 1, 3, 4, 5, 6, 6, 8, 11, 13, 14, 14, 15, 17, 18, 18}; static const UINT iidDeltaFreqCoarse_Code[] = { 0x0001fffb, 0x0001fffc, 0x0001fffd, 0x0001fffa, 0x0000fffc, 0x00007ffc, 0x00001ffd, 0x000003fe, 0x000001fe, 0x0000007e, 0x0000003c, 0x0000001d, 0x0000000d, 0x00000005, 0000000000, 0x00000004, 0x0000000c, 0x0000001c, 0x0000003d, 0x0000003e, 0x000000fe, 0x000007fe, 0x00001ffc, 0x00003ffc, 0x00003ffd, 0x00007ffd, 0x0001fffe, 0x0003fffe, 0x0003ffff}; /* PS Stereo Huffmantable: iidDeltaFreqFine */ static const UINT iidDeltaFreqFine_Length[] = { 18, 18, 18, 18, 18, 18, 18, 18, 18, 17, 18, 17, 17, 16, 16, 15, 14, 14, 13, 12, 12, 11, 10, 10, 8, 7, 6, 5, 4, 3, 1, 3, 4, 5, 6, 7, 8, 9, 10, 11, 11, 12, 13, 14, 14, 15, 16, 16, 17, 17, 18, 17, 18, 18, 18, 18, 18, 18, 18, 18, 18}; static const UINT iidDeltaFreqFine_Code[] = { 0x0001feb4, 0x0001feb5, 0x0001fd76, 0x0001fd77, 0x0001fd74, 0x0001fd75, 0x0001fe8a, 0x0001fe8b, 0x0001fe88, 0x0000fe80, 0x0001feb6, 0x0000fe82, 0x0000feb8, 0x00007f42, 0x00007fae, 0x00003faf, 0x00001fd1, 0x00001fe9, 0x00000fe9, 0x000007ea, 0x000007fb, 0x000003fb, 0x000001fb, 0x000001ff, 0x0000007c, 0x0000003c, 0x0000001c, 0x0000000c, 0000000000, 0x00000001, 0x00000001, 0x00000002, 0x00000001, 0x0000000d, 0x0000001d, 0x0000003d, 0x0000007d, 0x000000fc, 0x000001fc, 0x000003fc, 0x000003f4, 0x000007eb, 0x00000fea, 0x00001fea, 0x00001fd6, 0x00003fd0, 0x00007faf, 0x00007f43, 0x0000feb9, 0x0000fe83, 0x0001feb7, 0x0000fe81, 0x0001fe89, 0x0001fe8e, 0x0001fe8f, 0x0001fe8c, 0x0001fe8d, 0x0001feb2, 0x0001feb3, 0x0001feb0, 0x0001feb1}; /* PS Stereo Huffmantable: iidDeltaTimeCoarse */ static const UINT iidDeltaTimeCoarse_Length[] = { 19, 19, 19, 20, 20, 20, 17, 15, 12, 10, 8, 6, 4, 2, 1, 3, 5, 7, 9, 11, 13, 14, 17, 19, 20, 20, 20, 20, 20}; static const UINT iidDeltaTimeCoarse_Code[] = { 0x0007fff9, 0x0007fffa, 0x0007fffb, 0x000ffff8, 0x000ffff9, 0x000ffffa, 0x0001fffd, 0x00007ffe, 0x00000ffe, 0x000003fe, 0x000000fe, 0x0000003e, 0x0000000e, 0x00000002, 0000000000, 0x00000006, 0x0000001e, 0x0000007e, 0x000001fe, 0x000007fe, 0x00001ffe, 0x00003ffe, 0x0001fffc, 0x0007fff8, 0x000ffffb, 0x000ffffc, 0x000ffffd, 0x000ffffe, 0x000fffff}; /* PS Stereo Huffmantable: iidDeltaTimeFine */ static const UINT iidDeltaTimeFine_Length[] = { 16, 16, 16, 16, 16, 16, 16, 16, 16, 15, 15, 15, 15, 15, 15, 14, 14, 13, 13, 13, 12, 12, 11, 10, 9, 9, 7, 6, 5, 3, 1, 2, 5, 6, 7, 8, 9, 10, 11, 11, 12, 12, 13, 13, 14, 14, 15, 15, 15, 15, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16}; static const UINT iidDeltaTimeFine_Code[] = { 0x00004ed4, 0x00004ed5, 0x00004ece, 0x00004ecf, 0x00004ecc, 0x00004ed6, 0x00004ed8, 0x00004f46, 0x00004f60, 0x00002718, 0x00002719, 0x00002764, 0x00002765, 0x0000276d, 0x000027b1, 0x000013b7, 0x000013d6, 0x000009c7, 0x000009e9, 0x000009ed, 0x000004ee, 0x000004f7, 0x00000278, 0x00000139, 0x0000009a, 0x0000009f, 0x00000020, 0x00000011, 0x0000000a, 0x00000003, 0x00000001, 0000000000, 0x0000000b, 0x00000012, 0x00000021, 0x0000004c, 0x0000009b, 0x0000013a, 0x00000279, 0x00000270, 0x000004ef, 0x000004e2, 0x000009ea, 0x000009d8, 0x000013d7, 0x000013d0, 0x000027b2, 0x000027a2, 0x0000271a, 0x0000271b, 0x00004f66, 0x00004f67, 0x00004f61, 0x00004f47, 0x00004ed9, 0x00004ed7, 0x00004ecd, 0x00004ed2, 0x00004ed3, 0x00004ed0, 0x00004ed1}; static const INT iccDelta_Offset = 7; static const INT iccDelta_MaxVal = 14; /* PS Stereo Huffmantable: iccDeltaFreq */ static const UINT iccDeltaFreq_Length[] = {14, 14, 12, 10, 7, 5, 3, 1, 2, 4, 6, 8, 9, 11, 13}; static const UINT iccDeltaFreq_Code[] = { 0x00003fff, 0x00003ffe, 0x00000ffe, 0x000003fe, 0x0000007e, 0x0000001e, 0x00000006, 0000000000, 0x00000002, 0x0000000e, 0x0000003e, 0x000000fe, 0x000001fe, 0x000007fe, 0x00001ffe}; /* PS Stereo Huffmantable: iccDeltaTime */ static const UINT iccDeltaTime_Length[] = {14, 13, 11, 9, 7, 5, 3, 1, 2, 4, 6, 8, 10, 12, 14}; static const UINT iccDeltaTime_Code[] = { 0x00003ffe, 0x00001ffe, 0x000007fe, 0x000001fe, 0x0000007e, 0x0000001e, 0x00000006, 0000000000, 0x00000002, 0x0000000e, 0x0000003e, 0x000000fe, 0x000003fe, 0x00000ffe, 0x00003fff}; static const INT ipdDelta_Offset = 0; static const INT ipdDelta_MaxVal = 7; /* PS Stereo Huffmantable: ipdDeltaFreq */ static const UINT ipdDeltaFreq_Length[] = {1, 3, 4, 4, 4, 4, 4, 4}; static const UINT ipdDeltaFreq_Code[] = {0x00000001, 0000000000, 0x00000006, 0x00000004, 0x00000002, 0x00000003, 0x00000005, 0x00000007}; /* PS Stereo Huffmantable: ipdDeltaTime */ static const UINT ipdDeltaTime_Length[] = {1, 3, 4, 5, 5, 4, 4, 3}; static const UINT ipdDeltaTime_Code[] = {0x00000001, 0x00000002, 0x00000002, 0x00000003, 0x00000002, 0000000000, 0x00000003, 0x00000003}; static const INT opdDelta_Offset = 0; static const INT opdDelta_MaxVal = 7; /* PS Stereo Huffmantable: opdDeltaFreq */ static const UINT opdDeltaFreq_Length[] = {1, 3, 4, 4, 5, 5, 4, 3}; static const UINT opdDeltaFreq_Code[] = { 0x00000001, 0x00000001, 0x00000006, 0x00000004, 0x0000000f, 0x0000000e, 0x00000005, 0000000000, }; /* PS Stereo Huffmantable: opdDeltaTime */ static const UINT opdDeltaTime_Length[] = {1, 3, 4, 5, 5, 4, 4, 3}; static const UINT opdDeltaTime_Code[] = {0x00000001, 0x00000002, 0x00000001, 0x00000007, 0x00000006, 0000000000, 0x00000002, 0x00000003}; static INT getNoBands(const INT mode) { INT noBands = 0; switch (mode) { case 0: case 3: /* coarse */ noBands = PS_BANDS_COARSE; break; case 1: case 4: /* mid */ noBands = PS_BANDS_MID; break; case 2: case 5: /* fine not supported */ default: /* coarse as default */ noBands = PS_BANDS_COARSE; } return noBands; } static INT getIIDRes(INT iidMode) { if (iidMode < 3) return PS_IID_RES_COARSE; else return PS_IID_RES_FINE; } static INT encodeDeltaFreq(HANDLE_FDK_BITSTREAM hBitBuf, const INT *val, const INT nBands, const UINT *codeTable, const UINT *lengthTable, const INT tableOffset, const INT maxVal, INT *error) { INT bitCnt = 0; INT lastVal = 0; INT band; for (band = 0; band < nBands; band++) { INT delta = (val[band] - lastVal) + tableOffset; lastVal = val[band]; if ((delta > maxVal) || (delta < 0)) { *error = 1; delta = delta > 0 ? maxVal : 0; } bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, codeTable[delta], lengthTable[delta]); } return bitCnt; } static INT encodeDeltaTime(HANDLE_FDK_BITSTREAM hBitBuf, const INT *val, const INT *valLast, const INT nBands, const UINT *codeTable, const UINT *lengthTable, const INT tableOffset, const INT maxVal, INT *error) { INT bitCnt = 0; INT band; for (band = 0; band < nBands; band++) { INT delta = (val[band] - valLast[band]) + tableOffset; if ((delta > maxVal) || (delta < 0)) { *error = 1; delta = delta > 0 ? maxVal : 0; } bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, codeTable[delta], lengthTable[delta]); } return bitCnt; } INT FDKsbrEnc_EncodeIid(HANDLE_FDK_BITSTREAM hBitBuf, const INT *iidVal, const INT *iidValLast, const INT nBands, const PS_IID_RESOLUTION res, const PS_DELTA mode, INT *error) { const UINT *codeTable; const UINT *lengthTable; INT bitCnt = 0; bitCnt = 0; switch (mode) { case PS_DELTA_FREQ: switch (res) { case PS_IID_RES_COARSE: codeTable = iidDeltaFreqCoarse_Code; lengthTable = iidDeltaFreqCoarse_Length; bitCnt += encodeDeltaFreq(hBitBuf, iidVal, nBands, codeTable, lengthTable, iidDeltaCoarse_Offset, iidDeltaCoarse_MaxVal, error); break; case PS_IID_RES_FINE: codeTable = iidDeltaFreqFine_Code; lengthTable = iidDeltaFreqFine_Length; bitCnt += encodeDeltaFreq(hBitBuf, iidVal, nBands, codeTable, lengthTable, iidDeltaFine_Offset, iidDeltaFine_MaxVal, error); break; default: *error = 1; } break; case PS_DELTA_TIME: switch (res) { case PS_IID_RES_COARSE: codeTable = iidDeltaTimeCoarse_Code; lengthTable = iidDeltaTimeCoarse_Length; bitCnt += encodeDeltaTime( hBitBuf, iidVal, iidValLast, nBands, codeTable, lengthTable, iidDeltaCoarse_Offset, iidDeltaCoarse_MaxVal, error); break; case PS_IID_RES_FINE: codeTable = iidDeltaTimeFine_Code; lengthTable = iidDeltaTimeFine_Length; bitCnt += encodeDeltaTime(hBitBuf, iidVal, iidValLast, nBands, codeTable, lengthTable, iidDeltaFine_Offset, iidDeltaFine_MaxVal, error); break; default: *error = 1; } break; default: *error = 1; } return bitCnt; } INT FDKsbrEnc_EncodeIcc(HANDLE_FDK_BITSTREAM hBitBuf, const INT *iccVal, const INT *iccValLast, const INT nBands, const PS_DELTA mode, INT *error) { const UINT *codeTable; const UINT *lengthTable; INT bitCnt = 0; switch (mode) { case PS_DELTA_FREQ: codeTable = iccDeltaFreq_Code; lengthTable = iccDeltaFreq_Length; bitCnt += encodeDeltaFreq(hBitBuf, iccVal, nBands, codeTable, lengthTable, iccDelta_Offset, iccDelta_MaxVal, error); break; case PS_DELTA_TIME: codeTable = iccDeltaTime_Code; lengthTable = iccDeltaTime_Length; bitCnt += encodeDeltaTime(hBitBuf, iccVal, iccValLast, nBands, codeTable, lengthTable, iccDelta_Offset, iccDelta_MaxVal, error); break; default: *error = 1; } return bitCnt; } INT FDKsbrEnc_EncodeIpd(HANDLE_FDK_BITSTREAM hBitBuf, const INT *ipdVal, const INT *ipdValLast, const INT nBands, const PS_DELTA mode, INT *error) { const UINT *codeTable; const UINT *lengthTable; INT bitCnt = 0; switch (mode) { case PS_DELTA_FREQ: codeTable = ipdDeltaFreq_Code; lengthTable = ipdDeltaFreq_Length; bitCnt += encodeDeltaFreq(hBitBuf, ipdVal, nBands, codeTable, lengthTable, ipdDelta_Offset, ipdDelta_MaxVal, error); break; case PS_DELTA_TIME: codeTable = ipdDeltaTime_Code; lengthTable = ipdDeltaTime_Length; bitCnt += encodeDeltaTime(hBitBuf, ipdVal, ipdValLast, nBands, codeTable, lengthTable, ipdDelta_Offset, ipdDelta_MaxVal, error); break; default: *error = 1; } return bitCnt; } INT FDKsbrEnc_EncodeOpd(HANDLE_FDK_BITSTREAM hBitBuf, const INT *opdVal, const INT *opdValLast, const INT nBands, const PS_DELTA mode, INT *error) { const UINT *codeTable; const UINT *lengthTable; INT bitCnt = 0; switch (mode) { case PS_DELTA_FREQ: codeTable = opdDeltaFreq_Code; lengthTable = opdDeltaFreq_Length; bitCnt += encodeDeltaFreq(hBitBuf, opdVal, nBands, codeTable, lengthTable, opdDelta_Offset, opdDelta_MaxVal, error); break; case PS_DELTA_TIME: codeTable = opdDeltaTime_Code; lengthTable = opdDeltaTime_Length; bitCnt += encodeDeltaTime(hBitBuf, opdVal, opdValLast, nBands, codeTable, lengthTable, opdDelta_Offset, opdDelta_MaxVal, error); break; default: *error = 1; } return bitCnt; } static INT encodeIpdOpd(HANDLE_PS_OUT psOut, HANDLE_FDK_BITSTREAM hBitBuf) { INT bitCnt = 0; INT error = 0; INT env; FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->enableIpdOpd, 1); if (psOut->enableIpdOpd == 1) { INT *ipdLast = psOut->ipdLast; INT *opdLast = psOut->opdLast; for (env = 0; env < psOut->nEnvelopes; env++) { bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->deltaIPD[env], 1); bitCnt += FDKsbrEnc_EncodeIpd(hBitBuf, psOut->ipd[env], ipdLast, getNoBands(psOut->iidMode), psOut->deltaIPD[env], &error); bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->deltaOPD[env], 1); bitCnt += FDKsbrEnc_EncodeOpd(hBitBuf, psOut->opd[env], opdLast, getNoBands(psOut->iidMode), psOut->deltaOPD[env], &error); } /* reserved bit */ bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, 0, 1); } return bitCnt; } static INT getEnvIdx(const INT nEnvelopes, const INT frameClass) { INT envIdx = 0; switch (nEnvelopes) { case 0: envIdx = 0; break; case 1: if (frameClass == 0) envIdx = 1; else envIdx = 0; break; case 2: if (frameClass == 0) envIdx = 2; else envIdx = 1; break; case 3: envIdx = 2; break; case 4: envIdx = 3; break; default: /* unsupported number of envelopes */ envIdx = 0; } return envIdx; } static INT encodePSExtension(const HANDLE_PS_OUT psOut, HANDLE_FDK_BITSTREAM hBitBuf) { INT bitCnt = 0; if (psOut->enableIpdOpd == 1) { INT ipdOpdBits = 0; INT extSize = (2 + encodeIpdOpd(psOut, NULL) + 7) >> 3; if (extSize < 15) { bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, extSize, 4); } else { bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, 15, 4); bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, (extSize - 15), 8); } /* write ipd opd data */ ipdOpdBits += FDKsbrEnc_WriteBits_ps(hBitBuf, PS_EXT_ID_V0, 2); ipdOpdBits += encodeIpdOpd(psOut, hBitBuf); /* byte align the ipd opd data */ if (ipdOpdBits % 8) ipdOpdBits += FDKsbrEnc_WriteBits_ps(hBitBuf, 0, (8 - (ipdOpdBits % 8))); bitCnt += ipdOpdBits; } return (bitCnt); } INT FDKsbrEnc_WritePSBitstream(const HANDLE_PS_OUT psOut, HANDLE_FDK_BITSTREAM hBitBuf) { INT psExtEnable = 0; INT bitCnt = 0; INT error = 0; INT env; if (psOut != NULL) { /* PS HEADER */ bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->enablePSHeader, 1); if (psOut->enablePSHeader) { bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->enableIID, 1); if (psOut->enableIID) { bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->iidMode, 3); } bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->enableICC, 1); if (psOut->enableICC) { bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->iccMode, 3); } if (psOut->enableIpdOpd) { psExtEnable = 1; } bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psExtEnable, 1); } /* Frame class, number of envelopes */ bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->frameClass, 1); bitCnt += FDKsbrEnc_WriteBits_ps( hBitBuf, getEnvIdx(psOut->nEnvelopes, psOut->frameClass), 2); if (psOut->frameClass == 1) { for (env = 0; env < psOut->nEnvelopes; env++) { bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->frameBorder[env], 5); } } if (psOut->enableIID == 1) { INT *iidLast = psOut->iidLast; for (env = 0; env < psOut->nEnvelopes; env++) { bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->deltaIID[env], 1); bitCnt += FDKsbrEnc_EncodeIid( hBitBuf, psOut->iid[env], iidLast, getNoBands(psOut->iidMode), (PS_IID_RESOLUTION)getIIDRes(psOut->iidMode), psOut->deltaIID[env], &error); iidLast = psOut->iid[env]; } } if (psOut->enableICC == 1) { INT *iccLast = psOut->iccLast; for (env = 0; env < psOut->nEnvelopes; env++) { bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->deltaICC[env], 1); bitCnt += FDKsbrEnc_EncodeIcc(hBitBuf, psOut->icc[env], iccLast, getNoBands(psOut->iccMode), psOut->deltaICC[env], &error); iccLast = psOut->icc[env]; } } if (psExtEnable != 0) { bitCnt += encodePSExtension(psOut, hBitBuf); } } /* if(psOut != NULL) */ return bitCnt; } fdk-aac-2.0.2/libSBRenc/src/ps_bitenc.h000066400000000000000000000152721404225310700175170ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): N. Rettelbach Description: Parametric Stereo bitstream encoder *******************************************************************************/ #include "ps_main.h" #include "ps_const.h" #include "FDK_bitstream.h" #ifndef PS_BITENC_H #define PS_BITENC_H typedef struct T_PS_OUT { INT enablePSHeader; INT enableIID; INT iidMode; INT enableICC; INT iccMode; INT enableIpdOpd; INT frameClass; INT nEnvelopes; /* ENV data */ INT frameBorder[PS_MAX_ENVELOPES]; /* iid data */ PS_DELTA deltaIID[PS_MAX_ENVELOPES]; INT iid[PS_MAX_ENVELOPES][PS_MAX_BANDS]; INT iidLast[PS_MAX_BANDS]; /* icc data */ PS_DELTA deltaICC[PS_MAX_ENVELOPES]; INT icc[PS_MAX_ENVELOPES][PS_MAX_BANDS]; INT iccLast[PS_MAX_BANDS]; /* ipd data */ PS_DELTA deltaIPD[PS_MAX_ENVELOPES]; INT ipd[PS_MAX_ENVELOPES][PS_MAX_BANDS]; INT ipdLast[PS_MAX_BANDS]; /* opd data */ PS_DELTA deltaOPD[PS_MAX_ENVELOPES]; INT opd[PS_MAX_ENVELOPES][PS_MAX_BANDS]; INT opdLast[PS_MAX_BANDS]; } PS_OUT, *HANDLE_PS_OUT; #ifdef __cplusplus extern "C" { #endif /* __cplusplus */ INT FDKsbrEnc_EncodeIid(HANDLE_FDK_BITSTREAM hBitBuf, const INT *iidVal, const INT *iidValLast, const INT nBands, const PS_IID_RESOLUTION res, const PS_DELTA mode, INT *error); INT FDKsbrEnc_EncodeIcc(HANDLE_FDK_BITSTREAM hBitBuf, const INT *iccVal, const INT *iccValLast, const INT nBands, const PS_DELTA mode, INT *error); INT FDKsbrEnc_EncodeIpd(HANDLE_FDK_BITSTREAM hBitBuf, const INT *ipdVal, const INT *ipdValLast, const INT nBands, const PS_DELTA mode, INT *error); INT FDKsbrEnc_EncodeOpd(HANDLE_FDK_BITSTREAM hBitBuf, const INT *opdVal, const INT *opdValLast, const INT nBands, const PS_DELTA mode, INT *error); INT FDKsbrEnc_WritePSBitstream(const HANDLE_PS_OUT psOut, HANDLE_FDK_BITSTREAM hBitBuf); #ifdef __cplusplus } #endif /* __cplusplus */ #endif /* defined(PSENC_ENABLE) */ fdk-aac-2.0.2/libSBRenc/src/ps_const.h000066400000000000000000000136771404225310700174100ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): N. Rettelbach Description: Parametric Stereo constants *******************************************************************************/ #ifndef PS_CONST_H #define PS_CONST_H #define MAX_PS_CHANNELS (2) #define HYBRID_MAX_QMF_BANDS (3) #define HYBRID_FILTER_LENGTH (13) #define HYBRID_FILTER_DELAY ((HYBRID_FILTER_LENGTH - 1) / 2) #define HYBRID_FRAMESIZE (32) #define HYBRID_READ_OFFSET (10) #define MAX_HYBRID_BANDS ((64 - HYBRID_MAX_QMF_BANDS + 10)) typedef enum { PS_RES_COARSE = 0, PS_RES_MID = 1, PS_RES_FINE = 2 } PS_RESOLUTION; typedef enum { PS_BANDS_COARSE = 10, PS_BANDS_MID = 20, PS_MAX_BANDS = PS_BANDS_MID } PS_BANDS; typedef enum { PS_IID_RES_COARSE = 0, PS_IID_RES_FINE } PS_IID_RESOLUTION; typedef enum { PS_ICC_ROT_A = 0, PS_ICC_ROT_B } PS_ICC_ROTATION_MODE; typedef enum { PS_DELTA_FREQ, PS_DELTA_TIME } PS_DELTA; typedef enum { PS_MAX_ENVELOPES = 4 } PS_CONSTS; typedef enum { PSENC_OK = 0x0000, /*!< No error happened. All fine. */ PSENC_INVALID_HANDLE = 0x0020, /*!< Handle passed to function call was invalid. */ PSENC_MEMORY_ERROR = 0x0021, /*!< Memory allocation failed. */ PSENC_INIT_ERROR = 0x0040, /*!< General initialization error. */ PSENC_ENCODE_ERROR = 0x0060 /*!< The encoding process was interrupted by an unexpected error. */ } FDK_PSENC_ERROR; #endif fdk-aac-2.0.2/libSBRenc/src/ps_encode.cpp000066400000000000000000001074101404225310700200370ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): M. Neuendorf, N. Rettelbach, M. Multrus Description: PS parameter extraction, encoding *******************************************************************************/ /*! \file \brief PS parameter extraction, encoding functions $Revision: 96441 $ */ #include "ps_main.h" #include "ps_encode.h" #include "qmf.h" #include "sbr_misc.h" #include "sbrenc_ram.h" #include "genericStds.h" inline void FDKsbrEnc_addFIXP_DBL(const FIXP_DBL *X, const FIXP_DBL *Y, FIXP_DBL *Z, INT n) { for (INT i = 0; i < n; i++) Z[i] = (X[i] >> 1) + (Y[i] >> 1); } #define LOG10_2_10 3.01029995664f /* 10.0f*log10(2.f) */ static const INT iidGroupBordersLoRes[QMF_GROUPS_LO_RES + SUBQMF_GROUPS_LO_RES + 1] = { 0, 1, 2, 3, 4, 5, /* 6 subqmf subbands - 0th qmf subband */ 6, 7, /* 2 subqmf subbands - 1st qmf subband */ 8, 9, /* 2 subqmf subbands - 2nd qmf subband */ 10, 11, 12, 13, 14, 15, 16, 18, 21, 25, 30, 42, 71}; static const UCHAR iidGroupWidthLdLoRes[QMF_GROUPS_LO_RES + SUBQMF_GROUPS_LO_RES] = { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 1, 2, 2, 3, 4, 5}; static const INT subband2parameter20[QMF_GROUPS_LO_RES + SUBQMF_GROUPS_LO_RES] = {1, 0, 0, 1, 2, 3, /* 6 subqmf subbands - 0th qmf subband */ 4, 5, /* 2 subqmf subbands - 1st qmf subband */ 6, 7, /* 2 subqmf subbands - 2nd qmf subband */ 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19}; typedef enum { MAX_TIME_DIFF_FRAMES = 20, MAX_PS_NOHEADER_CNT = 10, MAX_NOENV_CNT = 10, DO_NOT_USE_THIS_MODE = 0x7FFFFF } __PS_CONSTANTS; static const FIXP_DBL iidQuant_fx[15] = { (FIXP_DBL)0xce000000, (FIXP_DBL)0xdc000000, (FIXP_DBL)0xe4000000, (FIXP_DBL)0xec000000, (FIXP_DBL)0xf2000000, (FIXP_DBL)0xf8000000, (FIXP_DBL)0xfc000000, (FIXP_DBL)0x00000000, (FIXP_DBL)0x04000000, (FIXP_DBL)0x08000000, (FIXP_DBL)0x0e000000, (FIXP_DBL)0x14000000, (FIXP_DBL)0x1c000000, (FIXP_DBL)0x24000000, (FIXP_DBL)0x32000000}; static const FIXP_DBL iidQuantFine_fx[31] = { (FIXP_DBL)0x9c000001, (FIXP_DBL)0xa6000001, (FIXP_DBL)0xb0000001, (FIXP_DBL)0xba000001, (FIXP_DBL)0xc4000000, (FIXP_DBL)0xce000000, (FIXP_DBL)0xd4000000, (FIXP_DBL)0xda000000, (FIXP_DBL)0xe0000000, (FIXP_DBL)0xe6000000, (FIXP_DBL)0xec000000, (FIXP_DBL)0xf0000000, (FIXP_DBL)0xf4000000, (FIXP_DBL)0xf8000000, (FIXP_DBL)0xfc000000, (FIXP_DBL)0x00000000, (FIXP_DBL)0x04000000, (FIXP_DBL)0x08000000, (FIXP_DBL)0x0c000000, (FIXP_DBL)0x10000000, (FIXP_DBL)0x14000000, (FIXP_DBL)0x1a000000, (FIXP_DBL)0x20000000, (FIXP_DBL)0x26000000, (FIXP_DBL)0x2c000000, (FIXP_DBL)0x32000000, (FIXP_DBL)0x3c000000, (FIXP_DBL)0x45ffffff, (FIXP_DBL)0x4fffffff, (FIXP_DBL)0x59ffffff, (FIXP_DBL)0x63ffffff}; static const FIXP_DBL iccQuant[8] = { (FIXP_DBL)0x7fffffff, (FIXP_DBL)0x77ef9d7f, (FIXP_DBL)0x6babc97f, (FIXP_DBL)0x4ceaf27f, (FIXP_DBL)0x2f0ed3c0, (FIXP_DBL)0x00000000, (FIXP_DBL)0xb49ba601, (FIXP_DBL)0x80000000}; static FDK_PSENC_ERROR InitPSData(HANDLE_PS_DATA hPsData) { FDK_PSENC_ERROR error = PSENC_OK; if (hPsData == NULL) { error = PSENC_INVALID_HANDLE; } else { int i, env; FDKmemclear(hPsData, sizeof(PS_DATA)); for (i = 0; i < PS_MAX_BANDS; i++) { hPsData->iidIdxLast[i] = 0; hPsData->iccIdxLast[i] = 0; } hPsData->iidEnable = hPsData->iidEnableLast = 0; hPsData->iccEnable = hPsData->iccEnableLast = 0; hPsData->iidQuantMode = hPsData->iidQuantModeLast = PS_IID_RES_COARSE; hPsData->iccQuantMode = hPsData->iccQuantModeLast = PS_ICC_ROT_A; for (env = 0; env < PS_MAX_ENVELOPES; env++) { hPsData->iccDiffMode[env] = PS_DELTA_FREQ; hPsData->iccDiffMode[env] = PS_DELTA_FREQ; for (i = 0; i < PS_MAX_BANDS; i++) { hPsData->iidIdx[env][i] = 0; hPsData->iccIdx[env][i] = 0; } } hPsData->nEnvelopesLast = 0; hPsData->headerCnt = MAX_PS_NOHEADER_CNT; hPsData->iidTimeCnt = MAX_TIME_DIFF_FRAMES; hPsData->iccTimeCnt = MAX_TIME_DIFF_FRAMES; hPsData->noEnvCnt = MAX_NOENV_CNT; } return error; } static FIXP_DBL quantizeCoef(const FIXP_DBL *RESTRICT input, const INT nBands, const FIXP_DBL *RESTRICT quantTable, const INT idxOffset, const INT nQuantSteps, INT *RESTRICT quantOut) { INT idx, band; FIXP_DBL quantErr = FL2FXCONST_DBL(0.f); for (band = 0; band < nBands; band++) { for (idx = 0; idx < nQuantSteps - 1; idx++) { if (fixp_abs((input[band] >> 1) - (quantTable[idx + 1] >> 1)) > fixp_abs((input[band] >> 1) - (quantTable[idx] >> 1))) { break; } } quantErr += (fixp_abs(input[band] - quantTable[idx]) >> PS_QUANT_SCALE); /* don't scale before subtraction; diff smaller (64-25)/64 */ quantOut[band] = idx - idxOffset; } return quantErr; } static INT getICCMode(const INT nBands, const INT rotType) { INT mode = 0; switch (nBands) { case PS_BANDS_COARSE: mode = PS_RES_COARSE; break; case PS_BANDS_MID: mode = PS_RES_MID; break; default: mode = 0; } if (rotType == PS_ICC_ROT_B) { mode += 3; } return mode; } static INT getIIDMode(const INT nBands, const INT iidRes) { INT mode = 0; switch (nBands) { case PS_BANDS_COARSE: mode = PS_RES_COARSE; break; case PS_BANDS_MID: mode = PS_RES_MID; break; default: mode = 0; break; } if (iidRes == PS_IID_RES_FINE) { mode += 3; } return mode; } static INT envelopeReducible(FIXP_DBL iid[PS_MAX_ENVELOPES][PS_MAX_BANDS], FIXP_DBL icc[PS_MAX_ENVELOPES][PS_MAX_BANDS], INT psBands, INT nEnvelopes) { #define THRESH_SCALE 7 INT reducible = 1; /* true */ INT e = 0, b = 0; FIXP_DBL dIid = FL2FXCONST_DBL(0.f); FIXP_DBL dIcc = FL2FXCONST_DBL(0.f); FIXP_DBL iidErrThreshold, iccErrThreshold; FIXP_DBL iidMeanError, iccMeanError; /* square values to prevent sqrt, multiply bands to prevent division; bands shifted DFRACT_BITS instead (DFRACT_BITS-1) because fMultDiv2 used*/ iidErrThreshold = fMultDiv2(FL2FXCONST_DBL(6.5f * 6.5f / (IID_SCALE_FT * IID_SCALE_FT)), (FIXP_DBL)(psBands << ((DFRACT_BITS)-THRESH_SCALE))); iccErrThreshold = fMultDiv2(FL2FXCONST_DBL(0.75f * 0.75f), (FIXP_DBL)(psBands << ((DFRACT_BITS)-THRESH_SCALE))); if (nEnvelopes <= 1) { reducible = 0; } else { /* mean error criterion */ for (e = 0; (e < nEnvelopes / 2) && (reducible != 0); e++) { iidMeanError = iccMeanError = FL2FXCONST_DBL(0.f); for (b = 0; b < psBands; b++) { dIid = (iid[2 * e][b] >> 1) - (iid[2 * e + 1][b] >> 1); /* scale 1 bit; squared -> 2 bit */ dIcc = (icc[2 * e][b] >> 1) - (icc[2 * e + 1][b] >> 1); iidMeanError += fPow2Div2(dIid) >> (5 - 1); /* + (bands=20) scale = 5 */ iccMeanError += fPow2Div2(dIcc) >> (5 - 1); } /* --> scaling = 7 bit = THRESH_SCALE !! */ /* instead sqrt values are squared! instead of division, multiply threshold with psBands scaling necessary!! */ /* quit as soon as threshold is reached */ if ((iidMeanError > (iidErrThreshold)) || (iccMeanError > (iccErrThreshold))) { reducible = 0; } } } /* nEnvelopes != 1 */ return reducible; } static void processIidData(PS_DATA *psData, FIXP_DBL iid[PS_MAX_ENVELOPES][PS_MAX_BANDS], const INT psBands, const INT nEnvelopes, const FIXP_DBL quantErrorThreshold) { INT iidIdxFine[PS_MAX_ENVELOPES][PS_MAX_BANDS]; INT iidIdxCoarse[PS_MAX_ENVELOPES][PS_MAX_BANDS]; FIXP_DBL errIID = FL2FXCONST_DBL(0.f); FIXP_DBL errIIDFine = FL2FXCONST_DBL(0.f); INT bitsIidFreq = 0; INT bitsIidTime = 0; INT bitsFineTot = 0; INT bitsCoarseTot = 0; INT error = 0; INT env, band; INT diffMode[PS_MAX_ENVELOPES], diffModeFine[PS_MAX_ENVELOPES]; INT loudnDiff = 0; INT iidTransmit = 0; /* Quantize IID coefficients */ for (env = 0; env < nEnvelopes; env++) { errIID += quantizeCoef(iid[env], psBands, iidQuant_fx, 7, 15, iidIdxCoarse[env]); errIIDFine += quantizeCoef(iid[env], psBands, iidQuantFine_fx, 15, 31, iidIdxFine[env]); } /* normalize error to number of envelopes, ps bands errIID /= psBands*nEnvelopes; errIIDFine /= psBands*nEnvelopes; */ /* Check if IID coefficients should be used in this frame */ psData->iidEnable = 0; for (env = 0; env < nEnvelopes; env++) { for (band = 0; band < psBands; band++) { loudnDiff += fixp_abs(iidIdxCoarse[env][band]); iidTransmit++; } } if (loudnDiff > fMultI(FL2FXCONST_DBL(0.7f), iidTransmit)) { /* 0.7f empiric value */ psData->iidEnable = 1; } /* if iid not active -> RESET data */ if (psData->iidEnable == 0) { psData->iidTimeCnt = MAX_TIME_DIFF_FRAMES; for (env = 0; env < nEnvelopes; env++) { psData->iidDiffMode[env] = PS_DELTA_FREQ; FDKmemclear(psData->iidIdx[env], sizeof(INT) * psBands); } return; } /* count COARSE quantization bits for first envelope*/ bitsIidFreq = FDKsbrEnc_EncodeIid(NULL, iidIdxCoarse[0], NULL, psBands, PS_IID_RES_COARSE, PS_DELTA_FREQ, &error); if ((psData->iidTimeCnt >= MAX_TIME_DIFF_FRAMES) || (psData->iidQuantModeLast == PS_IID_RES_FINE)) { bitsIidTime = DO_NOT_USE_THIS_MODE; } else { bitsIidTime = FDKsbrEnc_EncodeIid(NULL, iidIdxCoarse[0], psData->iidIdxLast, psBands, PS_IID_RES_COARSE, PS_DELTA_TIME, &error); } /* decision DELTA_FREQ vs DELTA_TIME */ if (bitsIidTime > bitsIidFreq) { diffMode[0] = PS_DELTA_FREQ; bitsCoarseTot = bitsIidFreq; } else { diffMode[0] = PS_DELTA_TIME; bitsCoarseTot = bitsIidTime; } /* count COARSE quantization bits for following envelopes*/ for (env = 1; env < nEnvelopes; env++) { bitsIidFreq = FDKsbrEnc_EncodeIid(NULL, iidIdxCoarse[env], NULL, psBands, PS_IID_RES_COARSE, PS_DELTA_FREQ, &error); bitsIidTime = FDKsbrEnc_EncodeIid(NULL, iidIdxCoarse[env], iidIdxCoarse[env - 1], psBands, PS_IID_RES_COARSE, PS_DELTA_TIME, &error); /* decision DELTA_FREQ vs DELTA_TIME */ if (bitsIidTime > bitsIidFreq) { diffMode[env] = PS_DELTA_FREQ; bitsCoarseTot += bitsIidFreq; } else { diffMode[env] = PS_DELTA_TIME; bitsCoarseTot += bitsIidTime; } } /* count FINE quantization bits for first envelope*/ bitsIidFreq = FDKsbrEnc_EncodeIid(NULL, iidIdxFine[0], NULL, psBands, PS_IID_RES_FINE, PS_DELTA_FREQ, &error); if ((psData->iidTimeCnt >= MAX_TIME_DIFF_FRAMES) || (psData->iidQuantModeLast == PS_IID_RES_COARSE)) { bitsIidTime = DO_NOT_USE_THIS_MODE; } else { bitsIidTime = FDKsbrEnc_EncodeIid(NULL, iidIdxFine[0], psData->iidIdxLast, psBands, PS_IID_RES_FINE, PS_DELTA_TIME, &error); } /* decision DELTA_FREQ vs DELTA_TIME */ if (bitsIidTime > bitsIidFreq) { diffModeFine[0] = PS_DELTA_FREQ; bitsFineTot = bitsIidFreq; } else { diffModeFine[0] = PS_DELTA_TIME; bitsFineTot = bitsIidTime; } /* count FINE quantization bits for following envelopes*/ for (env = 1; env < nEnvelopes; env++) { bitsIidFreq = FDKsbrEnc_EncodeIid(NULL, iidIdxFine[env], NULL, psBands, PS_IID_RES_FINE, PS_DELTA_FREQ, &error); bitsIidTime = FDKsbrEnc_EncodeIid(NULL, iidIdxFine[env], iidIdxFine[env - 1], psBands, PS_IID_RES_FINE, PS_DELTA_TIME, &error); /* decision DELTA_FREQ vs DELTA_TIME */ if (bitsIidTime > bitsIidFreq) { diffModeFine[env] = PS_DELTA_FREQ; bitsFineTot += bitsIidFreq; } else { diffModeFine[env] = PS_DELTA_TIME; bitsFineTot += bitsIidTime; } } if (bitsFineTot == bitsCoarseTot) { /* if same number of bits is needed, use the quantization with lower error */ if (errIIDFine < errIID) { bitsCoarseTot = DO_NOT_USE_THIS_MODE; } else { bitsFineTot = DO_NOT_USE_THIS_MODE; } } else { /* const FIXP_DBL minThreshold = * FL2FXCONST_DBL(0.2f/(IID_SCALE_FT*PS_QUANT_SCALE_FT)*(psBands*nEnvelopes)); */ const FIXP_DBL minThreshold = (FIXP_DBL)((LONG)0x00019999 * (psBands * nEnvelopes)); /* decision RES_FINE vs RES_COARSE */ /* test if errIIDFine*quantErrorThreshold < errIID */ /* shiftVal 2 comes from scaling of quantErrorThreshold */ if (fixMax(((errIIDFine >> 1) + (minThreshold >> 1)) >> 1, fMult(quantErrorThreshold, errIIDFine)) < (errIID >> 2)) { bitsCoarseTot = DO_NOT_USE_THIS_MODE; } else if (fixMax(((errIID >> 1) + (minThreshold >> 1)) >> 1, fMult(quantErrorThreshold, errIID)) < (errIIDFine >> 2)) { bitsFineTot = DO_NOT_USE_THIS_MODE; } } /* decision RES_FINE vs RES_COARSE */ if (bitsFineTot < bitsCoarseTot) { psData->iidQuantMode = PS_IID_RES_FINE; for (env = 0; env < nEnvelopes; env++) { psData->iidDiffMode[env] = diffModeFine[env]; FDKmemcpy(psData->iidIdx[env], iidIdxFine[env], psBands * sizeof(INT)); } } else { psData->iidQuantMode = PS_IID_RES_COARSE; for (env = 0; env < nEnvelopes; env++) { psData->iidDiffMode[env] = diffMode[env]; FDKmemcpy(psData->iidIdx[env], iidIdxCoarse[env], psBands * sizeof(INT)); } } /* Count DELTA_TIME encoding streaks */ for (env = 0; env < nEnvelopes; env++) { if (psData->iidDiffMode[env] == PS_DELTA_TIME) psData->iidTimeCnt++; else psData->iidTimeCnt = 0; } } static INT similarIid(PS_DATA *psData, const INT psBands, const INT nEnvelopes) { const INT diffThr = (psData->iidQuantMode == PS_IID_RES_COARSE) ? 2 : 3; const INT sumDiffThr = diffThr * psBands / 4; INT similar = 0; INT diff = 0; INT sumDiff = 0; INT env = 0; INT b = 0; if ((nEnvelopes == psData->nEnvelopesLast) && (nEnvelopes == 1)) { similar = 1; for (env = 0; env < nEnvelopes; env++) { sumDiff = 0; b = 0; do { diff = fixp_abs(psData->iidIdx[env][b] - psData->iidIdxLast[b]); sumDiff += diff; if ((diff > diffThr) /* more than x quantization steps in any band */ || (sumDiff > sumDiffThr)) { /* more than x quantisations steps overall difference */ similar = 0; } b++; } while ((b < psBands) && (similar > 0)); } } /* nEnvelopes==1 */ return similar; } static INT similarIcc(PS_DATA *psData, const INT psBands, const INT nEnvelopes) { const INT diffThr = 2; const INT sumDiffThr = diffThr * psBands / 4; INT similar = 0; INT diff = 0; INT sumDiff = 0; INT env = 0; INT b = 0; if ((nEnvelopes == psData->nEnvelopesLast) && (nEnvelopes == 1)) { similar = 1; for (env = 0; env < nEnvelopes; env++) { sumDiff = 0; b = 0; do { diff = fixp_abs(psData->iccIdx[env][b] - psData->iccIdxLast[b]); sumDiff += diff; if ((diff > diffThr) /* more than x quantisation step in any band */ || (sumDiff > sumDiffThr)) { /* more than x quantisations steps overall difference */ similar = 0; } b++; } while ((b < psBands) && (similar > 0)); } } /* nEnvelopes==1 */ return similar; } static void processIccData( PS_DATA *psData, FIXP_DBL icc[PS_MAX_ENVELOPES][PS_MAX_BANDS], /* const input values: unable to declare as const, since it does not poINT to const memory */ const INT psBands, const INT nEnvelopes) { FIXP_DBL errICC = FL2FXCONST_DBL(0.f); INT env, band; INT bitsIccFreq, bitsIccTime; INT error = 0; INT inCoherence = 0, iccTransmit = 0; INT *iccIdxLast; iccIdxLast = psData->iccIdxLast; /* Quantize ICC coefficients */ for (env = 0; env < nEnvelopes; env++) { errICC += quantizeCoef(icc[env], psBands, iccQuant, 0, 8, psData->iccIdx[env]); } /* Check if ICC coefficients should be used */ psData->iccEnable = 0; for (env = 0; env < nEnvelopes; env++) { for (band = 0; band < psBands; band++) { inCoherence += psData->iccIdx[env][band]; iccTransmit++; } } if (inCoherence > fMultI(FL2FXCONST_DBL(0.5f), iccTransmit)) { /* 0.5f empiric value */ psData->iccEnable = 1; } if (psData->iccEnable == 0) { psData->iccTimeCnt = MAX_TIME_DIFF_FRAMES; for (env = 0; env < nEnvelopes; env++) { psData->iccDiffMode[env] = PS_DELTA_FREQ; FDKmemclear(psData->iccIdx[env], sizeof(INT) * psBands); } return; } for (env = 0; env < nEnvelopes; env++) { bitsIccFreq = FDKsbrEnc_EncodeIcc(NULL, psData->iccIdx[env], NULL, psBands, PS_DELTA_FREQ, &error); if (psData->iccTimeCnt < MAX_TIME_DIFF_FRAMES) { bitsIccTime = FDKsbrEnc_EncodeIcc(NULL, psData->iccIdx[env], iccIdxLast, psBands, PS_DELTA_TIME, &error); } else { bitsIccTime = DO_NOT_USE_THIS_MODE; } if (bitsIccFreq > bitsIccTime) { psData->iccDiffMode[env] = PS_DELTA_TIME; psData->iccTimeCnt++; } else { psData->iccDiffMode[env] = PS_DELTA_FREQ; psData->iccTimeCnt = 0; } iccIdxLast = psData->iccIdx[env]; } } static void calculateIID(FIXP_DBL ldPwrL[PS_MAX_ENVELOPES][PS_MAX_BANDS], FIXP_DBL ldPwrR[PS_MAX_ENVELOPES][PS_MAX_BANDS], FIXP_DBL iid[PS_MAX_ENVELOPES][PS_MAX_BANDS], INT nEnvelopes, INT psBands) { INT i = 0; INT env = 0; for (env = 0; env < nEnvelopes; env++) { for (i = 0; i < psBands; i++) { /* iid[env][i] = 10.0f*(float)log10(pwrL[env][i]/pwrR[env][i]); */ FIXP_DBL IID = fMultDiv2(FL2FXCONST_DBL(LOG10_2_10 / IID_SCALE_FT), (ldPwrL[env][i] - ldPwrR[env][i])); IID = fixMin(IID, (FIXP_DBL)(MAXVAL_DBL >> (LD_DATA_SHIFT + 1))); IID = fixMax(IID, (FIXP_DBL)(MINVAL_DBL >> (LD_DATA_SHIFT + 1))); iid[env][i] = IID << (LD_DATA_SHIFT + 1); } } } static void calculateICC(FIXP_DBL pwrL[PS_MAX_ENVELOPES][PS_MAX_BANDS], FIXP_DBL pwrR[PS_MAX_ENVELOPES][PS_MAX_BANDS], FIXP_DBL pwrCr[PS_MAX_ENVELOPES][PS_MAX_BANDS], FIXP_DBL pwrCi[PS_MAX_ENVELOPES][PS_MAX_BANDS], FIXP_DBL icc[PS_MAX_ENVELOPES][PS_MAX_BANDS], INT nEnvelopes, INT psBands) { INT i = 0; INT env = 0; INT border = psBands; switch (psBands) { case PS_BANDS_COARSE: border = 5; break; case PS_BANDS_MID: border = 11; break; default: break; } for (env = 0; env < nEnvelopes; env++) { for (i = 0; i < border; i++) { /* icc[env][i] = min( pwrCr[env][i] / (float) sqrt(pwrL[env][i] * * pwrR[env][i]) , 1.f); */ int scale; FIXP_DBL invNrg = invSqrtNorm2( fMax(fMult(pwrL[env][i], pwrR[env][i]), (FIXP_DBL)1), &scale); icc[env][i] = SATURATE_LEFT_SHIFT(fMult(pwrCr[env][i], invNrg), scale, DFRACT_BITS); } for (; i < psBands; i++) { int denom_e; FIXP_DBL denom_m = fMultNorm(pwrL[env][i], pwrR[env][i], &denom_e); if (denom_m == (FIXP_DBL)0) { icc[env][i] = (FIXP_DBL)MAXVAL_DBL; } else { int num_e, result_e; FIXP_DBL num_m, result_m; num_e = CountLeadingBits( fixMax(fixp_abs(pwrCr[env][i]), fixp_abs(pwrCi[env][i]))); num_m = fPow2Div2((pwrCr[env][i] << num_e)) + fPow2Div2((pwrCi[env][i] << num_e)); result_m = fDivNorm(num_m, denom_m, &result_e); result_e += (-2 * num_e + 1) - denom_e; icc[env][i] = scaleValueSaturate(sqrtFixp(result_m >> (result_e & 1)), (result_e + (result_e & 1)) >> 1); } } } } void FDKsbrEnc_initPsBandNrgScale(HANDLE_PS_ENCODE hPsEncode) { INT group, bin; INT nIidGroups = hPsEncode->nQmfIidGroups + hPsEncode->nSubQmfIidGroups; FDKmemclear(hPsEncode->psBandNrgScale, PS_MAX_BANDS * sizeof(SCHAR)); for (group = 0; group < nIidGroups; group++) { /* Translate group to bin */ bin = hPsEncode->subband2parameterIndex[group]; /* Translate from 20 bins to 10 bins */ if (hPsEncode->psEncMode == PS_BANDS_COARSE) { bin = bin >> 1; } hPsEncode->psBandNrgScale[bin] = (hPsEncode->psBandNrgScale[bin] == 0) ? (hPsEncode->iidGroupWidthLd[group] + 5) : (fixMax(hPsEncode->iidGroupWidthLd[group], hPsEncode->psBandNrgScale[bin]) + 1); } } FDK_PSENC_ERROR FDKsbrEnc_CreatePSEncode(HANDLE_PS_ENCODE *phPsEncode) { FDK_PSENC_ERROR error = PSENC_OK; if (phPsEncode == NULL) { error = PSENC_INVALID_HANDLE; } else { HANDLE_PS_ENCODE hPsEncode = NULL; if (NULL == (hPsEncode = GetRam_PsEncode())) { error = PSENC_MEMORY_ERROR; goto bail; } FDKmemclear(hPsEncode, sizeof(PS_ENCODE)); *phPsEncode = hPsEncode; /* return allocated handle */ } bail: return error; } FDK_PSENC_ERROR FDKsbrEnc_InitPSEncode(HANDLE_PS_ENCODE hPsEncode, const PS_BANDS psEncMode, const FIXP_DBL iidQuantErrorThreshold) { FDK_PSENC_ERROR error = PSENC_OK; if (NULL == hPsEncode) { error = PSENC_INVALID_HANDLE; } else { if (PSENC_OK != (InitPSData(&hPsEncode->psData))) { goto bail; } switch (psEncMode) { case PS_BANDS_COARSE: case PS_BANDS_MID: hPsEncode->nQmfIidGroups = QMF_GROUPS_LO_RES; hPsEncode->nSubQmfIidGroups = SUBQMF_GROUPS_LO_RES; FDKmemcpy(hPsEncode->iidGroupBorders, iidGroupBordersLoRes, (hPsEncode->nQmfIidGroups + hPsEncode->nSubQmfIidGroups + 1) * sizeof(INT)); FDKmemcpy(hPsEncode->subband2parameterIndex, subband2parameter20, (hPsEncode->nQmfIidGroups + hPsEncode->nSubQmfIidGroups) * sizeof(INT)); FDKmemcpy(hPsEncode->iidGroupWidthLd, iidGroupWidthLdLoRes, (hPsEncode->nQmfIidGroups + hPsEncode->nSubQmfIidGroups) * sizeof(UCHAR)); break; default: error = PSENC_INIT_ERROR; goto bail; } hPsEncode->psEncMode = psEncMode; hPsEncode->iidQuantErrorThreshold = iidQuantErrorThreshold; FDKsbrEnc_initPsBandNrgScale(hPsEncode); } bail: return error; } FDK_PSENC_ERROR FDKsbrEnc_DestroyPSEncode(HANDLE_PS_ENCODE *phPsEncode) { FDK_PSENC_ERROR error = PSENC_OK; if (NULL != phPsEncode) { FreeRam_PsEncode(phPsEncode); } return error; } typedef struct { FIXP_DBL pwrL[PS_MAX_ENVELOPES][PS_MAX_BANDS]; FIXP_DBL pwrR[PS_MAX_ENVELOPES][PS_MAX_BANDS]; FIXP_DBL ldPwrL[PS_MAX_ENVELOPES][PS_MAX_BANDS]; FIXP_DBL ldPwrR[PS_MAX_ENVELOPES][PS_MAX_BANDS]; FIXP_DBL pwrCr[PS_MAX_ENVELOPES][PS_MAX_BANDS]; FIXP_DBL pwrCi[PS_MAX_ENVELOPES][PS_MAX_BANDS]; } PS_PWR_DATA; FDK_PSENC_ERROR FDKsbrEnc_PSEncode( HANDLE_PS_ENCODE hPsEncode, HANDLE_PS_OUT hPsOut, UCHAR *dynBandScale, UINT maxEnvelopes, FIXP_DBL *hybridData[HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2], const INT frameSize, const INT sendHeader) { FDK_PSENC_ERROR error = PSENC_OK; HANDLE_PS_DATA hPsData = &hPsEncode->psData; FIXP_DBL iid[PS_MAX_ENVELOPES][PS_MAX_BANDS]; FIXP_DBL icc[PS_MAX_ENVELOPES][PS_MAX_BANDS]; int envBorder[PS_MAX_ENVELOPES + 1]; int group, bin, col, subband, band; int i = 0; int env = 0; int psBands = (int)hPsEncode->psEncMode; int nIidGroups = hPsEncode->nQmfIidGroups + hPsEncode->nSubQmfIidGroups; int nEnvelopes = fixMin(maxEnvelopes, (UINT)PS_MAX_ENVELOPES); C_ALLOC_SCRATCH_START(pwrData, PS_PWR_DATA, 1) for (env = 0; env < nEnvelopes + 1; env++) { envBorder[env] = fMultI(GetInvInt(nEnvelopes), frameSize * env); } for (env = 0; env < nEnvelopes; env++) { /* clear energy array */ for (band = 0; band < psBands; band++) { pwrData->pwrL[env][band] = pwrData->pwrR[env][band] = pwrData->pwrCr[env][band] = pwrData->pwrCi[env][band] = FIXP_DBL(1); } /**** calculate energies and correlation ****/ /* start with hybrid data */ for (group = 0; group < nIidGroups; group++) { /* Translate group to bin */ bin = hPsEncode->subband2parameterIndex[group]; /* Translate from 20 bins to 10 bins */ if (hPsEncode->psEncMode == PS_BANDS_COARSE) { bin >>= 1; } /* determine group border */ int bScale = hPsEncode->psBandNrgScale[bin]; FIXP_DBL pwrL_env_bin = pwrData->pwrL[env][bin]; FIXP_DBL pwrR_env_bin = pwrData->pwrR[env][bin]; FIXP_DBL pwrCr_env_bin = pwrData->pwrCr[env][bin]; FIXP_DBL pwrCi_env_bin = pwrData->pwrCi[env][bin]; int scale = (int)dynBandScale[bin]; for (col = envBorder[env]; col < envBorder[env + 1]; col++) { for (subband = hPsEncode->iidGroupBorders[group]; subband < hPsEncode->iidGroupBorders[group + 1]; subband++) { FIXP_DBL l_real = (hybridData[col][0][0][subband]) << scale; FIXP_DBL l_imag = (hybridData[col][0][1][subband]) << scale; FIXP_DBL r_real = (hybridData[col][1][0][subband]) << scale; FIXP_DBL r_imag = (hybridData[col][1][1][subband]) << scale; pwrL_env_bin += (fPow2Div2(l_real) + fPow2Div2(l_imag)) >> bScale; pwrR_env_bin += (fPow2Div2(r_real) + fPow2Div2(r_imag)) >> bScale; pwrCr_env_bin += (fMultDiv2(l_real, r_real) + fMultDiv2(l_imag, r_imag)) >> bScale; pwrCi_env_bin += (fMultDiv2(r_real, l_imag) - fMultDiv2(l_real, r_imag)) >> bScale; } } /* assure, nrg's of left and right channel are not negative; necessary on * 16 bit multiply units */ pwrData->pwrL[env][bin] = fixMax((FIXP_DBL)0, pwrL_env_bin); pwrData->pwrR[env][bin] = fixMax((FIXP_DBL)0, pwrR_env_bin); pwrData->pwrCr[env][bin] = pwrCr_env_bin; pwrData->pwrCi[env][bin] = pwrCi_env_bin; } /* nIidGroups */ /* calc logarithmic energy */ LdDataVector(pwrData->pwrL[env], pwrData->ldPwrL[env], psBands); LdDataVector(pwrData->pwrR[env], pwrData->ldPwrR[env], psBands); } /* nEnvelopes */ /* calculate iid and icc */ calculateIID(pwrData->ldPwrL, pwrData->ldPwrR, iid, nEnvelopes, psBands); calculateICC(pwrData->pwrL, pwrData->pwrR, pwrData->pwrCr, pwrData->pwrCi, icc, nEnvelopes, psBands); /*** Envelope Reduction ***/ while (envelopeReducible(iid, icc, psBands, nEnvelopes)) { int e = 0; /* sum energies of two neighboring envelopes */ nEnvelopes >>= 1; for (e = 0; e < nEnvelopes; e++) { FDKsbrEnc_addFIXP_DBL(pwrData->pwrL[2 * e], pwrData->pwrL[2 * e + 1], pwrData->pwrL[e], psBands); FDKsbrEnc_addFIXP_DBL(pwrData->pwrR[2 * e], pwrData->pwrR[2 * e + 1], pwrData->pwrR[e], psBands); FDKsbrEnc_addFIXP_DBL(pwrData->pwrCr[2 * e], pwrData->pwrCr[2 * e + 1], pwrData->pwrCr[e], psBands); FDKsbrEnc_addFIXP_DBL(pwrData->pwrCi[2 * e], pwrData->pwrCi[2 * e + 1], pwrData->pwrCi[e], psBands); /* calc logarithmic energy */ LdDataVector(pwrData->pwrL[e], pwrData->ldPwrL[e], psBands); LdDataVector(pwrData->pwrR[e], pwrData->ldPwrR[e], psBands); /* reduce number of envelopes and adjust borders */ envBorder[e] = envBorder[2 * e]; } envBorder[nEnvelopes] = envBorder[2 * nEnvelopes]; /* re-calculate iid and icc */ calculateIID(pwrData->ldPwrL, pwrData->ldPwrR, iid, nEnvelopes, psBands); calculateICC(pwrData->pwrL, pwrData->pwrR, pwrData->pwrCr, pwrData->pwrCi, icc, nEnvelopes, psBands); } /* */ if (sendHeader) { hPsData->headerCnt = MAX_PS_NOHEADER_CNT; hPsData->iidTimeCnt = MAX_TIME_DIFF_FRAMES; hPsData->iccTimeCnt = MAX_TIME_DIFF_FRAMES; hPsData->noEnvCnt = MAX_NOENV_CNT; } /*** Parameter processing, quantisation etc ***/ processIidData(hPsData, iid, psBands, nEnvelopes, hPsEncode->iidQuantErrorThreshold); processIccData(hPsData, icc, psBands, nEnvelopes); /*** Initialize output struct ***/ /* PS Header on/off ? */ if ((hPsData->headerCnt < MAX_PS_NOHEADER_CNT) && ((hPsData->iidQuantMode == hPsData->iidQuantModeLast) && (hPsData->iccQuantMode == hPsData->iccQuantModeLast)) && ((hPsData->iidEnable == hPsData->iidEnableLast) && (hPsData->iccEnable == hPsData->iccEnableLast))) { hPsOut->enablePSHeader = 0; } else { hPsOut->enablePSHeader = 1; hPsData->headerCnt = 0; } /* nEnvelopes = 0 ? */ if ((hPsData->noEnvCnt < MAX_NOENV_CNT) && (similarIid(hPsData, psBands, nEnvelopes)) && (similarIcc(hPsData, psBands, nEnvelopes))) { hPsOut->nEnvelopes = nEnvelopes = 0; hPsData->noEnvCnt++; } else { hPsData->noEnvCnt = 0; } if (nEnvelopes > 0) { hPsOut->enableIID = hPsData->iidEnable; hPsOut->iidMode = getIIDMode(psBands, hPsData->iidQuantMode); hPsOut->enableICC = hPsData->iccEnable; hPsOut->iccMode = getICCMode(psBands, hPsData->iccQuantMode); hPsOut->enableIpdOpd = 0; hPsOut->frameClass = 0; hPsOut->nEnvelopes = nEnvelopes; for (env = 0; env < nEnvelopes; env++) { hPsOut->frameBorder[env] = envBorder[env + 1]; hPsOut->deltaIID[env] = (PS_DELTA)hPsData->iidDiffMode[env]; hPsOut->deltaICC[env] = (PS_DELTA)hPsData->iccDiffMode[env]; for (band = 0; band < psBands; band++) { hPsOut->iid[env][band] = hPsData->iidIdx[env][band]; hPsOut->icc[env][band] = hPsData->iccIdx[env][band]; } } /* IPD OPD not supported right now */ FDKmemclear(hPsOut->ipd, PS_MAX_ENVELOPES * PS_MAX_BANDS * sizeof(PS_DELTA)); for (env = 0; env < PS_MAX_ENVELOPES; env++) { hPsOut->deltaIPD[env] = PS_DELTA_FREQ; hPsOut->deltaOPD[env] = PS_DELTA_FREQ; } FDKmemclear(hPsOut->ipdLast, PS_MAX_BANDS * sizeof(INT)); FDKmemclear(hPsOut->opdLast, PS_MAX_BANDS * sizeof(INT)); for (band = 0; band < PS_MAX_BANDS; band++) { hPsOut->iidLast[band] = hPsData->iidIdxLast[band]; hPsOut->iccLast[band] = hPsData->iccIdxLast[band]; } /* save iids and iccs for differential time coding in the next frame */ hPsData->nEnvelopesLast = nEnvelopes; hPsData->iidEnableLast = hPsData->iidEnable; hPsData->iccEnableLast = hPsData->iccEnable; hPsData->iidQuantModeLast = hPsData->iidQuantMode; hPsData->iccQuantModeLast = hPsData->iccQuantMode; for (i = 0; i < psBands; i++) { hPsData->iidIdxLast[i] = hPsData->iidIdx[nEnvelopes - 1][i]; hPsData->iccIdxLast[i] = hPsData->iccIdx[nEnvelopes - 1][i]; } } /* Envelope > 0 */ C_ALLOC_SCRATCH_END(pwrData, PS_PWR_DATA, 1) return error; } fdk-aac-2.0.2/libSBRenc/src/ps_encode.h000066400000000000000000000157761404225310700175210ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): M. Neuendorf, N. Rettelbach, M. Multrus Description: PS Parameter extraction, encoding *******************************************************************************/ /*! \file \brief PS parameter extraction, encoding functions $Revision: 92790 $ */ #ifndef PS_ENCODE_H #define PS_ENCODE_H #include "ps_const.h" #include "ps_bitenc.h" #define IID_SCALE_FT (64.f) /* maxVal in Quant tab is +/- 50 */ #define IID_SCALE 6 /* maxVal in Quant tab is +/- 50 */ #define IID_MAXVAL (1 << IID_SCALE) #define PS_QUANT_SCALE_FT \ (64.f) /* error smaller (64-25)/64 * 20 bands * 4 env -> QuantScale 64 */ #define PS_QUANT_SCALE \ 6 /* error smaller (64-25)/64 * 20 bands * 4 env -> QuantScale 6 bit */ #define QMF_GROUPS_LO_RES 12 #define SUBQMF_GROUPS_LO_RES 10 #define QMF_GROUPS_HI_RES 18 #define SUBQMF_GROUPS_HI_RES 30 typedef struct T_PS_DATA { INT iidEnable; INT iidEnableLast; INT iidQuantMode; INT iidQuantModeLast; INT iidDiffMode[PS_MAX_ENVELOPES]; INT iidIdx[PS_MAX_ENVELOPES][PS_MAX_BANDS]; INT iidIdxLast[PS_MAX_BANDS]; INT iccEnable; INT iccEnableLast; INT iccQuantMode; INT iccQuantModeLast; INT iccDiffMode[PS_MAX_ENVELOPES]; INT iccIdx[PS_MAX_ENVELOPES][PS_MAX_BANDS]; INT iccIdxLast[PS_MAX_BANDS]; INT nEnvelopesLast; INT headerCnt; INT iidTimeCnt; INT iccTimeCnt; INT noEnvCnt; } PS_DATA, *HANDLE_PS_DATA; typedef struct T_PS_ENCODE { PS_DATA psData; PS_BANDS psEncMode; INT nQmfIidGroups; INT nSubQmfIidGroups; INT iidGroupBorders[QMF_GROUPS_HI_RES + SUBQMF_GROUPS_HI_RES + 1]; INT subband2parameterIndex[QMF_GROUPS_HI_RES + SUBQMF_GROUPS_HI_RES]; UCHAR iidGroupWidthLd[QMF_GROUPS_HI_RES + SUBQMF_GROUPS_HI_RES]; FIXP_DBL iidQuantErrorThreshold; UCHAR psBandNrgScale[PS_MAX_BANDS]; } PS_ENCODE; typedef struct T_PS_ENCODE *HANDLE_PS_ENCODE; FDK_PSENC_ERROR FDKsbrEnc_CreatePSEncode(HANDLE_PS_ENCODE *phPsEncode); FDK_PSENC_ERROR FDKsbrEnc_InitPSEncode(HANDLE_PS_ENCODE hPsEncode, const PS_BANDS psEncMode, const FIXP_DBL iidQuantErrorThreshold); FDK_PSENC_ERROR FDKsbrEnc_DestroyPSEncode(HANDLE_PS_ENCODE *phPsEncode); FDK_PSENC_ERROR FDKsbrEnc_PSEncode( HANDLE_PS_ENCODE hPsEncode, HANDLE_PS_OUT hPsOut, UCHAR *dynBandScale, UINT maxEnvelopes, FIXP_DBL *hybridData[HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2], const INT frameSize, const INT sendHeader); #endif fdk-aac-2.0.2/libSBRenc/src/ps_main.cpp000066400000000000000000000545571404225310700175430ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): M. Multrus Description: PS Wrapper, Downmix *******************************************************************************/ #include "ps_main.h" /* Includes ******************************************************************/ #include "ps_bitenc.h" #include "sbrenc_ram.h" /*--------------- function declarations --------------------*/ static void psFindBestScaling( HANDLE_PARAMETRIC_STEREO hParametricStereo, FIXP_DBL *hybridData[HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2], UCHAR *dynBandScale, FIXP_DBL *maxBandValue, SCHAR *dmxScale); /*------------- function definitions ----------------*/ FDK_PSENC_ERROR PSEnc_Create(HANDLE_PARAMETRIC_STEREO *phParametricStereo) { FDK_PSENC_ERROR error = PSENC_OK; HANDLE_PARAMETRIC_STEREO hParametricStereo = NULL; if (phParametricStereo == NULL) { error = PSENC_INVALID_HANDLE; } else { int i; if (NULL == (hParametricStereo = GetRam_ParamStereo())) { error = PSENC_MEMORY_ERROR; goto bail; } FDKmemclear(hParametricStereo, sizeof(PARAMETRIC_STEREO)); if (PSENC_OK != (error = FDKsbrEnc_CreatePSEncode(&hParametricStereo->hPsEncode))) { error = PSENC_MEMORY_ERROR; goto bail; } for (i = 0; i < MAX_PS_CHANNELS; i++) { if (FDKhybridAnalysisOpen( &hParametricStereo->fdkHybAnaFilter[i], hParametricStereo->__staticHybAnaStatesLF[i], sizeof(hParametricStereo->__staticHybAnaStatesLF[i]), hParametricStereo->__staticHybAnaStatesHF[i], sizeof(hParametricStereo->__staticHybAnaStatesHF[i])) != 0) { error = PSENC_MEMORY_ERROR; goto bail; } } } bail: if (phParametricStereo != NULL) { *phParametricStereo = hParametricStereo; /* return allocated handle */ } if (error != PSENC_OK) { PSEnc_Destroy(phParametricStereo); } return error; } FDK_PSENC_ERROR PSEnc_Init(HANDLE_PARAMETRIC_STEREO hParametricStereo, const HANDLE_PSENC_CONFIG hPsEncConfig, INT noQmfSlots, INT noQmfBands, UCHAR *dynamic_RAM) { FDK_PSENC_ERROR error = PSENC_OK; if ((NULL == hParametricStereo) || (NULL == hPsEncConfig)) { error = PSENC_INVALID_HANDLE; } else { int ch, i; hParametricStereo->initPS = 1; hParametricStereo->noQmfSlots = noQmfSlots; hParametricStereo->noQmfBands = noQmfBands; /* clear delay lines */ FDKmemclear(hParametricStereo->qmfDelayLines, sizeof(hParametricStereo->qmfDelayLines)); hParametricStereo->qmfDelayScale = FRACT_BITS - 1; /* create configuration for hybrid filter bank */ for (ch = 0; ch < MAX_PS_CHANNELS; ch++) { FDKhybridAnalysisInit(&hParametricStereo->fdkHybAnaFilter[ch], THREE_TO_TEN, 64, 64, 1); } /* ch */ FDKhybridSynthesisInit(&hParametricStereo->fdkHybSynFilter, THREE_TO_TEN, 64, 64); /* determine average delay */ hParametricStereo->psDelay = (HYBRID_FILTER_DELAY * hParametricStereo->noQmfBands); if ((hPsEncConfig->maxEnvelopes < PSENC_NENV_1) || (hPsEncConfig->maxEnvelopes > PSENC_NENV_MAX)) { hPsEncConfig->maxEnvelopes = PSENC_NENV_DEFAULT; } hParametricStereo->maxEnvelopes = hPsEncConfig->maxEnvelopes; if (PSENC_OK != (error = FDKsbrEnc_InitPSEncode( hParametricStereo->hPsEncode, (PS_BANDS)hPsEncConfig->nStereoBands, hPsEncConfig->iidQuantErrorThreshold))) { goto bail; } for (ch = 0; ch < MAX_PS_CHANNELS; ch++) { FIXP_DBL *pDynReal = GetRam_Sbr_envRBuffer(ch, dynamic_RAM); FIXP_DBL *pDynImag = GetRam_Sbr_envIBuffer(ch, dynamic_RAM); for (i = 0; i < HYBRID_FRAMESIZE; i++) { hParametricStereo->pHybridData[i + HYBRID_READ_OFFSET][ch][0] = &pDynReal[i * MAX_HYBRID_BANDS]; hParametricStereo->pHybridData[i + HYBRID_READ_OFFSET][ch][1] = &pDynImag[i * MAX_HYBRID_BANDS]; ; } for (i = 0; i < HYBRID_READ_OFFSET; i++) { hParametricStereo->pHybridData[i][ch][0] = hParametricStereo->__staticHybridData[i][ch][0]; hParametricStereo->pHybridData[i][ch][1] = hParametricStereo->__staticHybridData[i][ch][1]; } } /* ch */ /* clear static hybrid buffer */ FDKmemclear(hParametricStereo->__staticHybridData, sizeof(hParametricStereo->__staticHybridData)); /* clear bs buffer */ FDKmemclear(hParametricStereo->psOut, sizeof(hParametricStereo->psOut)); hParametricStereo->psOut[0].enablePSHeader = 1; /* write ps header in first frame */ /* clear scaling buffer */ FDKmemclear(hParametricStereo->dynBandScale, sizeof(UCHAR) * PS_MAX_BANDS); FDKmemclear(hParametricStereo->maxBandValue, sizeof(FIXP_DBL) * PS_MAX_BANDS); } /* valid handle */ bail: return error; } FDK_PSENC_ERROR PSEnc_Destroy(HANDLE_PARAMETRIC_STEREO *phParametricStereo) { FDK_PSENC_ERROR error = PSENC_OK; if (NULL != phParametricStereo) { HANDLE_PARAMETRIC_STEREO hParametricStereo = *phParametricStereo; if (hParametricStereo != NULL) { FDKsbrEnc_DestroyPSEncode(&hParametricStereo->hPsEncode); FreeRam_ParamStereo(phParametricStereo); } } return error; } static FDK_PSENC_ERROR ExtractPSParameters( HANDLE_PARAMETRIC_STEREO hParametricStereo, const int sendHeader, FIXP_DBL *hybridData[HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2]) { FDK_PSENC_ERROR error = PSENC_OK; if (hParametricStereo == NULL) { error = PSENC_INVALID_HANDLE; } else { /* call ps encode function */ if (hParametricStereo->initPS) { hParametricStereo->psOut[1] = hParametricStereo->psOut[0]; } hParametricStereo->psOut[0] = hParametricStereo->psOut[1]; if (PSENC_OK != (error = FDKsbrEnc_PSEncode( hParametricStereo->hPsEncode, &hParametricStereo->psOut[1], hParametricStereo->dynBandScale, hParametricStereo->maxEnvelopes, hybridData, hParametricStereo->noQmfSlots, sendHeader))) { goto bail; } if (hParametricStereo->initPS) { hParametricStereo->psOut[0] = hParametricStereo->psOut[1]; hParametricStereo->initPS = 0; } } bail: return error; } static FDK_PSENC_ERROR DownmixPSQmfData( HANDLE_PARAMETRIC_STEREO hParametricStereo, HANDLE_QMF_FILTER_BANK sbrSynthQmf, FIXP_DBL **RESTRICT mixRealQmfData, FIXP_DBL **RESTRICT mixImagQmfData, INT_PCM *downsampledOutSignal, const UINT downsampledOutSignalBufSize, FIXP_DBL *hybridData[HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2], const INT noQmfSlots, const INT psQmfScale[MAX_PS_CHANNELS], SCHAR *qmfScale) { FDK_PSENC_ERROR error = PSENC_OK; if (hParametricStereo == NULL) { error = PSENC_INVALID_HANDLE; } else { int n, k; C_AALLOC_SCRATCH_START(pWorkBuffer, FIXP_DBL, 2 * 64) /* define scalings */ int dynQmfScale = fixMax( 0, hParametricStereo->dmxScale - 1); /* scale one bit more for addition of left and right */ int downmixScale = psQmfScale[0] - dynQmfScale; const FIXP_DBL maxStereoScaleFactor = MAXVAL_DBL; /* 2.f/2.f */ for (n = 0; n < noQmfSlots; n++) { FIXP_DBL tmpHybrid[2][MAX_HYBRID_BANDS]; for (k = 0; k < 71; k++) { int dynScale, sc; /* scaling */ FIXP_DBL tmpLeftReal, tmpRightReal, tmpLeftImag, tmpRightImag; FIXP_DBL tmpScaleFactor, stereoScaleFactor; tmpLeftReal = hybridData[n][0][0][k]; tmpLeftImag = hybridData[n][0][1][k]; tmpRightReal = hybridData[n][1][0][k]; tmpRightImag = hybridData[n][1][1][k]; sc = fixMax( 0, CntLeadingZeros(fixMax( fixMax(fixp_abs(tmpLeftReal), fixp_abs(tmpLeftImag)), fixMax(fixp_abs(tmpRightReal), fixp_abs(tmpRightImag)))) - 2); tmpLeftReal <<= sc; tmpLeftImag <<= sc; tmpRightReal <<= sc; tmpRightImag <<= sc; dynScale = fixMin(sc - dynQmfScale, DFRACT_BITS - 1); /* calc stereo scale factor to avoid loss of energy in bands */ /* stereo scale factor = min(2.0f, sqrt( (abs(l(k, n)^2 + abs(r(k, n)^2 * )))/(0.5f*abs(l(k, n) + r(k, n))) )) */ stereoScaleFactor = fPow2Div2(tmpLeftReal) + fPow2Div2(tmpLeftImag) + fPow2Div2(tmpRightReal) + fPow2Div2(tmpRightImag); /* might be that tmpScaleFactor becomes negative, so fabs(.) */ tmpScaleFactor = fixp_abs(stereoScaleFactor + fMult(tmpLeftReal, tmpRightReal) + fMult(tmpLeftImag, tmpRightImag)); /* min(2.0f, sqrt(stereoScaleFactor/(0.5f*tmpScaleFactor))) */ if ((stereoScaleFactor >> 1) < fMult(maxStereoScaleFactor, tmpScaleFactor)) { int sc_num = CountLeadingBits(stereoScaleFactor); int sc_denum = CountLeadingBits(tmpScaleFactor); sc = -(sc_num - sc_denum); tmpScaleFactor = schur_div((stereoScaleFactor << (sc_num)) >> 1, tmpScaleFactor << sc_denum, 16); /* prevent odd scaling for next sqrt calculation */ if (sc & 0x1) { sc++; tmpScaleFactor >>= 1; } stereoScaleFactor = sqrtFixp(tmpScaleFactor); stereoScaleFactor <<= (sc >> 1); } else { stereoScaleFactor = maxStereoScaleFactor; } /* write data to hybrid output */ tmpHybrid[0][k] = fMultDiv2(stereoScaleFactor, (FIXP_DBL)(tmpLeftReal + tmpRightReal)) >> dynScale; tmpHybrid[1][k] = fMultDiv2(stereoScaleFactor, (FIXP_DBL)(tmpLeftImag + tmpRightImag)) >> dynScale; } /* hybrid bands - k */ FDKhybridSynthesisApply(&hParametricStereo->fdkHybSynFilter, tmpHybrid[0], tmpHybrid[1], mixRealQmfData[n], mixImagQmfData[n]); qmfSynthesisFilteringSlot( sbrSynthQmf, mixRealQmfData[n], mixImagQmfData[n], downmixScale - 7, downmixScale - 7, downsampledOutSignal + (n * sbrSynthQmf->no_channels), 1, pWorkBuffer); } /* slots */ *qmfScale = -downmixScale + 7; C_AALLOC_SCRATCH_END(pWorkBuffer, FIXP_DBL, 2 * 64) { const INT noQmfSlots2 = hParametricStereo->noQmfSlots >> 1; const int noQmfBands = hParametricStereo->noQmfBands; INT scale, i, j, slotOffset; FIXP_DBL tmp[2][64]; for (i = 0; i < noQmfSlots2; i++) { FDKmemcpy(tmp[0], hParametricStereo->qmfDelayLines[0][i], noQmfBands * sizeof(FIXP_DBL)); FDKmemcpy(tmp[1], hParametricStereo->qmfDelayLines[1][i], noQmfBands * sizeof(FIXP_DBL)); FDKmemcpy(hParametricStereo->qmfDelayLines[0][i], mixRealQmfData[i + noQmfSlots2], noQmfBands * sizeof(FIXP_DBL)); FDKmemcpy(hParametricStereo->qmfDelayLines[1][i], mixImagQmfData[i + noQmfSlots2], noQmfBands * sizeof(FIXP_DBL)); FDKmemcpy(mixRealQmfData[i + noQmfSlots2], mixRealQmfData[i], noQmfBands * sizeof(FIXP_DBL)); FDKmemcpy(mixImagQmfData[i + noQmfSlots2], mixImagQmfData[i], noQmfBands * sizeof(FIXP_DBL)); FDKmemcpy(mixRealQmfData[i], tmp[0], noQmfBands * sizeof(FIXP_DBL)); FDKmemcpy(mixImagQmfData[i], tmp[1], noQmfBands * sizeof(FIXP_DBL)); } if (hParametricStereo->qmfDelayScale > *qmfScale) { scale = hParametricStereo->qmfDelayScale - *qmfScale; slotOffset = 0; } else { scale = *qmfScale - hParametricStereo->qmfDelayScale; slotOffset = noQmfSlots2; } for (i = 0; i < noQmfSlots2; i++) { for (j = 0; j < noQmfBands; j++) { mixRealQmfData[i + slotOffset][j] >>= scale; mixImagQmfData[i + slotOffset][j] >>= scale; } } scale = *qmfScale; *qmfScale = fMin(*qmfScale, hParametricStereo->qmfDelayScale); hParametricStereo->qmfDelayScale = scale; } } /* valid handle */ return error; } INT FDKsbrEnc_PSEnc_WritePSData(HANDLE_PARAMETRIC_STEREO hParametricStereo, HANDLE_FDK_BITSTREAM hBitstream) { return ( (hParametricStereo != NULL) ? FDKsbrEnc_WritePSBitstream(&hParametricStereo->psOut[0], hBitstream) : 0); } FDK_PSENC_ERROR FDKsbrEnc_PSEnc_ParametricStereoProcessing( HANDLE_PARAMETRIC_STEREO hParametricStereo, INT_PCM *samples[2], UINT samplesBufSize, QMF_FILTER_BANK **hQmfAnalysis, FIXP_DBL **RESTRICT downmixedRealQmfData, FIXP_DBL **RESTRICT downmixedImagQmfData, INT_PCM *downsampledOutSignal, HANDLE_QMF_FILTER_BANK sbrSynthQmf, SCHAR *qmfScale, const int sendHeader) { FDK_PSENC_ERROR error = PSENC_OK; INT psQmfScale[MAX_PS_CHANNELS] = {0}; int psCh, i; C_AALLOC_SCRATCH_START(pWorkBuffer, FIXP_DBL, 4 * 64) for (psCh = 0; psCh < MAX_PS_CHANNELS; psCh++) { for (i = 0; i < hQmfAnalysis[psCh]->no_col; i++) { qmfAnalysisFilteringSlot( hQmfAnalysis[psCh], &pWorkBuffer[2 * 64], /* qmfReal[64] */ &pWorkBuffer[3 * 64], /* qmfImag[64] */ samples[psCh] + i * hQmfAnalysis[psCh]->no_channels, 1, &pWorkBuffer[0 * 64] /* qmf workbuffer 2*64 */ ); FDKhybridAnalysisApply( &hParametricStereo->fdkHybAnaFilter[psCh], &pWorkBuffer[2 * 64], /* qmfReal[64] */ &pWorkBuffer[3 * 64], /* qmfImag[64] */ hParametricStereo->pHybridData[i + HYBRID_READ_OFFSET][psCh][0], hParametricStereo->pHybridData[i + HYBRID_READ_OFFSET][psCh][1]); } /* no_col loop i */ psQmfScale[psCh] = hQmfAnalysis[psCh]->outScalefactor; } /* for psCh */ C_AALLOC_SCRATCH_END(pWorkBuffer, FIXP_DBL, 4 * 64) /* find best scaling in new QMF and Hybrid data */ psFindBestScaling( hParametricStereo, &hParametricStereo->pHybridData[HYBRID_READ_OFFSET], hParametricStereo->dynBandScale, hParametricStereo->maxBandValue, &hParametricStereo->dmxScale); /* extract the ps parameters */ if (PSENC_OK != (error = ExtractPSParameters(hParametricStereo, sendHeader, &hParametricStereo->pHybridData[0]))) { goto bail; } /* save hybrid date for next frame */ for (i = 0; i < HYBRID_READ_OFFSET; i++) { FDKmemcpy( hParametricStereo->pHybridData[i][0][0], hParametricStereo->pHybridData[hParametricStereo->noQmfSlots + i][0][0], MAX_HYBRID_BANDS * sizeof(FIXP_DBL)); /* left, real */ FDKmemcpy( hParametricStereo->pHybridData[i][0][1], hParametricStereo->pHybridData[hParametricStereo->noQmfSlots + i][0][1], MAX_HYBRID_BANDS * sizeof(FIXP_DBL)); /* left, imag */ FDKmemcpy( hParametricStereo->pHybridData[i][1][0], hParametricStereo->pHybridData[hParametricStereo->noQmfSlots + i][1][0], MAX_HYBRID_BANDS * sizeof(FIXP_DBL)); /* right, real */ FDKmemcpy( hParametricStereo->pHybridData[i][1][1], hParametricStereo->pHybridData[hParametricStereo->noQmfSlots + i][1][1], MAX_HYBRID_BANDS * sizeof(FIXP_DBL)); /* right, imag */ } /* downmix and hybrid synthesis */ if (PSENC_OK != (error = DownmixPSQmfData( hParametricStereo, sbrSynthQmf, downmixedRealQmfData, downmixedImagQmfData, downsampledOutSignal, samplesBufSize, &hParametricStereo->pHybridData[HYBRID_READ_OFFSET], hParametricStereo->noQmfSlots, psQmfScale, qmfScale))) { goto bail; } bail: return error; } static void psFindBestScaling( HANDLE_PARAMETRIC_STEREO hParametricStereo, FIXP_DBL *hybridData[HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2], UCHAR *dynBandScale, FIXP_DBL *maxBandValue, SCHAR *dmxScale) { HANDLE_PS_ENCODE hPsEncode = hParametricStereo->hPsEncode; INT group, bin, col, band; const INT frameSize = hParametricStereo->noQmfSlots; const INT psBands = (INT)hPsEncode->psEncMode; const INT nIidGroups = hPsEncode->nQmfIidGroups + hPsEncode->nSubQmfIidGroups; /* group wise scaling */ FIXP_DBL maxVal[2][PS_MAX_BANDS]; FIXP_DBL maxValue = FL2FXCONST_DBL(0.f); FDKmemclear(maxVal, sizeof(maxVal)); /* start with hybrid data */ for (group = 0; group < nIidGroups; group++) { /* Translate group to bin */ bin = hPsEncode->subband2parameterIndex[group]; /* Translate from 20 bins to 10 bins */ if (hPsEncode->psEncMode == PS_BANDS_COARSE) { bin >>= 1; } /* QMF downmix scaling */ for (col = 0; col < frameSize; col++) { int i, section = (col < frameSize - HYBRID_READ_OFFSET) ? 0 : 1; FIXP_DBL tmp = maxVal[section][bin]; for (i = hPsEncode->iidGroupBorders[group]; i < hPsEncode->iidGroupBorders[group + 1]; i++) { tmp = fixMax(tmp, (FIXP_DBL)fixp_abs(hybridData[col][0][0][i])); tmp = fixMax(tmp, (FIXP_DBL)fixp_abs(hybridData[col][0][1][i])); tmp = fixMax(tmp, (FIXP_DBL)fixp_abs(hybridData[col][1][0][i])); tmp = fixMax(tmp, (FIXP_DBL)fixp_abs(hybridData[col][1][1][i])); } maxVal[section][bin] = tmp; } } /* nIidGroups */ /* convert maxSpec to maxScaling, find scaling space */ for (band = 0; band < psBands; band++) { #ifndef MULT_16x16 dynBandScale[band] = CountLeadingBits(fixMax(maxVal[0][band], maxBandValue[band])); #else dynBandScale[band] = fixMax( 0, CountLeadingBits(fixMax(maxVal[0][band], maxBandValue[band])) - FRACT_BITS); #endif maxValue = fixMax(maxValue, fixMax(maxVal[0][band], maxVal[1][band])); maxBandValue[band] = fixMax(maxVal[0][band], maxVal[1][band]); } /* calculate maximal scaling for QMF downmix */ #ifndef MULT_16x16 *dmxScale = fixMin(DFRACT_BITS, CountLeadingBits(maxValue)); #else *dmxScale = fixMax(0, fixMin(FRACT_BITS, CountLeadingBits((maxValue)))); #endif } fdk-aac-2.0.2/libSBRenc/src/ps_main.h000066400000000000000000000243011404225310700171700ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Markus Multrus Description: PS Wrapper, Downmix header file *******************************************************************************/ #ifndef PS_MAIN_H #define PS_MAIN_H /* Includes ******************************************************************/ #include "sbr_def.h" #include "qmf.h" #include "ps_encode.h" #include "FDK_bitstream.h" #include "FDK_hybrid.h" /* Data Types ****************************************************************/ typedef enum { PSENC_STEREO_BANDS_INVALID = 0, PSENC_STEREO_BANDS_10 = 10, PSENC_STEREO_BANDS_20 = 20 } PSENC_STEREO_BANDS_CONFIG; typedef enum { PSENC_NENV_1 = 1, PSENC_NENV_2 = 2, PSENC_NENV_4 = 4, PSENC_NENV_DEFAULT = PSENC_NENV_2, PSENC_NENV_MAX = PSENC_NENV_4 } PSENC_NENV_CONFIG; typedef struct { UINT bitrateFrom; /* inclusive */ UINT bitrateTo; /* exclusive */ PSENC_STEREO_BANDS_CONFIG nStereoBands; PSENC_NENV_CONFIG nEnvelopes; LONG iidQuantErrorThreshold; /* quantization threshold to switch between coarse and fine iid quantization */ } psTuningTable_t; /* Function / Class Declarations *********************************************/ typedef struct T_PARAMETRIC_STEREO { HANDLE_PS_ENCODE hPsEncode; PS_OUT psOut[2]; FIXP_DBL __staticHybridData[HYBRID_READ_OFFSET][MAX_PS_CHANNELS][2] [MAX_HYBRID_BANDS]; FIXP_DBL *pHybridData[HYBRID_READ_OFFSET + HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2]; FIXP_DBL qmfDelayLines[2][32 >> 1][64]; int qmfDelayScale; INT psDelay; UINT maxEnvelopes; UCHAR dynBandScale[PS_MAX_BANDS]; FIXP_DBL maxBandValue[PS_MAX_BANDS]; SCHAR dmxScale; INT initPS; INT noQmfSlots; INT noQmfBands; FIXP_DBL __staticHybAnaStatesLF[MAX_PS_CHANNELS][2 * HYBRID_FILTER_LENGTH * HYBRID_MAX_QMF_BANDS]; FIXP_DBL __staticHybAnaStatesHF[MAX_PS_CHANNELS][2 * HYBRID_FILTER_DELAY * (64 - HYBRID_MAX_QMF_BANDS)]; FDK_ANA_HYB_FILTER fdkHybAnaFilter[MAX_PS_CHANNELS]; FDK_SYN_HYB_FILTER fdkHybSynFilter; } PARAMETRIC_STEREO; typedef struct T_PSENC_CONFIG { INT frameSize; INT qmfFilterMode; INT sbrPsDelay; PSENC_STEREO_BANDS_CONFIG nStereoBands; PSENC_NENV_CONFIG maxEnvelopes; FIXP_DBL iidQuantErrorThreshold; } PSENC_CONFIG, *HANDLE_PSENC_CONFIG; typedef struct T_PARAMETRIC_STEREO *HANDLE_PARAMETRIC_STEREO; /** * \brief Create a parametric stereo encoder instance. * * \param phParametricStereo A pointer to a parametric stereo handle to be * allocated. Initialized on return. * * \return * - PSENC_OK, on succes. * - PSENC_INVALID_HANDLE, PSENC_MEMORY_ERROR, on failure. */ FDK_PSENC_ERROR PSEnc_Create(HANDLE_PARAMETRIC_STEREO *phParametricStereo); /** * \brief Initialize a parametric stereo encoder instance. * * \param hParametricStereo Meta Data handle. * \param hPsEncConfig Filled parametric stereo configuration * structure. * \param noQmfSlots Number of slots within one audio frame. * \param noQmfBands Number of QMF bands. * \param dynamic_RAM Pointer to preallocated workbuffer. * * \return * - PSENC_OK, on succes. * - PSENC_INVALID_HANDLE, PSENC_INIT_ERROR, on failure. */ FDK_PSENC_ERROR PSEnc_Init(HANDLE_PARAMETRIC_STEREO hParametricStereo, const HANDLE_PSENC_CONFIG hPsEncConfig, INT noQmfSlots, INT noQmfBands, UCHAR *dynamic_RAM); /** * \brief Destroy parametric stereo encoder instance. * * Deallocate instance and free whole memory. * * \param phParametricStereo Pointer to the parametric stereo handle to be * deallocated. * * \return * - PSENC_OK, on succes. * - PSENC_INVALID_HANDLE, on failure. */ FDK_PSENC_ERROR PSEnc_Destroy(HANDLE_PARAMETRIC_STEREO *phParametricStereo); /** * \brief Apply parametric stereo processing. * * \param hParametricStereo Meta Data handle. * \param samples Pointer to 2 channel audio input signal. * \param timeInStride, Stride factor of input buffer. * \param hQmfAnalysis, Pointer to QMF analysis filterbanks. * \param downmixedRealQmfData Pointer to real QMF buffer to be written to. * \param downmixedImagQmfData Pointer to imag QMF buffer to be written to. * \param downsampledOutSignal Pointer to buffer where to write downmixed * timesignal. * \param sbrSynthQmf Pointer to QMF synthesis filterbank. * \param qmfScale Return scaling factor of the qmf data. * \param sendHeader Signal whether to write header data. * * \return * - PSENC_OK, on succes. * - PSENC_INVALID_HANDLE, PSENC_ENCODE_ERROR, on failure. */ FDK_PSENC_ERROR FDKsbrEnc_PSEnc_ParametricStereoProcessing( HANDLE_PARAMETRIC_STEREO hParametricStereo, INT_PCM *samples[2], UINT timeInStride, QMF_FILTER_BANK **hQmfAnalysis, FIXP_DBL **RESTRICT downmixedRealQmfData, FIXP_DBL **RESTRICT downmixedImagQmfData, INT_PCM *downsampledOutSignal, HANDLE_QMF_FILTER_BANK sbrSynthQmf, SCHAR *qmfScale, const int sendHeader); /** * \brief Write parametric stereo bitstream. * * Write ps_data() element to bitstream and return number of written bits. * Returns number of written bits only, if hBitstream == NULL. * * \param hParametricStereo Meta Data handle. * \param hBitstream Bitstream buffer handle. * * \return * - number of written bits. */ INT FDKsbrEnc_PSEnc_WritePSData(HANDLE_PARAMETRIC_STEREO hParametricStereo, HANDLE_FDK_BITSTREAM hBitstream); #endif /* PS_MAIN_H */ fdk-aac-2.0.2/libSBRenc/src/resampler.cpp000066400000000000000000000413061404225310700200730ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief FDK resampler tool box:$Revision: 91655 $ \author M. Werner */ #include "resampler.h" #include "genericStds.h" /**************************************************************************/ /* BIQUAD Filter Specifications */ /**************************************************************************/ #define B1 0 #define B2 1 #define A1 2 #define A2 3 #define BQC(x) FL2FXCONST_SGL(x / 2) struct FILTER_PARAM { const FIXP_SGL *coeffa; /*! SOS matrix One row/section. Scaled using BQC(). Order of coefficients: B1,B2,A1,A2. B0=A0=1.0 */ FIXP_DBL g; /*! overall gain */ int Wc; /*! normalized passband bandwidth at input samplerate * 1000 */ int noCoeffs; /*! number of filter coeffs */ int delay; /*! delay in samples at input samplerate */ }; #define BIQUAD_COEFSTEP 4 /** *\brief Low Pass Wc = 0,5, order 30, Stop Band -96dB. Wc criteria is "almost 0dB passband", not the usual -3db gain point. [b,a]=cheby2(30,96,0.505) [sos,g]=tf2sos(b,a) bandwidth 0.48 */ static const FIXP_SGL sos48[] = { BQC(1.98941075681938), BQC(0.999999996890811), BQC(0.863264527201963), BQC(0.189553799960663), BQC(1.90733804822445), BQC(1.00000001736189), BQC(0.836321575841691), BQC(0.203505809266564), BQC(1.75616665495325), BQC(0.999999946079721), BQC(0.784699225121588), BQC(0.230471265506986), BQC(1.55727745512726), BQC(1.00000011737815), BQC(0.712515423588351), BQC(0.268752723900498), BQC(1.33407591943643), BQC(0.999999795953228), BQC(0.625059117330989), BQC(0.316194685288965), BQC(1.10689898412458), BQC(1.00000035057114), BQC(0.52803514366398), BQC(0.370517843224669), BQC(0.89060371078454), BQC(0.999999343962822), BQC(0.426920462165257), BQC(0.429608200207746), BQC(0.694438261209433), BQC(1.0000008629792), BQC(0.326530699561716), BQC(0.491714450654174), BQC(0.523237800935322), BQC(1.00000101349782), BQC(0.230829556274851), BQC(0.555559034843281), BQC(0.378631165929563), BQC(0.99998986482665), BQC(0.142906422036095), BQC(0.620338874442411), BQC(0.260786911308437), BQC(1.00003261460178), BQC(0.0651008576256505), BQC(0.685759923926262), BQC(0.168409429188098), BQC(0.999933049695828), BQC(-0.000790067789975562), BQC(0.751905896602325), BQC(0.100724533818628), BQC(1.00009472669872), BQC(-0.0533772830257041), BQC(0.81930744384525), BQC(0.0561434357867363), BQC(0.999911636304276), BQC(-0.0913550299236405), BQC(0.88883625875915), BQC(0.0341680678662057), BQC(1.00003667508676), BQC(-0.113405185536697), BQC(0.961756638268446)}; static const FIXP_DBL g48 = FL2FXCONST_DBL(0.002712866530047) - (FIXP_DBL)0x8000; static const struct FILTER_PARAM param_set48 = { sos48, g48, 480, 15, 4 /* LF 2 */ }; /** *\brief Low Pass Wc = 0,5, order 24, Stop Band -96dB. Wc criteria is "almost 0dB passband", not the usual -3db gain point. [b,a]=cheby2(24,96,0.5) [sos,g]=tf2sos(b,a) bandwidth 0.45 */ static const FIXP_SGL sos45[] = { BQC(1.982962601444), BQC(1.00000000007504), BQC(0.646113303737836), BQC(0.10851149979981), BQC(1.85334094281111), BQC(0.999999999677192), BQC(0.612073220102006), BQC(0.130022141698044), BQC(1.62541051415425), BQC(1.00000000080398), BQC(0.547879702855959), BQC(0.171165825133192), BQC(1.34554656923247), BQC(0.9999999980169), BQC(0.460373914508491), BQC(0.228677463376354), BQC(1.05656568503116), BQC(1.00000000569363), BQC(0.357891894038287), BQC(0.298676843912185), BQC(0.787967587877312), BQC(0.999999984415017), BQC(0.248826893211877), BQC(0.377441803512978), BQC(0.555480971120497), BQC(1.00000003583307), BQC(0.140614263345315), BQC(0.461979302213679), BQC(0.364986207070964), BQC(0.999999932084303), BQC(0.0392669446074516), BQC(0.55033451180825), BQC(0.216827267631558), BQC(1.00000010534682), BQC(-0.0506232228865103), BQC(0.641691581560946), BQC(0.108951672277119), BQC(0.999999871167516), BQC(-0.125584840183225), BQC(0.736367748771803), BQC(0.0387988607229035), BQC(1.00000011205574), BQC(-0.182814849097974), BQC(0.835802108714964), BQC(0.0042866175809225), BQC(0.999999954830813), BQC(-0.21965740617151), BQC(0.942623047782363)}; static const FIXP_DBL g45 = FL2FXCONST_DBL(0.00242743980909524) - (FIXP_DBL)0x8000; static const struct FILTER_PARAM param_set45 = { sos45, g45, 450, 12, 4 /* LF 2 */ }; /* Created by Octave 2.1.73, Mon Oct 13 17:31:32 2008 CEST Wc = 0,5, order 16, Stop Band -96dB damping. [b,a]=cheby2(16,96,0.5) [sos,g]=tf2sos(b,a) bandwidth = 0.41 */ static const FIXP_SGL sos41[] = { BQC(1.96193625292), BQC(0.999999999999964), BQC(0.169266178786789), BQC(0.0128823300475907), BQC(1.68913437662092), BQC(1.00000000000053), BQC(0.124751503206552), BQC(0.0537472273950989), BQC(1.27274692366017), BQC(0.999999999995674), BQC(0.0433108625178357), BQC(0.131015753236317), BQC(0.85214175088395), BQC(1.00000000001813), BQC(-0.0625658152550408), BQC(0.237763778993806), BQC(0.503841579939009), BQC(0.999999999953223), BQC(-0.179176128722865), BQC(0.367475236424474), BQC(0.249990711986162), BQC(1.00000000007952), BQC(-0.294425165824676), BQC(0.516594857170212), BQC(0.087971668680286), BQC(0.999999999915528), BQC(-0.398956566777928), BQC(0.686417767801123), BQC(0.00965373325350294), BQC(1.00000000003744), BQC(-0.48579173764817), BQC(0.884931534239068)}; static const FIXP_DBL g41 = FL2FXCONST_DBL(0.00155956951169248); static const struct FILTER_PARAM param_set41 = { sos41, g41, 410, 8, 5 /* LF 3 */ }; /* # Created by Octave 2.1.73, Mon Oct 13 17:55:33 2008 CEST Wc = 0,5, order 12, Stop Band -96dB damping. [b,a]=cheby2(12,96,0.5); [sos,g]=tf2sos(b,a) */ static const FIXP_SGL sos35[] = { BQC(1.93299325235762), BQC(0.999999999999985), BQC(-0.140733187246596), BQC(0.0124139497836062), BQC(1.4890416764109), BQC(1.00000000000011), BQC(-0.198215402588504), BQC(0.0746730616584138), BQC(0.918450161309795), BQC(0.999999999999619), BQC(-0.30133912791941), BQC(0.192276468839529), BQC(0.454877024246818), BQC(1.00000000000086), BQC(-0.432337328809815), BQC(0.356852933642815), BQC(0.158017147118507), BQC(0.999999999998876), BQC(-0.574817494249777), BQC(0.566380436970833), BQC(0.0171834649478749), BQC(1.00000000000055), BQC(-0.718581178041165), BQC(0.83367484487889)}; static const FIXP_DBL g35 = FL2FXCONST_DBL(0.00162580994125131); static const struct FILTER_PARAM param_set35 = {sos35, g35, 350, 6, 4}; /* # Created by Octave 2.1.73, Mon Oct 13 18:15:38 2008 CEST Wc = 0,5, order 8, Stop Band -96dB damping. [b,a]=cheby2(8,96,0.5); [sos,g]=tf2sos(b,a) */ static const FIXP_SGL sos25[] = { BQC(1.85334094301225), BQC(1.0), BQC(-0.702127214212663), BQC(0.132452403998767), BQC(1.056565682167), BQC(0.999999999999997), BQC(-0.789503667880785), BQC(0.236328693569128), BQC(0.364986307455489), BQC(0.999999999999996), BQC(-0.955191189843375), BQC(0.442966457936379), BQC(0.0387985751642125), BQC(1.0), BQC(-1.19817786088084), BQC(0.770493895456328)}; static const FIXP_DBL g25 = FL2FXCONST_DBL(0.000945182835294559); static const struct FILTER_PARAM param_set25 = {sos25, g25, 250, 4, 5}; /* Must be sorted in descending order */ static const struct FILTER_PARAM *const filter_paramSet[] = { ¶m_set48, ¶m_set45, ¶m_set41, ¶m_set35, ¶m_set25}; /**************************************************************************/ /* Resampler Functions */ /**************************************************************************/ /*! \brief Reset downsampler instance and clear delay lines \return success of operation */ INT FDKaacEnc_InitDownsampler( DOWNSAMPLER *DownSampler, /*!< pointer to downsampler instance */ int Wc, /*!< normalized cutoff freq * 1000* */ int ratio) /*!< downsampler ratio */ { UINT i; const struct FILTER_PARAM *currentSet = NULL; FDKmemclear(DownSampler->downFilter.states, sizeof(DownSampler->downFilter.states)); DownSampler->downFilter.ptr = 0; /* find applicable parameter set */ currentSet = filter_paramSet[0]; for (i = 1; i < sizeof(filter_paramSet) / sizeof(struct FILTER_PARAM *); i++) { if (filter_paramSet[i]->Wc <= Wc) { break; } currentSet = filter_paramSet[i]; } DownSampler->downFilter.coeffa = currentSet->coeffa; DownSampler->downFilter.gain = currentSet->g; FDK_ASSERT(currentSet->noCoeffs <= MAXNR_SECTIONS * 2); DownSampler->downFilter.noCoeffs = currentSet->noCoeffs; DownSampler->delay = currentSet->delay; DownSampler->downFilter.Wc = currentSet->Wc; DownSampler->ratio = ratio; DownSampler->pending = ratio - 1; return (1); } /*! \brief faster simple folding operation Filter: H(z) = A(z)/B(z) with A(z) = a[0]*z^0 + a[1]*z^1 + a[2]*z^2 ... a[n]*z^n \return filtered value */ static inline INT_PCM AdvanceFilter( LP_FILTER *downFilter, /*!< pointer to iir filter instance */ INT_PCM *pInput, /*!< input of filter */ int downRatio) { INT_PCM output; int i, n; #define BIQUAD_SCALE 12 FIXP_DBL y = FL2FXCONST_DBL(0.0f); FIXP_DBL input; for (n = 0; n < downRatio; n++) { FIXP_BQS(*states)[2] = downFilter->states; const FIXP_SGL *coeff = downFilter->coeffa; int s1, s2; s1 = downFilter->ptr; s2 = s1 ^ 1; #if (SAMPLE_BITS == 16) input = ((FIXP_DBL)pInput[n]) << (DFRACT_BITS - SAMPLE_BITS - BIQUAD_SCALE); #elif (SAMPLE_BITS == 32) input = pInput[n] >> BIQUAD_SCALE; #else #error NOT IMPLEMENTED #endif FIXP_BQS state1, state2, state1b, state2b; state1 = states[0][s1]; state2 = states[0][s2]; /* Loop over sections */ for (i = 0; i < downFilter->noCoeffs; i++) { FIXP_DBL state0; /* Load merged states (from next section) */ state1b = states[i + 1][s1]; state2b = states[i + 1][s2]; state0 = input + fMult(state1, coeff[B1]) + fMult(state2, coeff[B2]); y = state0 - fMult(state1b, coeff[A1]) - fMult(state2b, coeff[A2]); /* Store new feed forward merge state */ states[i + 1][s2] = y << 1; /* Store new feed backward state */ states[i][s2] = input << 1; /* Feedback output to next section. */ input = y; /* Transfer merged states */ state1 = state1b; state2 = state2b; /* Step to next coef set */ coeff += BIQUAD_COEFSTEP; } downFilter->ptr ^= 1; } /* Apply global gain */ y = fMult(y, downFilter->gain); /* Apply final gain/scaling to output */ #if (SAMPLE_BITS == 16) output = (INT_PCM)SATURATE_RIGHT_SHIFT( y + (FIXP_DBL)(1 << (DFRACT_BITS - SAMPLE_BITS - BIQUAD_SCALE - 1)), DFRACT_BITS - SAMPLE_BITS - BIQUAD_SCALE, SAMPLE_BITS); // output = (INT_PCM) SATURATE_RIGHT_SHIFT(y, // DFRACT_BITS-SAMPLE_BITS-BIQUAD_SCALE, SAMPLE_BITS); #else output = SATURATE_LEFT_SHIFT(y, BIQUAD_SCALE, SAMPLE_BITS); #endif return output; } /*! \brief FDKaacEnc_Downsample numInSamples of type INT_PCM Returns number of output samples in numOutSamples \return success of operation */ INT FDKaacEnc_Downsample( DOWNSAMPLER *DownSampler, /*!< pointer to downsampler instance */ INT_PCM *inSamples, /*!< pointer to input samples */ INT numInSamples, /*!< number of input samples */ INT_PCM *outSamples, /*!< pointer to output samples */ INT *numOutSamples /*!< pointer tp number of output samples */ ) { INT i; *numOutSamples = 0; for (i = 0; i < numInSamples; i += DownSampler->ratio) { *outSamples = AdvanceFilter(&(DownSampler->downFilter), &inSamples[i], DownSampler->ratio); outSamples++; } *numOutSamples = numInSamples / DownSampler->ratio; return 0; } fdk-aac-2.0.2/libSBRenc/src/resampler.h000066400000000000000000000155071404225310700175440ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Description: *******************************************************************************/ #ifndef RESAMPLER_H #define RESAMPLER_H /*! \file \brief Fixed Point Resampler Tool Box $Revision: 92790 $ */ #include "common_fix.h" /**************************************************************************/ /* BIQUAD Filter Structure */ /**************************************************************************/ #define MAXNR_SECTIONS (15) typedef FIXP_DBL FIXP_BQS; typedef struct { FIXP_BQS states[MAXNR_SECTIONS + 1][2]; /*! state buffer */ const FIXP_SGL *coeffa; /*! pointer to filter coeffs */ FIXP_DBL gain; /*! overall gain factor */ int Wc; /*! normalized cutoff freq * 1000 */ int noCoeffs; /*! number of filter coeffs sets */ int ptr; /*! index to rinbuffers */ } LP_FILTER; /**************************************************************************/ /* Downsampler Structure */ /**************************************************************************/ typedef struct { LP_FILTER downFilter; /*! filter instance */ int ratio; /*! downsampling ration */ int delay; /*! downsampling delay (source fs) */ int pending; /*! number of pending output samples */ } DOWNSAMPLER; /** * \brief Initialized a given downsampler structure. */ INT FDKaacEnc_InitDownsampler( DOWNSAMPLER *DownSampler, /*!< pointer to downsampler instance */ INT Wc, /*!< normalized cutoff freq * 1000 */ INT ratio); /*!< downsampler ratio */ /** * \brief Downsample a set of audio samples. numInSamples must be at least equal * to the downsampler ratio. */ INT FDKaacEnc_Downsample( DOWNSAMPLER *DownSampler, /*!< pointer to downsampler instance */ INT_PCM *inSamples, /*!< pointer to input samples */ INT numInSamples, /*!< number of input samples */ INT_PCM *outSamples, /*!< pointer to output samples */ INT *numOutSamples); /*!< pointer tp number of output samples */ #endif /* RESAMPLER_H */ fdk-aac-2.0.2/libSBRenc/src/sbr.h000066400000000000000000000170601404225310700163340ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Main SBR structs definitions $Revision: 92790 $ */ #ifndef SBR_H #define SBR_H #include "fram_gen.h" #include "bit_sbr.h" #include "tran_det.h" #include "code_env.h" #include "env_est.h" #include "cmondata.h" #include "qmf.h" #include "resampler.h" #include "ton_corr.h" /* SBR bitstream delay */ #define MAX_DELAY_FRAMES 2 /* sbr encoder downsampling type */ typedef enum { SBRENC_DS_NONE, SBRENC_DS_TIME, SBRENC_DS_QMF } SBRENC_DS_TYPE; typedef struct SBR_CHANNEL { struct ENV_CHANNEL hEnvChannel; // INT_PCM *pDSOutBuffer; /**< Pointer to // downsampled audio output of SBR encoder */ DOWNSAMPLER downSampler; } SBR_CHANNEL; typedef SBR_CHANNEL* HANDLE_SBR_CHANNEL; typedef struct SBR_ELEMENT { HANDLE_SBR_CHANNEL sbrChannel[2]; QMF_FILTER_BANK* hQmfAnalysis[2]; SBR_CONFIG_DATA sbrConfigData; SBR_HEADER_DATA sbrHeaderData; SBR_BITSTREAM_DATA sbrBitstreamData; COMMON_DATA CmonData; INT dynXOverFreqDelay[5]; /**< to delay a frame (I don't like it that much that way - hrc) */ SBR_ELEMENT_INFO elInfo; UCHAR payloadDelayLine[1 + MAX_DELAY_FRAMES][MAX_PAYLOAD_SIZE]; UINT payloadDelayLineSize[1 + MAX_DELAY_FRAMES]; /* Sizes in bits */ } SBR_ELEMENT, *HANDLE_SBR_ELEMENT; typedef struct SBR_ENCODER { HANDLE_SBR_ELEMENT sbrElement[(8)]; HANDLE_SBR_CHANNEL pSbrChannel[(8)]; QMF_FILTER_BANK QmfAnalysis[(8)]; DOWNSAMPLER lfeDownSampler; int lfeChIdx; /* -1 default for no lfe, else assign channel index. */ int noElements; /* Number of elements. */ int nChannels; /* Total channel count across all elements. */ int frameSize; /* SBR framelength. */ int bufferOffset; /* Offset for SBR parameter extraction in time domain input buffer. */ int downsampledOffset; /* Offset of downsampled/mixed output for core encoder. */ int downmixSize; /* Size in samples of downsampled/mixed output for core encoder. */ INT downSampleFactor; /* Sampling rate relation between the SBR and the core encoder. */ SBRENC_DS_TYPE downsamplingMethod; /* Method of downsmapling, time-domain, QMF or none. */ int nBitstrDelay; /* Amount of SBR frames to be delayed in bitstream domain. */ int sbrDecDelay; /* SBR decoder delay in samples */ INT estimateBitrate; /* Estimate bitrate of SBR encoder. */ INT inputDataDelay; /* Delay caused by downsampler, in/out buffer at sbrEncoder_EncodeFrame. */ UCHAR* dynamicRam; UCHAR* pSBRdynamic_RAM; HANDLE_PARAMETRIC_STEREO hParametricStereo; QMF_FILTER_BANK qmfSynthesisPS; /* parameters describing allocation volume of present instance */ INT maxElements; INT maxChannels; INT supportPS; } SBR_ENCODER; #endif /* SBR_H */ fdk-aac-2.0.2/libSBRenc/src/sbr_def.h000066400000000000000000000223041404225310700171470ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief SBR main definitions $Revision: 92790 $ */ #ifndef SBR_DEF_H #define SBR_DEF_H #include "common_fix.h" #define noError 0 #define HANDLE_ERROR_INFO INT #define ERROR(a, b) 1 /* #define SBR_ENV_STATISTICS_BITRATE */ #undef SBR_ENV_STATISTICS_BITRATE /* #define SBR_ENV_STATISTICS */ #undef SBR_ENV_STATISTICS /* #define SBR_PAYLOAD_MONITOR */ #undef SBR_PAYLOAD_MONITOR #define SWAP(a, b) tempr = a, a = b, b = tempr #define TRUE 1 #define FALSE 0 /* Constants */ #define EPS 1e-12 #define LOG2 0.69314718056f /* natural logarithm of 2 */ #define ILOG2 1.442695041f /* 1/LOG2 */ #define RELAXATION_FLOAT (1e-6f) #define RELAXATION (FL2FXCONST_DBL(RELAXATION_FLOAT)) #define RELAXATION_FRACT \ (FL2FXCONST_DBL(0.524288f)) /* 0.524288f is fractional part of RELAXATION */ #define RELAXATION_SHIFT (19) #define RELAXATION_LD64 \ (FL2FXCONST_DBL(0.31143075889f)) /* (ld64(RELAXATION) \ */ /************ Definitions ***************/ #define SBR_COMP_MODE_DELTA 0 #define SBR_COMP_MODE_CTS 1 #define SBR_MAX_ENERGY_VALUES 5 #define SBR_GLOBAL_TONALITY_VALUES 2 #define MAX_NUM_CHANNELS 2 #define MAX_NOISE_ENVELOPES 2 #define MAX_NUM_NOISE_COEFFS 5 #define MAX_NUM_NOISE_VALUES (MAX_NUM_NOISE_COEFFS * MAX_NOISE_ENVELOPES) #define MAX_NUM_ENVELOPE_VALUES (MAX_ENVELOPES * MAX_FREQ_COEFFS) #define MAX_ENVELOPES 5 #define MAX_FREQ_COEFFS 48 #define MAX_FREQ_COEFFS_FS44100 35 #define MAX_FREQ_COEFFS_FS48000 32 #define NO_OF_ESTIMATES_LC 4 #define NO_OF_ESTIMATES_LD 3 #define MAX_NO_OF_ESTIMATES 4 #define NOISE_FLOOR_OFFSET 6 #define NOISE_FLOOR_OFFSET_64 (FL2FXCONST_DBL(0.09375f)) #define LOW_RES 0 #define HIGH_RES 1 #define LO 0 #define HI 1 #define LENGTH_SBR_FRAME_INFO 35 /* 19 */ #define SBR_NSFB_LOW_RES 9 /* 8 */ #define SBR_NSFB_HIGH_RES 18 /* 16 */ #define SBR_XPOS_CTRL_DEFAULT 2 #define SBR_FREQ_SCALE_DEFAULT 2 #define SBR_ALTER_SCALE_DEFAULT 1 #define SBR_NOISE_BANDS_DEFAULT 2 #define SBR_LIMITER_BANDS_DEFAULT 2 #define SBR_LIMITER_GAINS_DEFAULT 2 #define SBR_LIMITER_GAINS_INFINITE 3 #define SBR_INTERPOL_FREQ_DEFAULT 1 #define SBR_SMOOTHING_LENGTH_DEFAULT 0 /* sbr_header */ #define SI_SBR_AMP_RES_BITS 1 #define SI_SBR_COUPLING_BITS 1 #define SI_SBR_START_FREQ_BITS 4 #define SI_SBR_STOP_FREQ_BITS 4 #define SI_SBR_XOVER_BAND_BITS 3 #define SI_SBR_RESERVED_BITS 2 #define SI_SBR_DATA_EXTRA_BITS 1 #define SI_SBR_HEADER_EXTRA_1_BITS 1 #define SI_SBR_HEADER_EXTRA_2_BITS 1 /* sbr_header extra 1 */ #define SI_SBR_FREQ_SCALE_BITS 2 #define SI_SBR_ALTER_SCALE_BITS 1 #define SI_SBR_NOISE_BANDS_BITS 2 /* sbr_header extra 2 */ #define SI_SBR_LIMITER_BANDS_BITS 2 #define SI_SBR_LIMITER_GAINS_BITS 2 #define SI_SBR_INTERPOL_FREQ_BITS 1 #define SI_SBR_SMOOTHING_LENGTH_BITS 1 /* sbr_grid */ #define SBR_CLA_BITS 2 /*!< size of bs_frame_class */ #define SBR_CLA_BITS_LD 1 /*!< size of bs_frame_class */ #define SBR_ENV_BITS 2 /*!< size of bs_num_env_raw */ #define SBR_ABS_BITS 2 /*!< size of bs_abs_bord_raw for HE-AAC */ #define SBR_NUM_BITS 2 /*!< size of bs_num_rel */ #define SBR_REL_BITS 2 /*!< size of bs_rel_bord_raw */ #define SBR_RES_BITS 1 /*!< size of bs_freq_res_flag */ #define SBR_DIR_BITS 1 /*!< size of bs_df_flag */ /* sbr_data */ #define SI_SBR_INVF_MODE_BITS 2 #define SI_SBR_START_ENV_BITS_AMP_RES_3_0 6 #define SI_SBR_START_ENV_BITS_BALANCE_AMP_RES_3_0 5 #define SI_SBR_START_NOISE_BITS_AMP_RES_3_0 5 #define SI_SBR_START_NOISE_BITS_BALANCE_AMP_RES_3_0 5 #define SI_SBR_START_ENV_BITS_AMP_RES_1_5 7 #define SI_SBR_START_ENV_BITS_BALANCE_AMP_RES_1_5 6 #define SI_SBR_EXTENDED_DATA_BITS 1 #define SI_SBR_EXTENSION_SIZE_BITS 4 #define SI_SBR_EXTENSION_ESC_COUNT_BITS 8 #define SI_SBR_EXTENSION_ID_BITS 2 #define SBR_EXTENDED_DATA_MAX_CNT (15 + 255) #define EXTENSION_ID_PS_CODING 2 /* Envelope coding constants */ #define FREQ 0 #define TIME 1 /* qmf data scaling */ #define QMF_SCALE_OFFSET 7 /* huffman tables */ #define CODE_BOOK_SCF_LAV00 60 #define CODE_BOOK_SCF_LAV01 31 #define CODE_BOOK_SCF_LAV10 60 #define CODE_BOOK_SCF_LAV11 31 #define CODE_BOOK_SCF_LAV_BALANCE11 12 #define CODE_BOOK_SCF_LAV_BALANCE10 24 typedef enum { SBR_AMP_RES_1_5 = 0, SBR_AMP_RES_3_0 } AMP_RES; typedef enum { XPOS_MDCT, XPOS_MDCT_CROSS, XPOS_LC, XPOS_RESERVED, XPOS_SWITCHED /* not a real choice but used here to control behaviour */ } XPOS_MODE; typedef enum { INVF_OFF = 0, INVF_LOW_LEVEL, INVF_MID_LEVEL, INVF_HIGH_LEVEL, INVF_SWITCHED /* not a real choice but used here to control behaviour */ } INVF_MODE; #endif fdk-aac-2.0.2/libSBRenc/src/sbr_encoder.cpp000066400000000000000000002647241404225310700204010ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Andreas Ehret, Tobias Chalupka Description: SBR encoder top level processing. *******************************************************************************/ #include "sbr_encoder.h" #include "sbrenc_ram.h" #include "sbrenc_rom.h" #include "sbrenc_freq_sca.h" #include "env_bit.h" #include "cmondata.h" #include "sbr_misc.h" #include "sbr.h" #include "qmf.h" #include "ps_main.h" #define SBRENCODER_LIB_VL0 4 #define SBRENCODER_LIB_VL1 0 #define SBRENCODER_LIB_VL2 0 /***************************************************************************/ /* * SBR Delay balancing definitions. */ /* input buffer (1ch) |------------ 1537 -------------|-----|---------- 2048 -------------| (core2sbr delay ) ds (read, core and ds area) */ #define SFB(dwnsmp) \ (32 << (dwnsmp - \ 1)) /* SBR Frequency bands: 64 for dual-rate, 32 for single-rate */ #define STS(fl) \ (((fl) == 1024) ? 32 \ : 30) /* SBR Time Slots: 32 for core frame length 1024, 30 \ for core frame length 960 */ #define DELAY_QMF_ANA(dwnsmp) \ ((320 << ((dwnsmp)-1)) - (32 << ((dwnsmp)-1))) /* Full bandwidth */ #define DELAY_HYB_ANA (10 * 64) /* + 0.5 */ /* */ #define DELAY_HYB_SYN (6 * 64 - 32) /* */ #define DELAY_QMF_POSTPROC(dwnsmp) \ (32 * (dwnsmp)) /* QMF postprocessing delay */ #define DELAY_DEC_QMF(dwnsmp) (6 * SFB(dwnsmp)) /* Decoder QMF overlap */ #define DELAY_QMF_SYN(dwnsmp) \ (1 << (dwnsmp - \ 1)) /* QMF_NO_POLY/2=2.5, rounded down to 2, half for single-rate */ #define DELAY_QMF_DS (32) /* QMF synthesis for downsampled time signal */ /* Delay in QMF paths */ #define DELAY_SBR(fl, dwnsmp) \ (DELAY_QMF_ANA(dwnsmp) + (SFB(dwnsmp) * STS(fl) - 1) + DELAY_QMF_SYN(dwnsmp)) #define DELAY_PS(fl, dwnsmp) \ (DELAY_QMF_ANA(dwnsmp) + DELAY_HYB_ANA + DELAY_DEC_QMF(dwnsmp) + \ (SFB(dwnsmp) * STS(fl) - 1) + DELAY_HYB_SYN + DELAY_QMF_SYN(dwnsmp)) #define DELAY_ELDSBR(fl, dwnsmp) \ ((((fl) / 2) * (dwnsmp)) - 1 + DELAY_QMF_POSTPROC(dwnsmp)) #define DELAY_ELDv2SBR(fl, dwnsmp) \ ((((fl) / 2) * (dwnsmp)) - 1 + 80 * (dwnsmp)) /* 80 is the delay caused \ by the sum of the CLD \ analysis and the MPSLD \ synthesis filterbank */ /* Delay in core path (core and downsampler not taken into account) */ #define DELAY_COREPATH_SBR(fl, dwnsmp) \ ((DELAY_QMF_ANA(dwnsmp) + DELAY_DEC_QMF(dwnsmp) + DELAY_QMF_SYN(dwnsmp))) #define DELAY_COREPATH_ELDSBR(fl, dwnsmp) ((DELAY_QMF_POSTPROC(dwnsmp))) #define DELAY_COREPATH_ELDv2SBR(fl, dwnsmp) (128 * (dwnsmp)) /* 4 slots */ #define DELAY_COREPATH_PS(fl, dwnsmp) \ ((DELAY_QMF_ANA(dwnsmp) + DELAY_QMF_DS + \ /*(DELAY_AAC(fl)*2) + */ DELAY_QMF_ANA(dwnsmp) + DELAY_DEC_QMF(dwnsmp) + \ DELAY_HYB_SYN + DELAY_QMF_SYN(dwnsmp))) /* 2048 - 463*2 */ /* Delay differences between SBR- and downsampled path for SBR and SBR+PS */ #define DELAY_AAC2SBR(fl, dwnsmp) \ ((DELAY_COREPATH_SBR(fl, dwnsmp)) - DELAY_SBR((fl), (dwnsmp))) #define DELAY_ELD2SBR(fl, dwnsmp) \ ((DELAY_COREPATH_ELDSBR(fl, dwnsmp)) - DELAY_ELDSBR(fl, dwnsmp)) #define DELAY_AAC2PS(fl, dwnsmp) \ ((DELAY_COREPATH_PS(fl, dwnsmp)) - DELAY_PS(fl, dwnsmp)) /* 2048 - 463*2 */ /* Assumption: The sample delay resulting of of DELAY_AAC2PS is always smaller * than the sample delay implied by DELAY_AAC2SBR */ #define MAX_DS_FILTER_DELAY \ (5) /* the additional max downsampler filter delay (source fs) */ #define MAX_SAMPLE_DELAY \ (DELAY_AAC2SBR(1024, 2) + MAX_DS_FILTER_DELAY) /* maximum delay: frame \ length of 1024 and \ dual-rate sbr */ /***************************************************************************/ /*************** Delay parameters for sbrEncoder_Init_delay() **************/ typedef struct { int dsDelay; /* the delay of the (time-domain) downsampler itself */ int delay; /* overall delay / samples */ int sbrDecDelay; /* SBR decoder's delay */ int corePathOffset; /* core path offset / samples; added by sbrEncoder_Init_delay() */ int sbrPathOffset; /* SBR path offset / samples; added by sbrEncoder_Init_delay() */ int bitstrDelay; /* bitstream delay / frames; added by sbrEncoder_Init_delay() */ int delayInput2Core; /* delay of the input to the core / samples */ } DELAY_PARAM; /***************************************************************************/ #define INVALID_TABLE_IDX -1 /***************************************************************************/ /*! \brief Selects the SBR tuning settings to use dependent on number of channels, bitrate, sample rate and core coder \return Index to the appropriate table ****************************************************************************/ #define DISTANCE_CEIL_VALUE 5000000 static INT getSbrTuningTableIndex( UINT bitrate, /*! the total bitrate in bits/sec */ UINT numChannels, /*! the number of channels for the core coder */ UINT sampleRate, /*! the sampling rate of the core coder */ AUDIO_OBJECT_TYPE core, UINT *pBitRateClosest) { int i, bitRateClosestLowerIndex = -1, bitRateClosestUpperIndex = -1, found = 0; UINT bitRateClosestUpper = 0, bitRateClosestLower = DISTANCE_CEIL_VALUE; #define isForThisCore(i) \ ((sbrTuningTable[i].coreCoder == CODEC_AACLD && core == AOT_ER_AAC_ELD) || \ (sbrTuningTable[i].coreCoder == CODEC_AAC && core != AOT_ER_AAC_ELD)) for (i = 0; i < sbrTuningTableSize; i++) { if (isForThisCore(i)) /* tuning table is for this core codec */ { if (numChannels == sbrTuningTable[i].numChannels && sampleRate == sbrTuningTable[i].sampleRate) { found = 1; if ((bitrate >= sbrTuningTable[i].bitrateFrom) && (bitrate < sbrTuningTable[i].bitrateTo)) { return i; } else { if (sbrTuningTable[i].bitrateFrom > bitrate) { if (sbrTuningTable[i].bitrateFrom < bitRateClosestLower) { bitRateClosestLower = sbrTuningTable[i].bitrateFrom; bitRateClosestLowerIndex = i; } } if (sbrTuningTable[i].bitrateTo <= bitrate) { if (sbrTuningTable[i].bitrateTo > bitRateClosestUpper) { bitRateClosestUpper = sbrTuningTable[i].bitrateTo - 1; bitRateClosestUpperIndex = i; } } } } } } if (bitRateClosestUpperIndex >= 0) { return bitRateClosestUpperIndex; } if (pBitRateClosest != NULL) { /* If there was at least one matching tuning entry pick the least distance * bit rate */ if (found) { int distanceUpper = DISTANCE_CEIL_VALUE, distanceLower = DISTANCE_CEIL_VALUE; if (bitRateClosestLowerIndex >= 0) { distanceLower = sbrTuningTable[bitRateClosestLowerIndex].bitrateFrom - bitrate; } if (bitRateClosestUpperIndex >= 0) { distanceUpper = bitrate - sbrTuningTable[bitRateClosestUpperIndex].bitrateTo; } if (distanceUpper < distanceLower) { *pBitRateClosest = bitRateClosestUpper; } else { *pBitRateClosest = bitRateClosestLower; } } else { *pBitRateClosest = 0; } } return INVALID_TABLE_IDX; } /***************************************************************************/ /*! \brief Selects the PS tuning settings to use dependent on bitrate and core coder \return Index to the appropriate table ****************************************************************************/ static INT getPsTuningTableIndex(UINT bitrate, UINT *pBitRateClosest) { INT i, paramSets = sizeof(psTuningTable) / sizeof(psTuningTable[0]); int bitRateClosestLowerIndex = -1, bitRateClosestUpperIndex = -1; UINT bitRateClosestUpper = 0, bitRateClosestLower = DISTANCE_CEIL_VALUE; for (i = 0; i < paramSets; i++) { if ((bitrate >= psTuningTable[i].bitrateFrom) && (bitrate < psTuningTable[i].bitrateTo)) { return i; } else { if (psTuningTable[i].bitrateFrom > bitrate) { if (psTuningTable[i].bitrateFrom < bitRateClosestLower) { bitRateClosestLower = psTuningTable[i].bitrateFrom; bitRateClosestLowerIndex = i; } } if (psTuningTable[i].bitrateTo <= bitrate) { if (psTuningTable[i].bitrateTo > bitRateClosestUpper) { bitRateClosestUpper = psTuningTable[i].bitrateTo - 1; bitRateClosestUpperIndex = i; } } } } if (bitRateClosestUpperIndex >= 0) { return bitRateClosestUpperIndex; } if (pBitRateClosest != NULL) { int distanceUpper = DISTANCE_CEIL_VALUE, distanceLower = DISTANCE_CEIL_VALUE; if (bitRateClosestLowerIndex >= 0) { distanceLower = sbrTuningTable[bitRateClosestLowerIndex].bitrateFrom - bitrate; } if (bitRateClosestUpperIndex >= 0) { distanceUpper = bitrate - sbrTuningTable[bitRateClosestUpperIndex].bitrateTo; } if (distanceUpper < distanceLower) { *pBitRateClosest = bitRateClosestUpper; } else { *pBitRateClosest = bitRateClosestLower; } } return INVALID_TABLE_IDX; } /***************************************************************************/ /*! \brief In case of downsampled SBR we may need to lower the stop freq of a tuning setting to fit into the lower half of the spectrum ( which is sampleRate/4 ) \return the adapted stop frequency index (-1 -> error) \ingroup SbrEncCfg ****************************************************************************/ static INT FDKsbrEnc_GetDownsampledStopFreq(const INT sampleRateCore, const INT startFreq, INT stopFreq, const INT downSampleFactor) { INT maxStopFreqRaw = sampleRateCore / 2; INT startBand, stopBand; HANDLE_ERROR_INFO err; while (stopFreq > 0 && FDKsbrEnc_getSbrStopFreqRAW(stopFreq, sampleRateCore) > maxStopFreqRaw) { stopFreq--; } if (FDKsbrEnc_getSbrStopFreqRAW(stopFreq, sampleRateCore) > maxStopFreqRaw) return -1; err = FDKsbrEnc_FindStartAndStopBand( sampleRateCore << (downSampleFactor - 1), sampleRateCore, 32 << (downSampleFactor - 1), startFreq, stopFreq, &startBand, &stopBand); if (err) return -1; return stopFreq; } /***************************************************************************/ /*! \brief tells us, if for the given coreCoder, bitrate, number of channels and input sampling rate an SBR setting is available. If yes, it tells us also the core sampling rate we would need to run with \return a flag indicating success: yes (1) or no (0) ****************************************************************************/ static UINT FDKsbrEnc_IsSbrSettingAvail( UINT bitrate, /*! the total bitrate in bits/sec */ UINT vbrMode, /*! the vbr paramter, 0 means constant bitrate */ UINT numOutputChannels, /*! the number of channels for the core coder */ UINT sampleRateInput, /*! the input sample rate [in Hz] */ UINT sampleRateCore, /*! the core's sampling rate */ AUDIO_OBJECT_TYPE core) { INT idx = INVALID_TABLE_IDX; if (sampleRateInput < 16000) return 0; if (bitrate == 0) { /* map vbr quality to bitrate */ if (vbrMode < 30) bitrate = 24000; else if (vbrMode < 40) bitrate = 28000; else if (vbrMode < 60) bitrate = 32000; else if (vbrMode < 75) bitrate = 40000; else bitrate = 48000; bitrate *= numOutputChannels; } idx = getSbrTuningTableIndex(bitrate, numOutputChannels, sampleRateCore, core, NULL); return (idx == INVALID_TABLE_IDX ? 0 : 1); } /***************************************************************************/ /*! \brief Adjusts the SBR settings according to the chosen core coder settings which are accessible via config->codecSettings \return A flag indicating success: yes (1) or no (0) ****************************************************************************/ static UINT FDKsbrEnc_AdjustSbrSettings( const sbrConfigurationPtr config, /*! output, modified */ UINT bitRate, /*! the total bitrate in bits/sec */ UINT numChannels, /*! the core coder number of channels */ UINT sampleRateCore, /*! the core coder sampling rate in Hz */ UINT sampleRateSbr, /*! the sbr coder sampling rate in Hz */ UINT transFac, /*! the short block to long block ratio */ UINT standardBitrate, /*! the standard bitrate per channel in bits/sec */ UINT vbrMode, /*! the vbr paramter, 0 poor quality .. 100 high quality*/ UINT useSpeechConfig, /*!< adapt tuning parameters for speech ? */ UINT lcsMode, /*! the low complexity stereo mode */ UINT bParametricStereo, /*!< use parametric stereo */ AUDIO_OBJECT_TYPE core) /* Core audio codec object type */ { INT idx = INVALID_TABLE_IDX; /* set the core codec settings */ config->codecSettings.bitRate = bitRate; config->codecSettings.nChannels = numChannels; config->codecSettings.sampleFreq = sampleRateCore; config->codecSettings.transFac = transFac; config->codecSettings.standardBitrate = standardBitrate; if (bitRate < 28000) { config->threshold_AmpRes_FF_m = (FIXP_DBL)MAXVAL_DBL; config->threshold_AmpRes_FF_e = 7; } else if (bitRate >= 28000 && bitRate <= 48000) { /* The float threshold is 75 0.524288f is fractional part of RELAXATION, the quotaMatrix and therefore tonality are scaled by this 2/3 is because the original implementation divides the tonality values by 3, here it's divided by 2 128 compensates the necessary shiftfactor of 7 */ config->threshold_AmpRes_FF_m = FL2FXCONST_DBL(75.0f * 0.524288f / (2.0f / 3.0f) / 128.0f); config->threshold_AmpRes_FF_e = 7; } else if (bitRate > 48000) { config->threshold_AmpRes_FF_m = FL2FXCONST_DBL(0); config->threshold_AmpRes_FF_e = 0; } if (bitRate == 0) { /* map vbr quality to bitrate */ if (vbrMode < 30) bitRate = 24000; else if (vbrMode < 40) bitRate = 28000; else if (vbrMode < 60) bitRate = 32000; else if (vbrMode < 75) bitRate = 40000; else bitRate = 48000; bitRate *= numChannels; /* fix to enable mono vbrMode<40 @ 44.1 of 48kHz */ if (numChannels == 1) { if (sampleRateSbr == 44100 || sampleRateSbr == 48000) { if (vbrMode < 40) bitRate = 32000; } } } idx = getSbrTuningTableIndex(bitRate, numChannels, sampleRateCore, core, NULL); if (idx != INVALID_TABLE_IDX) { config->startFreq = sbrTuningTable[idx].startFreq; config->stopFreq = sbrTuningTable[idx].stopFreq; if (useSpeechConfig) { config->startFreq = sbrTuningTable[idx].startFreqSpeech; config->stopFreq = sbrTuningTable[idx].stopFreqSpeech; } /* Adapt stop frequency in case of downsampled SBR - only 32 bands then */ if (1 == config->downSampleFactor) { INT dsStopFreq = FDKsbrEnc_GetDownsampledStopFreq( sampleRateCore, config->startFreq, config->stopFreq, config->downSampleFactor); if (dsStopFreq < 0) { return 0; } config->stopFreq = dsStopFreq; } config->sbr_noise_bands = sbrTuningTable[idx].numNoiseBands; if (core == AOT_ER_AAC_ELD) config->init_amp_res_FF = SBR_AMP_RES_1_5; config->noiseFloorOffset = sbrTuningTable[idx].noiseFloorOffset; config->ana_max_level = sbrTuningTable[idx].noiseMaxLevel; config->stereoMode = sbrTuningTable[idx].stereoMode; config->freqScale = sbrTuningTable[idx].freqScale; if (numChannels == 1) { /* stereo case */ switch (core) { case AOT_AAC_LC: if (bitRate <= (useSpeechConfig ? 24000U : 20000U)) { config->freq_res_fixfix[0] = FREQ_RES_LOW; /* set low frequency resolution for non-split frames */ config->freq_res_fixfix[1] = FREQ_RES_LOW; /* set low frequency resolution for split frames */ } break; case AOT_ER_AAC_ELD: if (bitRate < 36000) config->freq_res_fixfix[1] = FREQ_RES_LOW; /* set low frequency resolution for split frames */ if (bitRate < 26000) { config->freq_res_fixfix[0] = FREQ_RES_LOW; /* set low frequency resolution for non-split frames */ config->fResTransIsLow = 1; /* for transient frames, set low frequency resolution */ } break; default: break; } } else { /* stereo case */ switch (core) { case AOT_AAC_LC: if (bitRate <= 28000) { config->freq_res_fixfix[0] = FREQ_RES_LOW; /* set low frequency resolution for non-split frames */ config->freq_res_fixfix[1] = FREQ_RES_LOW; /* set low frequency resolution for split frames */ } break; case AOT_ER_AAC_ELD: if (bitRate < 72000) { config->freq_res_fixfix[1] = FREQ_RES_LOW; /* set low frequency resolution for split frames */ } if (bitRate < 52000) { config->freq_res_fixfix[0] = FREQ_RES_LOW; /* set low frequency resolution for non-split frames */ config->fResTransIsLow = 1; /* for transient frames, set low frequency resolution */ } break; default: break; } if (bitRate <= 28000) { /* additionally restrict frequency resolution in FIXFIX frames to further reduce SBR payload size */ config->freq_res_fixfix[0] = FREQ_RES_LOW; config->freq_res_fixfix[1] = FREQ_RES_LOW; } } /* adjust usage of parametric coding dependent on bitrate and speech config * flag */ if (useSpeechConfig) config->parametricCoding = 0; if (core == AOT_ER_AAC_ELD) { if (bitRate < 28000) config->init_amp_res_FF = SBR_AMP_RES_3_0; config->SendHeaderDataTime = -1; } if (numChannels == 1) { if (bitRate < 16000) { config->parametricCoding = 0; } } else { if (bitRate < 20000) { config->parametricCoding = 0; } } config->useSpeechConfig = useSpeechConfig; /* PS settings */ config->bParametricStereo = bParametricStereo; return 1; } else { return 0; } } /***************************************************************************** functionname: FDKsbrEnc_InitializeSbrDefaults description: initializes the SBR configuration returns: error status input: - core codec type, - factor of SBR to core frame length, - core frame length output: initialized SBR configuration *****************************************************************************/ static UINT FDKsbrEnc_InitializeSbrDefaults(sbrConfigurationPtr config, INT downSampleFactor, UINT codecGranuleLen, const INT isLowDelay) { if ((downSampleFactor < 1 || downSampleFactor > 2) || (codecGranuleLen * downSampleFactor > 64 * 32)) return (0); /* error */ config->SendHeaderDataTime = 1000; config->useWaveCoding = 0; config->crcSbr = 0; config->dynBwSupported = 1; if (isLowDelay) config->tran_thr = 6000; else config->tran_thr = 13000; config->parametricCoding = 1; config->sbrFrameSize = codecGranuleLen * downSampleFactor; config->downSampleFactor = downSampleFactor; /* sbr default parameters */ config->sbr_data_extra = 0; config->amp_res = SBR_AMP_RES_3_0; config->tran_fc = 0; config->tran_det_mode = 1; config->spread = 1; config->stat = 0; config->e = 1; config->deltaTAcrossFrames = 1; config->dF_edge_1stEnv = FL2FXCONST_DBL(0.3f); config->dF_edge_incr = FL2FXCONST_DBL(0.3f); config->sbr_invf_mode = INVF_SWITCHED; config->sbr_xpos_mode = XPOS_LC; config->sbr_xpos_ctrl = SBR_XPOS_CTRL_DEFAULT; config->sbr_xpos_level = 0; config->useSaPan = 0; config->dynBwEnabled = 0; /* the following parameters are overwritten by the FDKsbrEnc_AdjustSbrSettings() function since they are included in the tuning table */ config->stereoMode = SBR_SWITCH_LRC; config->ana_max_level = 6; config->noiseFloorOffset = 0; config->startFreq = 5; /* 5.9 respectively 6.0 kHz at fs = 44.1/48 kHz */ config->stopFreq = 9; /* 16.2 respectively 16.8 kHz at fs = 44.1/48 kHz */ config->freq_res_fixfix[0] = FREQ_RES_HIGH; /* non-split case */ config->freq_res_fixfix[1] = FREQ_RES_HIGH; /* split case */ config->fResTransIsLow = 0; /* for transient frames, set variable frequency resolution according to freqResTable */ /* header_extra_1 */ config->freqScale = SBR_FREQ_SCALE_DEFAULT; config->alterScale = SBR_ALTER_SCALE_DEFAULT; config->sbr_noise_bands = SBR_NOISE_BANDS_DEFAULT; /* header_extra_2 */ config->sbr_limiter_bands = SBR_LIMITER_BANDS_DEFAULT; config->sbr_limiter_gains = SBR_LIMITER_GAINS_DEFAULT; config->sbr_interpol_freq = SBR_INTERPOL_FREQ_DEFAULT; config->sbr_smoothing_length = SBR_SMOOTHING_LENGTH_DEFAULT; return 1; } /***************************************************************************** functionname: DeleteEnvChannel description: frees memory of one SBR channel returns: - input: handle of channel output: released handle *****************************************************************************/ static void deleteEnvChannel(HANDLE_ENV_CHANNEL hEnvCut) { if (hEnvCut) { FDKsbrEnc_DeleteTonCorrParamExtr(&hEnvCut->TonCorr); FDKsbrEnc_deleteExtractSbrEnvelope(&hEnvCut->sbrExtractEnvelope); } } /***************************************************************************** functionname: sbrEncoder_ChannelClose description: close the channel coding handle returns: input: phSbrChannel output: *****************************************************************************/ static void sbrEncoder_ChannelClose(HANDLE_SBR_CHANNEL hSbrChannel) { if (hSbrChannel != NULL) { deleteEnvChannel(&hSbrChannel->hEnvChannel); } } /***************************************************************************** functionname: sbrEncoder_ElementClose description: close the channel coding handle returns: input: phSbrChannel output: *****************************************************************************/ static void sbrEncoder_ElementClose(HANDLE_SBR_ELEMENT *phSbrElement) { HANDLE_SBR_ELEMENT hSbrElement = *phSbrElement; if (hSbrElement != NULL) { if (hSbrElement->sbrConfigData.v_k_master) FreeRam_Sbr_v_k_master(&hSbrElement->sbrConfigData.v_k_master); if (hSbrElement->sbrConfigData.freqBandTable[LO]) FreeRam_Sbr_freqBandTableLO( &hSbrElement->sbrConfigData.freqBandTable[LO]); if (hSbrElement->sbrConfigData.freqBandTable[HI]) FreeRam_Sbr_freqBandTableHI( &hSbrElement->sbrConfigData.freqBandTable[HI]); FreeRam_SbrElement(phSbrElement); } return; } void sbrEncoder_Close(HANDLE_SBR_ENCODER *phSbrEncoder) { HANDLE_SBR_ENCODER hSbrEncoder = *phSbrEncoder; if (hSbrEncoder != NULL) { int el, ch; for (el = 0; el < (8); el++) { if (hSbrEncoder->sbrElement[el] != NULL) { sbrEncoder_ElementClose(&hSbrEncoder->sbrElement[el]); } } /* Close sbr Channels */ for (ch = 0; ch < (8); ch++) { if (hSbrEncoder->pSbrChannel[ch]) { sbrEncoder_ChannelClose(hSbrEncoder->pSbrChannel[ch]); FreeRam_SbrChannel(&hSbrEncoder->pSbrChannel[ch]); } if (hSbrEncoder->QmfAnalysis[ch].FilterStates) FreeRam_Sbr_QmfStatesAnalysis( (FIXP_QAS **)&hSbrEncoder->QmfAnalysis[ch].FilterStates); } if (hSbrEncoder->hParametricStereo) PSEnc_Destroy(&hSbrEncoder->hParametricStereo); if (hSbrEncoder->qmfSynthesisPS.FilterStates) FreeRam_PsQmfStatesSynthesis( (FIXP_DBL **)&hSbrEncoder->qmfSynthesisPS.FilterStates); /* Release Overlay */ if (hSbrEncoder->pSBRdynamic_RAM) FreeRam_SbrDynamic_RAM((FIXP_DBL **)&hSbrEncoder->pSBRdynamic_RAM); FreeRam_SbrEncoder(phSbrEncoder); } } /***************************************************************************** functionname: updateFreqBandTable description: updates vk_master returns: - input: config handle output: error info *****************************************************************************/ static INT updateFreqBandTable(HANDLE_SBR_CONFIG_DATA sbrConfigData, HANDLE_SBR_HEADER_DATA sbrHeaderData, const INT downSampleFactor) { INT k0, k2; if (FDKsbrEnc_FindStartAndStopBand( sbrConfigData->sampleFreq, sbrConfigData->sampleFreq >> (downSampleFactor - 1), sbrConfigData->noQmfBands, sbrHeaderData->sbr_start_frequency, sbrHeaderData->sbr_stop_frequency, &k0, &k2)) return (1); if (FDKsbrEnc_UpdateFreqScale( sbrConfigData->v_k_master, &sbrConfigData->num_Master, k0, k2, sbrHeaderData->freqScale, sbrHeaderData->alterScale)) return (1); sbrHeaderData->sbr_xover_band = 0; if (FDKsbrEnc_UpdateHiRes(sbrConfigData->freqBandTable[HI], &sbrConfigData->nSfb[HI], sbrConfigData->v_k_master, sbrConfigData->num_Master, &sbrHeaderData->sbr_xover_band)) return (1); FDKsbrEnc_UpdateLoRes( sbrConfigData->freqBandTable[LO], &sbrConfigData->nSfb[LO], sbrConfigData->freqBandTable[HI], sbrConfigData->nSfb[HI]); sbrConfigData->xOverFreq = (sbrConfigData->freqBandTable[LOW_RES][0] * sbrConfigData->sampleFreq / sbrConfigData->noQmfBands + 1) >> 1; return (0); } /***************************************************************************** functionname: resetEnvChannel description: resets parameters and allocates memory returns: error status input: output: hEnv *****************************************************************************/ static INT resetEnvChannel(HANDLE_SBR_CONFIG_DATA sbrConfigData, HANDLE_SBR_HEADER_DATA sbrHeaderData, HANDLE_ENV_CHANNEL hEnv) { /* note !!! hEnv->encEnvData.noOfnoisebands will be updated later in function * FDKsbrEnc_extractSbrEnvelope !!!*/ hEnv->TonCorr.sbrNoiseFloorEstimate.noiseBands = sbrHeaderData->sbr_noise_bands; if (FDKsbrEnc_ResetTonCorrParamExtr( &hEnv->TonCorr, sbrConfigData->xposCtrlSwitch, sbrConfigData->freqBandTable[HI][0], sbrConfigData->v_k_master, sbrConfigData->num_Master, sbrConfigData->sampleFreq, sbrConfigData->freqBandTable, sbrConfigData->nSfb, sbrConfigData->noQmfBands)) return (1); hEnv->sbrCodeNoiseFloor.nSfb[LO] = hEnv->TonCorr.sbrNoiseFloorEstimate.noNoiseBands; hEnv->sbrCodeNoiseFloor.nSfb[HI] = hEnv->TonCorr.sbrNoiseFloorEstimate.noNoiseBands; hEnv->sbrCodeEnvelope.nSfb[LO] = sbrConfigData->nSfb[LO]; hEnv->sbrCodeEnvelope.nSfb[HI] = sbrConfigData->nSfb[HI]; hEnv->encEnvData.noHarmonics = sbrConfigData->nSfb[HI]; hEnv->sbrCodeEnvelope.upDate = 0; hEnv->sbrCodeNoiseFloor.upDate = 0; return (0); } /* ****************************** FDKsbrEnc_SbrGetXOverFreq * ******************************/ /** * @fn * @brief calculates the closest possible crossover frequency * @return the crossover frequency SBR accepts * */ static INT FDKsbrEnc_SbrGetXOverFreq( HANDLE_SBR_ELEMENT hEnv, /*!< handle to SBR encoder instance */ INT xoverFreq) /*!< from core coder suggested crossover frequency */ { INT band; INT lastDiff, newDiff; INT cutoffSb; UCHAR *RESTRICT pVKMaster = hEnv->sbrConfigData.v_k_master; /* Check if there is a matching cutoff frequency in the master table */ cutoffSb = (4 * xoverFreq * hEnv->sbrConfigData.noQmfBands / hEnv->sbrConfigData.sampleFreq + 1) >> 1; lastDiff = cutoffSb; for (band = 0; band < hEnv->sbrConfigData.num_Master; band++) { newDiff = fixp_abs((INT)pVKMaster[band] - cutoffSb); if (newDiff >= lastDiff) { band--; break; } lastDiff = newDiff; } return ((pVKMaster[band] * hEnv->sbrConfigData.sampleFreq / hEnv->sbrConfigData.noQmfBands + 1) >> 1); } /***************************************************************************** functionname: FDKsbrEnc_EnvEncodeFrame description: performs the sbr envelope calculation for one element returns: input: output: *****************************************************************************/ INT FDKsbrEnc_EnvEncodeFrame( HANDLE_SBR_ENCODER hEnvEncoder, int iElement, INT_PCM *samples, /*!< time samples, always deinterleaved */ UINT samplesBufSize, /*!< time buffer channel stride */ UINT *sbrDataBits, /*!< Size of SBR payload */ UCHAR *sbrData, /*!< SBR payload */ int clearOutput /*!< Do not consider any input signal */ ) { HANDLE_SBR_ELEMENT hSbrElement = NULL; FDK_CRCINFO crcInfo; INT crcReg; INT ch; INT band; INT cutoffSb; INT newXOver; if (hEnvEncoder == NULL) return -1; hSbrElement = hEnvEncoder->sbrElement[iElement]; if (hSbrElement == NULL) return -1; /* header bitstream handling */ HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData = &hSbrElement->sbrBitstreamData; INT psHeaderActive = 0; sbrBitstreamData->HeaderActive = 0; /* Anticipate PS header because of internal PS bitstream delay in order to be * in sync with SBR header. */ if (sbrBitstreamData->CountSendHeaderData == (sbrBitstreamData->NrSendHeaderData - 1)) { psHeaderActive = 1; } /* Signal SBR header to be written into bitstream */ if (sbrBitstreamData->CountSendHeaderData == 0) { sbrBitstreamData->HeaderActive = 1; } /* Increment header interval counter */ if (sbrBitstreamData->NrSendHeaderData == 0) { sbrBitstreamData->CountSendHeaderData = 1; } else { if (sbrBitstreamData->CountSendHeaderData >= 0) { sbrBitstreamData->CountSendHeaderData++; sbrBitstreamData->CountSendHeaderData %= sbrBitstreamData->NrSendHeaderData; } } if (hSbrElement->CmonData.dynBwEnabled) { INT i; for (i = 4; i > 0; i--) hSbrElement->dynXOverFreqDelay[i] = hSbrElement->dynXOverFreqDelay[i - 1]; hSbrElement->dynXOverFreqDelay[0] = hSbrElement->CmonData.dynXOverFreqEnc; if (hSbrElement->dynXOverFreqDelay[1] > hSbrElement->dynXOverFreqDelay[2]) newXOver = hSbrElement->dynXOverFreqDelay[2]; else newXOver = hSbrElement->dynXOverFreqDelay[1]; /* has the crossover frequency changed? */ if (hSbrElement->sbrConfigData.dynXOverFreq != newXOver) { /* get corresponding master band */ cutoffSb = ((4 * newXOver * hSbrElement->sbrConfigData.noQmfBands / hSbrElement->sbrConfigData.sampleFreq) + 1) >> 1; for (band = 0; band < hSbrElement->sbrConfigData.num_Master; band++) { if (cutoffSb == hSbrElement->sbrConfigData.v_k_master[band]) break; } FDK_ASSERT(band < hSbrElement->sbrConfigData.num_Master); hSbrElement->sbrConfigData.dynXOverFreq = newXOver; hSbrElement->sbrHeaderData.sbr_xover_band = band; hSbrElement->sbrBitstreamData.HeaderActive = 1; psHeaderActive = 1; /* ps header is one frame delayed */ /* update vk_master table */ if (updateFreqBandTable(&hSbrElement->sbrConfigData, &hSbrElement->sbrHeaderData, hEnvEncoder->downSampleFactor)) return (1); /* reset SBR channels */ INT nEnvCh = hSbrElement->sbrConfigData.nChannels; for (ch = 0; ch < nEnvCh; ch++) { if (resetEnvChannel(&hSbrElement->sbrConfigData, &hSbrElement->sbrHeaderData, &hSbrElement->sbrChannel[ch]->hEnvChannel)) return (1); } } } /* allocate space for dummy header and crc */ crcReg = FDKsbrEnc_InitSbrBitstream( &hSbrElement->CmonData, hSbrElement->payloadDelayLine[hEnvEncoder->nBitstrDelay], MAX_PAYLOAD_SIZE * sizeof(UCHAR), &crcInfo, hSbrElement->sbrConfigData.sbrSyntaxFlags); /* Temporal Envelope Data */ SBR_FRAME_TEMP_DATA _fData; SBR_FRAME_TEMP_DATA *fData = &_fData; SBR_ENV_TEMP_DATA eData[MAX_NUM_CHANNELS]; /* Init Temporal Envelope Data */ { int i; FDKmemclear(&eData[0], sizeof(SBR_ENV_TEMP_DATA)); FDKmemclear(&eData[1], sizeof(SBR_ENV_TEMP_DATA)); FDKmemclear(fData, sizeof(SBR_FRAME_TEMP_DATA)); for (i = 0; i < MAX_NUM_NOISE_VALUES; i++) fData->res[i] = FREQ_RES_HIGH; } if (!clearOutput) { /* * Transform audio data into QMF domain */ for (ch = 0; ch < hSbrElement->sbrConfigData.nChannels; ch++) { HANDLE_ENV_CHANNEL h_envChan = &hSbrElement->sbrChannel[ch]->hEnvChannel; HANDLE_SBR_EXTRACT_ENVELOPE sbrExtrEnv = &h_envChan->sbrExtractEnvelope; if (hSbrElement->elInfo.fParametricStereo == 0) { QMF_SCALE_FACTOR tmpScale; FIXP_DBL **pQmfReal, **pQmfImag; C_AALLOC_SCRATCH_START(qmfWorkBuffer, FIXP_DBL, 64 * 2) /* Obtain pointers to QMF buffers. */ pQmfReal = sbrExtrEnv->rBuffer; pQmfImag = sbrExtrEnv->iBuffer; qmfAnalysisFiltering( hSbrElement->hQmfAnalysis[ch], pQmfReal, pQmfImag, &tmpScale, samples + hSbrElement->elInfo.ChannelIndex[ch] * samplesBufSize, 0, 1, qmfWorkBuffer); h_envChan->qmfScale = tmpScale.lb_scale + 7; C_AALLOC_SCRATCH_END(qmfWorkBuffer, FIXP_DBL, 64 * 2) } /* fParametricStereo == 0 */ /* Parametric Stereo processing */ if (hSbrElement->elInfo.fParametricStereo) { INT error = noError; /* Limit Parametric Stereo to one instance */ FDK_ASSERT(ch == 0); if (error == noError) { /* parametric stereo processing: - input: o left and right time domain samples - processing: o stereo qmf analysis o stereo hybrid analysis o ps parameter extraction o downmix + hybrid synthesis - output: o downmixed qmf data is written to sbrExtrEnv->rBuffer and sbrExtrEnv->iBuffer */ SCHAR qmfScale; INT_PCM *pSamples[2] = { samples + hSbrElement->elInfo.ChannelIndex[0] * samplesBufSize, samples + hSbrElement->elInfo.ChannelIndex[1] * samplesBufSize}; error = FDKsbrEnc_PSEnc_ParametricStereoProcessing( hEnvEncoder->hParametricStereo, pSamples, samplesBufSize, hSbrElement->hQmfAnalysis, sbrExtrEnv->rBuffer, sbrExtrEnv->iBuffer, samples + hSbrElement->elInfo.ChannelIndex[ch] * samplesBufSize, &hEnvEncoder->qmfSynthesisPS, &qmfScale, psHeaderActive); h_envChan->qmfScale = (int)qmfScale; } } /* if (hEnvEncoder->hParametricStereo) */ /* Extract Envelope relevant things from QMF data */ FDKsbrEnc_extractSbrEnvelope1(&hSbrElement->sbrConfigData, &hSbrElement->sbrHeaderData, &hSbrElement->sbrBitstreamData, h_envChan, &hSbrElement->CmonData, &eData[ch], fData); } /* hEnvEncoder->sbrConfigData.nChannels */ } /* Process Envelope relevant things and calculate envelope data and write payload */ FDKsbrEnc_extractSbrEnvelope2( &hSbrElement->sbrConfigData, &hSbrElement->sbrHeaderData, (hSbrElement->elInfo.fParametricStereo) ? hEnvEncoder->hParametricStereo : NULL, &hSbrElement->sbrBitstreamData, &hSbrElement->sbrChannel[0]->hEnvChannel, (hSbrElement->sbrConfigData.stereoMode != SBR_MONO) ? &hSbrElement->sbrChannel[1]->hEnvChannel : NULL, &hSbrElement->CmonData, eData, fData, clearOutput); hSbrElement->sbrBitstreamData.rightBorderFIX = 0; /* format payload, calculate crc */ FDKsbrEnc_AssembleSbrBitstream(&hSbrElement->CmonData, &crcInfo, crcReg, hSbrElement->sbrConfigData.sbrSyntaxFlags); /* save new payload, set to zero length if greater than MAX_PAYLOAD_SIZE */ hSbrElement->payloadDelayLineSize[hEnvEncoder->nBitstrDelay] = FDKgetValidBits(&hSbrElement->CmonData.sbrBitbuf); if (hSbrElement->payloadDelayLineSize[hEnvEncoder->nBitstrDelay] > (MAX_PAYLOAD_SIZE << 3)) hSbrElement->payloadDelayLineSize[hEnvEncoder->nBitstrDelay] = 0; /* While filling the Delay lines, sbrData is NULL */ if (sbrData) { *sbrDataBits = hSbrElement->payloadDelayLineSize[0]; FDKmemcpy(sbrData, hSbrElement->payloadDelayLine[0], (hSbrElement->payloadDelayLineSize[0] + 7) >> 3); } /* delay header active flag */ if (hSbrElement->sbrBitstreamData.HeaderActive == 1) { hSbrElement->sbrBitstreamData.HeaderActiveDelay = 1 + hEnvEncoder->nBitstrDelay; } else { if (hSbrElement->sbrBitstreamData.HeaderActiveDelay > 0) { hSbrElement->sbrBitstreamData.HeaderActiveDelay--; } } return (0); } /***************************************************************************** functionname: FDKsbrEnc_Downsample description: performs downsampling and delay compensation of the core path returns: input: output: *****************************************************************************/ INT FDKsbrEnc_Downsample( HANDLE_SBR_ENCODER hSbrEncoder, INT_PCM *samples, /*!< time samples, always deinterleaved */ UINT samplesBufSize, /*!< time buffer size per channel */ UINT numChannels, /*!< number of channels */ UINT *sbrDataBits, /*!< Size of SBR payload */ UCHAR *sbrData, /*!< SBR payload */ int clearOutput /*!< Do not consider any input signal */ ) { HANDLE_SBR_ELEMENT hSbrElement = NULL; INT nOutSamples; int el; if (hSbrEncoder->downSampleFactor > 1) { /* Do downsampling */ /* Loop over elements (LFE is handled later) */ for (el = 0; el < hSbrEncoder->noElements; el++) { hSbrElement = hSbrEncoder->sbrElement[el]; if (hSbrEncoder->sbrElement[el] != NULL) { if (hSbrEncoder->downsamplingMethod == SBRENC_DS_TIME) { int ch; int nChannels = hSbrElement->sbrConfigData.nChannels; for (ch = 0; ch < nChannels; ch++) { FDKaacEnc_Downsample( &hSbrElement->sbrChannel[ch]->downSampler, samples + hSbrElement->elInfo.ChannelIndex[ch] * samplesBufSize + hSbrEncoder->bufferOffset / numChannels, hSbrElement->sbrConfigData.frameSize, samples + hSbrElement->elInfo.ChannelIndex[ch] * samplesBufSize, &nOutSamples); } } } } /* Handle LFE (if existing) */ if (hSbrEncoder->lfeChIdx != -1) { /* lfe downsampler */ FDKaacEnc_Downsample(&hSbrEncoder->lfeDownSampler, samples + hSbrEncoder->lfeChIdx * samplesBufSize + hSbrEncoder->bufferOffset / numChannels, hSbrEncoder->frameSize, samples + hSbrEncoder->lfeChIdx * samplesBufSize, &nOutSamples); } } else { /* No downsampling. Still, some buffer shifting for correct delay */ int samples2Copy = hSbrEncoder->frameSize; if (hSbrEncoder->bufferOffset / (int)numChannels < samples2Copy) { for (int c = 0; c < (int)numChannels; c++) { /* Do memmove while taking care of overlapping memory areas. (memcpy does not necessarily take care) Distinguish between oeverlapping and non overlapping version due to reasons of complexity. */ FDKmemmove(samples + c * samplesBufSize, samples + c * samplesBufSize + hSbrEncoder->bufferOffset / numChannels, samples2Copy * sizeof(INT_PCM)); } } else { for (int c = 0; c < (int)numChannels; c++) { /* Simple memcpy since the memory areas are not overlapping */ FDKmemcpy(samples + c * samplesBufSize, samples + c * samplesBufSize + hSbrEncoder->bufferOffset / numChannels, samples2Copy * sizeof(INT_PCM)); } } } return 0; } /***************************************************************************** functionname: createEnvChannel description: initializes parameters and allocates memory returns: error status input: output: hEnv *****************************************************************************/ static INT createEnvChannel(HANDLE_ENV_CHANNEL hEnv, INT channel, UCHAR *dynamic_RAM) { FDKmemclear(hEnv, sizeof(struct ENV_CHANNEL)); if (FDKsbrEnc_CreateTonCorrParamExtr(&hEnv->TonCorr, channel)) { return (1); } if (FDKsbrEnc_CreateExtractSbrEnvelope(&hEnv->sbrExtractEnvelope, channel, /*chan*/ 0, dynamic_RAM)) { return (1); } return 0; } /***************************************************************************** functionname: initEnvChannel description: initializes parameters returns: error status input: output: *****************************************************************************/ static INT initEnvChannel(HANDLE_SBR_CONFIG_DATA sbrConfigData, HANDLE_SBR_HEADER_DATA sbrHeaderData, HANDLE_ENV_CHANNEL hEnv, sbrConfigurationPtr params, ULONG statesInitFlag, INT chanInEl, UCHAR *dynamic_RAM) { int frameShift, tran_off = 0; INT e; INT tran_fc; INT timeSlots, timeStep, startIndex; INT noiseBands[2] = {3, 3}; e = 1 << params->e; FDK_ASSERT(params->e >= 0); hEnv->encEnvData.freq_res_fixfix[0] = params->freq_res_fixfix[0]; hEnv->encEnvData.freq_res_fixfix[1] = params->freq_res_fixfix[1]; hEnv->encEnvData.fResTransIsLow = params->fResTransIsLow; hEnv->fLevelProtect = 0; hEnv->encEnvData.ldGrid = (sbrConfigData->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) ? 1 : 0; hEnv->encEnvData.sbr_xpos_mode = (XPOS_MODE)params->sbr_xpos_mode; if (hEnv->encEnvData.sbr_xpos_mode == XPOS_SWITCHED) { /* no other type than XPOS_MDCT or XPOS_SPEECH allowed, but enable switching */ sbrConfigData->switchTransposers = TRUE; hEnv->encEnvData.sbr_xpos_mode = XPOS_MDCT; } else { sbrConfigData->switchTransposers = FALSE; } hEnv->encEnvData.sbr_xpos_ctrl = params->sbr_xpos_ctrl; /* extended data */ if (params->parametricCoding) { hEnv->encEnvData.extended_data = 1; } else { hEnv->encEnvData.extended_data = 0; } hEnv->encEnvData.extension_size = 0; startIndex = QMF_FILTER_PROTOTYPE_SIZE - sbrConfigData->noQmfBands; switch (params->sbrFrameSize) { case 2304: timeSlots = 18; break; case 2048: case 1024: case 512: timeSlots = 16; break; case 1920: case 960: case 480: timeSlots = 15; break; case 1152: timeSlots = 9; break; default: return (1); /* Illegal frame size */ } timeStep = sbrConfigData->noQmfSlots / timeSlots; if (FDKsbrEnc_InitTonCorrParamExtr( params->sbrFrameSize, &hEnv->TonCorr, sbrConfigData, timeSlots, params->sbr_xpos_ctrl, params->ana_max_level, sbrHeaderData->sbr_noise_bands, params->noiseFloorOffset, params->useSpeechConfig)) return (1); hEnv->encEnvData.noOfnoisebands = hEnv->TonCorr.sbrNoiseFloorEstimate.noNoiseBands; noiseBands[0] = hEnv->encEnvData.noOfnoisebands; noiseBands[1] = hEnv->encEnvData.noOfnoisebands; hEnv->encEnvData.sbr_invf_mode = (INVF_MODE)params->sbr_invf_mode; if (hEnv->encEnvData.sbr_invf_mode == INVF_SWITCHED) { hEnv->encEnvData.sbr_invf_mode = INVF_MID_LEVEL; hEnv->TonCorr.switchInverseFilt = TRUE; } else { hEnv->TonCorr.switchInverseFilt = FALSE; } tran_fc = params->tran_fc; if (tran_fc == 0) { tran_fc = fixMin( 5000, FDKsbrEnc_getSbrStartFreqRAW(sbrHeaderData->sbr_start_frequency, params->codecSettings.sampleFreq)); } tran_fc = (tran_fc * 4 * sbrConfigData->noQmfBands / sbrConfigData->sampleFreq + 1) >> 1; if (sbrConfigData->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { frameShift = LD_PRETRAN_OFF; tran_off = LD_PRETRAN_OFF + FRAME_MIDDLE_SLOT_512LD * timeStep; } else { frameShift = 0; switch (timeSlots) { /* The factor of 2 is by definition. */ case NUMBER_TIME_SLOTS_2048: tran_off = 8 + FRAME_MIDDLE_SLOT_2048 * timeStep; break; case NUMBER_TIME_SLOTS_1920: tran_off = 7 + FRAME_MIDDLE_SLOT_1920 * timeStep; break; default: return 1; } } if (FDKsbrEnc_InitExtractSbrEnvelope( &hEnv->sbrExtractEnvelope, sbrConfigData->noQmfSlots, sbrConfigData->noQmfBands, startIndex, timeSlots, timeStep, tran_off, statesInitFlag, chanInEl, dynamic_RAM, sbrConfigData->sbrSyntaxFlags)) return (1); if (FDKsbrEnc_InitSbrCodeEnvelope(&hEnv->sbrCodeEnvelope, sbrConfigData->nSfb, params->deltaTAcrossFrames, params->dF_edge_1stEnv, params->dF_edge_incr)) return (1); if (FDKsbrEnc_InitSbrCodeEnvelope(&hEnv->sbrCodeNoiseFloor, noiseBands, params->deltaTAcrossFrames, 0, 0)) return (1); sbrConfigData->initAmpResFF = params->init_amp_res_FF; if (FDKsbrEnc_InitSbrHuffmanTables(&hEnv->encEnvData, &hEnv->sbrCodeEnvelope, &hEnv->sbrCodeNoiseFloor, sbrHeaderData->sbr_amp_res)) return (1); FDKsbrEnc_initFrameInfoGenerator( &hEnv->SbrEnvFrame, params->spread, e, params->stat, timeSlots, hEnv->encEnvData.freq_res_fixfix, hEnv->encEnvData.fResTransIsLow, hEnv->encEnvData.ldGrid); if (sbrConfigData->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { INT bandwidth_qmf_slot = (sbrConfigData->sampleFreq >> 1) / (sbrConfigData->noQmfBands); if (FDKsbrEnc_InitSbrFastTransientDetector( &hEnv->sbrFastTransientDetector, sbrConfigData->noQmfSlots, bandwidth_qmf_slot, sbrConfigData->noQmfBands, sbrConfigData->freqBandTable[0][0])) return (1); } /* The transient detector has to be initialized also if the fast transient detector was active, because the values from the transient detector structure are used. */ if (FDKsbrEnc_InitSbrTransientDetector( &hEnv->sbrTransientDetector, sbrConfigData->sbrSyntaxFlags, sbrConfigData->frameSize, sbrConfigData->sampleFreq, params, tran_fc, sbrConfigData->noQmfSlots, sbrConfigData->noQmfBands, hEnv->sbrExtractEnvelope.YBufferWriteOffset, hEnv->sbrExtractEnvelope.YBufferSzShift, frameShift, tran_off)) return (1); sbrConfigData->xposCtrlSwitch = params->sbr_xpos_ctrl; hEnv->encEnvData.noHarmonics = sbrConfigData->nSfb[HI]; hEnv->encEnvData.addHarmonicFlag = 0; return (0); } INT sbrEncoder_Open(HANDLE_SBR_ENCODER *phSbrEncoder, INT nElements, INT nChannels, INT supportPS) { INT i; INT errorStatus = 1; HANDLE_SBR_ENCODER hSbrEncoder = NULL; if (phSbrEncoder == NULL) { goto bail; } hSbrEncoder = GetRam_SbrEncoder(); if (hSbrEncoder == NULL) { goto bail; } FDKmemclear(hSbrEncoder, sizeof(SBR_ENCODER)); if (NULL == (hSbrEncoder->pSBRdynamic_RAM = (UCHAR *)GetRam_SbrDynamic_RAM())) { goto bail; } hSbrEncoder->dynamicRam = hSbrEncoder->pSBRdynamic_RAM; /* Create SBR elements */ for (i = 0; i < nElements; i++) { hSbrEncoder->sbrElement[i] = GetRam_SbrElement(i); if (hSbrEncoder->sbrElement[i] == NULL) { goto bail; } FDKmemclear(hSbrEncoder->sbrElement[i], sizeof(SBR_ELEMENT)); hSbrEncoder->sbrElement[i]->sbrConfigData.freqBandTable[LO] = GetRam_Sbr_freqBandTableLO(i); hSbrEncoder->sbrElement[i]->sbrConfigData.freqBandTable[HI] = GetRam_Sbr_freqBandTableHI(i); hSbrEncoder->sbrElement[i]->sbrConfigData.v_k_master = GetRam_Sbr_v_k_master(i); if ((hSbrEncoder->sbrElement[i]->sbrConfigData.freqBandTable[LO] == NULL) || (hSbrEncoder->sbrElement[i]->sbrConfigData.freqBandTable[HI] == NULL) || (hSbrEncoder->sbrElement[i]->sbrConfigData.v_k_master == NULL)) { goto bail; } } /* Create SBR channels */ for (i = 0; i < nChannels; i++) { hSbrEncoder->pSbrChannel[i] = GetRam_SbrChannel(i); if (hSbrEncoder->pSbrChannel[i] == NULL) { goto bail; } if (createEnvChannel(&hSbrEncoder->pSbrChannel[i]->hEnvChannel, i, hSbrEncoder->dynamicRam)) { goto bail; } } /* Create QMF States */ for (i = 0; i < fixMax(nChannels, (supportPS) ? 2 : 0); i++) { hSbrEncoder->QmfAnalysis[i].FilterStates = GetRam_Sbr_QmfStatesAnalysis(i); if (hSbrEncoder->QmfAnalysis[i].FilterStates == NULL) { goto bail; } } /* Create Parametric Stereo handle */ if (supportPS) { if (PSEnc_Create(&hSbrEncoder->hParametricStereo)) { goto bail; } hSbrEncoder->qmfSynthesisPS.FilterStates = GetRam_PsQmfStatesSynthesis(); if (hSbrEncoder->qmfSynthesisPS.FilterStates == NULL) { goto bail; } } /* supportPS */ *phSbrEncoder = hSbrEncoder; errorStatus = 0; return errorStatus; bail: /* Close SBR encoder instance */ sbrEncoder_Close(&hSbrEncoder); return errorStatus; } static INT FDKsbrEnc_Reallocate(HANDLE_SBR_ENCODER hSbrEncoder, SBR_ELEMENT_INFO elInfo[(8)], const INT noElements) { INT totalCh = 0; INT totalQmf = 0; INT coreEl; INT el = -1; hSbrEncoder->lfeChIdx = -1; /* default value, until lfe found */ for (coreEl = 0; coreEl < noElements; coreEl++) { /* SBR only handles SCE and CPE's */ if (elInfo[coreEl].elType == ID_SCE || elInfo[coreEl].elType == ID_CPE) { el++; } else { if (elInfo[coreEl].elType == ID_LFE) { hSbrEncoder->lfeChIdx = elInfo[coreEl].ChannelIndex[0]; } continue; } SBR_ELEMENT_INFO *pelInfo = &elInfo[coreEl]; HANDLE_SBR_ELEMENT hSbrElement = hSbrEncoder->sbrElement[el]; int ch; for (ch = 0; ch < pelInfo->nChannelsInEl; ch++) { hSbrElement->sbrChannel[ch] = hSbrEncoder->pSbrChannel[totalCh]; totalCh++; } /* analysis QMF */ for (ch = 0; ch < ((pelInfo->fParametricStereo) ? 2 : pelInfo->nChannelsInEl); ch++) { hSbrElement->elInfo.ChannelIndex[ch] = pelInfo->ChannelIndex[ch]; hSbrElement->hQmfAnalysis[ch] = &hSbrEncoder->QmfAnalysis[totalQmf++]; } /* Copy Element info */ hSbrElement->elInfo.elType = pelInfo->elType; hSbrElement->elInfo.instanceTag = pelInfo->instanceTag; hSbrElement->elInfo.nChannelsInEl = pelInfo->nChannelsInEl; hSbrElement->elInfo.fParametricStereo = pelInfo->fParametricStereo; hSbrElement->elInfo.fDualMono = pelInfo->fDualMono; } /* coreEl */ return 0; } /***************************************************************************** functionname: FDKsbrEnc_bsBufInit description: initializes bitstream buffer returns: initialized bitstream buffer in env encoder input: output: hEnv *****************************************************************************/ static INT FDKsbrEnc_bsBufInit(HANDLE_SBR_ELEMENT hSbrElement, int nBitstrDelay) { UCHAR *bitstreamBuffer; /* initialize the bitstream buffer */ bitstreamBuffer = hSbrElement->payloadDelayLine[nBitstrDelay]; FDKinitBitStream(&hSbrElement->CmonData.sbrBitbuf, bitstreamBuffer, MAX_PAYLOAD_SIZE * sizeof(UCHAR), 0, BS_WRITER); return (0); } /***************************************************************************** functionname: FDKsbrEnc_EnvInit description: initializes parameters returns: error status input: output: hEnv *****************************************************************************/ static INT FDKsbrEnc_EnvInit(HANDLE_SBR_ELEMENT hSbrElement, sbrConfigurationPtr params, INT *coreBandWith, AUDIO_OBJECT_TYPE aot, int nElement, const int headerPeriod, ULONG statesInitFlag, const SBRENC_DS_TYPE downsamplingMethod, UCHAR *dynamic_RAM) { int ch, i; if ((params->codecSettings.nChannels < 1) || (params->codecSettings.nChannels > MAX_NUM_CHANNELS)) { return (1); } /* init and set syntax flags */ hSbrElement->sbrConfigData.sbrSyntaxFlags = 0; switch (aot) { case AOT_ER_AAC_ELD: hSbrElement->sbrConfigData.sbrSyntaxFlags |= SBR_SYNTAX_LOW_DELAY; break; default: break; } if (params->crcSbr) { hSbrElement->sbrConfigData.sbrSyntaxFlags |= SBR_SYNTAX_CRC; } hSbrElement->sbrConfigData.noQmfBands = 64 >> (2 - params->downSampleFactor); switch (hSbrElement->sbrConfigData.noQmfBands) { case 64: hSbrElement->sbrConfigData.noQmfSlots = params->sbrFrameSize >> 6; break; case 32: hSbrElement->sbrConfigData.noQmfSlots = params->sbrFrameSize >> 5; break; default: hSbrElement->sbrConfigData.noQmfSlots = params->sbrFrameSize >> 6; return (2); } /* now initialize sbrConfigData, sbrHeaderData and sbrBitstreamData, */ hSbrElement->sbrConfigData.nChannels = params->codecSettings.nChannels; if (params->codecSettings.nChannels == 2) { if ((hSbrElement->elInfo.elType == ID_CPE) && ((hSbrElement->elInfo.fDualMono == 1))) { hSbrElement->sbrConfigData.stereoMode = SBR_LEFT_RIGHT; } else { hSbrElement->sbrConfigData.stereoMode = params->stereoMode; } } else { hSbrElement->sbrConfigData.stereoMode = SBR_MONO; } hSbrElement->sbrConfigData.frameSize = params->sbrFrameSize; hSbrElement->sbrConfigData.sampleFreq = params->downSampleFactor * params->codecSettings.sampleFreq; hSbrElement->sbrBitstreamData.CountSendHeaderData = 0; if (params->SendHeaderDataTime > 0) { if (headerPeriod == -1) { hSbrElement->sbrBitstreamData.NrSendHeaderData = (INT)( params->SendHeaderDataTime * hSbrElement->sbrConfigData.sampleFreq / (1000 * hSbrElement->sbrConfigData.frameSize)); hSbrElement->sbrBitstreamData.NrSendHeaderData = fixMax(hSbrElement->sbrBitstreamData.NrSendHeaderData, 1); } else { /* assure header period at least once per second */ hSbrElement->sbrBitstreamData.NrSendHeaderData = fixMin( fixMax(headerPeriod, 1), (hSbrElement->sbrConfigData.sampleFreq / hSbrElement->sbrConfigData.frameSize)); } } else { hSbrElement->sbrBitstreamData.NrSendHeaderData = 0; } hSbrElement->sbrHeaderData.sbr_data_extra = params->sbr_data_extra; hSbrElement->sbrBitstreamData.HeaderActive = 0; hSbrElement->sbrBitstreamData.rightBorderFIX = 0; hSbrElement->sbrHeaderData.sbr_start_frequency = params->startFreq; hSbrElement->sbrHeaderData.sbr_stop_frequency = params->stopFreq; hSbrElement->sbrHeaderData.sbr_xover_band = 0; hSbrElement->sbrHeaderData.sbr_lc_stereo_mode = 0; /* data_extra */ if (params->sbr_xpos_ctrl != SBR_XPOS_CTRL_DEFAULT) hSbrElement->sbrHeaderData.sbr_data_extra = 1; hSbrElement->sbrHeaderData.sbr_amp_res = (AMP_RES)params->amp_res; /* header_extra_1 */ hSbrElement->sbrHeaderData.freqScale = params->freqScale; hSbrElement->sbrHeaderData.alterScale = params->alterScale; hSbrElement->sbrHeaderData.sbr_noise_bands = params->sbr_noise_bands; hSbrElement->sbrHeaderData.header_extra_1 = 0; if ((params->freqScale != SBR_FREQ_SCALE_DEFAULT) || (params->alterScale != SBR_ALTER_SCALE_DEFAULT) || (params->sbr_noise_bands != SBR_NOISE_BANDS_DEFAULT)) { hSbrElement->sbrHeaderData.header_extra_1 = 1; } /* header_extra_2 */ hSbrElement->sbrHeaderData.sbr_limiter_bands = params->sbr_limiter_bands; hSbrElement->sbrHeaderData.sbr_limiter_gains = params->sbr_limiter_gains; if ((hSbrElement->sbrConfigData.sampleFreq > 48000) && (hSbrElement->sbrHeaderData.sbr_start_frequency >= 9)) { hSbrElement->sbrHeaderData.sbr_limiter_gains = SBR_LIMITER_GAINS_INFINITE; } hSbrElement->sbrHeaderData.sbr_interpol_freq = params->sbr_interpol_freq; hSbrElement->sbrHeaderData.sbr_smoothing_length = params->sbr_smoothing_length; hSbrElement->sbrHeaderData.header_extra_2 = 0; if ((params->sbr_limiter_bands != SBR_LIMITER_BANDS_DEFAULT) || (params->sbr_limiter_gains != SBR_LIMITER_GAINS_DEFAULT) || (params->sbr_interpol_freq != SBR_INTERPOL_FREQ_DEFAULT) || (params->sbr_smoothing_length != SBR_SMOOTHING_LENGTH_DEFAULT)) { hSbrElement->sbrHeaderData.header_extra_2 = 1; } /* other switches */ hSbrElement->sbrConfigData.useWaveCoding = params->useWaveCoding; hSbrElement->sbrConfigData.useParametricCoding = params->parametricCoding; hSbrElement->sbrConfigData.thresholdAmpResFF_m = params->threshold_AmpRes_FF_m; hSbrElement->sbrConfigData.thresholdAmpResFF_e = params->threshold_AmpRes_FF_e; /* init freq band table */ if (updateFreqBandTable(&hSbrElement->sbrConfigData, &hSbrElement->sbrHeaderData, params->downSampleFactor)) { return (1); } /* now create envelope ext and QMF for each available channel */ for (ch = 0; ch < hSbrElement->sbrConfigData.nChannels; ch++) { if (initEnvChannel(&hSbrElement->sbrConfigData, &hSbrElement->sbrHeaderData, &hSbrElement->sbrChannel[ch]->hEnvChannel, params, statesInitFlag, ch, dynamic_RAM)) { return (1); } } /* nChannels */ /* reset and intialize analysis qmf */ for (ch = 0; ch < ((hSbrElement->elInfo.fParametricStereo) ? 2 : hSbrElement->sbrConfigData.nChannels); ch++) { int err; UINT qmfFlags = (hSbrElement->sbrConfigData.sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) ? QMF_FLAG_CLDFB : 0; if (statesInitFlag) qmfFlags &= ~QMF_FLAG_KEEP_STATES; else qmfFlags |= QMF_FLAG_KEEP_STATES; err = qmfInitAnalysisFilterBank( hSbrElement->hQmfAnalysis[ch], (FIXP_QAS *)hSbrElement->hQmfAnalysis[ch]->FilterStates, hSbrElement->sbrConfigData.noQmfSlots, hSbrElement->sbrConfigData.noQmfBands, hSbrElement->sbrConfigData.noQmfBands, hSbrElement->sbrConfigData.noQmfBands, qmfFlags); if (0 != err) { return err; } } /* */ hSbrElement->CmonData.xOverFreq = hSbrElement->sbrConfigData.xOverFreq; hSbrElement->CmonData.dynBwEnabled = (params->dynBwSupported && params->dynBwEnabled); hSbrElement->CmonData.dynXOverFreqEnc = FDKsbrEnc_SbrGetXOverFreq(hSbrElement, hSbrElement->CmonData.xOverFreq); for (i = 0; i < 5; i++) hSbrElement->dynXOverFreqDelay[i] = hSbrElement->CmonData.dynXOverFreqEnc; hSbrElement->CmonData.sbrNumChannels = hSbrElement->sbrConfigData.nChannels; hSbrElement->sbrConfigData.dynXOverFreq = hSbrElement->CmonData.xOverFreq; /* Update Bandwith to be passed to the core encoder */ *coreBandWith = hSbrElement->CmonData.xOverFreq; return (0); } INT sbrEncoder_GetInBufferSize(int noChannels) { INT temp; temp = (2048); temp += 1024 + MAX_SAMPLE_DELAY; temp *= noChannels; temp *= sizeof(INT_PCM); return temp; } /* * Encode Dummy SBR payload frames to fill the delay lines. */ static INT FDKsbrEnc_DelayCompensation(HANDLE_SBR_ENCODER hEnvEnc, INT_PCM *timeBuffer, UINT timeBufferBufSize) { int n, el; for (n = hEnvEnc->nBitstrDelay; n > 0; n--) { for (el = 0; el < hEnvEnc->noElements; el++) { if (FDKsbrEnc_EnvEncodeFrame( hEnvEnc, el, timeBuffer + hEnvEnc->downsampledOffset / hEnvEnc->nChannels, timeBufferBufSize, NULL, NULL, 1)) return -1; } sbrEncoder_UpdateBuffers(hEnvEnc, timeBuffer, timeBufferBufSize); } return 0; } UINT sbrEncoder_LimitBitRate(UINT bitRate, UINT numChannels, UINT coreSampleRate, AUDIO_OBJECT_TYPE aot) { UINT newBitRate = bitRate; INT index; FDK_ASSERT(numChannels > 0 && numChannels <= 2); if (aot == AOT_PS) { if (numChannels == 1) { index = getPsTuningTableIndex(bitRate, &newBitRate); if (index == INVALID_TABLE_IDX) { bitRate = newBitRate; } } else { return 0; } } index = getSbrTuningTableIndex(bitRate, numChannels, coreSampleRate, aot, &newBitRate); if (index != INVALID_TABLE_IDX) { newBitRate = bitRate; } return newBitRate; } UINT sbrEncoder_IsSingleRatePossible(AUDIO_OBJECT_TYPE aot) { UINT isPossible = (AOT_PS == aot) ? 0 : 1; return isPossible; } /*****************************************************************************/ /* */ /*functionname: sbrEncoder_Init_delay */ /*description: Determine Delay balancing and new encoder delay */ /* */ /*returns: - error status */ /*input: - frame length of the core (i.e. e.g. AAC) */ /* - number of channels */ /* - downsample factor (1 for downsampled, 2 for dual-rate SBR) */ /* - low delay presence */ /* - ps presence */ /* - downsampling method: QMF-, time domain or no downsampling */ /* - various delay values (see DELAY_PARAM struct description) */ /* */ /*Example: Delay balancing for a HE-AACv1 encoder (time-domain downsampling) */ /*========================================================================== */ /* */ /* +--------+ +--------+ +--------+ +--------+ +--------+ */ /* |core | |ds 2:1 | |AAC | |QMF | |QMF | */ /* +-+path +------------+ +-+core +-+analysis+-+overlap +-+ */ /* | |offset | | | | | |32 bands| | | | */ /* | +--------+ +--------+ +--------+ +--------+ +--------+ | */ /* | core path +-------++ */ /* | |QMF | */ /*->+ +synth. +-> */ /* | |64 bands| */ /* | +-------++ */ /* | +--------+ +--------+ +--------+ +--------+ | */ /* | |SBR path| |QMF | |subband | |bs delay| | */ /* +-+offset +-+analysis+-+sample +-+(full +-----------------------+ */ /* | | |64 bands| |buffer | | frames)| */ /* +--------+ +--------+ +--------+ +--------+ */ /* SBR path */ /* */ /*****************************************************************************/ static INT sbrEncoder_Init_delay( const int coreFrameLength, /* input */ const int numChannels, /* input */ const int downSampleFactor, /* input */ const int lowDelay, /* input */ const int usePs, /* input */ const int is212, /* input */ const SBRENC_DS_TYPE downsamplingMethod, /* input */ DELAY_PARAM *hDelayParam /* input/output */ ) { int delayCorePath = 0; /* delay in core path */ int delaySbrPath = 0; /* delay difference in QMF aka SBR path */ int delayInput2Core = 0; /* delay from the input to the core */ int delaySbrDec = 0; /* delay of the decoder's SBR module */ int delayCore = hDelayParam->delay; /* delay of the core */ /* Added delay by the SBR delay initialization */ int corePathOffset = 0; /* core path */ int sbrPathOffset = 0; /* sbr path */ int bitstreamDelay = 0; /* sbr path, framewise */ int flCore = coreFrameLength; /* core frame length */ int returnValue = 0; /* return value - 0 means: no error */ /* 1) Calculate actual delay for core and SBR path */ if (is212) { delayCorePath = DELAY_COREPATH_ELDv2SBR(flCore, downSampleFactor); delaySbrPath = DELAY_ELDv2SBR(flCore, downSampleFactor); delaySbrDec = ((flCore) / 2) * (downSampleFactor); } else if (lowDelay) { delayCorePath = DELAY_COREPATH_ELDSBR(flCore, downSampleFactor); delaySbrPath = DELAY_ELDSBR(flCore, downSampleFactor); delaySbrDec = DELAY_QMF_POSTPROC(downSampleFactor); } else if (usePs) { delayCorePath = DELAY_COREPATH_PS(flCore, downSampleFactor); delaySbrPath = DELAY_PS(flCore, downSampleFactor); delaySbrDec = DELAY_COREPATH_SBR(flCore, downSampleFactor); } else { delayCorePath = DELAY_COREPATH_SBR(flCore, downSampleFactor); delaySbrPath = DELAY_SBR(flCore, downSampleFactor); delaySbrDec = DELAY_COREPATH_SBR(flCore, downSampleFactor); } delayCorePath += delayCore * downSampleFactor; delayCorePath += (downsamplingMethod == SBRENC_DS_TIME) ? hDelayParam->dsDelay : 0; /* 2) Manage coupling of paths */ if (downsamplingMethod == SBRENC_DS_QMF && delayCorePath > delaySbrPath) { /* In case of QMF downsampling, both paths are coupled, i.e. the SBR path offset would be added to both the SBR path and to the core path as well, thus making it impossible to achieve delay balancing. To overcome that problem, a framewise delay is added to the SBR path first, until the overall delay of the core path is shorter than the delay of the SBR path. When this is achieved, the missing delay difference can be added as downsampled offset to the core path. */ while (delayCorePath > delaySbrPath) { /* Add one frame delay to SBR path */ delaySbrPath += flCore * downSampleFactor; bitstreamDelay += 1; } } /* 3) Calculate necessary additional delay to balance the paths */ if (delayCorePath > delaySbrPath) { /* Delay QMF input */ while (delayCorePath > delaySbrPath + (int)flCore * (int)downSampleFactor) { /* Do bitstream frame-wise delay balancing if there are more than SBR framelength samples delay difference */ delaySbrPath += flCore * downSampleFactor; bitstreamDelay += 1; } /* Multiply input offset by input channels */ corePathOffset = 0; sbrPathOffset = (delayCorePath - delaySbrPath) * numChannels; } else { /* Delay AAC data */ /* Multiply downsampled offset by AAC core channels. Divide by 2 because of half samplerate of downsampled data. */ corePathOffset = ((delaySbrPath - delayCorePath) * numChannels) >> (downSampleFactor - 1); sbrPathOffset = 0; } /* 4) Calculate delay from input to core */ if (usePs) { delayInput2Core = (DELAY_QMF_ANA(downSampleFactor) + DELAY_QMF_DS + DELAY_HYB_SYN) + (downSampleFactor * corePathOffset) + 1; } else if (downsamplingMethod == SBRENC_DS_TIME) { delayInput2Core = corePathOffset + hDelayParam->dsDelay; } else { delayInput2Core = corePathOffset; } /* 6) Set output parameters */ hDelayParam->delay = FDKmax(delayCorePath, delaySbrPath); /* overall delay */ hDelayParam->sbrDecDelay = delaySbrDec; /* SBR decoder delay */ hDelayParam->delayInput2Core = delayInput2Core; /* delay input - core */ hDelayParam->bitstrDelay = bitstreamDelay; /* bitstream delay, in frames */ hDelayParam->corePathOffset = corePathOffset; /* offset added to core path */ hDelayParam->sbrPathOffset = sbrPathOffset; /* offset added to SBR path */ return returnValue; } /***************************************************************************** functionname: sbrEncoder_Init description: initializes the SBR encoder returns: error status *****************************************************************************/ INT sbrEncoder_Init(HANDLE_SBR_ENCODER hSbrEncoder, SBR_ELEMENT_INFO elInfo[(8)], int noElements, INT_PCM *inputBuffer, UINT inputBufferBufSize, INT *coreBandwidth, INT *inputBufferOffset, INT *numChannels, const UINT syntaxFlags, INT *coreSampleRate, UINT *downSampleFactor, INT *frameLength, AUDIO_OBJECT_TYPE aot, int *delay, int transformFactor, const int headerPeriod, ULONG statesInitFlag) { HANDLE_ERROR_INFO errorInfo = noError; sbrConfiguration sbrConfig[(8)]; INT error = 0; INT lowestBandwidth; /* Save input parameters */ INT inputSampleRate = *coreSampleRate; int coreFrameLength = *frameLength; int inputBandWidth = *coreBandwidth; int inputChannels = *numChannels; SBRENC_DS_TYPE downsamplingMethod = SBRENC_DS_NONE; int highestSbrStartFreq, highestSbrStopFreq; int lowDelay = 0; int usePs = 0; int is212 = 0; DELAY_PARAM delayParam; /* check whether SBR setting is available for the current encoder * configuration (bitrate, samplerate) */ if (!sbrEncoder_IsSingleRatePossible(aot)) { *downSampleFactor = 2; } if (aot == AOT_PS) { usePs = 1; } if (aot == AOT_ER_AAC_ELD) { lowDelay = 1; } else if (aot == AOT_ER_AAC_LD) { error = 1; goto bail; } /* Parametric Stereo */ if (usePs) { if (*numChannels == 2 && noElements == 1) { /* Override Element type in case of Parametric stereo */ elInfo[0].elType = ID_SCE; elInfo[0].fParametricStereo = 1; elInfo[0].nChannelsInEl = 1; /* core encoder gets downmixed mono signal */ *numChannels = 1; } else { error = 1; goto bail; } } /* usePs */ /* set the core's sample rate */ switch (*downSampleFactor) { case 1: *coreSampleRate = inputSampleRate; downsamplingMethod = SBRENC_DS_NONE; break; case 2: *coreSampleRate = inputSampleRate >> 1; downsamplingMethod = usePs ? SBRENC_DS_QMF : SBRENC_DS_TIME; break; default: *coreSampleRate = inputSampleRate >> 1; return 0; /* return error */ } /* check whether SBR setting is available for the current encoder * configuration (bitrate, coreSampleRate) */ { int el, coreEl; /* Check if every element config is feasible */ for (coreEl = 0; coreEl < noElements; coreEl++) { /* SBR only handles SCE and CPE's */ if (elInfo[coreEl].elType != ID_SCE && elInfo[coreEl].elType != ID_CPE) { continue; } /* check if desired configuration is available */ if (!FDKsbrEnc_IsSbrSettingAvail(elInfo[coreEl].bitRate, 0, elInfo[coreEl].nChannelsInEl, inputSampleRate, *coreSampleRate, aot)) { error = 1; goto bail; } } hSbrEncoder->nChannels = *numChannels; hSbrEncoder->frameSize = coreFrameLength * *downSampleFactor; hSbrEncoder->downsamplingMethod = downsamplingMethod; hSbrEncoder->downSampleFactor = *downSampleFactor; hSbrEncoder->estimateBitrate = 0; hSbrEncoder->inputDataDelay = 0; is212 = ((aot == AOT_ER_AAC_ELD) && (syntaxFlags & AC_LD_MPS)) ? 1 : 0; /* Open SBR elements */ el = -1; highestSbrStartFreq = highestSbrStopFreq = 0; lowestBandwidth = 99999; /* Loop through each core encoder element and get a matching SBR element * config */ for (coreEl = 0; coreEl < noElements; coreEl++) { /* SBR only handles SCE and CPE's */ if (elInfo[coreEl].elType == ID_SCE || elInfo[coreEl].elType == ID_CPE) { el++; } else { continue; } /* Set parametric Stereo Flag. */ if (usePs) { elInfo[coreEl].fParametricStereo = 1; } else { elInfo[coreEl].fParametricStereo = 0; } /* * Init sbrConfig structure */ if (!FDKsbrEnc_InitializeSbrDefaults(&sbrConfig[el], *downSampleFactor, coreFrameLength, IS_LOWDELAY(aot))) { error = 1; goto bail; } /* * Modify sbrConfig structure according to Element parameters */ if (!FDKsbrEnc_AdjustSbrSettings( &sbrConfig[el], elInfo[coreEl].bitRate, elInfo[coreEl].nChannelsInEl, *coreSampleRate, inputSampleRate, transformFactor, 24000, 0, 0, /* useSpeechConfig */ 0, /* lcsMode */ usePs, /* bParametricStereo */ aot)) { error = 1; goto bail; } /* Find common frequency border for all SBR elements */ highestSbrStartFreq = fixMax(highestSbrStartFreq, sbrConfig[el].startFreq); highestSbrStopFreq = fixMax(highestSbrStopFreq, sbrConfig[el].stopFreq); } /* first element loop */ /* Set element count (can be less than core encoder element count) */ hSbrEncoder->noElements = el + 1; FDKsbrEnc_Reallocate(hSbrEncoder, elInfo, noElements); for (el = 0; el < hSbrEncoder->noElements; el++) { int bandwidth = *coreBandwidth; /* Use lowest common bandwidth */ sbrConfig[el].startFreq = highestSbrStartFreq; sbrConfig[el].stopFreq = highestSbrStopFreq; /* initialize SBR element, and get core bandwidth */ error = FDKsbrEnc_EnvInit(hSbrEncoder->sbrElement[el], &sbrConfig[el], &bandwidth, aot, el, headerPeriod, statesInitFlag, hSbrEncoder->downsamplingMethod, hSbrEncoder->dynamicRam); if (error != 0) { error = 2; goto bail; } /* Get lowest core encoder bandwidth to be returned later. */ lowestBandwidth = fixMin(lowestBandwidth, bandwidth); } /* second element loop */ /* Initialize a downsampler for each channel in each SBR element */ if (hSbrEncoder->downsamplingMethod == SBRENC_DS_TIME) { for (el = 0; el < hSbrEncoder->noElements; el++) { HANDLE_SBR_ELEMENT hSbrEl = hSbrEncoder->sbrElement[el]; INT Wc, ch; Wc = 500; /* Cutoff frequency with full bandwidth */ for (ch = 0; ch < hSbrEl->elInfo.nChannelsInEl; ch++) { FDKaacEnc_InitDownsampler(&hSbrEl->sbrChannel[ch]->downSampler, Wc, *downSampleFactor); FDK_ASSERT(hSbrEl->sbrChannel[ch]->downSampler.delay <= MAX_DS_FILTER_DELAY); } } /* third element loop */ /* lfe */ FDKaacEnc_InitDownsampler(&hSbrEncoder->lfeDownSampler, 0, *downSampleFactor); } /* Get delay information */ delayParam.dsDelay = hSbrEncoder->sbrElement[0]->sbrChannel[0]->downSampler.delay; delayParam.delay = *delay; error = sbrEncoder_Init_delay(coreFrameLength, *numChannels, *downSampleFactor, lowDelay, usePs, is212, downsamplingMethod, &delayParam); if (error != 0) { error = 3; goto bail; } hSbrEncoder->nBitstrDelay = delayParam.bitstrDelay; hSbrEncoder->sbrDecDelay = delayParam.sbrDecDelay; hSbrEncoder->inputDataDelay = delayParam.delayInput2Core; /* Assign core encoder Bandwidth */ *coreBandwidth = lowestBandwidth; /* Estimate sbr bitrate, 2.5 kBit/s per sbr channel */ hSbrEncoder->estimateBitrate += 2500 * (*numChannels); /* Initialize bitstream buffer for each element */ for (el = 0; el < hSbrEncoder->noElements; el++) { FDKsbrEnc_bsBufInit(hSbrEncoder->sbrElement[el], delayParam.bitstrDelay); } /* initialize parametric stereo */ if (usePs) { PSENC_CONFIG psEncConfig; FDK_ASSERT(hSbrEncoder->noElements == 1); INT psTuningTableIdx = getPsTuningTableIndex(elInfo[0].bitRate, NULL); psEncConfig.frameSize = coreFrameLength; // sbrConfig.sbrFrameSize; psEncConfig.qmfFilterMode = 0; psEncConfig.sbrPsDelay = 0; /* tuning parameters */ if (psTuningTableIdx != INVALID_TABLE_IDX) { psEncConfig.nStereoBands = psTuningTable[psTuningTableIdx].nStereoBands; psEncConfig.maxEnvelopes = psTuningTable[psTuningTableIdx].nEnvelopes; psEncConfig.iidQuantErrorThreshold = (FIXP_DBL)psTuningTable[psTuningTableIdx].iidQuantErrorThreshold; /* calculation is not quite linear, increased number of envelopes causes * more bits */ /* assume avg. 50 bits per frame for 10 stereo bands / 1 envelope * configuration */ hSbrEncoder->estimateBitrate += ((((*coreSampleRate) * 5 * psEncConfig.nStereoBands * psEncConfig.maxEnvelopes) / hSbrEncoder->frameSize)); } else { error = ERROR(CDI, "Invalid ps tuning table index."); goto bail; } qmfInitSynthesisFilterBank( &hSbrEncoder->qmfSynthesisPS, (FIXP_DBL *)hSbrEncoder->qmfSynthesisPS.FilterStates, hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfSlots, hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfBands >> 1, hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfBands >> 1, hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfBands >> 1, (statesInitFlag) ? 0 : QMF_FLAG_KEEP_STATES); if (errorInfo == noError) { /* update delay */ psEncConfig.sbrPsDelay = FDKsbrEnc_GetEnvEstDelay(&hSbrEncoder->sbrElement[0] ->sbrChannel[0] ->hEnvChannel.sbrExtractEnvelope); errorInfo = PSEnc_Init(hSbrEncoder->hParametricStereo, &psEncConfig, hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfSlots, hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfBands, hSbrEncoder->dynamicRam); } } hSbrEncoder->downsampledOffset = delayParam.corePathOffset; hSbrEncoder->bufferOffset = delayParam.sbrPathOffset; *delay = delayParam.delay; { hSbrEncoder->downmixSize = coreFrameLength * (*numChannels); } /* Delay Compensation: fill bitstream delay buffer with zero input signal */ if (hSbrEncoder->nBitstrDelay > 0) { error = FDKsbrEnc_DelayCompensation(hSbrEncoder, inputBuffer, inputBufferBufSize); if (error != 0) goto bail; } /* Set Output frame length */ *frameLength = coreFrameLength * *downSampleFactor; /* Input buffer offset */ *inputBufferOffset = fixMax(delayParam.sbrPathOffset, delayParam.corePathOffset); } return error; bail: /* Restore input settings */ *coreSampleRate = inputSampleRate; *frameLength = coreFrameLength; *numChannels = inputChannels; *coreBandwidth = inputBandWidth; return error; } INT sbrEncoder_EncodeFrame(HANDLE_SBR_ENCODER hSbrEncoder, INT_PCM *samples, UINT samplesBufSize, UINT sbrDataBits[(8)], UCHAR sbrData[(8)][MAX_PAYLOAD_SIZE]) { INT error; int el; for (el = 0; el < hSbrEncoder->noElements; el++) { if (hSbrEncoder->sbrElement[el] != NULL) { error = FDKsbrEnc_EnvEncodeFrame( hSbrEncoder, el, samples + hSbrEncoder->downsampledOffset / hSbrEncoder->nChannels, samplesBufSize, &sbrDataBits[el], sbrData[el], 0); if (error) return error; } } error = FDKsbrEnc_Downsample( hSbrEncoder, samples + hSbrEncoder->downsampledOffset / hSbrEncoder->nChannels, samplesBufSize, hSbrEncoder->nChannels, &sbrDataBits[el], sbrData[el], 0); if (error) return error; return 0; } INT sbrEncoder_UpdateBuffers(HANDLE_SBR_ENCODER hSbrEncoder, INT_PCM *timeBuffer, UINT timeBufferBufSize) { if (hSbrEncoder->downsampledOffset > 0) { int c; int nd = hSbrEncoder->downmixSize / hSbrEncoder->nChannels; for (c = 0; c < hSbrEncoder->nChannels; c++) { /* Move delayed downsampled data */ FDKmemcpy(timeBuffer + timeBufferBufSize * c, timeBuffer + timeBufferBufSize * c + nd, sizeof(INT_PCM) * (hSbrEncoder->downsampledOffset / hSbrEncoder->nChannels)); } } else { int c; for (c = 0; c < hSbrEncoder->nChannels; c++) { /* Move delayed input data */ FDKmemcpy( timeBuffer + timeBufferBufSize * c, timeBuffer + timeBufferBufSize * c + hSbrEncoder->frameSize, sizeof(INT_PCM) * hSbrEncoder->bufferOffset / hSbrEncoder->nChannels); } } if (hSbrEncoder->nBitstrDelay > 0) { int el; for (el = 0; el < hSbrEncoder->noElements; el++) { FDKmemmove( hSbrEncoder->sbrElement[el]->payloadDelayLine[0], hSbrEncoder->sbrElement[el]->payloadDelayLine[1], sizeof(UCHAR) * (hSbrEncoder->nBitstrDelay * MAX_PAYLOAD_SIZE)); FDKmemmove(&hSbrEncoder->sbrElement[el]->payloadDelayLineSize[0], &hSbrEncoder->sbrElement[el]->payloadDelayLineSize[1], sizeof(UINT) * (hSbrEncoder->nBitstrDelay)); } } return 0; } INT sbrEncoder_SendHeader(HANDLE_SBR_ENCODER hSbrEncoder) { INT error = -1; if (hSbrEncoder) { int el; for (el = 0; el < hSbrEncoder->noElements; el++) { if ((hSbrEncoder->noElements == 1) && (hSbrEncoder->sbrElement[0]->elInfo.fParametricStereo == 1)) { hSbrEncoder->sbrElement[el]->sbrBitstreamData.CountSendHeaderData = hSbrEncoder->sbrElement[el]->sbrBitstreamData.NrSendHeaderData - 1; } else { hSbrEncoder->sbrElement[el]->sbrBitstreamData.CountSendHeaderData = 0; } } error = 0; } return error; } INT sbrEncoder_ContainsHeader(HANDLE_SBR_ENCODER hSbrEncoder) { INT sbrHeader = 1; if (hSbrEncoder) { int el; for (el = 0; el < hSbrEncoder->noElements; el++) { sbrHeader &= (hSbrEncoder->sbrElement[el]->sbrBitstreamData.HeaderActiveDelay == 1) ? 1 : 0; } } return sbrHeader; } INT sbrEncoder_GetHeaderDelay(HANDLE_SBR_ENCODER hSbrEncoder) { INT delay = -1; if (hSbrEncoder) { if ((hSbrEncoder->noElements == 1) && (hSbrEncoder->sbrElement[0]->elInfo.fParametricStereo == 1)) { delay = hSbrEncoder->nBitstrDelay + 1; } else { delay = hSbrEncoder->nBitstrDelay; } } return delay; } INT sbrEncoder_GetBsDelay(HANDLE_SBR_ENCODER hSbrEncoder) { INT delay = -1; if (hSbrEncoder) { delay = hSbrEncoder->nBitstrDelay; } return delay; } INT sbrEncoder_SAPPrepare(HANDLE_SBR_ENCODER hSbrEncoder) { INT error = -1; if (hSbrEncoder) { int el; for (el = 0; el < hSbrEncoder->noElements; el++) { hSbrEncoder->sbrElement[el]->sbrBitstreamData.rightBorderFIX = 1; } error = 0; } return error; } INT sbrEncoder_GetEstimateBitrate(HANDLE_SBR_ENCODER hSbrEncoder) { INT estimateBitrate = 0; if (hSbrEncoder) { estimateBitrate += hSbrEncoder->estimateBitrate; } return estimateBitrate; } INT sbrEncoder_GetInputDataDelay(HANDLE_SBR_ENCODER hSbrEncoder) { INT delay = -1; if (hSbrEncoder) { delay = hSbrEncoder->inputDataDelay; } return delay; } INT sbrEncoder_GetSbrDecDelay(HANDLE_SBR_ENCODER hSbrEncoder) { INT delay = -1; if (hSbrEncoder) { delay = hSbrEncoder->sbrDecDelay; } return delay; } INT sbrEncoder_GetLibInfo(LIB_INFO *info) { int i; if (info == NULL) { return -1; } /* search for next free tab */ for (i = 0; i < FDK_MODULE_LAST; i++) { if (info[i].module_id == FDK_NONE) break; } if (i == FDK_MODULE_LAST) { return -1; } info += i; info->module_id = FDK_SBRENC; info->version = LIB_VERSION(SBRENCODER_LIB_VL0, SBRENCODER_LIB_VL1, SBRENCODER_LIB_VL2); LIB_VERSION_STRING(info); #ifdef SUPPRESS_BUILD_DATE_INFO info->build_date = ""; info->build_time = ""; #else info->build_date = __DATE__; info->build_time = __TIME__; #endif info->title = "SBR Encoder"; /* Set flags */ info->flags = 0 | CAPF_SBR_HQ | CAPF_SBR_PS_MPEG; /* End of flags */ return 0; } fdk-aac-2.0.2/libSBRenc/src/sbr_misc.cpp000066400000000000000000000217571404225310700177120ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Sbr miscellaneous helper functions $Revision: 36750 $ */ #include "sbr_misc.h" void FDKsbrEnc_Shellsort_fract(FIXP_DBL *in, INT n) { FIXP_DBL v; INT i, j; INT inc = 1; do inc = 3 * inc + 1; while (inc <= n); do { inc = inc / 3; for (i = inc + 1; i <= n; i++) { v = in[i - 1]; j = i; while (in[j - inc - 1] > v) { in[j - 1] = in[j - inc - 1]; j -= inc; if (j <= inc) break; } in[j - 1] = v; } } while (inc > 1); } /* Sorting routine */ void FDKsbrEnc_Shellsort_int(INT *in, INT n) { INT i, j, v; INT inc = 1; do inc = 3 * inc + 1; while (inc <= n); do { inc = inc / 3; for (i = inc + 1; i <= n; i++) { v = in[i - 1]; j = i; while (in[j - inc - 1] > v) { in[j - 1] = in[j - inc - 1]; j -= inc; if (j <= inc) break; } in[j - 1] = v; } } while (inc > 1); } /******************************************************************************* Functionname: FDKsbrEnc_AddVecLeft ******************************************************************************* Description: Arguments: INT* dst, INT* length_dst, INT* src, INT length_src Return: none *******************************************************************************/ void FDKsbrEnc_AddVecLeft(INT *dst, INT *length_dst, INT *src, INT length_src) { INT i; for (i = length_src - 1; i >= 0; i--) FDKsbrEnc_AddLeft(dst, length_dst, src[i]); } /******************************************************************************* Functionname: FDKsbrEnc_AddLeft ******************************************************************************* Description: Arguments: INT* vector, INT* length_vector, INT value Return: none *******************************************************************************/ void FDKsbrEnc_AddLeft(INT *vector, INT *length_vector, INT value) { INT i; for (i = *length_vector; i > 0; i--) vector[i] = vector[i - 1]; vector[0] = value; (*length_vector)++; } /******************************************************************************* Functionname: FDKsbrEnc_AddRight ******************************************************************************* Description: Arguments: INT* vector, INT* length_vector, INT value Return: none *******************************************************************************/ void FDKsbrEnc_AddRight(INT *vector, INT *length_vector, INT value) { vector[*length_vector] = value; (*length_vector)++; } /******************************************************************************* Functionname: FDKsbrEnc_AddVecRight ******************************************************************************* Description: Arguments: INT* dst, INT* length_dst, INT* src, INT length_src) Return: none *******************************************************************************/ void FDKsbrEnc_AddVecRight(INT *dst, INT *length_dst, INT *src, INT length_src) { INT i; for (i = 0; i < length_src; i++) FDKsbrEnc_AddRight(dst, length_dst, src[i]); } /***************************************************************************** functionname: FDKsbrEnc_LSI_divide_scale_fract description: Calculates division with best precision and scales the result. return: num*scale/denom *****************************************************************************/ FIXP_DBL FDKsbrEnc_LSI_divide_scale_fract(FIXP_DBL num, FIXP_DBL denom, FIXP_DBL scale) { FIXP_DBL tmp = FL2FXCONST_DBL(0.0f); if (num != FL2FXCONST_DBL(0.0f)) { INT shiftCommon; INT shiftNum = CountLeadingBits(num); INT shiftDenom = CountLeadingBits(denom); INT shiftScale = CountLeadingBits(scale); num = num << shiftNum; scale = scale << shiftScale; tmp = fMultDiv2(num, scale); if (denom > (tmp >> fixMin(shiftNum + shiftScale - 1, (DFRACT_BITS - 1)))) { denom = denom << shiftDenom; tmp = schur_div(tmp, denom, 15); shiftCommon = fixMin((shiftNum - shiftDenom + shiftScale - 1), (DFRACT_BITS - 1)); if (shiftCommon < 0) tmp <<= -shiftCommon; else tmp >>= shiftCommon; } else { tmp = /*FL2FXCONST_DBL(1.0)*/ (FIXP_DBL)MAXVAL_DBL; } } return (tmp); } fdk-aac-2.0.2/libSBRenc/src/sbr_misc.h000066400000000000000000000126761404225310700173570ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Sbr miscellaneous helper functions prototypes $Revision: 92790 $ \author */ #ifndef SBR_MISC_H #define SBR_MISC_H #include "sbr_encoder.h" /* Sorting routines */ void FDKsbrEnc_Shellsort_fract(FIXP_DBL *in, INT n); void FDKsbrEnc_Shellsort_int(INT *in, INT n); void FDKsbrEnc_AddLeft(INT *vector, INT *length_vector, INT value); void FDKsbrEnc_AddRight(INT *vector, INT *length_vector, INT value); void FDKsbrEnc_AddVecLeft(INT *dst, INT *length_dst, INT *src, INT length_src); void FDKsbrEnc_AddVecRight(INT *dst, INT *length_vector_dst, INT *src, INT length_src); FIXP_DBL FDKsbrEnc_LSI_divide_scale_fract(FIXP_DBL num, FIXP_DBL denom, FIXP_DBL scale); #endif fdk-aac-2.0.2/libSBRenc/src/sbrenc_freq_sca.cpp000066400000000000000000000536071404225310700212270ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief frequency scale $Revision: 95225 $ */ #include "sbrenc_freq_sca.h" #include "sbr_misc.h" #include "genericStds.h" /* StartFreq */ static INT getStartFreq(INT fsCore, const INT start_freq); /* StopFreq */ static INT getStopFreq(INT fsCore, const INT stop_freq); static INT numberOfBands(INT b_p_o, INT start, INT stop, FIXP_DBL warp_factor); static void CalcBands(INT *diff, INT start, INT stop, INT num_bands); static INT modifyBands(INT max_band, INT *diff, INT length); static void cumSum(INT start_value, INT *diff, INT length, UCHAR *start_adress); /******************************************************************************* Functionname: FDKsbrEnc_getSbrStartFreqRAW ******************************************************************************* Description: Arguments: Return: *******************************************************************************/ INT FDKsbrEnc_getSbrStartFreqRAW(INT startFreq, INT fsCore) { INT result; if (startFreq < 0 || startFreq > 15) { return -1; } /* Update startFreq struct */ result = getStartFreq(fsCore, startFreq); result = (result * (fsCore >> 5) + 1) >> 1; /* (result*fsSBR/QMFbands+1)>>1; */ return (result); } /* End FDKsbrEnc_getSbrStartFreqRAW */ /******************************************************************************* Functionname: getSbrStopFreq ******************************************************************************* Description: Arguments: Return: *******************************************************************************/ INT FDKsbrEnc_getSbrStopFreqRAW(INT stopFreq, INT fsCore) { INT result; if (stopFreq < 0 || stopFreq > 13) return -1; /* Uppdate stopFreq struct */ result = getStopFreq(fsCore, stopFreq); result = (result * (fsCore >> 5) + 1) >> 1; /* (result*fsSBR/QMFbands+1)>>1; */ return (result); } /* End getSbrStopFreq */ /******************************************************************************* Functionname: getStartFreq ******************************************************************************* Description: Arguments: fsCore - core sampling rate Return: *******************************************************************************/ static INT getStartFreq(INT fsCore, const INT start_freq) { INT k0_min; switch (fsCore) { case 8000: k0_min = 24; /* (3000 * nQmfChannels / fsSBR ) + 0.5 */ break; case 11025: k0_min = 17; /* (3000 * nQmfChannels / fsSBR ) + 0.5 */ break; case 12000: k0_min = 16; /* (3000 * nQmfChannels / fsSBR ) + 0.5 */ break; case 16000: k0_min = 16; /* (4000 * nQmfChannels / fsSBR ) + 0.5 */ break; case 22050: k0_min = 12; /* (4000 * nQmfChannels / fsSBR ) + 0.5 */ break; case 24000: k0_min = 11; /* (4000 * nQmfChannels / fsSBR ) + 0.5 */ break; case 32000: k0_min = 10; /* (5000 * nQmfChannels / fsSBR ) + 0.5 */ break; case 44100: k0_min = 7; /* (5000 * nQmfChannels / fsSBR ) + 0.5 */ break; case 48000: k0_min = 7; /* (5000 * nQmfChannels / fsSBR ) + 0.5 */ break; case 96000: k0_min = 3; /* (5000 * nQmfChannels / fsSBR ) + 0.5 */ break; default: k0_min = 11; /* illegal fs */ } switch (fsCore) { case 8000: { INT v_offset[] = {-8, -7, -6, -5, -4, -3, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7}; return (k0_min + v_offset[start_freq]); } case 11025: { INT v_offset[] = {-5, -4, -3, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13}; return (k0_min + v_offset[start_freq]); } case 12000: { INT v_offset[] = {-5, -3, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16}; return (k0_min + v_offset[start_freq]); } case 16000: { INT v_offset[] = {-6, -4, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16}; return (k0_min + v_offset[start_freq]); } case 22050: case 24000: case 32000: { INT v_offset[] = {-4, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16, 20}; return (k0_min + v_offset[start_freq]); } case 44100: case 48000: case 96000: { INT v_offset[] = {-2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16, 20, 24}; return (k0_min + v_offset[start_freq]); } default: { INT v_offset[] = {0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16, 20, 24, 28, 33}; return (k0_min + v_offset[start_freq]); } } } /* End getStartFreq */ /******************************************************************************* Functionname: getStopFreq ******************************************************************************* Description: Arguments: Return: *******************************************************************************/ static INT getStopFreq(INT fsCore, const INT stop_freq) { INT result, i; INT k1_min; INT v_dstop[13]; INT *v_stop_freq = NULL; INT v_stop_freq_16[14] = {48, 49, 50, 51, 52, 54, 55, 56, 57, 59, 60, 61, 63, 64}; INT v_stop_freq_22[14] = {35, 37, 38, 40, 42, 44, 46, 48, 51, 53, 56, 58, 61, 64}; INT v_stop_freq_24[14] = {32, 34, 36, 38, 40, 42, 44, 46, 49, 52, 55, 58, 61, 64}; INT v_stop_freq_32[14] = {32, 34, 36, 38, 40, 42, 44, 46, 49, 52, 55, 58, 61, 64}; INT v_stop_freq_44[14] = {23, 25, 27, 29, 32, 34, 37, 40, 43, 47, 51, 55, 59, 64}; INT v_stop_freq_48[14] = {21, 23, 25, 27, 30, 32, 35, 38, 42, 45, 49, 54, 59, 64}; INT v_stop_freq_64[14] = {20, 22, 24, 26, 29, 31, 34, 37, 41, 45, 49, 54, 59, 64}; INT v_stop_freq_88[14] = {15, 17, 19, 21, 23, 26, 29, 33, 37, 41, 46, 51, 57, 64}; INT v_stop_freq_96[14] = {13, 15, 17, 19, 21, 24, 27, 31, 35, 39, 44, 50, 57, 64}; INT v_stop_freq_192[14] = {7, 8, 10, 12, 14, 16, 19, 23, 27, 32, 38, 46, 54, 64}; switch (fsCore) { case 8000: k1_min = 48; v_stop_freq = v_stop_freq_16; break; case 11025: k1_min = 35; v_stop_freq = v_stop_freq_22; break; case 12000: k1_min = 32; v_stop_freq = v_stop_freq_24; break; case 16000: k1_min = 32; v_stop_freq = v_stop_freq_32; break; case 22050: k1_min = 23; v_stop_freq = v_stop_freq_44; break; case 24000: k1_min = 21; v_stop_freq = v_stop_freq_48; break; case 32000: k1_min = 20; v_stop_freq = v_stop_freq_64; break; case 44100: k1_min = 15; v_stop_freq = v_stop_freq_88; break; case 48000: k1_min = 13; v_stop_freq = v_stop_freq_96; break; case 96000: k1_min = 7; v_stop_freq = v_stop_freq_192; break; default: k1_min = 21; /* illegal fs */ } /* Ensure increasing bandwidth */ for (i = 0; i <= 12; i++) { v_dstop[i] = v_stop_freq[i + 1] - v_stop_freq[i]; } FDKsbrEnc_Shellsort_int(v_dstop, 13); /* Sort bandwidth changes */ result = k1_min; for (i = 0; i < stop_freq; i++) { result = result + v_dstop[i]; } return (result); } /* End getStopFreq */ /******************************************************************************* Functionname: FDKsbrEnc_FindStartAndStopBand ******************************************************************************* Description: Arguments: srSbr SBR sampling freqency srCore AAC core sampling freqency noChannels Number of QMF channels startFreq SBR start frequency in QMF bands stopFreq SBR start frequency in QMF bands *k0 Output parameter *k2 Output parameter Return: Error code (0 is OK) *******************************************************************************/ INT FDKsbrEnc_FindStartAndStopBand(const INT srSbr, const INT srCore, const INT noChannels, const INT startFreq, const INT stopFreq, INT *k0, INT *k2) { /* Update startFreq struct */ *k0 = getStartFreq(srCore, startFreq); /* Test if start freq is outside corecoder range */ if (srSbr * noChannels < *k0 * srCore) { return ( 1); /* raise the cross-over frequency and/or lower the number of target bands per octave (or lower the sampling frequency) */ } /*Update stopFreq struct */ if (stopFreq < 14) { *k2 = getStopFreq(srCore, stopFreq); } else if (stopFreq == 14) { *k2 = 2 * *k0; } else { *k2 = 3 * *k0; } /* limit to Nyqvist */ if (*k2 > noChannels) { *k2 = noChannels; } /* Test for invalid k0 k2 combinations */ if ((srCore == 22050) && ((*k2 - *k0) > MAX_FREQ_COEFFS_FS44100)) return (1); /* Number of bands exceeds valid range of MAX_FREQ_COEFFS for fs=44.1kHz */ if ((srCore >= 24000) && ((*k2 - *k0) > MAX_FREQ_COEFFS_FS48000)) return (1); /* Number of bands exceeds valid range of MAX_FREQ_COEFFS for fs>=48kHz */ if ((*k2 - *k0) > MAX_FREQ_COEFFS) return (1); /*Number of bands exceeds valid range of MAX_FREQ_COEFFS */ if ((*k2 - *k0) < 0) return (1); /* Number of bands is negative */ return (0); } /******************************************************************************* Functionname: FDKsbrEnc_UpdateFreqScale ******************************************************************************* Description: Arguments: Return: *******************************************************************************/ INT FDKsbrEnc_UpdateFreqScale(UCHAR *v_k_master, INT *h_num_bands, const INT k0, const INT k2, const INT freqScale, const INT alterScale) { INT b_p_o = 0; /* bands_per_octave */ FIXP_DBL warp = FL2FXCONST_DBL(0.0f); INT dk = 0; /* Internal variables */ INT k1 = 0, i; INT num_bands0; INT num_bands1; INT diff_tot[MAX_OCTAVE + MAX_SECOND_REGION]; INT *diff0 = diff_tot; INT *diff1 = diff_tot + MAX_OCTAVE; INT k2_achived; INT k2_diff; INT incr = 0; /* Init */ if (freqScale == 1) b_p_o = 12; if (freqScale == 2) b_p_o = 10; if (freqScale == 3) b_p_o = 8; if (freqScale > 0) /*Bark*/ { if (alterScale == 0) warp = FL2FXCONST_DBL(0.5f); /* 1.0/(1.0*2.0) */ else warp = FL2FXCONST_DBL(1.0f / 2.6f); /* 1.0/(1.3*2.0); */ if (4 * k2 >= 9 * k0) /*two or more regions (how many times the basis band is copied)*/ { k1 = 2 * k0; num_bands0 = numberOfBands(b_p_o, k0, k1, FL2FXCONST_DBL(0.5f)); num_bands1 = numberOfBands(b_p_o, k1, k2, warp); CalcBands(diff0, k0, k1, num_bands0); /*CalcBands1 => diff0 */ FDKsbrEnc_Shellsort_int(diff0, num_bands0); /*SortBands sort diff0 */ if (diff0[0] == 0) /* too wide FB bands for target tuning */ { return (1); /* raise the cross-over frequency and/or lower the number of target bands per octave (or lower the sampling frequency */ } cumSum(k0, diff0, num_bands0, v_k_master); /* cumsum */ CalcBands(diff1, k1, k2, num_bands1); /* CalcBands2 => diff1 */ FDKsbrEnc_Shellsort_int(diff1, num_bands1); /* SortBands sort diff1 */ if (diff0[num_bands0 - 1] > diff1[0]) /* max(1) > min(2) */ { if (modifyBands(diff0[num_bands0 - 1], diff1, num_bands1)) return (1); } /* Add 2'nd region */ cumSum(k1, diff1, num_bands1, &v_k_master[num_bands0]); *h_num_bands = num_bands0 + num_bands1; /* Output nr of bands */ } else /* one region */ { k1 = k2; num_bands0 = numberOfBands(b_p_o, k0, k1, FL2FXCONST_DBL(0.5f)); CalcBands(diff0, k0, k1, num_bands0); /* CalcBands1 => diff0 */ FDKsbrEnc_Shellsort_int(diff0, num_bands0); /* SortBands sort diff0 */ if (diff0[0] == 0) /* too wide FB bands for target tuning */ { return (1); /* raise the cross-over frequency and/or lower the number of target bands per octave (or lower the sampling frequency */ } cumSum(k0, diff0, num_bands0, v_k_master); /* cumsum */ *h_num_bands = num_bands0; /* Output nr of bands */ } } else /* Linear mode */ { if (alterScale == 0) { dk = 1; num_bands0 = 2 * ((k2 - k0) / 2); /* FLOOR to get to few number of bands*/ } else { dk = 2; num_bands0 = 2 * (((k2 - k0) / dk + 1) / 2); /* ROUND to get closest fit */ } k2_achived = k0 + num_bands0 * dk; k2_diff = k2 - k2_achived; for (i = 0; i < num_bands0; i++) diff_tot[i] = dk; /* If linear scale wasn't achived */ /* and we got wide SBR are */ if (k2_diff < 0) { incr = 1; i = 0; } /* If linear scale wasn't achived */ /* and we got small SBR are */ if (k2_diff > 0) { incr = -1; i = num_bands0 - 1; } /* Adjust diff vector to get sepc. SBR range */ while (k2_diff != 0) { diff_tot[i] = diff_tot[i] - incr; i = i + incr; k2_diff = k2_diff + incr; } cumSum(k0, diff_tot, num_bands0, v_k_master); /* cumsum */ *h_num_bands = num_bands0; /* Output nr of bands */ } if (*h_num_bands < 1) return (1); /*To small sbr area */ return (0); } /* End FDKsbrEnc_UpdateFreqScale */ static INT numberOfBands(INT b_p_o, INT start, INT stop, FIXP_DBL warp_factor) { INT result = 0; /* result = 2* (INT) ( (double)b_p_o * * (double)(FDKlog((double)stop/(double)start)/FDKlog((double)2)) * * (double)FX_DBL2FL(warp_factor) + 0.5); */ result = ((b_p_o * fMult((CalcLdInt(stop) - CalcLdInt(start)), warp_factor) + (FL2FX_DBL(0.5f) >> LD_DATA_SHIFT)) >> ((DFRACT_BITS - 1) - LD_DATA_SHIFT)) << 1; /* do not optimize anymore (rounding!!) */ return (result); } static void CalcBands(INT *diff, INT start, INT stop, INT num_bands) { INT i, qb, qe, qtmp; INT previous; INT current; FIXP_DBL base, exp, tmp; previous = start; for (i = 1; i <= num_bands; i++) { base = fDivNorm((FIXP_DBL)stop, (FIXP_DBL)start, &qb); exp = fDivNorm((FIXP_DBL)i, (FIXP_DBL)num_bands, &qe); tmp = fPow(base, qb, exp, qe, &qtmp); tmp = fMult(tmp, (FIXP_DBL)(start << 24)); current = (INT)scaleValue(tmp, qtmp - 23); current = (current + 1) >> 1; /* rounding*/ diff[i - 1] = current - previous; previous = current; } } /* End CalcBands */ static void cumSum(INT start_value, INT *diff, INT length, UCHAR *start_adress) { INT i; start_adress[0] = start_value; for (i = 1; i <= length; i++) start_adress[i] = start_adress[i - 1] + diff[i - 1]; } /* End cumSum */ static INT modifyBands(INT max_band_previous, INT *diff, INT length) { INT change = max_band_previous - diff[0]; /* Limit the change so that the last band cannot get narrower than the first * one */ if (change > (diff[length - 1] - diff[0]) / 2) change = (diff[length - 1] - diff[0]) / 2; diff[0] += change; diff[length - 1] -= change; FDKsbrEnc_Shellsort_int(diff, length); return (0); } /* End modifyBands */ /******************************************************************************* Functionname: FDKsbrEnc_UpdateHiRes ******************************************************************************* Description: Arguments: Return: *******************************************************************************/ INT FDKsbrEnc_UpdateHiRes(UCHAR *h_hires, INT *num_hires, UCHAR *v_k_master, INT num_master, INT *xover_band) { INT i; INT max1, max2; if ((v_k_master[*xover_band] > 32) || /* v_k_master[*xover_band] > noQMFChannels(dualRate)/divider */ (*xover_band > num_master)) { /* xover_band error, too big for this startFreq. Will be clipped */ /* Calculate maximum value for xover_band */ max1 = 0; max2 = num_master; while ((v_k_master[max1 + 1] < 32) && /* noQMFChannels(dualRate)/divider */ ((max1 + 1) < max2)) { max1++; } *xover_band = max1; } *num_hires = num_master - *xover_band; for (i = *xover_band; i <= num_master; i++) { h_hires[i - *xover_band] = v_k_master[i]; } return (0); } /* End FDKsbrEnc_UpdateHiRes */ /******************************************************************************* Functionname: FDKsbrEnc_UpdateLoRes ******************************************************************************* Description: Arguments: Return: *******************************************************************************/ void FDKsbrEnc_UpdateLoRes(UCHAR *h_lores, INT *num_lores, UCHAR *h_hires, INT num_hires) { INT i; if (num_hires % 2 == 0) /* if even number of hires bands */ { *num_lores = num_hires / 2; /* Use every second lores=hires[0,2,4...] */ for (i = 0; i <= *num_lores; i++) h_lores[i] = h_hires[i * 2]; } else /* odd number of hires which means xover is odd */ { *num_lores = (num_hires + 1) / 2; /* Use lores=hires[0,1,3,5 ...] */ h_lores[0] = h_hires[0]; for (i = 1; i <= *num_lores; i++) { h_lores[i] = h_hires[i * 2 - 1]; } } } /* End FDKsbrEnc_UpdateLoRes */ fdk-aac-2.0.2/libSBRenc/src/sbrenc_freq_sca.h000066400000000000000000000133161404225310700206650ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief frequency scale prototypes $Revision: 92790 $ */ #ifndef SBRENC_FREQ_SCA_H #define SBRENC_FREQ_SCA_H #include "sbr_encoder.h" #include "sbr_def.h" #define MAX_OCTAVE 29 #define MAX_SECOND_REGION 50 INT FDKsbrEnc_UpdateFreqScale(UCHAR *v_k_master, INT *h_num_bands, const INT k0, const INT k2, const INT freq_scale, const INT alter_scale); INT FDKsbrEnc_UpdateHiRes(UCHAR *h_hires, INT *num_hires, UCHAR *v_k_master, INT num_master, INT *xover_band); void FDKsbrEnc_UpdateLoRes(UCHAR *v_lores, INT *num_lores, UCHAR *v_hires, INT num_hires); INT FDKsbrEnc_FindStartAndStopBand(const INT srSbr, const INT srCore, const INT noChannels, const INT startFreq, const INT stop_freq, INT *k0, INT *k2); INT FDKsbrEnc_getSbrStartFreqRAW(INT startFreq, INT fsCore); INT FDKsbrEnc_getSbrStopFreqRAW(INT stopFreq, INT fsCore); #endif fdk-aac-2.0.2/libSBRenc/src/sbrenc_ram.cpp000066400000000000000000000225601404225310700202150ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Memory layout $Revision: 92864 $ This module declares all static and dynamic memory spaces */ #include "sbrenc_ram.h" #include "sbr.h" #include "genericStds.h" C_AALLOC_MEM(Ram_SbrDynamic_RAM, FIXP_DBL, ((SBR_ENC_DYN_RAM_SIZE) / sizeof(FIXP_DBL))) /*! \name StaticSbrData Static memory areas, must not be overwritten in other sections of the encoder */ /* @{ */ /*! static sbr encoder instance for one encoder (2 channels) all major static and dynamic memory areas are located in module sbr_ram and sbr rom */ C_ALLOC_MEM(Ram_SbrEncoder, SBR_ENCODER, 1) C_ALLOC_MEM2(Ram_SbrChannel, SBR_CHANNEL, 1, (8)) C_ALLOC_MEM2(Ram_SbrElement, SBR_ELEMENT, 1, (8)) /*! Filter states for QMF-analysis.
Dimension: #MAXNRSBRCHANNELS * #SBR_QMF_FILTER_LENGTH */ C_AALLOC_MEM2_L(Ram_Sbr_QmfStatesAnalysis, FIXP_QAS, 640, (8), SECT_DATA_L1) /*! Matrix holding the quota values for all estimates, all channels Dimension #MAXNRSBRCHANNELS * +#SBR_QMF_CHANNELS* #MAX_NO_OF_ESTIMATES */ C_ALLOC_MEM2_L(Ram_Sbr_quotaMatrix, FIXP_DBL, (MAX_NO_OF_ESTIMATES * 64), (8), SECT_DATA_L1) /*! Matrix holding the sign values for all estimates, all channels Dimension #MAXNRSBRCHANNELS * +#SBR_QMF_CHANNELS* #MAX_NO_OF_ESTIMATES */ C_ALLOC_MEM2(Ram_Sbr_signMatrix, INT, (MAX_NO_OF_ESTIMATES * 64), (8)) /*! Frequency band table (low res)
Dimension #MAX_FREQ_COEFFS/2+1 */ C_ALLOC_MEM2(Ram_Sbr_freqBandTableLO, UCHAR, (MAX_FREQ_COEFFS / 2 + 1), (8)) /*! Frequency band table (high res)
Dimension #MAX_FREQ_COEFFS +1 */ C_ALLOC_MEM2(Ram_Sbr_freqBandTableHI, UCHAR, (MAX_FREQ_COEFFS + 1), (8)) /*! vk matser table
Dimension #MAX_FREQ_COEFFS +1 */ C_ALLOC_MEM2(Ram_Sbr_v_k_master, UCHAR, (MAX_FREQ_COEFFS + 1), (8)) /* Missing harmonics detection */ /*! sbr_detectionVectors
Dimension #MAX_NUM_CHANNELS*#MAX_NO_OF_ESTIMATES*#MAX_FREQ_COEFFS] */ C_ALLOC_MEM2(Ram_Sbr_detectionVectors, UCHAR, (MAX_NO_OF_ESTIMATES * MAX_FREQ_COEFFS), (8)) /*! sbr_prevCompVec[
Dimension #MAX_NUM_CHANNELS*#MAX_FREQ_COEFFS] */ C_ALLOC_MEM2(Ram_Sbr_prevEnvelopeCompensation, UCHAR, MAX_FREQ_COEFFS, (8)) /*! sbr_guideScfb[
Dimension #MAX_NUM_CHANNELS*#MAX_FREQ_COEFFS] */ C_ALLOC_MEM2(Ram_Sbr_guideScfb, UCHAR, MAX_FREQ_COEFFS, (8)) /*! sbr_guideVectorDetected
Dimension #MAX_NUM_CHANNELS*#MAX_NO_OF_ESTIMATES*#MAX_FREQ_COEFFS] */ C_ALLOC_MEM2(Ram_Sbr_guideVectorDetected, UCHAR, (MAX_NO_OF_ESTIMATES * MAX_FREQ_COEFFS), (8)) C_ALLOC_MEM2(Ram_Sbr_guideVectorDiff, FIXP_DBL, (MAX_NO_OF_ESTIMATES * MAX_FREQ_COEFFS), (8)) C_ALLOC_MEM2(Ram_Sbr_guideVectorOrig, FIXP_DBL, (MAX_NO_OF_ESTIMATES * MAX_FREQ_COEFFS), (8)) /* Static Parametric Stereo memory */ C_AALLOC_MEM_L(Ram_PsQmfStatesSynthesis, FIXP_DBL, 640 / 2, SECT_DATA_L1) C_ALLOC_MEM_L(Ram_PsEncode, PS_ENCODE, 1, SECT_DATA_L1) C_ALLOC_MEM(Ram_ParamStereo, PARAMETRIC_STEREO, 1) /* @} */ /*! \name DynamicSbrData Dynamic memory areas, might be reused in other algorithm sections, e.g. the core encoder. */ /* @{ */ /*! Energy buffer for envelope extraction
Dimension #MAXNRSBRCHANNELS * +#SBR_QMF_SLOTS * #SBR_QMF_CHANNELS */ C_ALLOC_MEM2(Ram_Sbr_envYBuffer, FIXP_DBL, (32 / 2 * 64), (8)) FIXP_DBL* GetRam_Sbr_envYBuffer(int n, UCHAR* dynamic_RAM) { FDK_ASSERT(dynamic_RAM != 0); /* The reinterpret_cast is used to suppress a compiler warning. We know that * (dynamic_RAM + OFFSET_NRG + (n*Y_2_BUF_BYTE)) is sufficiently aligned, so * the cast is safe */ return reinterpret_cast( reinterpret_cast(dynamic_RAM + OFFSET_NRG + (n * Y_2_BUF_BYTE))); } /* * QMF data */ /* The SBR encoder uses a single channel overlapping buffer set (always n=0), * but PS does not. */ FIXP_DBL* GetRam_Sbr_envRBuffer(int n, UCHAR* dynamic_RAM) { FDK_ASSERT(dynamic_RAM != 0); /* The reinterpret_cast is used to suppress a compiler warning. We know that * (dynamic_RAM + OFFSET_QMF + (n*(ENV_R_BUFF_BYTE+ENV_I_BUFF_BYTE))) is * sufficiently aligned, so the cast is safe */ return reinterpret_cast(reinterpret_cast( dynamic_RAM + OFFSET_QMF + (n * (ENV_R_BUFF_BYTE + ENV_I_BUFF_BYTE)))); } FIXP_DBL* GetRam_Sbr_envIBuffer(int n, UCHAR* dynamic_RAM) { FDK_ASSERT(dynamic_RAM != 0); /* The reinterpret_cast is used to suppress a compiler warning. We know that * (dynamic_RAM + OFFSET_QMF + (ENV_R_BUFF_BYTE) + * (n*(ENV_R_BUFF_BYTE+ENV_I_BUFF_BYTE))) is sufficiently aligned, so the cast * is safe */ return reinterpret_cast( reinterpret_cast(dynamic_RAM + OFFSET_QMF + (ENV_R_BUFF_BYTE) + (n * (ENV_R_BUFF_BYTE + ENV_I_BUFF_BYTE)))); } /* @} */ fdk-aac-2.0.2/libSBRenc/src/sbrenc_ram.h000066400000000000000000000173111404225310700176600ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Memory layout $Revision: 92790 $ */ #ifndef SBRENC_RAM_H #define SBRENC_RAM_H #include "sbr_def.h" #include "env_est.h" #include "sbr_encoder.h" #include "sbr.h" #include "ps_main.h" #include "ps_encode.h" #define ENV_TRANSIENTS_BYTE ((sizeof(FIXP_DBL) * (MAX_NUM_CHANNELS * 3 * 32))) #define ENV_R_BUFF_BYTE ((sizeof(FIXP_DBL) * ((32) * MAX_HYBRID_BANDS))) #define ENV_I_BUFF_BYTE ((sizeof(FIXP_DBL) * ((32) * MAX_HYBRID_BANDS))) #define Y_BUF_CH_BYTE \ ((2 * sizeof(FIXP_DBL) * (((32) - (32 / 2)) * MAX_HYBRID_BANDS))) #define ENV_R_BUF_PS_BYTE ((sizeof(FIXP_DBL) * 32 * 64 / 2)) #define ENV_I_BUF_PS_BYTE ((sizeof(FIXP_DBL) * 32 * 64 / 2)) #define TON_BUF_CH_BYTE \ ((sizeof(FIXP_DBL) * (MAX_NO_OF_ESTIMATES * MAX_FREQ_COEFFS))) #define Y_2_BUF_BYTE (Y_BUF_CH_BYTE) /* Workbuffer RAM - Allocation */ /* ++++++++++++++++++++++++++++++++++++++++++++++++++++ | OFFSET_QMF | OFFSET_NRG | ++++++++++++++++++++++++++++++++++++++++++++++++++++ ------------------------- ------------------------- | | 0.5 * | | sbr_envRBuffer | sbr_envYBuffer_size | | sbr_envIBuffer | | ------------------------- ------------------------- */ #define BUF_NRG_SIZE ((MAX_NUM_CHANNELS * Y_2_BUF_BYTE)) #define BUF_QMF_SIZE (ENV_R_BUFF_BYTE + ENV_I_BUFF_BYTE) /* Size of the shareable memory region than can be reused */ #define SBR_ENC_DYN_RAM_SIZE (BUF_QMF_SIZE + BUF_NRG_SIZE) #define OFFSET_QMF (0) #define OFFSET_NRG (OFFSET_QMF + BUF_QMF_SIZE) /* ***************************************************************************************************** */ H_ALLOC_MEM(Ram_SbrDynamic_RAM, FIXP_DBL) H_ALLOC_MEM(Ram_SbrEncoder, SBR_ENCODER) H_ALLOC_MEM(Ram_SbrChannel, SBR_CHANNEL) H_ALLOC_MEM(Ram_SbrElement, SBR_ELEMENT) H_ALLOC_MEM(Ram_Sbr_quotaMatrix, FIXP_DBL) H_ALLOC_MEM(Ram_Sbr_signMatrix, INT) H_ALLOC_MEM(Ram_Sbr_QmfStatesAnalysis, FIXP_QAS) H_ALLOC_MEM(Ram_Sbr_freqBandTableLO, UCHAR) H_ALLOC_MEM(Ram_Sbr_freqBandTableHI, UCHAR) H_ALLOC_MEM(Ram_Sbr_v_k_master, UCHAR) H_ALLOC_MEM(Ram_Sbr_detectionVectors, UCHAR) H_ALLOC_MEM(Ram_Sbr_prevEnvelopeCompensation, UCHAR) H_ALLOC_MEM(Ram_Sbr_guideScfb, UCHAR) H_ALLOC_MEM(Ram_Sbr_guideVectorDetected, UCHAR) /* Dynamic Memory Allocation */ H_ALLOC_MEM(Ram_Sbr_envYBuffer, FIXP_DBL) FIXP_DBL* GetRam_Sbr_envYBuffer(int n, UCHAR* dynamic_RAM); FIXP_DBL* GetRam_Sbr_envRBuffer(int n, UCHAR* dynamic_RAM); FIXP_DBL* GetRam_Sbr_envIBuffer(int n, UCHAR* dynamic_RAM); H_ALLOC_MEM(Ram_Sbr_guideVectorDiff, FIXP_DBL) H_ALLOC_MEM(Ram_Sbr_guideVectorOrig, FIXP_DBL) H_ALLOC_MEM(Ram_PsQmfStatesSynthesis, FIXP_DBL) H_ALLOC_MEM(Ram_PsEncode, PS_ENCODE) FIXP_DBL* FDKsbrEnc_SliceRam_PsRqmf(FIXP_DBL* rQmfData, UCHAR* dynamic_RAM, int n, int i, int qmfSlots); FIXP_DBL* FDKsbrEnc_SliceRam_PsIqmf(FIXP_DBL* iQmfData, UCHAR* dynamic_RAM, int n, int i, int qmfSlots); H_ALLOC_MEM(Ram_ParamStereo, PARAMETRIC_STEREO) #endif fdk-aac-2.0.2/libSBRenc/src/sbrenc_rom.cpp000066400000000000000000001225351404225310700202360ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Tobias Chalupka Description: Definition of constant tables *******************************************************************************/ /*! \file \brief Definition of constant tables $Revision: 95404 $ This module contains most of the constant data that can be stored in ROM. */ #include "sbrenc_rom.h" #include "genericStds.h" //@{ /******************************************************************************* Table Overview: o envelope level, 1.5 dB: 1a) v_Huff_envelopeLevelC10T[121] 1b) v_Huff_envelopeLevelL10T[121] 2a) v_Huff_envelopeLevelC10F[121] 2b) v_Huff_envelopeLevelL10F[121] o envelope balance, 1.5 dB: 3a) bookSbrEnvBalanceC10T[49] 3b) bookSbrEnvBalanceL10T[49] 4a) bookSbrEnvBalanceC10F[49] 4b) bookSbrEnvBalanceL10F[49] o envelope level, 3.0 dB: 5a) v_Huff_envelopeLevelC11T[63] 5b) v_Huff_envelopeLevelL11T[63] 6a) v_Huff_envelopeLevelC11F[63] 6b) v_Huff_envelopeLevelC11F[63] o envelope balance, 3.0 dB: 7a) bookSbrEnvBalanceC11T[25] 7b) bookSbrEnvBalanceL11T[25] 8a) bookSbrEnvBalanceC11F[25] 8b) bookSbrEnvBalanceL11F[25] o noise level, 3.0 dB: 9a) v_Huff_NoiseLevelC11T[63] 9b) v_Huff_NoiseLevelL11T[63] - ) (v_Huff_envelopeLevelC11F[63] is used for freq dir) - ) (v_Huff_envelopeLevelL11F[63] is used for freq dir) o noise balance, 3.0 dB: 10a) bookSbrNoiseBalanceC11T[25] 10b) bookSbrNoiseBalanceL11T[25] - ) (bookSbrEnvBalanceC11F[25] is used for freq dir) - ) (bookSbrEnvBalanceL11F[25] is used for freq dir) (1.5 dB is never used for noise) ********************************************************************************/ /*******************************************************************************/ /* table : envelope level, 1.5 dB */ /* theor range : [-58,58], CODE_BOOK_SCF_LAV = 58 */ /* implem range: [-60,60], CODE_BOOK_SCF_LAV10 = 60 */ /* raw stats : envelopeLevel_00 (yes, wrong suffix in name) KK 01-03-09 */ /*******************************************************************************/ /* direction: time contents : codewords raw table: HuffCode3C2FIX.m/envelopeLevel_00T_cF.mat/v_nChex_cF built by : FH 01-07-05 */ const INT v_Huff_envelopeLevelC10T[121] = { 0x0003FFD6, 0x0003FFD7, 0x0003FFD8, 0x0003FFD9, 0x0003FFDA, 0x0003FFDB, 0x0007FFB8, 0x0007FFB9, 0x0007FFBA, 0x0007FFBB, 0x0007FFBC, 0x0007FFBD, 0x0007FFBE, 0x0007FFBF, 0x0007FFC0, 0x0007FFC1, 0x0007FFC2, 0x0007FFC3, 0x0007FFC4, 0x0007FFC5, 0x0007FFC6, 0x0007FFC7, 0x0007FFC8, 0x0007FFC9, 0x0007FFCA, 0x0007FFCB, 0x0007FFCC, 0x0007FFCD, 0x0007FFCE, 0x0007FFCF, 0x0007FFD0, 0x0007FFD1, 0x0007FFD2, 0x0007FFD3, 0x0001FFE6, 0x0003FFD4, 0x0000FFF0, 0x0001FFE9, 0x0003FFD5, 0x0001FFE7, 0x0000FFF1, 0x0000FFEC, 0x0000FFED, 0x0000FFEE, 0x00007FF4, 0x00003FF9, 0x00003FF7, 0x00001FFA, 0x00001FF9, 0x00000FFB, 0x000007FC, 0x000003FC, 0x000001FD, 0x000000FD, 0x0000007D, 0x0000003D, 0x0000001D, 0x0000000D, 0x00000005, 0x00000001, 0x00000000, 0x00000004, 0x0000000C, 0x0000001C, 0x0000003C, 0x0000007C, 0x000000FC, 0x000001FC, 0x000003FD, 0x00000FFA, 0x00001FF8, 0x00003FF6, 0x00003FF8, 0x00007FF5, 0x0000FFEF, 0x0001FFE8, 0x0000FFF2, 0x0007FFD4, 0x0007FFD5, 0x0007FFD6, 0x0007FFD7, 0x0007FFD8, 0x0007FFD9, 0x0007FFDA, 0x0007FFDB, 0x0007FFDC, 0x0007FFDD, 0x0007FFDE, 0x0007FFDF, 0x0007FFE0, 0x0007FFE1, 0x0007FFE2, 0x0007FFE3, 0x0007FFE4, 0x0007FFE5, 0x0007FFE6, 0x0007FFE7, 0x0007FFE8, 0x0007FFE9, 0x0007FFEA, 0x0007FFEB, 0x0007FFEC, 0x0007FFED, 0x0007FFEE, 0x0007FFEF, 0x0007FFF0, 0x0007FFF1, 0x0007FFF2, 0x0007FFF3, 0x0007FFF4, 0x0007FFF5, 0x0007FFF6, 0x0007FFF7, 0x0007FFF8, 0x0007FFF9, 0x0007FFFA, 0x0007FFFB, 0x0007FFFC, 0x0007FFFD, 0x0007FFFE, 0x0007FFFF}; /* direction: time contents : codeword lengths raw table: HuffCode3C2FIX.m/envelopeLevel_00T_cF.mat/v_nLhex_cF built by : FH 01-07-05 */ const UCHAR v_Huff_envelopeLevelL10T[121] = { 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x11, 0x12, 0x10, 0x11, 0x12, 0x11, 0x10, 0x10, 0x10, 0x10, 0x0F, 0x0E, 0x0E, 0x0D, 0x0D, 0x0C, 0x0B, 0x0A, 0x09, 0x08, 0x07, 0x06, 0x05, 0x04, 0x03, 0x02, 0x02, 0x03, 0x04, 0x05, 0x06, 0x07, 0x08, 0x09, 0x0A, 0x0C, 0x0D, 0x0E, 0x0E, 0x0F, 0x10, 0x11, 0x10, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13}; /* direction: freq contents : codewords raw table: HuffCode3C2FIX.m/envelopeLevel_00F_cF.mat/v_nChex_cF built by : FH 01-07-05 */ const INT v_Huff_envelopeLevelC10F[121] = { 0x0007FFE7, 0x0007FFE8, 0x000FFFD2, 0x000FFFD3, 0x000FFFD4, 0x000FFFD5, 0x000FFFD6, 0x000FFFD7, 0x000FFFD8, 0x0007FFDA, 0x000FFFD9, 0x000FFFDA, 0x000FFFDB, 0x000FFFDC, 0x0007FFDB, 0x000FFFDD, 0x0007FFDC, 0x0007FFDD, 0x000FFFDE, 0x0003FFE4, 0x000FFFDF, 0x000FFFE0, 0x000FFFE1, 0x0007FFDE, 0x000FFFE2, 0x000FFFE3, 0x000FFFE4, 0x0007FFDF, 0x000FFFE5, 0x0007FFE0, 0x0003FFE8, 0x0007FFE1, 0x0003FFE0, 0x0003FFE9, 0x0001FFEF, 0x0003FFE5, 0x0001FFEC, 0x0001FFED, 0x0001FFEE, 0x0000FFF4, 0x0000FFF3, 0x0000FFF0, 0x00007FF7, 0x00007FF6, 0x00003FFA, 0x00001FFA, 0x00001FF9, 0x00000FFA, 0x00000FF8, 0x000007F9, 0x000003FB, 0x000001FC, 0x000001FA, 0x000000FB, 0x0000007C, 0x0000003C, 0x0000001C, 0x0000000C, 0x00000005, 0x00000001, 0x00000000, 0x00000004, 0x0000000D, 0x0000001D, 0x0000003D, 0x000000FA, 0x000000FC, 0x000001FB, 0x000003FA, 0x000007F8, 0x000007FA, 0x000007FB, 0x00000FF9, 0x00000FFB, 0x00001FF8, 0x00001FFB, 0x00003FF8, 0x00003FF9, 0x0000FFF1, 0x0000FFF2, 0x0001FFEA, 0x0001FFEB, 0x0003FFE1, 0x0003FFE2, 0x0003FFEA, 0x0003FFE3, 0x0003FFE6, 0x0003FFE7, 0x0003FFEB, 0x000FFFE6, 0x0007FFE2, 0x000FFFE7, 0x000FFFE8, 0x000FFFE9, 0x000FFFEA, 0x000FFFEB, 0x000FFFEC, 0x0007FFE3, 0x000FFFED, 0x000FFFEE, 0x000FFFEF, 0x000FFFF0, 0x0007FFE4, 0x000FFFF1, 0x0003FFEC, 0x000FFFF2, 0x000FFFF3, 0x0007FFE5, 0x0007FFE6, 0x000FFFF4, 0x000FFFF5, 0x000FFFF6, 0x000FFFF7, 0x000FFFF8, 0x000FFFF9, 0x000FFFFA, 0x000FFFFB, 0x000FFFFC, 0x000FFFFD, 0x000FFFFE, 0x000FFFFF}; /* direction: freq contents : codeword lengths raw table: HuffCode3C2FIX.m/envelopeLevel_00F_cF.mat/v_nLhex_cF built by : FH 01-07-05 */ const UCHAR v_Huff_envelopeLevelL10F[121] = { 0x13, 0x13, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x13, 0x14, 0x14, 0x14, 0x14, 0x13, 0x14, 0x13, 0x13, 0x14, 0x12, 0x14, 0x14, 0x14, 0x13, 0x14, 0x14, 0x14, 0x13, 0x14, 0x13, 0x12, 0x13, 0x12, 0x12, 0x11, 0x12, 0x11, 0x11, 0x11, 0x10, 0x10, 0x10, 0x0F, 0x0F, 0x0E, 0x0D, 0x0D, 0x0C, 0x0C, 0x0B, 0x0A, 0x09, 0x09, 0x08, 0x07, 0x06, 0x05, 0x04, 0x03, 0x02, 0x02, 0x03, 0x04, 0x05, 0x06, 0x08, 0x08, 0x09, 0x0A, 0x0B, 0x0B, 0x0B, 0x0C, 0x0C, 0x0D, 0x0D, 0x0E, 0x0E, 0x10, 0x10, 0x11, 0x11, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x14, 0x13, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x13, 0x14, 0x14, 0x14, 0x14, 0x13, 0x14, 0x12, 0x14, 0x14, 0x13, 0x13, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14}; /*******************************************************************************/ /* table : envelope balance, 1.5 dB */ /* theor range : [-48,48], CODE_BOOK_SCF_LAV = 48 */ /* implem range: same but mapped to [-24,24], CODE_BOOK_SCF_LAV_BALANCE10 = 24 */ /* raw stats : envelopePan_00 (yes, wrong suffix in name) KK 01-03-09 */ /*******************************************************************************/ /* direction: time contents : codewords raw table: HuffCode3C.m/envelopePan_00T.mat/v_nBhex built by : FH 01-05-15 */ const INT bookSbrEnvBalanceC10T[49] = { 0x0000FFE4, 0x0000FFE5, 0x0000FFE6, 0x0000FFE7, 0x0000FFE8, 0x0000FFE9, 0x0000FFEA, 0x0000FFEB, 0x0000FFEC, 0x0000FFED, 0x0000FFEE, 0x0000FFEF, 0x0000FFF0, 0x0000FFF1, 0x0000FFF2, 0x0000FFF3, 0x0000FFF4, 0x0000FFE2, 0x00000FFC, 0x000007FC, 0x000001FE, 0x0000007E, 0x0000001E, 0x00000006, 0x00000000, 0x00000002, 0x0000000E, 0x0000003E, 0x000000FE, 0x000007FD, 0x00000FFD, 0x00007FF0, 0x0000FFE3, 0x0000FFF5, 0x0000FFF6, 0x0000FFF7, 0x0000FFF8, 0x0000FFF9, 0x0000FFFA, 0x0001FFF6, 0x0001FFF7, 0x0001FFF8, 0x0001FFF9, 0x0001FFFA, 0x0001FFFB, 0x0001FFFC, 0x0001FFFD, 0x0001FFFE, 0x0001FFFF}; /* direction: time contents : codeword lengths raw table: HuffCode3C.m/envelopePan_00T.mat/v_nLhex built by : FH 01-05-15 */ const UCHAR bookSbrEnvBalanceL10T[49] = { 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x0C, 0x0B, 0x09, 0x07, 0x05, 0x03, 0x01, 0x02, 0x04, 0x06, 0x08, 0x0B, 0x0C, 0x0F, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x11, 0x11, 0x11, 0x11, 0x11, 0x11, 0x11, 0x11, 0x11, 0x11}; /* direction: freq contents : codewords raw table: HuffCode3C.m/envelopePan_00F.mat/v_nBhex built by : FH 01-05-15 */ const INT bookSbrEnvBalanceC10F[49] = { 0x0003FFE2, 0x0003FFE3, 0x0003FFE4, 0x0003FFE5, 0x0003FFE6, 0x0003FFE7, 0x0003FFE8, 0x0003FFE9, 0x0003FFEA, 0x0003FFEB, 0x0003FFEC, 0x0003FFED, 0x0003FFEE, 0x0003FFEF, 0x0003FFF0, 0x0000FFF7, 0x0001FFF0, 0x00003FFC, 0x000007FE, 0x000007FC, 0x000000FE, 0x0000007E, 0x0000000E, 0x00000002, 0x00000000, 0x00000006, 0x0000001E, 0x0000003E, 0x000001FE, 0x000007FD, 0x00000FFE, 0x00007FFA, 0x0000FFF6, 0x0003FFF1, 0x0003FFF2, 0x0003FFF3, 0x0003FFF4, 0x0003FFF5, 0x0003FFF6, 0x0003FFF7, 0x0003FFF8, 0x0003FFF9, 0x0003FFFA, 0x0003FFFB, 0x0003FFFC, 0x0003FFFD, 0x0003FFFE, 0x0007FFFE, 0x0007FFFF}; /* direction: freq contents : codeword lengths raw table: HuffCode3C.m/envelopePan_00F.mat/v_nLhex built by : FH 01-05-15 */ const UCHAR bookSbrEnvBalanceL10F[49] = { 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x10, 0x11, 0x0E, 0x0B, 0x0B, 0x08, 0x07, 0x04, 0x02, 0x01, 0x03, 0x05, 0x06, 0x09, 0x0B, 0x0C, 0x0F, 0x10, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x13, 0x13}; /*******************************************************************************/ /* table : envelope level, 3.0 dB */ /* theor range : [-29,29], CODE_BOOK_SCF_LAV = 29 */ /* implem range: [-31,31], CODE_BOOK_SCF_LAV11 = 31 */ /* raw stats : envelopeLevel_11 KK 00-02-03 */ /*******************************************************************************/ /* direction: time contents : codewords raw table: HuffCode2.m built by : FH 00-02-04 */ const INT v_Huff_envelopeLevelC11T[63] = { 0x0003FFED, 0x0003FFEE, 0x0007FFDE, 0x0007FFDF, 0x0007FFE0, 0x0007FFE1, 0x0007FFE2, 0x0007FFE3, 0x0007FFE4, 0x0007FFE5, 0x0007FFE6, 0x0007FFE7, 0x0007FFE8, 0x0007FFE9, 0x0007FFEA, 0x0007FFEB, 0x0007FFEC, 0x0001FFF4, 0x0000FFF7, 0x0000FFF9, 0x0000FFF8, 0x00003FFB, 0x00003FFA, 0x00003FF8, 0x00001FFA, 0x00000FFC, 0x000007FC, 0x000000FE, 0x0000003E, 0x0000000E, 0x00000002, 0x00000000, 0x00000006, 0x0000001E, 0x0000007E, 0x000001FE, 0x000007FD, 0x00001FFB, 0x00003FF9, 0x00003FFC, 0x00007FFA, 0x0000FFF6, 0x0001FFF5, 0x0003FFEC, 0x0007FFED, 0x0007FFEE, 0x0007FFEF, 0x0007FFF0, 0x0007FFF1, 0x0007FFF2, 0x0007FFF3, 0x0007FFF4, 0x0007FFF5, 0x0007FFF6, 0x0007FFF7, 0x0007FFF8, 0x0007FFF9, 0x0007FFFA, 0x0007FFFB, 0x0007FFFC, 0x0007FFFD, 0x0007FFFE, 0x0007FFFF}; /* direction: time contents : codeword lengths raw table: HuffCode2.m built by : FH 00-02-04 */ const UCHAR v_Huff_envelopeLevelL11T[63] = { 0x12, 0x12, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x11, 0x10, 0x10, 0x10, 0x0E, 0x0E, 0x0E, 0x0D, 0x0C, 0x0B, 0x08, 0x06, 0x04, 0x02, 0x01, 0x03, 0x05, 0x07, 0x09, 0x0B, 0x0D, 0x0E, 0x0E, 0x0F, 0x10, 0x11, 0x12, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13}; /* direction: freq contents : codewords raw table: HuffCode2.m built by : FH 00-02-04 */ const INT v_Huff_envelopeLevelC11F[63] = { 0x000FFFF0, 0x000FFFF1, 0x000FFFF2, 0x000FFFF3, 0x000FFFF4, 0x000FFFF5, 0x000FFFF6, 0x0003FFF3, 0x0007FFF5, 0x0007FFEE, 0x0007FFEF, 0x0007FFF6, 0x0003FFF4, 0x0003FFF2, 0x000FFFF7, 0x0007FFF0, 0x0001FFF5, 0x0003FFF0, 0x0001FFF4, 0x0000FFF7, 0x0000FFF6, 0x00007FF8, 0x00003FFB, 0x00000FFD, 0x000007FD, 0x000003FD, 0x000001FD, 0x000000FD, 0x0000003E, 0x0000000E, 0x00000002, 0x00000000, 0x00000006, 0x0000001E, 0x000000FC, 0x000001FC, 0x000003FC, 0x000007FC, 0x00000FFC, 0x00001FFC, 0x00003FFA, 0x00007FF9, 0x00007FFA, 0x0000FFF8, 0x0000FFF9, 0x0001FFF6, 0x0001FFF7, 0x0003FFF5, 0x0003FFF6, 0x0003FFF1, 0x000FFFF8, 0x0007FFF1, 0x0007FFF2, 0x0007FFF3, 0x000FFFF9, 0x0007FFF7, 0x0007FFF4, 0x000FFFFA, 0x000FFFFB, 0x000FFFFC, 0x000FFFFD, 0x000FFFFE, 0x000FFFFF}; /* direction: freq contents : codeword lengths raw table: HuffCode2.m built by : FH 00-02-04 */ const UCHAR v_Huff_envelopeLevelL11F[63] = { 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x12, 0x13, 0x13, 0x13, 0x13, 0x12, 0x12, 0x14, 0x13, 0x11, 0x12, 0x11, 0x10, 0x10, 0x0F, 0x0E, 0x0C, 0x0B, 0x0A, 0x09, 0x08, 0x06, 0x04, 0x02, 0x01, 0x03, 0x05, 0x08, 0x09, 0x0A, 0x0B, 0x0C, 0x0D, 0x0E, 0x0F, 0x0F, 0x10, 0x10, 0x11, 0x11, 0x12, 0x12, 0x12, 0x14, 0x13, 0x13, 0x13, 0x14, 0x13, 0x13, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14}; /*******************************************************************************/ /* table : envelope balance, 3.0 dB */ /* theor range : [-24,24], CODE_BOOK_SCF_LAV = 24 */ /* implem range: same but mapped to [-12,12], CODE_BOOK_SCF_LAV_BALANCE11 = 12 */ /* raw stats : envelopeBalance_11 KK 00-02-03 */ /*******************************************************************************/ /* direction: time contents : codewords raw table: HuffCode3C.m/envelopeBalance_11T.mat/v_nBhex built by : FH 01-05-15 */ const INT bookSbrEnvBalanceC11T[25] = { 0x00001FF2, 0x00001FF3, 0x00001FF4, 0x00001FF5, 0x00001FF6, 0x00001FF7, 0x00001FF8, 0x00000FF8, 0x000000FE, 0x0000007E, 0x0000000E, 0x00000006, 0x00000000, 0x00000002, 0x0000001E, 0x0000003E, 0x000001FE, 0x00001FF9, 0x00001FFA, 0x00001FFB, 0x00001FFC, 0x00001FFD, 0x00001FFE, 0x00003FFE, 0x00003FFF}; /* direction: time contents : codeword lengths raw table: HuffCode3C.m/envelopeBalance_11T.mat/v_nLhex built by : FH 01-05-15 */ const UCHAR bookSbrEnvBalanceL11T[25] = { 0x0D, 0x0D, 0x0D, 0x0D, 0x0D, 0x0D, 0x0D, 0x0C, 0x08, 0x07, 0x04, 0x03, 0x01, 0x02, 0x05, 0x06, 0x09, 0x0D, 0x0D, 0x0D, 0x0D, 0x0D, 0x0D, 0x0E, 0x0E}; /* direction: freq contents : codewords raw table: HuffCode3C.m/envelopeBalance_11F.mat/v_nBhex built by : FH 01-05-15 */ const INT bookSbrEnvBalanceC11F[25] = { 0x00001FF7, 0x00001FF8, 0x00001FF9, 0x00001FFA, 0x00001FFB, 0x00003FF8, 0x00003FF9, 0x000007FC, 0x000000FE, 0x0000007E, 0x0000000E, 0x00000002, 0x00000000, 0x00000006, 0x0000001E, 0x0000003E, 0x000001FE, 0x00000FFA, 0x00001FF6, 0x00003FFA, 0x00003FFB, 0x00003FFC, 0x00003FFD, 0x00003FFE, 0x00003FFF}; /* direction: freq contents : codeword lengths raw table: HuffCode3C.m/envelopeBalance_11F.mat/v_nLhex built by : FH 01-05-15 */ const UCHAR bookSbrEnvBalanceL11F[25] = { 0x0D, 0x0D, 0x0D, 0x0D, 0x0D, 0x0E, 0x0E, 0x0B, 0x08, 0x07, 0x04, 0x02, 0x01, 0x03, 0x05, 0x06, 0x09, 0x0C, 0x0D, 0x0E, 0x0E, 0x0E, 0x0E, 0x0E, 0x0E}; /*******************************************************************************/ /* table : noise level, 3.0 dB */ /* theor range : [-29,29], CODE_BOOK_SCF_LAV = 29 */ /* implem range: [-31,31], CODE_BOOK_SCF_LAV11 = 31 */ /* raw stats : noiseLevel_11 KK 00-02-03 */ /*******************************************************************************/ /* direction: time contents : codewords raw table: HuffCode2.m built by : FH 00-02-04 */ const INT v_Huff_NoiseLevelC11T[63] = { 0x00001FCE, 0x00001FCF, 0x00001FD0, 0x00001FD1, 0x00001FD2, 0x00001FD3, 0x00001FD4, 0x00001FD5, 0x00001FD6, 0x00001FD7, 0x00001FD8, 0x00001FD9, 0x00001FDA, 0x00001FDB, 0x00001FDC, 0x00001FDD, 0x00001FDE, 0x00001FDF, 0x00001FE0, 0x00001FE1, 0x00001FE2, 0x00001FE3, 0x00001FE4, 0x00001FE5, 0x00001FE6, 0x00001FE7, 0x000007F2, 0x000000FD, 0x0000003E, 0x0000000E, 0x00000006, 0x00000000, 0x00000002, 0x0000001E, 0x000000FC, 0x000003F8, 0x00001FCC, 0x00001FE8, 0x00001FE9, 0x00001FEA, 0x00001FEB, 0x00001FEC, 0x00001FCD, 0x00001FED, 0x00001FEE, 0x00001FEF, 0x00001FF0, 0x00001FF1, 0x00001FF2, 0x00001FF3, 0x00001FF4, 0x00001FF5, 0x00001FF6, 0x00001FF7, 0x00001FF8, 0x00001FF9, 0x00001FFA, 0x00001FFB, 0x00001FFC, 0x00001FFD, 0x00001FFE, 0x00003FFE, 0x00003FFF}; /* direction: time contents : codeword lengths raw table: HuffCode2.m built by : FH 00-02-04 */ const UCHAR v_Huff_NoiseLevelL11T[63] = { 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000B, 0x00000008, 0x00000006, 0x00000004, 0x00000003, 0x00000001, 0x00000002, 0x00000005, 0x00000008, 0x0000000A, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000E, 0x0000000E}; /*******************************************************************************/ /* table : noise balance, 3.0 dB */ /* theor range : [-24,24], CODE_BOOK_SCF_LAV = 24 */ /* implem range: same but mapped to [-12,12], CODE_BOOK_SCF_LAV_BALANCE11 = 12 */ /* raw stats : noiseBalance_11 KK 00-02-03 */ /*******************************************************************************/ /* direction: time contents : codewords raw table: HuffCode3C.m/noiseBalance_11.mat/v_nBhex built by : FH 01-05-15 */ const INT bookSbrNoiseBalanceC11T[25] = { 0x000000EC, 0x000000ED, 0x000000EE, 0x000000EF, 0x000000F0, 0x000000F1, 0x000000F2, 0x000000F3, 0x000000F4, 0x000000F5, 0x0000001C, 0x00000002, 0x00000000, 0x00000006, 0x0000003A, 0x000000F6, 0x000000F7, 0x000000F8, 0x000000F9, 0x000000FA, 0x000000FB, 0x000000FC, 0x000000FD, 0x000000FE, 0x000000FF}; /* direction: time contents : codeword lengths raw table: HuffCode3C.m/noiseBalance_11.mat/v_nLhex built by : FH 01-05-15 */ const UCHAR bookSbrNoiseBalanceL11T[25] = { 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x05, 0x02, 0x01, 0x03, 0x06, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08}; /* tuningTable */ const sbrTuningTable_t sbrTuningTable[] = { /* Some of the low bitrates are commented out here, this is because the encoder could lose frames at those bitrates and throw an error because it has insufficient bits to encode for some test items. */ /*** HE-AAC section ***/ /* sf,sfsp,sf,sfsp,nnb,nfo,saml,SM,FS*/ /*** mono ***/ /* 8/16 kHz dual rate */ {CODEC_AAC, 8000, 10000, 8000, 1, 7, 6, 11, 10, 1, 0, 6, SBR_MONO, 3}, {CODEC_AAC, 10000, 12000, 8000, 1, 11, 7, 13, 12, 1, 0, 6, SBR_MONO, 3}, {CODEC_AAC, 12000, 16001, 8000, 1, 14, 10, 13, 13, 1, 0, 6, SBR_MONO, 3}, {CODEC_AAC, 16000, 24000, 8000, 1, 14, 10, 14, 14, 2, 0, 3, SBR_MONO, 2}, {CODEC_AAC, 24000, 32000, 8000, 1, 14, 10, 14, 14, 2, 0, 3, SBR_MONO, 2}, {CODEC_AAC, 32000, 48001, 8000, 1, 14, 11, 15, 15, 2, 0, 3, SBR_MONO, 2}, /* 11/22 kHz dual rate */ {CODEC_AAC, 8000, 10000, 11025, 1, 5, 4, 6, 6, 1, 0, 6, SBR_MONO, 3}, {CODEC_AAC, 10000, 12000, 11025, 1, 8, 5, 12, 9, 1, 0, 6, SBR_MONO, 3}, {CODEC_AAC, 12000, 16000, 11025, 1, 12, 8, 13, 8, 1, 0, 6, SBR_MONO, 3}, {CODEC_AAC, 16000, 20000, 11025, 1, 12, 8, 13, 8, 1, 0, 6, SBR_MONO, 3}, {CODEC_AAC, 20000, 24001, 11025, 1, 13, 9, 13, 8, 1, 0, 6, SBR_MONO, 3}, {CODEC_AAC, 24000, 32000, 11025, 1, 14, 10, 14, 9, 2, 0, 3, SBR_MONO, 2}, {CODEC_AAC, 32000, 48000, 11025, 1, 15, 11, 15, 10, 2, 0, 3, SBR_MONO, 2}, {CODEC_AAC, 48000, 64001, 11025, 1, 15, 11, 15, 10, 2, 0, 3, SBR_MONO, 1}, /* 12/24 kHz dual rate */ {CODEC_AAC, 8000, 10000, 12000, 1, 4, 3, 6, 6, 1, 0, 6, SBR_MONO, 3}, {CODEC_AAC, 10000, 12000, 12000, 1, 7, 4, 11, 8, 1, 0, 6, SBR_MONO, 3}, {CODEC_AAC, 12000, 16000, 12000, 1, 11, 7, 12, 8, 1, 0, 6, SBR_MONO, 3}, {CODEC_AAC, 16000, 20000, 12000, 1, 11, 7, 12, 8, 1, 0, 6, SBR_MONO, 3}, {CODEC_AAC, 20000, 24001, 12000, 1, 12, 8, 12, 8, 1, 0, 6, SBR_MONO, 3}, {CODEC_AAC, 24000, 32000, 12000, 1, 13, 9, 13, 9, 2, 0, 3, SBR_MONO, 2}, {CODEC_AAC, 32000, 48000, 12000, 1, 14, 10, 14, 10, 2, 0, 3, SBR_MONO, 2}, {CODEC_AAC, 48000, 64001, 12000, 1, 14, 11, 15, 11, 2, 0, 3, SBR_MONO, 1}, /* 16/32 kHz dual rate */ {CODEC_AAC, 8000, 10000, 16000, 1, 1, 1, 0, 0, 1, 0, 6, SBR_MONO, 3}, {CODEC_AAC, 10000, 12000, 16000, 1, 2, 1, 6, 0, 1, 0, 6, SBR_MONO, 3}, {CODEC_AAC, 12000, 16000, 16000, 1, 4, 2, 6, 0, 1, 0, 6, SBR_MONO, 3}, {CODEC_AAC, 16000, 18000, 16000, 1, 4, 2, 8, 3, 1, 0, 6, SBR_MONO, 3}, {CODEC_AAC, 18000, 22000, 16000, 1, 6, 5, 11, 7, 2, 0, 6, SBR_MONO, 2}, {CODEC_AAC, 22000, 28000, 16000, 1, 10, 9, 12, 8, 2, 0, 6, SBR_MONO, 2}, {CODEC_AAC, 28000, 36000, 16000, 1, 12, 12, 13, 13, 2, 0, 3, SBR_MONO, 2}, {CODEC_AAC, 36000, 44000, 16000, 1, 14, 14, 13, 13, 2, 0, 3, SBR_MONO, 1}, {CODEC_AAC, 44000, 64001, 16000, 1, 14, 14, 13, 13, 2, 0, 3, SBR_MONO, 1}, /* 22.05/44.1 kHz dual rate */ /* { CODEC_AAC, 8000, 11369, 22050, 1, 1, 1, 1, 1, 1, 0, 6, SBR_MONO, 3 }, */ {CODEC_AAC, 11369, 16000, 22050, 1, 3, 1, 4, 4, 1, 0, 6, SBR_MONO, 3}, {CODEC_AAC, 16000, 18000, 22050, 1, 3, 1, 5, 4, 1, 0, 6, SBR_MONO, 3}, {CODEC_AAC, 18000, 22000, 22050, 1, 4, 4, 8, 5, 2, 0, 6, SBR_MONO, 2}, {CODEC_AAC, 22000, 28000, 22050, 1, 7, 6, 8, 6, 2, 0, 6, SBR_MONO, 2}, {CODEC_AAC, 28000, 36000, 22050, 1, 10, 10, 9, 9, 2, 0, 3, SBR_MONO, 2}, {CODEC_AAC, 36000, 44000, 22050, 1, 11, 11, 10, 10, 2, 0, 3, SBR_MONO, 1}, {CODEC_AAC, 44000, 64001, 22050, 1, 13, 13, 12, 12, 2, 0, 3, SBR_MONO, 1}, /* 24/48 kHz dual rate */ /* { CODEC_AAC, 8000, 12000, 24000, 1, 1, 1, 1, 1, 1, 0, 6, SBR_MONO, 3 }, */ {CODEC_AAC, 12000, 16000, 24000, 1, 3, 1, 4, 4, 1, 0, 6, SBR_MONO, 3}, {CODEC_AAC, 16000, 18000, 24000, 1, 3, 1, 5, 4, 1, 0, 6, SBR_MONO, 3}, {CODEC_AAC, 18000, 22000, 24000, 1, 4, 3, 8, 5, 2, 0, 6, SBR_MONO, 2}, {CODEC_AAC, 22000, 28000, 24000, 1, 7, 6, 8, 6, 2, 0, 6, SBR_MONO, 2}, {CODEC_AAC, 28000, 36000, 24000, 1, 10, 10, 9, 9, 2, 0, 3, SBR_MONO, 2}, {CODEC_AAC, 36000, 44000, 24000, 1, 11, 11, 10, 10, 2, 0, 3, SBR_MONO, 1}, {CODEC_AAC, 44000, 64001, 24000, 1, 13, 13, 11, 11, 2, 0, 3, SBR_MONO, 1}, /* 32/64 kHz dual rate */ {CODEC_AAC, 24000, 36000, 32000, 1, 4, 4, 4, 4, 2, 0, 3, SBR_MONO, 3}, {CODEC_AAC, 36000, 60000, 32000, 1, 7, 7, 6, 6, 2, 0, 3, SBR_MONO, 2}, {CODEC_AAC, 60000, 72000, 32000, 1, 9, 9, 8, 8, 2, 0, 3, SBR_MONO, 1}, {CODEC_AAC, 72000, 100000, 32000, 1, 11, 11, 10, 10, 2, 0, 3, SBR_MONO, 1}, {CODEC_AAC, 100000, 160001, 32000, 1, 13, 13, 11, 11, 2, 0, 3, SBR_MONO, 1}, /* 44.1/88.2 kHz dual rate */ {CODEC_AAC, 24000, 36000, 44100, 1, 4, 4, 4, 4, 2, 0, 3, SBR_MONO, 3}, {CODEC_AAC, 36000, 60000, 44100, 1, 7, 7, 6, 6, 2, 0, 3, SBR_MONO, 2}, {CODEC_AAC, 60000, 72000, 44100, 1, 9, 9, 8, 8, 2, 0, 3, SBR_MONO, 1}, {CODEC_AAC, 72000, 100000, 44100, 1, 11, 11, 10, 10, 2, 0, 3, SBR_MONO, 1}, {CODEC_AAC, 100000, 160001, 44100, 1, 13, 13, 11, 11, 2, 0, 3, SBR_MONO, 1}, /* 48/96 kHz dual rate */ {CODEC_AAC, 32000, 36000, 48000, 1, 4, 4, 9, 9, 2, 0, 3, SBR_MONO, 3}, {CODEC_AAC, 36000, 60000, 48000, 1, 7, 7, 10, 10, 2, 0, 3, SBR_MONO, 2}, {CODEC_AAC, 60000, 72000, 48000, 1, 9, 9, 10, 10, 2, 0, 3, SBR_MONO, 1}, {CODEC_AAC, 72000, 100000, 48000, 1, 11, 11, 11, 11, 2, 0, 3, SBR_MONO, 1}, {CODEC_AAC, 100000, 160001, 48000, 1, 13, 13, 11, 11, 2, 0, 3, SBR_MONO, 1}, /*** stereo ***/ /* 08/16 kHz dual rate */ {CODEC_AAC, 16000, 24000, 8000, 2, 6, 6, 9, 7, 1, 0, -3, SBR_SWITCH_LRC, 3}, {CODEC_AAC, 24000, 28000, 8000, 2, 9, 9, 11, 9, 1, 0, -3, SBR_SWITCH_LRC, 3}, {CODEC_AAC, 28000, 36000, 8000, 2, 11, 9, 11, 9, 2, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AAC, 36000, 44000, 8000, 2, 13, 11, 13, 11, 2, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AAC, 44000, 52000, 8000, 2, 14, 12, 13, 12, 2, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AAC, 52000, 60000, 8000, 2, 14, 14, 13, 13, 3, 0, -3, SBR_SWITCH_LRC, 1}, {CODEC_AAC, 60000, 76000, 8000, 2, 14, 14, 13, 13, 3, 0, -3, SBR_LEFT_RIGHT, 1}, {CODEC_AAC, 76000, 128001, 8000, 2, 14, 14, 13, 13, 3, 0, -3, SBR_LEFT_RIGHT, 1}, /* 11/22 kHz dual rate */ {CODEC_AAC, 16000, 24000, 11025, 2, 7, 5, 9, 7, 1, 0, -3, SBR_SWITCH_LRC, 3}, {CODEC_AAC, 24000, 28000, 11025, 2, 10, 8, 10, 8, 1, 0, -3, SBR_SWITCH_LRC, 3}, {CODEC_AAC, 28000, 36000, 11025, 2, 12, 8, 12, 8, 2, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AAC, 36000, 44000, 11025, 2, 13, 9, 13, 9, 2, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AAC, 44000, 52000, 11025, 2, 14, 11, 13, 11, 2, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AAC, 52000, 60000, 11025, 2, 15, 15, 13, 13, 3, 0, -3, SBR_SWITCH_LRC, 1}, {CODEC_AAC, 60000, 76000, 11025, 2, 15, 15, 13, 13, 3, 0, -3, SBR_LEFT_RIGHT, 1}, {CODEC_AAC, 76000, 128001, 11025, 2, 15, 15, 13, 13, 3, 0, -3, SBR_LEFT_RIGHT, 1}, /* 12/24 kHz dual rate */ {CODEC_AAC, 16000, 24000, 12000, 2, 6, 4, 9, 7, 1, 0, -3, SBR_SWITCH_LRC, 3}, {CODEC_AAC, 24000, 28000, 12000, 2, 9, 7, 10, 8, 1, 0, -3, SBR_SWITCH_LRC, 3}, {CODEC_AAC, 28000, 36000, 12000, 2, 11, 7, 12, 8, 2, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AAC, 36000, 44000, 12000, 2, 12, 9, 12, 9, 2, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AAC, 44000, 52000, 12000, 2, 13, 12, 13, 12, 2, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AAC, 52000, 60000, 12000, 2, 14, 14, 13, 13, 3, 0, -3, SBR_SWITCH_LRC, 1}, {CODEC_AAC, 60000, 76000, 12000, 2, 14, 14, 13, 13, 3, 0, -3, SBR_LEFT_RIGHT, 1}, {CODEC_AAC, 76000, 128001, 12000, 2, 14, 14, 13, 13, 3, 0, -3, SBR_LEFT_RIGHT, 1}, /* 16/32 kHz dual rate */ {CODEC_AAC, 16000, 24000, 16000, 2, 4, 2, 1, 0, 1, 0, -3, SBR_SWITCH_LRC, 3}, {CODEC_AAC, 24000, 28000, 16000, 2, 8, 7, 10, 8, 1, 0, -3, SBR_SWITCH_LRC, 3}, {CODEC_AAC, 28000, 36000, 16000, 2, 10, 9, 12, 11, 2, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AAC, 36000, 44000, 16000, 2, 13, 13, 13, 13, 2, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AAC, 44000, 52000, 16000, 2, 14, 14, 13, 13, 2, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AAC, 52000, 60000, 16000, 2, 14, 14, 13, 13, 3, 0, -3, SBR_SWITCH_LRC, 1}, {CODEC_AAC, 60000, 76000, 16000, 2, 14, 14, 13, 13, 3, 0, -3, SBR_LEFT_RIGHT, 1}, {CODEC_AAC, 76000, 128001, 16000, 2, 14, 14, 13, 13, 3, 0, -3, SBR_LEFT_RIGHT, 1}, /* 22.05/44.1 kHz dual rate */ {CODEC_AAC, 16000, 24000, 22050, 2, 2, 1, 1, 0, 1, 0, -3, SBR_SWITCH_LRC, 3}, {CODEC_AAC, 24000, 28000, 22050, 2, 5, 4, 6, 5, 1, 0, -3, SBR_SWITCH_LRC, 3}, {CODEC_AAC, 28000, 32000, 22050, 2, 5, 4, 8, 7, 2, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AAC, 32000, 36000, 22050, 2, 7, 6, 8, 7, 2, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AAC, 36000, 44000, 22050, 2, 10, 10, 9, 9, 2, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AAC, 44000, 52000, 22050, 2, 12, 12, 9, 9, 3, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AAC, 52000, 60000, 22050, 2, 13, 13, 10, 10, 3, 0, -3, SBR_SWITCH_LRC, 1}, {CODEC_AAC, 60000, 76000, 22050, 2, 14, 14, 12, 12, 3, 0, -3, SBR_LEFT_RIGHT, 1}, {CODEC_AAC, 76000, 128001, 22050, 2, 14, 14, 12, 12, 3, 0, -3, SBR_LEFT_RIGHT, 1}, /* 24/48 kHz dual rate */ {CODEC_AAC, 16000, 24000, 24000, 2, 2, 1, 1, 0, 1, 0, -3, SBR_SWITCH_LRC, 3}, {CODEC_AAC, 24000, 28000, 24000, 2, 5, 5, 6, 6, 1, 0, -3, SBR_SWITCH_LRC, 3}, {CODEC_AAC, 28000, 36000, 24000, 2, 7, 6, 8, 7, 2, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AAC, 36000, 44000, 24000, 2, 10, 10, 9, 9, 2, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AAC, 44000, 52000, 24000, 2, 12, 12, 9, 9, 3, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AAC, 52000, 60000, 24000, 2, 13, 13, 10, 10, 3, 0, -3, SBR_SWITCH_LRC, 1}, {CODEC_AAC, 60000, 76000, 24000, 2, 14, 14, 12, 12, 3, 0, -3, SBR_LEFT_RIGHT, 1}, {CODEC_AAC, 76000, 128001, 24000, 2, 14, 14, 12, 12, 3, 0, -3, SBR_LEFT_RIGHT, 1}, /* 32/64 kHz dual rate */ {CODEC_AAC, 32000, 60000, 32000, 2, 4, 4, 4, 4, 2, 0, -3, SBR_SWITCH_LRC, 3}, {CODEC_AAC, 60000, 80000, 32000, 2, 7, 7, 6, 6, 3, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AAC, 80000, 112000, 32000, 2, 9, 9, 8, 8, 3, 0, -3, SBR_LEFT_RIGHT, 1}, {CODEC_AAC, 112000, 144000, 32000, 2, 11, 11, 10, 10, 3, 0, -3, SBR_LEFT_RIGHT, 1}, {CODEC_AAC, 144000, 256001, 32000, 2, 13, 13, 11, 11, 3, 0, -3, SBR_LEFT_RIGHT, 1}, /* 44.1/88.2 kHz dual rate */ {CODEC_AAC, 32000, 60000, 44100, 2, 4, 4, 4, 4, 2, 0, -3, SBR_SWITCH_LRC, 3}, {CODEC_AAC, 60000, 80000, 44100, 2, 7, 7, 6, 6, 3, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AAC, 80000, 112000, 44100, 2, 9, 9, 8, 8, 3, 0, -3, SBR_LEFT_RIGHT, 1}, {CODEC_AAC, 112000, 144000, 44100, 2, 11, 11, 10, 10, 3, 0, -3, SBR_LEFT_RIGHT, 1}, {CODEC_AAC, 144000, 256001, 44100, 2, 13, 13, 11, 11, 3, 0, -3, SBR_LEFT_RIGHT, 1}, /* 48/96 kHz dual rate */ {CODEC_AAC, 36000, 60000, 48000, 2, 4, 4, 9, 9, 2, 0, -3, SBR_SWITCH_LRC, 3}, {CODEC_AAC, 60000, 80000, 48000, 2, 7, 7, 9, 9, 3, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AAC, 80000, 112000, 48000, 2, 9, 9, 10, 10, 3, 0, -3, SBR_LEFT_RIGHT, 1}, {CODEC_AAC, 112000, 144000, 48000, 2, 11, 11, 11, 11, 3, 0, -3, SBR_LEFT_RIGHT, 1}, {CODEC_AAC, 144000, 256001, 48000, 2, 13, 13, 11, 11, 3, 0, -3, SBR_LEFT_RIGHT, 1}, /** AAC LOW DELAY SECTION **/ /* 24 kHz dual rate - 12kHz singlerate is not allowed (deactivated in FDKsbrEnc_IsSbrSettingAvail()) */ {CODEC_AACLD, 8000, 32000, 12000, 1, 1, 1, 0, 0, 1, 0, 6, SBR_MONO, 3}, /*** mono ***/ /* 16/32 kHz dual rate */ {CODEC_AACLD, 16000, 18000, 16000, 1, 4, 5, 9, 7, 1, 0, 6, SBR_MONO, 3}, {CODEC_AACLD, 18000, 22000, 16000, 1, 7, 7, 12, 12, 1, 6, 9, SBR_MONO, 3}, {CODEC_AACLD, 22000, 28000, 16000, 1, 6, 6, 9, 9, 2, 3, 6, SBR_MONO, 3}, {CODEC_AACLD, 28000, 36000, 16000, 1, 8, 8, 12, 7, 2, 9, 12, SBR_MONO, 3}, {CODEC_AACLD, 36000, 44000, 16000, 1, 10, 14, 12, 13, 2, 0, 3, SBR_MONO, 1}, {CODEC_AACLD, 44000, 64001, 16000, 1, 11, 14, 13, 13, 2, 0, 3, SBR_MONO, 1}, /* 22.05/44.1 kHz dual rate */ {CODEC_AACLD, 18000, 22000, 22050, 1, 4, 4, 5, 5, 2, 0, 6, SBR_MONO, 3}, {CODEC_AACLD, 22000, 28000, 22050, 1, 5, 5, 6, 6, 2, 0, 6, SBR_MONO, 2}, {CODEC_AACLD, 28000, 36000, 22050, 1, 7, 8, 8, 8, 2, 0, 3, SBR_MONO, 2}, {CODEC_AACLD, 36000, 44000, 22050, 1, 9, 9, 9, 9, 2, 0, 3, SBR_MONO, 1}, {CODEC_AACLD, 44000, 52000, 22050, 1, 12, 11, 11, 11, 2, 0, 3, SBR_MONO, 1}, {CODEC_AACLD, 52000, 64001, 22050, 1, 13, 11, 11, 10, 2, 0, 3, SBR_MONO, 1}, /* 24/48 kHz dual rate */ {CODEC_AACLD, 20000, 22000, 24000, 1, 3, 4, 8, 8, 2, 0, 6, SBR_MONO, 2}, {CODEC_AACLD, 22000, 28000, 24000, 1, 3, 8, 8, 7, 2, 0, 3, SBR_MONO, 2}, {CODEC_AACLD, 28000, 36000, 24000, 1, 4, 8, 8, 7, 2, 0, 3, SBR_MONO, 2}, {CODEC_AACLD, 36000, 56000, 24000, 1, 8, 9, 9, 8, 2, 0, 3, SBR_MONO, 1}, {CODEC_AACLD, 56000, 64001, 24000, 1, 13, 11, 11, 10, 2, 0, 3, SBR_MONO, 1}, /* 32/64 kHz dual rate */ {CODEC_AACLD, 24000, 36000, 32000, 1, 4, 4, 4, 4, 2, 0, 3, SBR_MONO, 3}, {CODEC_AACLD, 36000, 60000, 32000, 1, 7, 7, 6, 6, 2, 0, 3, SBR_MONO, 2}, {CODEC_AACLD, 60000, 72000, 32000, 1, 9, 9, 8, 8, 2, 0, 3, SBR_MONO, 1}, {CODEC_AACLD, 72000, 100000, 32000, 1, 11, 11, 10, 10, 2, 0, 3, SBR_MONO, 1}, {CODEC_AACLD, 100000, 160001, 32000, 1, 13, 13, 11, 11, 2, 0, 3, SBR_MONO, 1}, /* 44/88 kHz dual rate */ {CODEC_AACLD, 36000, 60000, 44100, 1, 8, 7, 6, 9, 2, 0, 3, SBR_MONO, 2}, {CODEC_AACLD, 60000, 72000, 44100, 1, 9, 9, 10, 10, 2, 0, 3, SBR_MONO, 1}, {CODEC_AACLD, 72000, 100000, 44100, 1, 11, 11, 11, 11, 2, 0, 3, SBR_MONO, 1}, {CODEC_AACLD, 100000, 160001, 44100, 1, 13, 13, 11, 11, 2, 0, 3, SBR_MONO, 1}, /* 48/96 kHz dual rate */ /* 32 and 40kbps line tuned for dual-rate SBR */ {CODEC_AACLD, 36000, 60000, 48000, 1, 4, 7, 4, 4, 2, 0, 3, SBR_MONO, 3}, {CODEC_AACLD, 60000, 72000, 48000, 1, 9, 9, 10, 10, 2, 0, 3, SBR_MONO, 1}, {CODEC_AACLD, 72000, 100000, 48000, 1, 11, 11, 11, 11, 2, 0, 3, SBR_MONO, 1}, {CODEC_AACLD, 100000, 160001, 48000, 1, 13, 13, 11, 11, 2, 0, 3, SBR_MONO, 1}, /*** stereo ***/ /* 16/32 kHz dual rate */ {CODEC_AACLD, 32000, 36000, 16000, 2, 10, 9, 12, 11, 2, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AACLD, 36000, 44000, 16000, 2, 13, 13, 13, 13, 2, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AACLD, 44000, 52000, 16000, 2, 10, 9, 11, 9, 2, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AACLD, 52000, 60000, 16000, 2, 14, 14, 13, 13, 3, 0, -3, SBR_SWITCH_LRC, 1}, {CODEC_AACLD, 60000, 76000, 16000, 2, 14, 14, 13, 13, 3, 0, -3, SBR_LEFT_RIGHT, 1}, {CODEC_AACLD, 76000, 128001, 16000, 2, 14, 14, 13, 13, 3, 0, -3, SBR_LEFT_RIGHT, 1}, /* 22.05/44.1 kHz dual rate */ {CODEC_AACLD, 32000, 36000, 22050, 2, 5, 4, 7, 6, 2, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AACLD, 36000, 44000, 22050, 2, 5, 8, 8, 8, 2, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AACLD, 44000, 52000, 22050, 2, 7, 10, 8, 8, 3, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AACLD, 52000, 60000, 22050, 2, 9, 11, 9, 9, 3, 0, -3, SBR_SWITCH_LRC, 1}, {CODEC_AACLD, 60000, 76000, 22050, 2, 10, 12, 10, 11, 3, 0, -3, SBR_LEFT_RIGHT, 1}, {CODEC_AACLD, 76000, 82000, 22050, 2, 12, 12, 11, 11, 3, 0, -3, SBR_LEFT_RIGHT, 1}, {CODEC_AACLD, 82000, 128001, 22050, 2, 13, 12, 11, 11, 3, 0, -3, SBR_LEFT_RIGHT, 1}, /* 24/48 kHz dual rate */ {CODEC_AACLD, 32000, 36000, 24000, 2, 5, 4, 7, 6, 2, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AACLD, 36000, 44000, 24000, 2, 4, 8, 8, 8, 2, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AACLD, 44000, 52000, 24000, 2, 6, 10, 8, 8, 3, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AACLD, 52000, 60000, 24000, 2, 9, 11, 9, 9, 3, 0, -3, SBR_SWITCH_LRC, 1}, {CODEC_AACLD, 60000, 76000, 24000, 2, 11, 12, 10, 11, 3, 0, -3, SBR_LEFT_RIGHT, 1}, {CODEC_AACLD, 76000, 88000, 24000, 2, 12, 13, 11, 11, 3, 0, -3, SBR_LEFT_RIGHT, 1}, {CODEC_AACLD, 88000, 128001, 24000, 2, 13, 13, 11, 11, 3, 0, -3, SBR_LEFT_RIGHT, 1}, /* 32/64 kHz dual rate */ {CODEC_AACLD, 60000, 80000, 32000, 2, 7, 7, 6, 6, 3, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AACLD, 80000, 112000, 32000, 2, 9, 9, 8, 8, 3, 0, -3, SBR_LEFT_RIGHT, 1}, {CODEC_AACLD, 112000, 144000, 32000, 2, 11, 11, 10, 10, 3, 0, -3, SBR_LEFT_RIGHT, 1}, {CODEC_AACLD, 144000, 256001, 32000, 2, 13, 13, 11, 11, 3, 0, -3, SBR_LEFT_RIGHT, 1}, /* 44.1/88.2 kHz dual rate */ {CODEC_AACLD, 60000, 80000, 44100, 2, 7, 7, 6, 6, 3, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AACLD, 80000, 112000, 44100, 2, 10, 10, 8, 8, 3, 0, -3, SBR_LEFT_RIGHT, 1}, {CODEC_AACLD, 112000, 144000, 44100, 2, 12, 12, 10, 10, 3, 0, -3, SBR_LEFT_RIGHT, 1}, {CODEC_AACLD, 144000, 256001, 44100, 2, 13, 13, 11, 11, 3, 0, -3, SBR_LEFT_RIGHT, 1}, /* 48/96 kHz dual rate */ {CODEC_AACLD, 60000, 80000, 48000, 2, 7, 7, 10, 10, 2, 0, -3, SBR_SWITCH_LRC, 2}, {CODEC_AACLD, 80000, 112000, 48000, 2, 9, 9, 10, 10, 3, 0, -3, SBR_LEFT_RIGHT, 1}, {CODEC_AACLD, 112000, 144000, 48000, 2, 11, 11, 11, 11, 3, 0, -3, SBR_LEFT_RIGHT, 1}, {CODEC_AACLD, 144000, 176000, 48000, 2, 12, 12, 11, 11, 3, 0, -3, SBR_LEFT_RIGHT, 1}, {CODEC_AACLD, 176000, 256001, 48000, 2, 13, 13, 11, 11, 3, 0, -3, SBR_LEFT_RIGHT, 1}, }; const int sbrTuningTableSize = sizeof(sbrTuningTable) / sizeof(sbrTuningTable[0]); const psTuningTable_t psTuningTable[4] = { {8000, 22000, PSENC_STEREO_BANDS_10, PSENC_NENV_1, FL2FXCONST_DBL(3.0f / 4.0f)}, {22000, 28000, PSENC_STEREO_BANDS_20, PSENC_NENV_1, FL2FXCONST_DBL(2.0f / 4.0f)}, {28000, 36000, PSENC_STEREO_BANDS_20, PSENC_NENV_2, FL2FXCONST_DBL(1.5f / 4.0f)}, {36000, 160001, PSENC_STEREO_BANDS_20, PSENC_NENV_4, FL2FXCONST_DBL(1.1f / 4.0f)}, }; //@} fdk-aac-2.0.2/libSBRenc/src/sbrenc_rom.h000066400000000000000000000137041404225310700177000ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Declaration of constant tables $Revision: 92790 $ */ #ifndef SBRENC_ROM_H #define SBRENC_ROM_H #include "sbr_def.h" #include "sbr_encoder.h" #include "ps_main.h" /* huffman tables */ extern const INT v_Huff_envelopeLevelC10T[121]; extern const UCHAR v_Huff_envelopeLevelL10T[121]; extern const INT v_Huff_envelopeLevelC10F[121]; extern const UCHAR v_Huff_envelopeLevelL10F[121]; extern const INT bookSbrEnvBalanceC10T[49]; extern const UCHAR bookSbrEnvBalanceL10T[49]; extern const INT bookSbrEnvBalanceC10F[49]; extern const UCHAR bookSbrEnvBalanceL10F[49]; extern const INT v_Huff_envelopeLevelC11T[63]; extern const UCHAR v_Huff_envelopeLevelL11T[63]; extern const INT v_Huff_envelopeLevelC11F[63]; extern const UCHAR v_Huff_envelopeLevelL11F[63]; extern const INT bookSbrEnvBalanceC11T[25]; extern const UCHAR bookSbrEnvBalanceL11T[25]; extern const INT bookSbrEnvBalanceC11F[25]; extern const UCHAR bookSbrEnvBalanceL11F[25]; extern const INT v_Huff_NoiseLevelC11T[63]; extern const UCHAR v_Huff_NoiseLevelL11T[63]; extern const INT bookSbrNoiseBalanceC11T[25]; extern const UCHAR bookSbrNoiseBalanceL11T[25]; extern const sbrTuningTable_t sbrTuningTable[]; extern const int sbrTuningTableSize; extern const psTuningTable_t psTuningTable[4]; #endif fdk-aac-2.0.2/libSBRenc/src/ton_corr.cpp000066400000000000000000001005161404225310700177250ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Description: *******************************************************************************/ #include "ton_corr.h" #include "sbrenc_ram.h" #include "sbr_misc.h" #include "genericStds.h" #include "autocorr2nd.h" #define BAND_V_SIZE 32 #define NUM_V_COMBINE \ 8 /* Must be a divisor of 64 and fulfill the ASSERTs below */ /**************************************************************************/ /*! \brief Calculates the tonal to noise ration for different frequency bands and time segments. The ratio between the predicted energy (tonal energy A) and the total energy (A + B) is calculated. This is converted to the ratio between the predicted energy (tonal energy A) and the non-predictable energy (noise energy B). Hence the quota-matrix contains A/B = q/(1-q). The samples in nrgVector are scaled by 1.0/16.0 The samples in pNrgVectorFreq are scaled by 1.0/2.0 The samples in quotaMatrix are scaled by RELAXATION \return none. */ /**************************************************************************/ void FDKsbrEnc_CalculateTonalityQuotas( HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct. */ FIXP_DBL **RESTRICT sourceBufferReal, /*!< The real part of the QMF-matrix. */ FIXP_DBL **RESTRICT sourceBufferImag, /*!< The imaginary part of the QMF-matrix. */ INT usb, /*!< upper side band, highest + 1 QMF band in the SBR range. */ INT qmfScale /*!< sclefactor of QMF subsamples */ ) { INT i, k, r, r2, timeIndex, autoCorrScaling; INT startIndexMatrix = hTonCorr->startIndexMatrix; INT totNoEst = hTonCorr->numberOfEstimates; INT noEstPerFrame = hTonCorr->numberOfEstimatesPerFrame; INT move = hTonCorr->move; INT noQmfChannels = hTonCorr->noQmfChannels; /* Number of Bands */ INT buffLen = hTonCorr->bufferLength; /* Number of Slots */ INT stepSize = hTonCorr->stepSize; INT *pBlockLength = hTonCorr->lpcLength; INT **RESTRICT signMatrix = hTonCorr->signMatrix; FIXP_DBL *RESTRICT nrgVector = hTonCorr->nrgVector; FIXP_DBL **RESTRICT quotaMatrix = hTonCorr->quotaMatrix; FIXP_DBL *RESTRICT pNrgVectorFreq = hTonCorr->nrgVectorFreq; FIXP_DBL *realBuf; FIXP_DBL *imagBuf; FIXP_DBL alphar[2], alphai[2], fac; C_ALLOC_SCRATCH_START(ac, ACORR_COEFS, 1) C_ALLOC_SCRATCH_START(realBufRef, FIXP_DBL, 2 * BAND_V_SIZE * NUM_V_COMBINE) realBuf = realBufRef; imagBuf = realBuf + BAND_V_SIZE * NUM_V_COMBINE; FDK_ASSERT(buffLen <= BAND_V_SIZE); FDK_ASSERT(sizeof(FIXP_DBL) * NUM_V_COMBINE * BAND_V_SIZE * 2 < (1024 * sizeof(FIXP_DBL) - sizeof(ACORR_COEFS))); /* * Buffering of the quotaMatrix and the quotaMatrixTransp. *********************************************************/ for (i = 0; i < move; i++) { FDKmemcpy(quotaMatrix[i], quotaMatrix[i + noEstPerFrame], noQmfChannels * sizeof(FIXP_DBL)); FDKmemcpy(signMatrix[i], signMatrix[i + noEstPerFrame], noQmfChannels * sizeof(INT)); } FDKmemmove(nrgVector, nrgVector + noEstPerFrame, move * sizeof(FIXP_DBL)); FDKmemclear(nrgVector + startIndexMatrix, (totNoEst - startIndexMatrix) * sizeof(FIXP_DBL)); FDKmemclear(pNrgVectorFreq, noQmfChannels * sizeof(FIXP_DBL)); /* * Calculate the quotas for the current time steps. **************************************************/ for (r = 0; r < usb; r++) { int blockLength; k = hTonCorr->nextSample; /* startSample */ timeIndex = startIndexMatrix; /* Copy as many as possible Band across all Slots at once */ if (realBuf != realBufRef) { realBuf -= BAND_V_SIZE; imagBuf -= BAND_V_SIZE; } else { realBuf += BAND_V_SIZE * (NUM_V_COMBINE - 1); imagBuf += BAND_V_SIZE * (NUM_V_COMBINE - 1); for (i = 0; i < buffLen; i++) { int v; FIXP_DBL *ptr; ptr = realBuf + i; for (v = 0; v < NUM_V_COMBINE; v++) { ptr[0] = sourceBufferReal[i][r + v]; ptr[0 + BAND_V_SIZE * NUM_V_COMBINE] = sourceBufferImag[i][r + v]; ptr -= BAND_V_SIZE; } } } blockLength = pBlockLength[0]; while (k <= buffLen - blockLength) { autoCorrScaling = fixMin( getScalefactor(&realBuf[k - LPC_ORDER], LPC_ORDER + blockLength), getScalefactor(&imagBuf[k - LPC_ORDER], LPC_ORDER + blockLength)); autoCorrScaling = fixMax(0, autoCorrScaling - 1); scaleValues(&realBuf[k - LPC_ORDER], LPC_ORDER + blockLength, autoCorrScaling); scaleValues(&imagBuf[k - LPC_ORDER], LPC_ORDER + blockLength, autoCorrScaling); autoCorrScaling <<= 1; /* consider qmf buffer scaling twice */ autoCorrScaling += autoCorr2nd_cplx(ac, realBuf + k, imagBuf + k, blockLength); if (ac->det == FL2FXCONST_DBL(0.0f)) { alphar[1] = alphai[1] = FL2FXCONST_DBL(0.0f); alphar[0] = (ac->r01r) >> 2; alphai[0] = (ac->r01i) >> 2; fac = fMultDiv2(ac->r00r, ac->r11r) >> 1; } else { alphar[1] = (fMultDiv2(ac->r01r, ac->r12r) >> 1) - (fMultDiv2(ac->r01i, ac->r12i) >> 1) - (fMultDiv2(ac->r02r, ac->r11r) >> 1); alphai[1] = (fMultDiv2(ac->r01i, ac->r12r) >> 1) + (fMultDiv2(ac->r01r, ac->r12i) >> 1) - (fMultDiv2(ac->r02i, ac->r11r) >> 1); alphar[0] = (fMultDiv2(ac->r01r, ac->det) >> (ac->det_scale + 1)) + fMult(alphar[1], ac->r12r) + fMult(alphai[1], ac->r12i); alphai[0] = (fMultDiv2(ac->r01i, ac->det) >> (ac->det_scale + 1)) + fMult(alphai[1], ac->r12r) - fMult(alphar[1], ac->r12i); fac = fMultDiv2(ac->r00r, fMult(ac->det, ac->r11r)) >> (ac->det_scale + 1); } if (fac == FL2FXCONST_DBL(0.0f)) { quotaMatrix[timeIndex][r] = FL2FXCONST_DBL(0.0f); signMatrix[timeIndex][r] = 0; } else { /* quotaMatrix is scaled with the factor RELAXATION parse RELAXATION in fractional part and shift factor: 1/(1/0.524288 * 2^RELAXATION_SHIFT) */ FIXP_DBL tmp, num, denom; INT numShift, denomShift, commonShift; INT sign; num = fMultDiv2(alphar[0], ac->r01r) + fMultDiv2(alphai[0], ac->r01i) - fMultDiv2(alphar[1], fMult(ac->r02r, ac->r11r)) - fMultDiv2(alphai[1], fMult(ac->r02i, ac->r11r)); num = fixp_abs(num); denom = (fac >> 1) + (fMultDiv2(fac, RELAXATION_FRACT) >> RELAXATION_SHIFT) - num; denom = fixp_abs(denom); num = fMult(num, RELAXATION_FRACT); numShift = CountLeadingBits(num) - 2; num = scaleValue(num, numShift); denomShift = CountLeadingBits(denom); denom = (FIXP_DBL)denom << denomShift; if ((num > FL2FXCONST_DBL(0.0f)) && (denom != FL2FXCONST_DBL(0.0f))) { commonShift = fixMin(numShift - denomShift + RELAXATION_SHIFT, DFRACT_BITS - 1); if (commonShift < 0) { commonShift = -commonShift; tmp = schur_div(num, denom, 16); commonShift = fixMin(commonShift, CountLeadingBits(tmp)); quotaMatrix[timeIndex][r] = tmp << commonShift; } else { quotaMatrix[timeIndex][r] = schur_div(num, denom, 16) >> commonShift; } } else { quotaMatrix[timeIndex][r] = FL2FXCONST_DBL(0.0f); } if (ac->r11r != FL2FXCONST_DBL(0.0f)) { if (((ac->r01r >= FL2FXCONST_DBL(0.0f)) && (ac->r11r >= FL2FXCONST_DBL(0.0f))) || ((ac->r01r < FL2FXCONST_DBL(0.0f)) && (ac->r11r < FL2FXCONST_DBL(0.0f)))) { sign = 1; } else { sign = -1; } } else { sign = 1; } if (sign < 0) { r2 = r; /* (INT) pow(-1, band); */ } else { r2 = r + 1; /* (INT) pow(-1, band+1); */ } signMatrix[timeIndex][r] = 1 - 2 * (r2 & 0x1); } nrgVector[timeIndex] += ((ac->r00r) >> fixMin(DFRACT_BITS - 1, (2 * qmfScale + autoCorrScaling + SCALE_NRGVEC))); /* pNrgVectorFreq[r] finally has to be divided by noEstPerFrame, replaced * division by shifting with one */ pNrgVectorFreq[r] = pNrgVectorFreq[r] + ((ac->r00r) >> fixMin(DFRACT_BITS - 1, (2 * qmfScale + autoCorrScaling + SCALE_NRGVEC))); blockLength = pBlockLength[1]; k += stepSize; timeIndex++; } } C_ALLOC_SCRATCH_END(realBufRef, FIXP_DBL, 2 * BAND_V_SIZE * NUM_V_COMBINE) C_ALLOC_SCRATCH_END(ac, ACORR_COEFS, 1) } /**************************************************************************/ /*! \brief Extracts the parameters required in the decoder to obtain the correct tonal to noise ratio after SBR. Estimates the tonal to noise ratio of the original signal (using LPC). Predicts the tonal to noise ration of the SBR signal (in the decoder) by patching the tonal to noise ratio values similar to the patching of the lowband in the decoder. Given the tonal to noise ratio of the original and the SBR signal, it estimates the required amount of inverse filtering, additional noise as well as any additional sines. \return none. */ /**************************************************************************/ void FDKsbrEnc_TonCorrParamExtr( HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct. */ INVF_MODE *infVec, /*!< Vector where the inverse filtering levels will be stored. */ FIXP_DBL *noiseLevels, /*!< Vector where the noise levels will be stored. */ INT *missingHarmonicFlag, /*!< Flag set to one or zero, dependent on if any strong sines are missing.*/ UCHAR *missingHarmonicsIndex, /*!< Vector indicating where sines are missing. */ UCHAR *envelopeCompensation, /*!< Vector to store compensation values for the energies in. */ const SBR_FRAME_INFO *frameInfo, /*!< Frame info struct, contains the time and frequency grid of the current frame.*/ UCHAR *transientInfo, /*!< Transient info.*/ UCHAR *freqBandTable, /*!< Frequency band tables for high-res.*/ INT nSfb, /*!< Number of scalefactor bands for high-res. */ XPOS_MODE xposType, /*!< Type of transposer used in the decoder.*/ UINT sbrSyntaxFlags) { INT band; INT transientFlag = transientInfo[1]; /*!< Flag indicating if a transient is present in the current frame. */ INT transientPos = transientInfo[0]; /*!< Position of the transient.*/ INT transientFrame, transientFrameInvfEst; INVF_MODE *infVecPtr; /* Determine if this is a frame where a transient starts... The detection of noise-floor, missing harmonics and invf_est, is not in sync for the non-buf-opt decoder such as AAC. Hence we need to keep track on the transient in the present frame as well as in the next. */ transientFrame = 0; if (hTonCorr->transientNextFrame) { /* The transient was detected in the previous frame, but is actually */ transientFrame = 1; hTonCorr->transientNextFrame = 0; if (transientFlag) { if (transientPos + hTonCorr->transientPosOffset >= frameInfo->borders[frameInfo->nEnvelopes]) { hTonCorr->transientNextFrame = 1; } } } else { if (transientFlag) { if (transientPos + hTonCorr->transientPosOffset < frameInfo->borders[frameInfo->nEnvelopes]) { transientFrame = 1; hTonCorr->transientNextFrame = 0; } else { hTonCorr->transientNextFrame = 1; } } } transientFrameInvfEst = transientFrame; /* Estimate the required invese filtereing level. */ if (hTonCorr->switchInverseFilt) FDKsbrEnc_qmfInverseFilteringDetector( &hTonCorr->sbrInvFilt, hTonCorr->quotaMatrix, hTonCorr->nrgVector, hTonCorr->indexVector, hTonCorr->frameStartIndexInvfEst, hTonCorr->numberOfEstimatesPerFrame + hTonCorr->frameStartIndexInvfEst, transientFrameInvfEst, infVec); /* Detect what tones will be missing. */ if (xposType == XPOS_LC) { FDKsbrEnc_SbrMissingHarmonicsDetectorQmf( &hTonCorr->sbrMissingHarmonicsDetector, hTonCorr->quotaMatrix, hTonCorr->signMatrix, hTonCorr->indexVector, frameInfo, transientInfo, missingHarmonicFlag, missingHarmonicsIndex, freqBandTable, nSfb, envelopeCompensation, hTonCorr->nrgVectorFreq); } else { *missingHarmonicFlag = 0; FDKmemclear(missingHarmonicsIndex, nSfb * sizeof(UCHAR)); } /* Noise floor estimation */ infVecPtr = hTonCorr->sbrInvFilt.prevInvfMode; FDKsbrEnc_sbrNoiseFloorEstimateQmf( &hTonCorr->sbrNoiseFloorEstimate, frameInfo, noiseLevels, hTonCorr->quotaMatrix, hTonCorr->indexVector, *missingHarmonicFlag, hTonCorr->frameStartIndex, hTonCorr->numberOfEstimatesPerFrame, transientFrame, infVecPtr, sbrSyntaxFlags); /* Store the invfVec data for the next frame...*/ for (band = 0; band < hTonCorr->sbrInvFilt.noDetectorBands; band++) { hTonCorr->sbrInvFilt.prevInvfMode[band] = infVec[band]; } } /**************************************************************************/ /*! \brief Searches for the closest match in the frequency master table. \return closest entry. */ /**************************************************************************/ static INT findClosestEntry(INT goalSb, UCHAR *v_k_master, INT numMaster, INT direction) { INT index; if (goalSb <= v_k_master[0]) return v_k_master[0]; if (goalSb >= v_k_master[numMaster]) return v_k_master[numMaster]; if (direction) { index = 0; while (v_k_master[index] < goalSb) { index++; } } else { index = numMaster; while (v_k_master[index] > goalSb) { index--; } } return v_k_master[index]; } /**************************************************************************/ /*! \brief resets the patch \return errorCode, noError if successful. */ /**************************************************************************/ static INT resetPatch( HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct. */ INT xposctrl, /*!< Different patch modes. */ INT highBandStartSb, /*!< Start band of the SBR range. */ UCHAR *v_k_master, /*!< Master frequency table from which all other table are derived.*/ INT numMaster, /*!< Number of elements in the master table. */ INT fs, /*!< Sampling frequency. */ INT noChannels) /*!< Number of QMF-channels. */ { INT patch, k, i; INT targetStopBand; PATCH_PARAM *patchParam = hTonCorr->patchParam; INT sbGuard = hTonCorr->guard; INT sourceStartBand; INT patchDistance; INT numBandsInPatch; INT lsb = v_k_master[0]; /* Lowest subband related to the synthesis filterbank */ INT usb = v_k_master[numMaster]; /* Stop subband related to the synthesis filterbank */ INT xoverOffset = highBandStartSb - v_k_master[0]; /* Calculate distance in subbands between k0 and kx */ INT goalSb; /* * Initialize the patching parameter */ if (xposctrl == 1) { lsb += xoverOffset; xoverOffset = 0; } goalSb = (INT)((2 * noChannels * 16000 + (fs >> 1)) / fs); /* 16 kHz band */ goalSb = findClosestEntry(goalSb, v_k_master, numMaster, 1); /* Adapt region to master-table */ /* First patch */ sourceStartBand = hTonCorr->shiftStartSb + xoverOffset; targetStopBand = lsb + xoverOffset; /* even (odd) numbered channel must be patched to even (odd) numbered channel */ patch = 0; while (targetStopBand < usb) { /* To many patches */ if (patch >= MAX_NUM_PATCHES) return (1); /*Number of patches to high */ patchParam[patch].guardStartBand = targetStopBand; targetStopBand += sbGuard; patchParam[patch].targetStartBand = targetStopBand; numBandsInPatch = goalSb - targetStopBand; /* get the desired range of the patch */ if (numBandsInPatch >= lsb - sourceStartBand) { /* desired number bands are not available -> patch whole source range */ patchDistance = targetStopBand - sourceStartBand; /* get the targetOffset */ patchDistance = patchDistance & ~1; /* rounding off odd numbers and make all even */ numBandsInPatch = lsb - (targetStopBand - patchDistance); numBandsInPatch = findClosestEntry(targetStopBand + numBandsInPatch, v_k_master, numMaster, 0) - targetStopBand; /* Adapt region to master-table */ } /* desired number bands are available -> get the minimal even patching * distance */ patchDistance = numBandsInPatch + targetStopBand - lsb; /* get minimal distance */ patchDistance = (patchDistance + 1) & ~1; /* rounding up odd numbers and make all even */ if (numBandsInPatch <= 0) { patch--; } else { patchParam[patch].sourceStartBand = targetStopBand - patchDistance; patchParam[patch].targetBandOffs = patchDistance; patchParam[patch].numBandsInPatch = numBandsInPatch; patchParam[patch].sourceStopBand = patchParam[patch].sourceStartBand + numBandsInPatch; targetStopBand += patchParam[patch].numBandsInPatch; } /* All patches but first */ sourceStartBand = hTonCorr->shiftStartSb; /* Check if we are close to goalSb */ if (fixp_abs(targetStopBand - goalSb) < 3) { goalSb = usb; } patch++; } patch--; /* if highest patch contains less than three subband: skip it */ if (patchParam[patch].numBandsInPatch < 3 && patch > 0) { patch--; } hTonCorr->noOfPatches = patch + 1; /* Assign the index-vector, so we know where to look for the high-band. -1 represents a guard-band. */ for (k = 0; k < hTonCorr->patchParam[0].guardStartBand; k++) hTonCorr->indexVector[k] = k; for (i = 0; i < hTonCorr->noOfPatches; i++) { INT sourceStart = hTonCorr->patchParam[i].sourceStartBand; INT targetStart = hTonCorr->patchParam[i].targetStartBand; INT numberOfBands = hTonCorr->patchParam[i].numBandsInPatch; INT startGuardBand = hTonCorr->patchParam[i].guardStartBand; for (k = 0; k < (targetStart - startGuardBand); k++) hTonCorr->indexVector[startGuardBand + k] = -1; for (k = 0; k < numberOfBands; k++) hTonCorr->indexVector[targetStart + k] = sourceStart + k; } return (0); } /**************************************************************************/ /*! \brief Creates an instance of the tonality correction parameter module. The module includes modules for inverse filtering level estimation, missing harmonics detection and noise floor level estimation. \return errorCode, noError if successful. */ /**************************************************************************/ INT FDKsbrEnc_CreateTonCorrParamExtr( HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Pointer to handle to SBR_TON_CORR struct. */ INT chan) /*!< Channel index, needed for mem allocation */ { INT i; FIXP_DBL *quotaMatrix = GetRam_Sbr_quotaMatrix(chan); INT *signMatrix = GetRam_Sbr_signMatrix(chan); if ((NULL == quotaMatrix) || (NULL == signMatrix)) { goto bail; } FDKmemclear(hTonCorr, sizeof(SBR_TON_CORR_EST)); for (i = 0; i < MAX_NO_OF_ESTIMATES; i++) { hTonCorr->quotaMatrix[i] = quotaMatrix + (i * 64); hTonCorr->signMatrix[i] = signMatrix + (i * 64); } if (0 != FDKsbrEnc_CreateSbrMissingHarmonicsDetector( &hTonCorr->sbrMissingHarmonicsDetector, chan)) { goto bail; } return 0; bail: hTonCorr->quotaMatrix[0] = quotaMatrix; hTonCorr->signMatrix[0] = signMatrix; FDKsbrEnc_DeleteTonCorrParamExtr(hTonCorr); return -1; } /**************************************************************************/ /*! \brief Initialize an instance of the tonality correction parameter module. The module includes modules for inverse filtering level estimation, missing harmonics detection and noise floor level estimation. \return errorCode, noError if successful. */ /**************************************************************************/ INT FDKsbrEnc_InitTonCorrParamExtr( INT frameSize, /*!< Current SBR frame size. */ HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Pointer to handle to SBR_TON_CORR struct. */ HANDLE_SBR_CONFIG_DATA sbrCfg, /*!< Pointer to SBR configuration parameters. */ INT timeSlots, /*!< Number of time-slots per frame */ INT xposCtrl, /*!< Different patch modes. */ INT ana_max_level, /*!< Maximum level of the adaptive noise. */ INT noiseBands, /*!< Number of noise bands per octave. */ INT noiseFloorOffset, /*!< Noise floor offset. */ UINT useSpeechConfig) /*!< Speech or music tuning. */ { INT nCols = sbrCfg->noQmfSlots; INT fs = sbrCfg->sampleFreq; INT noQmfChannels = sbrCfg->noQmfBands; INT highBandStartSb = sbrCfg->freqBandTable[LOW_RES][0]; UCHAR *v_k_master = sbrCfg->v_k_master; INT numMaster = sbrCfg->num_Master; UCHAR **freqBandTable = sbrCfg->freqBandTable; INT *nSfb = sbrCfg->nSfb; INT i; /* Reset the patching and allocate memory for the quota matrix. Assuming parameters for the LPC analysis. */ if (sbrCfg->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { switch (timeSlots) { case NUMBER_TIME_SLOTS_1920: hTonCorr->lpcLength[0] = 8 - LPC_ORDER; hTonCorr->lpcLength[1] = 7 - LPC_ORDER; hTonCorr->numberOfEstimates = NO_OF_ESTIMATES_LD; hTonCorr->numberOfEstimatesPerFrame = 2; /* sbrCfg->noQmfSlots / 7 */ hTonCorr->frameStartIndexInvfEst = 0; hTonCorr->transientPosOffset = FRAME_MIDDLE_SLOT_512LD; break; case NUMBER_TIME_SLOTS_2048: hTonCorr->lpcLength[0] = 8 - LPC_ORDER; hTonCorr->lpcLength[1] = 8 - LPC_ORDER; hTonCorr->numberOfEstimates = NO_OF_ESTIMATES_LD; hTonCorr->numberOfEstimatesPerFrame = 2; /* sbrCfg->noQmfSlots / 8 */ hTonCorr->frameStartIndexInvfEst = 0; hTonCorr->transientPosOffset = FRAME_MIDDLE_SLOT_512LD; break; } } else switch (timeSlots) { case NUMBER_TIME_SLOTS_2048: hTonCorr->lpcLength[0] = 16 - LPC_ORDER; /* blockLength[0] */ hTonCorr->lpcLength[1] = 16 - LPC_ORDER; /* blockLength[0] */ hTonCorr->numberOfEstimates = NO_OF_ESTIMATES_LC; hTonCorr->numberOfEstimatesPerFrame = sbrCfg->noQmfSlots / 16; hTonCorr->frameStartIndexInvfEst = 0; hTonCorr->transientPosOffset = FRAME_MIDDLE_SLOT_2048; break; case NUMBER_TIME_SLOTS_1920: hTonCorr->lpcLength[0] = 15 - LPC_ORDER; /* blockLength[0] */ hTonCorr->lpcLength[1] = 15 - LPC_ORDER; /* blockLength[0] */ hTonCorr->numberOfEstimates = NO_OF_ESTIMATES_LC; hTonCorr->numberOfEstimatesPerFrame = sbrCfg->noQmfSlots / 15; hTonCorr->frameStartIndexInvfEst = 0; hTonCorr->transientPosOffset = FRAME_MIDDLE_SLOT_1920; break; default: return -1; } hTonCorr->bufferLength = nCols; hTonCorr->stepSize = hTonCorr->lpcLength[0] + LPC_ORDER; /* stepSize[0] implicitly 0. */ hTonCorr->nextSample = LPC_ORDER; /* firstSample */ hTonCorr->move = hTonCorr->numberOfEstimates - hTonCorr->numberOfEstimatesPerFrame; /* Number of estimates to move when buffering.*/ if (hTonCorr->move < 0) { return -1; } hTonCorr->startIndexMatrix = hTonCorr->numberOfEstimates - hTonCorr->numberOfEstimatesPerFrame; /* Where to store the latest estimations in the tonality Matrix.*/ hTonCorr->frameStartIndex = 0; /* Where in the tonality matrix the current frame (to be sent to the decoder) starts. */ hTonCorr->prevTransientFlag = 0; hTonCorr->transientNextFrame = 0; hTonCorr->noQmfChannels = noQmfChannels; for (i = 0; i < hTonCorr->numberOfEstimates; i++) { FDKmemclear(hTonCorr->quotaMatrix[i], sizeof(FIXP_DBL) * noQmfChannels); FDKmemclear(hTonCorr->signMatrix[i], sizeof(INT) * noQmfChannels); } /* Reset the patch.*/ hTonCorr->guard = 0; hTonCorr->shiftStartSb = 1; if (resetPatch(hTonCorr, xposCtrl, highBandStartSb, v_k_master, numMaster, fs, noQmfChannels)) return (1); if (FDKsbrEnc_InitSbrNoiseFloorEstimate( &hTonCorr->sbrNoiseFloorEstimate, ana_max_level, freqBandTable[LO], nSfb[LO], noiseBands, noiseFloorOffset, timeSlots, useSpeechConfig)) return (1); if (FDKsbrEnc_initInvFiltDetector( &hTonCorr->sbrInvFilt, hTonCorr->sbrNoiseFloorEstimate.freqBandTableQmf, hTonCorr->sbrNoiseFloorEstimate.noNoiseBands, useSpeechConfig)) return (1); if (FDKsbrEnc_InitSbrMissingHarmonicsDetector( &hTonCorr->sbrMissingHarmonicsDetector, fs, frameSize, nSfb[HI], noQmfChannels, hTonCorr->numberOfEstimates, hTonCorr->move, hTonCorr->numberOfEstimatesPerFrame, sbrCfg->sbrSyntaxFlags)) return (1); return (0); } /**************************************************************************/ /*! \brief resets tonality correction parameter module. \return errorCode, noError if successful. */ /**************************************************************************/ INT FDKsbrEnc_ResetTonCorrParamExtr( HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct. */ INT xposctrl, /*!< Different patch modes. */ INT highBandStartSb, /*!< Start band of the SBR range. */ UCHAR *v_k_master, /*!< Master frequency table from which all other table are derived.*/ INT numMaster, /*!< Number of elements in the master table. */ INT fs, /*!< Sampling frequency (of the SBR part). */ UCHAR * *freqBandTable, /*!< Frequency band table for low-res and high-res. */ INT *nSfb, /*!< Number of frequency bands (hig-res and low-res). */ INT noQmfChannels /*!< Number of QMF channels. */ ) { /* Reset the patch.*/ hTonCorr->guard = 0; hTonCorr->shiftStartSb = 1; if (resetPatch(hTonCorr, xposctrl, highBandStartSb, v_k_master, numMaster, fs, noQmfChannels)) return (1); /* Reset the noise floor estimate.*/ if (FDKsbrEnc_resetSbrNoiseFloorEstimate(&hTonCorr->sbrNoiseFloorEstimate, freqBandTable[LO], nSfb[LO])) return (1); /* Reset the inveerse filtereing detector. */ if (FDKsbrEnc_resetInvFiltDetector( &hTonCorr->sbrInvFilt, hTonCorr->sbrNoiseFloorEstimate.freqBandTableQmf, hTonCorr->sbrNoiseFloorEstimate.noNoiseBands)) return (1); /* Reset the missing harmonics detector. */ if (FDKsbrEnc_ResetSbrMissingHarmonicsDetector( &hTonCorr->sbrMissingHarmonicsDetector, nSfb[HI])) return (1); return (0); } /**************************************************************************/ /*! \brief Deletes the tonality correction paramtere module. \return none */ /**************************************************************************/ void FDKsbrEnc_DeleteTonCorrParamExtr( HANDLE_SBR_TON_CORR_EST hTonCorr) /*!< Handle to SBR_TON_CORR struct. */ { if (hTonCorr) { FreeRam_Sbr_quotaMatrix(hTonCorr->quotaMatrix); FreeRam_Sbr_signMatrix(hTonCorr->signMatrix); FDKsbrEnc_DeleteSbrMissingHarmonicsDetector( &hTonCorr->sbrMissingHarmonicsDetector); } } fdk-aac-2.0.2/libSBRenc/src/ton_corr.h000066400000000000000000000302741404225310700173750ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief General tonality correction detector module. */ #ifndef TON_CORR_H #define TON_CORR_H #include "sbr_encoder.h" #include "mh_det.h" #include "nf_est.h" #include "invf_est.h" #define MAX_NUM_PATCHES 6 #define SCALE_NRGVEC 4 /** parameter set for one single patch */ typedef struct { INT sourceStartBand; /*!< first band in lowbands where to take the samples from */ INT sourceStopBand; /*!< first band in lowbands which is not included in the patch anymore */ INT guardStartBand; /*!< first band in highbands to be filled with zeros in order to reduce interferences between patches */ INT targetStartBand; /*!< first band in highbands to be filled with whitened lowband signal */ INT targetBandOffs; /*!< difference between 'startTargetBand' and 'startSourceBand' */ INT numBandsInPatch; /*!< number of consecutive bands in this one patch */ } PATCH_PARAM; typedef struct { INT switchInverseFilt; /*!< Flag to enable dynamic adaption of invf. detection */ INT noQmfChannels; INT bufferLength; /*!< Length of the r and i buffers. */ INT stepSize; /*!< Stride for the lpc estimate. */ INT numberOfEstimates; /*!< The total number of estiamtes, available in the quotaMatrix.*/ UINT numberOfEstimatesPerFrame; /*!< The number of estimates per frame available in the quotaMatrix.*/ INT lpcLength[2]; /*!< Segment length used for second order LPC analysis.*/ INT nextSample; /*!< Where to start the LPC analysis of the current frame.*/ INT move; /*!< How many estimates to move in the quotaMatrix, when buffering. */ INT frameStartIndex; /*!< The start index for the current frame in the r and i buffers. */ INT startIndexMatrix; /*!< The start index for the current frame in the quotaMatrix. */ INT frameStartIndexInvfEst; /*!< The start index of the inverse filtering, not the same as the others, dependent on what decoder is used (buffer opt, or no buffer opt). */ INT prevTransientFlag; /*!< The transisent flag (from the transient detector) for the previous frame. */ INT transientNextFrame; /*!< Flag to indicate that the transient will show up in the next frame. */ INT transientPosOffset; /*!< An offset value to match the transient pos as calculated by the transient detector with the actual position in the frame.*/ INT* signMatrix[MAX_NO_OF_ESTIMATES]; /*!< Matrix holding the sign of each channe, i.e. indicating in what part of a QMF channel a possible sine is. */ FIXP_DBL* quotaMatrix[MAX_NO_OF_ESTIMATES]; /*!< Matrix holding the quota values for all estimates, all channels. */ FIXP_DBL nrgVector[MAX_NO_OF_ESTIMATES]; /*!< Vector holding the averaged energies for every QMF band. */ FIXP_DBL nrgVectorFreq[64]; /*!< Vector holding the averaged energies for every QMF channel */ SCHAR indexVector[64]; /*!< Index vector poINTing to the correct lowband channel, when indexing a highband channel, -1 represents a guard band */ PATCH_PARAM patchParam[MAX_NUM_PATCHES]; /*!< new parameter set for patching */ INT guard; /*!< number of guardbands between every patch */ INT shiftStartSb; /*!< lowest subband of source range to be included in the patches */ INT noOfPatches; /*!< number of patches */ SBR_MISSING_HARMONICS_DETECTOR sbrMissingHarmonicsDetector; /*!< SBR_MISSING_HARMONICS_DETECTOR struct. */ SBR_NOISE_FLOOR_ESTIMATE sbrNoiseFloorEstimate; /*!< SBR_NOISE_FLOOR_ESTIMATE struct. */ SBR_INV_FILT_EST sbrInvFilt; /*!< SBR_INV_FILT_EST struct. */ } SBR_TON_CORR_EST; typedef SBR_TON_CORR_EST* HANDLE_SBR_TON_CORR_EST; void FDKsbrEnc_TonCorrParamExtr( HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct. */ INVF_MODE* infVec, /*!< Vector where the inverse filtering levels will be stored. */ FIXP_DBL* noiseLevels, /*!< Vector where the noise levels will be stored. */ INT* missingHarmonicFlag, /*!< Flag set to one or zero, dependent on if any strong sines are missing.*/ UCHAR* missingHarmonicsIndex, /*!< Vector indicating where sines are missing. */ UCHAR* envelopeCompensation, /*!< Vector to store compensation values for the energies in. */ const SBR_FRAME_INFO* frameInfo, /*!< Frame info struct, contains the time and frequency grid of the current frame.*/ UCHAR* transientInfo, /*!< Transient info.*/ UCHAR* freqBandTable, /*!< Frequency band tables for high-res.*/ INT nSfb, /*!< Number of scalefactor bands for high-res. */ XPOS_MODE xposType, /*!< Type of transposer used in the decoder.*/ UINT sbrSyntaxFlags); INT FDKsbrEnc_CreateTonCorrParamExtr( HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Pointer to handle to SBR_TON_CORR struct. */ INT chan); /*!< Channel index, needed for mem allocation */ INT FDKsbrEnc_InitTonCorrParamExtr( INT frameSize, /*!< Current SBR frame size. */ HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Pointer to handle to SBR_TON_CORR struct. */ HANDLE_SBR_CONFIG_DATA sbrCfg, /*!< Pointer to SBR configuration parameters. */ INT timeSlots, /*!< Number of time-slots per frame */ INT xposCtrl, /*!< Different patch modes. */ INT ana_max_level, /*!< Maximum level of the adaptive noise. */ INT noiseBands, /*!< Number of noise bands per octave. */ INT noiseFloorOffset, /*!< Noise floor offset. */ UINT useSpeechConfig /*!< Speech or music tuning. */ ); void FDKsbrEnc_DeleteTonCorrParamExtr( HANDLE_SBR_TON_CORR_EST hTonCorr); /*!< Handle to SBR_TON_CORR struct. */ void FDKsbrEnc_CalculateTonalityQuotas( HANDLE_SBR_TON_CORR_EST hTonCorr, FIXP_DBL** sourceBufferReal, FIXP_DBL** sourceBufferImag, INT usb, INT qmfScale /*!< sclefactor of QMF subsamples */ ); INT FDKsbrEnc_ResetTonCorrParamExtr( HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct. */ INT xposctrl, /*!< Different patch modes. */ INT highBandStartSb, /*!< Start band of the SBR range. */ UCHAR* v_k_master, /*!< Master frequency table from which all other table are derived.*/ INT numMaster, /*!< Number of elements in the master table. */ INT fs, /*!< Sampling frequency (of the SBR part). */ UCHAR** freqBandTable, /*!< Frequency band table for low-res and high-res. */ INT* nSfb, /*!< Number of frequency bands (hig-res and low-res). */ INT noQmfChannels /*!< Number of QMF channels. */ ); #endif fdk-aac-2.0.2/libSBRenc/src/tran_det.cpp000066400000000000000000001214361404225310700177040ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Tobias Chalupka Description: SBR encoder transient detector *******************************************************************************/ #include "tran_det.h" #include "fram_gen.h" #include "sbrenc_ram.h" #include "sbr_misc.h" #include "genericStds.h" #define NORM_QMF_ENERGY 9.31322574615479E-10 /* 2^-30 */ /* static FIXP_DBL ABS_THRES = fixMax( FL2FXCONST_DBL(1.28e5 * * NORM_QMF_ENERGY), (FIXP_DBL)1) Minimum threshold for detecting changes */ #define ABS_THRES ((FIXP_DBL)16) /******************************************************************************* Functionname: spectralChange ******************************************************************************* \brief Calculates a measure for the spectral change within the frame The function says how good it would be to split the frame at the given border position into 2 envelopes. The return value delta_sum is scaled with the factor 1/64 \return calculated value *******************************************************************************/ #define NRG_SHIFT 3 /* for energy summation */ static FIXP_DBL spectralChange( FIXP_DBL Energies[NUMBER_TIME_SLOTS_2304][MAX_FREQ_COEFFS], INT *scaleEnergies, FIXP_DBL EnergyTotal, INT nSfb, INT start, INT border, INT YBufferWriteOffset, INT stop, INT *result_e) { INT i, j; INT len1, len2; SCHAR energies_e_diff[NUMBER_TIME_SLOTS_2304], energies_e, energyTotal_e = 19, energies_e_add; SCHAR prevEnergies_e_diff, newEnergies_e_diff; FIXP_DBL tmp0, tmp1; FIXP_DBL delta, delta_sum; INT accu_e, tmp_e; delta_sum = FL2FXCONST_DBL(0.0f); *result_e = 0; len1 = border - start; len2 = stop - border; /* prefer borders near the middle of the frame */ FIXP_DBL pos_weight; pos_weight = FL2FXCONST_DBL(0.5f) - (len1 * GetInvInt(len1 + len2)); pos_weight = /*FL2FXCONST_DBL(1.0)*/ (FIXP_DBL)MAXVAL_DBL - (fMult(pos_weight, pos_weight) << 2); /*** Calc scaling for energies ***/ FDK_ASSERT(scaleEnergies[0] >= 0); FDK_ASSERT(scaleEnergies[1] >= 0); energies_e = 19 - fMin(scaleEnergies[0], scaleEnergies[1]); /* limit shift for energy accumulation, energies_e can be -10 min. */ if (energies_e < -10) { energies_e_add = -10 - energies_e; energies_e = -10; } else if (energies_e > 17) { energies_e_add = energies_e - 17; energies_e = 17; } else { energies_e_add = 0; } /* compensate scaling differences between scaleEnergies[0] and * scaleEnergies[1] */ prevEnergies_e_diff = scaleEnergies[0] - fMin(scaleEnergies[0], scaleEnergies[1]) + energies_e_add + NRG_SHIFT; newEnergies_e_diff = scaleEnergies[1] - fMin(scaleEnergies[0], scaleEnergies[1]) + energies_e_add + NRG_SHIFT; prevEnergies_e_diff = fMin(prevEnergies_e_diff, DFRACT_BITS - 1); newEnergies_e_diff = fMin(newEnergies_e_diff, DFRACT_BITS - 1); for (i = start; i < YBufferWriteOffset; i++) { energies_e_diff[i] = prevEnergies_e_diff; } for (i = YBufferWriteOffset; i < stop; i++) { energies_e_diff[i] = newEnergies_e_diff; } /* Sum up energies of all QMF-timeslots for both halfs */ FDK_ASSERT(len1 <= 8); /* otherwise an overflow is possible */ FDK_ASSERT(len2 <= 8); /* otherwise an overflow is possible */ for (j = 0; j < nSfb; j++) { FIXP_DBL accu1 = FL2FXCONST_DBL(0.f); FIXP_DBL accu2 = FL2FXCONST_DBL(0.f); accu_e = energies_e + 3; /* Sum up energies in first half */ for (i = start; i < border; i++) { accu1 += scaleValue(Energies[i][j], -energies_e_diff[i]); } /* Sum up energies in second half */ for (i = border; i < stop; i++) { accu2 += scaleValue(Energies[i][j], -energies_e_diff[i]); } /* Ensure certain energy to prevent division by zero and to prevent * splitting for very low levels */ accu1 = fMax(accu1, (FIXP_DBL)len1); accu2 = fMax(accu2, (FIXP_DBL)len2); /* Energy change in current band */ #define LN2 FL2FXCONST_DBL(0.6931471806f) /* ln(2) */ tmp0 = fLog2(accu2, accu_e) - fLog2(accu1, accu_e); tmp1 = fLog2((FIXP_DBL)len1, 31) - fLog2((FIXP_DBL)len2, 31); delta = fMult(LN2, (tmp0 + tmp1)); delta = (FIXP_DBL)fAbs(delta); /* Weighting with amplitude ratio of this band */ accu_e++; /* scale at least one bit due to (accu1+accu2) */ accu1 >>= 1; accu2 >>= 1; if (accu_e & 1) { accu_e++; /* for a defined square result exponent, the exponent has to be even */ accu1 >>= 1; accu2 >>= 1; } delta_sum += fMult(sqrtFixp(accu1 + accu2), delta); *result_e = ((accu_e >> 1) + LD_DATA_SHIFT); } if (energyTotal_e & 1) { energyTotal_e += 1; /* for a defined square result exponent, the exponent has to be even */ EnergyTotal >>= 1; } delta_sum = fMult(delta_sum, invSqrtNorm2(EnergyTotal, &tmp_e)); *result_e = *result_e + (tmp_e - (energyTotal_e >> 1)); return fMult(delta_sum, pos_weight); } /******************************************************************************* Functionname: addLowbandEnergies ******************************************************************************* \brief Calculates total lowband energy The input values Energies[0] (low-band) are scaled by the factor 2^(14-*scaleEnergies[0]) The input values Energies[1] (high-band) are scaled by the factor 2^(14-*scaleEnergies[1]) \return total energy in the lowband, scaled by the factor 2^19 *******************************************************************************/ static FIXP_DBL addLowbandEnergies(FIXP_DBL **Energies, int *scaleEnergies, int YBufferWriteOffset, int nrgSzShift, int tran_off, UCHAR *freqBandTable, int slots) { INT nrgTotal_e; FIXP_DBL nrgTotal_m; FIXP_DBL accu1 = FL2FXCONST_DBL(0.0f); FIXP_DBL accu2 = FL2FXCONST_DBL(0.0f); int tran_offdiv2 = tran_off >> nrgSzShift; const int sc1 = DFRACT_BITS - fNormz((FIXP_DBL)fMax( 1, (freqBandTable[0] * (YBufferWriteOffset - tran_offdiv2) - 1))); const int sc2 = DFRACT_BITS - fNormz((FIXP_DBL)fMax( 1, (freqBandTable[0] * (tran_offdiv2 + (slots >> nrgSzShift) - YBufferWriteOffset) - 1))); int ts, k; /* Sum up lowband energy from one frame at offset tran_off */ /* freqBandTable[LORES] has MAX_FREQ_COEFFS/2 +1 coeefs max. */ for (ts = tran_offdiv2; ts < YBufferWriteOffset; ts++) { for (k = 0; k < freqBandTable[0]; k++) { accu1 += Energies[ts][k] >> sc1; } } for (; ts < tran_offdiv2 + (slots >> nrgSzShift); ts++) { for (k = 0; k < freqBandTable[0]; k++) { accu2 += Energies[ts][k] >> sc2; } } nrgTotal_m = fAddNorm(accu1, (sc1 - 5) - scaleEnergies[0], accu2, (sc2 - 5) - scaleEnergies[1], &nrgTotal_e); nrgTotal_m = scaleValueSaturate(nrgTotal_m, nrgTotal_e); return (nrgTotal_m); } /******************************************************************************* Functionname: addHighbandEnergies ******************************************************************************* \brief Add highband energies Highband energies are mapped to an array with smaller dimension: Its time resolution is only 1 SBR-timeslot and its frequency resolution is 1 SBR-band. Therefore the data to be fed into the spectralChange function is reduced. The values EnergiesM are scaled by the factor (2^19-scaleEnergies[0]) for slots=YBufferWriteOffset. \return total energy in the highband, scaled by factor 2^19 *******************************************************************************/ static FIXP_DBL addHighbandEnergies( FIXP_DBL **RESTRICT Energies, /*!< input */ INT *scaleEnergies, INT YBufferWriteOffset, FIXP_DBL EnergiesM[NUMBER_TIME_SLOTS_2304] [MAX_FREQ_COEFFS], /*!< Combined output */ UCHAR *RESTRICT freqBandTable, INT nSfb, INT sbrSlots, INT timeStep) { INT i, j, k, slotIn, slotOut, scale[2]; INT li, ui; FIXP_DBL nrgTotal; FIXP_DBL accu = FL2FXCONST_DBL(0.0f); /* Combine QMF-timeslots to SBR-timeslots, combine QMF-bands to SBR-bands, combine Left and Right channel */ for (slotOut = 0; slotOut < sbrSlots; slotOut++) { /* Note: Below slotIn = slotOut and not slotIn = timeStep*slotOut because the Energies[] time resolution is always the SBR slot resolution regardless of the timeStep. */ slotIn = slotOut; for (j = 0; j < nSfb; j++) { accu = FL2FXCONST_DBL(0.0f); li = freqBandTable[j]; ui = freqBandTable[j + 1]; for (k = li; k < ui; k++) { for (i = 0; i < timeStep; i++) { accu += Energies[slotIn][k] >> 5; } } EnergiesM[slotOut][j] = accu; } } /* scale energies down before add up */ scale[0] = fixMin(8, scaleEnergies[0]); scale[1] = fixMin(8, scaleEnergies[1]); if ((scaleEnergies[0] - scale[0]) > (DFRACT_BITS - 1) || (scaleEnergies[1] - scale[1]) > (DFRACT_BITS - 1)) nrgTotal = FL2FXCONST_DBL(0.0f); else { /* Now add all energies */ accu = FL2FXCONST_DBL(0.0f); for (slotOut = 0; slotOut < YBufferWriteOffset; slotOut++) { for (j = 0; j < nSfb; j++) { accu += (EnergiesM[slotOut][j] >> scale[0]); } } nrgTotal = accu >> (scaleEnergies[0] - scale[0]); for (slotOut = YBufferWriteOffset; slotOut < sbrSlots; slotOut++) { for (j = 0; j < nSfb; j++) { accu += (EnergiesM[slotOut][j] >> scale[0]); } } nrgTotal = fAddSaturate(nrgTotal, accu >> (scaleEnergies[1] - scale[1])); } return (nrgTotal); } /******************************************************************************* Functionname: FDKsbrEnc_frameSplitter ******************************************************************************* \brief Decides if a FIXFIX-frame shall be splitted into 2 envelopes If no transient has been detected before, the frame can still be splitted into 2 envelopes. *******************************************************************************/ void FDKsbrEnc_frameSplitter( FIXP_DBL **Energies, INT *scaleEnergies, HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientDetector, UCHAR *freqBandTable, UCHAR *tran_vector, int YBufferWriteOffset, int YBufferSzShift, int nSfb, int timeStep, int no_cols, FIXP_DBL *tonality) { if (tran_vector[1] == 0) /* no transient was detected */ { FIXP_DBL delta; INT delta_e; FIXP_DBL(*EnergiesM)[MAX_FREQ_COEFFS]; FIXP_DBL EnergyTotal, newLowbandEnergy, newHighbandEnergy; INT border; INT sbrSlots = fMultI(GetInvInt(timeStep), no_cols); C_ALLOC_SCRATCH_START(_EnergiesM, FIXP_DBL, NUMBER_TIME_SLOTS_2304 * MAX_FREQ_COEFFS) FDK_ASSERT(sbrSlots * timeStep == no_cols); EnergiesM = (FIXP_DBL(*)[MAX_FREQ_COEFFS])_EnergiesM; /* Get Lowband-energy over a range of 2 frames (Look half a frame back and ahead). */ newLowbandEnergy = addLowbandEnergies( Energies, scaleEnergies, YBufferWriteOffset, YBufferSzShift, h_sbrTransientDetector->tran_off, freqBandTable, no_cols); newHighbandEnergy = addHighbandEnergies(Energies, scaleEnergies, YBufferWriteOffset, EnergiesM, freqBandTable, nSfb, sbrSlots, timeStep); { /* prevLowBandEnergy: Corresponds to 1 frame, starting with half a frame look-behind newLowbandEnergy: Corresponds to 1 frame, starting in the middle of the current frame */ EnergyTotal = (newLowbandEnergy >> 1) + (h_sbrTransientDetector->prevLowBandEnergy >> 1); /* mean of new and prev LB NRG */ EnergyTotal = fAddSaturate(EnergyTotal, newHighbandEnergy); /* Add HB NRG */ /* The below border should specify the same position as the middle border of a FIXFIX-frame with 2 envelopes. */ border = (sbrSlots + 1) >> 1; if ((INT)EnergyTotal & 0xffffffe0 && (scaleEnergies[0] < 32 || scaleEnergies[1] < 32)) /* i.e. > 31 */ { delta = spectralChange(EnergiesM, scaleEnergies, EnergyTotal, nSfb, 0, border, YBufferWriteOffset, sbrSlots, &delta_e); } else { delta = FL2FXCONST_DBL(0.0f); delta_e = 0; /* set tonality to 0 when energy is very low, since the amplitude resolution should then be low as well */ *tonality = FL2FXCONST_DBL(0.0f); } if (fIsLessThan(h_sbrTransientDetector->split_thr_m, h_sbrTransientDetector->split_thr_e, delta, delta_e)) { tran_vector[0] = 1; /* Set flag for splitting */ } else { tran_vector[0] = 0; } } /* Update prevLowBandEnergy */ h_sbrTransientDetector->prevLowBandEnergy = newLowbandEnergy; h_sbrTransientDetector->prevHighBandEnergy = newHighbandEnergy; C_ALLOC_SCRATCH_END(_EnergiesM, FIXP_DBL, NUMBER_TIME_SLOTS_2304 * MAX_FREQ_COEFFS) } } /* * Calculate transient energy threshold for each QMF band */ static void calculateThresholds(FIXP_DBL **RESTRICT Energies, INT *RESTRICT scaleEnergies, FIXP_DBL *RESTRICT thresholds, int YBufferWriteOffset, int YBufferSzShift, int noCols, int noRows, int tran_off) { FIXP_DBL mean_val, std_val, temp; FIXP_DBL i_noCols; FIXP_DBL i_noCols1; FIXP_DBL accu, accu0, accu1; int scaleFactor0, scaleFactor1, commonScale; int i, j; i_noCols = GetInvInt(noCols + tran_off) << YBufferSzShift; i_noCols1 = GetInvInt(noCols + tran_off - 1) << YBufferSzShift; /* calc minimum scale of energies of previous and current frame */ commonScale = fixMin(scaleEnergies[0], scaleEnergies[1]); /* calc scalefactors to adapt energies to common scale */ scaleFactor0 = fixMin((scaleEnergies[0] - commonScale), (DFRACT_BITS - 1)); scaleFactor1 = fixMin((scaleEnergies[1] - commonScale), (DFRACT_BITS - 1)); FDK_ASSERT((scaleFactor0 >= 0) && (scaleFactor1 >= 0)); /* calculate standard deviation in every subband */ for (i = 0; i < noRows; i++) { int startEnergy = (tran_off >> YBufferSzShift); int endEnergy = ((noCols >> YBufferSzShift) + tran_off); int shift; /* calculate mean value over decimated energy values (downsampled by 2). */ accu0 = accu1 = FL2FXCONST_DBL(0.0f); for (j = startEnergy; j < YBufferWriteOffset; j++) accu0 = fMultAddDiv2(accu0, Energies[j][i], i_noCols); for (; j < endEnergy; j++) accu1 = fMultAddDiv2(accu1, Energies[j][i], i_noCols); mean_val = ((accu0 << 1) >> scaleFactor0) + ((accu1 << 1) >> scaleFactor1); /* average */ shift = fixMax( 0, CountLeadingBits(mean_val) - 6); /* -6 to keep room for accumulating upto N = 24 values */ /* calculate standard deviation */ accu = FL2FXCONST_DBL(0.0f); /* summe { ((mean_val-nrg)^2) * i_noCols1 } */ for (j = startEnergy; j < YBufferWriteOffset; j++) { temp = ((FIXP_DBL)mean_val - ((FIXP_DBL)Energies[j][i] >> scaleFactor0)) << shift; temp = fPow2Div2(temp); accu = fMultAddDiv2(accu, temp, i_noCols1); } for (; j < endEnergy; j++) { temp = ((FIXP_DBL)mean_val - ((FIXP_DBL)Energies[j][i] >> scaleFactor1)) << shift; temp = fPow2Div2(temp); accu = fMultAddDiv2(accu, temp, i_noCols1); } accu <<= 2; std_val = sqrtFixp(accu) >> shift; /* standard deviation */ /* Take new threshold as average of calculated standard deviation ratio and old threshold if greater than absolute threshold */ temp = (commonScale <= (DFRACT_BITS - 1)) ? fMult(FL2FXCONST_DBL(0.66f), thresholds[i]) + (fMult(FL2FXCONST_DBL(0.34f), std_val) >> commonScale) : (FIXP_DBL)0; thresholds[i] = fixMax(ABS_THRES, temp); FDK_ASSERT(commonScale >= 0); } } /* * Calculate transient levels for each QMF time slot. */ static void extractTransientCandidates( FIXP_DBL **RESTRICT Energies, INT *RESTRICT scaleEnergies, FIXP_DBL *RESTRICT thresholds, FIXP_DBL *RESTRICT transients, int YBufferWriteOffset, int YBufferSzShift, int noCols, int start_band, int stop_band, int tran_off, int addPrevSamples) { FIXP_DBL i_thres; C_ALLOC_SCRATCH_START(EnergiesTemp, FIXP_DBL, 2 * 32) int tmpScaleEnergies0, tmpScaleEnergies1; int endCond; int startEnerg, endEnerg; int i, j, jIndex, jpBM; tmpScaleEnergies0 = scaleEnergies[0]; tmpScaleEnergies1 = scaleEnergies[1]; /* Scale value for first energies, upto YBufferWriteOffset */ tmpScaleEnergies0 = fixMin(tmpScaleEnergies0, MAX_SHIFT_DBL); /* Scale value for first energies, from YBufferWriteOffset upwards */ tmpScaleEnergies1 = fixMin(tmpScaleEnergies1, MAX_SHIFT_DBL); FDK_ASSERT((tmpScaleEnergies0 >= 0) && (tmpScaleEnergies1 >= 0)); /* Keep addPrevSamples extra previous transient candidates. */ FDKmemmove(transients, transients + noCols - addPrevSamples, (tran_off + addPrevSamples) * sizeof(FIXP_DBL)); FDKmemclear(transients + tran_off + addPrevSamples, noCols * sizeof(FIXP_DBL)); endCond = noCols; /* Amount of new transient values to be calculated. */ startEnerg = (tran_off - 3) >> YBufferSzShift; /* >>YBufferSzShift because of amount of energy values. -3 because of neighbors being watched. */ endEnerg = ((noCols + (YBufferWriteOffset << YBufferSzShift)) - 1) >> YBufferSzShift; /* YBufferSzShift shifts because of half energy values. */ /* Compute differential values with two different weightings in every subband */ for (i = start_band; i < stop_band; i++) { FIXP_DBL thres = thresholds[i]; if ((LONG)thresholds[i] >= 256) i_thres = (LONG)((LONG)MAXVAL_DBL / ((((LONG)thresholds[i])) + 1)) << (32 - 24); else i_thres = (LONG)MAXVAL_DBL; /* Copy one timeslot and de-scale and de-squish */ if (YBufferSzShift == 1) { for (j = startEnerg; j < YBufferWriteOffset; j++) { FIXP_DBL tmp = Energies[j][i]; EnergiesTemp[(j << 1) + 1] = EnergiesTemp[j << 1] = tmp >> tmpScaleEnergies0; } for (; j <= endEnerg; j++) { FIXP_DBL tmp = Energies[j][i]; EnergiesTemp[(j << 1) + 1] = EnergiesTemp[j << 1] = tmp >> tmpScaleEnergies1; } } else { for (j = startEnerg; j < YBufferWriteOffset; j++) { FIXP_DBL tmp = Energies[j][i]; EnergiesTemp[j] = tmp >> tmpScaleEnergies0; } for (; j <= endEnerg; j++) { FIXP_DBL tmp = Energies[j][i]; EnergiesTemp[j] = tmp >> tmpScaleEnergies1; } } /* Detect peaks in energy values. */ jIndex = tran_off; jpBM = jIndex + addPrevSamples; for (j = endCond; j--; jIndex++, jpBM++) { FIXP_DBL delta, tran; int d; delta = (FIXP_DBL)0; tran = (FIXP_DBL)0; for (d = 1; d < 4; d++) { delta += EnergiesTemp[jIndex + d]; /* R */ delta -= EnergiesTemp[jIndex - d]; /* L */ delta -= thres; if (delta > (FIXP_DBL)0) { tran = fMultAddDiv2(tran, i_thres, delta); } } transients[jpBM] += (tran << 1); } } C_ALLOC_SCRATCH_END(EnergiesTemp, FIXP_DBL, 2 * 32) } void FDKsbrEnc_transientDetect(HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTran, FIXP_DBL **Energies, INT *scaleEnergies, UCHAR *transient_info, int YBufferWriteOffset, int YBufferSzShift, int timeStep, int frameMiddleBorder) { int no_cols = h_sbrTran->no_cols; int qmfStartSample; int addPrevSamples; int timeStepShift = 0; int i, cond; /* Where to start looking for transients in the transient candidate buffer */ qmfStartSample = timeStep * frameMiddleBorder; /* We need to look one value backwards in the transients, so we might need one * more previous value. */ addPrevSamples = (qmfStartSample > 0) ? 0 : 1; switch (timeStep) { case 1: timeStepShift = 0; break; case 2: timeStepShift = 1; break; case 4: timeStepShift = 2; break; } calculateThresholds(Energies, scaleEnergies, h_sbrTran->thresholds, YBufferWriteOffset, YBufferSzShift, h_sbrTran->no_cols, h_sbrTran->no_rows, h_sbrTran->tran_off); extractTransientCandidates( Energies, scaleEnergies, h_sbrTran->thresholds, h_sbrTran->transients, YBufferWriteOffset, YBufferSzShift, h_sbrTran->no_cols, 0, h_sbrTran->no_rows, h_sbrTran->tran_off, addPrevSamples); transient_info[0] = 0; transient_info[1] = 0; transient_info[2] = 0; /* Offset by the amount of additional previous transient candidates being * kept. */ qmfStartSample += addPrevSamples; /* Check for transients in second granule (pick the last value of subsequent * values) */ for (i = qmfStartSample; i < qmfStartSample + no_cols; i++) { cond = (h_sbrTran->transients[i] < fMult(FL2FXCONST_DBL(0.9f), h_sbrTran->transients[i - 1])) && (h_sbrTran->transients[i - 1] > h_sbrTran->tran_thr); if (cond) { transient_info[0] = (i - qmfStartSample) >> timeStepShift; transient_info[1] = 1; break; } } if (h_sbrTran->frameShift != 0) { /* transient prediction for LDSBR */ /* Check for transients in first qmf-slots of second frame */ for (i = qmfStartSample + no_cols; i < qmfStartSample + no_cols + h_sbrTran->frameShift; i++) { cond = (h_sbrTran->transients[i] < fMult(FL2FXCONST_DBL(0.9f), h_sbrTran->transients[i - 1])) && (h_sbrTran->transients[i - 1] > h_sbrTran->tran_thr); if (cond) { int pos = (int)((i - qmfStartSample - no_cols) >> timeStepShift); if ((pos < 3) && (transient_info[1] == 0)) { transient_info[2] = 1; } break; } } } } int FDKsbrEnc_InitSbrTransientDetector( HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientDetector, UINT sbrSyntaxFlags, /* SBR syntax flags derived from AOT. */ INT frameSize, INT sampleFreq, sbrConfigurationPtr params, int tran_fc, int no_cols, int no_rows, int YBufferWriteOffset, int YBufferSzShift, int frameShift, int tran_off) { INT totalBitrate = params->codecSettings.standardBitrate * params->codecSettings.nChannels; INT codecBitrate = params->codecSettings.bitRate; FIXP_DBL bitrateFactor_m, framedur_fix; INT bitrateFactor_e, tmp_e; FDKmemclear(h_sbrTransientDetector, sizeof(SBR_TRANSIENT_DETECTOR)); h_sbrTransientDetector->frameShift = frameShift; h_sbrTransientDetector->tran_off = tran_off; if (codecBitrate) { bitrateFactor_m = fDivNorm((FIXP_DBL)totalBitrate, (FIXP_DBL)(codecBitrate << 2), &bitrateFactor_e); bitrateFactor_e += 2; } else { bitrateFactor_m = FL2FXCONST_DBL(1.0 / 4.0); bitrateFactor_e = 2; } framedur_fix = fDivNorm(frameSize, sampleFreq); /* The longer the frames, the more often should the FIXFIX- case transmit 2 envelopes instead of 1. Frame durations below 10 ms produce the highest threshold so that practically always only 1 env is transmitted. */ FIXP_DBL tmp = framedur_fix - FL2FXCONST_DBL(0.010); tmp = fixMax(tmp, FL2FXCONST_DBL(0.0001)); tmp = fDivNorm(FL2FXCONST_DBL(0.000075), fPow2(tmp), &tmp_e); bitrateFactor_e = (tmp_e + bitrateFactor_e); if (sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { bitrateFactor_e--; /* divide by 2 */ } FDK_ASSERT(no_cols <= 32); FDK_ASSERT(no_rows <= 64); h_sbrTransientDetector->no_cols = no_cols; h_sbrTransientDetector->tran_thr = (FIXP_DBL)((params->tran_thr << (32 - 24 - 1)) / no_rows); h_sbrTransientDetector->tran_fc = tran_fc; h_sbrTransientDetector->split_thr_m = fMult(tmp, bitrateFactor_m); h_sbrTransientDetector->split_thr_e = bitrateFactor_e; h_sbrTransientDetector->no_rows = no_rows; h_sbrTransientDetector->mode = params->tran_det_mode; h_sbrTransientDetector->prevLowBandEnergy = FL2FXCONST_DBL(0.0f); return (0); } #define ENERGY_SCALING_SIZE 32 INT FDKsbrEnc_InitSbrFastTransientDetector( HANDLE_FAST_TRAN_DET h_sbrFastTransientDetector, const INT time_slots_per_frame, const INT bandwidth_qmf_slot, const INT no_qmf_channels, const INT sbr_qmf_1st_band) { int i; int buff_size; FIXP_DBL myExp; FIXP_DBL myExpSlot; h_sbrFastTransientDetector->lookahead = TRAN_DET_LOOKAHEAD; h_sbrFastTransientDetector->nTimeSlots = time_slots_per_frame; buff_size = h_sbrFastTransientDetector->nTimeSlots + h_sbrFastTransientDetector->lookahead; for (i = 0; i < buff_size; i++) { h_sbrFastTransientDetector->delta_energy[i] = FL2FXCONST_DBL(0.0f); h_sbrFastTransientDetector->energy_timeSlots[i] = FL2FXCONST_DBL(0.0f); h_sbrFastTransientDetector->lowpass_energy[i] = FL2FXCONST_DBL(0.0f); h_sbrFastTransientDetector->transientCandidates[i] = 0; } FDK_ASSERT(bandwidth_qmf_slot > 0.f); h_sbrFastTransientDetector->stopBand = fMin(TRAN_DET_STOP_FREQ / bandwidth_qmf_slot, no_qmf_channels); h_sbrFastTransientDetector->startBand = fMin(sbr_qmf_1st_band, h_sbrFastTransientDetector->stopBand - TRAN_DET_MIN_QMFBANDS); FDK_ASSERT(h_sbrFastTransientDetector->startBand < no_qmf_channels); FDK_ASSERT(h_sbrFastTransientDetector->startBand < h_sbrFastTransientDetector->stopBand); FDK_ASSERT(h_sbrFastTransientDetector->startBand > 1); FDK_ASSERT(h_sbrFastTransientDetector->stopBand > 1); /* the energy weighting and adding up has a headroom of 6 Bits, so up to 64 bands can be added without potential overflow. */ FDK_ASSERT(h_sbrFastTransientDetector->stopBand - h_sbrFastTransientDetector->startBand <= 64); /* QMF_HP_dB_SLOPE_FIX says that we want a 20 dB per 16 kHz HP filter. The following lines map this to the QMF bandwidth. */ #define EXP_E 7 /* 64 (=64) multiplications max, max. allowed sum is 0.5 */ myExp = fMultNorm(QMF_HP_dBd_SLOPE_FIX, 0, (FIXP_DBL)bandwidth_qmf_slot, DFRACT_BITS - 1, EXP_E); myExpSlot = myExp; for (i = 0; i < 64; i++) { /* Calculate dBf over all qmf bands: dBf = (10^(0.002266f/10*bw(slot)))^(band) = = 2^(log2(10)*0.002266f/10*bw(slot)*band) = = 2^(0.00075275f*bw(slot)*band) */ FIXP_DBL dBf_m; /* dBf mantissa */ INT dBf_e; /* dBf exponent */ INT tmp; INT dBf_int; /* dBf integer part */ FIXP_DBL dBf_fract; /* dBf fractional part */ /* myExp*(i+1) = myExp_int - myExp_fract myExp*(i+1) is split up here for better accuracy of CalcInvLdData(), for its result can be split up into an integer and a fractional part */ /* Round up to next integer */ FIXP_DBL myExp_int = (myExpSlot & (FIXP_DBL)0xfe000000) + (FIXP_DBL)0x02000000; /* This is the fractional part that needs to be substracted */ FIXP_DBL myExp_fract = myExp_int - myExpSlot; /* Calc integer part */ dBf_int = CalcInvLdData(myExp_int); /* The result needs to be re-scaled. The ld(myExp_int) had been scaled by EXP_E, the CalcInvLdData expects the operand to be scaled by LD_DATA_SHIFT. Therefore, the correctly scaled result is dBf_int^(2^(EXP_E-LD_DATA_SHIFT)), which is dBf_int^2 */ if (dBf_int <= 46340) { /* compare with maximum allowed value for signed integer multiplication, 46340 = (INT)floor(sqrt((double)(((UINT)1<<(DFRACT_BITS-1))-1))) */ dBf_int *= dBf_int; /* Calc fractional part */ dBf_fract = CalcInvLdData(-myExp_fract); /* The result needs to be re-scaled. The ld(myExp_fract) had been scaled by EXP_E, the CalcInvLdData expects the operand to be scaled by LD_DATA_SHIFT. Therefore, the correctly scaled result is dBf_fract^(2^(EXP_E-LD_DATA_SHIFT)), which is dBf_fract^2 */ dBf_fract = fMultNorm(dBf_fract, dBf_fract, &tmp); /* Get worst case scaling of multiplication result */ dBf_e = (DFRACT_BITS - 1 - tmp) - CountLeadingBits(dBf_int); /* Now multiply integer with fractional part of the result, thus resulting in the overall accurate fractional result */ dBf_m = fMultNorm(dBf_int, DFRACT_BITS - 1, dBf_fract, tmp, dBf_e); myExpSlot += myExp; } else { dBf_m = (FIXP_DBL)0; dBf_e = 0; } /* Keep the results */ h_sbrFastTransientDetector->dBf_m[i] = dBf_m; h_sbrFastTransientDetector->dBf_e[i] = dBf_e; } /* Make sure that dBf is greater than 1.0 (because it should be a highpass) */ /* ... */ return 0; } void FDKsbrEnc_fastTransientDetect( const HANDLE_FAST_TRAN_DET h_sbrFastTransientDetector, const FIXP_DBL *const *Energies, const int *const scaleEnergies, const INT YBufferWriteOffset, UCHAR *const tran_vector) { int timeSlot, band; FIXP_DBL max_delta_energy; /* helper to store maximum energy ratio */ int max_delta_energy_scale; /* helper to store scale of maximum energy ratio */ int ind_max = 0; /* helper to store index of maximum energy ratio */ int isTransientInFrame = 0; const int nTimeSlots = h_sbrFastTransientDetector->nTimeSlots; const int lookahead = h_sbrFastTransientDetector->lookahead; const int startBand = h_sbrFastTransientDetector->startBand; const int stopBand = h_sbrFastTransientDetector->stopBand; int *transientCandidates = h_sbrFastTransientDetector->transientCandidates; FIXP_DBL *energy_timeSlots = h_sbrFastTransientDetector->energy_timeSlots; int *energy_timeSlots_scale = h_sbrFastTransientDetector->energy_timeSlots_scale; FIXP_DBL *delta_energy = h_sbrFastTransientDetector->delta_energy; int *delta_energy_scale = h_sbrFastTransientDetector->delta_energy_scale; const FIXP_DBL thr = TRAN_DET_THRSHLD; const INT thr_scale = TRAN_DET_THRSHLD_SCALE; /*reset transient info*/ tran_vector[2] = 0; /* reset transient candidates */ FDKmemclear(transientCandidates + lookahead, nTimeSlots * sizeof(int)); for (timeSlot = lookahead; timeSlot < nTimeSlots + lookahead; timeSlot++) { int i, norm; FIXP_DBL tmpE = FL2FXCONST_DBL(0.0f); int headroomEnSlot = DFRACT_BITS - 1; FIXP_DBL smallNRG = FL2FXCONST_DBL(1e-2f); FIXP_DBL denominator; INT denominator_scale; /* determine minimum headroom of energy values for this timeslot */ for (band = startBand; band < stopBand; band++) { int tmp_headroom = fNormz(Energies[timeSlot][band]) - 1; if (tmp_headroom < headroomEnSlot) { headroomEnSlot = tmp_headroom; } } for (i = 0, band = startBand; band < stopBand; band++, i++) { /* energy is weighted by weightingfactor stored in dBf_m array */ /* dBf_m index runs from 0 to stopBand-startband */ /* energy shifted by calculated headroom for maximum precision */ FIXP_DBL weightedEnergy = fMult(Energies[timeSlot][band] << headroomEnSlot, h_sbrFastTransientDetector->dBf_m[i]); /* energy is added up */ /* shift by 6 to have a headroom for maximum 64 additions */ /* shift by dBf_e to handle weighting factor dependent scale factors */ tmpE += weightedEnergy >> (6 + (10 - h_sbrFastTransientDetector->dBf_e[i])); } /* store calculated energy for timeslot */ energy_timeSlots[timeSlot] = tmpE; /* calculate overall scale factor for energy of this timeslot */ /* = original scale factor of energies * (-scaleEnergies[0]+2*QMF_SCALE_OFFSET or * -scaleEnergies[1]+2*QMF_SCALE_OFFSET */ /* depending on YBufferWriteOffset) */ /* + weighting factor scale (10) */ /* + adding up scale factor ( 6) */ /* - headroom of energy value (headroomEnSlot) */ if (timeSlot < YBufferWriteOffset) { energy_timeSlots_scale[timeSlot] = (-scaleEnergies[0] + 2 * QMF_SCALE_OFFSET) + (10 + 6) - headroomEnSlot; } else { energy_timeSlots_scale[timeSlot] = (-scaleEnergies[1] + 2 * QMF_SCALE_OFFSET) + (10 + 6) - headroomEnSlot; } /* Add a small energy to the denominator, thus making the transient detection energy-dependent. Loud transients are being detected, silent ones not. */ /* make sure that smallNRG does not overflow */ if (-energy_timeSlots_scale[timeSlot - 1] + 1 > 5) { denominator = smallNRG; denominator_scale = 0; } else { /* Leave an additional headroom of 1 bit for this addition. */ smallNRG = scaleValue(smallNRG, -(energy_timeSlots_scale[timeSlot - 1] + 1)); denominator = (energy_timeSlots[timeSlot - 1] >> 1) + smallNRG; denominator_scale = energy_timeSlots_scale[timeSlot - 1] + 1; } delta_energy[timeSlot] = fDivNorm(energy_timeSlots[timeSlot], denominator, &norm); delta_energy_scale[timeSlot] = energy_timeSlots_scale[timeSlot] - denominator_scale + norm; } /*get transient candidates*/ /* For every timeslot, check if delta(E) exceeds the threshold. If it did, it could potentially be marked as a transient candidate. However, the 2 slots before the current one must not be transients with an energy higher than 1.4*E(current). If both aren't transients or if the energy of the current timesolot is more than 1.4 times higher than the energy in the last or the one before the last slot, it is marked as a transient.*/ FDK_ASSERT(lookahead >= 2); for (timeSlot = lookahead; timeSlot < nTimeSlots + lookahead; timeSlot++) { FIXP_DBL energy_cur_slot_weighted = fMult(energy_timeSlots[timeSlot], FL2FXCONST_DBL(1.0f / 1.4f)); if (!fIsLessThan(delta_energy[timeSlot], delta_energy_scale[timeSlot], thr, thr_scale) && (((transientCandidates[timeSlot - 2] == 0) && (transientCandidates[timeSlot - 1] == 0)) || !fIsLessThan(energy_cur_slot_weighted, energy_timeSlots_scale[timeSlot], energy_timeSlots[timeSlot - 1], energy_timeSlots_scale[timeSlot - 1]) || !fIsLessThan(energy_cur_slot_weighted, energy_timeSlots_scale[timeSlot], energy_timeSlots[timeSlot - 2], energy_timeSlots_scale[timeSlot - 2]))) { /* in case of strong transients, subsequent * qmf slots might be recognized as transients. */ transientCandidates[timeSlot] = 1; } } /*get transient with max energy*/ max_delta_energy = FL2FXCONST_DBL(0.0f); max_delta_energy_scale = 0; ind_max = 0; isTransientInFrame = 0; for (timeSlot = 0; timeSlot < nTimeSlots; timeSlot++) { int scale = fMax(delta_energy_scale[timeSlot], max_delta_energy_scale); if (transientCandidates[timeSlot] && ((delta_energy[timeSlot] >> (scale - delta_energy_scale[timeSlot])) > (max_delta_energy >> (scale - max_delta_energy_scale)))) { max_delta_energy = delta_energy[timeSlot]; max_delta_energy_scale = scale; ind_max = timeSlot; isTransientInFrame = 1; } } /*from all transient candidates take the one with the biggest energy*/ if (isTransientInFrame) { tran_vector[0] = ind_max; tran_vector[1] = 1; } else { /*reset transient info*/ tran_vector[0] = tran_vector[1] = 0; } /*check for transients in lookahead*/ for (timeSlot = nTimeSlots; timeSlot < nTimeSlots + lookahead; timeSlot++) { if (transientCandidates[timeSlot]) { tran_vector[2] = 1; } } /*update buffers*/ for (timeSlot = 0; timeSlot < lookahead; timeSlot++) { transientCandidates[timeSlot] = transientCandidates[nTimeSlots + timeSlot]; /* fixpoint stuff */ energy_timeSlots[timeSlot] = energy_timeSlots[nTimeSlots + timeSlot]; energy_timeSlots_scale[timeSlot] = energy_timeSlots_scale[nTimeSlots + timeSlot]; delta_energy[timeSlot] = delta_energy[nTimeSlots + timeSlot]; delta_energy_scale[timeSlot] = delta_energy_scale[nTimeSlots + timeSlot]; } } fdk-aac-2.0.2/libSBRenc/src/tran_det.h000066400000000000000000000175321404225310700173520ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /**************************** SBR encoder library ****************************** Author(s): Description: *******************************************************************************/ /*! \file \brief Transient detector prototypes $Revision: 95111 $ */ #ifndef TRAN_DET_H #define TRAN_DET_H #include "sbr_encoder.h" #include "sbr_def.h" typedef struct { FIXP_DBL transients[32 + (32 / 2)]; FIXP_DBL thresholds[64]; FIXP_DBL tran_thr; /* Master threshold for transient signals */ FIXP_DBL split_thr_m; /* Threshold for splitting FIXFIX-frames into 2 env */ INT split_thr_e; /* Scale for splitting threshold */ FIXP_DBL prevLowBandEnergy; /* Energy of low band */ FIXP_DBL prevHighBandEnergy; /* Energy of high band */ INT tran_fc; /* Number of lowband subbands to discard */ INT no_cols; INT no_rows; INT mode; int frameShift; int tran_off; /* Offset for reading energy values. */ } SBR_TRANSIENT_DETECTOR; typedef SBR_TRANSIENT_DETECTOR *HANDLE_SBR_TRANSIENT_DETECTOR; #define TRAN_DET_LOOKAHEAD 2 #define TRAN_DET_START_FREQ 4500 /*start frequency for transient detection*/ #define TRAN_DET_STOP_FREQ 13500 /*stop frequency for transient detection*/ #define TRAN_DET_MIN_QMFBANDS \ 4 /* minimum qmf bands for transient detection \ */ #define QMF_HP_dBd_SLOPE_FIX \ FL2FXCONST_DBL(0.00075275f) /* 0.002266f/10 * log2(10) */ #define TRAN_DET_THRSHLD FL2FXCONST_DBL(5.0f / 8.0f) #define TRAN_DET_THRSHLD_SCALE (3) typedef struct { INT transientCandidates[32 + TRAN_DET_LOOKAHEAD]; INT nTimeSlots; INT lookahead; INT startBand; INT stopBand; FIXP_DBL dBf_m[64]; INT dBf_e[64]; FIXP_DBL energy_timeSlots[32 + TRAN_DET_LOOKAHEAD]; INT energy_timeSlots_scale[32 + TRAN_DET_LOOKAHEAD]; FIXP_DBL delta_energy[32 + TRAN_DET_LOOKAHEAD]; INT delta_energy_scale[32 + TRAN_DET_LOOKAHEAD]; FIXP_DBL lowpass_energy[32 + TRAN_DET_LOOKAHEAD]; INT lowpass_energy_scale[32 + TRAN_DET_LOOKAHEAD]; } FAST_TRAN_DETECTOR; typedef FAST_TRAN_DETECTOR *HANDLE_FAST_TRAN_DET; INT FDKsbrEnc_InitSbrFastTransientDetector( HANDLE_FAST_TRAN_DET h_sbrFastTransientDetector, const INT time_slots_per_frame, const INT bandwidth_qmf_slot, const INT no_qmf_channels, const INT sbr_qmf_1st_band); void FDKsbrEnc_fastTransientDetect( const HANDLE_FAST_TRAN_DET h_sbrFastTransientDetector, const FIXP_DBL *const *Energies, const int *const scaleEnergies, const INT YBufferWriteOffset, UCHAR *const tran_vector); void FDKsbrEnc_transientDetect( HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientDetector, FIXP_DBL **Energies, INT *scaleEnergies, UCHAR *tran_vector, int YBufferWriteOffset, int YBufferSzShift, int timeStep, int frameMiddleBorder); int FDKsbrEnc_InitSbrTransientDetector( HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientDetector, UINT sbrSyntaxFlags, /* SBR syntax flags derived from AOT. */ INT frameSize, INT sampleFreq, sbrConfigurationPtr params, int tran_fc, int no_cols, int no_rows, int YBufferWriteOffset, int YBufferSzShift, int frameShift, int tran_off); void FDKsbrEnc_frameSplitter( FIXP_DBL **Energies, INT *scaleEnergies, HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientDetector, UCHAR *freqBandTable, UCHAR *tran_vector, int YBufferWriteOffset, int YBufferSzShift, int nSfb, int timeStep, int no_cols, FIXP_DBL *tonality); #endif fdk-aac-2.0.2/libSYS/000077500000000000000000000000001404225310700141445ustar00rootroot00000000000000fdk-aac-2.0.2/libSYS/include/000077500000000000000000000000001404225310700155675ustar00rootroot00000000000000fdk-aac-2.0.2/libSYS/include/FDK_audio.h000066400000000000000000001021351404225310700175270ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /************************* System integration library ************************** Author(s): Manuel Jander Description: *******************************************************************************/ /** \file FDK_audio.h * \brief Global audio struct and constant definitions. */ #ifndef FDK_AUDIO_H #define FDK_AUDIO_H #include "machine_type.h" #include "genericStds.h" #include "syslib_channelMapDescr.h" #ifdef __cplusplus extern "C" { #endif /** * File format identifiers. */ typedef enum { FF_UNKNOWN = -1, /**< Unknown format. */ FF_RAW = 0, /**< No container, bit stream data conveyed "as is". */ FF_MP4_3GPP = 3, /**< 3GPP file format. */ FF_MP4_MP4F = 4, /**< MPEG-4 File format. */ FF_RAWPACKETS = 5 /**< Proprietary raw packet file. */ } FILE_FORMAT; /** * Transport type identifiers. */ typedef enum { TT_UNKNOWN = -1, /**< Unknown format. */ TT_MP4_RAW = 0, /**< "as is" access units (packet based since there is obviously no sync layer) */ TT_MP4_ADIF = 1, /**< ADIF bitstream format. */ TT_MP4_ADTS = 2, /**< ADTS bitstream format. */ TT_MP4_LATM_MCP1 = 6, /**< Audio Mux Elements with muxConfigPresent = 1 */ TT_MP4_LATM_MCP0 = 7, /**< Audio Mux Elements with muxConfigPresent = 0, out of band StreamMuxConfig */ TT_MP4_LOAS = 10, /**< Audio Sync Stream. */ TT_DRM = 12 /**< Digital Radio Mondial (DRM30/DRM+) bitstream format. */ } TRANSPORT_TYPE; #define TT_IS_PACKET(x) \ (((x) == TT_MP4_RAW) || ((x) == TT_DRM) || ((x) == TT_MP4_LATM_MCP0) || \ ((x) == TT_MP4_LATM_MCP1)) /** * Audio Object Type definitions. */ typedef enum { AOT_NONE = -1, AOT_NULL_OBJECT = 0, AOT_AAC_MAIN = 1, /**< Main profile */ AOT_AAC_LC = 2, /**< Low Complexity object */ AOT_AAC_SSR = 3, AOT_AAC_LTP = 4, AOT_SBR = 5, AOT_AAC_SCAL = 6, AOT_TWIN_VQ = 7, AOT_CELP = 8, AOT_HVXC = 9, AOT_RSVD_10 = 10, /**< (reserved) */ AOT_RSVD_11 = 11, /**< (reserved) */ AOT_TTSI = 12, /**< TTSI Object */ AOT_MAIN_SYNTH = 13, /**< Main Synthetic object */ AOT_WAV_TAB_SYNTH = 14, /**< Wavetable Synthesis object */ AOT_GEN_MIDI = 15, /**< General MIDI object */ AOT_ALG_SYNTH_AUD_FX = 16, /**< Algorithmic Synthesis and Audio FX object */ AOT_ER_AAC_LC = 17, /**< Error Resilient(ER) AAC Low Complexity */ AOT_RSVD_18 = 18, /**< (reserved) */ AOT_ER_AAC_LTP = 19, /**< Error Resilient(ER) AAC LTP object */ AOT_ER_AAC_SCAL = 20, /**< Error Resilient(ER) AAC Scalable object */ AOT_ER_TWIN_VQ = 21, /**< Error Resilient(ER) TwinVQ object */ AOT_ER_BSAC = 22, /**< Error Resilient(ER) BSAC object */ AOT_ER_AAC_LD = 23, /**< Error Resilient(ER) AAC LowDelay object */ AOT_ER_CELP = 24, /**< Error Resilient(ER) CELP object */ AOT_ER_HVXC = 25, /**< Error Resilient(ER) HVXC object */ AOT_ER_HILN = 26, /**< Error Resilient(ER) HILN object */ AOT_ER_PARA = 27, /**< Error Resilient(ER) Parametric object */ AOT_RSVD_28 = 28, /**< might become SSC */ AOT_PS = 29, /**< PS, Parametric Stereo (includes SBR) */ AOT_MPEGS = 30, /**< MPEG Surround */ AOT_ESCAPE = 31, /**< Signal AOT uses more than 5 bits */ AOT_MP3ONMP4_L1 = 32, /**< MPEG-Layer1 in mp4 */ AOT_MP3ONMP4_L2 = 33, /**< MPEG-Layer2 in mp4 */ AOT_MP3ONMP4_L3 = 34, /**< MPEG-Layer3 in mp4 */ AOT_RSVD_35 = 35, /**< might become DST */ AOT_RSVD_36 = 36, /**< might become ALS */ AOT_AAC_SLS = 37, /**< AAC + SLS */ AOT_SLS = 38, /**< SLS */ AOT_ER_AAC_ELD = 39, /**< AAC Enhanced Low Delay */ AOT_USAC = 42, /**< USAC */ AOT_SAOC = 43, /**< SAOC */ AOT_LD_MPEGS = 44, /**< Low Delay MPEG Surround */ /* Pseudo AOTs */ AOT_MP2_AAC_LC = 129, /**< Virtual AOT MP2 Low Complexity profile */ AOT_MP2_SBR = 132, /**< Virtual AOT MP2 Low Complexity Profile with SBR */ AOT_DRM_AAC = 143, /**< Virtual AOT for DRM (ER-AAC-SCAL without SBR) */ AOT_DRM_SBR = 144, /**< Virtual AOT for DRM (ER-AAC-SCAL with SBR) */ AOT_DRM_MPEG_PS = 145, /**< Virtual AOT for DRM (ER-AAC-SCAL with SBR and MPEG-PS) */ AOT_DRM_SURROUND = 146, /**< Virtual AOT for DRM Surround (ER-AAC-SCAL (+SBR) +MPS) */ AOT_DRM_USAC = 147 /**< Virtual AOT for DRM with USAC */ } AUDIO_OBJECT_TYPE; #define CAN_DO_PS(aot) \ ((aot) == AOT_AAC_LC || (aot) == AOT_SBR || (aot) == AOT_PS || \ (aot) == AOT_ER_BSAC || (aot) == AOT_DRM_AAC) #define IS_USAC(aot) ((aot) == AOT_USAC) #define IS_LOWDELAY(aot) ((aot) == AOT_ER_AAC_LD || (aot) == AOT_ER_AAC_ELD) /** Channel Mode ( 1-7 equals MPEG channel configurations, others are * arbitrary). */ typedef enum { MODE_INVALID = -1, MODE_UNKNOWN = 0, MODE_1 = 1, /**< C */ MODE_2 = 2, /**< L+R */ MODE_1_2 = 3, /**< C, L+R */ MODE_1_2_1 = 4, /**< C, L+R, Rear */ MODE_1_2_2 = 5, /**< C, L+R, LS+RS */ MODE_1_2_2_1 = 6, /**< C, L+R, LS+RS, LFE */ MODE_1_2_2_2_1 = 7, /**< C, LC+RC, L+R, LS+RS, LFE */ MODE_6_1 = 11, /**< C, L+R, LS+RS, Crear, LFE */ MODE_7_1_BACK = 12, /**< C, L+R, LS+RS, Lrear+Rrear, LFE */ MODE_7_1_TOP_FRONT = 14, /**< C, L+R, LS+RS, LFE, Ltop+Rtop */ MODE_7_1_REAR_SURROUND = 33, /**< C, L+R, LS+RS, Lrear+Rrear, LFE */ MODE_7_1_FRONT_CENTER = 34, /**< C, LC+RC, L+R, LS+RS, LFE */ MODE_212 = 128 /**< 212 configuration, used in ELDv2 */ } CHANNEL_MODE; /** * Speaker description tags. * Do not change the enumeration values unless it keeps the following * segmentation: * - Bit 0-3: Horizontal postion (0: none, 1: front, 2: side, 3: back, 4: lfe) * - Bit 4-7: Vertical position (0: normal, 1: top, 2: bottom) */ typedef enum { ACT_NONE = 0x00, ACT_FRONT = 0x01, /*!< Front speaker position (at normal height) */ ACT_SIDE = 0x02, /*!< Side speaker position (at normal height) */ ACT_BACK = 0x03, /*!< Back speaker position (at normal height) */ ACT_LFE = 0x04, /*!< Low frequency effect speaker postion (front) */ ACT_TOP = 0x10, /*!< Top speaker area (for combination with speaker positions) */ ACT_FRONT_TOP = 0x11, /*!< Top front speaker = (ACT_FRONT|ACT_TOP) */ ACT_SIDE_TOP = 0x12, /*!< Top side speaker = (ACT_SIDE |ACT_TOP) */ ACT_BACK_TOP = 0x13, /*!< Top back speaker = (ACT_BACK |ACT_TOP) */ ACT_BOTTOM = 0x20, /*!< Bottom speaker area (for combination with speaker positions) */ ACT_FRONT_BOTTOM = 0x21, /*!< Bottom front speaker = (ACT_FRONT|ACT_BOTTOM) */ ACT_SIDE_BOTTOM = 0x22, /*!< Bottom side speaker = (ACT_SIDE |ACT_BOTTOM) */ ACT_BACK_BOTTOM = 0x23 /*!< Bottom back speaker = (ACT_BACK |ACT_BOTTOM) */ } AUDIO_CHANNEL_TYPE; typedef enum { SIG_UNKNOWN = -1, SIG_IMPLICIT = 0, SIG_EXPLICIT_BW_COMPATIBLE = 1, SIG_EXPLICIT_HIERARCHICAL = 2 } SBR_PS_SIGNALING; /** * Audio Codec flags. */ #define AC_ER_VCB11 \ 0x000001 /*!< aacSectionDataResilienceFlag flag (from ASC): 1 means use \ virtual codebooks */ #define AC_ER_RVLC \ 0x000002 /*!< aacSpectralDataResilienceFlag flag (from ASC): 1 means use \ huffman codeword reordering */ #define AC_ER_HCR \ 0x000004 /*!< aacSectionDataResilienceFlag flag (from ASC): 1 means use \ virtual codebooks */ #define AC_SCALABLE 0x000008 /*!< AAC Scalable*/ #define AC_ELD 0x000010 /*!< AAC-ELD */ #define AC_LD 0x000020 /*!< AAC-LD */ #define AC_ER 0x000040 /*!< ER syntax */ #define AC_BSAC 0x000080 /*!< BSAC */ #define AC_USAC 0x000100 /*!< USAC */ #define AC_RSV603DA 0x000200 /*!< RSVD60 3D audio */ #define AC_HDAAC 0x000400 /*!< HD-AAC */ #define AC_RSVD50 0x004000 /*!< Rsvd50 */ #define AC_SBR_PRESENT 0x008000 /*!< SBR present flag (from ASC) */ #define AC_SBRCRC \ 0x010000 /*!< SBR CRC present flag. Only relevant for AAC-ELD for now. */ #define AC_PS_PRESENT 0x020000 /*!< PS present flag (from ASC or implicit) */ #define AC_MPS_PRESENT \ 0x040000 /*!< MPS present flag (from ASC or implicit) \ */ #define AC_DRM 0x080000 /*!< DRM bit stream syntax */ #define AC_INDEP 0x100000 /*!< Independency flag */ #define AC_MPEGD_RES 0x200000 /*!< MPEG-D residual individual channel data. */ #define AC_SAOC_PRESENT 0x400000 /*!< SAOC Present Flag */ #define AC_DAB 0x800000 /*!< DAB bit stream syntax */ #define AC_ELD_DOWNSCALE 0x1000000 /*!< ELD Downscaled playout */ #define AC_LD_MPS 0x2000000 /*!< Low Delay MPS. */ #define AC_DRC_PRESENT \ 0x4000000 /*!< Dynamic Range Control (DRC) data found. \ */ #define AC_USAC_SCFGI3 \ 0x8000000 /*!< USAC flag: If stereoConfigIndex is 3 the flag is set. */ /** * Audio Codec flags (reconfiguration). */ #define AC_CM_DET_CFG_CHANGE \ 0x000001 /*!< Config mode signalizes the callback to work in config change \ detection mode */ #define AC_CM_ALLOC_MEM \ 0x000002 /*!< Config mode signalizes the callback to work in memory \ allocation mode */ /** * Audio Codec flags (element specific). */ #define AC_EL_USAC_TW 0x000001 /*!< USAC time warped filter bank is active */ #define AC_EL_USAC_NOISE 0x000002 /*!< USAC noise filling is active */ #define AC_EL_USAC_ITES 0x000004 /*!< USAC SBR inter-TES tool is active */ #define AC_EL_USAC_PVC \ 0x000008 /*!< USAC SBR predictive vector coding tool is active */ #define AC_EL_USAC_MPS212 0x000010 /*!< USAC MPS212 tool is active */ #define AC_EL_USAC_LFE 0x000020 /*!< USAC element is LFE */ #define AC_EL_USAC_CP_POSSIBLE \ 0x000040 /*!< USAC may use Complex Stereo Prediction in this channel element \ */ #define AC_EL_ENHANCED_NOISE 0x000080 /*!< Enhanced noise filling*/ #define AC_EL_IGF_AFTER_TNS 0x000100 /*!< IGF after TNS */ #define AC_EL_IGF_INDEP_TILING 0x000200 /*!< IGF independent tiling */ #define AC_EL_IGF_USE_ENF 0x000400 /*!< IGF use enhanced noise filling */ #define AC_EL_FULLBANDLPD 0x000800 /*!< enable fullband LPD tools */ #define AC_EL_LPDSTEREOIDX 0x001000 /*!< LPD-stereo-tool stereo index */ #define AC_EL_LFE 0x002000 /*!< The element is of type LFE. */ /* CODER_CONFIG::flags */ #define CC_MPEG_ID 0x00100000 #define CC_IS_BASELAYER 0x00200000 #define CC_PROTECTION 0x00400000 #define CC_SBR 0x00800000 #define CC_SBRCRC 0x00010000 #define CC_SAC 0x00020000 #define CC_RVLC 0x01000000 #define CC_VCB11 0x02000000 #define CC_HCR 0x04000000 #define CC_PSEUDO_SURROUND 0x08000000 #define CC_USAC_NOISE 0x10000000 #define CC_USAC_TW 0x20000000 #define CC_USAC_HBE 0x40000000 /** Generic audio coder configuration structure. */ typedef struct { AUDIO_OBJECT_TYPE aot; /**< Audio Object Type (AOT). */ AUDIO_OBJECT_TYPE extAOT; /**< Extension Audio Object Type (SBR). */ CHANNEL_MODE channelMode; /**< Channel mode. */ UCHAR channelConfigZero; /**< Use channel config zero + pce although a standard channel config could be signaled. */ INT samplingRate; /**< Sampling rate. */ INT extSamplingRate; /**< Extended samplerate (SBR). */ INT downscaleSamplingRate; /**< Downscale sampling rate (ELD downscaled mode) */ INT bitRate; /**< Average bitrate. */ int samplesPerFrame; /**< Number of PCM samples per codec frame and audio channel. */ int noChannels; /**< Number of audio channels. */ int bitsFrame; int nSubFrames; /**< Amount of encoder subframes. 1 means no subframing. */ int BSACnumOfSubFrame; /**< The number of the sub-frames which are grouped and transmitted in a super-frame (BSAC). */ int BSAClayerLength; /**< The average length of the large-step layers in bytes (BSAC). */ UINT flags; /**< flags */ UCHAR matrixMixdownA; /**< Matrix mixdown index to put into PCE. Default value 0 means no mixdown coefficient, valid values are 1-4 which correspond to matrix_mixdown_idx 0-3. */ UCHAR headerPeriod; /**< Frame period for sending in band configuration buffers in the transport layer. */ UCHAR stereoConfigIndex; /**< USAC MPS stereo mode */ UCHAR sbrMode; /**< USAC SBR mode */ SBR_PS_SIGNALING sbrSignaling; /**< 0: implicit signaling, 1: backwards compatible explicit signaling, 2: hierarcical explicit signaling */ UCHAR rawConfig[64]; /**< raw codec specific config as bit stream */ int rawConfigBits; /**< Size of rawConfig in bits */ UCHAR sbrPresent; UCHAR psPresent; } CODER_CONFIG; #define USAC_ID_BIT 16 /** USAC element IDs start at USAC_ID_BIT */ /** MP4 Element IDs. */ typedef enum { /* mp4 element IDs */ ID_NONE = -1, /**< Invalid Element helper ID. */ ID_SCE = 0, /**< Single Channel Element. */ ID_CPE = 1, /**< Channel Pair Element. */ ID_CCE = 2, /**< Coupling Channel Element. */ ID_LFE = 3, /**< LFE Channel Element. */ ID_DSE = 4, /**< Currently one Data Stream Element for ancillary data is supported. */ ID_PCE = 5, /**< Program Config Element. */ ID_FIL = 6, /**< Fill Element. */ ID_END = 7, /**< Arnie (End Element = Terminator). */ ID_EXT = 8, /**< Extension Payload (ER only). */ ID_SCAL = 9, /**< AAC scalable element (ER only). */ /* USAC element IDs */ ID_USAC_SCE = 0 + USAC_ID_BIT, /**< Single Channel Element. */ ID_USAC_CPE = 1 + USAC_ID_BIT, /**< Channel Pair Element. */ ID_USAC_LFE = 2 + USAC_ID_BIT, /**< LFE Channel Element. */ ID_USAC_EXT = 3 + USAC_ID_BIT, /**< Extension Element. */ ID_USAC_END = 4 + USAC_ID_BIT, /**< Arnie (End Element = Terminator). */ ID_LAST } MP4_ELEMENT_ID; /* usacConfigExtType q.v. ISO/IEC DIS 23008-3 Table 52 and ISO/IEC FDIS * 23003-3:2011(E) Table 74*/ typedef enum { /* USAC and RSVD60 3DA */ ID_CONFIG_EXT_FILL = 0, /* RSVD60 3DA */ ID_CONFIG_EXT_DOWNMIX = 1, ID_CONFIG_EXT_LOUDNESS_INFO = 2, ID_CONFIG_EXT_AUDIOSCENE_INFO = 3, ID_CONFIG_EXT_HOA_MATRIX = 4, ID_CONFIG_EXT_SIG_GROUP_INFO = 6 /* 5-127 => reserved for ISO use */ /* > 128 => reserved for use outside of ISO scope */ } CONFIG_EXT_ID; #define IS_CHANNEL_ELEMENT(elementId) \ ((elementId) == ID_SCE || (elementId) == ID_CPE || (elementId) == ID_LFE || \ (elementId) == ID_USAC_SCE || (elementId) == ID_USAC_CPE || \ (elementId) == ID_USAC_LFE) #define IS_MP4_CHANNEL_ELEMENT(elementId) \ ((elementId) == ID_SCE || (elementId) == ID_CPE || (elementId) == ID_LFE) #define EXT_ID_BITS 4 /**< Size in bits of extension payload type tags. */ /** Extension payload types. */ typedef enum { EXT_FIL = 0x00, EXT_FILL_DATA = 0x01, EXT_DATA_ELEMENT = 0x02, EXT_DATA_LENGTH = 0x03, EXT_UNI_DRC = 0x04, EXT_LDSAC_DATA = 0x09, EXT_SAOC_DATA = 0x0a, EXT_DYNAMIC_RANGE = 0x0b, EXT_SAC_DATA = 0x0c, EXT_SBR_DATA = 0x0d, EXT_SBR_DATA_CRC = 0x0e } EXT_PAYLOAD_TYPE; #define IS_USAC_CHANNEL_ELEMENT(elementId) \ ((elementId) == ID_USAC_SCE || (elementId) == ID_USAC_CPE || \ (elementId) == ID_USAC_LFE) /** MPEG-D USAC & RSVD60 3D audio Extension Element Types. */ typedef enum { /* usac */ ID_EXT_ELE_FILL = 0x00, ID_EXT_ELE_MPEGS = 0x01, ID_EXT_ELE_SAOC = 0x02, ID_EXT_ELE_AUDIOPREROLL = 0x03, ID_EXT_ELE_UNI_DRC = 0x04, /* rsv603da */ ID_EXT_ELE_OBJ_METADATA = 0x05, ID_EXT_ELE_SAOC_3D = 0x06, ID_EXT_ELE_HOA = 0x07, ID_EXT_ELE_FMT_CNVRTR = 0x08, ID_EXT_ELE_MCT = 0x09, ID_EXT_ELE_ENHANCED_OBJ_METADATA = 0x0d, /* reserved for use outside of ISO scope */ ID_EXT_ELE_VR_METADATA = 0x81, ID_EXT_ELE_UNKNOWN = 0xFF } USAC_EXT_ELEMENT_TYPE; /** * Proprietary raw packet file configuration data type identifier. */ typedef enum { TC_NOTHING = 0, /* No configuration available -> in-band configuration. */ TC_RAW_ADTS = 2, /* Transfer type is ADTS. */ TC_RAW_LATM_MCP1 = 6, /* Transfer type is LATM with SMC present. */ TC_RAW_SDC = 21 /* Configuration data field is Drm SDC. */ } TP_CONFIG_TYPE; /* * ############################################################################################## * Library identification and error handling * ############################################################################################## */ /* \cond */ typedef enum { FDK_NONE = 0, FDK_TOOLS = 1, FDK_SYSLIB = 2, FDK_AACDEC = 3, FDK_AACENC = 4, FDK_SBRDEC = 5, FDK_SBRENC = 6, FDK_TPDEC = 7, FDK_TPENC = 8, FDK_MPSDEC = 9, FDK_MPEGFILEREAD = 10, FDK_MPEGFILEWRITE = 11, FDK_PCMDMX = 31, FDK_MPSENC = 34, FDK_TDLIMIT = 35, FDK_UNIDRCDEC = 38, FDK_MODULE_LAST } FDK_MODULE_ID; /* AAC capability flags */ #define CAPF_AAC_LC 0x00000001 /**< Support flag for AAC Low Complexity. */ #define CAPF_ER_AAC_LD \ 0x00000002 /**< Support flag for AAC Low Delay with Error Resilience tools. \ */ #define CAPF_ER_AAC_SCAL 0x00000004 /**< Support flag for AAC Scalable. */ #define CAPF_ER_AAC_LC \ 0x00000008 /**< Support flag for AAC Low Complexity with Error Resilience \ tools. */ #define CAPF_AAC_480 \ 0x00000010 /**< Support flag for AAC with 480 framelength. */ #define CAPF_AAC_512 \ 0x00000020 /**< Support flag for AAC with 512 framelength. */ #define CAPF_AAC_960 \ 0x00000040 /**< Support flag for AAC with 960 framelength. */ #define CAPF_AAC_1024 \ 0x00000080 /**< Support flag for AAC with 1024 framelength. */ #define CAPF_AAC_HCR \ 0x00000100 /**< Support flag for AAC with Huffman Codeword Reordering. */ #define CAPF_AAC_VCB11 \ 0x00000200 /**< Support flag for AAC Virtual Codebook 11. */ #define CAPF_AAC_RVLC \ 0x00000400 /**< Support flag for AAC Reversible Variable Length Coding. */ #define CAPF_AAC_MPEG4 0x00000800 /**< Support flag for MPEG file format. */ #define CAPF_AAC_DRC \ 0x00001000 /**< Support flag for AAC Dynamic Range Control. */ #define CAPF_AAC_CONCEALMENT \ 0x00002000 /**< Support flag for AAC concealment. */ #define CAPF_AAC_DRM_BSFORMAT \ 0x00004000 /**< Support flag for AAC DRM bistream format. */ #define CAPF_ER_AAC_ELD \ 0x00008000 /**< Support flag for AAC Enhanced Low Delay with Error \ Resilience tools. */ #define CAPF_ER_AAC_BSAC \ 0x00010000 /**< Support flag for AAC BSAC. */ #define CAPF_AAC_ELD_DOWNSCALE \ 0x00040000 /**< Support flag for AAC-ELD Downscaling */ #define CAPF_AAC_USAC_LP \ 0x00100000 /**< Support flag for USAC low power mode. */ #define CAPF_AAC_USAC \ 0x00200000 /**< Support flag for Unified Speech and Audio Coding (USAC). */ #define CAPF_ER_AAC_ELDV2 \ 0x00800000 /**< Support flag for AAC Enhanced Low Delay with MPS 212. */ #define CAPF_AAC_UNIDRC \ 0x01000000 /**< Support flag for MPEG-D Dynamic Range Control (uniDrc). */ /* Transport capability flags */ #define CAPF_ADTS \ 0x00000001 /**< Support flag for ADTS transport format. */ #define CAPF_ADIF \ 0x00000002 /**< Support flag for ADIF transport format. */ #define CAPF_LATM \ 0x00000004 /**< Support flag for LATM transport format. */ #define CAPF_LOAS \ 0x00000008 /**< Support flag for LOAS transport format. */ #define CAPF_RAWPACKETS \ 0x00000010 /**< Support flag for RAW PACKETS transport format. */ #define CAPF_DRM \ 0x00000020 /**< Support flag for DRM/DRM+ transport format. */ #define CAPF_RSVD50 \ 0x00000040 /**< Support flag for RSVD50 transport format */ /* SBR capability flags */ #define CAPF_SBR_LP \ 0x00000001 /**< Support flag for SBR Low Power mode. */ #define CAPF_SBR_HQ \ 0x00000002 /**< Support flag for SBR High Quality mode. */ #define CAPF_SBR_DRM_BS \ 0x00000004 /**< Support flag for */ #define CAPF_SBR_CONCEALMENT \ 0x00000008 /**< Support flag for SBR concealment. */ #define CAPF_SBR_DRC \ 0x00000010 /**< Support flag for SBR Dynamic Range Control. */ #define CAPF_SBR_PS_MPEG \ 0x00000020 /**< Support flag for MPEG Parametric Stereo. */ #define CAPF_SBR_PS_DRM \ 0x00000040 /**< Support flag for DRM Parametric Stereo. */ #define CAPF_SBR_ELD_DOWNSCALE \ 0x00000080 /**< Support flag for ELD reduced delay mode */ #define CAPF_SBR_HBEHQ \ 0x00000100 /**< Support flag for HQ HBE */ /* PCM utils capability flags */ #define CAPF_DMX_BLIND \ 0x00000001 /**< Support flag for blind downmixing. */ #define CAPF_DMX_PCE \ 0x00000002 /**< Support flag for guided downmix with data from MPEG-2/4 \ Program Config Elements (PCE). */ #define CAPF_DMX_ARIB \ 0x00000004 /**< Support flag for PCE guided downmix with slightly different \ equations and levels to fulfill ARIB standard. */ #define CAPF_DMX_DVB \ 0x00000008 /**< Support flag for guided downmix with data from DVB ancillary \ data fields. */ #define CAPF_DMX_CH_EXP \ 0x00000010 /**< Support flag for simple upmixing by dublicating channels or \ adding zero channels. */ #define CAPF_DMX_6_CH \ 0x00000020 /**< Support flag for 5.1 channel configuration (input and \ output). */ #define CAPF_DMX_8_CH \ 0x00000040 /**< Support flag for 6 and 7.1 channel configurations (input and \ output). */ #define CAPF_DMX_24_CH \ 0x00000080 /**< Support flag for 22.2 channel configuration (input and \ output). */ #define CAPF_LIMITER \ 0x00002000 /**< Support flag for signal level limiting. \ */ /* MPEG Surround capability flags */ #define CAPF_MPS_STD \ 0x00000001 /**< Support flag for MPEG Surround. */ #define CAPF_MPS_LD \ 0x00000002 /**< Support flag for Low Delay MPEG Surround. \ */ #define CAPF_MPS_USAC \ 0x00000004 /**< Support flag for USAC MPEG Surround. */ #define CAPF_MPS_HQ \ 0x00000010 /**< Support flag indicating if high quality processing is \ supported */ #define CAPF_MPS_LP \ 0x00000020 /**< Support flag indicating if partially complex (low power) \ processing is supported */ #define CAPF_MPS_BLIND \ 0x00000040 /**< Support flag indicating if blind processing is supported */ #define CAPF_MPS_BINAURAL \ 0x00000080 /**< Support flag indicating if binaural output is possible */ #define CAPF_MPS_2CH_OUT \ 0x00000100 /**< Support flag indicating if 2ch output is possible */ #define CAPF_MPS_6CH_OUT \ 0x00000200 /**< Support flag indicating if 6ch output is possible */ #define CAPF_MPS_8CH_OUT \ 0x00000400 /**< Support flag indicating if 8ch output is possible */ #define CAPF_MPS_1CH_IN \ 0x00001000 /**< Support flag indicating if 1ch dmx input is possible */ #define CAPF_MPS_2CH_IN \ 0x00002000 /**< Support flag indicating if 2ch dmx input is possible */ #define CAPF_MPS_6CH_IN \ 0x00004000 /**< Support flag indicating if 5ch dmx input is possible */ /* \endcond */ /* * ############################################################################################## * Library versioning * ############################################################################################## */ /** * Convert each member of version numbers to one single numeric version * representation. * \param lev0 1st level of version number. * \param lev1 2nd level of version number. * \param lev2 3rd level of version number. */ #define LIB_VERSION(lev0, lev1, lev2) \ ((lev0 << 24 & 0xff000000) | (lev1 << 16 & 0x00ff0000) | \ (lev2 << 8 & 0x0000ff00)) /** * Build text string of version. */ #define LIB_VERSION_STRING(info) \ FDKsprintf((info)->versionStr, "%d.%d.%d", (((info)->version >> 24) & 0xff), \ (((info)->version >> 16) & 0xff), \ (((info)->version >> 8) & 0xff)) /** * Library information. */ typedef struct LIB_INFO { const char* title; const char* build_date; const char* build_time; FDK_MODULE_ID module_id; INT version; UINT flags; char versionStr[32]; } LIB_INFO; #ifdef __cplusplus #define FDK_AUDIO_INLINE inline #else #define FDK_AUDIO_INLINE #endif /** Initialize library info. */ static FDK_AUDIO_INLINE void FDKinitLibInfo(LIB_INFO* info) { int i; for (i = 0; i < FDK_MODULE_LAST; i++) { info[i].module_id = FDK_NONE; } } /** Aquire supported features of library. */ static FDK_AUDIO_INLINE UINT FDKlibInfo_getCapabilities(const LIB_INFO* info, FDK_MODULE_ID module_id) { int i; for (i = 0; i < FDK_MODULE_LAST; i++) { if (info[i].module_id == module_id) { return info[i].flags; } } return 0; } /** Search for next free tab. */ static FDK_AUDIO_INLINE INT FDKlibInfo_lookup(const LIB_INFO* info, FDK_MODULE_ID module_id) { int i = -1; for (i = 0; i < FDK_MODULE_LAST; i++) { if (info[i].module_id == module_id) return -1; if (info[i].module_id == FDK_NONE) break; } if (i == FDK_MODULE_LAST) return -1; return i; } /* * ############################################################################################## * Buffer description * ############################################################################################## */ /** * I/O buffer descriptor. */ typedef struct FDK_bufDescr { void** ppBase; /*!< Pointer to an array containing buffer base addresses. Set to NULL for buffer requirement info. */ UINT* pBufSize; /*!< Pointer to an array containing the number of elements that can be placed in the specific buffer. */ UINT* pEleSize; /*!< Pointer to an array containing the element size for each buffer in bytes. That is mostly the number returned by the sizeof() operator for the data type used for the specific buffer. */ UINT* pBufType; /*!< Pointer to an array of bit fields containing a description for each buffer. See XXX below for more details. */ UINT numBufs; /*!< Total number of buffers. */ } FDK_bufDescr; /** * Buffer type description field. */ #define FDK_BUF_TYPE_MASK_IO ((UINT)0x03 << 30) #define FDK_BUF_TYPE_MASK_DESCR ((UINT)0x3F << 16) #define FDK_BUF_TYPE_MASK_ID ((UINT)0xFF) #define FDK_BUF_TYPE_INPUT ((UINT)0x1 << 30) #define FDK_BUF_TYPE_OUTPUT ((UINT)0x2 << 30) #define FDK_BUF_TYPE_PCM_DATA ((UINT)0x1 << 16) #define FDK_BUF_TYPE_ANC_DATA ((UINT)0x2 << 16) #define FDK_BUF_TYPE_BS_DATA ((UINT)0x4 << 16) #ifdef __cplusplus } #endif #endif /* FDK_AUDIO_H */ fdk-aac-2.0.2/libSYS/include/genericStds.h000066400000000000000000000615141404225310700202210ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /************************* System integration library ************************** Author(s): Description: *******************************************************************************/ /** \file genericStds.h \brief Generic Run-Time Support function wrappers and heap allocation monitoring. */ #if !defined(GENERICSTDS_H) #define GENERICSTDS_H #include "machine_type.h" #ifndef M_PI #define M_PI 3.14159265358979323846 /*!< Pi. Only used in example projects. */ #endif /** * Identifiers for various memory locations. They are used along with memory * allocation functions like FDKcalloc_L() to specify the requested memory's * location. */ typedef enum { /* Internal */ SECT_DATA_L1 = 0x2000, SECT_DATA_L2, SECT_DATA_L1_A, SECT_DATA_L1_B, SECT_CONSTDATA_L1, /* External */ SECT_DATA_EXTERN = 0x4000, SECT_CONSTDATA_EXTERN } MEMORY_SECTION; /*! \addtogroup SYSLIB_MEMORY_MACROS FDK memory macros * * The \c H_ prefix indicates that the macro is to be used in a header file, the * \c C_ prefix indicates that the macro is to be used in a source file. * * Declaring memory areas requires to specify a unique name and a data type. * * For defining a memory area you require additionally one or two sizes, * depending if the memory should be organized into one or two dimensions. * * The macros containing the keyword \c AALLOC instead of \c ALLOC additionally * take care of returning aligned memory addresses (beyond the natural alignment * of its type). The preprocesor macro * ::ALIGNMENT_DEFAULT indicates the aligment to be used (this is hardware * specific). * * The \c _L suffix indicates that the memory will be located in a specific * section. This is useful to allocate critical memory section into fast * internal SRAM for example. * * @{ */ /** See \ref SYSLIB_MEMORY_MACROS for description. */ #define H_ALLOC_MEM(name, type) \ type *Get##name(int n = 0); \ void Free##name(type **p); \ UINT GetRequiredMem##name(void); /** See \ref SYSLIB_MEMORY_MACROS for description. */ #define H_ALLOC_MEM_OVERLAY(name, type) \ type *Get##name(int n = 0); \ void Free##name(type **p); \ UINT GetRequiredMem##name(void); /** See \ref SYSLIB_MEMORY_MACROS for description. */ #define C_ALLOC_MEM(name, type, num) \ type *Get##name(int n) { \ FDK_ASSERT((n) == 0); \ return ((type *)FDKcalloc(num, sizeof(type))); \ } \ void Free##name(type **p) { \ if (p != NULL) { \ FDKfree(*p); \ *p = NULL; \ } \ } \ UINT GetRequiredMem##name(void) { \ return ALGN_SIZE_EXTRES((num) * sizeof(type)); \ } /** See \ref SYSLIB_MEMORY_MACROS for description. */ #define C_ALLOC_MEM2(name, type, n1, n2) \ type *Get##name(int n) { \ FDK_ASSERT((n) < (n2)); \ return ((type *)FDKcalloc(n1, sizeof(type))); \ } \ void Free##name(type **p) { \ if (p != NULL) { \ FDKfree(*p); \ *p = NULL; \ } \ } \ UINT GetRequiredMem##name(void) { \ return ALGN_SIZE_EXTRES((n1) * sizeof(type)) * (n2); \ } /** See \ref SYSLIB_MEMORY_MACROS for description. */ #define C_AALLOC_MEM(name, type, num) \ type *Get##name(int n) { \ type *ap; \ FDK_ASSERT((n) == 0); \ ap = ((type *)FDKaalloc((num) * sizeof(type), ALIGNMENT_DEFAULT)); \ return ap; \ } \ void Free##name(type **p) { \ if (p != NULL) { \ FDKafree(*p); \ *p = NULL; \ } \ } \ UINT GetRequiredMem##name(void) { \ return ALGN_SIZE_EXTRES((num) * sizeof(type) + ALIGNMENT_DEFAULT + \ sizeof(void *)); \ } /** See \ref SYSLIB_MEMORY_MACROS for description. */ #define C_AALLOC_MEM2(name, type, n1, n2) \ type *Get##name(int n) { \ type *ap; \ FDK_ASSERT((n) < (n2)); \ ap = ((type *)FDKaalloc((n1) * sizeof(type), ALIGNMENT_DEFAULT)); \ return ap; \ } \ void Free##name(type **p) { \ if (p != NULL) { \ FDKafree(*p); \ *p = NULL; \ } \ } \ UINT GetRequiredMem##name(void) { \ return ALGN_SIZE_EXTRES((n1) * sizeof(type) + ALIGNMENT_DEFAULT + \ sizeof(void *)) * \ (n2); \ } /** See \ref SYSLIB_MEMORY_MACROS for description. */ #define C_ALLOC_MEM_L(name, type, num, s) \ type *Get##name(int n) { \ FDK_ASSERT((n) == 0); \ return ((type *)FDKcalloc_L(num, sizeof(type), s)); \ } \ void Free##name(type **p) { \ if (p != NULL) { \ FDKfree_L(*p); \ *p = NULL; \ } \ } \ UINT GetRequiredMem##name(void) { \ return ALGN_SIZE_EXTRES((num) * sizeof(type)); \ } /** See \ref SYSLIB_MEMORY_MACROS for description. */ #define C_ALLOC_MEM2_L(name, type, n1, n2, s) \ type *Get##name(int n) { \ FDK_ASSERT((n) < (n2)); \ return (type *)FDKcalloc_L(n1, sizeof(type), s); \ } \ void Free##name(type **p) { \ if (p != NULL) { \ FDKfree_L(*p); \ *p = NULL; \ } \ } \ UINT GetRequiredMem##name(void) { \ return ALGN_SIZE_EXTRES((n1) * sizeof(type)) * (n2); \ } /** See \ref SYSLIB_MEMORY_MACROS for description. */ #define C_AALLOC_MEM_L(name, type, num, s) \ type *Get##name(int n) { \ type *ap; \ FDK_ASSERT((n) == 0); \ ap = ((type *)FDKaalloc_L((num) * sizeof(type), ALIGNMENT_DEFAULT, s)); \ return ap; \ } \ void Free##name(type **p) { \ if (p != NULL) { \ FDKafree_L(*p); \ *p = NULL; \ } \ } \ UINT GetRequiredMem##name(void) { \ return ALGN_SIZE_EXTRES((num) * sizeof(type) + ALIGNMENT_DEFAULT + \ sizeof(void *)); \ } /** See \ref SYSLIB_MEMORY_MACROS for description. */ #define C_AALLOC_MEM2_L(name, type, n1, n2, s) \ type *Get##name(int n) { \ type *ap; \ FDK_ASSERT((n) < (n2)); \ ap = ((type *)FDKaalloc_L((n1) * sizeof(type), ALIGNMENT_DEFAULT, s)); \ return ap; \ } \ void Free##name(type **p) { \ if (p != NULL) { \ FDKafree_L(*p); \ *p = NULL; \ } \ } \ UINT GetRequiredMem##name(void) { \ return ALGN_SIZE_EXTRES((n1) * sizeof(type) + ALIGNMENT_DEFAULT + \ sizeof(void *)) * \ (n2); \ } /** See \ref SYSLIB_MEMORY_MACROS for description. */ #define C_ALLOC_MEM_OVERLAY(name, type, num, sect, tag) \ C_AALLOC_MEM_L(name, type, num, sect) /** See \ref SYSLIB_MEMORY_MACROS for description. */ #define C_AALLOC_SCRATCH_START(name, type, n) \ type _##name[(n) + (ALIGNMENT_DEFAULT + sizeof(type) - 1)]; \ type *name = (type *)ALIGN_PTR(_##name); \ C_ALLOC_ALIGNED_REGISTER(name, (n) * sizeof(type)); /** See \ref SYSLIB_MEMORY_MACROS for description. */ #define C_ALLOC_SCRATCH_START(name, type, n) type name[n]; /** See \ref SYSLIB_MEMORY_MACROS for description. */ #define C_AALLOC_SCRATCH_END(name, type, n) C_ALLOC_ALIGNED_UNREGISTER(name); /** See \ref SYSLIB_MEMORY_MACROS for description. */ #define C_ALLOC_SCRATCH_END(name, type, n) /** See \ref SYSLIB_MEMORY_MACROS for description. */ #define C_AALLOC_STACK_START(name, type, n) \ type _##name[(n) + (ALIGNMENT_DEFAULT + sizeof(type) - 1)]; \ type *name = (type *)ALIGN_PTR(_##name); \ C_ALLOC_ALIGNED_REGISTER(name, (n) * sizeof(type)); /** See \ref SYSLIB_MEMORY_MACROS for description. */ #define C_AALLOC_STACK_END(name, type, n) C_ALLOC_ALIGNED_UNREGISTER(name); /*! @} */ #define C_ALLOC_ALIGNED_REGISTER(x, size) #define C_ALLOC_ALIGNED_UNREGISTER(x) #define C_ALLOC_ALIGNED_CHECK(x) #define C_ALLOC_ALIGNED_CHECK2(x, y) #define FDK_showBacktrace(a, b) /*! \addtogroup SYSLIB_EXITCODES Unified exit codes * Exit codes to be used as return values of FDK software test and * demonstration applications. Not as return values of product modules and/or * libraries. * @{ */ #define FDK_EXITCODE_OK 0 /*!< Successful termination. No errors. */ #define FDK_EXITCODE_USAGE \ 64 /*!< The command/application was used incorrectly, e.g. with the wrong \ number of arguments, a bad flag, a bad syntax in a parameter, or \ whatever. */ #define FDK_EXITCODE_DATAERROR \ 65 /*!< The input data was incorrect in some way. This should only be used \ for user data and not system files. */ #define FDK_EXITCODE_NOINPUT \ 66 /*!< An input file (not a system file) did not exist or was not readable. \ */ #define FDK_EXITCODE_UNAVAILABLE \ 69 /*!< A service is unavailable. This can occur if a support program or \ file does not exist. This can also be used as a catchall message when \ something you wanted to do doesn't work, but you don't know why. */ #define FDK_EXITCODE_SOFTWARE \ 70 /*!< An internal software error has been detected. This should be limited \ to non- operating system related errors as possible. */ #define FDK_EXITCODE_CANTCREATE \ 73 /*!< A (user specified) output file cannot be created. */ #define FDK_EXITCODE_IOERROR \ 74 /*!< An error occurred while doing I/O on some file. */ /*! @} */ /*-------------------------------------------- * Runtime support declarations *---------------------------------------------*/ #ifdef __cplusplus extern "C" { #endif void FDKprintf(const char *szFmt, ...); void FDKprintfErr(const char *szFmt, ...); /** Wrapper for 's getchar(). */ int FDKgetchar(void); INT FDKfprintf(void *stream, const char *format, ...); INT FDKsprintf(char *str, const char *format, ...); char *FDKstrchr(char *s, INT c); const char *FDKstrstr(const char *haystack, const char *needle); char *FDKstrcpy(char *dest, const char *src); char *FDKstrncpy(char *dest, const char *src, const UINT n); #define FDK_MAX_OVERLAYS 8 /**< Maximum number of memory overlays. */ void *FDKcalloc(const UINT n, const UINT size); void *FDKmalloc(const UINT size); void FDKfree(void *ptr); /** * Allocate and clear an aligned memory area. Use FDKafree() instead of * FDKfree() for these memory areas. * * \param size Size of requested memory in bytes. * \param alignment Alignment of requested memory in bytes. * \return Pointer to allocated memory. */ void *FDKaalloc(const UINT size, const UINT alignment); /** * Free an aligned memory area. * * \param ptr Pointer to be freed. */ void FDKafree(void *ptr); /** * Allocate memory in a specific memory section. * Requests can be made for internal or external memory. If internal memory is * requested, FDKcalloc_L() first tries to use L1 memory, which sizes are * defined by ::DATA_L1_A_SIZE and ::DATA_L1_B_SIZE. If no L1 memory is * available, then FDKcalloc_L() tries to use L2 memory. If that fails as well, * the requested memory is allocated at an extern location using the fallback * FDKcalloc(). * * \param n See MSDN documentation on calloc(). * \param size See MSDN documentation on calloc(). * \param s Memory section. * \return See MSDN documentation on calloc(). */ void *FDKcalloc_L(const UINT n, const UINT size, MEMORY_SECTION s); /** * Allocate aligned memory in a specific memory section. * See FDKcalloc_L() description for details - same applies here. */ void *FDKaalloc_L(const UINT size, const UINT alignment, MEMORY_SECTION s); /** * Free memory that was allocated in a specific memory section. */ void FDKfree_L(void *ptr); /** * Free aligned memory that was allocated in a specific memory section. */ void FDKafree_L(void *ptr); /** * Copy memory. Source and destination memory must not overlap. * Either use implementation from a Standard Library, or, if no Standard Library * is available, a generic implementation. * The define ::USE_BUILTIN_MEM_FUNCTIONS in genericStds.cpp controls what to * use. The function arguments correspond to the standard memcpy(). Please see * MSDN documentation for details on how to use it. */ void FDKmemcpy(void *dst, const void *src, const UINT size); /** * Copy memory. Source and destination memory are allowed to overlap. * Either use implementation from a Standard Library, or, if no Standard Library * is available, a generic implementation. * The define ::USE_BUILTIN_MEM_FUNCTIONS in genericStds.cpp controls what to * use. The function arguments correspond to the standard memmove(). Please see * MSDN documentation for details on how to use it. */ void FDKmemmove(void *dst, const void *src, const UINT size); /** * Clear memory. * Either use implementation from a Standard Library, or, if no Standard Library * is available, a generic implementation. * The define ::USE_BUILTIN_MEM_FUNCTIONS in genericStds.cpp controls what to * use. The function arguments correspond to the standard memclear(). Please see * MSDN documentation for details on how to use it. */ void FDKmemclear(void *memPtr, const UINT size); /** * Fill memory with values. * The function arguments correspond to the standard memset(). Please see MSDN * documentation for details on how to use it. */ void FDKmemset(void *memPtr, const INT value, const UINT size); /* Compare function wrappers */ INT FDKmemcmp(const void *s1, const void *s2, const UINT size); INT FDKstrcmp(const char *s1, const char *s2); INT FDKstrncmp(const char *s1, const char *s2, const UINT size); UINT FDKstrlen(const char *s); #define FDKmax(a, b) ((a) > (b) ? (a) : (b)) #define FDKmin(a, b) ((a) < (b) ? (a) : (b)) #define FDK_INT_MAX ((INT)0x7FFFFFFF) #define FDK_INT_MIN ((INT)0x80000000) /* FILE I/O */ /*! * Check platform for endianess. * * \return 1 if platform is little endian, non-1 if platform is big endian. */ int IS_LITTLE_ENDIAN(void); /*! * Convert input value to little endian format. * * \param val Value to be converted. It may be in both big or little endian. * \return Value in little endian format. */ UINT TO_LITTLE_ENDIAN(UINT val); /*! * \fn FDKFILE *FDKfopen(const char *filename, const char *mode); * Standard fopen() wrapper. * \fn INT FDKfclose(FDKFILE *FP); * Standard fclose() wrapper. * \fn INT FDKfseek(FDKFILE *FP, LONG OFFSET, int WHENCE); * Standard fseek() wrapper. * \fn INT FDKftell(FDKFILE *FP); * Standard ftell() wrapper. * \fn INT FDKfflush(FDKFILE *fp); * Standard fflush() wrapper. * \fn UINT FDKfwrite(const void *ptrf, INT size, UINT nmemb, FDKFILE *fp); * Standard fwrite() wrapper. * \fn UINT FDKfread(void *dst, INT size, UINT nmemb, FDKFILE *fp); * Standard fread() wrapper. */ typedef void FDKFILE; extern const INT FDKSEEK_SET, FDKSEEK_CUR, FDKSEEK_END; FDKFILE *FDKfopen(const char *filename, const char *mode); INT FDKfclose(FDKFILE *FP); INT FDKfseek(FDKFILE *FP, LONG OFFSET, int WHENCE); INT FDKftell(FDKFILE *FP); INT FDKfflush(FDKFILE *fp); UINT FDKfwrite(const void *ptrf, INT size, UINT nmemb, FDKFILE *fp); UINT FDKfread(void *dst, INT size, UINT nmemb, FDKFILE *fp); char *FDKfgets(void *dst, INT size, FDKFILE *fp); void FDKrewind(FDKFILE *fp); INT FDKfeof(FDKFILE *fp); /** * \brief Write each member in little endian order. Convert automatically * to host endianess. * \param ptrf Pointer to memory where to read data from. * \param size Size of each item to be written. * \param nmemb Number of items to be written. * \param fp File pointer of type FDKFILE. * \return Number of items read on success and fread() error on failure. */ UINT FDKfwrite_EL(const void *ptrf, INT size, UINT nmemb, FDKFILE *fp); /** * \brief Read variable of size "size" as little endian. Convert * automatically to host endianess. 4-byte alignment is enforced for 24 bit * data, at 32 bit full scale. * \param dst Pointer to memory where to store data into. * \param size Size of each item to be read. * \param nmemb Number of items to be read. * \param fp File pointer of type FDKFILE. * \return Number of items read on success and fread() error on failure. */ UINT FDKfread_EL(void *dst, INT size, UINT nmemb, FDKFILE *fp); /** * \brief Print FDK software disclaimer. */ void FDKprintDisclaimer(void); #ifdef __cplusplus } #endif #endif /* GENERICSTDS_H */ fdk-aac-2.0.2/libSYS/include/machine_type.h000066400000000000000000000334531404225310700204150ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /************************* System integration library ************************** Author(s): Description: *******************************************************************************/ /** \file machine_type.h * \brief Type defines for various processors and compiler tools. */ #if !defined(MACHINE_TYPE_H) #define MACHINE_TYPE_H #include /* Needed to define size_t */ #if defined(__ANDROID__) && (__GNUC__ == 4) && (__GNUC_MINOR__ == 4) && \ (__GNUC_GNU_INLINE__ == 1) typedef unsigned long long uint64_t; #include #endif /* Library calling convention spec. __cdecl and friends might be added here as * required. */ #define LINKSPEC_H #define LINKSPEC_CPP /* for doxygen the following docu parts must be separated */ /** \var SCHAR * Data type representing at least 1 byte signed integer on all supported * platforms. */ /** \var UCHAR * Data type representing at least 1 byte unsigned integer on all * supported platforms. */ /** \var INT * Data type representing at least 4 byte signed integer on all supported * platforms. */ /** \var UINT * Data type representing at least 4 byte unsigned integer on all * supported platforms. */ /** \var LONG * Data type representing 4 byte signed integer on all supported * platforms. */ /** \var ULONG * Data type representing 4 byte unsigned integer on all supported * platforms. */ /** \var SHORT * Data type representing 2 byte signed integer on all supported * platforms. */ /** \var USHORT * Data type representing 2 byte unsigned integer on all supported * platforms. */ /** \var INT64 * Data type representing 8 byte signed integer on all supported * platforms. */ /** \var UINT64 * Data type representing 8 byte unsigned integer on all supported * platforms. */ /** \def SHORT_BITS * Number of bits the data type short represents. sizeof() is not suited * to get this info, because a byte is not always defined as 8 bits. */ /** \def CHAR_BITS * Number of bits the data type char represents. sizeof() is not suited * to get this info, because a byte is not always defined as 8 bits. */ /** \var INT_PCM * Data type representing the width of input and output PCM samples. */ typedef signed int INT; typedef unsigned int UINT; #ifdef __LP64__ /* force FDK long-datatypes to 4 byte */ /* Use defines to avoid type alias problems on 64 bit machines. */ #define LONG INT #define ULONG UINT #else /* __LP64__ */ typedef signed long LONG; typedef unsigned long ULONG; #endif /* __LP64__ */ typedef signed short SHORT; typedef unsigned short USHORT; typedef signed char SCHAR; typedef unsigned char UCHAR; #define SHORT_BITS 16 #define CHAR_BITS 8 /* Define 64 bit base integer type. */ #ifdef _MSC_VER typedef __int64 INT64; typedef unsigned __int64 UINT64; #else typedef long long INT64; typedef unsigned long long UINT64; #endif #ifndef NULL #ifdef __cplusplus #define NULL 0 #else #define NULL ((void *)0) #endif #endif #if ((defined(__i686__) || defined(__i586__) || defined(__i386__) || \ defined(__x86_64__)) || \ (defined(_MSC_VER) && (defined(_M_IX86) || defined(_M_X64)))) && \ !defined(FDK_ASSERT_ENABLE) #define FDK_ASSERT_ENABLE #endif #if defined(FDK_ASSERT_ENABLE) #include #define FDK_ASSERT(x) assert(x) #else #define FDK_ASSERT(ignore) #endif typedef SHORT INT_PCM; #define MAXVAL_PCM MAXVAL_SGL #define MINVAL_PCM MINVAL_SGL #define WAV_BITS 16 #define SAMPLE_BITS 16 #define SAMPLE_MAX ((INT_PCM)(((ULONG)1 << (SAMPLE_BITS - 1)) - 1)) #define SAMPLE_MIN (~SAMPLE_MAX) /*! * \def RAM_ALIGN * Used to align memory as prefix before memory declaration. For example: \code RAM_ALIGN int myArray[16]; \endcode Note, that not all platforms support this mechanism. For example with TI compilers a preprocessor pragma is used, but to do something like \code #define RAM_ALIGN #pragma DATA_ALIGN(x) \endcode would require the preprocessor to process this line twice to fully resolve it. Hence, a fully platform-independant way to use alignment is not supported. * \def ALIGNMENT_DEFAULT * Default alignment in bytes. */ #define ALIGNMENT_DEFAULT 8 /* RAM_ALIGN keyword causes memory alignment of global variables. */ #if defined(_MSC_VER) #define RAM_ALIGN __declspec(align(ALIGNMENT_DEFAULT)) #elif defined(__GNUC__) #define RAM_ALIGN __attribute__((aligned(ALIGNMENT_DEFAULT))) #else #define RAM_ALIGN #endif /*! * \def RESTRICT * The restrict keyword is supported by some platforms and RESTRICT maps * to either the corresponding keyword on each platform or to void if the * compiler does not provide such feature. It tells the compiler that a * pointer points to memory that does not overlap with other memories pointed to * by other pointers. If this keyword is used and the assumption of no * overlap is not true the resulting code might crash. * * \def WORD_ALIGNED(x) * Tells the compiler that pointer x is 16 bit aligned. It does not cause * the address itself to be aligned, but serves as a hint to the optimizer. The * alignment of the pointer must be guarranteed, if not the code might * crash. * * \def DWORD_ALIGNED(x) * Tells the compiler that pointer x is 32 bit aligned. It does not cause * the address itself to be aligned, but serves as a hint to the optimizer. The * alignment of the pointer must be guarranteed, if not the code might * crash. * */ #define RESTRICT #define WORD_ALIGNED(x) C_ALLOC_ALIGNED_CHECK2((const void *)(x), 2); #define DWORD_ALIGNED(x) C_ALLOC_ALIGNED_CHECK2((const void *)(x), 4); /*----------------------------------------------------------------------------------- * ALIGN_SIZE *-----------------------------------------------------------------------------------*/ /*! * \brief This macro aligns a given value depending on ::ALIGNMENT_DEFAULT. * * For example if #ALIGNMENT_DEFAULT equals 8, then: * - ALIGN_SIZE(3) returns 8 * - ALIGN_SIZE(8) returns 8 * - ALIGN_SIZE(9) returns 16 */ #define ALIGN_SIZE(a) \ ((a) + (((INT)ALIGNMENT_DEFAULT - ((size_t)(a) & (ALIGNMENT_DEFAULT - 1))) & \ (ALIGNMENT_DEFAULT - 1))) /*! * \brief This macro aligns a given address depending on ::ALIGNMENT_DEFAULT. */ #define ALIGN_PTR(a) \ ((void *)((unsigned char *)(a) + \ ((((INT)ALIGNMENT_DEFAULT - \ ((size_t)(a) & (ALIGNMENT_DEFAULT - 1))) & \ (ALIGNMENT_DEFAULT - 1))))) /* Alignment macro for libSYS heap implementation */ #define ALIGNMENT_EXTRES (ALIGNMENT_DEFAULT) #define ALGN_SIZE_EXTRES(a) \ ((a) + (((INT)ALIGNMENT_EXTRES - ((INT)(a) & (ALIGNMENT_EXTRES - 1))) & \ (ALIGNMENT_EXTRES - 1))) /*! * \def FDK_FORCEINLINE * Sometimes compiler do not do what they are told to do, and in case of * inlining some additional command might be necessary depending on the * platform. * * \def FDK_INLINE * Defines how the compiler is told to inline stuff. */ #ifndef FDK_FORCEINLINE #if defined(__GNUC__) && !defined(__SDE_MIPS__) #define FDK_FORCEINLINE inline __attribute((always_inline)) #else #define FDK_FORCEINLINE inline #endif #endif #define FDK_INLINE static inline /*! * \def LNK_SECTION_DATA_L1 * The LNK_SECTION_* defines allow memory to be drawn from specific memory * sections. Used as prefix before variable declaration. * * \def LNK_SECTION_DATA_L2 * See ::LNK_SECTION_DATA_L1 * \def LNK_SECTION_L1_DATA_A * See ::LNK_SECTION_DATA_L1 * \def LNK_SECTION_L1_DATA_B * See ::LNK_SECTION_DATA_L1 * \def LNK_SECTION_CONSTDATA_L1 * See ::LNK_SECTION_DATA_L1 * \def LNK_SECTION_CONSTDATA * See ::LNK_SECTION_DATA_L1 * \def LNK_SECTION_CODE_L1 * See ::LNK_SECTION_DATA_L1 * \def LNK_SECTION_CODE_L2 * See ::LNK_SECTION_DATA_L1 * \def LNK_SECTION_INITCODE * See ::LNK_SECTION_DATA_L1 */ /************************************************** * Code Section macros **************************************************/ #define LNK_SECTION_CODE_L1 #define LNK_SECTION_CODE_L2 #define LNK_SECTION_INITCODE /* Memory section macros. */ /* default fall back */ #define LNK_SECTION_DATA_L1 #define LNK_SECTION_DATA_L2 #define LNK_SECTION_CONSTDATA #define LNK_SECTION_CONSTDATA_L1 #define LNK_SECTION_L1_DATA_A #define LNK_SECTION_L1_DATA_B /************************************************** * Macros regarding static code analysis **************************************************/ #ifdef __cplusplus #if !defined(__has_cpp_attribute) #define __has_cpp_attribute(x) 0 #endif #if defined(__clang__) && __has_cpp_attribute(clang::fallthrough) #define FDK_FALLTHROUGH [[clang::fallthrough]] #endif #endif #ifndef FDK_FALLTHROUGH #if defined(__GNUC__) && (__GNUC__ >= 7) #define FDK_FALLTHROUGH __attribute__((fallthrough)) #else #define FDK_FALLTHROUGH #endif #endif #ifdef _MSC_VER /* * Sometimes certain features are excluded from compilation and therefore the * warning 4065 may occur: "switch statement contains 'default' but no 'case' * labels" We consider this warning irrelevant and disable it. */ #pragma warning(disable : 4065) #endif #endif /* MACHINE_TYPE_H */ fdk-aac-2.0.2/libSYS/include/syslib_channelMapDescr.h000066400000000000000000000203351404225310700223570ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /************************* System integration library ************************** Author(s): Thomas Dietzen Description: *******************************************************************************/ /** \file syslib_channelMapDescr.h * \brief Function and structure declarations for the channel map descriptor implementation. */ #ifndef SYSLIB_CHANNELMAPDESCR_H #define SYSLIB_CHANNELMAPDESCR_H #include "machine_type.h" /** * \brief Contains information needed for a single channel map. */ typedef struct { const UCHAR* pChannelMap; /*!< Actual channel mapping for one single configuration. */ UCHAR numChannels; /*!< The number of channels for the channel map which is the maximum used channel index+1. */ } CHANNEL_MAP_INFO; /** * \brief This is the main data struct. It contains the mapping for all * channel configurations such as administration information. * * CAUTION: Do not access this structure directly from a algorithm specific * library. Always use one of the API access functions below! */ typedef struct { const CHANNEL_MAP_INFO* pMapInfoTab; /*!< Table of channel maps. */ UINT mapInfoTabLen; /*!< Length of the channel map table array. */ UINT fPassThrough; /*!< Flag that defines whether the specified mapping shall be applied (value: 0) or the input just gets passed through (MPEG mapping). */ } FDK_channelMapDescr; #ifdef __cplusplus extern "C" { #endif /** * \brief Initialize a given channel map descriptor. * * \param pMapDescr Pointer to a channel map descriptor to be initialized. * \param pMapInfoTab Table of channel maps to initizalize the descriptor with. * If a NULL pointer is given a default table for WAV-like mapping will be used. * \param mapInfoTabLen Length of the channel map table array (pMapInfoTab). If a zero length is given a default table for WAV-like mapping will be used. * \param fPassThrough If the flag is set the reordering (given by pMapInfoTab) will be bypassed. */ void FDK_chMapDescr_init(FDK_channelMapDescr* const pMapDescr, const CHANNEL_MAP_INFO* const pMapInfoTab, const UINT mapInfoTabLen, const UINT fPassThrough); /** * \brief Change the channel reordering state of a given channel map * descriptor. * * \param pMapDescr Pointer to a (initialized) channel map descriptor. * \param fPassThrough If the flag is set the reordering (given by * pMapInfoTab) will be bypassed. * \return Value unequal to zero if set operation was not * successful. And zero on success. */ int FDK_chMapDescr_setPassThrough(FDK_channelMapDescr* const pMapDescr, UINT fPassThrough); /** * \brief Get the mapping value for a specific channel and map index. * * \param pMapDescr Pointer to channel map descriptor. * \param chIdx Channel index. * \param mapIdx Mapping index (corresponding to the channel configuration * index). * \return Mapping value. */ UCHAR FDK_chMapDescr_getMapValue(const FDK_channelMapDescr* const pMapDescr, const UCHAR chIdx, const UINT mapIdx); /** * \brief Evaluate whether channel map descriptor is reasonable or not. * * \param pMapDescr Pointer to channel map descriptor. * \return Value unequal to zero if descriptor is valid, otherwise * zero. */ int FDK_chMapDescr_isValid(const FDK_channelMapDescr* const pMapDescr); /** * Extra variables for setting up Wg4 channel mapping. */ extern const CHANNEL_MAP_INFO FDK_mapInfoTabWg4[]; extern const UINT FDK_mapInfoTabLenWg4; #ifdef __cplusplus } #endif #endif /* !defined(SYSLIB_CHANNELMAPDESCR_H) */ fdk-aac-2.0.2/libSYS/src/000077500000000000000000000000001404225310700147335ustar00rootroot00000000000000fdk-aac-2.0.2/libSYS/src/genericStds.cpp000066400000000000000000000327451404225310700177240ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /************************* System integration library ************************** Author(s): Description: - Generic memory, stdio, string, etc. function wrappers or builtins. - OS dependant function wrappers. *******************************************************************************/ #ifndef _CRT_SECURE_NO_WARNINGS #define _CRT_SECURE_NO_WARNINGS #endif #define __GENERICSTDS_CPP__ #include "genericStds.h" /* library info */ #define SYS_LIB_VL0 2 #define SYS_LIB_VL1 0 #define SYS_LIB_VL2 0 #define SYS_LIB_TITLE "System Integration Library" #ifdef __ANDROID__ #define SYS_LIB_BUILD_DATE "" #define SYS_LIB_BUILD_TIME "" #else #define SYS_LIB_BUILD_DATE __DATE__ #define SYS_LIB_BUILD_TIME __TIME__ #endif #include #include #include #include /*************************************************************** * memory allocation monitoring variables ***************************************************************/ /* Include OS/System specific implementations. */ #include #include #include void FDKprintf(const char *szFmt, ...) { va_list ap; va_start(ap, szFmt); vprintf(szFmt, ap); va_end(ap); } void FDKprintfErr(const char *szFmt, ...) { va_list ap; va_start(ap, szFmt); vfprintf(stderr, szFmt, ap); va_end(ap); } int FDKgetchar(void) { return getchar(); } INT FDKfprintf(FDKFILE *stream, const char *format, ...) { INT chars = 0; va_list ap; va_start(ap, format); chars += vfprintf((FILE *)stream, format, ap); va_end(ap); return chars; } INT FDKsprintf(char *str, const char *format, ...) { INT chars = 0; va_list ap; va_start(ap, format); chars += vsprintf(str, format, ap); va_end(ap); return chars; } /************************************************************************************************/ /************************************************************************************************/ char *FDKstrchr(char *s, INT c) { return strchr(s, c); } const char *FDKstrstr(const char *haystack, const char *needle) { return strstr(haystack, needle); } char *FDKstrcpy(char *dest, const char *src) { return strcpy(dest, src); } char *FDKstrncpy(char *dest, const char *src, UINT n) { return strncpy(dest, src, n); } /************************************************************************* * DYNAMIC MEMORY management (heap) *************************************************************************/ void *FDKcalloc(const UINT n, const UINT size) { void *ptr; ptr = calloc(n, size); return ptr; } void *FDKmalloc(const UINT size) { void *ptr; ptr = malloc(size); return ptr; } void FDKfree(void *ptr) { free((INT *)ptr); } void *FDKaalloc(const UINT size, const UINT alignment) { void *addr, *result = NULL; addr = FDKcalloc(1, size + alignment + (UINT)sizeof(void *)); /* Malloc and clear memory. */ if (addr != NULL) { result = ALIGN_PTR((unsigned char *)addr + sizeof(void *)); /* Get aligned memory base address. */ *(((void **)result) - 1) = addr; /* Save malloc'ed memory pointer. */ C_ALLOC_ALIGNED_REGISTER(result, size); } return result; /* Return aligned address. */ } void FDKafree(void *ptr) { void *addr; addr = *(((void **)ptr) - 1); /* Get pointer to malloc'ed memory. */ C_ALLOC_ALIGNED_UNREGISTER(ptr); FDKfree(addr); /* Free malloc'ed memory area. */ } /*--------------------------------------------------------------------------* * DATA MEMORY L1/L2 (fallback) *--------------------------------------------------------------------------*/ /*--------------------------------------------------------------------------* * FDKcalloc_L *--------------------------------------------------------------------------*/ void *FDKcalloc_L(const UINT dim, const UINT size, MEMORY_SECTION s) { return FDKcalloc(dim, size); } void FDKfree_L(void *p) { FDKfree(p); } void *FDKaalloc_L(const UINT size, const UINT alignment, MEMORY_SECTION s) { void *addr, *result = NULL; addr = FDKcalloc_L(1, size + alignment + (UINT)sizeof(void *), s); /* Malloc and clear memory. */ if (addr != NULL) { result = ALIGN_PTR((unsigned char *)addr + sizeof(void *)); /* Get aligned memory base address. */ *(((void **)result) - 1) = addr; /* Save malloc'ed memory pointer. */ C_ALLOC_ALIGNED_REGISTER(result, size); } return result; /* Return aligned address. */ } void FDKafree_L(void *ptr) { void *addr; addr = *(((void **)ptr) - 1); /* Get pointer to malloc'ed memory. */ C_ALLOC_ALIGNED_UNREGISTER(ptr); FDKfree_L(addr); /* Free malloc'ed memory area. */ } /*--------------------------------------------------------------------------------------- * FUNCTION: FDKmemcpy * DESCRIPTION: - copies memory from "src" to "dst" with length "size" bytes * - compiled with FDK_DEBUG will give you warnings *---------------------------------------------------------------------------------------*/ void FDKmemcpy(void *dst, const void *src, const UINT size) { /* -- check for overlapping memory areas -- */ FDK_ASSERT(((const unsigned char *)dst - (const unsigned char *)src) >= (ptrdiff_t)size || ((const unsigned char *)src - (const unsigned char *)dst) >= (ptrdiff_t)size); /* do the copy */ memcpy(dst, src, size); } void FDKmemmove(void *dst, const void *src, const UINT size) { memmove(dst, src, size); } void FDKmemset(void *memPtr, const INT value, const UINT size) { memset(memPtr, value, size); } void FDKmemclear(void *memPtr, const UINT size) { FDKmemset(memPtr, 0, size); } UINT FDKstrlen(const char *s) { return (UINT)strlen(s); } /* Compare function wrappers */ INT FDKmemcmp(const void *s1, const void *s2, const UINT size) { return memcmp(s1, s2, size); } INT FDKstrcmp(const char *s1, const char *s2) { return strcmp(s1, s2); } INT FDKstrncmp(const char *s1, const char *s2, const UINT size) { return strncmp(s1, s2, size); } int IS_LITTLE_ENDIAN(void) { int __dummy = 1; return (*((UCHAR *)(&(__dummy)))); } UINT TO_LITTLE_ENDIAN(UINT val) { return IS_LITTLE_ENDIAN() ? val : (((val & 0xff) << 24) | ((val & 0xff00) << 8) | ((val & 0xff0000) >> 8) | ((val & 0xff000000) >> 24)); } /* ==================== FILE I/O ====================== */ FDKFILE *FDKfopen(const char *filename, const char *mode) { return fopen(filename, mode); } INT FDKfclose(FDKFILE *fp) { return fclose((FILE *)fp); } INT FDKfseek(FDKFILE *fp, LONG OFFSET, int WHENCE) { return fseek((FILE *)fp, OFFSET, WHENCE); } INT FDKftell(FDKFILE *fp) { return ftell((FILE *)fp); } INT FDKfflush(FDKFILE *fp) { return fflush((FILE *)fp); } const INT FDKSEEK_SET = SEEK_SET; const INT FDKSEEK_CUR = SEEK_CUR; const INT FDKSEEK_END = SEEK_END; UINT FDKfwrite(const void *ptrf, INT size, UINT nmemb, FDKFILE *fp) { return (UINT)fwrite(ptrf, size, nmemb, (FILE *)fp); } UINT FDKfread(void *dst, INT size, UINT nmemb, FDKFILE *fp) { return (UINT)fread(dst, size, nmemb, (FILE *)fp); } char *FDKfgets(void *dst, INT size, FDKFILE *fp) { return fgets((char *)dst, size, (FILE *)fp); } void FDKrewind(FDKFILE *fp) { FDKfseek((FILE *)fp, 0, FDKSEEK_SET); } UINT FDKfwrite_EL(const void *ptrf, INT size, UINT nmemb, FDKFILE *fp) { if (IS_LITTLE_ENDIAN()) { FDKfwrite(ptrf, size, nmemb, fp); } else { UINT n; INT s; const UCHAR *ptr = (const UCHAR *)ptrf; for (n = 0; n < nmemb; n++) { for (s = size - 1; s >= 0; s--) { FDKfwrite(ptr + s, 1, 1, fp); } ptr = ptr + size; } } return nmemb; } UINT FDKfread_EL(void *dst, INT size, UINT nmemb, FDKFILE *fp) { UINT n, s0, s1, err; UCHAR tmp, *ptr; UCHAR tmp24[3]; /* Enforce alignment of 24 bit data. */ if (size == 3) { ptr = (UCHAR *)dst; for (n = 0; n < nmemb; n++) { if ((err = FDKfread(tmp24, 1, 3, fp)) != 3) { return err; } *ptr++ = tmp24[0]; *ptr++ = tmp24[1]; *ptr++ = tmp24[2]; /* Sign extension */ if (tmp24[2] & 0x80) { *ptr++ = 0xff; } else { *ptr++ = 0; } } err = nmemb; size = sizeof(LONG); } else { if ((err = FDKfread(dst, size, nmemb, fp)) != nmemb) { return err; } } if (!IS_LITTLE_ENDIAN() && size > 1) { ptr = (UCHAR *)dst; for (n = 0; n < nmemb; n++) { for (s0 = 0, s1 = size - 1; s0 < s1; s0++, s1--) { tmp = ptr[s0]; ptr[s0] = ptr[s1]; ptr[s1] = tmp; } ptr += size; } } return err; } INT FDKfeof(FDKFILE *fp) { return feof((FILE *)fp); } /* Global initialization/cleanup */ void FDKprintDisclaimer(void) { FDKprintf( "This program is protected by copyright law and international treaties.\n" "Any reproduction or distribution of this program, or any portion\n" "of it, may result in severe civil and criminal penalties, and will be\n" "prosecuted to the maximum extent possible under law.\n\n"); } fdk-aac-2.0.2/libSYS/src/syslib_channelMapDescr.cpp000066400000000000000000000272051404225310700220610ustar00rootroot00000000000000/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. This FDK AAC Codec software is intended to be used on a wide variety of Android devices. AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part of the MPEG specifications. Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners individually for the purpose of encoding or decoding bit streams in products that are compliant with the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec software may already be covered under those patent licenses when it is used for those licensed purposes only. Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional applications information and documentation. 2. COPYRIGHT LICENSE Redistribution and use in source and binary forms, with or without modification, are permitted without payment of copyright license fees provided that you satisfy the following conditions: You must retain the complete text of this software license in redistributions of the FDK AAC Codec or your modifications thereto in source code form. You must retain the complete text of this software license in the documentation and/or other materials provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. You must make available free of charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. The name of Fraunhofer may not be used to endorse or promote products derived from this library without prior written permission. You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto. Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software and the date of any change. For modified versions of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 3. NO PATENT LICENSE NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with respect to this software. You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, including but not limited to procurement of substitute goods or services; loss of use, data, or profits, or business interruption, however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence), arising in any way out of the use of this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION Fraunhofer Institute for Integrated Circuits IIS Attention: Audio and Multimedia Departments - FDK AAC LL Am Wolfsmantel 33 91058 Erlangen, Germany www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ----------------------------------------------------------------------------- */ /************************* System integration library ************************** Author(s): Thomas Dietzen Description: *******************************************************************************/ /** \file syslib_channelMapDescr.cpp * \brief Implementation of routines that handle the channel map descriptor. */ #include "syslib_channelMapDescr.h" #define DFLT_CH_MAP_TAB_LEN \ (15) /* Length of the default channel map info table. */ /** * \brief The following arrays provide a channel map for each channel config (0 * to 14). * * The i-th channel will be mapped to the postion a[i-1]+1 * with i>0 and a[] is one of the following mapping arrays. */ static const UCHAR mapFallback[] = {0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23}; static const UCHAR mapCfg1[] = {0, 1}; static const UCHAR mapCfg2[] = {0, 1}; static const UCHAR mapCfg3[] = {2, 0, 1}; static const UCHAR mapCfg4[] = {2, 0, 1, 3}; static const UCHAR mapCfg5[] = {2, 0, 1, 3, 4}; static const UCHAR mapCfg6[] = {2, 0, 1, 4, 5, 3}; static const UCHAR mapCfg7[] = {2, 6, 7, 0, 1, 4, 5, 3}; static const UCHAR mapCfg11[] = {2, 0, 1, 4, 5, 6, 3}; static const UCHAR mapCfg12[] = {2, 0, 1, 6, 7, 4, 5, 3}; static const UCHAR mapCfg13[] = {2, 6, 7, 0, 1, 10, 11, 4, 5, 8, 3, 9, 14, 12, 13, 18, 19, 15, 16, 17, 20, 21, 22, 23}; static const UCHAR mapCfg14[] = {2, 0, 1, 4, 5, 3, 6, 7}; /** * \brief Default table comprising channel map information for each channel * config (0 to 14). */ static const CHANNEL_MAP_INFO mapInfoTabDflt[DFLT_CH_MAP_TAB_LEN] = {/* chCfg, map, numCh */ /* 0 */ {mapFallback, 24}, /* 1 */ {mapCfg1, 2}, /* 2 */ {mapCfg2, 2}, /* 3 */ {mapCfg3, 3}, /* 4 */ {mapCfg4, 4}, /* 5 */ {mapCfg5, 5}, /* 6 */ {mapCfg6, 6}, /* 7 */ {mapCfg7, 8}, /* 8 */ {mapFallback, 24}, /* 9 */ {mapFallback, 24}, /* 10 */ {mapFallback, 24}, /* 11 */ {mapCfg11, 7}, /* 12 */ {mapCfg12, 8}, /* 13 */ {mapCfg13, 24}, /* 14 */ {mapCfg14, 8}}; static const UCHAR mapWg4Cfg1[] = {0, 1}; static const UCHAR mapWg4Cfg2[] = {0, 1}; static const UCHAR mapWg4Cfg3[] = {2, 0, 1}; static const UCHAR mapWg4Cfg4[] = {3, 0, 1, 2}; static const UCHAR mapWg4Cfg5[] = {4, 0, 1, 2, 3}; static const UCHAR mapWg4Cfg6[] = {4, 0, 1, 2, 3, 5}; static const UCHAR mapWg4Cfg7[] = {6, 0, 1, 2, 3, 4, 5, 7}; static const UCHAR mapWg4Cfg14[] = {6, 0, 1, 2, 3, 4, 5, 7}; const CHANNEL_MAP_INFO FDK_mapInfoTabWg4[] = {/* chCfg, map, numCh */ /* 0 */ {mapFallback, 24}, /* 1 */ {mapWg4Cfg1, 2}, /* 2 */ {mapWg4Cfg2, 2}, /* 3 */ {mapWg4Cfg3, 3}, /* 4 */ {mapWg4Cfg4, 4}, /* 5 */ {mapWg4Cfg5, 5}, /* 6 */ {mapWg4Cfg6, 6}, /* 7 */ {mapWg4Cfg7, 8}, /* 8 */ {mapFallback, 24}, /* 9 */ {mapFallback, 24}, /* 10 */ {mapFallback, 24}, /* 11 */ {mapFallback, 24}, // Unhandled for Wg4 yet /* 12 */ {mapFallback, 24}, // Unhandled for Wg4 yet /* 13 */ {mapFallback, 24}, // Unhandled for Wg4 yet /* 14 */ {mapFallback, 24}}; // Unhandled for Wg4 yet const UINT FDK_mapInfoTabLenWg4 = sizeof(FDK_mapInfoTabWg4)/sizeof(FDK_mapInfoTabWg4[0]); /** * Get the mapping value for a specific channel and map index. */ UCHAR FDK_chMapDescr_getMapValue(const FDK_channelMapDescr* const pMapDescr, const UCHAR chIdx, const UINT mapIdx) { UCHAR mapValue = chIdx; /* Pass through by default. */ FDK_ASSERT(pMapDescr != NULL); if ((pMapDescr->fPassThrough == 0) && (pMapDescr->pMapInfoTab != NULL) && (pMapDescr->mapInfoTabLen > mapIdx)) { /* Nest sanity check to avoid possible memory access violation. */ if (chIdx < pMapDescr->pMapInfoTab[mapIdx].numChannels) { mapValue = pMapDescr->pMapInfoTab[mapIdx].pChannelMap[chIdx]; } } return mapValue; } /** * \brief Evaluate whether single channel map is reasonable or not. * * \param pMapInfo Pointer to channel map. * \return Value unequal to zero if map is valid, otherwise zero. */ static int fdk_chMapDescr_isValidMap(const CHANNEL_MAP_INFO* const pMapInfo) { int result = 1; UINT i; if (pMapInfo == NULL) { result = 0; } else { UINT numChannels = pMapInfo->numChannels; /* Check for all map values if they are inside the range 0 to numChannels-1 * and unique. */ if (numChannels < 32) { /* Optimized version for less than 32 channels. Needs only one loop. */ UINT mappedChMask = 0x0; for (i = 0; i < numChannels; i += 1) { mappedChMask |= 1 << pMapInfo->pChannelMap[i]; } if (mappedChMask != (((UINT)1 << numChannels) - 1)) { result = 0; } } else { /* General case that can handle all number of channels but needs one more loop. */ for (i = 0; (i < numChannels) && result; i += 1) { UINT j; UCHAR value0 = pMapInfo->pChannelMap[i]; if (value0 > numChannels - 1) { /* out of range? */ result = 0; } for (j = numChannels - 1; (j > i) && result; j -= 1) { if (value0 == pMapInfo->pChannelMap[j]) { /* not unique */ result = 0; } } } } } return result; } /** * Evaluate whether channel map descriptor is reasonable or not. */ int FDK_chMapDescr_isValid(const FDK_channelMapDescr* const pMapDescr) { int result = 0; UINT i; if (pMapDescr != NULL) { result = 1; for (i = 0; (i < pMapDescr->mapInfoTabLen) && result; i += 1) { if (!fdk_chMapDescr_isValidMap(&pMapDescr->pMapInfoTab[i])) { result = 0; } } } return result; } /** * Initialize the complete channel map descriptor. */ void FDK_chMapDescr_init(FDK_channelMapDescr* const pMapDescr, const CHANNEL_MAP_INFO* const pMapInfoTab, const UINT mapInfoTabLen, const UINT fPassThrough) { if (pMapDescr != NULL) { int useDefaultTab = 1; pMapDescr->fPassThrough = (fPassThrough == 0) ? 0 : 1; if ((pMapInfoTab != NULL) && (mapInfoTabLen > 0)) { /* Set the valid custom mapping table. */ pMapDescr->pMapInfoTab = pMapInfoTab; pMapDescr->mapInfoTabLen = mapInfoTabLen; /* Validate the complete descriptor. */ useDefaultTab = (FDK_chMapDescr_isValid(pMapDescr) == 0) ? 1 : 0; } if (useDefaultTab != 0) { /* Set default table. */ pMapDescr->pMapInfoTab = mapInfoTabDflt; pMapDescr->mapInfoTabLen = DFLT_CH_MAP_TAB_LEN; } } } /** * Set channel mapping bypass flag in a given channel map descriptor. */ int FDK_chMapDescr_setPassThrough(FDK_channelMapDescr* const pMapDescr, UINT fPassThrough) { int err = 1; if (pMapDescr != NULL) { if ((pMapDescr->pMapInfoTab != NULL) && (pMapDescr->mapInfoTabLen > 0)) { pMapDescr->fPassThrough = (fPassThrough == 0) ? 0 : 1; err = 0; } } return err; } fdk-aac-2.0.2/m4/000077500000000000000000000000001404225310700133175ustar00rootroot00000000000000fdk-aac-2.0.2/m4/.gitkeep000066400000000000000000000000001404225310700147360ustar00rootroot00000000000000fdk-aac-2.0.2/wavreader.c000066400000000000000000000113061404225310700151240ustar00rootroot00000000000000/* ------------------------------------------------------------------ * Copyright (C) 2009 Martin Storsjo * * Licensed under the Apache License, Version 2.0 (the "License"); * you may not use this file except in compliance with the License. * You may obtain a copy of the License at * * http://www.apache.org/licenses/LICENSE-2.0 * * Unless required by applicable law or agreed to in writing, software * distributed under the License is distributed on an "AS IS" BASIS, * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either * express or implied. * See the License for the specific language governing permissions * and limitations under the License. * ------------------------------------------------------------------- */ #include "wavreader.h" #include #include #include #include #define TAG(a, b, c, d) (((a) << 24) | ((b) << 16) | ((c) << 8) | (d)) struct wav_reader { FILE *wav; uint32_t data_length; int format; int sample_rate; int bits_per_sample; int channels; int byte_rate; int block_align; int streamed; }; static uint32_t read_tag(struct wav_reader* wr) { uint32_t tag = 0; tag = (tag << 8) | fgetc(wr->wav); tag = (tag << 8) | fgetc(wr->wav); tag = (tag << 8) | fgetc(wr->wav); tag = (tag << 8) | fgetc(wr->wav); return tag; } static uint32_t read_int32(struct wav_reader* wr) { uint32_t value = 0; value |= fgetc(wr->wav) << 0; value |= fgetc(wr->wav) << 8; value |= fgetc(wr->wav) << 16; value |= fgetc(wr->wav) << 24; return value; } static uint16_t read_int16(struct wav_reader* wr) { uint16_t value = 0; value |= fgetc(wr->wav) << 0; value |= fgetc(wr->wav) << 8; return value; } static void skip(FILE *f, int n) { int i; for (i = 0; i < n; i++) fgetc(f); } void* wav_read_open(const char *filename) { struct wav_reader* wr = (struct wav_reader*) malloc(sizeof(*wr)); long data_pos = 0; memset(wr, 0, sizeof(*wr)); if (!strcmp(filename, "-")) wr->wav = stdin; else wr->wav = fopen(filename, "rb"); if (wr->wav == NULL) { free(wr); return NULL; } while (1) { uint32_t tag, tag2, length; tag = read_tag(wr); if (feof(wr->wav)) break; length = read_int32(wr); if (!length || length >= 0x7fff0000) { wr->streamed = 1; length = ~0; } if (tag != TAG('R', 'I', 'F', 'F') || length < 4) { fseek(wr->wav, length, SEEK_CUR); continue; } tag2 = read_tag(wr); length -= 4; if (tag2 != TAG('W', 'A', 'V', 'E')) { fseek(wr->wav, length, SEEK_CUR); continue; } // RIFF chunk found, iterate through it while (length >= 8) { uint32_t subtag, sublength; subtag = read_tag(wr); if (feof(wr->wav)) break; sublength = read_int32(wr); length -= 8; if (length < sublength) break; if (subtag == TAG('f', 'm', 't', ' ')) { if (sublength < 16) { // Insufficient data for 'fmt ' break; } wr->format = read_int16(wr); wr->channels = read_int16(wr); wr->sample_rate = read_int32(wr); wr->byte_rate = read_int32(wr); wr->block_align = read_int16(wr); wr->bits_per_sample = read_int16(wr); if (wr->format == 0xfffe) { if (sublength < 28) { // Insufficient data for waveformatex break; } skip(wr->wav, 8); wr->format = read_int32(wr); skip(wr->wav, sublength - 28); } else { skip(wr->wav, sublength - 16); } } else if (subtag == TAG('d', 'a', 't', 'a')) { data_pos = ftell(wr->wav); wr->data_length = sublength; if (!wr->data_length || wr->streamed) { wr->streamed = 1; return wr; } fseek(wr->wav, sublength, SEEK_CUR); } else { skip(wr->wav, sublength); } length -= sublength; } if (length > 0) { // Bad chunk? fseek(wr->wav, length, SEEK_CUR); } } fseek(wr->wav, data_pos, SEEK_SET); return wr; } void wav_read_close(void* obj) { struct wav_reader* wr = (struct wav_reader*) obj; if (wr->wav != stdin) fclose(wr->wav); free(wr); } int wav_get_header(void* obj, int* format, int* channels, int* sample_rate, int* bits_per_sample, unsigned int* data_length) { struct wav_reader* wr = (struct wav_reader*) obj; if (format) *format = wr->format; if (channels) *channels = wr->channels; if (sample_rate) *sample_rate = wr->sample_rate; if (bits_per_sample) *bits_per_sample = wr->bits_per_sample; if (data_length) *data_length = wr->data_length; return wr->format && wr->sample_rate; } int wav_read_data(void* obj, unsigned char* data, unsigned int length) { struct wav_reader* wr = (struct wav_reader*) obj; int n; if (wr->wav == NULL) return -1; if (length > wr->data_length && !wr->streamed) length = wr->data_length; n = fread(data, 1, length, wr->wav); wr->data_length -= length; return n; } fdk-aac-2.0.2/wavreader.h000066400000000000000000000021621404225310700151310ustar00rootroot00000000000000/* ------------------------------------------------------------------ * Copyright (C) 2009 Martin Storsjo * * Licensed under the Apache License, Version 2.0 (the "License"); * you may not use this file except in compliance with the License. * You may obtain a copy of the License at * * http://www.apache.org/licenses/LICENSE-2.0 * * Unless required by applicable law or agreed to in writing, software * distributed under the License is distributed on an "AS IS" BASIS, * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either * express or implied. * See the License for the specific language governing permissions * and limitations under the License. * ------------------------------------------------------------------- */ #ifndef WAVREADER_H #define WAVREADER_H #ifdef __cplusplus extern "C" { #endif void* wav_read_open(const char *filename); void wav_read_close(void* obj); int wav_get_header(void* obj, int* format, int* channels, int* sample_rate, int* bits_per_sample, unsigned int* data_length); int wav_read_data(void* obj, unsigned char* data, unsigned int length); #ifdef __cplusplus } #endif #endif fdk-aac-2.0.2/win32/000077500000000000000000000000001404225310700137415ustar00rootroot00000000000000fdk-aac-2.0.2/win32/getopt.h000066400000000000000000000702621404225310700154230ustar00rootroot00000000000000#ifndef __GETOPT_H__ /** * DISCLAIMER * This file has no copyright assigned and is placed in the Public Domain. * This file is part of the mingw-w64 runtime package. * * The mingw-w64 runtime package and its code is distributed in the hope that it * will be useful but WITHOUT ANY WARRANTY. ALL WARRANTIES, EXPRESSED OR * IMPLIED ARE HEREBY DISCLAIMED. This includes but is not limited to * warranties of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. */ /* * Implementation of the `getopt', `getopt_long' and `getopt_long_only' * APIs, for inclusion in the MinGW runtime library. * * This file is part of the MinGW32 package set. * * Written by Keith Marshall * Copyright (C) 2008, 2009, 2011, 2012, MinGW.org Project. * * --------------------------------------------------------------------------- * * Permission is hereby granted, free of charge, to any person obtaining a * copy of this software and associated documentation files (the "Software"), * to deal in the Software without restriction, including without limitation * the rights to use, copy, modify, merge, publish, distribute, sublicense, * and/or sell copies of the Software, and to permit persons to whom the * Software is furnished to do so, subject to the following conditions: * * The above copyright notice, this permission notice, and the following * disclaimer shall be included in all copies or substantial portions of * the Software. * * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS * OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER * DEALINGS IN THE SOFTWARE. * * --------------------------------------------------------------------------- * */ #define __GETOPT_H__ /* All the headers include this file. */ #include #include #include #include #ifdef __cplusplus extern "C" { #endif extern int optind; /* index of first non-option in argv */ extern int optopt; /* single option character, as parsed */ extern int opterr; /* flag to enable built-in diagnostics... */ /* (user may set to zero, to suppress) */ extern char *optarg; /* pointer to argument of current option */ /* Identify how to get the calling program name, for use in messages... */ #ifdef __CYGWIN__ /* * CYGWIN uses this DLL reference... */ # define PROGNAME __progname extern char __declspec(dllimport) *__progname; #else /* * ...while elsewhere, we simply use the first argument passed. */ # define PROGNAME *argv #endif extern int getopt(int nargc, char * const *nargv, const char *options); #ifdef _BSD_SOURCE /* * BSD adds the non-standard `optreset' feature, for reinitialisation * of `getopt' parsing. We support this feature, for applications which * proclaim their BSD heritage, before including this header; however, * to maintain portability, developers are advised to avoid it. */ # define optreset __mingw_optreset extern int optreset; #endif #ifdef __cplusplus } #endif /* * POSIX requires the `getopt' API to be specified in `unistd.h'; * thus, `unistd.h' includes this header. However, we do not want * to expose the `getopt_long' or `getopt_long_only' APIs, when * included in this manner. Thus, close the standard __GETOPT_H__ * declarations block, and open an additional __GETOPT_LONG_H__ * specific block, only when *not* __UNISTD_H_SOURCED__, in which * to declare the extended API. */ #endif /* !defined(__GETOPT_H__) */ #if !defined(__UNISTD_H_SOURCED__) && !defined(__GETOPT_LONG_H__) #define __GETOPT_LONG_H__ #ifdef __cplusplus extern "C" { #endif struct option /* specification for a long form option... */ { const char *name; /* option name, without leading hyphens */ int has_arg; /* does it take an argument? */ int *flag; /* where to save its status, or NULL */ int val; /* its associated status value */ }; enum /* permitted values for its `has_arg' field... */ { no_argument = 0, /* option never takes an argument */ required_argument, /* option always requires an argument */ optional_argument /* option may take an argument */ }; extern int getopt_long(int nargc, char * const *nargv, const char *options, const struct option *long_options, int *idx); extern int getopt_long_only(int nargc, char * const *nargv, const char *options, const struct option *long_options, int *idx); /* * Previous MinGW implementation had... */ #ifndef HAVE_DECL_GETOPT /* * ...for the long form API only; keep this for compatibility. */ # define HAVE_DECL_GETOPT 1 #endif /* Identify how to get the calling program name, for use in messages... */ #ifdef __CYGWIN__ /* * CYGWIN uses this DLL reference... */ # define PROGNAME __progname extern char __declspec(dllimport) *__progname; #else /* * ...while elsewhere, we simply use the first argument passed. */ # define PROGNAME *argv #endif /* Initialise the public variables. */ int optind = 1; /* index for first non-option arg */ int opterr = 1; /* enable built-in error messages */ char *optarg = NULL; /* pointer to current option argument */ #define CHAR char /* argument type selector */ #define getopt_switchar '-' /* option prefix character in argv */ #define getopt_pluschar '+' /* prefix for POSIX mode in optstring */ #define getopt_takes_argument ':' /* marker for optarg in optstring */ #define getopt_arg_assign '=' /* longopt argument field separator */ #define getopt_unknown '?' /* return code for unmatched option */ #define getopt_ordered 1 /* return code for ordered non-option */ #define getopt_all_done -1 /* return code to indicate completion */ enum { /* All `getopt' API functions are implemented via calls to the * common static function `getopt_parse()'; these `mode' selectors * determine the behaviour of `getopt_parse()', to deliver the * appropriate result in each case. */ getopt_mode_standard = 0, /* getopt() */ getopt_mode_long, /* getopt_long() */ getopt_mode_long_only /* getopt_long_only() */ }; enum { /* When attempting to match a command line argument to a long form option, * these indicate the status of the match. */ getopt_no_match = 0, /* no successful match */ getopt_abbreviated_match, /* argument is an abbreviation for an option */ getopt_exact_match /* argument matches the full option name */ }; int optopt = getopt_unknown; /* return value for option being evaluated */ /* Some BSD applications expect to be able to reinitialise `getopt' parsing * by setting a global variable called `optreset'. We provide an obfuscated * API, which allows applications to emulate this brain damage; however, any * use of this is non-portable, and is strongly discouraged. */ #define optreset __mingw_optreset int optreset = 0; static int getopt_missing_arg( const CHAR *optstring ) { /* Helper function to determine the appropriate return value, * for the case where a required option argument is missing. */ if( (*optstring == getopt_pluschar) || (*optstring == getopt_switchar) ) ++optstring; return (*optstring == getopt_takes_argument) ? getopt_takes_argument : getopt_unknown; } /* `complain' macro facilitates the generation of simple built-in * error messages, displayed on various fault conditions, provided * `opterr' is non-zero. */ #define complain( MSG, ARG ) if( opterr ) \ fprintf( stderr, "%s: "MSG"\n", PROGNAME, ARG ) static int getopt_argerror( int mode, char *fmt, CHAR *prog, struct option *opt, int retval ) { /* Helper function, to generate more complex built-in error * messages, for invalid arguments to long form options ... */ if( opterr ) { /* ... but, displayed only if `opterr' is non-zero. */ char flag[] = "--"; if( mode != getopt_mode_long ) /* * only display one hyphen, for implicit long form options, * improperly resolved by `getopt_long_only()'. */ flag[1] = 0; /* * always preface the program name ... */ fprintf( stderr, "%s: ", prog ); /* * to the appropriate, option specific message. */ fprintf( stderr, fmt, flag, opt->name ); } /* Whether displaying the message, or not, always set `optopt' * to identify the faulty option ... */ optopt = opt->val; /* * and return the `invalid option' indicator. */ return retval; } /* `getopt_conventions' establish behavioural options, to control * the operation of `getopt_parse()', e.g. to select between POSIX * and GNU style argument parsing behaviour. */ #define getopt_set_conventions 0x1000 #define getopt_posixly_correct 0x0010 static int getopt_conventions( int flags ) { static int conventions = 0; if( (conventions == 0) && ((flags & getopt_set_conventions) == 0) ) { /* default conventions have not yet been established; * initialise them now! */ conventions = getopt_set_conventions; if( flags == getopt_pluschar ) conventions |= getopt_posixly_correct; } else if( flags & getopt_set_conventions ) /* * default conventions may have already been established, * but this is a specific request to augment them. */ conventions |= flags; /* in any event, return the currently established conventions. */ return conventions; } static int is_switchar( CHAR flag ) { /* A simple helper function, used to identify the switch character * introducing an optional command line argument. */ return flag == getopt_switchar; } static const CHAR *getopt_match( CHAR lookup, const CHAR *opt_string ) { /* Helper function, used to identify short form options. */ if( (*opt_string == getopt_pluschar) || (*opt_string == getopt_switchar) ) ++opt_string; if( *opt_string == getopt_takes_argument ) ++opt_string; do if( lookup == *opt_string ) return opt_string; while( *++opt_string ); return NULL; } static int getopt_match_long( const CHAR *nextchar, const CHAR *optname ) { /* Helper function, used to identify potential matches for * long form options. */ CHAR matchchar; while( (matchchar = *nextchar++) && (matchchar == *optname) ) /* * skip over initial substring which DOES match. */ ++optname; if( matchchar ) { /* did NOT match the entire argument to an initial substring * of a defined option name ... */ if( matchchar != getopt_arg_assign ) /* * ... and didn't stop at an `=' internal field separator, * so this is NOT a possible match. */ return getopt_no_match; /* DID stop at an `=' internal field separator, * so this IS a possible match, and what follows is an * argument to the possibly matched option. */ optarg = (char *)(nextchar); } return *optname /* * if we DIDN'T match the ENTIRE text of the option name, * then it's a possible abbreviated match ... */ ? getopt_abbreviated_match /* * but if we DID match the entire option name, * then it's a DEFINITE EXACT match. */ : getopt_exact_match; } static int getopt_resolved( int mode, int argc, CHAR *const *argv, int *argind, struct option *opt, int index, int *retindex, const CHAR *optstring ) { /* Helper function to establish appropriate return conditions, * on resolution of a long form option. */ if( retindex != NULL ) *retindex = index; /* On return, `optind' should normally refer to the argument, if any, * which follows the current one; it is convenient to set this, before * checking for the presence of any `optarg'. */ optind = *argind + 1; if( optarg && (opt[index].has_arg == no_argument) ) /* * it is an error for the user to specify an option specific argument * with an option which doesn't expect one! */ return getopt_argerror( mode, "option `%s%s' doesn't accept an argument\n", PROGNAME, opt + index, getopt_unknown ); else if( (optarg == NULL) && (opt[index].has_arg == required_argument) ) { /* similarly, it is an error if no argument is specified * with an option which requires one ... */ if( optind < argc ) /* * ... except that the requirement may be satisfied from * the following command line argument, if any ... */ optarg = argv[*argind = optind++]; else /* so fail this case, only if no such argument exists! */ return getopt_argerror( mode, "option `%s%s' requires an argument\n", PROGNAME, opt + index, getopt_missing_arg( optstring ) ); } /* when the caller has provided a return buffer ... */ if( opt[index].flag != NULL ) { /* ... then we place the proper return value there, * and return a status code of zero ... */ *(opt[index].flag) = opt[index].val; return 0; } /* ... otherwise, the return value becomes the status code. */ return opt[index].val; } static int getopt_verify( const CHAR *nextchar, const CHAR *optstring ) { /* Helper function, called by getopt_parse() when invoked * by getopt_long_only(), to verify when an unmatched or an * ambiguously matched long form option string is valid as * a short form option specification. */ if( ! (nextchar && *nextchar && optstring && *optstring) ) /* * There are no characters to be matched, or there are no * valid short form option characters to which they can be * matched, so this can never be valid. */ return 0; while( *nextchar ) { /* For each command line character in turn ... */ const CHAR *test; if( (test = getopt_match( *nextchar++, optstring )) == NULL ) /* * ... there is no short form option to match the current * candidate, so the entire argument fails. */ return 0; if( test[1] == getopt_takes_argument ) /* * The current candidate is valid, and it matches an option * which takes an argument, so this command line argument is * a valid short form option specification; accept it. */ return 1; } /* If we get to here, then every character in the command line * argument was valid as a short form option; accept it. */ return 1; } static #define getopt_std_args int argc, CHAR *const argv[], const CHAR *optstring int getopt_parse( int mode, getopt_std_args, ... ) { /* Common core implementation for ALL `getopt' functions. */ static int argind = 0; static int optbase = 0; static const CHAR *nextchar = NULL; static int optmark = 0; if( (optreset |= (optind < 1)) || (optind < optbase) ) { /* POSIX does not prescribe any definitive mechanism for restarting * a `getopt' scan, but some applications may require such capability. * We will support it, by allowing the caller to adjust the value of * `optind' downwards, (nominally setting it to zero). Since POSIX * wants `optind' to have an initial value of one, but we want all * of our internal place holders to be initialised to zero, when we * are called for the first time, we will handle such a reset by * adjusting all of the internal place holders to one less than * the adjusted `optind' value, (but never to less than zero). */ if( optreset ) { /* User has explicitly requested reinitialisation... * We need to reset `optind' to it's normal initial value of 1, * to avoid a potential infinitely recursive loop; by doing this * up front, we also ensure that the remaining place holders * will be correctly reinitialised to no less than zero. */ optind = 1; /* We also need to clear the `optreset' request... */ optreset = 0; } /* Now, we may safely reinitialise the internal place holders, to * one less than `optind', without fear of making them negative. */ optmark = optbase = argind = optind - 1; nextchar = NULL; } /* From a POSIX perspective, the following is `undefined behaviour'; * we implement it thus, for compatibility with GNU and BSD getopt. */ else if( optind > (argind + 1) ) { /* Some applications expect to be able to manipulate `optind', * causing `getopt' to skip over one or more elements of `argv'; * POSIX doesn't require us to support this brain-damaged concept; * (indeed, POSIX defines no particular behaviour, in the event of * such usage, so it must be considered a bug for an application * to rely on any particular outcome); nonetheless, Mac-OS-X and * BSD actually provide *documented* support for this capability, * so we ensure that our internal place holders keep track of * external `optind' increments; (`argind' must lag by one). */ argind = optind - 1; /* When `optind' is misused, in this fashion, we also abandon any * residual text in the argument we had been parsing; this is done * without any further processing of such abandoned text, assuming * that the caller is equipped to handle it appropriately. */ nextchar = NULL; } if( nextchar && *nextchar ) { /* we are parsing a standard, or short format, option argument ... */ const CHAR *optchar; if( (optchar = getopt_match( optopt = *nextchar++, optstring )) != NULL ) { /* we have identified it as valid ... */ if( optchar[1] == getopt_takes_argument ) { /* and determined that it requires an associated argument ... */ if( ! *(optarg = (char *)(nextchar)) ) { /* the argument is NOT attached ... */ if( optchar[2] == getopt_takes_argument ) /* * but this GNU extension marks it as optional, * so we don't provide one on this occasion. */ optarg = NULL; /* otherwise this option takes a mandatory argument, * so, provided there is one available ... */ else if( (argc - argind) > 1 ) /* * we take the following command line argument, * as the appropriate option argument. */ optarg = argv[++argind]; /* but if no further argument is available, * then there is nothing we can do, except for * issuing the requisite diagnostic message. */ else { complain( "option requires an argument -- %c", optopt ); return getopt_missing_arg( optstring ); } } optind = argind + 1; nextchar = NULL; } else optarg = NULL; optind = (nextchar && *nextchar) ? argind : argind + 1; return optopt; } /* if we didn't find a valid match for the specified option character, * then we fall through to here, so take appropriate diagnostic action. */ if( mode == getopt_mode_long_only ) { complain( "unrecognised option `-%s'", --nextchar ); nextchar = NULL; optopt = 0; } else complain( "invalid option -- %c", optopt ); optind = (nextchar && *nextchar) ? argind : argind + 1; return getopt_unknown; } if( optmark > optbase ) { /* This can happen, in GNU parsing mode ONLY, when we have * skipped over non-option arguments, and found a subsequent * option argument; in this case we permute the arguments. */ int index; /* * `optspan' specifies the number of contiguous arguments * which are spanned by the current option, and so must be * moved together during permutation. */ const int optspan = argind - optmark + 1; /* * we use `this_arg' to store these temporarily. */ CHAR **this_arg = malloc(sizeof(CHAR*) * optspan); /* * we cannot manipulate `argv' directly, since the `getopt' * API prototypes it as `read-only'; this cast to `arglist' * allows us to work around that restriction. */ CHAR **arglist = (char **)(argv); /* save temporary copies of the arguments which are associated * with the current option ... */ for( index = 0; index < optspan; ++index ) this_arg[index] = arglist[optmark + index]; /* move all preceding non-option arguments to the right, * overwriting these saved arguments, while making space * to replace them in their permuted location. */ for( --optmark; optmark >= optbase; --optmark ) arglist[optmark + optspan] = arglist[optmark]; /* restore the temporarily saved option arguments to * their permuted location. */ for( index = 0; index < optspan; ++index ) arglist[optbase + index] = this_arg[index]; /* adjust `optbase', to account for the relocated option. */ optbase += optspan; free(this_arg); } else /* no permutation occurred ... * simply adjust `optbase' for all options parsed so far. */ optbase = argind + 1; /* enter main parsing loop ... */ while( argc > ++argind ) { /* inspect each argument in turn, identifying possible options ... */ if( is_switchar( *(nextchar = argv[optmark = argind]) ) && *++nextchar ) { /* we've found a candidate option argument ... */ if( is_switchar( *nextchar ) ) { /* it's a double hyphen argument ... */ const CHAR *refchar = nextchar; if( *++refchar ) { /* and it looks like a long format option ... * `getopt_long' mode must be active to accept it as such, * `getopt_long_only' also qualifies, but we must downgrade * it to force explicit handling as a long format option. */ if( mode >= getopt_mode_long ) { nextchar = refchar; mode = getopt_mode_long; } } else { /* this is an explicit `--' end of options marker, so wrap up now! */ if( optmark > optbase ) { /* permuting the argument list as necessary ... * (note use of `this_arg' and `arglist', as above). */ CHAR *this_arg = argv[optmark]; CHAR **arglist = (CHAR **)(argv); /* move all preceding non-option arguments to the right ... */ do arglist[optmark] = arglist[optmark - 1]; while( optmark-- > optbase ); /* reinstate the `--' marker, in its permuted location. */ arglist[optbase] = this_arg; } /* ... before finally bumping `optbase' past the `--' marker, * and returning the `all done' completion indicator. */ optind = ++optbase; return getopt_all_done; } } else if( mode < getopt_mode_long_only ) { /* it's not an explicit long option, and `getopt_long_only' isn't active, * so we must explicitly try to match it as a short option. */ mode = getopt_mode_standard; } if( mode >= getopt_mode_long ) { /* the current argument is a long form option, (either explicitly, * introduced by a double hyphen, or implicitly because we were called * by `getopt_long_only'); this is where we parse it. */ int lookup; int matched = -1; /* we need to fetch the `extra' function arguments, which are * specified for the `getopt_long' APIs. */ va_list refptr; struct option *longopts; int *optindex; va_start( refptr, optstring ); longopts = va_arg( refptr, struct option * ); optindex = va_arg( refptr, int * ); va_end( refptr ); /* ensuring that `optarg' does not inherit any junk, from parsing * preceding arguments ... */ optarg = NULL; for( lookup = 0; longopts && longopts[lookup].name; ++lookup ) { /* scan the list of defined long form options ... */ switch( getopt_match_long( nextchar, longopts[lookup].name ) ) { /* looking for possible matches for the current argument. */ case getopt_exact_match: /* * when an exact match is found, * return it immediately, setting `nextchar' to NULL, * to ensure we don't mistakenly try to match any * subsequent characters as short form options. */ nextchar = NULL; return getopt_resolved( mode, argc, argv, &argind, longopts, lookup, optindex, optstring ); case getopt_abbreviated_match: /* * but, for a partial (initial substring) match ... */ if( matched >= 0 ) { /* if this is not the first, then we have an ambiguity ... */ if( (mode == getopt_mode_long_only) /* * However, in the case of getopt_long_only(), if * the entire ambiguously matched string represents * a valid short option specification, then we may * proceed to interpret it as such. */ && getopt_verify( nextchar, optstring ) ) return getopt_parse( mode, argc, argv, optstring ); /* If we get to here, then the ambiguously matched * partial long option isn't valid for short option * evaluation; reset parser context to resume with * the following command line argument, diagnose * ambiguity, and bail out. */ optopt = 0; nextchar = NULL; optind = argind + 1; complain( "option `%s' is ambiguous", argv[argind] ); return getopt_unknown; } /* otherwise just note that we've found a possible match ... */ matched = lookup; } } if( matched >= 0 ) { /* if we get to here, then we found exactly one partial match, * so return it, as for an exact match. */ nextchar = NULL; return getopt_resolved( mode, argc, argv, &argind, longopts, matched, optindex, optstring ); } /* if here, then we had what SHOULD have been a long form option, * but it is unmatched ... */ if( (mode < getopt_mode_long_only) /* * ... although paradoxically, `mode == getopt_mode_long_only' * allows us to still try to match it as a short form option. */ || (getopt_verify( nextchar, optstring ) == 0) ) { /* When it cannot be matched, reset the parsing context to * resume from the next argument, diagnose the failed match, * and bail out. */ optopt = 0; nextchar = NULL; optind = argind + 1; complain( "unrecognised option `%s'", argv[argind] ); return getopt_unknown; } } /* fall through to handle standard short form options... * when the option argument format is neither explictly identified * as long, nor implicitly matched as such, and the argument isn't * just a bare hyphen, (which isn't an option), then we make one * recursive call to explicitly interpret it as short format. */ if( *nextchar ) return getopt_parse( mode, argc, argv, optstring ); } /* if we get to here, then we've parsed a non-option argument ... * in GNU compatibility mode, we step over it, so we can permute * any subsequent option arguments, but ... */ if( *optstring == getopt_switchar ) { /* if `optstring' begins with a `-' character, this special * GNU specific behaviour requires us to return the non-option * arguments in strict order, as pseudo-arguments to a special * option, with return value defined as `getopt_ordered'. */ nextchar = NULL; optind = argind + 1; optarg = argv[argind]; return getopt_ordered; } if( getopt_conventions( *optstring ) & getopt_posixly_correct ) /* * otherwise ... * for POSIXLY_CORRECT behaviour, or if `optstring' begins with * a `+' character, then we break out of the parsing loop, so that * the scan ends at the current argument, with no permutation. */ break; } /* fall through when all arguments have been evaluated, */ optind = optbase; return getopt_all_done; } /* All three public API entry points are trivially defined, * in terms of the internal `getopt_parse' function. */ int getopt( getopt_std_args ) { return getopt_parse( getopt_mode_standard, argc, argv, optstring ); } int getopt_long( getopt_std_args, const struct option *opts, int *index ) { return getopt_parse( getopt_mode_long, argc, argv, optstring, opts, index ); } int getopt_long_only( getopt_std_args, const struct option *opts, int *index ) { return getopt_parse( getopt_mode_long_only, argc, argv, optstring, opts, index ); } #ifdef __weak_alias /* * These Microsnot style uglified aliases are provided for compatibility * with the previous MinGW implementation of the getopt API. */ __weak_alias( getopt, _getopt ) __weak_alias( getopt_long, _getopt_long ) __weak_alias( getopt_long_only, _getopt_long_only ) #endif #ifdef __cplusplus } #endif #endif /* !defined(__UNISTD_H_SOURCED__) && !defined(__GETOPT_LONG_H__) */